diff --git a/.lastclean b/.lastclean new file mode 100644 index 0000000000..425151f3a4 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +40 diff --git a/.version b/.version new file mode 100644 index 0000000000..f4ddbad08d --- /dev/null +++ b/.version @@ -0,0 +1 @@ +20.2.0-rc1 \ No newline at end of file diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000000..1221fcaf47 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,106118 @@ +2023-03-02 16:45 +0000 Asterisk Development Team + + * asterisk 20.2.0-rc1 Released. + +2023-03-02 10:37 +0000 [93813c9dca] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 20.2.0 +2023-02-16 10:05 +0000 [ceda5a9859] George Joseph + + * res_pjsip: Replace invalid UTF-8 sequences in callerid name + + * Added a new function ast_utf8_replace_invalid_chars() to + utf8.c that copies a string replacing any invalid UTF-8 + sequences with the Unicode specified U+FFFD replacement + character. For example: "abc\xffdef" becomes "abc\uFFFDdef". + Any UTF-8 compliant implementation will show that character + as a � character. + + * Updated res_pjsip:set_id_from_hdr() to use + ast_utf8_replace_invalid_chars and print a warning if any + invalid sequences were found during the copy. + + * Updated stasis_channels:ast_channel_publish_varset to use + ast_utf8_replace_invalid_chars and print a warning if any + invalid sequences were found during the copy. + + ASTERISK-27830 + + Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8 + +2023-02-27 18:35 +0000 [e5c5cd6e25] Sean Bright + + * test.c: Avoid passing -1 to FD_* family of functions. + + This avoids buffer overflow errors when running tests that capture + output from child processes. + + This also corrects a copypasta in an off-nominal error message. + + Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d + +2022-12-14 10:00 +0000 [ede67a99be] Naveen Albert + + * chan_iax2: Fix jitterbuffer regression prior to receiving audio. + + ASTERISK_29392 (a security fix) introduced a regression by + not processing frames when we don't have an audio format. + + Currently, chan_iax2 only calls jb_get to read frames from + the jitterbuffer when the voiceformat has been set on the pvt. + However, this only happens when we receive a voice frame, which + means that prior to receiving voice frames, other types of frames + get stalled completely in the jitterbuffer. + + To fix this, we now fallback to using the format negotiated during + call setup until we've actually received a voice frame with a format. + This ensures we're always able to read from the jitterbuffer. + + ASTERISK-30354 #close + ASTERISK-30162 #close + + Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9 + +2023-02-27 15:35 +0000 [827222d607] Sean Bright + + * test_crypto.c: Fix getcwd(…) build error. + + `getcwd(…)` is decorated with the `warn_unused_result` attribute and + therefore needs its return value checked. + + Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93 + +2023-02-11 06:58 +0000 [200dc7d0e8] Nick French + + * pjproject_bundled: Fix cross-compilation with SSL libs. + + Asterisk makefiles auto-detect SSL library availability, + then they assume that pjproject makefiles will also autodetect + an SSL library at the same time, so they do not pass on the + autodetection result to pjproject. + + This normally works, except the pjproject makefiles disables + autodetection when cross-compiling. + + Fix by explicitly configuring pjproject to use SSL if we + have been told to use it or it was autodetected + + ASTERISK-30424 #close + + Change-Id: I8fe2999ea46710e21d1d55a1bed92769c6ebded9 + +2023-01-30 17:14 +0000 [5c11d7adea] Mike Bradeen + + * app_read: Add an option to return terminator on empty digits. + + Adds 'e' option to allow Read() to return the terminator as the + dialed digits in the case where only the terminator is entered. + + ie; if "#" is entered, return "#" if the 'e' option is set and "" + if it is not. + + ASTERISK-30411 + + Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5 + +2023-01-07 23:04 +0000 [5b0e3444c3] cmaj + + * res_phoneprov.c: Multihomed SERVER cache prevention + + Phones moving between subnets on multi-homed server have their + initially connected interface IP cached in the SERVER variable, + even when it is not specified in the configuration files. This + prevents phones from obtaining the correct SERVER variable value + when they move to another subnet. + + ASTERISK-30388 #close + Reported-by: cmaj + + Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92 + +2023-01-27 14:23 +0000 [2308afed8e] Mike Bradeen + + * app_directory: Add a 'skip call' option. + + Adds 's' option to skip calling the extension and instead set the + extension as DIRECTORY_EXTEN channel variable. + + ASTERISK-30405 + + Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299 + +2023-02-06 09:54 +0000 [98742388b6] Mike Bradeen + + * app_senddtmf: Add option to answer target channel. + + Adds a new option to SendDTMF() which will answer the specified + channel if it is not already up. If no channel is specified, the + current channel will be answered instead. + + ASTERISK-30422 + + Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085 + +2023-02-21 14:25 +0000 [37e558f6ef] Mike Bradeen + + * res_pjsip: Prevent SEGV in pjsip_evsub_send_request + + contributed pjproject - patch to check sub->pending_notify + in evsub.c:on_tsx_state before calling + pjsip_evsub_send_request() + + res_pjsip_pubsub - change post pjsip 2.13 behavior to use + pubsub_on_refresh_timeout to avoid the ao2_cleanup call on + the sub_tree. This is is because the final NOTIFY send is no + longer the last place the sub_tree is referenced. + + ASTERISK-30419 + + Change-Id: Ib5cc662ce578e9adcda312e16c58a10b6453e438 + +2023-02-02 08:19 +0000 [aeb16aa7d8] Sean Bright + + * app_queue: Minor docs and logging fixes for UnpauseQueueMember. + + ASTERISK-30417 #close + + Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe + +2023-01-31 08:40 +0000 [aef0c0ce0e] Sean Bright + + * app_queue: Reset all queue defaults before reload. + + Several queue fields were not being set to their default value during + a reload. + + Additionally added some sample configuration options that were missing + from queues.conf.sample. + + Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4 + +2023-01-20 16:50 +0000 [58636a6ea6] Mike Bradeen + + * res_pjsip: Upgraded bundled pjsip to 2.13 + + Removed multiple patches. + + Code chages in res_pjsip_pubsub due to changes in evsub. + + Pjsip now calls on_evsub_state() before on_rx_refresh(), + so the sub tree deletion that used to take place in + on_evsub_state() now must take place in on_rx_refresh(). + + Additionally, pjsip now requires that you send the NOTIFY + from within on_rx_refresh(), otherwise it will assert + when going to send the 200 OK. The idea is that it will + look for this NOTIFY and cache it until after sending the + response in order to deal with the self-imposed message + mis-order. Asterisk previously dealt with this by pushing + the NOTIFY in on_rx_refresh(), but pjsip now forces us + to use it's method. + + Changes were required to configure in order to detect + which way pjsip handles this as the two are not + compatible for the reasons mentioned above. + + A corresponding change in testsuite is required in order + to deal with the small interal timing changes caused by + moving the NOTIFY send. + + ASTERISK-30325 + + Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f + +2023-01-30 15:17 +0000 [96d9ad51ac] Sean Bright + + * doxygen: Fix doxygen errors. + + Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85 + +2022-01-06 16:11 +0000 [88b2c741ca] Naveen Albert + + * app_signal: Add signaling applications + + Adds the Signal and WaitForSignal + applications, which can be used for inter-channel + signaling in the dialplan. + + Signal supports sending a signal to other channels + listening for a signal of the same name, with an + optional data payload. The signal is received by + all channels waiting for that named signal. + + ASTERISK-29810 #close + + Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3 + +2023-01-25 16:27 +0000 [70856e865f] Mike Bradeen + + * app_directory: add ability to specify configuration file + + Adds option to app_directory to specify a filename from which to + read configuration instead of voicemail.conf ie; + + same => n,Directory(,,c(directory.conf)) + + This configuration should contain a list of extensions using the + voicemail.conf format, ie; + + 2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no + + ASTERISK-30404 + + Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13 + +2022-02-12 15:59 +0000 [8a45cd7af4] Naveen Albert + + * func_json: Enhance parsing capabilities of JSON_DECODE + + Adds support for arrays to JSON_DECODE by allowing the + user to print out entire arrays or index a particular + key or print the number of keys in a JSON array. + + Additionally, adds support for recursively iterating a + JSON tree in a single function call, making it easier + to parse JSON results with multiple levels. A maximum + depth is imposed to prevent potentially blowing + the stack. + + Also fixes a bug with the unit tests causing an empty + string to be printed instead of the actual test result. + + ASTERISK-29913 #close + + Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b + +2023-01-04 06:35 +0000 [f99849f8d5] sungtae kim + + * res_stasis_snoop: Fix snoop crash + + Added NULL pointer check and channel lock to prevent resource release + while the chanspy is processing. + + ASTERISK-29604 + + Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24 + +2023-01-26 14:18 +0000 [56051d1ac5] Sean Bright + + * pbx_ael: Global variables are not expanded. + + Variable references within global variable assignments are now + expanded rather than being included literally. + + ASTERISK-30406 #close + + Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d + +2022-10-13 08:45 +0000 [a1da8042d1] Naveen Albert + + * res_pjsip_session: Add overlap_context option. + + Adds the overlap_context option, which can be used + to explicitly specify a context to use for overlap + dialing extension matches, rather than forcibly + using the context configured for the endpoint. + + ASTERISK-30262 #close + + Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db + +2023-01-05 10:41 +0000 [ef16eaee36] Sean Bright + + * app_playback.c: Fix PLAYBACKSTATUS regression. + + In Asterisk 11, if a channel was redirected away during Playback(), + the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 + (specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that + behavior was inadvertently changed and the same operation would result + in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 + behavior has been restored. + + Partial fix for ASTERISK~25661. + + Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff + +2023-01-11 11:17 +0000 [2f5aece0c9] George Joseph + + * res_rtp_asterisk: Don't use double math to generate timestamps + + Rounding issues with double math were causing rtp timestamp + slips in outgoing packets. We're now back to integer math + and are getting no more slips. + + ASTERISK-30391 + + Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4 + +2023-01-06 10:06 +0000 [e86d5d7fda] Alexei Gradinari + + * format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...) + + Each playback of WAV files results in logging + "Skipping unknown block 'LIST'". + + To prevent unnecessary flooding of this DEBUG log this patch replaces + ast_log(LOG_DEBUG, ...) by ast_debug(1, ...). + + Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0 + +2022-11-17 20:16 +0000 [3526441e41] Igor Goncharovsky + + * res_pjsip_rfc3326: Add SIP causes support for RFC3326 + + Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). + + ASTERISK-30319 #close + + Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645 + +2022-10-28 05:57 +0000 [4710f37ef6] George Joseph + + * res_rtp_asterisk: Asterisk Media Experience Score (MES) + + ----------------- + + This commit reinstates MES with some casting fixes to the + functions in time.h that convert between doubles and timeval + structures. The casting issues were causing incorrect + timestamps to be calculated which caused transcoding from/to + G722 to produce bad or no audio. + + ASTERISK-30391 + + ----------------- + + This module has been updated to provide additional + quality statistics in the form of an Asterisk + Media Experience Score. The score is avilable using + the same mechanisms you'd use to retrieve jitter, loss, + and rtt statistics. For more information about the + score and how to retrieve it, see + https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score + + * Updated chan_pjsip to set quality channel variables when a + call ends. + * Updated channels/pjsip/dialplan_functions.c to add the ability + to retrieve the MES along with the existing rtcp stats when + using the CHANNEL dialplan function. + * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed + checks for debugging purposes. + * Added several function to time.h for manipulating time-in-samples + and times represented as double seconds. + * Updated rtp_engine.c to pass through the MES when stats are + requested. Also debug output that dumps the stats when an + rtp instance is destroyed. + * Updated res_rtp_asterisk.c to implement the calculation of the + MES. In the process, also had to update the calculation of + jitter. Many debugging statements were also changed to be + more informative. + * Added a unit test for internal testing. The test should not be + run during normal operation and is disabled by default. + + Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038 + +2023-01-09 07:21 +0000 [62ca063fca] George Joseph + + * Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" + + This reverts commit d454801c2ddba89f7925c847012db2866e271f68. + + Reason for revert: Issue when transcoding to/from g722 + + Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86 + +2022-12-08 15:44 +0000 [d33bd6d67a] Naveen Albert + + * loader: Allow declined modules to be unloaded. + + Currently, if a module declines to load, dlopen is called + to register the module but dlclose never gets called. + Furthermore, loader.c currently doesn't allow dlclose + to ever get called on the module, since it declined to + load and the unload function bails early in this case. + + This can be problematic if a module is updated, since the + new module cannot be loaded into memory since we haven't + closed all references to it. To fix this, we now allow + modules to be unloaded, even if they never "loaded" in + Asterisk itself, so that dlclose is called and the module + can be properly cleaned up, allowing the updated module + to be loaded from scratch next time. + + ASTERISK-30345 #close + + Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5 + +2022-08-15 15:04 +0000 [e06fe8e344] Naveen Albert + + * app_broadcast: Add Broadcast application + + Adds a new application, Broadcast, which can be used for + one-to-many transmission and many-to-one reception of + channel audio in Asterisk. This is similar to ChanSpy, + except it is designed for multiple channel targets instead + of a single one. This can make certain kinds of audio + manipulation more efficient and streamlined. New kinds + of audio injection impossible with ChanSpy are also made + possible. + + ASTERISK-30180 #close + + Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726 + +2022-12-13 14:35 +0000 [68e345286b] Naveen Albert + + * func_frame_trace: Print text for text frames. + + Since text frames contain a text body, make FRAME_TRACE + more useful for text frames by actually printing the text. + + ASTERISK-30353 #close + + Change-Id: Ia6ce3d15cecd7a673a528d34faac86854a2bab50 + +2022-12-16 12:25 +0000 [3b3fef2347] Naveen Albert + + * json.h: Add ast_json_object_real_get. + + json.h contains macros to get a string and an integer + from a JSON object. However, the macro to do this for + JSON reals is missing. This adds that. + + ASTERISK-30361 #close + + Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a + +2022-12-21 19:01 +0000 [7b8f7428da] Naveen Albert + + * manager: Fix appending variables. + + The if statement here is always false after the for + loop finishes, so variables are never appended. + This removes that to properly append to the end + of the variable list. + + ASTERISK-30351 #close + Reported by: Sebastian Gutierrez + + Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199 + +2022-12-23 06:02 +0000 [24102ba236] George Joseph + + * res_pjsip_transport_websocket: Add remote port to transport + + When Asterisk receives a new websocket conenction, it creates a new + pjsip transport for it and copies connection data into it. The + transport manager then uses the remote IP address and port on the + transport to create a monitor for each connection. However, the + remote port wasn't being copied, only the IP address which meant + that the transport manager was creating only 1 monitoring entry for + all websocket connections from the same IP address. Therefore, if + one of those connections failed, it deleted the transport taking + all the the connections from that same IP address with it. + + * We now copy the remote port into the created transport and the + transport manager behaves correctly. + + ASTERISK-30369 + + Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1 + +2022-12-28 13:33 +0000 [edc90c96ac] Boris P. Korzun + + * http.c: Fix NULL pointer dereference bug + + If native HTTP is disabled but HTTPS is enabled and status page enabled + too, Core/HTTP crashes while loading. 'global_http_server' references + to NULL, but the status page tries to dereference it. + + The patch adds a check for HTTP is enabled. + + ASTERISK-30379 #close + + Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249 + +2022-12-16 01:00 +0000 [3d9b9a2b16] Holger Hans Peter Freyther + + * res_http_media_cache: Do not crash when there is no extension + + Do not crash when a URL has no path component as in this case the + ast_uri_path function will return NULL. Make the code cope with not + having a path. + + The below would crash + > media cache create http://google.com /tmp/foo.wav + + Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault. + 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 + (gdb) bt + #0 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 + #1 0x0000ffff43d43a78 in file_extension_from_string (str=, buffer=buffer@entry=0xffffca9973c0 "", + capacity=capacity@entry=64) at res_http_media_cache.c:288 + #2 0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568, + buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378 + #3 0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392 + #4 0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555 + #5 0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=, data=, object=) + at res_http_media_cache.c:613 + #6 0x0000000000487638 in bucket_file_wizard_create (sorcery=, data=, object=) + at bucket.c:191 + #7 0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718, + details=details@entry=0xffffca9974a8) at sorcery.c:2027 + #8 0x0000000000559698 in ast_sorcery_create (sorcery=, object=object@entry=0x3bf96568) at sorcery.c:2077 + #9 0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727 + #10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com", + file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335 + #11 0x00000000004f88ec in media_cache_handle_create_item (e=, cmd=, a=0xffffca9976b8) + at media_cache.c:640 + + ASTERISK-30375 #close + + Change-Id: I6a9433688cb5d3d4be8758b7642d923bdde6c273 + +2022-10-28 05:57 +0000 [d454801c2d] George Joseph + + * res_rtp_asterisk: Asterisk Media Experience Score (MES) + + This module has been updated to provide additional + quality statistics in the form of an Asterisk + Media Experience Score. The score is avilable using + the same mechanisms you'd use to retrieve jitter, loss, + and rtt statistics. For more information about the + score and how to retrieve it, see + https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score + + * Updated chan_pjsip to set quality channel variables when a + call ends. + * Updated channels/pjsip/dialplan_functions.c to add the ability + to retrieve the MES along with the existing rtcp stats when + using the CHANNEL dialplan function. + * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed + checks for debugging purposes. + * Added several function to time.h for manipulating time-in-samples + and times represented as double seconds. + * Updated rtp_engine.c to pass through the MES when stats are + requested. Also debug output that dumps the stats when an + rtp instance is destroyed. + * Updated res_rtp_asterisk.c to implement the calculation of the + MES. In the process, also had to update the calculation of + jitter. Many debugging statements were also changed to be + more informative. + * Added a unit test for internal testing. The test should not be + run during normal operation and is disabled by default. + + ASTERISK-30280 + + Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a + +2022-12-21 09:01 +0000 [cc8d9b947b] Naveen Albert + + * pbx_app: Update outdated pbx_exec channel snapshots. + + pbx_exec makes a channel snapshot before executing applications. + This doesn't cause an issue during normal dialplan execution + where pbx_exec is called over and over again in succession. + However, if pbx_exec is called "one off", e.g. using + ast_pbx_exec_application, then a channel snapshot never ends + up getting made after the executed application returns, and + inaccurate snapshot information will linger for a while, causing + "core show channels", etc. to show erroneous info. + + This is fixed by manually making a channel snapshot at the end + of ast_pbx_exec_application, since we anticipate that pbx_exec + might not get called again immediately. + + ASTERISK-30367 #close + + Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086 + +2022-11-26 06:54 +0000 [c7598ee947] Naveen Albert + + * res_pjsip_session: Use Caller ID for extension matching. + + Currently, there is no Caller ID available to us when + checking for an extension match when handling INVITEs. + As a result, extension patterns that depend on the Caller ID + are not matched and calls may be incorrectly rejected. + + The Caller ID is not available because the supplement that + adds Caller ID to the session does not execute until after + this check. Supplement callbacks cannot yet be executed + at this point since the session is not yet in the appropriate + state. + + To fix this without impacting existing behavior, the Caller ID + number is now retrieved before attempting to pattern match. + This ensures pattern matching works correctly and there is + no behavior change to the way supplements are called. + + ASTERISK-28767 #close + + Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7 + +2022-12-12 12:42 +0000 [881faf544f] Ben Ford + + * res_pjsip_sdp_rtp.c: Use correct timeout when put on hold. + + When a call is put on hold and it has moh_passthrough and rtp_timeout + set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is + expected to be used, but rtp_timeout is used instead. This change adds a + couple of checks for locally_held to determine if rtp_timeout_hold needs + to be used instead of rtp_timeout. + + ASTERISK-30350 + + Change-Id: I7b106fc244332014216d12bba851cefe884cc25f + +2022-11-14 07:12 +0000 [20d4775d0a] Naveen Albert + + * app_voicemail_odbc: Fix string overflow warning. + + Fixes a negative offset warning by initializing + the buffer to empty. + + Additionally, although it doesn't currently complain + about it, the size of a buffer is increased to + accomodate the maximum size contents it could have. + + ASTERISK-30240 #close + + Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877 + +2022-11-25 18:03 +0000 [cbb1fd2cb9] Naveen Albert + + * func_callerid: Warn about invalid redirecting reason. + + Currently, if a user attempts to set a Caller ID related + function to an invalid value, a warning is emitted, + except for when setting the redirecting reason. + We now emit a warning if we were unable to successfully + parse the user-provided reason. + + ASTERISK-30332 #close + + Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a + +2022-11-04 05:11 +0000 [115a1b4f0a] Igor Goncharovsky + + * res_pjsip: Fix path usage in case dialing with '@' + + Fix aor lookup on sip path addition. Issue happens in case of dialing + with @ and overriding user part of RURI. + + ASTERISK-30100 #close + Reported-by: Yury Kirsanov + + Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13 + +2022-11-21 21:37 +0000 [58404b5c22] Peter Fern + + * streams: Ensure that stream is closed in ast_stream_and_wait on error + + When ast_stream_and_wait returns an error (for example, when attempting + to stream to a channel after hangup) the stream is not closed, and + callers typically do not check the return code. This results in leaking + file descriptors, leading to resource exhaustion. + + This change ensures that the stream is closed in case of error. + + ASTERISK-30198 #close + Reported-by: Julien Alie + + Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803 + +2022-12-10 16:51 +0000 [36bea9ad33] Naveen Albert + + * app_sendtext: Remove references to removed applications. + + Removes see-also references to applications that don't + exist anymore (removed in Asterisk 19), + so these dead links don't show up on the wiki. + + ASTERISK-30347 #close + + Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909 + +2022-12-15 12:55 +0000 Asterisk Development Team + + * asterisk 20.1.0-rc1 Released. + +2022-12-15 06:40 +0000 [fefc236e7c] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 20.1.0 +2022-12-09 13:37 +0000 [01b3962201] Alexandre Fournier + + * res_geoloc: fix NULL pointer dereference bug + + The `ast_geoloc_datastore_add_eprofile` function does not return 0 on + success, it returns the size of the underlying datastore. This means + that the datastore will be freed and its pointer set to NULL when no + error occured at all. + + ASTERISK-30346 + + Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9 + +2022-12-13 09:25 +0000 [b6855755ce] Joshua C. Colp + + * res_pjsip_aoc: Don't assume a body exists on responses. + + When adding AOC to an outgoing response the code + assumed that a body would exist for comparing the + Content-Type. This isn't always true. + + The code now checks to make sure the response has + a body before checking the Content-Type. + + ASTERISK-21502 + + Change-Id: Iaead371434fc3bc693dad487228106a7d7a5ac76 + +2022-12-12 09:16 +0000 [2f9cdfbc50] Naveen Albert + + * app_if: Fix format truncation errors. + + Fixes format truncation warnings in gcc 12.2.1. + + ASTERISK-30349 #close + + Change-Id: I42be4edf0284358b906e765d1966b6b9d66e1d3c + +2022-11-01 15:37 +0000 [5c114dcb4a] Michael Kuron + + * manager: AOC-S support for AOCMessage + + ASTERISK-21502 + + Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08 + +2022-10-23 04:42 +0000 [fee9012fe1] Michael Kuron + + * res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip + + chan_sip supported sending AOC-D and AOC-E information in SIP INFO + messages in an "AOC" header in a format that was originally defined by + Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC + format that is supported by devices from multiple vendors, including + Snom phones with firmware >= 8.4.2 (released in 2010). + + This commit adds a new res_pjsip_aoc module that inserts AOC information + into outgoing messages or sends SIP INFO messages as described below. + It also fixes a small issue in res_pjsip_session which didn't always + call session supplements on outgoing_response. + + * AOC-S in the 180/183/200 responses to an INVITE request + * AOC-S in SIP INFO (if a 200 response has already been sent or if the + INVITE was sent by Asterisk) + * AOC-D in SIP INFO + * AOC-D in the 200 response to a BYE request (if the client hangs up) + * AOC-D in a BYE request (if Asterisk hangs up) + * AOC-E in the 200 response to a BYE request (if the client hangs up) + * AOC-E in a BYE request (if Asterisk hangs up) + + The specification defines one more, AOC-S in an INVITE request, which + is not implemented here because it is not currently possible in + Asterisk to have AOC data ready at this point in call setup. Once + specifying AOC-S via the dialplan or passing it through from another + SIP channel's INVITE is possible, that might be added. + + The SIP INFO requests are sent out immediately when the AOC indication + is received. The others are inserted into an appropriate outgoing + message whenever that is ready to be sent. In the latter case, the XML + is stored in a channel variable at the time the AOC indication is + received. Depending on where the AOC indications are coming from (e.g. + PRI or AMI), it may not always be possible to guarantee that the AOC-E + is available in time for the BYE. + + Successfully tested AOC-D and both variants of AOC-E with a Snom D735 + running firmware 10.1.127.10. It does not appear to properly support + AOC-S however, so that could only be tested by inspecting SIP traces. + + ASTERISK-21502 #close + Reported-by: Matt Jordan + + Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333 + +2022-12-08 04:33 +0000 [564349ff5d] Joshua C. Colp + + * ari: Destroy body variables in channel create. + + When passing a JSON body to the 'create' channel route + it would be converted into Asterisk variables, but never + freed resulting in a memory leak. + + This change makes it so that the variables are freed in + all cases. + + ASTERISK-30344 + + Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d + +2022-11-03 15:28 +0000 [b9c031c1f8] Naveen Albert + + * app_voicemail: Fix missing email in msg_create_from_file. + + msg_create_from_file currently does not dispatch emails, + which means that applications using this function, such + as MixMonitor, will not trigger notifications to users + (only AMI events are sent our currently). This is inconsistent + with other ways users can receive voicemail. + + This is fixed by adding an option that attempts to send + an email and falling back to just the notifications as + done now if that fails. The existing behavior remains + the default. + + ASTERISK-30283 #close + + Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7 + +2022-11-25 03:59 +0000 [58534b309f] Marcel Wagner + + * res_pjsip: Fix typo in from_domain documentation + + This fixes a small typo in the from_domain documentation on the endpoint documentation + + ASTERISK-30328 #close + + Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21 + +2022-11-21 12:53 +0000 [531eacd6c9] Naveen Albert + + * res_hep: Add support for named capture agents. + + Adds support for the capture agent name field + of the Homer protocol to Asterisk by allowing + users to specify a name that will be sent to + the HEP server. + + ASTERISK-30322 #close + + Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b + +2021-06-28 11:56 +0000 [b365ea8601] Naveen Albert + + * app_if: Adds conditional branch applications + + Adds the If, ElseIf, Else, ExitIf, and EndIf + applications for conditional execution + of a block of dialplan, similar to the While, + EndWhile, and ExitWhile applications. The + appropriate branch is executed at most once + if available and may be broken out of while + inside. + + ASTERISK-29497 + + Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49 + +2022-10-16 19:33 +0000 [0d6003fa9a] Naveen Albert + + * res_pjsip_session.c: Map empty extensions in INVITEs to s. + + Some SIP devices use an empty extension for PLAR functionality. + + Rather than rejecting these empty extensions, we now use the s + extension for such calls to mirror the existing PLAR functionality + in Asterisk (e.g. chan_dahdi). + + ASTERISK-30265 #close + + Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590 + +2022-11-17 13:30 +0000 [b83af13f65] Marcel Wagner + + * res_pjsip: Update contact_user to point out default + + Updates the documentation for the 'contact_user' field to point out the + default outbound contact if no contact_user is specified 's' + + ASTERISK-30316 #close + + Change-Id: I61f24fb9164e4d07e05908a2511805281874c876 + +2022-11-23 16:59 +0000 [80e6205bb0] Naveen Albert + + * res_adsi: Fix major regression caused by media format rearchitecture. + + The commit that rearchitected media formats, + a2c912e9972c91973ea66902d217746133f96026 (ASTERISK_23114) + introduced a regression by improperly translating code in res_adsi.c. + In particular, the pointer to the frame buffer was initialized + at the top of adsi_careful_send, rather than dynamically updating it + for each frame, as is required. + + This resulted in the first frame being repeatedly sent, + rather than advancing through the frames. + This corrupted the transmission of the CAS to the CPE, + which meant that CPE would never respond with the DTMF acknowledgment, + effectively completely breaking ADSI functionality. + + This issue is now fixed, and ADSI now works properly again. + + ASTERISK-29793 #close + + Change-Id: Icdeddf733eda2981c98712d1ac9cddc0db507dbe + +2022-07-21 14:07 +0000 [406143ae61] Naveen Albert + + * res_pjsip_header_funcs: Add custom parameter support. + + Adds support for custom URI and header parameters + in the From header in PJSIP. Parameters can be + both set and read using this function. + + ASTERISK-30150 #close + + Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428 + +2022-11-13 16:15 +0000 [83eb113e0f] Naveen Albert + + * func_presencestate: Fix invalid memory access. + + When parsing information from AstDB while loading, + it is possible that certain pointers are never + set, which leads to invalid memory access and + then, fatally, invalid free attempts on this memory. + We now initialize to NULL to prevent this. + + ASTERISK-30311 #close + + Change-Id: I6120681d04fd2c12a9473f35ce95a1f8e74e3929 + +2022-12-01 05:54 +0000 [b90e57758b] Naveen Albert + + * sig_analog: Fix no timeout duration. + + ASTERISK_28702 previously attempted to fix an + issue with flash hook hold timing out after + just under 17 minutes, when it should have never + been timing out. It fixed this by changing 999999 + to INT_MAX, but it did so in chan_dahdi, which + is the wrong place since ss_thread is now in + sig_analog and the one in chan_dahdi is mostly + dead code. + + This fixes this by porting the fix to sig_analog. + + ASTERISK-30336 #close + + Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15 + +2022-11-05 07:11 +0000 [52c7d3ed07] Naveen Albert + + * xmldoc: Allow XML docs to be reloaded. + + The XML docs are currently only loaded on + startup with no way to update them during runtime. + This makes it impossible to load modules that + use ACO/Sorcery (which require documentation) + if they are added to the source tree and built while + Asterisk is running (e.g. external modules). + + This adds a CLI command to reload the XML docs + during runtime so that documentation can be updated + without a full restart of Asterisk. + + ASTERISK-30289 #close + + Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07 + +2022-11-24 09:56 +0000 [a4bcdce1db] Naveen Albert + + * rtp_engine.h: Update examples using ast_format_set. + + This file includes some doxygen comments referencing + ast_format_set. This is an obsolete API that was + removed years back, but documentation was not fully + updated to reflect that. These examples are + updated to the current way of doing things + (using the format cache). + + ASTERISK-30327 #close + + Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7 + +2022-11-04 06:04 +0000 [691178c48e] Naveen Albert + + * app_mixmonitor: Add option to use real Caller ID for voicemail. + + MixMonitor currently uses the Connected Line as the Caller ID + for voicemails. This is due to the implementation being written + this way for use with Digium phones. However, in general this + is not correct for generic usage in the dialplan, and people + may need the real Caller ID instead. This adds an option to do that. + + ASTERISK-30286 #close + + Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c + +2022-11-29 14:02 +0000 [d476994768] Ben Ford + + * pjproject: 2.13 security fixes + + Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13. + + ASTERISK-30338 + + Change-Id: I86fdc003d5d22cb66e7cc6dc3313a8194f27eb69 + +2022-10-10 09:35 +0000 [7684c9e907] George Joseph + + * pjsip_transport_events: Fix possible use after free on transport + + It was possible for a module that registered for transport monitor + events to pass in a pjsip_transport that had already been freed. + This caused pjsip_transport_events to crash when looking up the + monitor for the transport. The fix is a two pronged approach. + + 1. We now increment the reference count on pjsip_transports when we + create monitors for them, then decrement the count when the + transport is going to be destroyed. + + 2. There are now APIs to register and unregister monitor callbacks + by "transport key" which is a string concatenation of the remote ip + address and port. This way the module needing to monitor the + transport doesn't have to hold on to the transport object itself to + unregister. It just has to save the transport_key. + + * Added the pjsip_transport reference increment and decrement. + + * Changed the internal transport monitor container key from the + transport->obj_name (which may not be unique anyway) to the + transport_key. + + * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that + fills a buffer with the transport_key using a passed-in + pjsip_transport. + + * Added the following functions: + ast_sip_transport_monitor_register_key + ast_sip_transport_monitor_register_replace_key + ast_sip_transport_monitor_unregister_key + and marked their non-key counterparts as deprecated. + + * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use + the new "key" monitor functions. + + NOTE: res_pjsip_registrar also uses the transport monitor + functionality but doesn't have a persistent object other than + contact to store a transport key. At this time, it continues to + use the non-key monitor functions. + + ASTERISK-30244 + + Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b + +2022-10-03 13:54 +0000 [81f10e847e] Mike Bradeen + + * manager: prevent file access outside of config dir + + Add live_dangerously flag to manager and use this flag to + determine if a configuation file outside of AST_CONFIG_DIR + should be read. + + ASTERISK-30176 + + Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404 + +2022-06-06 18:11 +0000 [eb1d7ab53c] Mike Bradeen + + * ooh323c: not checking for IE minimum length + + When decoding q.931 encoded calling/called number + now checking for length being less than minimum required. + + ASTERISK-30103 + + Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7 + +2022-11-11 14:30 +0000 [c7df5ee7c1] Naveen Albert + + * pbx_builtins: Allow Answer to return immediately. + + The Answer application currently waits for up to 500ms + for media, even if users specify a different timeout. + + This adds an option to not wait for media on the channel + by doing a raw answer instead. The default 500ms threshold + is also documented. + + ASTERISK-30308 #close + + Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4 + +2022-11-10 18:47 +0000 [5ede4e217a] Naveen Albert + + * chan_dahdi: Allow FXO channels to start immediately. + + Currently, chan_dahdi will wait for at least one + ring before an incoming call can enter the dialplan. + This is generally necessary in order to receive + the Caller ID spill and/or distinctive ringing + detection. + + However, if neither of these is required, then there + is nothing gained by waiting for one ring and this + unnecessarily delays call setup. Users can now + use immediate=yes to make FXO channels (FXS signaled) + begin processing dialplan as soon as Asterisk receives + the call. + + ASTERISK-30305 #close + + Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5 + +2022-09-07 07:06 +0000 [60b81eabe0] Maximilian Fridrich + + * core & res_pjsip: Improve topology change handling. + + This PR contains two relatively separate changes in channel.c and + res_pjsip_session.c which ensure that topology changes are not ignored + in cases where they should be handled. + + For channel.c: + + The function ast_channel_request_stream_topology_change only triggers a + stream topology request change indication, if the channel's topology + does not equal the requested topology. However, a channel could be in a + state where it is currently "negotiating" a new topology but hasn't + updated it yet, so the topology request change would be lost. Channels + need to be able to handle such situations internally and stream + topology requests should therefore always be passed on. + + In the case of chan_pjsip for example, it queues a session refresh + (re-INVITE) if it is currently in the middle of a transaction or has + pending requests (among other reasons). + + Now, ast_channel_request_stream_topology_change always indicates a + stream topology request change even if the requested topology equals the + channel's topology. + + For res_pjsip_session.c: + + The function resolve_refresh_media_states does not process stream state + changes if the delayed active state differs from the current active + state. I.e. if the currently active stream state has changed between the + time the sip session refresh request was queued and the time it is being + processed, the session refresh is ignored. However, res_pjsip_session + contains logic that ensures that session refreshes are queued and + re-queued correctly if a session refresh is currently not possible. So + this check is not necessary and led to some session refreshes being + lost. + + Now, a session refresh is done even if the delayed active state differs + from the current active state and it is checked whether the delayed + pending state differs from the current active - because that means a + refresh is necessary. + + Further, the unit test of resolve_refresh_media_states was adapted to + reflect the new behavior. I.e. the changes to delayed pending are + prioritized over the changes to current active because we want to + preserve the original intention of the pending state. + + ASTERISK-30184 + + Change-Id: Icd0703295271089057717006730b555b9a1d4e5a + +2022-09-24 05:15 +0000 [2efa290d3c] Naveen Albert + + * sla: Prevent deadlock and crash due to autoservicing. + + SLAStation currently autoservices the station channel before + creating a thread to actually dial the trunk. This leads + to duplicate servicing of the channel which causes assertions, + deadlocks, crashes, and moreover not the correct behavior. + + Removing the autoservice prevents the crash, but if the station + hangs up before the trunk answers, the call hangs since the hangup + was never serviced on the channel. + + This is fixed by not autoservicing the channel, but instead + servicing it in the thread dialing the trunk, since it is doing + so synchronously to begin with. Instead of sleeping for 100ms + in a loop, we simply use the channel for timing, and abort + if it disappears. + + The same issue also occurs with SLATrunk when a call is answered, + because ast_answer invokes ast_waitfor_nandfds. Thus, we use + ast_raw_answer instead which does not cause any conflict and allows + the call to be answered normally without thread blocking issues. + + ASTERISK-29998 #close + + Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8 + +2022-11-07 09:30 +0000 [ce2153fc5a] Jaco Kroon + + * Build system: Avoid executable stack. + + Found in res_geolocation, but I believe others may have similar issues, + thus not linking to a specific issue. + + Essentially gcc doesn't mark the stack for being non-executable unless + it's compiling the source, this informs ld via gcc to mark the object as + not requiring an executable stack (which a binary blob obviously + doesn't). + + ASTERISK-30321 + + Change-Id: I71bcc2fd1fe0c82a28b3257405d6f2b566fd9bfc + Signed-off-by: Jaco Kroon + +2022-11-10 06:11 +0000 [002afc3f2a] Naveen Albert + + * func_json: Fix memory leak. + + A memory leak was present in func_json due to + using ast_json_free, which just calls ast_free, + as opposed to recursively freeing the JSON + object as needed. This is now fixed to use the + right free functions. + + ASTERISK-30293 #close + + Change-Id: I982324dde841dc9147c8d8ad35c8719daf418b49 + +2022-11-10 06:20 +0000 [1e77b8c473] Naveen Albert + + * test_json: Remove duplicated static function. + + Removes the function mkstemp_file and uses + ast_file_mkftemp from file.h instead. + + ASTERISK-30295 #close + + Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609 + +2022-11-16 05:40 +0000 [61922d2934] Joshua C. Colp + + * res_agi: Respect "transmit_silence" option for "RECORD FILE". + + The "RECORD FILE" command in res_agi has its own + implementation for actually doing the recording. This + has resulted in it not actually obeying the option + "transmit_silence" when recording. + + This change causes it to now send silence if the + option is enabled. + + ASTERISK-30314 + + Change-Id: Ib3a85601ff35d1b904f836691bad8a4b7e957174 + +2022-11-03 15:56 +0000 [6e59b01e1a] Naveen Albert + + * app_mixmonitor: Add option to delete files on exit. + + Adds an option that allows MixMonitor to delete + its copy of any recording files before exiting. + + This can be handy in conjunction with options + like m, which copy the file elsewhere, and the + original files may no longer be needed. + + ASTERISK-30284 #close + + Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e + +2022-11-03 17:01 +0000 [49cfdbbdff] Naveen Albert + + * manager: Update ModuleCheck documentation. + + The ModuleCheck XML documentation falsely + claims that the module's version number is returned. + This has not been the case since 14, since the version + number is not available anymore, but the documentation + was not changed at the time. It is now updated to + reflect this. + + ASTERISK-30285 #close + + Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91 + +2022-11-06 10:39 +0000 [8142b313c3] Naveen Albert + + * file.c: Don't emit warnings on winks. + + Adds an ignore case for wink since it should + pass through with no warning. + + ASTERISK-30290 #close + + Change-Id: Ieb7e34daa717357ac5c93efb0059f6c2321f16ad + +2022-11-02 09:24 +0000 [0c1c623dee] George Joseph + + * runUnittests.sh: Save coredumps to proper directory + + Fixed the specification of "outputdir" when calling ast_coredumper + so the txt files are saved in the correct place. + + ASTERISK-30282 + + Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6 + +2022-10-01 14:49 +0000 [dfe2f38642] Naveen Albert + + * app_stack: Print proper exit location for PBXless channels. + + When gosub is executed on channels without a PBX, the context, + extension, and priority are initialized to the channel driver's + default location for that endpoint. As a result, the last Return + will restore this location and the Gosub logs will print out bogus + information about our exit point. + + To fix this, on channels that don't have a PBX, the execution + location is left intact on the last return if there are no + further stack frames left. This allows the correct location + to be printed out to the user, rather than the bogus default + context. + + ASTERISK-30076 #close + + Change-Id: I1d42a99c9aa9e3708d32718863175158a894e414 + +2022-11-02 07:41 +0000 [f723b465e5] George Joseph + + * chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer + + unicast_rtp_request() was setting the channel variables like this: + + pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", + ast_sockaddr_stringify_addr(&local_address)); + ast_rtp_instance_get_local_address(instance, &local_address); + pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", + ast_sockaddr_stringify_port(&local_address)); + + ...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being + set before local_address was set. In fact, the address part of + local_address was set earlier in the function, just not the port. + This was confusing however so ast_rtp_instance_get_local_address() + is now being called before setting UNICASTRTP_LOCAL_ADDRESS. + + ASTERISK-30281 + + Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4 + +2022-10-13 11:19 +0000 [50e2921a48] Mike Bradeen + + * res_pjsip: prevent crash on websocket disconnect + + When a websocket (or potentially any stateful connection) is quickly + created then destroyed, it is possible that the qualify thread will + destroy the transaction before the initialzing thread is finished + with it. + + Depending on the timing, this can cause an assertion within pjsip. + + To prevent this, ast_send_stateful_response will now create the group + lock and add a reference to it before creating the transaction. + + While this should resolve the crash, there is still the potential that + the contact will not be cleaned up properly, see:ASTERISK~29286. As a + result, the contact has to 'time out' before it will be removed. + + ASTERISK-28689 + + Change-Id: Id050fded2247a04d8f0fc5b8a2cf3e5482cb8cee + +2022-10-27 06:32 +0000 [afd86b47c1] Naveen Albert + + * tcptls: Prevent crash when freeing OpenSSL errors. + + write_openssl_error_to_log has been erroneously + using ast_free instead of free, which will + cause a crash when MALLOC_DEBUG is enabled since + the memory was not allocated by Asterisk's memory + manager. This changes it to use the actual free + function directly to avoid this. + + ASTERISK-30278 #close + + Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d + +2022-09-09 12:20 +0000 [096529d33f] Igor Goncharovsky + + * res_pjsip_outbound_registration: Allow to use multiple proxies for registration + + Current registration code use pjsip_parse_uri to verify outbound_proxy + that is different from the reading this option for the endpoint. This + made value with multiple proxies invalid for registration pjsip settings. + Removing URI validation helps to use registration through multiple proxies. + + ASTERISK-30217 #close + + Change-Id: I064558e66f04b9f3260c46181812a01349761357 + +2022-10-23 10:16 +0000 [ca8900b0f6] Naveen Albert + + * tests: Fix compilation errors on 32-bit. + + Fix compilation errors caused by using size_t + instead of uintmax_t and non-portable format + specifiers. + + ASTERISK-30273 #close + + Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee + +2022-08-26 03:59 +0000 [12445040d3] Henning Westerholt + + * res_pjsip: return all codecs on a re-INVITE without SDP + + Currently chan_pjsip on receiving a re-INVITE without SDP will only + return the codecs that are previously negotiated and not offering + all enabled codecs. + + This causes interoperability issues with different equipment (e.g. + from Cisco) for some of our customers and probably also in other + scenarios involving 3PCC infrastructure. + + According to RFC 3261, section 14.2 we SHOULD return all codecs + on a re-INVITE without SDP + + The PR proposes a new parameter to configure this behaviour: + all_codecs_on_empty_reinvite. It includes the code, documentation, + alembic migrations, CHANGES file and example configuration additions. + + ASTERISK-30193 #close + + Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148 + +2022-10-14 17:24 +0000 [40b52322e5] Naveen Albert + + * res_pjsip_notify: Add option support for AMI. + + The PJSIP notify CLI commands allow for using + "options" configured in pjsip_notify.conf. + + This allows these same options to be used in + AMI actions as well. + + Additionally, as part of this improvement, + some repetitive common code is refactored. + + ASTERISK-30263 #close + + Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5 + +2022-07-20 19:03 +0000 [c32b39d123] Naveen Albert + + * res_pjsip_logger: Add method-based logging option. + + Expands the pjsip logger to support the ability to filter + by SIP message method. This can make certain types of SIP debugging + easier by only logging messages of particular method(s). + + ASTERISK-30146 #close + + Co-authored-by: Sean Bright + Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f + +2022-10-06 11:51 +0000 [50a4495799] Frederic LE FOLL + + * Dialing API: Cancel a running async thread, may not cancel all calls + + race condition: ast_dial_join() may not cancel outgoing call, if + function is called just after called party answer and before + application execution (bit is_running_app not yet set). + + This fix adds ast_softhangup() calls in addition to existing + pthread_kill() when is_running_app is not set. + + ASTERISK-30258 + + Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4 + +2022-10-23 17:46 +0000 [180ca32565] Naveen Albert + + * chan_dahdi: Fix unavailable channels returning busy. + + This fixes dahdi_request to properly set the cause + code to CONGESTION instead of BUSY if no channels + were actually available. + + Currently, the cause is erroneously set to busy + if the channel itself is found, regardless of its + current state. However, if the channel is not available + (e.g. T1 down, card not operable, etc.), then the + channel itself may not be in a functional state, + in which case CHANUNAVAIL is the correct cause to use. + + This adds a simple check to ensure that busy tone + is only returned if a channel is encountered that + has an owner, since that is the only possible way + that a channel could actually be busy. + + ASTERISK-30274 #close + + Change-Id: Iad5870223c081240c925b19df8d6af136953b994 + +2022-10-16 15:35 +0000 [9258d8212a] Naveen Albert + + * res_pjsip_pubsub: Prevent removing subscriptions. + + pjproject does not provide any mechanism of removing + event packages, which means that once a subscription + handler is registered, it is effectively permanent. + + pjproject will assert if the same event package is + ever registered again, so currently unloading and + loading any Asterisk modules that use subscriptions + will cause a crash that is beyond our control. + + For that reason, we now prevent users from being + able to unload these modules, to prevent them + from ever being loaded twice. + + ASTERISK-30264 #close + + Change-Id: I7fdcb1a5e44d38b7ba10c44259fe98f0ae9bc12c + +2022-09-28 07:38 +0000 [407216a0a5] Naveen Albert + + * say: Don't prepend ampersand erroneously. + + Some logic in say.c for determining if we need + to also add an ampersand for file seperation was faulty, + as non-successful files would increment the count, causing + a leading ampersand to be added improperly. + + This is fixed, and a unit test that captures this regression + is also added. + + ASTERISK-30248 #close + + Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b + +2022-09-16 13:45 +0000 [d0bea5a725] Philip Prindeville + + * res_crypto: handle unsafe private key files + + ASTERISK-30213 #close + + Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a + +2022-09-29 15:55 +0000 [907d7e7d7d] Mike Bradeen + + * audiohook: add directional awareness + + Add enum to allow setting optional direction. If set to only one + direction, only feed matching-direction frames to the associated + slin factory. + + This prevents mangling the transcoder on non-mixed frames when the + READ and WRITE frames would have otherwise required it. Also + removes the need to mute or discard the un-wanted frames as they + are no longer added in the first place. + + res_stasis_snoop is changed to use this addition to set direction + on audiohook based on spy direction. + + If no direction is set, the ast_audiohook_init will init this enum + to BOTH which maintains existing functionality. + + ASTERISK-30252 + + Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb + +2022-06-01 11:06 +0000 [b331caca30] Naveen Albert + + * cdr: Allow bridging and dial state changes to be ignored. + + Allows bridging, parking, and dial messages to be globally + ignored for all CDRs such that only a single CDR record + is generated per channel. + + This is useful when CDRs should endure for the lifetime of + an entire channel and bridging and dial updates in the + dialplan should not result in multiple CDR records being + created for the call. With the ignore bridging option, + bridging changes have no impact on the channel's CDRs. + With the ignore dial state option, multiple Dials and their + outcomes have no impact on the channel's CDRs. The + last disposition on the channel is preserved in the CDR, + so the actual disposition of the call remains available. + + These two options can reduce the amount of "CDR hacks" that + have hitherto been necessary to ensure that CDR was not + "spoiled" by these messages if that was undesired, such as + putting a dummy optimization-disabled local channel between + the caller and the actual call and putting the CDR on the channel + in the middle to ensure that CDR would persist for the entire + call and properly record start, answer, and end times. + Enabling these options is desirable when calls correspond + to the entire lifetime of channels and the CDR should + reflect that. + + Current default behavior remains unchanged. + + ASTERISK-30091 #close + + Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832 + +2022-09-30 06:24 +0000 [e0e7f35730] Naveen Albert + + * res_tonedetect: Add ringback support to TONE_DETECT. + + Adds support for detecting audible ringback tone + to the TONE_DETECT function using the p option. + + ASTERISK-30254 #close + + Change-Id: Ie2329ff245248768367d26749c285fbe823f6414 + +2022-10-01 17:08 +0000 [98fc05f13b] Naveen Albert + + * chan_dahdi: Resolve format truncation warning. + + Fixes a format truncation warning in notify_message. + + ASTERISK-30256 #close + + Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1 + +2022-09-16 18:29 +0000 [ef74ecacc7] Philip Prindeville + + * res_crypto: don't modify fname in try_load_key() + + "fname" is passed in as a const char *, but strstr() mangles that + into a char *, and we were attempting to modify the string in place. + This is an unwanted (and undocumented) side-effect. + + ASTERISK-30213 + + Change-Id: Ifa36d352aafeb7f9beec3f746332865c7d21e629 + +2022-09-15 22:45 +0000 [5e2485b5c0] Philip Prindeville + + * res_crypto: use ast_file_read_dirs() to iterate + + ASTERISK-30213 + + Change-Id: I115f5f8942ffcfb23cd2559a55bac8a2eba081e0 + +2022-09-27 09:35 +0000 [2a500b325a] George Joseph + + * res_geolocation: Update wiki documentation + + Also added a note to the geolocation.conf.sample file + and added a README to the res/res_geolocation/wiki + directory. + + Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc + +2022-07-26 07:01 +0000 [0d2e140123] Maximilian Fridrich + + * res_pjsip: Add mediasec capabilities. + + This patch adds support for mediasec SIP headers and SDP attributes. + These are defined in RFC 3329, 3GPP TS 24.229 and + draft-dawes-sipcore-mediasec-parameter. The new features are + implemented so that a backbone for RFC 3329 is present to streamline + future work on RFC 3329. + + With this patch, Asterisk can communicate with Deutsche Telekom trunks + which require these fields. + + ASTERISK-30032 + + Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2 + +2022-09-28 07:44 +0000 [7f80830ced] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 20.0.0 +2022-09-19 19:53 +0000 [62881c668b] Holger Hans Peter Freyther + + * res_prometheus: Do not crash on invisible bridges + + Avoid crashing by skipping invisible bridges and checking the + snapshot for a null pointer. In effect this is how the bridges + are enumerated in res/ari/resource_bridges.c already. + + ASTERISK-30239 + ASTERISK-30237 + + Change-Id: I58ef9f44036feded5966b5fc70ae754f8182883d + +2022-09-19 12:35 +0000 [8afb313a43] Naveen Albert + + * res_pjsip_geolocation: Change some notices to debugs. + + If geolocation is not in use for an endpoint, the NOTICE + log level is currently spammed with messages about this, + even though nothing is wrong and these messages provide + no real value. These log messages are therefore changed + to debugs. + + ASTERISK-30241 #close + + Change-Id: I656b355d812f67cc0f0fdf09b00b0e1458598bb4 + +2022-09-24 07:08 +0000 [7335b0cffe] Birger Harzenetter (license 5870) + + * db: Fix incorrect DB tree count for AMI. + + The DBGetTree AMI action's ListItem previously + always reported 1, regardless of the count. This + is corrected to report the actual count. + + ASTERISK-30245 #close + patches: + gettreecount.diff submitted by Birger Harzenetter (license 5870) + + Change-Id: I46d8992710f1b8524426b1255f57d1ef4a4934d4 + +2022-09-21 18:17 +0000 [407167cc28] Naveen Albert + + * func_logic: Don't emit warning if both IF branches are empty. + + The IF function currently emits warnings if both IF branches + are empty. However, there is no actual necessity that either + branch be non-empty as, unlike other conditional applications/ + functions, nothing is inherently done with IF, and both + sides could legitimately be empty. The warning is thus turned + into a debug message. + + ASTERISK-30243 #close + + Change-Id: I5250625dd720f95e1859b5dfb933905d7e7a730e + +2022-09-11 17:13 +0000 [a5ec60e6c6] Naveen Albert + + * features: Add no answer option to Bridge. + + Adds the n "no answer" option to the Bridge application + so that answer supervision can not automatically + be provided when Bridge is executed. + + Additionally, a mechanism (dialplan variable) + is added to prevent bridge targets (typically the + target of a masquerade) from answering the channel + when they enter the bridge. + + ASTERISK-30223 #close + + Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d + +2022-09-08 19:12 +0000 [1e29607b5c] Naveen Albert + + * app_bridgewait: Add option to not answer channel. + + Adds the n option to not answer the channel when calling + BridgeWait, so the application can be used without + forcing answer supervision. + + ASTERISK-30216 #close + + Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a + +2022-08-15 14:59 +0000 [8c791f9a65] Naveen Albert + + * app_amd: Add option to play audio during AMD. + + Adds an option that will play an audio file + to the party while AMD is running on the + channel, so the called party does not just + hear silence. + + ASTERISK-30179 #close + + Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6 + +2022-09-15 22:35 +0000 [3e7ce90f9c] Philip Prindeville + + * test: initialize capture structure before freeing + + ASTERISK-30232 #close + + Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902 + +2021-05-17 09:19 +0000 [1ed4518328] Naveen Albert + + * func_export: Add EXPORT function + + Adds the EXPORT function, which allows write + access to variables and functions on other + channels. + + ASTERISK-29432 #close + + Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf + +2022-07-26 08:40 +0000 [5bbad0d27c] Maximilian Fridrich + + * res_pjsip: Add 100rel option "peer_supported". + + This patch adds a new option to the 100rel parameter for pjsip + endpoints called "peer_supported". When an endpoint with this option + receives an incoming request and the request indicated support for the + 100rel extension, then Asterisk will send 1xx responses reliably. If + the request did not indicate 100rel support, Asterisk sends 1xx + responses normally. + + ASTERISK-30158 + + Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad + +2022-09-10 10:15 +0000 [8aae0b9f08] Naveen Albert + + * func_scramble: Fix null pointer dereference. + + Fix segfault due to null pointer dereference + inside the audiohook callback. + + ASTERISK-30220 #close + + Change-Id: Ideb80f606974366e89d619d908744230b5a5a259 + +2022-09-05 01:59 +0000 [278c5726ca] Jaco Kroon + + * manager: be more aggressive about purging http sessions. + + If we find that n_max (currently hard wired to 1) sessions were purged, + schedule the next purge for 1ms into the future rather than 5000ms (as + per current). This way we will purge up to 1000 sessions per second + rather than 1 every 5 seconds. + + This mitigates a build-up of sessions should http sessions gets + established faster than 1 per 5 seconds. + + Change-Id: I9820d39aa080109df44fe98c1325cafae48d54f5 + Signed-off-by: Jaco Kroon + +2022-09-11 15:40 +0000 [ab1dbfef75] Naveen Albert + + * func_strings: Add trim functions. + + Adds TRIM, LTRIM, and RTRIM, which can be used + for trimming leading and trailing whitespace + from strings. + + ASTERISK-30222 #close + + Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554 + +2022-09-16 09:58 +0000 [e25b690d10] George Joseph + + * res_crypto: Memory issues and uninitialized variable errors + + ASTERISK-30235 + + Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd + +2022-09-16 08:43 +0000 [e33f2dcc0f] George Joseph + + * res_geolocation: Fix issues exposed by compiling with -O2 + + Fixed "may be used uninitialized" errors in geoloc_config.c. + + ASTERISK-30234 + + Change-Id: I1ea336bf7abbc16fa59b75720f0db8f1d960b3d4 + +2022-09-13 14:41 +0000 [026dc08529] Philip Prindeville + + * res_crypto: don't complain about directories + + ASTERISK-30226 #close + + Change-Id: I5695fb0c9521f112f754b8362cff2a8f3eff05c5 + +2022-09-14 14:50 +0000 Asterisk Development Team + + * asterisk 20.0.0-rc1 Released. + +2022-09-14 09:25 +0000 [f01ed3eea4] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 20.0.0 +2022-08-15 14:30 +0000 [7a44296ca9] Mike Bradeen + + * res_pjsip: Add user=phone on From and PAID for usereqphone=yes + + Adding user=phone to local-side uri's when user_eq_phone=yes is set for + an endpoint. Previously this would only add the header to the To and R-URI. + + ASTERISK-30178 + + Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600 + +2022-09-13 08:14 +0000 [8cbea1c7ef] George Joseph + + * res_geolocation: Fix segfault when there's an empty element + + Fixed a segfault caused by var_list_from_loc_info() encountering + an empty location info element. + + Fixed an issue in ast_strsep() where a value with only whitespace + wasn't being preserved. + + Fixed an issue in ast_variable_list_from_quoted_string() where + an empty value was considered a failure. + + ASTERISK-30215 + Reported by: Dan Cropp + + Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a + +2022-08-13 11:32 +0000 [80bc844fd6] sungtae kim + + * res_musiconhold: Add option to not play music on hold on unanswered channels + + This change adds an option, answeredonly, that will prevent music on + hold on channels that are not answered. + + ASTERISK-30135 + + Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08 + +2022-08-02 12:15 +0000 [881a3f2306] Ben Ford + + * res_pjsip: Add TEL URI support for basic calls. + + This change allows TEL URI requests to come through for basic calls. The + allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To + headers will now allow TEL URIs, as well as the request URI. + + Support is only for TEL URIs present in traffic from a remote party. + Asterisk does not generate any TEL URIs on its own. + + ASTERISK-26894 + + Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a + +2022-03-24 14:22 +0000 [3e054c9ebc] Philip Prindeville + + * res_crypto: Use EVP API's instead of legacy API's + + ASTERISK-30046 #close + + Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2 + +2022-05-03 19:27 +0000 [736cdf84f4] Philip Prindeville + + * test: Add coverage for res_crypto + + We're validating the following functionality: + + encrypting a block of data with RSA + decrypting a block of data with RSA + signing a block of data with RSA + verifying a signature with RSA + encrypting a block of data with AES-ECB + encrypting a block of data with AES-ECB + + as well as accessing test keys from the keystore. + + ASTERISK-30045 #close + + Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d + +2022-07-26 12:38 +0000 [2d7656cb50] Philip Prindeville + + * res_crypto: make keys reloadable on demand for testing + + ASTERISK-30045 + + Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8 + +2022-05-03 13:12 +0000 [5809d879b0] Philip Prindeville + + * test: Add test coverage for capture child process output + + ASTERISK-30037 #close + + Change-Id: I0273e85eeeb6b8e46703f24cd74d84f3daf0a69a + +2022-07-26 14:38 +0000 [2c4c44ca64] Philip Prindeville + + * main/utils: allow checking for command in $PATH + + ASTERISK-30037 + + Change-Id: I4b6f7264c8c737c476c798d2352f3232b263bbdf + +2022-05-02 23:49 +0000 [b9df2c481b] Philip Prindeville + + * test: Add ability to capture child process output + + ASTERISK-30037 + + Change-Id: Icbf84ce05addb197a458361c35d784e460d8d6c2 + +2022-04-26 20:44 +0000 [d13afaf302] Philip Prindeville + + * res_crypto: Don't load non-regular files in keys directory + + ASTERISK-30046 + + Change-Id: Ie77e0648f8b0b1c2159fb24662d1989cfd4cc36d + +2022-09-08 09:12 +0000 [2dac2bf8dc] Naveen Albert + + * func_frame_trace: Remove bogus assertion. + + The FRAME_TRACE function currently asserts if it sees + a MASQUERADE_NOTIFY. However, this is a legitimate thing + that can happen so asserting is inappropriate, as there + are no clear negative ramifications of such a thing. This + is adjusted to be like the other frames to print out + the subclass. + + ASTERISK-30210 #close + + Change-Id: I8ecbdcf17e35f64bdeab42868471f581ad1d1a56 + +2022-07-27 14:54 +0000 [c487425620] Naveen Albert + + * lock.c: Add AMI event for deadlocks. + + Adds an AMI event to indicate that a deadlock + has likely started, when Asterisk is compiled + with DETECT_DEADLOCKS enabled. This can make + it easier to perform automated deadlock detection + and take appropriate action (such as doing a core + dump). Unlike the deadlock warnings, the AMI event + is emitted only once per deadlock. + + ASTERISK-30161 #close + + Change-Id: Ifc6ed3e390f8b4cff7f8077a50e4d7a5b54e42fb + +2022-09-04 14:38 +0000 [205c7c8d21] Naveen Albert + + * app_confbridge: Add end_marked_any option. + + Adds the end_marked_any option, which can be used + to kick a user from a conference if any marked user + leaves. + + ASTERISK-30211 #close + + Change-Id: I9e8da7ccb892e522546c0f2b5476d172e022c2f5 + +2022-09-03 18:19 +0000 [2de016b181] Naveen Albert + + * pbx_variables: Use const char if possible. + + Use const char for char arguments to + pbx_substitute_variables_helper_full_location + that can do so (context and exten). + + ASTERISK-30209 #close + + Change-Id: I001357177e9c3dca2b2b4eebc5650c1095b3da6f + +2022-08-25 08:00 +0000 [05f42806cc] George Joseph + + * res_geolocation: Add two new options to GEOLOC_PROFILE + + Added an 'a' option to the GEOLOC_PROFILE function to allow + variable lists like location_info_refinement to be appended + to instead of replacing the entire list. + + Added an 'r' option to the GEOLOC_PROFILE function to resolve all + variables before a read operation and after a Set operation. + + Added a few missing parameters to the ones allowed for writing + with GEOLOC_PROFILE. + + Fixed a bug where calling GEOLOC_PROFILE to read a parameter + might actually update the profile object. + + Cleaned up XML documentation a bit. + + ASTERISK-30190 + + Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c + +2022-08-18 07:29 +0000 [c799db6a21] George Joseph + + * res_geolocation: Allow location parameters on the profile object + + You can now specify the location object's format, location_info, + method, location_source and confidence parameters directly on + a profile object for simple scenarios where the location + information isn't common with any other profiles. This is + mutually exclusive with setting location_reference on the + profile. + + Updated appdocsxml.dtd to allow xi:include in a configObject + element. This makes it easier to link to complete configOptions + in another object. This is used to add the above fields to the + profile object without having to maintain the option descriptions + in two places. + + ASTERISK-30185 + + Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569 + +2022-08-17 08:15 +0000 [4ffc5561c4] George Joseph + + * res_geolocation: Add profile parameter suppress_empty_ca_elements + + Added profile parameter "suppress_empty_ca_elements" that + will cause Civic Address elements that are empty to be + suppressed from the outgoing PIDF-LO document. + + Fixed a possible SEGV if a sub-parameter value didn't have a + value. + + ASTERISK-30177 + + Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd + +2022-08-16 07:25 +0000 [2d5a6498dd] George Joseph + + * res_geolocation: Add built-in profiles + + The trigger to perform outgoing geolocation processing is the + presence of a geoloc_outgoing_call_profile on an endpoint. This + is intentional so as to not leak location information to + destinations that shouldn't receive it. In a totally dynamic + configuration scenario however, there may not be any profiles + defined in geolocation.conf. This makes it impossible to do + outgoing processing without defining a "dummy" profile in the + config file. + + This commit adds 4 built-in profiles: + "" + "" + "" + "" + The profiles are empty except for having their precedence + set and can be set on an endpoint to allow processing without + entries in geolocation.conf. "" is actually the + best one to use in this situation. + + ASTERISK-30182 + + Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a + +2022-08-30 08:01 +0000 [f3de933b16] Joshua C. Colp + + * res_pjsip_sdp_rtp: Skip formats without SDP details. + + When producing an outgoing SDP we iterate through the configured + formats and produce SDP information. It is possible for some + configured formats to not have SDP information available. If this + is the case we skip over them to allow the SDP to still be + produced. + + ASTERISK-29185 + + Change-Id: I3e37569aa4ca341260e6ca5904dc2f75e46a1749 + +2022-05-03 07:53 +0000 [c7612521be] Naveen Albert + + * cli: Prevent assertions on startup from bad ao2 refs. + + If "core show channels" is run before startup has completed, it + is possible for bad ao2 refs to occur because the system is not + yet fully initialized. This will lead to an assertion failing. + + To prevent this, initialization of CLI builtins is moved to be + later along in the main load sequence. Core CLI commands are + loaded at the same time, but channel-related commands are loaded + later on. + + ASTERISK-29846 #close + + Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6 + +2022-08-19 08:24 +0000 [a0713a9f70] Joshua C. Colp + + * pjsip: Add TLS transport reload support for certificate and key. + + This change adds support using the pjsip_tls_transport_restart + function for reloading the TLS certificate and key, if the filenames + remain unchanged. This is useful for Let's Encrypt and other + situations. Note that no restart of the transport will occur if + the certificate and key remain unchanged. + + ASTERISK-30186 + + Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0 + +2022-08-25 06:51 +0000 [754346a4a9] Naveen Albert + + * res_tonedetect: Fix typos referring to wrong variables. + + Fixes two typos that cause fax detection to not work. + One refers to the wrong frame variable, and the other + refers to the subclass.integer instead of the frametype + as it should. + + ASTERISK-30192 #close + + Change-Id: I7b35fdb7bcf25a29a212eee37c20812c64ab3ef1 + +2022-08-17 13:30 +0000 [46776c77c4] Mike Bradeen + + * alembic: add missing ps_endpoints columns + + The following required columns were missing, + now added to the ps_endpoints table: + + incoming_call_offer_pref + outgoing_call_offer_pref + stir_shaken_profile + + ASTERISK-29453 + + Change-Id: I5cf565edf30195844d6acbc1e1de8c5f0d837568 + +2022-08-19 11:02 +0000 [583e017f34] Sean Bright + + * chan_dahdi.c: Resolve a format-truncation build warning. + + With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0: + + > chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated + > writing up to 255 bytes into a region of size between 242 and 252 + > [-Werror=format-truncation=] + + This removes the error-prone sizeof(...) calculations in favor of just + doubling the size of the base buffer. + + Change-Id: I2d276785286730d3d5d0a921bcea2e065dbf27c5 + +2022-08-03 09:55 +0000 [12c4c1bf5f] Alexei Gradinari + + * res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update + + Set termination state to old subscriptions to prevent queueing and sending + NOTIFY messages on exten/device state changes. + + Postpone destruction of old subscriptions until all already queued tasks + that may be using old subscriptions have completed. + + ASTERISK-29906 + + Change-Id: I96582aad3a26515ca73a8460ee6756f56f6ba23b + +2022-08-15 07:34 +0000 [155c796203] Sean Bright + + * channel.h: Remove redundant declaration. + + The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set() + for us, so no need to declare it separately. + + Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a + +2022-02-05 06:13 +0000 [3fa66c92b5] Naveen Albert + + * features: Add transfer initiation options. + + Adds additional control options over the transfer + feature functionality to give users more control + in how the transfer feature sounds and works. + + First, the "transfer" sound that plays when a transfer is + initiated can now be customized by the user in + features.conf, just as with the other transfer sounds. + + Secondly, the user can now specify the transfer extension + in advance by using the TRANSFER_EXTEN variable. If + a valid extension is contained in this variable, the call + will automatically be transferred to this destination. + Otherwise, it will fall back to collecting the extension + from the user as is always done now. + + ASTERISK-29899 #close + + Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3 + +2022-08-31 14:16 +0000 [adffb975dc] Mike Bradeen + + * CI: Fixing path issue on venv check + + ASTERISK-26826 + + Change-Id: I07388d16f74452cebc9c981f99044eb6b77df792 + +2022-08-11 13:39 +0000 [4fc9e06db1] Mike Bradeen + + * CI: use Python3 virtual environment + + Requires Python3 testsuite changes + + ASTERISK-26826 + + Change-Id: I92ec7dec751ad455503a584d6e860db88c56d6bc + +2022-07-28 16:12 +0000 [e2e049e473] Naveen Albert + + * general: Very minor coding guideline fixes. + + Fixes a few coding guideline violations: + * Use of C99 comments + * Opening brace on same line as function prototype + + ASTERISK-30163 #close + + Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7 + +2022-08-05 08:50 +0000 [8a8416e365] George Joseph + + * res_geolocation: Address user issues, remove complexity, plug leaks + + * Added processing for the 'confidence' element. + * Added documentation to some APIs. + * removed a lot of complex code related to the very-off-nominal + case of needing to process multiple location info sources. + * Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes + one eprofile instead of a datastore of multiples. + * Plugged a huge leak in XML processing that arose from + insufficient documentation by the libxml/libxslt authors. + * Refactored stylesheets to be more efficient. + * Renamed 'profile_action' to 'profile_precedence' to better + reflect it's purpose. + * Added the config option for 'allow_routing_use' which + sets the value of the 'Geolocation-Routing' header. + * Removed the GeolocProfileCreate and GeolocProfileDelete + dialplan apps. + * Changed the GEOLOC_PROFILE dialplan function as follows: + * Removed the 'profile' argument. + * Automatically create a profile if it doesn't exist. + * Delete a profile if 'inheritable' is set to no. + * Fixed various bugs and leaks + * Updated Asterisk WiKi documentation. + + ASTERISK-30167 + + Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0 + +2022-07-30 16:15 +0000 [ff044c222b] Naveen Albert + + * chan_iax2: Add missing options documentation. + + Adds missing dial resource option documentation. + + ASTERISK-30164 #close + + Change-Id: I674e1fc9b1e5d67a20599bd4b418ce294d48fc83 + +2022-07-31 19:30 +0000 [dc7ec11c26] Naveen Albert + + * app_confbridge: Fix memory leak on updated menu options. + + If the CONFBRIDGE function is used to dynamically set + menu options, a memory leak occurs when a menu option + that has been set is overridden, since the menu entry + is not destroyed before being freed. This ensures that + it is. + + Additionally, logic that duplicates the destroy function + is removed in lieu of the destroy function itself. + + ASTERISK-28422 #close + + Change-Id: I71cfb5c24e636984d41086d1333a416dc12ff995 + +2022-07-19 09:05 +0000 [30d7a212b0] George Joseph + + * Geolocation: Wiki Documentation + + Change-Id: I68ba22db0a69d9e2eabcc2141b48a2395f7f1a23 + +2022-07-28 06:10 +0000 [f4a020a45b] Naveen Albert + + * manager: Remove documentation for nonexistent action. + + The manager XML documentation documents a "FilterList" + action, but there is no such action. Therefore, this can + lead to confusion when people try to use a documented + action that does not, in fact, exist. This is removed + as the action never did exist in the past, nor would it + be trivial to add since we only store the regex_t + objects, so the filter list can't actually be provided + without storing that separately. Most likely, the + documentation was originally added (around version 10) + in anticipation of something that never happened. + + ASTERISK-29917 #close + + Change-Id: I846b16fd6f80a91d4ddc5d8a861b522d7c6f8f97 + +2022-07-22 15:57 +0000 [c654486547] Naveen Albert + + * general: Improve logging levels of some log messages. + + Adjusts some logging levels to be more or less important, + that is more prominent when actual problems occur and less + prominent for less noteworthy things. + + ASTERISK-30153 #close + + Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b + +2022-07-27 13:34 +0000 [5feebc0857] Naveen Albert + + * cdr.conf: Remove obsolete app_mysql reference. + + The CDR sample config still mentions that app_mysql + is available in the addons directory, but this is + incorrect as it was removed as of 19. This removes + that to avoid confusion. + + ASTERISK-30160 #close + + Change-Id: Ie5293ccb4f2b365896981811b480544e67bb9cd7 + +2022-07-27 13:28 +0000 [165368bf0b] Naveen Albert + + * general: Remove obsolete SVN references. + + There are a handful of files in the tree that + reference an SVN link for the coding guidelines. + + This removes these because the links are dead + and the vast majority of source files do not + contain these links, so this is more consistent. + + app_skel still maintains an (up to date) link + to the coding guidelines. + + ASTERISK-30159 #close + + Change-Id: I35bbb20f66982e98099cff3029ede20091ffdac7 + +2022-07-23 18:14 +0000 [2d8f2696b2] Naveen Albert + + * app_confbridge: Add missing AMI documentation. + + Documents the ConfbridgeListRooms AMI response, + which is currently not documented. + + ASTERISK-30020 #close + + Change-Id: Id6fff7a936244bae7b52686301eb740c1169cdea + +2022-07-23 18:07 +0000 [4af881506e] Naveen Albert + + * app_meetme: Add missing AMI documentation. + + The MeetmeList and MeetmeListRooms AMI + responses are currently completely undocumented. + This adds documentation for these responses. + + ASTERISK-30018 #close + + Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8 + +2022-07-23 17:37 +0000 [83912496ab] Naveen Albert + + * func_srv: Document field parameter. + + Adds missing documentation for the field parameter + for the SRVRESULT function. + + ASTERISK-30151 + Reported by: Chris Young + + Change-Id: I4385a2e0892a07e30dea1a8a0588e2c1bea2b1f1 + +2022-07-23 17:17 +0000 [c771e2dd7a] Naveen Albert + + * pbx_functions.c: Manually update ast_str strlen. + + When ast_func_read2 is used to read a function using + its read function (as opposed to a native ast_str read2 + function), the result is copied directly by the function + into the ast_str buffer. As a result, the ast_str length + remains initialized to 0, which is a bug because this is + not the real string length. + + This can cascade and have issues elsewhere, such as when + reading substrings of functions that only register read + as opposed to read2 callbacks. In this case, since reading + ast_str_strlen returns 0, the returned substring is empty + as opposed to the actual substring. This has caused + the ast_str family of functions to behave inconsistently + and erroneously, in contrast to the pbx_variables substitution + functions which work correctly. + + This fixes this issue by manually updating the ast_str length + when the result is copied directly into the ast_str buffer. + + Additionally, an assertion and a unit test that previously + exposed these issues are added, now that the issue is fixed. + + ASTERISK-29966 #close + + Change-Id: I4e2dba41410f9d4dff61c995d2ca27718248e07f + +2022-02-18 15:59 +0000 [f645157a4b] Sergey V. Lobanov + + * build: fix bininstall launchd issue on cross-platform build + + configure script detects /sbin/launchd, but the result of this + check is not used in Makefile (bininstall). Makefile also detects + /sbin/launchd file to decide if it is required to install + safe_asterisk. + + configure script correctly detects cross compile build and sets + PBX_LAUNCHD=0 + + In case of building asterisk on MacOS host for Linux target using + external toolchain (e.g. OpenWrt toolchain), bininstall does not + install safe_asterisk (due to /sbin/launchd detection in Makefile), + but it is required on target (Linux). + + This patch adds HAVE_SBIN_LAUNCHD=@PBX_LAUNCHD@ to makeopts.in to + use the result of /sbin/launchd detection from configure script in + Makefile. + Also this patch uses HAVE_SBIN_LAUNCHD in Makefile (bininstall) to + decide if it is required to install safe_asterisk. + + ASTERISK-29905 #close + + Change-Id: Iff61217276cd188f43f51ef4cdbffe39d9f07f65 + +2022-07-11 06:32 +0000 [a9223f210e] Naveen Albert + + * db: Add AMI action to retrieve DB keys at prefix. + + Adds the DBGetTree action, which can be used to + retrieve all of the DB keys beginning with a + particular prefix, similar to the capability + provided by the database show CLI command. + + ASTERISK-30136 #close + + Change-Id: I3be9425e53be71f24303fdd4d2923c14e84337e6 + +2022-07-12 16:38 +0000 [ce18196280] Naveen Albert + + * manager: Fix incomplete filtering of AMI events. + + The global event filtering code was only in one + possible execution path, so not all events were + being properly filtered out if requested. This moves + that into the universal AMI handling code so all + events are properly handled. + + Additionally, the CLI listing of disabled events can + also get truncated, so we now print out everything. + + ASTERISK-30137 #close + + Change-Id: If8c42edcb2abc5158552da7eba2a8ff6b20e1959 + +2022-07-20 05:59 +0000 [f8000daff5] George Joseph + + * Update defaultbranch to 20 + + Change-Id: Ib91db9223a78188667e15053bcc73931b878414e + +2022-07-20 05:44 +0000 [a818b05ca1] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 20.0.0 +2022-06-14 04:12 +0000 [37c16f9eef] Michael Neuhauser + + * res_pjsip: delay contact pruning on Asterisk start + + Move the call to ast_sip_location_prune_boot_contacts() *after* the call + to ast_res_pjsip_init_options_handling() so that + res/res_pjsip/pjsip_options.c is informed about the contact deletion and + updates its sip_options_contact_statuses list. This allows for an AMI + event to be sent by res/res_pjsip/pjsip_options.c if the endpoint + registers again from the same remote address and port (i.e., same URI) + as used before the Asterisk restart. + + ASTERISK-30109 + Reported-by: Michael Neuhauser + + Change-Id: I1ba4478019e4931a7085f62708d9b66837e901a8 + +2022-03-28 20:35 +0000 [f2f397c1a8] Naveen Albert + + * chan_dahdi: Fix buggy and missing Caller ID parameters + + There are several things wrong with analog Caller ID + handling that are fixed by this commit: + + callerid.c's Caller ID generation function contains the + logic to use the presentation to properly send the proper + Caller ID. However, currently, DAHDI does not pass any + presentation information to the Caller ID module, which + means that presentation is completely ignored on all calls. + This means that lines could be getting Caller ID information + they aren't supposed to. + + Part of the reason this has been obscured is because the + simple switch logic for handling the built in *67 and *82 + is completely wrong. Rather than modifying the presentation + for the call accordingly (which is what it's supposed to do), + it simply blanks out the Caller ID or fills it in. This is + wrong, so wrong that it makes a mockery of the specification. + Additionally, it would leave to the "UNAVAILABLE" disposition + being used for Caller ID generation as opposed to the "PRIVATE" + disposition that it should have been using. This is now fixed + to only update the presentation and not modify the number and + name, so that the simple switch *67/*82 work correctly. + + Next, sig_analog currently only copies over the name and number, + nothing else, when it is filling in a duplicated caller id + structure. Thus, we also now copy over the presentation + information so that is available for the Caller ID spill. + Additionally, this meant that "valid" was implicitly 0, + and as such presentation would always fail to "Unavailable". + The validity is therefore also copied over so it can be used + by ast_party_id_presentation. + + As part of this fix, new API is added so that all the relevant + Caller ID information can be passed in to the Caller ID generation + functions. Parameters that are also completely missing from the + Caller ID spill have also been added, to enhance the compatibility, + correctness, and completeness of the Asterisk Caller ID implementation. + + ASTERISK-29991 #close + + Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15 + +2022-07-10 17:31 +0000 [be6a03f68c] Sam Banks + + * queues.conf.sample: Correction of typo + + ASTERISK-30126 #close + + Change-Id: I009c4dcbf9338a13e3baf87b52a5bbe4f9f81a42 + +2022-04-01 15:17 +0000 [8a21417095] Naveen Albert + + * chan_dahdi: Add POLARITY function. + + Adds a POLARITY function which can be used to + retrieve the current polarity of an FXS channel + as well as set the polarity of an FXS channel + to idle or reverse at any point during a call. + + ASTERISK-30000 #close + + Change-Id: If6f50998f723e4484bf68e2473f5cedfeaf9b8f1 + +2022-06-01 21:03 +0000 [7cc026b3fb] Mike Bradeen + + * Makefile: Avoid git-make user conflict + + make_version now silently checks if the required git commands will + fail. If they do, then return UNKNOWN__git_check_fail to + distinguish this failure from other UNKNOWN__ version failures + + Makefile checks for this value on install and exits out with + instructions + + ASTERISK-30029 + + Change-Id: If8f10cac8f509c08981120f17555762342020221 + +2022-06-18 07:17 +0000 [2843e5678d] Naveen Albert + + * app_confbridge: Always set minimum video update interval. + + Currently, if multiple video-enabled ConfBridges are + conferenced together, we immediately get into a scenario + where an infinite sequence of video updates fills up + the taskprocessor queue and causes memory consumption + to climb unabated until Asterisk is killed. This is due + to the core bridging mechanism that provides video updates + (softmix_bridge_write_control in bridge_softmix.c) + continously updating all the channels in the bridge with + video updates. + + The logic to do so in the core is that the video updates + should be provided if the video_update_discard property + for the bridge is 0, or if enough time has elapsed since + the last video update. Thus, we already have a safeguard + built in to ensure the scenario described above does not + happen. Currently, however, this safeguard is not being + adequately ensured. + + In app_confbridge, the video_update_discard property + defaults to 2000, which is a healthy value that should + completely prevent this issue. However, this value is + only set onto the bridge in the SFU video mode. This + leaves video modes such as follow_talker completely + vulnerable, since video_update_discard will actually + be 0, since the default or set value was never applied. + As a result, the core bridging mechanism will always + try to provide video updates regardless of when the last + one was sent. + + To prevent this issue from happening, we now always + set the video_update_discard property on the bridge + with the value from the bridge profile. The app_confbridge + defaults will thus ensure that infinite video updates + no longer happen in any video mode. + + ASTERISK-29907 #close + + Change-Id: I4accb2536ac62797950468e9930f12ef7dd486b2 + +2022-07-05 10:24 +0000 [d25bf55de5] Sean Bright + + * pbx.c: Simplify ast_context memory management. + + Allocate all of the ast_context's character data in the structure's + flexible array member and eliminate the clunky fake_context. This will + simplify future changes to ast_context. + + Change-Id: I98357de75d8ac2b3c4c9f201223632e6901021ea + +2022-07-13 13:38 +0000 [80d6f5eb20] George Joseph + + * geoloc_eprofile.c: Fix setting of loc_src in set_loc_src() + + line 196: loc_src = '\0'; + should have been + line 196: *loc_src = '\0'; + + The issue was caught by the gcc optimizer complaining that + loc_src had a zero length because the pointer itself was being + set to NULL instead of the _contents_ of the pointer being set + to the NULL terminator. + + ASTERISK-30138 + Reported-by: Sean Bright + + Change-Id: Id247be113cc8510f043ca053d5b4f5f3d32acd29 + +2022-07-07 10:32 +0000 [1fa568e76f] George Joseph + + * Geolocation: chan_pjsip Capability Preview + + This commit adds res_pjsip_geolocation which gives chan_pjsip + the ability to use the core geolocation capabilities. + + This commit message is intentionally short because this isn't + a simple capability. See the documentation at + https://wiki.asterisk.org/wiki/display/AST/Geolocation + for more information. + + THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON + USER FEEDBACK! + + ASTERISK-30128 + + Change-Id: Ie2e2bcd87243c2cfabc43eb823d4427c7086f4d9 + +2022-02-15 07:29 +0000 [639d72e98c] George Joseph + + * Geolocation: Core Capability Preview + + This commit adds res_geolocation which creates the core capabilities + to manipulate Geolocation information on SIP INVITEs. + + An upcoming commit will add res_pjsip_geolocation which will + allow the capabilities to be used with the pjsip channel driver. + + This commit message is intentionally short because this isn't + a simple capability. See the documentation at + https://wiki.asterisk.org/wiki/display/AST/Geolocation + for more information. + + THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON + USER FEEDBACK! + + ASTERISK-30127 + + Change-Id: Ibfde963121b1ecf57fd98ee7060c4f0808416303 + +2022-05-31 19:49 +0000 [bcc18ca9f5] Naveen Albert + + * general: Fix various typos. + + ASTERISK-30089 #close + + Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275 + +2022-06-17 12:15 +0000 [4cbe12d6d1] Morvai Szabolcs + + * cel_odbc & res_config_odbc: Add support for SQL_DATETIME field type + + See also: ASTERISK_30023 + + ASTERISK-30096 #close + patches: + inline on issue - submitted by Morvai Szabolcs + + Change-Id: I79c0b74862100acd9c8319dca5cc456a654d02eb + +2022-07-04 05:21 +0000 [5f60caa402] Naveen Albert + + * chan_iax2: Allow compiling without OpenSSL. + + ASTERISK_30007 accidentally made OpenSSL a + required depdendency. This adds an ifdef so + the relevant code is compiled only if OpenSSL + is available, since it only needs to be executed + if OpenSSL is available anyways. + + ASTERISK-30083 #close + + Change-Id: Iad05c1a9a8bd2a48e7edf8d234eaa9f80779e34d + +2022-06-28 06:59 +0000 [68bcf4c4c5] Joshua C. Colp + + * websocket / aeap: Handle poll() interruptions better. + + A sporadic test failure was happening when executing the AEAP + Websocket transport tests. It was originally thought this was + due to things not getting cleaned up fast enough, but upon further + investigation I determined the underlying cause was poll() + getting interrupted and this not being handled in all places. + + This change adds EINTR and EAGAIN handling to the Websocket + client connect code as well as the AEAP Websocket transport code. + If either occur then the code will just go back to waiting + for data. + + The originally disabled failure test case has also been + re-enabled. + + ASTERISK-30099 + + Change-Id: I1711a331ecf5d35cd542911dc6aaa9acf1e172ad + +2022-05-14 16:25 +0000 [f5680a7568] Naveen Albert + + * res_cliexec: Add dialplan exec CLI command. + + Adds a CLI command similar to "dialplan eval function" except for + applications: "dialplan exec application", useful for quickly + testing certain application behavior directly from the CLI + without writing any dialplan. + + ASTERISK-30062 #close + + Change-Id: I42e9fa9b60746c21450d40f99a026d48d2486dde + +2022-07-03 16:29 +0000 [938383aff3] Trevor Peirce + + * features: Update documentation for automon and automixmon + + The current documentation is out of date and does not reflect actual + behaviour. This change makes documentation clearer and accurately + reflect the purpose of relevant channel variables. + + ASTERISK-30123 + + Change-Id: I160d0b01fce862477ad55ac1aa708a730473eb6f + +2022-06-27 12:31 +0000 [5fe9887701] George Joseph + + * Geolocation: Base Asterisk Prereqs + + * Added ast_variable_list_from_quoted_string() + Parse a quoted string into an ast_variable list. + + * Added ast_str_substitute_variables_full2() + Perform variable/function/expression substitution on an ast_str. + + * Added ast_strsep_quoted() + Like ast_strsep except you can specify a specific quote character. + Also added unit test. + + * Added ast_xml_find_child_element() + Find a direct child element by name. + + * Added ast_xml_doc_dump_memory() + Dump the specified document to a buffer + + * ast_datastore_free() now checks for a NULL datastore + before attempting to destroy it. + + Change-Id: I5dcefed2f5f93a109e8b489e18d80d42e45244ec + +2022-06-24 06:09 +0000 [740c773781] Boris P. Korzun + + * pbx_lua: Remove compiler warnings + + Improved variable definitions (specified correct type) for avoiding + compiler warnings. + + ASTERISK-30117 #close + + Change-Id: I3b00c1befb658ee9379ddabd9a9132765ca9201a + +2022-04-08 05:34 +0000 [d52e2b0f1d] Jose Lopes + + * res_pjsip_header_funcs: Add functions PJSIP_RESPONSE_HEADER and PJSIP_RESPONSE_HEADERS + + These new functions allow retrieving information from headers on 200 OK + INVITE response. + + ASTERISK-29999 + + Change-Id: I264a610a9333359297a0825feb29a1bb4f4ad144 + +2022-06-09 02:10 +0000 [77f6c50814] Boris P. Korzun + + * res_prometheus: Optional load res_pjsip_outbound_registration.so + + Switched res_pjsip_outbound_registration.so dep to optional. Added + module loaded check before using it. + + ASTERISK-30101 #close + + Change-Id: Ia34f1684d984e821fbdd4de8911f930337703666 + +2022-04-30 11:44 +0000 [626fefdf7d] Naveen Albert + + * app_dial: Fix dial status regression. + + ASTERISK_28638 caused a regression by incorrectly aborting + early and overwriting the status on certain calls. + This was exhibited by certain technologies such as DAHDI, + where DAHDI returns NULL for the request if a line is busy. + This caused the BUSY condition to be incorrectly treated + as CHANUNAVAIL because the DIALSTATUS was getting incorrectly + overwritten and call handling was aborted early. + + This is fixed by instead checking if any valid peers have been + specified, as opposed to checking the list size of successful + requests. This is because the latter could be empty but this + does not indicate any kind of problem. This restores the + previous working behavior. + + ASTERISK-29989 #close + + Change-Id: I4d4b209b967816b1bc791534593ababa2b99bb88 + +2022-04-01 14:49 +0000 [350ffcb02b] Naveen Albert + + * db: Notify user if deleted DB entry didn't exist. + + Currently, if using the CLI to delete a DB entry, + "Database entry removed" is always returned, + regardless of whether or not the entry actually + existed in the first place. This meant that users + were never told if entries did not exist. + + The same issue occurs if trying to delete a DB key + using AMI. + + To address this, new API is added that is more stringent + in deleting values from AstDB, which will not return + success if the value did not exist in the first place, + and will print out specific error details if available. + + ASTERISK-30001 #close + + Change-Id: Ic84e3eddcd66c7a6ed7fea91cdfd402568378b18 + +2022-02-05 15:16 +0000 [b841845453] Naveen Albert + + * cli: Fix CLI blocking forever on terminating backslash + + A corner case exists in CLI parsing where if + a CLI user in a remote console ends with + a backslash and then invokes command completion + (using TAB or ?), then the console will freeze + forever until a SIGQUIT signal is sent to the + process, due to getting blocked forever + reading the command completion. CTRL+C + and other key combinations have no impact on + the CLI session. + + This occurs because, in such cases, the CLI + process is waiting for AST_CLI_COMPLETE_EOF + to appear in the buffer from the main process, + but instead the main process is confused by + the funny syntax and thus prints out the CLI help. + As a result, the CLI process is stuck on the + read call, waiting for the completion that + will never come. + + This prevents blocking forever by checking + if the data from the main process starts with + "Usage:". If it does, that means that CLI help + was sent instead of the tab complete vector, + and thus the CLI should bail out and not wait + any longer. + + ASTERISK-29822 #close + + Change-Id: I9810ac59304fec162da701653c9c834f0ec8f670 + +2022-06-18 12:13 +0000 [ae8a36a7d9] Naveen Albert + + * app_dial: Propagate outbound hook flashes. + + The Dial application currently stops hook flashes + dead in their tracks from propagating through on + outbound calls. This fixes that so they can go + down the wire. + + ASTERISK-30115 #close + + Change-Id: Id4e78b29a049f35c5b1e7520eaa10d0eb5b7f97c + +2022-06-20 16:00 +0000 [e5553fbd15] Naveen Albert + + * res_calendar_icalendar: Send user agent in request. + + Microsoft recently began rejecting all requests for + ICS calendars on Office 365 with 400 errors if + the request doesn't contain a user agent. See: + + https://docs.microsoft.com/en-us/answers/questions/883904/34the-remote-server-returned-an-error-400-bad-requ.html + + Accordingly, we now send a user agent on requests for + ICS files so that requests to Office 365 will work as + they did before. + + ASTERISK-30106 + + Change-Id: Ie9dcaef12ae8adf37533c684499eb11005fac8f7 + +2022-05-21 20:40 +0000 [0f0cc43e1b] Naveen Albert + + * say: Abort play loop if caller hangs up. + + If the caller has hung up, break out of the play loop so we don't try + to play remaining files and fail to do so. + + ASTERISK-30075 #close + + Change-Id: I55e85be28ee90b48c0fe4ce20ac136a7dbb49f14 + +2022-06-08 18:32 +0000 [a3b2daf127] Kevin Harwell + + * res_pjsip: allow TLS verification of wildcard cert-bearing servers + + Rightly the use of wildcards in certificates is disallowed in accordance + with RFC5922. However, RFC2818 does make some allowances with regards to + their use when using subject alt names with DNS name types. + + As such this patch creates a new setting for TLS transports called + 'allow_wildcard_certs', which when it and 'verify_server' are both enabled + allows DNS name types, as well as the common name that start with '*.' + to match as a wildcard. + + For instance: *.example.com + will match for: foo.example.com + + Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc... + And the starting wildcard only matches for a single level. + + For instance: *.example.com + will NOT match for: foo.bar.example.com + + The new setting is disabled by default. + + ASTERISK-30072 #close + + Change-Id: If0be3fdab2e09c2a66bb54824fca406ebaac3da4 + +2022-05-15 07:41 +0000 [4a11ae7ecf] Naveen Albert + + * pbx: Add helper function to execute applications. + + Finding an application and executing it if found is + a common task throughout Asterisk. This adds a helper + function around pbx_exec to do this, to eliminate + redundant code and make it easier for modules to + substitute variables and execute applications by name. + + ASTERISK-30061 #close + + Change-Id: Ifee4d2825df7545fb515d763d393065675140c84 + +2022-05-10 07:19 +0000 [d052418b94] Stanislav Abramenkov + + * pjsip: Upgrade bundled version to pjproject 2.12.1 + + More information: + https://github.com/pjsip/pjproject/releases/tag/2.12.1 + + Pull request to third-party + https://github.com/asterisk/third-party/pull/11 + + ASTERISK-30050 + + Change-Id: Icb4e86d4b85ef9b975355c91f3ed56a50b51c6bd + +2022-06-11 14:29 +0000 [2604a8352b] Naveen Albert + + * asterisk.c: Fix incompatibility warnings for remote console. + + A previous review fixing ASTERISK_22246 and ASTERISK_26582 + got a couple of the options mixed up as to whether or not + they are compatible with the remote console. This fixes + those to the best of my knowledge. + + ASTERISK-30097 #close + + Change-Id: Id54166991aa79f04fb02699cc499bedda854253b + +2022-06-07 16:03 +0000 [d9ce2a652b] Kevin Harwell + + * test_aeap_transport: disable part of failing unit test + + The 'transport_binary' test sporadically fails, but on a theory that the + problem is caused by a previously executed test, transport_connect_fail, + part of that test has been disabled until a solution is found. + + ASTERISK_30099 + + Change-Id: I48ed74d696aa9b6159f59661f3d535cac4c909e1 + +2022-05-13 07:33 +0000 [97f278a94a] Naveen Albert + + * sig_analog: Fix broken three-way conferencing. + + Three-way calling for analog lines is currently broken. + If party A is on a call with party B and initiates a + three-way call to party C, the behavior differs depending + on whether the call is conferenced prior to party C + answering. The post-answer case is correct. However, + if A flashes before C answers, then the next flash + disconnects B rather than C, which is incorrect. + + This error occurs because the subs are not swapped + in the misbehaving case. This is because the flash + handler only swaps the subs if C has answered already, + which is wrong. To fix this, we swap the subs regardless + of whether C has answered or not when the call is + conferenced. This ensures that C is disconnected + on the next hook flash, rather than B as can happen + currently. + + ASTERISK-30043 #close + + Change-Id: I96c5bf6c9b7eb2636136b716c677c82c079b6f06 + +2022-05-15 08:31 +0000 [cc8e098e1d] Naveen Albert + + * app_voicemail: Add option to prevent message deletion. + + Adds an option to VoiceMailMain that prevents the user + from deleting messages during that application invocation. + This can be useful for public or shared mailboxes, where + some users should be able to listen to messages but not + delete them. + + ASTERISK-30063 #close + + Change-Id: Icdfb8423ae8d1fce65a056b603eb84a672e80a26 + +2022-05-31 05:59 +0000 [ddc2cca659] Naveen Albert + + * res_parking: Add music on hold override option. + + An m option to Park and ParkAndAnnounce now allows + specifying a music on hold class override. + + ASTERISK-30087 + + Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e + +2022-05-31 20:43 +0000 [51d262af12] Naveen Albert + + * xmldocs: Improve examples. + + Use example tags instead of regular para tags + where possible. + + ASTERISK-30090 + + Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5 + +2022-03-12 12:27 +0000 [31dc28ab09] Naveen Albert + + * res_pjsip_outbound_registration: Make max random delay configurable. + + Currently, PJSIP will randomly wait up to 10 seconds for each + outbound registration's initial attempt. The reason for this + is to avoid having all outbound registrations attempt to register + simultaneously. + + This can create limitations with the test suite where we need to + be able to receive inbound calls potentially within 10 seconds of + starting up. For instance, we might register to another server + and then try to receive a call through the registration, but if + the registration hasn't happened yet, this will fail, and hence + this inconsistent behavior can cause tests to fail. Ultimately, + this requires a smaller random value because there may be no good + reason to wait for up to 10 seconds in these circumstances. + + To address this, a new config option is introduced which makes this + maximum delay configurable. This allows, for instance, this to be + set to a very small value in test systems to ensure that registrations + happen immediately without an unnecessary delay, and can be used more + generally to control how "tight" the initial outbound registrations + are. + + ASTERISK-29965 #close + + Change-Id: Iab989a8e94323e645f3a21cbb6082287c7b2f3fd + +2022-06-07 16:53 +0000 [5f0581c5f5] Trevor Peirce + + * res_pjsip: Actually enable session timers when timers=always + + When a pjsip endpoint is defined with timers=always, this has been a + functional noop. This patch correctly sets the feature bitmap to both + enable support for session timers and to enable them even when the + endpoint itself does not request or support timers. + + ASTERISK-29603 + Reported-By: Ray Crumrine + + Change-Id: I8b5eeaa9ec7f50cc6d96dd34c2b4aa9c53fb5440 + +2022-06-06 17:21 +0000 [044a08ae7b] Alexei Gradinari + + * res_pjsip_pubsub: delete scheduled notification on RLS update + + If there is scheduled notification, we must delete it + to avoid using destroyed subscriptions. + + ASTERISK-29906 + + Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f + +2022-06-07 08:48 +0000 [355c07e2e6] Alexei Gradinari + + * res_pjsip_pubsub: XML sanitized RLS display name + + ASTERISK-29891 + + Change-Id: Ic8c9697e616446e06e6302653eae902aa23372ad + +2022-06-01 14:59 +0000 [74df01009f] Christof Efkemann + + * app_sayunixtime: Use correct inflection for German time. + + In function ast_say_date_with_format_de(), take special + care when the hour is one o'clock. In this case, the + German number "eins" must be inflected to its neutrum form, + "ein". This is achieved by playing "digits/1N" instead of + "digits/1". Fixes both 12- and 24-hour formats. + + ASTERISK-30092 + + Change-Id: Ica9b80125c0b317e378d89c1ea786816e2635510 + +2022-05-16 08:01 +0000 [169e553320] Naveen Albert + + * chan_iax2: Prevent deadlock due to duplicate autoservice. + + If a switch is invoked using chan_iax2, deadlock can result + because the PBX core is autoservicing the channel while chan_iax2 + also then attempts to service it while waiting for the result + of the switch. This removes servicing of the channel to prevent + any conflicts. + + ASTERISK-30064 #close + + Change-Id: Ie92f206d32f9a36924af734ddde652b21106af22 + +2022-05-03 07:44 +0000 [3e8629454a] Naveen Albert + + * loader: Prevent deadlock using tab completion. + + If tab completion using ast_module_helper is attempted + during startup, deadlock will ensue because the CLI + will attempt to lock the module list while it is already + locked by the loader. This causes deadlock because when + the loader tries to acquire the CLI lock, they are blocked + on each other. + + Waiting for startup to complete is not feasible because + the CLI lock is acquired while waiting, so deadlock will + ensure regardless of whether or not a lock on the module + list is attempted. + + To prevent deadlock, we immediately abort if tab completion + is attempted on the module list before Asterisk is fully + booted. + + ASTERISK-30039 #close + + Change-Id: Idd468906c512bb196631e366a8f597a0e2e9271d + +2022-03-23 06:05 +0000 [64a764c33e] Naveen Albert + + * res_calendar: Prevent assertion if event ends in past. + + res_calendar will trigger an assertion currently + if the ending time is calculated to be in the past. + Unlike the reminder and start times, however, there + is currently no check to catch non-positive times + and set them to 1. As a result, if we get a negative + value by happenstance, this can cause a crash. + + To prevent the assertion from begin triggered, we now + use the same logic as the reminder and start events + to catch this issue before it can cause a problem. + + ASTERISK-29981 #close + + Change-Id: Idfb3204d195f350d2575fb4bc72a54a597d6e93c + +2022-05-30 15:55 +0000 [bae8092826] Naveen Albert + + * res_parking: Warn if out of bounds parking spot requested. + + Emits a warning if the user has requested a parking spot that + is out of bounds for the requested parking lot. + + ASTERISK-30086 + + Change-Id: I1080371e4f63e94724455003753014fbd3f95fbf + +2022-05-19 09:23 +0000 [a03b53bb7b] Maximilian Fridrich + + * chan_pjsip: Only set default audio stream on hold. + + When a PJSIP channel is set on hold or off hold, all streams were set + on/off hold. This is not the desired behaviour and caused issues + when there were multiple streams in the topology. + + Now, only the default audio stream is set on/off hold when a hold is + indicated. + + ASTERISK-30051 + + Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71 + +2022-05-26 16:29 +0000 [42b191ad64] Alexei Gradinari + + * res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI + + The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local" + Identity Element URI and Target Element URI to the same value - + the channel Caller Number. + For Identity Element it's ok to set as Caller ID. + But Local Target URI should be set as local URI. + + In this case the Local Target URI can be used for Directed Call Pickup + by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup). + + Also XML sanitized Display names. + + ASTERISK-24601 + + Change-Id: If130a2f2f3b2339b14dca0ec0ebeea3a87b34343 + +2022-05-11 14:48 +0000 [7dcea19ce8] Shloime Rosenblum + + * res_agi: Evaluate dialplan functions and variables in agi exec if enabled + + Agi commnad exec can now evaluate dialplan functions and + variables if variable AGIEXECFULL is set to yes. this can + be useful when executing Playback or Read from agi. + + ASTERISK-30058 #close + + Change-Id: I669991f540496e7bddd096fec82b52c083036832 + +2022-05-17 12:01 +0000 [a6c7524e0d] Sean Bright + + * ast_pkgconfig.m4: AST_PKG_CONFIG_CHECK() relies on sed. + + Make sure that we have a working sed before trying to use it. + + ASTERISK-30059 #close + + Change-Id: I9abad67a5df11b665d480feec304ab9d6f55cc76 + +2022-04-25 17:40 +0000 [4bf2473ac4] Moritz Fain + + * ari: expose channel driver's unique id to ARI channel resource + + This change exposes the channel driver's unique id (i.e. the Call-ID + for chan_sip/chan_pjsip based channels) to ARI channel resources + as `protocol_id`. + + ASTERISK-30027 + Reported by: Moritz Fain + Tested by: Moritz Fain + + Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87 + +2022-05-17 09:18 +0000 [8d7819482c] Sean Bright + + * loader.c: Use portable printf conversion specifier for int64. + + ASTERISK-30060 #close + + Change-Id: I88d47a1488be2f39017b8d562f993f081844fcb8 + +2022-05-17 07:18 +0000 [63ff0ccadf] Joshua C. Colp + + * res_pjsip_transport_websocket: Also set the remote name. + + As part of PJSIP 2.11 a behavior change was done to require + a matching remote hostname on an established transport for + secure transports. Since the Websocket transport is considered + a secure transport this caused the existing connection to not + be found and used. + + We now set the remote hostname and the transport can be found. + + ASTERISK-30065 + + Change-Id: Ia1cdef33e1411f927985b4b852c95e163c080e94 + +2022-05-04 10:37 +0000 [4848d6eeb9] Thomas Guebels + + * res_pjsip_transport_websocket: save the original contact host + + This is needed to be able to restore it in REGISTER responses, + otherwise the client won't be able to find the contact it created. + + ASTERISK-30042 + + Change-Id: I0c5823918199acf09246b3b206fbde66773688f6 + +2022-01-07 10:25 +0000 [604785f931] Naveen Albert + + * res_pjsip_outbound_registration: Show time until expiration + + Adjusts the pjsip show registration(s) commands to show + the amount of seconds remaining until a registration + expires. + + ASTERISK-29845 #close + + Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483 + +2022-04-29 11:42 +0000 [432a1d2d7e] Naveen Albert + + * app_confbridge: Add function to retrieve channels. + + Adds the CONFBRIDGE_CHANNELS function which can be used + to retrieve a comma-separated list of channels, filtered + by a particular type of participant category. This output + can then be used with functions like UNSHIFT, SHIFT, POP, + etc. + + ASTERISK-30036 #close + + Change-Id: I1950aff932437476dc1abab6f47fb4ac90520b83 + +2022-04-26 14:00 +0000 [a24979a2d7] Naveen Albert + + * chan_dahdi: Fix broken operator mode clearing. + + Currently, the operator services mode in DAHDI is broken and unusable. + The actual operator recall functionality works properly; however, + when the operator hangs up (which is the only way that such a call + is allowed to end), both lines are permanently taken out of service + until "dahdi restart" is run. This prevents this feature from being + used. + + Operator mode is one of the few factors that can cause the general + analog event handling in sig_analog not to be used. Several years + back, much of the analog handling was moved from chan_dahdi to + sig_analog. However, this was not done fully or consistently at + the time, and when operator mode is active, sig_analog does not + get used. Generally this is correct, but in the case of hangup + it should be using sig_analog regardless of the operator mode; + otherwise, the lines do not properly clear and they become unusable. + + This bug is fixed so the operator can now hang up and properly + release the call. It is treated just like any other hangup. The + operator mode functionality continues to work as it did before. + + ASTERISK-29993 #close + + Change-Id: Ib2e3ddb40d9c71e8801e0b4bb0a12e2b52f51d24 + +2022-05-03 07:57 +0000 [4aa541683b] George Joseph + + * GCC12: Fixes for 16+ + + Most issues were in stringfields and had to do with comparing + a pointer to an constant/interned string with NULL. Since the + string was a constant, a pointer to it could never be NULL so + the comparison was always "true". gcc now complains about that. + + There were also a few issues where determining if there was + enough space for a memcpy or s(n)printf which were fixed + by defining some of the involved variables as "volatile". + + There were also a few other miscellaneous fixes. + + ASTERISK-30044 + + Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570 + +2022-05-04 13:00 +0000 [49108810d1] George Joseph + + * GCC12: Fixes for 18+. state_id_by_topic comparing wrong value + + GCC 12 caught an issue in state_id_by_topic where we were + checking a pointer for NULL instead of the contents of + the pointer for '\0'. + + ASTERISK-30044 + + Change-Id: Ia0b04d4fff45c92acb7f07132a33622fa341148e + +2022-04-29 03:47 +0000 [8fdc6008a4] Maximilian Fridrich + + * core_unreal: Flip stream direction of second channel. + + When a new unreal (local) channel is created, a second (;2) channel is + created as a counterpart which clones the topology of the first + channel. This creates issues when an outgoing stream is sendonly or + recvonly as the stream state of the inbound channel will be the same + as the stream state of the outbound channel. + + Now the stream state is flipped for the streams of the 2nd channel in + ast_unreal_new_channels if the outgoing stream topology is recvonly or + sendonly. + + ASTERISK-29655 + Reported by: Michael Auracher + + ASTERISK-29638 + Reported by: Michael Auracher + + Change-Id: I0cea29635bb20b7bf7fd0fb95498cd44dab98fbf + +2022-03-27 07:33 +0000 [892c06564f] Naveen Albert + + * chan_dahdi: Document dial resource options. + + Documents the Dial syntax for DAHDI, namely the channel group, + distinctive ring, answer confirmation, and digital call options + that are specified in the resource itself. + + ASTERISK-24827 #close + + Change-Id: Ib95e78497fb00dc5cbfde1c93a69f034bfd08c30 + +2022-03-29 18:47 +0000 [0a8b3d3467] Naveen Albert + + * chan_dahdi: Don't allow MWI FSK if channel not idle. + + For lines that have mailboxes configured on them, with + FSK MWI, DAHDI will periodically try to dispatch FSK + to update MWI. However, this is never supposed to be + done when a channel is not idle. + + There is currently an edge case where MWI FSK can + extraneously get spooled for the channel if a caller + hook flashes and hangs up, which triggers a recall ring. + After one ring, the on hook time threshold in this if + condition has been satisfied and an MWI update is spooled. + This means that when the phone is picked up again, the + answerer gets an FSK spill before being reconnected to + the party on hold. + + To prevent this, we now explicitly check to ensure that + subchannel 0 has no owner. There is no owner when DAHDI + channels are idle, but if the channel is "in use" in some + way (such as in the aforementioned scenario), then there + is an owner, and we shouldn't process MWI at this time. + + ASTERISK-28518 #close + + Change-Id: Ia3904434fd81688d71742f7e84358b7e1c38e92a + +2022-02-23 10:29 +0000 [a2679b0ee2] Michael Cargile + + * apps/confbridge: Added hear_own_join_sound option to control who hears sound_join + + Added the hear_own_join_sound option to the confbridge user profile to + control who hears the sound_join audio file. When set to 'yes' the user + entering the conference and the participants already in the conference + will hear the sound_join audio file. When set to 'no' the user entering + the conference will not hear the sound_join audio file, but the + participants already in the conference will hear the sound_join audio + file. + + ASTERISK-29931 + Added by Michael Cargile + + Change-Id: I856bd66dc0dfa057323860a6418c1371d249abd2 + +2022-03-27 06:23 +0000 [19c841950b] Naveen Albert + + * chan_dahdi: Don't append cadences on dahdi restart. + + Currently, if any custom ring cadences are specified, they are + appended to the array of cadences from wherever we left off + last time. This works properly the first time, but on subsequent + dahdi restarts, it means that the existing cadences are left + alone and (most likely) the same cadences are then re-added + afterwards. In short order, the cadence array gets maxed out + and the user begins seeing warnings that the array is full + and no more cadences may be added. + + This buggy behavior persists until Asterisk is completely + restarted; however, if and when dahdi restart is run again, + then the same problem is reintroduced. + + This fixes this behavior so that cadence parsing is more + idempotent, that is so running dahdi restart multiple times + starts adding cadences from the beginning, rather than from + wherever the last cadence was added. + + As before, it is still not possible to revert to the default + cadences by simply removing all cadences in this manner, nor + is it possible to delete existing cadences. However, this + does make it possible to update existing cadences, which + was not possible before, and also ensures that the cadences + remain unchanged if the config remains unchanged. + + ASTERISK-29990 #close + + Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d + +2022-04-02 16:22 +0000 [fbe960ca42] Naveen Albert + + * chan_iax2: Prevent crash if dialing RSA-only call without outkey. + + Currently, if attempting to place a call to a peer that only allows + RSA authentication, if we fail to provide an outkey when placing + the call, Asterisk will crash. + + This exposes the broader issue that IAX2 is prone to causing a crash + if encryption or decryption is attempted but we never initialized + the encryption and decryption keys. In other words, if the logic + to use encryption in chan_iax2 is not perfectly aligned with the + decision to build keys in the first place, then a crash is not + only possible but probable. This was demonstrated by ASTERISK_29264, + for instance. + + This permanently prevents such events from causing a crash by explicitly + checking that keys are initialized properly before setting the flags + to use encryption for the call. Instead of crashing, the call will + now abort. + + ASTERISK-30007 #close + + Change-Id: If925c3d86099ceac7f621804f2532baac5050c9a + +2022-02-05 09:03 +0000 [fe6f7dcb13] Naveen Albert + + * menuselect: Don't erroneously recompile modules. + + A bug in menuselect can cause modules that are disabled + by default to be recompiled every time a recompilation + occurs. This occurs for module categories that are NOT + positive output, as for these categories, the modules + contained in the makeopts file indicate modules which + should NOT be selected. The existing procedure of iterating + through these modules to mark modules as present is thus + insufficient. This has led to modules with a default_enabled + tag of "no" to get deleted and recompiled every time, even + when they haven't changed. + + To fix this, we now modify the mark as present behavior + for module categories that are not positive output. For + these, we start by iterating through the module tree + and marking all modules as present, then go back and + mark anything contained in the makeopts file as not + present. This ensures that makeopt selections are actually + used properly, regardless of whether a module category + uses positive output or not. + + ASTERISK-29728 #close + + Change-Id: Idf2974c4ed8d0ba3738a92f08a6082b234277b95 + +2022-03-31 10:44 +0000 [b90650d8f4] Naveen Albert + + * app_meetme: Don't erroneously set global variables. + + The admin_exec function in app_meetme is used by the SLA + applications for internal bridging. However, in these cases, + chan is NULL. Currently, this function will set some status + variables that are intended for a channel, but since channel + is NULL, this is erroneously creating meaningless global + variables, which shouldn't be happening. This sets these + variables only if chan is not NULL. + + ASTERISK-30002 #close + + Change-Id: I817df6c26f5bda131678e56791b0b61ba64fc6f7 + +2022-03-05 05:43 +0000 [4585a9c3b8] Naveen Albert + + * asterisk.c: Warn of incompatibilities with remote console. + + Some command line options to Asterisk only apply when Asterisk + is started and cannot be used with remote console mode. If a + user tries to use any of these, they are currently simply + silently ignored. + + This prints out a warning if incompatible options are used, + informing users that an option used cannot be used with remote + console mode. Additionally, some clarifications are added to + the help text and man page. + + ASTERISK-22246 + ASTERISK-26582 + + Change-Id: I980a5380ef2c19e8ea348596396d5382893c4337 + +2022-03-14 20:41 +0000 [306ce09df2] Naveen Albert + + * func_db: Add function to return cardinality at prefix + + Adds the DB_KEYCOUNT function, which can be used to retrieve + the number of keys at a given prefix in AstDB. + + ASTERISK-29968 #close + + Change-Id: Ib2393b77b7e962dbaae6192f8576bc3f6ba92d09 + +2022-03-29 19:22 +0000 [fe50f049c4] Naveen Albert + + * chan_dahdi: Fix insufficient array size for round robin. + + According to chan_dahdi.conf, up to 64 groups (numbered + 0 through 63) can be used when dialing DAHDI channels. + + However, currently dialing round robin with a group number + greater than 31 fails because the array for the round robin + structure is only size 32, instead of 64 as it should be. + + This fixes that so the round robin array size is consistent + with the actual groups capacity. + + ASTERISK-29994 + + Change-Id: I4caa08d7025f78ac75a0539f71aaf3eb3e85b3b7 + +2022-02-26 03:07 +0000 [a3abc868db] Mark Petersen + + * chan_sip.c Session timers get removed on UPDATE + + If Asterisk receives a SIP REFER with Session-Timers UAC + maintain Session-Timers when sending UPDATE" + + ASTERISK-29843 + + Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c + +2021-06-21 07:49 +0000 [6ddb0ec939] Naveen Albert + + * func_evalexten: Extension evaluation function. + + This adds the EVAL_EXTEN function, which may be used to retrieve + the variable-substituted data at any extension. + + ASTERISK-29486 + + Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432 + +2022-02-28 19:29 +0000 [ce7846e658] Naveen Albert + + * file.c: Prevent formats from seeking negative offsets. + + Currently, if a user uses an application like ControlPlayback + to try to rewind a file past the beginning, this can throw + warnings when the file format (e.g. PCM) tries to seek to + a negative offset. + + Instead of letting file formats try (and fail) to seek a + negative offset, we instead now catch this in the rewind + function to ensure that we never seek an offset less than 0. + This prevents legitimate user actions from triggering warnings + from any particular file formats. + + ASTERISK-29943 #close + + Change-Id: Ia53f2623f57898f4b8e5c894b968b01e95426967 + +2022-02-26 06:37 +0000 [193b7a81fe] Naveen Albert + + * chan_pjsip: Add ability to send flash events. + + PJSIP currently is capable of receiving flash events + and converting them to FLASH control frames, but it + currently lacks support for doing the reverse: taking + a FLASH control frame and converting it into a flash + event in the SIP domain. + + This adds the ability for PJSIP to process flash control + frames by converting them into the appropriate SIP INFO + message, which can then be sent to the peer. This allows, + for example, flash events to be sent between Asterisk + systems using PJSIP. + + ASTERISK-29941 #close + + Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd + +2021-12-26 15:39 +0000 [92d408f293] Naveen Albert + + * cli: Add command to evaluate dialplan functions. + + Adds the dialplan eval function commands to evaluate a dialplan + function from the CLI. The return value and function result are + printed out and can be used for testing or debugging. + + ASTERISK-29820 #close + + Change-Id: I833e97ea54c49336aca145330a2adeebfad05209 + +2022-02-25 14:58 +0000 [0c70d497bc] Naveen Albert + + * documentation: Adds versioning information. + + Adds version information for applications, functions, + and manager events/actions. + + This is not completely exhaustive by any means but + covers most new things added that have release + versioning information in the issue tracker. + + ASTERISK-29940 #close + + Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131 + +2022-04-02 17:38 +0000 [bce722e60d] Naveen Albert + + * samples: Remove obsolete sample configs. + + Removes a couple sample config files for modules + which have since been removed from Asterisk. + + ASTERISK-30008 #close + + Change-Id: I6be89cafc6c575d98a5315e4912b61a833aacf52 + +2022-02-21 07:23 +0000 [1cdaeb8161] Mark Petersen + + * chan_pjsip: add allow_sending_180_after_183 option + + added new global config option "allow_sending_180_after_183" + that if enabled will preserve 180 after a 183 + + ASTERISK-29842 + + Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18 + +2022-03-07 08:11 +0000 [eab489b22e] Mark Petersen + + * chan_sip: SIP route header is missing on UPDATE + + if Asterisk need to send an UPDATE before answer + on a channel that uses Record-Route: + it will not include a Route header + + ASTERISK-29955 + + Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef + +2022-04-25 18:39 +0000 [f6062b17cc] Joshua C. Colp + + * manager: Terminate session on write error. + + On a write error to an AMI session a flag was set to + indicate that the write error had occurred, with the + expected result being that the session be terminated. + This was not actually happening and instead writing + would continue to be attempted. + + This change adds a check for the write error and causes + the session to actually terminate. + + ASTERISK-29948 + + Change-Id: Icaf5d413d4c0d5dc78292a17287fecc8720a31a5 + +2022-04-21 09:10 +0000 [e9355e66d1] Yury Kirsanov + + * bridge_simple.c: Unhold channels on join simple bridge. + + Patch provided inline by Yury Kirsanov on the linked issue and + approved by Josh Colp. + + ASTERISK-29253 #close + + Change-Id: I5b9ccc67ebf06e875ed061d9e7fc21f47b0a4e1f + +2021-06-18 12:54 +0000 [272bac70dd] Kevin Harwell + + * res_aeap & res_speech_aeap: Add Asterisk External Application Protocol + + Add framework to connect to, and read and write protocol based + messages from and to an external application using an Asterisk + External Application Protocol (AEAP). This has been divided into + several abstractions: + + 1. transport - base communication layer (currently websocket only) + 2. message - AEAP description and data (currently JSON only) + 3. transaction - links/binds requests and responses + 4. aeap - transport, message, and transaction handler/manager + + This patch also adds an AEAP implementation for speech to text. + Existing speech API callbacks for speech to text have been completed + making it possible for Asterisk to connect to a configured external + translator service and provide audio for STT. Results can also be + received from the external translator, and made available as speech + results in Asterisk. + + Unit tests have also been created that test the AEAP framework, and + also the speech to text implementation. + + ASTERISK-29726 #close + + Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2 + +2022-04-13 08:12 +0000 [53a3af6321] Maximilian Fridrich + + * app_dial: Flip stream direction of outgoing channel. + + When executing dial, the topology of the incoming channel is cloned and + used for the outgoing channel. This creates issues when an incoming + stream is sendonly or recvonly as the stream state of the outgoing + channel will be the same as the stream state of the incoming channel. + + Now the stream state is flipped for the outgoing stream in + dial_exec_full if the incoming stream topology is recvonly or sendonly. + + ASTERISK-29655 + Reported by: Michael Auracher + + ASTERISK-29638 + Reported by: Michael Auracher + + Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1 + +2022-04-21 10:26 +0000 [f593b1e93b] Ben Ford + + * res_pjsip_stir_shaken.c: Fix enabled when not configured. + + There was an issue with the conditional where STIR/SHAKEN would be + enabled even when not configured. It has been changed to ensure that if + a profile does not exist and stir_shaken is not set in pjsip.conf, then + the conditional will return from the function without performing + STIR/SHAKEN operations. + + ASTERISK-30024 + + Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6 + +2022-04-06 05:23 +0000 [fdc1c750f3] Joshua C. Colp + + * res_pjsip: Always set async_operations to 1. + + The async_operations setting on a transport configures how + many simultaneous incoming packets the transport can handle + when multiple threads are polling and waiting on the transport. + As we only use a single thread this was needlessly creating + incoming packets when set to a non-default value, wasting memory. + + ASTERISK-30006 + + Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f + +2022-04-19 10:36 +0000 [b1e0527bbd] Sean Bright + + * config.h: Don't use C++ keywords as argument names. + + ASTERISK-30021 #close + + Change-Id: I70eb59b782a4946b979942e21422746b7563029c + +2022-04-20 07:40 +0000 [283b09cf70] Joshua C. Colp + + * cdr_adaptive_odbc: Add support for SQL_DATETIME field type. + + ASTERISK-30023 + + Change-Id: I0e1697f6af044e9eab7e07bbaeeffd1bb68ac34a + +2022-04-11 04:30 +0000 [b3f39be0cc] Joshua C. Colp + + * pjsip: Increase maximum number of format attributes. + + Chrome has added more attributes, causing the limit to be + exceeded. This raises it up some more. + + ASTERISK-30015 + + Change-Id: I964957c005c4e6f7871b15ea1ccd9b4659c7ef32 + +2022-02-28 11:19 +0000 [0724b767a3] Ben Ford + + * AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header. + + Adds a new configuration option, stir_shaken_profile, in pjsip.conf that + can be specified on a per endpoint basis. This option will reference a + stir_shaken_profile that can be configured in stir_shaken.conf. The type + of this option must be 'profile'. The stir_shaken option can be + specified on this object with the same values as before (attest, verify, + on), but it cannot be off since having the profile itself implies wanting + STIR/SHAKEN support. You can also specify an ACL from acl.conf (along + with permit and deny lines in the object itself) that will be used to + limit what interfaces Asterisk will attempt to retrieve information from + when reading the Identity header. + + ASTERISK-29476 + + Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406 + +2022-01-07 08:50 +0000 [8f3dd86b8d] Ben Ford + + * AST-2022-001 - res_stir_shaken/curl: Limit file size and check start. + + Put checks in place to limit how much we will actually download, as well + as a check for the data we receive at the start to ensure it begins with + what we would expect a certificate to begin with. + + ASTERISK-29872 + + Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46 + +2022-02-10 06:02 +0000 [4aedaaadeb] Joshua C. Colp + + * func_odbc: Add SQL_ESC_BACKSLASHES dialplan function. + + Some databases depending on their configuration using backslashes + for escaping. When combined with the use of ' this can result in + a broken func_odbc query. + + This change adds a SQL_ESC_BACKSLASHES dialplan function which can + be used to escape the backslashes. + + This is done as a dialplan function instead of being always done + as some databases do not require this, and always doing it would + result in incorrect data being put into the database. + + ASTERISK-29838 + + Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d + +2022-03-04 19:41 +0000 [b87c5f5124] Naveen Albert + + * app_mf, app_sf: Return -1 if channel hangs up. + + The ReceiveMF and ReceiveSF applications currently always + return 0, even if a channel has hung up. The call will still + end but generally applications are expected to return -1 if + the channel has hung up. + + We now return -1 if a hangup occured to bring this behavior + in line with this norm. This has no functional impact, but + merely increases conformity with how these modules interact + with the PBX core. + + ASTERISK-29951 #close + + Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033 + +2022-01-22 09:53 +0000 [ede4e2099f] Naveen Albert + + * app_queue: Add music on hold option to Queue. + + Adds the m option to the Queue application, which allows a + music on hold class to be specified at runtime which will + override the class configured in queues.conf. + + This option functions like the m option to Dial. + + ASTERISK-29876 #close + + Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7 + +2022-03-05 09:40 +0000 [da44b848f5] Naveen Albert + + * app_meetme: Emit warning if conference not found. + + Currently, if a user tries to access a non-dynamic + MeetMe conference and the conference is not found, + the call simply silent hangs up. There is no indication + to the user that anything went wrong at all. + + This changes the relevant debug message to a warning + so that the user is notified of this invalidity. + + ASTERISK-29954 #close + + Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048 + +2022-02-24 13:33 +0000 [94df607771] Naveen Albert + + * build: Remove obsolete leftover build references. + + Removes some leftover build and config references to + modules that have since been removed from Asterisk. + + ASTERISK-29935 #close + + Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af + +2022-03-23 17:45 +0000 [0e31df6c93] Kevin Harwell + + * res_pjsip_header_funcs: wrong pool used tdata headers + + When adding headers to an outgoing request the headers were cloned using + the dialog's pool when they should have been cloned using tdata's pool. + Under certain circumstances it was possible for the dialog object, and + its pool to be freed while tdata is still active and available. Thus the + cloned header "disappeared", and when tdata tried to later access it a + crash would occur. + + This patch makes it so all added headers are cloned appropriately using + tdata's pool. + + ASTERISK-29411 #close + ASTERISK-29535 #close + + Change-Id: I9852025b5ee93ce1c038209150ee9dba1e0767c5 + +2022-03-25 10:46 +0000 [30cefc97a6] Kevin Harwell + + * deprecation cleanup: remove leftover files + + Several modules removal and deprecations occurred in 19.0.0 (initial + 19 release), but associated UPGRADE files were not removed from + staging for some reason in the master branch. + + This patch removes those files, and also removes a spurious leftover + header, chan_phone.h (associated module removed in 19). + + Change-Id: Ib92142c846b45c882d6b2b6caca7225253c83add + +2022-02-24 11:48 +0000 [fa0078fbe4] Joshua C. Colp + + * pjproject: Update bundled to 2.12 release. + + This change removes patches which have been merged into + upstream and updates some existing ones. It also adds + some additional config_site.h changes to restore previous + behavior, as well as a patch to allow multiple Authorization + headers. There seems to be some confusion or disagreement + on language in RFC 8760 in regards to whether multiple + Authorization headers are supported. The RFC implies it + is allowed, as does some past sipcore discussion. There is + also the catch all of "local policy" to allow it. In + the case of Asterisk we allow it. + + ASTERISK-29351 + + Change-Id: Id39ece02dedb7b9f739e0e37ea47d76854af7191 + +2022-03-05 10:26 +0000 [a7cf3979ec] Naveen Albert + + * pbx.c: Warn if there are too many includes in a context. + + The PBX core uses the stack when it comes to includes, which + means that a context can only contain strictly fewer than + AST_PBX_MAX_STACK includes. If this is exceeded, then warnings + will be emitted for each number of includes beyond this if + searching for an extension in the including context, and if + the extension's inclusion is beyond the stack size, it will + simply not be found. + + To address this, we now check if there are too many includes + in a context when the dialplan is reloaded so that if there + is an issue, the user is aware of at "compile time" as opposed + to "run time" only. Secondly, more details are printed out + when this message is encountered so it's clear what has happened. + + ASTERISK-26719 + + Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958 + +2022-03-25 14:00 +0000 [3e97156fd3] George Joseph + + * Makefile: Disable XML doc validation + + make_xml_documentation was being called with the --validate + flag set when it shouldn't have been. This was causing + build failures if neither xmllint nor xmlstarlet were installed. + The correct behavior is to simply print a message that either + one of those tools should be installed for validation and + continue with the build. + + ASTERISK-29988 + + Change-Id: Idc6c44114e7dd3fadae183a4e22f4fdba0b8a645 + +2022-03-25 09:33 +0000 [144b3c5453] George Joseph + + * make_xml_documentation: Remove usage of get_sourceable_makeopts + + get_sourceable_makeopts wasn't handling variables with embedded + double quotes in them very well. One example was the DOWNLOAD + variable when curl was being used instead of wget. Rather than + trying to fix get_sourceable_makeopts, it's just been removed. + + ASTERISK-29986 + Reported by: Stefan Ruijsenaars + + Change-Id: Idf2a90902228c2558daa5be7a4f8327556099cd2 + +2022-02-04 18:36 +0000 [0d11938e92] Birger Harzenetter (license 5870) + + * chan_iax2: Fix spacing in netstats command + + The iax2 show netstats command previously didn't contain + enough spacing in the header to properly align the table + header with the table body. This caused column headers + to not align with the values on longer channel names. + + Some spacing is added to account for the longest channel + names that display (before truncation occurs) so that + columns are always properly aligned. + + ASTERISK-29895 #close + patches: + 61205_misaligned2.patch submitted by Birger Harzenetter (license 5870) + + Change-Id: I450ce6bb81157b9d6d149007e53b749f237b6d9f + +2022-03-25 08:19 +0000 [5ac5c2b0ab] Sean Bright + + * openssl: Supress deprecation warnings from OpenSSL 3.0 + + There is work going on to update our OpenSSL usage to avoid the + deprecated functions but in the meantime make it possible to compile + in devmode. + + Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726 + +2022-03-23 16:04 +0000 [9b654d4e98] Marcel Wagner + + * documentation: Add information on running install_prereq script in readme + + Adding information in the readme about running the install_preqreq script to install components that the ./configure script might indicate as missing. + + ASTERISK-29976 #close + + Change-Id: Ic287b46300168729838bddd8f9265e98fc22bce6 + +2022-03-13 12:46 +0000 [7bc8ef2681] Naveen Albert + + * chan_iax2: Fix perceived showing host address. + + ASTERISK_22025 introduced a regression that shows + the host IP and port as the perceived IP and port + again, as opposed to showing the actual perceived + address. This fixes this by showing the correct + information. + + ASTERISK-29048 #close + + Change-Id: I0ad3e25bc6b449e83ce72ea5d1a1cdba72aa304a + +2022-02-22 14:51 +0000 [6624e34580] Boris P. Korzun + + * res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity + + Change RTP timer behavior for detecting RTP only after two-way + SDP channel establishment. Ignore detecting after receiving 183 + with SDP or while direct media is used. + Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout + and rtpholdtimeout options in chan_sip. + + ASTERISK-26689 #close + ASTERISK-29929 #close + + Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb + +2022-03-15 19:06 +0000 [64f11e0d18] Hugh McMaster + + * configure.ac: Use pkg-config to detect libxml2 + + Use pkg-config to detect libxml2, falling back to xml2-config if the + former is not available. + + This patch ensures Asterisk continues to build on systems without + xml2-config installed. + + The patch also updates the associated 'configure' files. + + ASTERISK-29970 #close + + Change-Id: I3c90dfe0b0590486cbb8e6d426a7c5c4199410c0 + +2022-02-13 13:06 +0000 [287a1a9126] Philip Prindeville + + * time: add support for time64 libcs + + Treat time_t's as entirely unique and use the POSIX API's for + converting to/from strings. + + Lastly, a 64-bit integer formats as 20 digits at most in base10. + Don't need to have any 100 byte buffers to hold that. + + ASTERISK-29674 #close + + Signed-off-by: Philip Prindeville + Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56 + +2022-03-15 12:24 +0000 [d1900d4a4c] Alexei Gradinari + + * res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request + + When asterisk generates the RLMI part of NOTIFY request, + the asterisk uses the local contact uri instead of the URI to which + the SUBSCRIBE request is sent. + Because of this mismatch some IP phones (for example Cisco 5XX) ignore + this list. + + According + https://datatracker.ietf.org/doc/html/rfc4662#section-5.2 + The first mandatory attribute is "uri", which contains the uri + that corresponds to the list. Typically, this is the URI to which + the SUBSCRIBE request was sent. + https://datatracker.ietf.org/doc/html/rfc4662#section-5.3 + The "uri" attribute identifies the resource to which the + element corresponds. Typically, this will be a SIP URI that, if + subscribed to, would return the state of the resource. + + This patch makes asterisk to generate URI using SUBSCRIBE request URI. + + ASTERISK-29961 #close + + Change-Id: I1fcfc08fd589677f40608c59a4e143c45ee05f6c + +2022-03-05 06:04 +0000 [1e87cadf8e] Naveen Albert + + * app_dial: Document DIALSTATUS return values. + + Adds documentation for all of the possible return values + for the DIALSTATUS variable in the Dial application. + + ASTERISK-25716 + + Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0 + +2022-03-10 11:07 +0000 [d3abdf0b8d] Sean Bright + + * stasis_recording: Perform a complete match on requested filename. + + Using the length of a file found on the filesystem rather than the + file being requested could result in filenames whose names are + substrings of another to be erroneously matched. + + We now ensure a complete comparison before returning a positive + result. + + ASTERISK-29960 #close + + Change-Id: Id3ffc77681b9b75b8569062f3d952a128a21c71a + +2022-03-22 09:01 +0000 [686c386b05] Sean Bright + + * download_externals: Use HTTPS for downloads + + ASTERISK-29980 #close + + Change-Id: I7b347665822ea2774dd322276c09be67914d2065 + +2022-03-04 14:26 +0000 [c33718a54d] Sean Bright + + * conversions.c: Specify that we only want to parse decimal numbers. + + Passing 0 as the last argument to strtoimax() or strtoumax() causes + octal and hexadecimal to be accepted which was not originally + intended. So we now force to only accept decimal. + + ASTERISK-29950 #close + + Change-Id: I93baf0f273441e8280354630a463df263a8c0edd + +2022-02-21 19:05 +0000 [2a87303ebd] Philip Prindeville + + * logger: workaround woefully small BUFSIZ in MUSL + + MUSL defines BUFSIZ as 1024 which is not reasonable for log messages. + + More broadly, BUFSIZ is the amount of buffering stdio.h does, which + is arbitrary and largely orthogonal to what logging should accept + as the maximum message size. + + ASTERISK-29928 + + Signed-off-by: Philip Prindeville + Change-Id: Iaa49fbbab029c64ae3d95e4b18270e0442cce170 + +2022-03-14 11:57 +0000 [fd29d28832] Naveen Albert + + * pbx_builtins: Add missing options documentation + + BackGround and WaitExten both accept options that are not + currently documented. This adds documentation for these + options to the xml documentation for each application. + + ASTERISK-29967 #close + + Change-Id: If812a9f1ccbba3e4d427a0e7a6dea923c2f905f7 + +2022-02-08 16:58 +0000 [edce853123] Alexei Gradinari + + * res_pjsip_pubsub: update RLS to reflect the changes to the lists + + This patch makes the Resource List Subscriptions (RLS) dynamic. + The asterisk updates the current subscriptions to reflect the changes + to the list on the subscriptions refresh. If list items are added, + removed, updated or do not exist anymore, the asterisk regenerates + the resource list. + + ASTERISK-29906 #close + + Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3 + +2022-02-25 11:01 +0000 [37ece75677] Naveen Albert + + * res_agi: Fix xmldocs bug with set music. + + The XML documentation for the SET MUSIC AGI + command is invalid, as the parameter does not + have a name and the on/off enum options for + the on/off argument are listed separately, which + is incorrect. The cumulative effect of these currently + is that the Asterisk Wiki documentation for SET MUSIC + is broken and external documentation generators crash + on SET MUSIC due to the malformed documentation. + + These issues are corrected so that the documentation + can be successfully parsed as with other similar AGI + commands. + + ASTERISK-29939 #close + ASTERISK-28891 #close + + Change-Id: I8c3d59897531bcbc401cbc7b00c9e2829dcb35f8 + +2022-02-18 03:19 +0000 [636d43caa3] Boris P. Korzun + + * res_config_pgsql: Add text-type column check in require_pgsql() + + Omit "unsupported column type 'text'" warning in logs while + using text-type column in the PgSQL backend. + + ASTERISK-29924 #close + + Change-Id: I48061a7d469426859670db07f1ed8af1eb814712 + +2022-02-09 04:28 +0000 [2be01ba40b] Kfir Itzhak + + * app_queue: Add QueueWithdrawCaller AMI action + + This adds a new AMI action called QueueWithdrawCaller. + This AMI action makes it possible to withdraw a caller from a queue, + in a safe and a generic manner. + This can be useful for retrieving a specific call and + dispatching it to a specific extension. + It works by signaling the caller to exit the queue application + whenever it can. Therefore, it is not guaranteed + that the call will leave the queue. + + ASTERISK-29909 #close + + Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec + +2022-02-24 10:55 +0000 [fbde0186c7] Naveen Albert + + * ami: Improve substring parsing for disabled events. + + ASTERISK_29853 added the ability to selectively disable + AMI events on a global basis, but the logic for this uses + strstr which means that events with names which are the prefix + of another event, if disabled, could disable those events as + well. + + Instead, we account for this possibility to prevent this + undesired behavior from occuring. + + ASTERISK_29853 + + Change-Id: Icccd1872602889806740971e4adf932f92466959 + +2022-03-02 08:57 +0000 [b40c4d59b1] George Joseph + + * xml.c, config,c: Add stylesheets and variable list string parsing + + Added functions to open, close, and apply XML Stylesheets + to XML documents. Although the presence of libxslt was already + being checked by configure, it was only happening if xmldoc was + enabled. Now it's checked regardless. + + Added ability to parse a string consisting of comma separated + name/value pairs into an ast_variable list. The reverse of + ast_variable_list_join(). + + Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e + +2022-03-01 10:58 +0000 [9c36c055c1] George Joseph + + * xmldoc: Fix issue with xmlstarlet validation + + Added the missing xml-stylesheet and Xinclude namespace + declarations in pjsip_config.xml and pjsip_manager.xml. + + Updated make_xml_documentation to show detailed errors when + xmlstarlet is the validator. It's now run once with the '-q' + option to suppress harmless/expected messages and if it actually + fails, it's run again without '-q' but with '-e' to show + the actual errors. + + Change-Id: I4bdc9d2ea6741e8d2e5eb82df60c68ccc59e1f5e + +2022-02-20 14:16 +0000 [b5391ff691] George Joseph + + * core: Config and XML tweaks needed for geolocation + + Added: + + Replace a variable in a list: + int ast_variable_list_replace_variable(struct ast_variable **head, + struct ast_variable *old, struct ast_variable *new); + Added test as well. + + Create a "name=value" string from a variable list: + 'name1="val1",name2="val2"', etc. + struct ast_str *ast_variable_list_join( + const struct ast_variable *head, const char *item_separator, + const char *name_value_separator, const char *quote_char, + struct ast_str **str); + Added test as well. + + Allow the name of an XML element to be changed. + void ast_xml_set_name(struct ast_xml_node *node, const char *name); + + Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb + +2022-02-14 07:31 +0000 [2e00b5edbd] George Joseph + + * Makefile: Allow XML documentation to exist outside source files + + Moved the xmldoc build logic from the top-level Makefile into + its own script "make_xml_documentation" in the build_tools + directory. + + Created a new utility script "get_sourceable_makeopts", also in + the build_tools directory, that dumps the top-level "makeopts" + file in a format that can be "sourced" from shell sscripts. + This allows scripts to easily get the values of common make + build variables such as the location of the GREP, SED, AWK, etc. + utilities as well as the AST* and library *_LIB and *_INCLUDE + variables. + + Besides moving logic out of the Makefile, some optimizations + were done like removing "third-party" from the list of + subdirectories to be searched for documentation and changing some + assignments from "=" to ":=" so they're only evaluated once. + The speed increase is noticeable. + + The makeopts.in file was updated to include the paths to + REALPATH and DIRNAME. The ./conifgure script was setting them + but makeopts.in wasn't including them. + + So... + + With this change, you can now place documentation in any"c" + source file AND you can now place it in a separate XML file + altogether. The following are examples of valid locations: + + res/res_pjsip.c + Using the existing /*** DOCUMENTATION ***/ fragment. + + res/res_pjsip/pjsip_configuration.c + Using the existing /*** DOCUMENTATION ***/ fragment. + + res/res_pjsip/pjsip_doc.xml + A fully-formed XML file. The "configInfo", "manager", + "managerEvent", etc. elements that would be in the "c" + file DOCUMENTATION fragment should be wrapped in proper + XML. Example for "somemodule.xml": + + + + + + ... + + + + It's the "appdocsxml.dtd" that tells make_xml_documentation + that this is a documentation XML file and not some other XML file. + It also allows many XML-capable editors to do formatting and + validation. + + Other than the ".xml" suffix, the name of the file is not + significant. + + As a start... This change also moves the documentation that was + in res_pjsip.c to 2 new XML files in res/res_pjsip: + pjsip_config.xml and pjsip_manager.xml. This cut the number of + lines in res_pjsip.c in half. :) + + Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade + +2022-02-17 10:26 +0000 [1950cec3fd] George Joseph + + * build: Refactor the earlier "basebranch" commit + + Recap from earlier commit: If you have a development branch for a + major project that will receive gerrit reviews it'll probably be + named something like "development/16/newproject" or a work branch + based on that "development" branch. That will necessitate + setting "defaultbranch=development/16/newproject" in .gitreview. + The make_version script uses that variable to construct the + asterisk version however, which results in versions + like "GIT-development/16/newproject-ee582a8c7b" which is probably + not what you want. It also constructs the URLs for downloading + external modules with that version, which will fail. + + Fast-forward: + + The earlier attempt at adding a "basebranch" variable to + .gitreview didn't work out too well in practice because changes + were made to .gitreview, which is a checked-in file. So, if + you wanted to rebase your work branch on the base branch, rebase + would attempt to overwrite your .gitreview with the one from + the base branch and complain about a conflict. + + This is a slighltly different approach that adds three methods to + determine the mainline branch: + + 1. --- MAINLINE_BRANCH from the environment + + If MAINLINE_BRANCH is already set in the environment, that will + be used. This is primarily for the Jenkins jobs. + + 2. --- .develvars + + Instead of storing the basebranch in .gitreview, it can now be + stored in a non-checked-in ".develvars" file and keyed by the + current branch. So, if you were working on a branch named + "new-feature-work" based on "development/16/new-feature" and wanted + to push to that branch in Gerrit but wanted to pull the external + modules for 16, you'd create the following .develvars file: + + [branch "new-feature-work"] + mainline-branch = 16 + + The .gitreview file would still look like: + + [gerrit] + defaultbranch=development/16/new-feature + + ...which would cause any reviews pushed from "new-feature-work" to + go to the "development/16/new-feature" branch in Gerrit. + + The key is that the .develvars file is NEVER checked in (it's been + added to .gitignore). + + 3. --- Well Known Development Branch + + If you're actually working in a branch named like + "development//some-feature", the mainline branch + will be parsed from it. + + 4. --- .gitreview + + If none of the earlier conditions exist, the .gitreview + "defaultbranch" variable will be used just as before. + + Change-Id: I1cdeeaa0944bba3f2e01d7a2039559d0c266f8c9 + +2022-02-23 07:58 +0000 [dd7db5c698] Joshua C. Colp + + * jansson: Update bundled to 2.14 version. + + ASTERISK-29353 + + Change-Id: I4ea43eda1691565563a4c03ef37166952d211b2b + +2022-01-06 07:57 +0000 [27fb4fd5bc] Naveen Albert + + * func_channel: Add lastcontext and lastexten. + + Adds the lastcontext and lastexten channel fields to allow users + to access previous dialplan execution locations. + + ASTERISK-29840 #close + + Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307 + +2022-02-04 19:27 +0000 [3a3b8fbd9f] Naveen Albert + + * channel.c: Clean up debug level 1. + + Although there are 10 debugs levels, over time, + many current debug calls have come to use + inappropriately low debug levels. In particular, + a select few debug calls (currently all debug 1) + can result in thousands of debug messages per minute + for a single call. + + This can adds a lot of noise to core debug + which dilutes the value in having different + debug levels in the first place, as these + log messages are from the core internals are + are better suited for higher debug levels. + + Some debugs levels are thus adjusted so that + debug level 1 is not inappropriately overloaded + with these extremely high-volume and general + debug messages. + + ASTERISK-29897 #close + + Change-Id: I55a71598993552d3d64a401a35ee99474770d4b4 + +2022-02-17 13:47 +0000 [2ba5da15b0] Naveen Albert + + * configs, LICENSE: remove pbx.digium.com. + + pbx.digium.com no longer accepts IAX2 calls and + there are no plans for it to come back. + + Accordingly, nonworking IAX2 URIs are removed from + both the LICENSE file and the sample config. + + ASTERISK-29923 #close + + Change-Id: I257c54d4d812ed6b4bd4cbec2cd7ebe2b87b5bad + +2022-02-04 19:11 +0000 [c35e205bef] Naveen Albert + + * documentation: Add since tag to xmldocs DTD + + Adds the since tag to the documentation DTD so + that individual applications, functions, etc. + can now specify when they were added to Asterisk. + + This tag is added at the individual application, + function, etc. level as opposed to at the module + level because modules can expand over time as new + functionality is added, and granularity only + to the module level would generally not be useful. + + This enables the ability to more easily determine + when new functionality was added to Asterisk, down + to minor version as opposed to just by major version. + This makes it easier for users to write more portable + dialplan if desired to not use functionality that may + not be widely available yet. + + ASTERISK-29896 #close + + Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5 + +2022-01-13 08:37 +0000 [e26b57984f] Naveen Albert + + * asterisk: Add macro for curl user agent. + + Currently, each module that uses libcurl duplicates the standard + Asterisk curl user agent. + + This adds a global macro for the Asterisk user agent used for + curl requests to eliminate this duplication. + + ASTERISK-29861 #close + + Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d + +2021-12-16 13:41 +0000 [1633410161] Naveen Albert + + * res_stir_shaken: refactor utility function + + Refactors temp file utility function into file.c. + + ASTERISK-29809 #close + + Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b + +2022-02-16 05:34 +0000 [39820e3561] Naveen Albert + + * app_voicemail: Emit warning if asking for nonexistent mailbox. + + Currently, if VoiceMailMain is called with a mailbox, if that + mailbox doesn't exist, then the application silently falls back + to prompting the user for the mailbox, as if no arguments were + provided. + + However, if a specific mailbox is requested and it doesn't exist, + then no warning at all is emitted. + + This fixes this behavior to now warn if a specifically + requested mailbox could not be accessed, before falling back to + prompting the user for the correct mailbox. + + ASTERISK-29920 #close + + Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4 + +2022-02-07 16:31 +0000 [a2aa881dcb] Alexei Gradinari + + * res_pjsip_pubsub: fix Batched Notifications stop working + + If Subscription refresh occurred between when the batched notification + was scheduled and the serialized notification was to be sent, + then new schedule notification task would never be added. + + There are 2 threads: + + thread #1. ast_sip_subscription_notify is called, + if notification_batch_interval then call schedule_notification. + 1.1. The schedule_notification checks notify_sched_id > -1 + not true, then + send_scheduled_notify = 1 + notify_sched_id = + ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb.... + 1.2. The sched_cb pushes task serialized_send_notify to serializer + and returns 0 which means no reschedule. + 1.3. The serialized_send_notify checks send_scheduled_notify if it's false + the just returns. BUT notify_sched_id is still set, so no more ast_sched_add. + + thread #2. pubsub_on_rx_refresh is called + 2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer + 2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout + which calls send_notify + 2.3. The send_notify set send_scheduled_notify = 0; + + The serialized_send_notify should always unset notify_sched_id. + + ASTERISK-29904 #close + + Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875 + +2022-02-01 09:59 +0000 [c12cb899de] Alexei Gradinari + + * res_pjsip_pubsub: provide a display name for RLS subscriptions + + Whereas BLFs allow to show a display name for each RLS entry, + the asterisk provides only the extension now. + This is not end user friendly. + + This commit adds a new resource_list option, resource_display_name, + to indicate whether display name of resource or the resource name being + provided for RLS entries. + If this option is enabled, the Display Name will be provided. + This option is disabled by default to remain the previous behavior. + If the 'event' set to 'presence' or 'dialog' the non-empty HINT name + will be set as the Display Name. + The 'message-summary' is not supported yet. + + ASTERISK-29891 #close + + Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681 + +2022-02-18 06:09 +0000 [b1765c93e4] Naveen Albert + + * func_db: Add validity check for key names when writing. + + Adds a simple sanity check for key names when users are + writing data to AstDB. This captures four cases indicating + malformed keynames that generally result in bad data going + into the DB that the user didn't intend: an empty key name, + a key name beginning or ending with a slash, and a key name + containing two slashes in a row. Generally, this is the + result of a variable being used in the key name being empty. + + If a malformed key name is detected, a warning is emitted + to indicate the bug in the dialplan. + + ASTERISK-29925 #close + + Change-Id: Ifc08a9fe532a519b1b80caca1aafed7611d573bf + +2022-01-13 19:37 +0000 [4722c8b70a] Naveen Albert + + * cli: Add core dump info to core show settings. + + Adds two pieces of information to the core show settings command + which are useful in the context of getting backtraces. + + The first is to display whether or not Asterisk would generate + a core dump if it were to crash. + + The second is to show the current running directory of Asterisk. + + ASTERISK-29866 #close + + Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83 + +2022-02-04 19:46 +0000 [335c69ead4] Naveen Albert + + * documentation: Adds missing default attributes. + + The configObject tag contains a default attribute which + allows the default value to be specified, if applicable. + This allows for the default value to show up specially on + the wiki in a way that is clear to users. + + There are a couple places in the tree where default values + are included in the description as opposed to as attributes, + which means these can't be parsed specially for the wiki. + These are changed to use the attribute instead of being + included in the text description. + + ASTERISK-29898 #close + + Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755 + +2022-02-05 06:39 +0000 [c9ef2b3b86] Naveen Albert + + * app_mp3: Document and warn about HTTPS incompatibility. + + mpg123 doesn't support HTTPS, but the MP3Player application + doesn't document this or warn the user about this. HTTPS + streams have become more common nowadays and users could + reasonably try to play them without being aware they should + use the HTTP stream instead. + + This adds documentation to note this limitation. It also + throws a warning if users try to use the HTTPS stream to + tell them to use the HTTP stream instead. + + ASTERISK-29900 #close + + Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827 + +2022-01-22 16:52 +0000 [0da713168d] Naveen Albert + + * app_mf: Add max digits option to ReceiveMF. + + Adds an option to the ReceiveMF application to allow specifying a + maximum number of digits. + + Originally, this capability was not added to ReceiveMF as it was + with ReceiveSF because typically a ST digit is used to denote that + sending of digits is complete. However, there are certain signaling + protocols which simply transmit a digit (such as Expanded In-Band + Signaling) and for these, it's necessary to be able to read a + certain number of digits, as opposed to until receiving a ST digit. + + This capability is added as an option, as opposed to as a parameter, + to remain compatible with existing usage (and not shift the + parameters). + + ASTERISK-29877 #close + + Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b + +2022-01-09 07:32 +0000 [585c2d17bb] Naveen Albert + + * ami: Allow events to be globally disabled. + + The disabledevents setting has been added to the general section + in manager.conf, which allows users to specify events that + should be globally disabled and not sent to any AMI listeners. + + This allows for processing of these AMI events to end sooner and, + for frequent AMI events such as Newexten which users may not have + any need for, allows them to not be processed. Additionally, it also + cleans up core debug as previously when debug was 3 or higher, + the debug was constantly spammed by "Analyzing AMI event" messages + along with a complete dump of the event contents (often for Newexten). + + ASTERISK-29853 #close + + Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205 + +2022-02-02 19:18 +0000 [3b1debb28b] Mike Bradeen + + * taskprocessor.c: Prevent crash on graceful shutdown + + When tps_shutdown is called as part of the cleanup process there is a + chance that one of the taskprocessors that references the + tps_singletons object is still running. The change is to allow for + tps_shutdown to check tps_singleton's container count and give the + running taskprocessors a chance to finish. If after + AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still + container references we shutdown anyway as this is most likely a bug + due to a taskprocessor not being unreferenced. + + ASTERISK-29365 + + Change-Id: Ia932fc003d316389b9c4fd15ad6594458c9727f1 + +2022-01-21 13:00 +0000 [b41440a179] Alexei Gradinari + + * app_queue: load queues and members from Realtime when needed + + There are a lot of Queue AMI actions and Queue applications + which do not load queue and queue members from Realtime. + + AMI actions + QueuePause - if queue not in memory - response "Interface not found". + QueueStatus/QueueSummary - if queue not in memory - empty response. + + Applications: + PauseQueueMember - if queue not in memory + Attempt to pause interface %s, not found + UnpauseQueueMember - if queue not in memory + Attempt to unpause interface xxxxx, not found + + This patch adds a new function load_realtime_queues + which loads queue and queue members for desired queue + or all queues and all members if param 'queuename' is NULL or empty. + Calls the function load_realtime_queues when needed. + + Also this patch fixes leak of ast_config in function set_member_value. + + Also this patch fixes incorrect LOG_WARNING when pausing/unpausing + already paused/unpaused member. + The function ast_update_realtime returns 0 when no record modified. + So 0 is not an error to warn about. + + ASTERISK-29873 #close + ASTERISK-18416 #close + ASTERISK-27597 #close + + Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c + +2022-02-07 10:55 +0000 [16fccf140d] Sean Bright + + * manager.c: Simplify AMI ModuleCheck handling + + This code was needlessly complex and would fail to properly delimit + the response message if LOW_MEMORY was defined. + + Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e + +2022-01-21 07:52 +0000 [427bee9beb] Mark Petersen + + * res_prometheus.c: missing module dependency + + added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO + which fixes issue with module crashes on load "FRACK!, Failed assertion" + + ASTERISK-29871 + + Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096 + +2022-02-03 15:48 +0000 [e1b050d8a3] Sean Bright + + * res_pjsip.c: Correct minor typos in 'realm' documentation. + + Change-Id: I886936b808def5540d40071321e72f6bfa19063a + +2022-01-31 12:52 +0000 [134cbebc1f] Sean Bright + + * manager.c: Generate valid XML if attribute names have leading digits. + + The XML Manager Event Interface (amxml) now generates attribute names + that are compliant with the XML 1.1 specification. Previously, an + attribute name that started with a digit would be rendered as-is, even + though attribute names must not begin with a digit. We now prefix + attribute names that start with a digit with an underscore ('_') to + prevent XML validation failures. + + This is not backwards compatible but my assumption is that compliant + XML parsers would already have been complaining about this. + + ASTERISK-29886 #close + + Change-Id: Icfaa56a131a082d803e9b7db5093806d455a0523 + +2022-02-01 10:09 +0000 [4126d703bf] Sean Bright + + * build_tools/make_version: Fix bashism in comparison. + + In POSIX sh (which we indicate in the shebang), there is no == + operator. + + Change-Id: Ic03d38214d14cdf329b0ba272279a815bb532965 + +2022-01-21 14:08 +0000 [38c3c7f498] George Joseph + + * bundled_pjproject: Add additional multipart search utils + + Added the following APIs: + pjsip_multipart_find_part_by_header() + pjsip_multipart_find_part_by_header_str() + pjsip_multipart_find_part_by_cid_str() + pjsip_multipart_find_part_by_cid_uri() + + Change-Id: I6aee3dcf59eb171f93aae0f0564ff907262ef40d + +2022-01-07 04:01 +0000 [e505337065] Mark Petersen + + * chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN + + resolve issue with pickup on device that uses "183" and not "180" + + ASTERISK-29832 + + Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841 + +2022-01-26 07:56 +0000 [bfc4d63d15] George Joseph + + * build: Add "basebranch" to .gitreview + + If you have a development branch for a major project that + will receive gerrit reviews it'll probably be named something + like "development/16/newproject". That will necessitate setting + "defaultbranch=development/16/newproject" in .gitreview. The + make_version script uses that variable to construct the asterisk + version however, which results in versions like + "GIT-development/16/newproject-ee582a8c7b" which is probably not + what you want. Worse, since the download_externals script uses + make_version to construct the URL to download the binary codecs + or DPMA. Since it's expecting a simple numeric version, the + downloads will fail. + + To get this to work, a new variable "basebranch" has been added + to .gitreview and make_version has been updated to use that instead + of defaultversion: + + .gitreview: + defaultbranch=development/16/myproject + basebranch=16 + + Now git-review will send the reviews to the proper branch + (development/16/myproject) but the version will still be + constructed using the simple branch number (16). + + If "basebranch" is missing from .gitreview, make_version will + fall back to using "defaultbranch". + + Change-Id: I2941a3b21e668febeb6cfbc1a7bb51a67726fcc4 + +2022-01-31 07:09 +0000 [8d571ea6b5] George Joseph + + * res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup + + In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess + that hasn't been initialized, it'll assert and abort. If + digest_create_request_with_auth() fails to find the proper + auth object however, it jumps to its cleanup which does exactly + that. So now we no longer attempt to call pjsip_auth_clt_deinit() + if we never actually initialized it. + + ASTERISK-29888 + + Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62 + +2021-12-15 12:36 +0000 [386c5e495f] Naveen Albert + + * cdr: allow disabling CDR by default on new channels + + Adds a new option, defaultenabled, to the CDR core to + control whether or not CDR is enabled on a newly created + channel. This allows CDR to be disabled by default on + new channels and require the user to explicitly enable + CDR if desired. Existing behavior remains unchanged. + + ASTERISK-29808 #close + + Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1 + +2022-01-11 13:19 +0000 [70f8ea0d1a] Naveen Albert + + * res_tonedetect: Fixes some logic issues and typos + + Fixes some minor logic issues with the module: + + Previously, the OPT_END_FILTER flag was getting + tested before options were parsed, so it could + never evaluate to true (wrong ordering). + + Additionally, the initially parsed timeout (float) + needs to be compared with 0, not the result int + which is set afterwards (wrong variable). + + ASTERISK-29857 #close + + Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c + +2022-01-11 12:33 +0000 [7ae8321925] Naveen Albert + + * func_frame_drop: Fix typo referencing wrong buffer + + In order to get around the issue of certain frames + having names that could overlap, func_frame_drop + surrounds names with commas for the purposes of + comparison. + + The buffer is allocated and printed to properly, + but the original buffer is used for comparison. + In most cases, this wouldn't have had any effect, + but that was not the intention behind the buffer. + This updates the code to reference the modified + buffer instead. + + ASTERISK-29854 #close + + Change-Id: I430b52e14e712d0e62a23aa3b5644fe958b684a7 + +2022-01-20 06:56 +0000 [7b15ced930] Torrey Searle + + * res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf + + When generating dtmfs, asterisk can incorrectly think packet loss + occured during the dtmf generation, resulting in a jump in sequence + numbers when forwarding voice frames resumes. This patch forces + asterisk to re-learn the expected sequence number after each DTMF + to avoid this + + ASTERISK-29869 #close + + Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853 + +2022-01-13 16:31 +0000 [851a759619] Kevin Harwell + + * res_http_websocket: Add a client connection timeout + + Previously there was no way to specify a connection timeout when + attempting to connect a websocket client to a server. This patch + makes it possible to now do such. + + Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d + +2022-01-21 10:34 +0000 [ce91a0fdbc] Sean Bright + + * build: Rebuild configure and autoconfig.h.in + + autoconfigh.h.in was missed in the original review for this + issue. Additionally it looks like I have newer pkg-config autoconf + macros on my development machine. + + ASTERISK-29817 + + Change-Id: I3c85a4de82c5d7d6e0e23dad4c33bb650a86a57b + +2021-12-08 15:14 +0000 [b79a571279] Mike Bradeen + + * sched: fix and test a double deref on delete of an executing call back + + sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an + executing call-back. This is done by adding a new variable 'rescheduled' + to the struct sched which is set in ast_sched_runq and checked in + ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for + now deprecated ast_sched_del which returns a new possible value -2 + if called on an executing call-back with rescheduled set. ast_sched_del + is modified to call ast_sched_del_nonrunning to maintain existing code. + AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it + will not throw a warning or invoke refcall. + test_sched: Add a new unit test sched_test_freebird that will check the + reference count in the resolved scenario. + + ASTERISK-29698 + + Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d + +2022-01-04 03:11 +0000 [93d090147f] Mark Petersen + + * app_queue.c: Queue don't play "thank-you" when here is no hold time announcements + + if holdtime is (0 min, 0 sec) there is no hold time announcements + we should then also not playing queue-thankyou + + ASTERISK-29831 + + Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c + +2022-01-19 16:33 +0000 [5875c7bb6c] Luke Escude + + * res_pjsip_sdp_rtp.c: Support keepalive for video streams. + + ASTERISK-28890 #close + + Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1 + +2021-11-10 21:40 +0000 [23be22abf4] Michał Górny + + * build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD + + Fix the sed(1) invocation used to process git-svn-id not to use "\s" + that is a GNU-ism and is not supported by NetBSD sed. As a result, + this call did not work properly and make_version did output the full + git-svn-id line rather than the revision. + + ASTERISK-29852 + + Change-Id: Ie4b406e2748920643446851a0a252a4ca7245772 + +2021-11-10 22:29 +0000 [2b490787eb] Michał Górny + + * main/utils: Implement ast_get_tid() for NetBSD + + Implement the ast_get_tid() function for NetBSD system. NetBSD supports + getting the TID via _lwp_self(). + + ASTERISK-29850 + + Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c + +2021-11-10 22:24 +0000 [dda02b8979] Michał Górny + + * main: Enable rdtsc support on NetBSD + + Enable the Linux rdtsc implementation on NetBSD as well. The assembly + works correctly there. + + ASTERISK-29851 + + Change-Id: I460ad9b4d971913420ecb84186f5ba5ab03f6f37 + +2021-11-10 20:05 +0000 [6a879eea31] Michał Górny + + * BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD + + Fix the configure script not to detect the presence of gethostbyname_r() + on NetBSD incorrectly. NetBSD includes it as an internal libc symbol + that is not exposed in system headers and that is incompatible with + other implementations. In order to avoid misdetecting it, perform + the symbol check only if the declaration is found in the public header + first. + + ASTERISK-29817 + + Change-Id: Iafa359b09908251bcd299ff54be003ea129b9eda + +2021-11-10 22:06 +0000 [710c8f8b29] Michał Górny + + * include: Remove unimplemented HMAC declarations + + Remove the HMAC declarations from the includes. They are + not implemented nor used anywhere, and their presence breaks the build + on NetBSD that delivers an incompatible hmac() function in . + + ASTERISK-29818 + + Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7 + +2022-01-11 12:41 +0000 [27502b6dd2] Naveen Albert + + * frame.h: Fix spelling typo + + Fixes CNG description from "noice" to "noise". + + ASTERISK-29855 #close + + Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d + +2022-01-11 12:46 +0000 [d35e292ae4] Naveen Albert + + * res_rtp_asterisk: Fix typo in flag test/set + + The code currently checks to see if an RFC3389 + warning flag is set, except if it is, it merely + sets the flag again, the logic of which doesn't + make any sense. + + This adjusts the if comparison to check if the + flag has NOT been set, and if so, emit a notice + log event and set the flag so that future frames + do not cause an event to be logged. + + ASTERISK-29856 #close + + Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1 + +2022-01-18 08:04 +0000 [97ace6b816] George Joseph + + * bundled_pjproject: Fix srtp detection + + Reverted recent change that set '--with-external-srtp' instead + of '--without-external-srtp'. Since Asterisk handles all SRTP, + we don't need it enabled in pjproject at all. + + ASTERISK-29867 + + Change-Id: I2ce1bdd30abd21c062eac8f8fefe9b898787b801 + +2022-01-10 07:44 +0000 [b1dfc9c805] George Joseph + + * res_pjsip: Make message_filter and session multipart aware + + Neither pjsip_message_filter's filter_on_tx_message() nor + res_pjsip_session's session_outgoing_nat_hook() were multipart + aware and just assumed that an SDP would be the only thing in + a message body. Both were changed to use the new + pjsip_get_sdp_info() function which searches for an sdp in + both single- and multi- part message bodies. + + ASTERISK-29813 + + Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56 + +2022-01-12 11:12 +0000 [5d1407aa06] George Joseph + + * build: Fix issues building pjproject + + The change to allow easier hacking on bundled pjproject created + a few issues: + + * The new Makefile was trying to run the bundled make even if + PJPROJECT_BUNDLED=no. third-party/Makefile now checks for + PJPROJECT_BUNDLED and JANSSON_BUNDLED and skips them if they + are "no". + + * When building with bundled, config_site.h was being copied + only if a full make or a "make main" was done. A "make res" + would fail all the pjsip modules because they couldn't find + config_site.h. The Makefile now copies config_site.h and + asterisk_malloc_debug.h into the pjproject source tree + when it's "configure" is performed. This is how it used + to be before the big change. + + ASTERISK-29858 + + Change-Id: I9427264fa3cb8b3f59a95e5f9693eac236a6f76d + +2022-01-06 13:05 +0000 [921ab52cf3] George Joseph + + * res_pjsip: Add utils for checking media types + + Added two new functions to assist checking media types... + + * ast_sip_are_media_types_equal compares two pjsip_media_types. + * ast_sip_is_media_type_in tests if one media type is in a list + of others. + + Added static definitions for commonly used media types to + res_pjsip.h. + + Changed several modules to use the new functions and static + definitions. + + ASTERISK_29813 + (not ready to close) + + Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9 + +2022-01-12 07:16 +0000 [0d1b9e6baf] George Joseph + + * bundled_pjproject: Create generic pjsip_hdr_find functions + + pjsip_msg_find_hdr(), pjsip_msg_find_hdr_by_name(), and + pjsip_msg_find_hdr_by_names() require a pjsip_msg to be passed in + so if you need to search a header list that's not in a pjsip_msg, + you have to do it yourself. This commit adds generic versions of + those 3 functions that take in the actual header list head instead + of a pjsip_msg so if you need to search a list of headers in + something like a pjsip_multipart_part, you can do so easily. + + Change-Id: I6f2c127170eafda48e5e0d5d4d187bcd52b4df07 + +2022-01-12 13:20 +0000 [65b2ddee26] Sean Bright + + * say.c: Prevent erroneous failures with 'say' family of functions. + + A regression was introduced in ASTERISK~29531 that caused 'say' + functions to fail with file lists that would previously have + succeeded. This caused affected channels to hang up where previously + they would have continued. + + We now explicitly check for the empty string to restore the previous + behavior. + + ASTERISK-29859 #close + + Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5 + +2022-01-08 14:35 +0000 [5f59e0d36f] Naveen Albert + + * documentation: Document built-in system and channel vars + + Documentation for built-in special system and channel + vars is currently outdated, and updating is a manual + process since there is no XML documentation for these + anywhere. + + This adds documentation for system vars to func_env + and for channel vars to func_channel so that they + appear along with the corresponding fields that would + be accessed using a function. + + ASTERISK-29848 #close + + Change-Id: I6997f925c4a45fffe71321861f5898a8b7182fa9 + +2022-01-08 09:09 +0000 [fbaf74bd3a] Naveen Albert + + * pbx_variables: add missing ASTSBINDIR variable + + Every config variable in the directories + section of asterisk.conf currently has a + counterpart built-in variable containing + the value of the config option, except + for the last one, astsbindir, which should + have an ASTSBINDIR variable. + + However, the actual corresponding ASTSBINDIR + variable is missing in pbx_variables.c. + + This adds the missing variable so that all + the config options have their corresponding + variable. + + ASTERISK-29847 #close + + Change-Id: I36006faf471825b36ebc8aa5e87a3bcb38d446fc + +2021-11-30 16:35 +0000 [bc59b66de3] George Joseph + + * bundled_pjproject: Make it easier to hack + + There are times when you need to troubleshoot issues with bundled + pjproject or add new features that need to be pushed upstream + but... + + * The source directory created by extracting the pjproject tarball + is not scanned for code changes so you have to keep forcing + rebuilds. + * The source directory isn't a git repo so you can't easily create + patches, do git bisects, etc. + * Accidentally doing a make distclean will ruin your day by wiping + out the source directory, and your changes. + * etc. + + This commit makes that easier. + See third-party/pjproject/README-hacking.md for the details. + + ASTERISK-29824 + + Change-Id: Idb1251040affdab31d27cd272dda68676da9b268 + +2021-12-24 10:26 +0000 [0d62735f99] Sean Bright + + * utils.c: Remove all usages of ast_gethostbyname() + + gethostbyname() and gethostbyname_r() are deprecated in favor of + getaddrinfo() which we use in the ast_sockaddr family of functions. + + ASTERISK-29819 #close + + Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7 + +2021-12-13 09:53 +0000 [262a4053ff] Naveen Albert + + * say.conf: fix 12pm noon logic + + Fixes 12pm noon incorrectly returning 0/a.m. + Also fixes a misspelling typo in the config. + + ASTERISK-29695 #close + + Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406 + +2022-01-04 08:08 +0000 [3616dda066] Sean Bright + + * pjproject: Fix incorrect unescaping of tokens during parsing + + ASTERISK-29664 #close + + Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5 + +2021-12-30 07:02 +0000 [dc7bcd68e4] Mark Petersen + + * app_queue.c: Support for Nordic syntax in announcements + + adding support for playing the correct en/et for nordic languages + by adding 'n' for neuter gender in the relevant ast_say_number + + ASTERISK-29827 + + Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1 + +2021-12-23 08:50 +0000 [138fbfa274] Naveen Albert + + * dsp: Add define macro for DTMF_MATRIX_SIZE + + Adds the macro DTMF_MATRIX_SIZE to replace + the magic number 4 sprinkled throughout + dsp.c. + + ASTERISK-29815 #close + + Change-Id: Ie3bddb92c6b16204ece0f758009e9490eb33b9ba + +2022-01-03 11:10 +0000 [68f1e5d508] Naveen Albert + + * ami: Add AMI event for Wink + + Adds an AMI event for a wink frame. + + ASTERISK-29830 #close + + Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56 + +2021-12-15 08:23 +0000 [5b8d68d678] Naveen Albert + + * cli: Add module refresh command + + Adds a command to the CLI to unload and then + load a module. This makes it easier to perform + these operations which are often done + subsequently to load a new version of a module. + + "module reload" already refers to reloading of + configuration, so the name "refresh" is chosen + instead. + + ASTERISK-29807 #close + + Change-Id: I595f6f11774a0de2565a1fba38da22309ce93a2c + +2022-01-02 19:13 +0000 [80766059ef] Naveen Albert + + * app_mp3: Throw warning on nonexistent stream + + Currently, the MP3Player application doesn't + emit a warning if attempting to play a stream + which no longer exists. This can be a common + scenario as many mp3 streams are valid at some + point but can disappear at any time. + + Now a warning is thrown if attempting to play + a nonexistent MP3 stream, instead of silently + exiting. + + ASTERISK-29829 #close + + Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535 + +2021-12-13 08:29 +0000 [70bc0ff9d0] Naveen Albert + + * documentation: Add missing AMI documentation + + Adds missing documentation for some channel, + bridge, and queue events. + + ASTERISK-24427 + ASTERISK-29515 + + Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64 + +2021-11-15 16:13 +0000 [1ddaedeaf5] Kevin Harwell + + * tcptls.c: refactor client connection to be more robust + + The current TCP client connect code, blocks and does not handle EINTR + error case. + + This patch makes the client socket non-blocking while connecting, + ensures a connect does not immediately fail due to EINTR "errors", + and adds a connect timeout option. + + The original client start call sets the new timeout option to + "infinite", thus making sure old, orginal behavior is retained. + + ASTERISK-29746 #close + + Change-Id: I907571843a83e43c0742b95a64785f4411f02671 + +2021-12-13 10:59 +0000 [f7c4a3800c] Naveen Albert + + * app_sf: Add full tech-agnostic SF support + + Adds tech-agnostic support for SF signaling + by adding SF sender and receiver applications + as well as Dial integration. + + ASTERISK-29802 #close + + Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad + +2021-12-15 06:23 +0000 [a2ea233a6d] Steve Davies + + * app_queue: Fix hint updates, allow dup. hints + + A previous patch for ASTERISK_29578 caused a 'leak' of + extension state information across queues, causing the + state of the first member of unrelated queues to be + updated in addition to the correct member. Which queues + and members depended on the order of queues in the + iterator. + + Additionally, it is possible to use the same 'hint:' on + multiple queue members, so the update cannot break out + of the update loop early when a match is found. + + ASTERISK-29806 #close + + Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad + +2021-12-23 15:57 +0000 [3fd12f1aa3] Sean Bright + + * say.c: Honor requests for DTMF interruption. + + SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal, + and SayPhonetic all claim to allow DTMF interruption if the + SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we + are failing to break out of a given 'say' application if DTMF actually + occurs. + + ASTERISK-29816 #close + + Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e + +2021-11-16 06:32 +0000 [dd41572f99] Florentin Mayer + + * res_pjsip_sdp_rtp: Preserve order of RTP codecs + + The ast_rtp_codecs_payloads functions do not preserve the order in which + the payloads were specified on an incoming SDP media line. This leads to + a problem with the codec negotiation functionality, as the format + capabilities of the stream are extracted from the ast_rtp_codecs. This + commit moves the ast_rtp_codec to ast_format conversion to the place + where the order is still known. + + ASTERISK-28863 + ASTERISK-29320 + + Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25 + +2021-12-27 07:28 +0000 [f9e67945da] Joshua C. Colp + + * bridge: Unlock channel during Local peer check. + + It's not safe to keep the channel locked while locking + the peer Local channel, as it can result in a deadlock. + + This change unlocks it during this time but keeps the + bridge locked to ensure nothing changes about the bridge. + + ASTERISK-29821 + + Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6 + +2021-11-07 09:32 +0000 [2b61440027] Josh Soref + + * test_time.c: Tolerate DST transitions + + When test_timezone_watch runs very near a DST transition, + two time zones that would otherwise be expected to report the same + time can differ because of the DST transition. + + Instead of having the test fail when this happens, report the + times, time zones, and dst flags. + + ASTERISK-29722 + + Change-Id: Id59bdac8b277e14343ccdf0c99b89e92f79f316a + +2021-12-14 11:39 +0000 [7728210352] George Joseph + + * bundled_pjproject: Add more support for multipart bodies + + Adding upstream patch for pull request... + https://github.com/pjsip/pjproject/pull/2920 + --------------------------------------------------------------- + + sip_inv: Additional multipart support (#2919) + + sip_inv.c:inv_check_sdp_in_incoming_msg() deals with multipart + message bodies in rdata correctly. In the case where early media is + involved though, the existing sdp has to be retrieved from the last + tdata sent in this transaction. This, however, always assumes that + the sdp sent is in a non-multipart body. While there's a function + to retrieve the sdp from multipart and non-multpart rdata bodies, + no similar function for tdata exists. So... + + * The existing pjsip_rdata_get_sdp_info2 was refactored to + find the sdp in any body, multipart or non-multipart, and + from either an rdata or tdata. The new function is + pjsip_get_sdp_info. This new function detects whether the + pjsip_msg->body->data is the text representation of the sdp + from an rdata or an existing pjmedia_sdp_session object + from a tdata, or whether pjsip_msg->body is a multipart + body containing either of the two sdp formats. + + * The exsting pjsip_rdata_get_sdp_info and pjsip_rdata_get_sdp_info2 + functions are now wrappers that get the body and Content-Type + header from the rdata and call pjsip_get_sdp_info. + + * Two new wrappers named pjsip_tdata_get_sdp_info and + pjsip_tdata_get_sdp_info2 have been created that get the body + from the tdata and call pjsip_get_sdp_info. + + * inv_offer_answer_test.c was updated to test multipart scenarios. + + ASTERISK-29804 + + Change-Id: I483c7c3d413280c9e247a96ad581278347f9c71b + +2021-12-09 02:55 +0000 [cb44ceadec] Frederic Van Espen + + * ast_coredumper: Fix deleting results when output dir is set + + When OUTPUTDIR is set to another directory and the + --delete-results-after is set, the resulting txt files are + not deleted. + + ASTERISK-29794 #close + + Change-Id: I1c0071f6809a1e3f5cfc455d6eb08378bc0d7286 + +2021-12-13 16:49 +0000 [cfcbf0adad] Naveen Albert + + * pbx_variables: initialize uninitialized variable + + The variable cp4 in a variable substitution function + can potentially be used without being initialized + currently. This causes Asterisk to no longer compile. + + This initializes cp4 to NULL to make the compiler + happy. + + ASTERISK-29803 #close + + Change-Id: I392579cbb76db2795d5820c9427cf55fbcee9e72 + +2021-12-08 05:24 +0000 [92cb1c0a59] Mark Petersen + + * app_queue.c: added DIALEDPEERNUMBER on outgoing channel + + added that we set DIALEDPEERNUMBER on the outgoing channels + so it is avalible in b(content^extension^line) + this add the same behaviour as Dial + + ASTERISK-29795 + + Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6 + +2021-11-15 15:35 +0000 [1c389faa31] Kevin Harwell + + * http.c: Add ability to create multiple HTTP servers + + Previously, it was only possible to have one HTTP server in Asterisk. + With this patch it is now possible to have multiple HTTP servers + listening on different addresses. + + Note, this behavior has only been made available through an API call + from within the TEST_FRAMEWORK. Specifically, this feature has been + added in order to allow unit test to create/start and stop servers, + if one has not been enabled through configuration. + + Change-Id: Ic5fb5f11e62c019a1c51310f4667b32a4dae52f5 + +2021-12-12 18:08 +0000 [b951821eb7] Naveen Albert + + * app.c: Throw warnings for nonexistent options + + Currently, Asterisk doesn't throw warnings if options + are passed into applications that don't accept them. + This can confuse users if they're unaware that they + are doing something wrong. + + This adds an additional check to parse_options so that + a warning is thrown anytime an option is parsed that + doesn't exist in the parsing application, so that users + are notified of the invalid usage. + + ASTERISK-29801 #close + + Change-Id: Id029274a57135caca193c913307a63fd75e24679 + +2021-12-08 12:07 +0000 [4f06de7cf8] Mark Petersen + + * app_voicemail.c: Support for Danish syntax in VM + + added support for playing the correct plural sound file + dependen on where you have 1 or multipe messages + based on the existing SE/NO code + + ASTERISK-29797 + + Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2 + +2021-11-17 15:39 +0000 [54761a41cd] Naveen Albert + + * app_sendtext: Add ReceiveText application + + Adds a ReceiveText application that can be used in + conjunction with SendText. Currently, there is no + way in Asterisk to receive text in the dialplan + (or anywhere else, really). This allows for Asterisk + to be the recipient of text instead of just the sender. + + ASTERISK-29759 #close + + Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d + +2021-12-11 20:11 +0000 [8ec13f06de] Naveen Albert + + * strings: Fix enum names in comment examples + + The enum values for ast_strsep_flags includes + AST_STRSEP_STRIP. However, some comments reference + AST_SEP_STRIP, which doesn't exist. This fixes + these comments to use the correct value. + + ASTERISK-29800 #close + + Change-Id: If7bbd0c0e6226a211d25ddf9d1629347e2674943 + +2021-11-20 14:37 +0000 [5c67a991c2] Naveen Albert + + * pbx_variables: Increase parsing capabilities of MSet + + Currently MSet can only parse a maximum of 24 variables. + If more variables are provided to MSet, the 24th variable + will simply contain the remainder of the string and the + remaining variables thereafter will never get set. + + This increases the number of variables that can be parsed + in one go from 24 to 99. Additionally, documentation is added + since this limitation is currently undocumented and is + confusing to users who encounter this limitation. + + ASTERISK-29766 #close + + Change-Id: I3fe35b462dedec0a452fd9ea7f92c920a3939f16 + +2021-11-23 20:21 +0000 [97f400100c] Naveen Albert + + * chan_sip: Fix crash when accessing RURI before initiating outgoing call + + Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before + initiating an outgoing call will cause Asterisk to crash. This is + because a null field is accessed, resulting in an offset from null and + subsequent memory access violation. + + Since RURI is not guaranteed to exist, we now check if the base + pointer is non-null before calculating an offset. + + ASTERISK-29772 + + Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83 + +2021-10-25 16:19 +0000 [b64e894650] Naveen Albert + + * func_json: Adds JSON_DECODE function + + Adds the JSON_DECODE function for parsing JSON in the + dialplan. JSON parsing already exists in the Asterisk + core and is used for many different things. This + function exposes the basic parsing capability to + the user in the dialplan, for instance, in conjunction + with CURL for using API responses. + + ASTERISK-29706 #close + + Change-Id: Iea60c49a7358dfdc2db60803cdc9a742f808ba2c + +2021-11-17 15:16 +0000 [c3ff464864] Naveen Albert + + * configs: Updates to sample configs + + Includes some minor updates to extensions.conf + and iax.conf. In particular, the demonstration + of macros in extensions.conf is removed, as + Macro is deprecated and will be removed soon. + These examples have been replaced with examples + demonstrating the usage of Gosub instead. + + The older exten => ...,n syntax is also mostly + replaced with the same keyword to demonstrate the + newer, more concise way of defining extensions. + + IAXTEL no longer exists, so this example is replaced + with something more generic. + + Some documentation is also added to extensions.conf + and iax.conf to clarify some of the new expanded + encryption capabilities with IAX2. + + ASTERISK-29758 #close + + Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46 + +2021-11-15 15:08 +0000 [23a4a12420] Naveen Albert + + * pbx: Add variable substitution API for extensions + + Currently, variable substitution involving dialplan + extensions is quite clunky since it entails obtaining + the current dialplan location, backing it up, storing + the desired variables for substitution on the channel, + performing substitution, then restoring the original + location. + + In addition to being clunky, things could also go wrong + if an async goto were to occur and change the dialplan + location during a substitution. + + Fundamentally, there's no reason it needs to be done this + way, so new API is added to allow for directly passing in + the dialplan location for the purposes of variable + substitution so we don't need to mess with the channel + information anymore. Existing API is not changed. + + ASTERISK-29745 #close + + Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4 + +2021-12-11 18:45 +0000 [6a6967bf0c] Sean Bright + + * CHANGES: Correct reference to configuration file. + + Change-Id: I22a788ebf11168fff7fbf9ea956ebcd705ab63dd + +2021-09-21 19:18 +0000 [ee9eef492c] Naveen Albert + + * app_mf: Add full tech-agnostic MF support + + Adds tech-agnostic support for MF signaling by adding + MF sender and receiver applications as well as Dial + integration. + + ASTERISK-29496-mf #do-not-close + + Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4 + +2021-12-06 04:25 +0000 [67c4661fb0] Alexander Traud + + * xmldoc: Avoid whitespace around value for parameter/required. + + Otherwise, the value 'false' was not found in the enumerated set of + the XML DTD for the XML attribute 'required' in the XML element + 'parameter'. Therefore, DTD validation of the runtime XML failed. + + ASTERISK-29790 + + Change-Id: Id13f230ad65a70dd8c2e3ae9ac85d1e841aed03e + +2021-12-04 02:36 +0000 [826233b550] Alexander Traud + + * progdocs: Fix Doxygen left-overs. + + Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016 + +2021-12-06 05:17 +0000 [12c45dd6a2] Alexander Traud + + * xmldoc: Correct definition for XML element 'matchInfo'. + + ASTERISK-29791 + + Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725 + +2021-11-23 08:05 +0000 [f3b29c6aa8] Alexander Traud + + * progdocs: Update Makefile. + + In developer mode, use internal documentation as well. + This should produce no warnings. Fix yours! + + In noisy mode, output all possible warnings of Doxygen. + This creates zillion of warnings. Double-check your current module! + + Any warnings are in the file './doxygen.log'. Beside that, this change + avoids deprecated parameters because the configuration file for Doxygen + contains only those parameters which differ from the default. This + avoids the need to update the file on each run. Furthermore, it adds + AST_VECTOR to be expanded. Finally, the default name for that file is + Doxyfile. Therefore, let us use that! + + ASTERISK-26991 + ASTERISK-20259 + + Change-Id: I4129092a199d5e24c319a09cd088614b121015af + +2021-12-03 07:38 +0000 [f6df28ce87] Alexander Traud + + * res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites. + + res_sdp_crypto_parse_offer(.) emits many log messages already. + + ASTERISK-29785 + + Change-Id: I1a191ebe4fec1102946d4e31887e5197ca02dfe8 + +2021-11-30 14:16 +0000 [e9cac5f4bf] Sean Bright + + * channel: Short-circuit ast_channel_get_by_name() on empty arg. + + We know that passing a NULL or empty argument to + ast_channel_get_by_name() will never result in a matching channel and + will always result in an error being emitted, so just short-circuit + out in that case. + + ASTERISK-28219 #close + + Change-Id: I88eadc748e9c6996fc17467b0a05881bbfd00bce + +2021-10-26 16:12 +0000 [59fcd1e7e2] Mike Bradeen + + * res_rtp_asterisk: Addressing possible rtp range issues + + res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined + that rtpstart was configured to be an odd value. Also adding a loop + counter to prevent a possible infinite loop when looking for a free + port. + + ASTERISK-27406 + + Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b + +2021-08-24 09:56 +0000 [a8b2692836] Mark Petersen + + * apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL_CLEARING + + changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING + + ASTERISK-28053 + Reported by: roadkill + + Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89 + +2021-11-19 02:54 +0000 [a85f2bf34d] Alexander Traud + + * res: Fix for Doxygen. + + These are the remaining issues found in /res. + + ASTERISK-29761 + + Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d + +2021-11-08 18:30 +0000 [e93fb874b4] Dustin Marquess + + * res_fax_spandsp: Add spandsp 3.0.0+ compatibility + + Newer versions of spandsp did refactoring of code to add new features + like color FAXing. This refactoring broke backwards compatibility. + Add support for the new version while retaining support for 0.0.6. + + ASTERISK-29729 #close + + Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1 + +2021-11-19 09:47 +0000 [9440f6ec58] Alexander Traud + + * main: Fix for Doxygen. + + ASTERISK-29763 + + Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e + +2021-11-27 13:11 +0000 [cc025026b7] Alexander Traud + + * progdocs: Fix for Doxygen, the hidden parts. + + ASTERISK-29779 + + Change-Id: If338163488498f65fa7248b60e80299c0a928e4b + +2021-11-12 10:05 +0000 [affe7ee879] Alexander Traud + + * progdocs: Fix grouping for latest Doxygen. + + Since Doxygen 1.8.16, a special comment block is required. Otherwise + (pure C comment), the group command is ignored. Additionally, several + unbalanced group commands were fixed. + + ASTERISK-29732 + + Change-Id: I4687857b9d56e6f44fd440b73af156691660202e + +2021-11-25 12:41 +0000 [24a04054ad] Naveen Albert + + * documentation: Standardize examples + + Most examples in the XML documentation use the + example tag to demonstrate examples, which gets + parsed specially in the Wiki to make it easier + to follow for users. + + This fixes a few modules to use the example + tag instead of vanilla para tags to bring them + in line with the standard syntax. + + ASTERISK-29777 #close + + Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b + +2021-11-28 14:52 +0000 [2478bfcff9] Sean Bright + + * config.c: Prevent UB in ast_realtime_require_field. + + A backend's implementation of the realtime 'require' function may call + va_arg() and then fail, leaving the va_list in an undefined + state. Pass a copy of the va_list instead. + + ASTERISK-29771 #close + + Change-Id: I555565a72af84e96d49f62fe8cb66ba5a78461f4 + +2021-11-01 10:40 +0000 [d374d63ef8] Naveen Albert + + * app_voicemail: Refactor email generation functions + + Refactors generic functions used for email generation + into utils.c so that they can be used by multiple + modules, including app_voicemail and app_minivm, + to avoid code duplication. + + ASTERISK-29715 #close + + Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee + +2021-11-25 10:34 +0000 [ecffdab059] Alexander Traud + + * stir/shaken: Avoid a compiler extension of GCC. + + ASTERISK-29776 + + Change-Id: I86e5eca66fb775a5744af0c929fb269e70575a73 + +2021-11-23 07:12 +0000 [1230369b71] Alexander Traud + + * progdocs: Remove outdated references in doxyref.h. + + ASTERISK-29773 + + Change-Id: Ica93160d9158cc0e80c5fda829b80d1b49a6b9b9 + +2021-10-28 02:28 +0000 [4b3c75ca31] Jaco Kroon + + * logger: use __FUNCTION__ instead of __PRETTY_FUNCTION__ + + This avoids a few long-name overflows, at the cost of less instructive + names in the case of C++ (specifically overloaded functions and class + methods). This in turn is offset against the fact that we're logging + the filename and line numbers in any case. + + Change-Id: I54101a0bb5f8cb9ef63ec12c5e0d4c8edafff9ed + Signed-off-by: Jaco Kroon + +2021-11-20 06:05 +0000 [38f9000fcb] Alexander Traud + + * xmldoc: Fix for Doxygen. + + ASTERISK-29765 + + Change-Id: I654ba0debe8351038d4433716434a09370f04c9d + +2021-11-16 16:34 +0000 [4a4f1a5c9a] Mike Bradeen + + * astobj2.c: Fix core when ref_log enabled + + In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj + structure is freed, but is then referenced later if ref_log is + enabled. The change is to store the obj->priv_data.options value + locally and reference it instead of the value from the freed obj + + ASTERISK-29730 + + Change-Id: I60cc5dc1f5a4330e7ad56976fc38a42de0ab6072 + +2021-11-19 03:46 +0000 [726d6dd166] Alexander Traud + + * channels: Fix for Doxygen. + + ASTERISK-29762 + + Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7 + +2021-11-16 04:06 +0000 [3a4c9ec0e2] Joshua C. Colp + + * bridge: Deny full Local channel pair in bridge. + + Local channels are made up of two pairs - the 1 and 2 + sides. When a frame goes in one side, it comes out the + other. Back and forth. When both halves are in a + bridge this creates an infinite loop of frames. + + This change makes it so that bridging no longer + allows both of these sides to exist in the same + bridge. + + ASTERISK-29748 + + Change-Id: I29928b6de87cd9be996a77daccefd7c360fef651 + +2021-11-06 18:35 +0000 [4468fc11d6] Naveen Albert + + * res_tonedetect: Add call progress tone detection + + Makes basic call progress tone detection available + in a tech-agnostic manner with the addition of the + ToneScan application. This can determine if the channel + has encountered a busy signal, SIT tones, dial tone, + modem, fax machine, etc. A few basic async progress + tone detect options are also added to the TONE_DETECT + function. + + ASTERISK-29720 #close + + Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90 + +2021-11-08 13:59 +0000 [f6aed7b8d1] Boris P. Korzun + + * rtp_engine: Add type field for JSON RTCP Report stasis messages + + ASTERISK-29727 #close + + Change-Id: I2eca8aeb591cb63ac2238d08eab662367453cb82 + +2021-11-17 03:24 +0000 [00fc7212bd] Alexander Traud + + * odbc: Fix for Doxygen. + + ASTERISK-29754 + + Change-Id: Ia09eb68d283d201d9a6fbeccfc0efe83fe0502a5 + +2021-11-17 02:54 +0000 [241dbb1ec0] Alexander Traud + + * parking: Fix for Doxygen. + + ASTERISK-29753 + + Change-Id: I7a61974584f6169502e6860fc711919fe7bbfaa7 + +2021-11-17 06:18 +0000 [634e3ebdb8] Alexander Traud + + * res_ari: Fix for Doxygen. + + ASTERISK-29756 + + Change-Id: I2f1c1eea1c902492b77b74de9950f20ebbb7e758 + +2021-11-17 04:26 +0000 [c30ed45c94] Alexander Traud + + * frame: Fix for Doxygen. + + ASTERISK-29755 + + Change-Id: I8240013ec3db0669c0acf67e26bf6c9cbb5b72af + +2021-11-17 05:43 +0000 [9ae084ff44] Alexander Traud + + * ari-stubs: Avoid 'is' as comparism with an literal. + + Python 3.9.7 gave a syntax warning. + + Change-Id: I3e3a982fe720726bc0015bcdb0e638a626ec89d4 + +2021-11-08 09:42 +0000 [5d8e0a6542] Alexander Traud + + * BuildSystem: Consistently allow 'ye' even for Jansson. + + Furthermore, consistently use not 'No' but ':' for non-existent file + paths. Finally, use the same pattern for checking file paths: + a) = ":" + b) != "x:" + + Change-Id: I0c80c76d2cc98b0e5c859131290f4e3141a1a544 + +2021-11-16 10:26 +0000 [acd1cd66b8] Alexander Traud + + * stasis: Fix for Doxygen. + + ASTERISK-29750 + + Change-Id: Iea50173e785b2e9d49bc24c0af7111cfd96d44a9 + +2021-11-17 02:30 +0000 [173bc6b4c3] Alexander Traud + + * app: Fix for Doxygen. + + ASTERISK-29752 + + Change-Id: If40cbd01d47a6cfd620b18206dedb8460216c8af + +2021-11-16 06:51 +0000 [845ece8bc4] Alexander Traud + + * res_xmpp: Fix for Doxygen. + + ASTERISK-29749 + + Change-Id: I7885793b63bdeaa883e76edb899bbba9660eb1c5 + +2021-11-16 12:07 +0000 [fa91010229] Alexander Traud + + * channel: Fix for Doxygen. + + ASTERISK-29751 + + Change-Id: Ie04da5029c57ebee44733bdf05013156abe80176 + +2021-11-13 06:04 +0000 [4051434be4] Alexander Traud + + * chan_iax2: Fix for Doxygen. + + ASTERISK-29737 + + Change-Id: I282003cc553989fd5c19ceeac9e478fa4ee06cec + +2021-11-16 03:40 +0000 [463f6c83e8] Alexander Traud + + * res_pjsip: Fix for Doxygen. + + ASTERISK-29747 + + Change-Id: Ic7a1e9453f805a6264fe86c96b7d18b87b376084 + +2021-11-15 08:12 +0000 [8944dc78d1] Alexander Traud + + * bridges: Fix for Doxygen. + + ASTERISK-29743 + + Change-Id: I6e1bbbaa5875e19994a328ab40a5d429c6010e8b + +2021-11-15 07:38 +0000 [2024c2e476] Alexander Traud + + * addons: Fix for Doxygen. + + ASTERISK-29742 + + Change-Id: Ie752cb9638ced1ebe3a55d710c6c18ef6bd0aafc + +2021-11-15 07:18 +0000 [196c24df22] Alexander Traud + + * apps: Fix for Doxygen. + + ASTERISK-29740 + + Change-Id: Icb6fbcfea0a5f1c82caa5001902b6a786adbf307 + +2021-11-15 07:29 +0000 [47ade30c6b] Alexander Traud + + * tests: Fix for Doxygen. + + ASTERISK-29741 + + Change-Id: I012d72b237bda2ef2d0f86307dfc6dc7add4b54b + +2021-11-12 13:52 +0000 [2b90194d63] Alexander Traud + + * progdocs: Avoid multiple use of section labels. + + ASTERISK-29735 + + Change-Id: I56935e73f7bd1d4ae2721d11040f4835da64b810 + +2021-11-12 13:17 +0000 [e79271cca4] Alexander Traud + + * progdocs: Use Doxygen \example correctly. + + ASTERISK-29734 + + Change-Id: I83b51e85cd71867645ab3a8a820f8fd1f065abd2 + +2021-11-13 04:40 +0000 [55110339ec] Alexander Traud + + * bridge_channel: Fix for Doxygen. + + ASTERISK-29736 + + Change-Id: Ia5370289e6526001a6b52754b533bcea1a9d7e5c + +2021-11-12 12:41 +0000 [57fef28dc9] Alexander Traud + + * progdocs: Avoid 'name' with Doxygen \file. + + Fixes four misuses of the parameter 'name'. Additionally, for + consistency and to avoid such an issue in future, those few other + places, which used '\file name', were changed just to '\file'. Then, + Doxygen uses the name of the current file. + + ASTERISK-29733 + + Change-Id: I0c18b4c863c6988b138c77448057349a9ee7052d + +2021-11-15 13:02 +0000 [ad67f6966e] Naveen Albert + + * app_morsecode: Fix deadlock + + Fixes a deadlock in app_morsecode caused by locking + the channel twice when reading variables from the + channel. The duplicate lock is simply removed. + + ASTERISK-29744 #close + + Change-Id: I204000701f123361d7f85e0498fedc90243c75e4 + +2021-10-25 12:51 +0000 [2320a96349] Naveen Albert + + * app_read: Fix custom terminator functionality regression + + Currently, when the t option is specified with no arguments, + the # character is still treated as a terminator, even though + no character should be treated as a terminator. + + This is because a previous regression fix was modified to + remove the use of NULL as a default altogether. However, + NULL and an empty string actually refer to different + arrangements and should be treated differently. NULL is the + default terminator (#), while an empty string removes the + terminator altogether. This is the behavior being used by + the rest of the core. + + Additionally, since S_OR catches empty strings as well as + NULL (not intended), this is changed to a ternary operator + instead, which fixes the behavior. + + ASTERISK-29705 #close + + Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086 + +2021-10-24 13:38 +0000 [126de2839b] Naveen Albert + + * res_pjsip_callerid: Fix OLI parsing + + Fix parsing of ANI2/OLI information, since it was previously + parsing the user, when it should have been parsing other_param. + + Also improves the parsing by using pjproject native functions + rather than trying to parse the parameters ourselves like + chan_sip did. A previous attempt at this caused a crash, but + this works correctly now. + + ASTERISK-29703 #close + + Change-Id: I8f3c79032d9ea1a21d16f8e11f22bd8d887738a1 + +2021-10-30 20:04 +0000 [b4966c4f2a] Josh Soref + + * build_tools: Spelling fixes + + Correct typos of the following word families: + + binutils + + ASTERISK-29714 + + Change-Id: I2f676ab48cd50edc400c43307cb53679e4c09b97 + +2021-10-30 20:04 +0000 [815e99d5ea] Josh Soref + + * contrib: Spelling fixes + + Correct typos of the following word families: + + standard + increase + comments + valgrind + promiscuous + editing + libtonezone + storage + aggressive + whitespace + russellbryant + consecutive + peternixon + + ASTERISK-29714 + + Change-Id: I9cafbf41b579c9c0c84c81719d2c4f900beec245 + +2021-10-30 20:04 +0000 [84556eb962] Josh Soref + + * codecs: Spelling fixes + + Correct typos of the following word families: + + voiced + denumerator + codeword + upsampling + constructed + residual + subroutine + conditional + quantizing + courtesy + number + + ASTERISK-29714 + + Change-Id: I471fb8086a5277d8f05047fedee22cfa97a4252d + +2021-10-30 20:04 +0000 [7285ba33ee] Josh Soref + + * formats: Spelling fixes + + Correct typos of the following word families: + + truncate + + ASTERISK-29714 + + Change-Id: I6507760c72b919873cff7cac22b3781036cd4955 + +2021-10-30 20:04 +0000 [623fece76d] Josh Soref + + * CREDITS: Spelling fixes + + Correct typos of the following word families: + + contributors + + ASTERISK-29714 + + Change-Id: I6f46dae8bf8125a21ce8ff318380b2b412d9d2f9 + +2021-10-30 20:04 +0000 [01697d4836] Josh Soref + + * addons: Spelling fixes + + Correct typos of the following word families: + + definition + listener + fastcopy + logical + registration + classify + documentation + explicitly + dialed + endpoint + elements + arithmetic + might + prepend + byte + terminal + inquiry + skipping + aliases + calling + absent + authentication + transmit + their + ericsson + disconnecting + redir + items + client + adapter + transmitter + existing + satisfies + pointer + interval + supplied + + ASTERISK-29714 + + Change-Id: I8548438246f7b718d88e0b9e0a1eb384bbec88e4 + +2021-10-30 20:04 +0000 [b9e888418e] Josh Soref + + * configs: Spelling fixes + + Correct typos of the following word families: + + password + excludes + undesirable + checksums + through + screening + interpreting + database + causes + initiation + member + busydetect + defined + severely + throughput + recognized + counter + require + indefinitely + accounts + + ASTERISK-29714 + + Change-Id: Ie8f2a7b274a162dd627ee6a2165f5e8a3876527e + +2021-10-30 20:04 +0000 [de6ab15e6a] Josh Soref + + * doc: Spelling fixes + + Correct typos of the following word families: + + transparent + roughly + + ASTERISK-29714 + + Change-Id: I2b90c68dfde4aa3f0d58f64f8187465336acb1b3 + +2021-10-30 20:04 +0000 [33a5c32bf6] Josh Soref + + * menuselect: Spelling fixes + + Correct typos of the following word families: + + dependency + unless + random + dependencies + delimited + randomly + modules + + ASTERISK-29714 + + Change-Id: I3920603a8dc7c0a1852d2f885e06b1144692d40e + +2021-10-30 20:04 +0000 [5d3a115bee] Josh Soref + + * include: Spelling fixes + + Correct typos of the following word families: + + activities + forward + occurs + unprepared + association + compress + extracted + doubly + callback + prometheus + underlying + keyframe + continue + convenience + calculates + ignorepattern + determine + subscribers + subsystem + synthetic + applies + example + manager + established + result + microseconds + occurrences + unsuccessful + accommodates + related + signifying + unsubscribe + greater + fastforward + itself + unregistering + using + translator + sorcery + implementation + serializers + asynchronous + unknowingly + initialization + determining + category + these + persistent + propagate + outputted + string + allocated + decremented + second + cacheability + destructor + impaired + decrypted + relies + signaling + based + suspended + retrieved + functions + search + auth + considered + + ASTERISK-29714 + + Change-Id: I542ce887a16603f886a915920d5710d4a0a1358d + +2021-10-30 20:04 +0000 [83a2e76671] Josh Soref + + * UPGRADE.txt: Spelling fixes + + Correct typos of the following word families: + + themselves + support + received + + ASTERISK-29714 + + Change-Id: Ibd0a7996d5801c754d3d44fba31fe788a13dba95 + +2021-10-30 20:04 +0000 [2a8fb4695e] Josh Soref + + * bridges: Spelling fixes + + Correct typos of the following word families: + + multiplication + potentially + iteration + interaction + virtual + synthesis + convolve + initializes + overlap + + ASTERISK-29714 + + Change-Id: Ia40f1aca8f2996ab407c6ed9d24cb10a67c6684b + +2021-10-30 20:04 +0000 [eb03b18ff9] Josh Soref + + * apps: Spelling fixes + + Correct typos of the following word families: + + simultaneously + administrator + directforward + attachfmt + dailplan + automatically + applicable + nouns + explicit + outside + sponsored + attachment + audio + spied + doesn't + counting + encoded + implements + recursively + emailaddress + arguments + queuerules + members + priority + output + advanced + silencethreshold + brazilian + debugging + argument + meadmin + formatting + integrated + sneakiness + + ASTERISK-29714 + + Change-Id: Ie5ecaec91c00b26309da4e51cfc0991a5bb7d092 + +2021-10-30 20:04 +0000 [d46ba42910] Josh Soref + + * channels: Spelling fixes + + Correct typos of the following word families: + + appease + permanently + overriding + residue + silliness + extension + channels + globally + reference + japanese + group + coordinate + registry + information + inconvenience + attempts + cadence + payloads + presence + provisioning + mimics + behavior + width + natively + syslabel + not owning + unquelch + mostly + constants + interesting + active + unequipped + brodmann + commanding + backlogged + without + bitstream + firmware + maintain + exclusive + practically + structs + appearance + range + retransmission + indication + provisional + associating + always + whether + cyrillic + distinctive + components + reinitialized + initialized + capability + switches + occurring + happened + outbound + + ASTERISK-29714 + + Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339 + +2021-10-30 20:04 +0000 [e54a9d31f1] Josh Soref + + * tests: Spelling fixes + + Correct typos of the following word families: + + mounting + jitterbuffer + thrashing + original + manipulating + entries + actual + possibility + tasks + options + positives + taskprocessor + other + dynamic + declarative + + ASTERISK-29714 + + Change-Id: I6b94659d045eec5d8d020fce2e9b6e2f593dfeb6 + +2021-10-30 20:04 +0000 [3bf314d643] Josh Soref + + * CHANGES: Spelling fixes + + Correct typos of the following word families: + + issuing + execution + bridging + alert + respective + unlikely + confbridge + offered + negotiation + announced + engineer + systems + inherited + passthrough + functionality + supporting + conflicts + semantically + monitor + specify + specifiable + + ASTERISK-29714 + + Change-Id: Ia6b1cf634f52c5f7b1b8769dc54dae78106ed98c + +2021-10-30 20:04 +0000 [1b1f5f9f67] Josh Soref + + * funcs: Spelling fixes + + Correct typos of the following word families: + + effectively + emitted + expect + anthony + + ASTERISK-29714 + + Change-Id: Ic16f9ec855bb6d14ec8e170b90af9a36b06d488a + +2021-10-30 20:04 +0000 [ccb8b8ffbf] Josh Soref + + * pbx: Spelling fixes + + Correct typos of the following word families: + + process + populate + with + africa + accessing + contexts + exercise + university + organizations + withhold + maintaining + independent + rotation + ignore + eventname + + ASTERISK-29714 + + Change-Id: I90eacc5bc3dcf75a9c898cfb85164f37dec08345 + +2021-10-30 20:04 +0000 [f382775241] Josh Soref + + * main: Spelling fixes + + Correct typos of the following word families: + + analysis + nuisance + converting + although + transaction + desctitle + acquire + update + evaluate + thousand + this + dissolved + management + integrity + reconstructed + decrement + further on + irrelevant + currently + constancy + anyway + unconstrained + featuregroups + right + larger + evaluated + encumbered + languages + digits + authoritative + framing + blindxfer + tolerate + traverser + exclamation + perform + permissions + rearrangement + performing + processing + declension + happily + duplicate + compound + hundred + returns + elicit + allocate + actually + paths + inheritance + atxferdropcall + earlier + synchronization + multiplier + acknowledge + across + against + thousands + joyous + manipulators + guaranteed + emulating + soundfile + + ASTERISK-29714 + + Change-Id: I926ba4b11e9f6dd3fdd93170ab1f9b997910be70 + +2021-10-30 20:04 +0000 [15c4814f55] Josh Soref + + * utils: Spelling fixes + + Correct typos of the following word families: + + command-line + immediately + extensions + momentarily + mustn't + numbered + bytes + caching + + ASTERISK-29714 + + Change-Id: I8b2b125c5d4d2f9e87a58515c97468ad47ca44f8 + +2021-10-30 20:04 +0000 [4490f0b962] Josh Soref + + * Makefile: Spelling fixes + + Correct typos of the following word families: + + libraries + install + overwrite + + ASTERISK-29714 + + Change-Id: I6488814f79186d6c23dfd7b7f9bba0a046126174 + +2021-10-30 20:04 +0000 [9ae9893c63] Josh Soref + + * res: Spelling fixes + + Correct typos of the following word families: + + identifying + structures + actcount + initializer + attributes + statement + enough + locking + declaration + userevent + provides + unregister + session + execute + searches + verification + suppressed + prepared + passwords + recipients + event + because + brief + unidentified + redundancy + character + the + module + reload + operation + backslashes + accurate + incorrect + collision + initializing + instance + interpreted + buddies + omitted + manually + requires + queries + generator + scheduler + configuration has + owner + resource + performed + masquerade + apparently + routable + + ASTERISK-29714 + + Change-Id: I88485116d2c59b776aa2e1f8b4ce8239a21decda + +2021-10-30 20:04 +0000 [ff11d74331] Josh Soref + + * rest-api-templates: Spelling fixes + + Correct typos of the following word families: + + overwritten + descendants + + ASTERISK-29714 + + Change-Id: I2307e35887a3437e50317a4b86f0893f25f9fd3b + +2021-10-30 20:04 +0000 [9641d15039] Josh Soref + + * agi: Spelling fixes + + Correct typos of the following word families: + + pretend + speech + + ASTERISK-29714 + + Change-Id: I7d0527c329cda07552247ea11b2d7db207a3d87d + +2021-11-08 07:00 +0000 [f1f23bbe4e] George Joseph + + * CI: Rename 'master' node to 'built-in' + + Jenkins renamed the 'master' node to 'built-in' in version + 2.319 so we have to adjust as well. + + Change-Id: Ice663c3a66d0eedf76e8e5fe530328455991ec25 + +2021-11-08 08:08 +0000 [b8db1daec6] Alexander Traud + + * BuildSystem: In POSIX sh, == in place of = is undefined. + + ASTERISK-29724 + + Change-Id: I59aa0e52effdc16992f3a736ccf73430a6ef135b + +2021-11-08 09:01 +0000 [a109b5aee0] Sean Bright + + * pbx.c: Don't remove dashes from hints on reload. + + When reloading dialplan, hints created dynamically would lose any dash + characters. Now we ignore those dashes if we are dealing with a hint + during a reload. + + ASTERISK-28040 #close + + Change-Id: I95e48f5a268efa3c6840ab69798525d3dce91636 + +2021-10-24 06:55 +0000 [f9ba1ee7c9] Naveen Albert + + * sig_analog: Fix truncated buffer copy + + Fixes compiler warning caused by a truncated copy of the ANI2 into a + buffer of size 10. This could prevent the null terminator from being + copied if the copy value exceeds the size of the buffer. This increases + the buffer size to 101 to ensure there is no way for truncation to occur. + + ASTERISK-29702 #close + + Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0 + +2021-10-24 07:31 +0000 [4e514419d9] Naveen Albert + + * app_voicemail: Fix phantom voicemail bug on rerecord + + If users are able to press # for options while leaving + a message and then press 3 to rerecord the message, if + the caller hangs up during the rerecord prompt but before + Asterisk starts recording a message, then an "empty" + voicemail gets processed whereby an email gets sent out + notifying the user of a 0:00 duration message. The file + doesn't actually exist, so playback will fail since there + was no message to begin with. + + This adds a check after the streaming of the rerecord + announcement to see if the caller has hung up. If so, + we bail out early so that we can clean up properly. + + ASTERISK-29391 #close + + Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31 + +2021-10-25 19:47 +0000 [df9aeea4c8] Naveen Albert + + * chan_iax2: Allow both secret and outkey at dial time + + Historically, the dial syntax for IAX2 has held that + an outkey (used only for RSA authenticated calls) + and a secret (used only for plain text and MD5 authenticated + calls, historically) were mutually exclusive, and thus + the same position in the dial string was used for both + values. + + Now that encryption is possible with RSA authentication, + this poses a limitation, since encryption requires a + secret and RSA authentication requires an outkey. Thus, + the dial syntax is extended so that both a secret and + an outkey can be specified. + + The new extended syntax is backwards compatible with the + old syntax. However, a secret can now be specified after + the outkey, or the outkey can be specified after the secret. + This makes it possible to spawn an encrypted RSA authenticated + call without a corresponding peer being predefined in iax.conf. + + ASTERISK-29707 #close + + Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e + +2021-10-28 07:09 +0000 [d1163653d1] Alexander Traud + + * res_snmp: As build tool, prefer pkg-config over net-snmp-config. + + ASTERISK-29709 + + Change-Id: Ie169df878bdfc3a06b3097c5c38d185b480f54d4 + +2021-11-04 05:22 +0000 [ee0ed3ae49] Alexander Traud + + * res_config_sqlite: Remove deprecated module. + + ASTERISK-29717 + + Change-Id: I64b914eef744542528f7d4396bd06715898fbc55 + +2021-10-28 07:41 +0000 [14709ae12d] Alexander Traud + + * stasis: Avoid 'dispatched' as unused variable in normal mode. + + ASTERISK-29710 + + Change-Id: Ia849f1172e4e694c5d5d7f0cad449f936ee12216 + +2021-10-29 10:05 +0000 [ce2d743d59] Sean Bright + + * various: Fix GCC 11.2 compilation issues. + + * Initialize some variables that are never used anyway. + + * Use valid pointers instead of integers cast to void pointers when + calling pthread_setspecific(). + + ASTERISK-29711 #close + ASTERISK-29713 #close + + Change-Id: I8728cd6f2f4b28e0e48113c5da450b768c2a6683 + +2021-09-09 09:39 +0000 [8aea2e5929] George Joseph + + * ast_coredumper: Refactor to better find things + + The search for a running asterisk when --running is used + has been greatly simplified and in the event it doesn't + work, you can now specify a pid to use on the command + line with --pid. + + The search for asterisk modules when --tarball-coredumps + is used has been enhanced to have a better chance of finding + them and in the event it doesn't work, you can now specify + --libdir on the command line to indicate the library directory + where they were installed. + + The DATEFORMAT variable was renamed to DATEOPTS and is now + passed to the 'date' utility rather than running DATEFORMAT + as a command. + + The coredump and output files are now renamed with DATEOPTS. + This can be disabled by specifying --no-rename. + + Several confusing and conflicting options were removed: + --append-coredumps + --conffile + --no-default-search + --tarball-uniqueid + + The script was re-structured to make it easier for follow. + + Change-Id: I674be64bdde3ef310b6a551d4911c3b600ffee59 + +2021-10-21 12:29 +0000 [67d1f881eb] Kevin Harwell + + * strings/json: Add string delimter match, and object create with vars methods + + Add a function to check if there is an exact match a one string between + delimiters in another string. + + Add a function that will create an ast_json object out of a list of + Asterisk variables. An excludes string can also optionally be passed + in. + + Also, add a macro to make it easier to get object integers. + + Change-Id: I5f34f18e102126aef3997f19a553a266d70d6226 + +2021-09-21 12:09 +0000 [1031a1805b] Ben Ford + + * STIR/SHAKEN: Option split and response codes. + + The stir_shaken configuration option now has 4 different choices to pick + from: off, attest, verify, and on. Off and on behave the same way they + do now. Attest will only perform attestation on the endpoint, and verify + will only perform verification on the endpoint. + + Certain responses are required to be sent based on certain conditions + for STIR/SHAKEN. For example, if we get a Date header that is outside of + the time range that is considered valid, a 403 Stale Date response + should be sent. This and several other responses have been added. + + Change-Id: I4ac1ecf652cd0e336006b0ca638dc826b5b1ebf7 + +2021-08-25 08:15 +0000 [56ecf7005b] Rodrigo Ramírez Norambuena + + * app_queue: Add LoginTime field for member in a queue. + + Add a time_t logintime to storage a time when a member is added into a + queue. + + Also, includes show this time (in seconds) using a 'queue show' command + and the field LoginTime for response for AMI events. + + ASTERISK-18069 #close + + Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190 + +2021-10-21 12:49 +0000 [8beac820c0] Kevin Harwell + + * res_speech: Add a type conversion, and new engine unregister methods + + Add a new function that converts a speech results type to a string. + Also add another function to unregister an engine, but returns a + pointer to the unregistered engine object instead of a success/fail + integer. + + Change-Id: I0f7de17cb411021c09fb03988bc2b904e1380192 + +2021-10-07 13:07 +0000 [99a1a427a9] Mike Bradeen + + * various: Fix GCC 11 compilation issues. + + test_voicemail_api: Use empty char* for empty_msg_ids. + chan_skinny: Fix size of calledParty to be maximum extension. + menuselect: Change Makefile to stop deprecated warnings. Added comments + test_linkedlist: 'bogus' variable was manually allocated from a macro + and the test fails if this happens but the compiler couldn't 'see' this + and returns a warning. memset to all 0's after allocation. + chan_ooh323: Fixed various indentation issues that triggered misleading + indentation warnings. + + ASTERISK-29682 + Reported by: George Joseph + + Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe + +2021-09-20 11:10 +0000 [cfae5224e3] Shloime Rosenblum + + * apps/app_playback.c: Add 'mix' option to app_playback + + I am adding a mix option that will play by filename and say.conf unlike + say option that will only play with say.conf. It + will look on the format of the name, if it is like say it play with + say.conf if not it will play the file name. + + ASTERISK-29662 + + Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a + +2021-10-19 11:35 +0000 [0adcdbd118] George Joseph + + * BuildSystem: Check for alternate openssl packages + + OpenSSL is one of those packages that often have alternatives + with later versions. For instance, CentOS/EL 7 has an + openssl package at version 1.0.2 but there's an openssl11 + package from the epel repository that has 1.1.1. This gets + installed to /usr/include/openssl11 and /usr/lib64/openssl11. + Unfortunately, the existing --with-ssl and --with-crypto + ./configure options expect to point to a source tree and + don't work in this situation. Also unfortunately, the + checks in ./configure don't use pkg-config. + + In order to make this work with the existing situation, you'd + have to run... + ./configure --with-ssl=/usr/lib64/openssl11 \ + --with-crypto=/usr/lib64/openssl11 \ + CFLAGS=-I/usr/include/openssl11 + + BUT... those options don't get passed down to bundled pjproject + so when you run make, you have to include the CFLAGS again + which is a big pain. + + Oh... To make matters worse, although you can specify + PJPROJECT_CONFIGURE_OPTS on the ./configure command line, + they don't get saved so if you do a make clean, which will + force a re-configure of bundled pjproject, those options + don't get used. + + So... + + * In configure.ac... Since pkg-config is installed by install_prereq + anyway, we now use it to check for the system openssl >= 1.1.0. + If that works, great. If not, we check for the openssl11 + package. If that works, great. If not, we fall back to just + checking for any openssl. If pkg-config isn't installed for some + reason, or --with-ssl= or --with-crypto= were specified + on the ./configure command line, we fall back to the existing + logic that uses AST_EXT_LIB_CHECK(). + + * The whole OpenSSL check process has been moved up before + THIRD_PARTY_CONFIGURE(), which does the initial pjproject + bundled configure, is run. This way the results of the above + checks, which may result in new include or library directories, + is included. + + * Although not strictly needed for openssl, We now save the value of + PJPROJECT_CONFIGURE_OPTS in the makeopts file so it can be used + again if a re-configure is triggered. + + ASTERISK-29693 + + Change-Id: I341ab7603e6b156aa15a66f43675ac5029d5fbde + +2021-10-14 14:38 +0000 [886983b114] Sean Bright + + * func_talkdetect.c: Fix logical errors in silence detection. + + There are 3 separate changes here: + + 1. The documentation erroneously stated that the dsp_talking_threshold + argument was a number of milliseconds when it is actually an energy + level used by the DSP code to classify talking vs. silence. + + 2. Fixes a copy paste error in the argument handling code. + + 3. Don't erroneously switch to the talking state if we aren't actively + handling a frame we've classified as talking. + + Patch inspired by one provided by Moritz Fain (License #6961). + + ASTERISK-27816 #close + + Change-Id: I5953fd570b98b49c41cee55bfe3b941753fb2511 + +2021-10-11 14:04 +0000 [44fd75fae2] Sean Bright + + * configure: Remove unused OpenSSL SRTP check. + + Discovered while looking at ASTERISK~29684. Usage was removed in change + I3c77c7b00b2ffa2e935632097fa057b9fdf480c0. + + Change-Id: Iaf2f7a16ea5a7eee6375319347e4b40b8e7b10e3 + +2021-10-12 13:17 +0000 [072f2ebb12] Mike Bradeen + + * build: prevent binary downloads for non x86 architectures + + download_externals: Add check for i686 and i386 (in addition + to the current x86_64) and exit if not one of the three. + + ASTERISK-26497 + + Change-Id: Ia4d429fcefa5b2f5b6e99159d4607de8e8325b2f + +2021-10-14 10:15 +0000 [51859252f7] Sebastien Duthil + + * main/stun.c: fix crash upon STUN request timeout + + Some ast_stun_request users do not provide a destination address when + sending to a connection-mode socket. + + ASTERISK-29691 + + Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc + +2021-10-07 12:50 +0000 [9fcd50a8c9] Sean Bright + + * Makefile: Use basename in a POSIX-compliant way. + + If you aren't using GNU coreutils, chances are that your basename + doesn't know about the -s argument. Luckily for us, basename does what + we need it do even without the -s argument. + + Change-Id: I8b81a429bb037b997ee6640ff8a2b5e860962bb7 + +2021-10-05 19:59 +0000 [7fc26e8617] Mark Murawski + + * pbx_ael: Fix crash and lockup issue regarding 'ael reload' + + Avoid infinite recursion and crash + + Change-Id: I8ed05ec3aa2806c50c77edc5dd0cd4e4fa08b3f4 + +2021-05-24 13:04 +0000 [7ff6c43760] Naveen Albert + + * chan_iax2: Add encryption for RSA authentication + + Adds support for encryption to RSA-authenticated + calls. Also prevents crashes if an RSA IAX2 call + is initiated to a switch requiring encryption + but no secret is provided. + + ASTERISK-20219 + + Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40 + +2021-07-19 11:34 +0000 [5e9799a42e] Matthew Kern + + * res_pjsip_t38: bind UDPTL sessions like RTP + + In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the + fallback use of the transport's bind address solve problems sending + media on systems that cannot send ipv4 packets on ipv6 sockets, and + certain other situations. This change extends both of these behaviors + to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific + problems on these systems, introducing a new option + endpoint/t38_bind_udptl_to_media_address. + + ASTERISK-29402 + + Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557 + +2021-09-29 12:58 +0000 [b40ca38c56] Naveen Albert + + * app_read: Fix null pointer crash + + If the terminator character is not explicitly specified + and an indications tone is used for reading a digit, + there is no null pointer check so Asterisk crashes. + This prevents null usage from occuring. + + ASTERISK-29673 #close + + Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac + +2021-09-29 04:32 +0000 [6bc747b639] Jean Aunis + + * res_rtp_asterisk: fix memory leak + + Add missing reference decrement in rtp_deallocate_transport() + + ASTERISK-29671 + + Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9 + +2021-09-19 15:08 +0000 [d20587250e] Shloime Rosenblum + + * main/say.c: Support future dates with Q and q format params + + The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today + + ASTERISK-29637 + + Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02 + +2021-07-21 16:36 +0000 [47cb177baf] Joseph Nadiv + + * res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts + + The behavior of max_contacts and remove_existing are connected. If + remove_existing is enabled, the soonest expiring contacts are removed. + This may occur when there is an unavailable contact. Similarly, + when remove_existing is not enabled, registrations from good + endpoints are rejected in favor of retaining unavailable contacts. + + This commit adds a new AOR option remove_unavailable, and the effect + of this setting will depend on remove_existing. If remove_existing + is set to no, we will still remove unavailable contacts when they + exceed max_contacts, if there are any. If remove_existing is set to + yes, we will prioritize the removal of unavailable contacts before + those that are expiring soonest. + + ASTERISK-29525 + + Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784 + +2021-09-23 09:13 +0000 [0aac38c0ac] Joshua C. Colp + + * ari: Ignore invisible bridges when listing bridges. + + When listing bridges we go through the ones present in + ARI, get their snapshot, turn it into JSON, and add it + to the payload we ultimately return. + + An invisible "dial bridge" exists within ARI that would + also try to be added to this payload if the channel + "create" and "dial" routes were used. This would ultimately + fail due to invisible bridges having no snapshot + resulting in the listing of bridges failing. + + This change makes it so that the listing of bridges + ignores invisible ones. + + ASTERISK-29668 + + Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a + +2021-09-19 06:14 +0000 [d900130021] Naveen Albert + + * func_vmcount: Add support for multiple mailboxes + + Allows multiple mailboxes to be specified for VMCOUNT + instead of just one. + + ASTERISK-29661 #close + + Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813 + +2021-09-21 09:58 +0000 [5ca9898dfb] Sean Bright + + * message.c: Support 'To' header override with AMI's MessageSend. + + The MessageSend AMI action has been updated to allow the Destination + and the To addresses to be provided separately. This brings the + MessageSend manager command in line with the capabilities of the + MessageSend dialplan application. + + ASTERISK-29663 #close + + Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c + +2021-09-15 13:21 +0000 [de6ecd5e34] Naveen Albert + + * func_channel: Add CHANNEL_EXISTS function. + + Adds a function to check for the existence of a channel by + name or by UNIQUEID. + + ASTERISK-29656 #close + + Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb + +2021-09-05 13:11 +0000 [5abf499d23] Naveen Albert + + * app_queue: Fix hint updates for included contexts + + Previously, if custom hints were used with the hint: + format in app_queue, when device state changes occured, + app_queue would only do a literal string comparison of + the context used for the hint in app_queue and the context + of the hint which just changed state. This caused hints + to not update and become stale if the context associated + with the agent included the context which actually changes + state, essentially completely breaking device state for + any such agents defined in this manner. + + This fix adds an additional check to ensure that included + contexts are also compared against the context which changed + state, so that the behavior is correct no matter whether the + context is specified to app_queue directly or indirectly. + + ASTERISK-29578 #close + + Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78 + +2021-09-10 09:40 +0000 [02f54e2751] Sean Bright + + * res_http_media_cache.c: Compare unaltered MIME types. + + Rather than stripping parameters from Content-Type headers before + comparison, first try to compare the whole string. If no match is + found, strip the parameters and try that way. + + ASTERISK-29275 #close + + Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f + +2021-07-25 17:19 +0000 [148f8355a0] Naveen Albert + + * logger: Add custom logging capabilities + + Adds the ability for users to log to custom log levels + by providing custom log level names in logger.conf. Also + adds a logger show levels CLI command. + + ASTERISK-29529 + + Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702 + +2021-09-17 10:57 +0000 [6698753b24] Sean Bright + + * app_externalivr.c: Fix mixed leading whitespace in source code. + + No functional changes. + + Change-Id: I46514152c0af67f395526374aaa847ccd6a85378 + +2021-09-17 14:58 +0000 [29ad5b18f1] Guido Falsi + + * res_rtp_asterisk.c: Fix build failure when not building with pjproject. + + Some code has been added referencing symbols defined in a block + protected by #ifdef HAVE_PJPROJECT. Protect those code parts in + ifdef blocks too. + + ASTERISK-29660 + + Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f + +2021-09-14 12:02 +0000 [54a9dbb2b8] George Joseph + + * pjproject: Add patch to fix trailing whitespace issue in rtpmap + + An issue was found where a particular manufacturer's phones add a + trailing space to the end of the rtpmap attribute when specifying + a payload type that has a "param" after the format name and clock + rate. For example: + + a=rtpmap:120 opus/48000/2 \r\n + + Because pjmedia_sdp_attr_get_rtpmap currently takes everything after + the second '/' up to the line end as the param, the space is + included in future comparisons, which then fail if the param being + compared to doesn't also have the space. + + We now use pj_scan_get() to parse the param part of rtpmap so + trailing whitespace is automatically stripped. + + ASTERISK-29654 + + Change-Id: Ibd0a4e243a69cde7ba9312275b13ab62ab86bc1b + +2021-09-13 10:18 +0000 [07c297d058] Carlos Oliva + + * app_mp3: Force output to 16 bits in mpg123 + + In new mpg123 versions (since 1.26) the default output is 32 bits + Asterisk expects the output in 16 bits, so we force the output to be on 16 bits. + It will work wit new and old versions of mpg123. + Thanks Thomas Orgis for giving the key! + + ASTERISK-29635 #close + + Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816 + +2021-06-08 15:44 +0000 [5b5c358e4b] Naveen Albert + + * res_pjsip_caller_id: Add ANI2/OLI parsing + + Adds parsing of ANI II digits (Originating + Line Information) to PJSIP, on par with + what currently exists in chan_sip. + + ASTERISK-29472 + + Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847 + +2021-06-28 10:37 +0000 [b760bad2b9] Naveen Albert + + * app_mf: Add channel agnostic MF sender + + Adds a SendMF application and PlayMF manager + event to send arbitrary R1 MF tones on the + current or specified channel. + + ASTERISK-29496 + + Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4 + +2021-09-02 18:20 +0000 [18c92353f8] Naveen Albert + + * app_stack: Include current location if branch fails + + Previously, the error emitted when app_stack tries + to branch to a dialplan location that doesn't exist + has included only the information about the attempted + branch in the error log. This adds the current location + as well so users can see where the branch failed in + the logs. + + ASTERISK-29626 + + Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce + +2021-09-10 09:56 +0000 [46afd61b75] Sean Bright + + * test_http_media_cache.c: Fix copy/paste error during test deregistration. + + Change-Id: I9a3a978b2f818be464e062d97b93831b127ef28c + +2021-09-03 13:27 +0000 [a1fa8df0ae] Sungtae Kim + + * resource_channels.c: Fix external media data option + + Fixed the external media creation handle to handle the 'data' option correctly. + + ASTERISK-29629 + + Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129 + +2021-09-02 18:57 +0000 [b8fc77a35b] Naveen Albert + + * func_strings: Add STRBETWEEN function + + Adds the STRBETWEEN function, which can be used to insert a + substring between each character in a string. For instance, + this can be used to insert pauses between DTMF tones in a + string of digits. + + ASTERISK-29627 + + Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258 + +2021-09-08 14:29 +0000 [c4037d4aa3] Sean Bright + + * test_abstract_jb.c: Fix put and put_out_of_order memory leaks. + + We can't rely on RAII_VAR(...) to properly clean up data that is + allocated within a loop. + + ASTERISK-27176 #close + + Change-Id: Ib575616101230c4f603519114ec62ebf3936882c + +2021-09-02 19:00 +0000 [e0111a56fa] Naveen Albert + + * func_env: Add DIRNAME and BASENAME functions + + Adds the DIRNAME and BASENAME functions, which are + wrappers around the corresponding C library functions. + These can be used to safely and conveniently work with + file paths and names in the dialplan. + + ASTERISK-29628 #close + + Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3 + +2021-07-26 12:46 +0000 [ddf6299b8d] Naveen Albert + + * func_sayfiles: Retrieve say file names + + Up until now, all of the logic used to translate + arguments to the Say applications has been + directly coupled to playback, preventing other + modules from using this logic. + + This refactors code in say.c and adds a SAYFILES + function that can be used to retrieve the file + names that would be played. These can then be + used in other applications or for other purposes. + + Additionally, a SayMoney application and a SayOrdinal + application are added. Both SayOrdinal and SayNumber + are also expanded to support integers greater than + one billion. + + ASTERISK-29531 + + Change-Id: If9718c89353b8e153d84add3cc4637b79585db19 + +2021-08-09 12:41 +0000 [7df69633cf] Naveen Albert + + * res_tonedetect: Tone detection module + + dsp.c contains arbitrary tone detection functionality + which is currently only used for fax tone recognition. + This change makes this functionality publicly + accessible so that other modules can take advantage + of this. + + Additionally, a WaitForTone and TONE_DETECT app and + function are included to allow users to do their + own tone detection operations in the dialplan. + + ASTERISK-29546 + + Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26 + +2021-09-08 09:36 +0000 [448962d056] George Joseph + + * res_snmp: Add -fPIC to _ASTCFLAGS + + With gcc 11, res/res_snmp.c and res/snmp/agent.c need the + -fPIC option added to its _ASTCFLAGS. + + ASTERISK-29634 + + Change-Id: I34649c85e075fd954e578378fabf798c3f038f50 + +2021-09-07 12:32 +0000 [26fc5f3c72] Sean Bright + + * app_voicemail.c: Ability to silence instructions if greeting is present. + + There is an option to silence voicemail instructions but it does not + take into consideration if a recorded greeting exists or not. Add a + new 'S' option that does that. + + ASTERISK-29632 #close + + Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4 + +2021-09-04 12:07 +0000 [605dd03b36] Sean Bright + + * term.c: Add support for extended number format terminfo files. + + ncurses 6.1 introduced an extended number format for terminfo files + which the terminfo parsing in Asterisk is not able to parse. This + results in some TERM values that do support color (screen-256color on + Ubuntu 20.04 for example) to not get a color console. + + ASTERISK-29630 #close + + Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3 + +2021-09-03 00:30 +0000 [c07d531191] Jasper Hafkenscheid + + * res_srtp: Disable parsing of not enabled cryptos + + When compiled without extended srtp crypto suites also disable parsing + these from received SDP. This prevents using these, as some client + implementations are not stable. + + ASTERISK-29625 + + Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a + +2021-09-06 11:37 +0000 [695fc3dbd7] Sean Bright + + * dns.c: Load IPv6 DNS resolvers if configured. + + IPv6 nameserver addresses are stored in different part of the + __res_state structure, so look there if we appear to have support for + it. + + ASTERISK-28004 #close + + Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d + +2021-09-08 07:52 +0000 [976521c9a2] George Joseph + + * bridge_softmix: Suppress error on topology change failure + + There are conditions under which a failure to change topology + is expected so there's no need to print an ERROR message. + + ASTERISK-29618 + Reported by: Alexander + + Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481 + +2021-08-31 02:50 +0000 [79d6d222d6] sungtae kim + + * resource_channels.c: Fix wrong external media parameter parse + + Fixed ARI external media handler to accept body parameters. + + ASTERISK-29622 + + Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d + +2021-08-25 10:21 +0000 [5029e78f39] Sean Bright + + * config_options: Handle ACO arrays correctly in generated XML docs. + + There are 3 separate changes here but they are all closely related: + + * Only try to set matchfield attributes on 'field' nodes + + * We need to adjust how we treat the category pointer based on the + value of the category_match, to avoid memory corruption. We now + generate a regex-like string when match types other than + ACO_WHITELIST and ACO_BLACKLIST are used. + + * Switch app_agent_pool from ACO_BLACKLIST_ARRAY to + ACO_BLACKLIST_EXACT since we only have one category we need to + ignore, not two. + + ASTERISK-29614 #close + + Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e + +2021-08-18 14:44 +0000 [3072c540bb] Naveen Albert + + * chan_iax2: Add ANI2/OLI information element + + Adds an information element for ANI2 so that + Originating Line Information can be transmitted + over IAX2 channels. + + ASTERISK-29605 #close + + Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2 + +2021-08-31 15:03 +0000 [bbf4f30059] Mark Murawski + + * pbx_ael: Fix crash and lockup issue regarding 'ael reload' + + Currently pbx_ael does not check if a reload is currently pending + before proceeding with a reload. This can cause multiple threads to + operate at the same time on what should be mutex protected data. This + change adds protection to reloading to ensure only one ael reload is + executing at a time. + + ASTERISK-29609 #close + + Change-Id: I5ed392ad226f6e4e7696ad742076d3e45c57af35 + +2021-08-25 06:49 +0000 [6cc004dc5a] Naveen Albert + + * app_read: Allow reading # as a digit + + Allows for the digit # to be read as a digit, + just like any other DTMF digit, as opposed to + forcing it to be used as an end of input + indicator. The default behavior remains + unchanged. + + ASTERISK-18454 #close + + Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b + +2021-04-05 14:06 +0000 [6fbf55ac11] Sebastien Duthil + + * res_rtp_asterisk: Automatically refresh stunaddr from DNS + + This allows the STUN server to change its IP address without having to + reload the res_rtp_asterisk module. + + The refresh of the name resolution occurs first when the module is + loaded, then recurringly, slightly after the previous DNS answer TTL + expires. + + ASTERISK-29508 #close + + Change-Id: I7955a046293f913ba121bbd82153b04439e3465f + +2021-08-24 20:04 +0000 [f01a0398f8] Naveen Albert + + * bridge_basic: Change warning to verbose if transfer cancelled + + The attended transfer feature will emit a warning if the user + cancels the transfer or the attended transfer doesn't complete + for any reason. Changes the warning to a verbose message, + since nothing is actually wrong here. + + ASTERISK-29612 #close + + Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d + +2021-08-20 15:35 +0000 [92f9ae32a8] Naveen Albert + + * app_queue: Don't reset queue stats on reload + + Prevents reloads of app_queue from also resetting + queue statistics. + + Also preserves individual queue agent statistics + if we're just reloading members. + + ASTERISK-28701 + + Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1 + +2021-08-25 09:23 +0000 [63d27af3ca] Alexander Traud + + * res_rtp_asterisk: sqrt(.) requires the header math.h. + + ASTERISK-29616 + + Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900 + +2021-08-25 09:29 +0000 [fbdd8a7f8a] Alexander Traud + + * dialplan: Add one static and fix two whitespace errors. + + Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83 + +2021-06-19 23:36 +0000 [466eb4a52b] Sarah Autumn + + * sig_analog: Changes to improve electromechanical signalling compatibility + + This changeset is intended to address compatibility issues encountered + when interfacing Asterisk to electromechanical telephone switches that + implement ANI-B, ANI-C, or ANI-D. + + In particular the behaviours that this impacts include: + + - FGC-CAMA did not work at all when using MF signaling. Modified the + switch case block to send calls to the correct part of the + signaling-handling state machine. + + - For FGC-CAMA operation, the delay between called number ST and + second wink for ANI spill has been made configurable; previously + all calls were made to wait for one full second. + + - After the ANI spill, previous behavior was to require a 'ST' tone + to advance the call. This has been changed to allow 'STP' 'ST2P' + or 'ST3P' as well, for compatibility with ANI-D. + + - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. + + - For calls with an ANI failure, No. 1 Crossbar switches will send + forward a single-digit failure code, with no calling number digits + and no ST pulse to terminate the spill. I've made the ANI timeout + configurable so to reduce dead air time on calls with ANI fail. + + - ANI info digits configurable. Modern digital switches will send 2 + digits, but ANI-B sends only a single info digit. This caused the + ANI reported by Asterisk to be misaligned. + + - Changed a confusing log message to be more informative. + + ASTERISK-29518 + + Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256 + +2021-08-05 11:55 +0000 [c4839c04b6] Andre Barbosa + + * media_cache: Don't lock when curl the remote file + + When playing a remote sound file, which is not in cache, first we need + to download it with ast_bucket_file_retrieve. + + This can take a while if the remote host is slow. The current CURL + timeout is 180secs, so in extreme situations, it can take 3 minutes to + return. + + Because ast_media_cache_retrieve has a lock on all function, while we + are waiting for the delayed download, Asterisk is not able to play any + more files, even the files already cached locally. + + ASTERISK-29544 #close + + Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408 + +2021-08-16 08:25 +0000 [84f2bf4307] George Joseph + + * res_pjproject: Allow mapping to Asterisk TRACE level + + Allow mapping pjproject log messages to the Asterisk TRACE + log level. The defaults were also changes to log pjproject + levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6 + all went to DEBUG. + + ASTERISK-29582 + + Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d + +2021-08-12 16:02 +0000 [314d8776dc] Naveen Albert + + * app_milliwatt: Timing fix + + The Milliwatt application uses incorrect tone timings + that cause it to play the 1004 Hz tone constantly. + + This adds an option to enable the correct timing + behavior, so that the Milliwatt application can + be used for milliwatt test lines. The default behavior + remains unchanged for compatability reasons, even + though it is incorrect. + + ASTERISK-29575 #close + + Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c + +2021-06-28 09:25 +0000 [85ef06d300] Naveen Albert + + * func_math: Return integer instead of float if possible + + The MIN, MAX, and ABS functions all support float + arguments, but currently return floats even if the + arguments are all integers and the response is + a whole number, in which case the user is likely + expecting an integer. This casts the float to an integer + before printing into the response buffer if possible. + + ASTERISK-29495 + + Change-Id: I902d29eacf3ecd0f8a6a5e433c97f0421d205488 + +2021-08-04 09:46 +0000 [5c9d7a0373] Naveen Albert + + * app_morsecode: Add American Morse code + + Previously, the Morsecode application only supported international + Morse code. This adds support for American Morse code and adds an + option to configure the frequency used in off intervals. + + Additionally, the application checks for hangup between tones + to prevent application execution from continuing after hangup. + + ASTERISK-29541 + + Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4 + +2021-08-04 14:16 +0000 [498db70884] Naveen Albert + + * func_scramble: Audio scrambler function + + Adds a function to scramble audio on a channel using + whole spectrum frequency inversion. This can be used + as a privacy enhancement with applications like + ChanSpy or other potentially sensitive audio. + + ASTERISK-29542 + + Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e + +2021-08-04 19:28 +0000 [a099f13a20] Naveen Albert + + * app_originate: Add ability to set codecs + + A list of codecs to use for dialplan-originated calls can + now be specified in Originate, similar to the ability + in call files and the manager action. + + Additionally, we now default to just using the slin codec + for originated calls, rather than all the slin* codecs up + through slin192, which has been known to cause issues + and inconsistencies from AMI and call file behavior. + + ASTERISK-29543 + + Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883 + +2021-08-16 11:11 +0000 [137bd7fe65] Alexander Traud + + * BuildSystem: Remove two dead exceptions for compiler Clang. + + Commit 305ce3d added -Wno-parentheses-equality to Makefile.rules, + turning the previous two warning suppressions from commit e9520db + redundant. Let us remove the latter. + + Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac + +2021-08-16 14:31 +0000 [0ca3ebe7cd] Naveen Albert + + * chan_alsa, chan_sip: Add replacement to moduleinfo + + Adds replacement modules to the moduleinfo for + chan_alsa and chan_sip. + + ASTERISK-29601 #close + + Change-Id: I7a4877b0d5c0c17e088e8fa8ebbfa9a195223cbc + +2021-08-17 08:11 +0000 [0ddeac0e36] Joshua C. Colp + + * res_monitor: Disable building by default. + + ASTERISK-29602 + + Change-Id: I6f0af0a959409cdbc6b185b1604301bafc872a5a + +2021-08-16 13:47 +0000 [fcbf0a6699] Joshua C. Colp + + * muted: Remove deprecated application. + + ASTERISK-29600 + + Change-Id: I0ae1c6a2996da43217126f094de90761314dcf82 + +2021-08-16 13:39 +0000 [6d5b66f5f3] Joshua C. Colp + + * conf2ael: Remove deprecated application. + + ASTERISK-29599 + + Change-Id: I75dc77162926fb17e7c6caf8f04e3aabd792fb0c + +2021-08-16 13:26 +0000 [800fd84af6] Joshua C. Colp + + * res_config_sqlite: Remove deprecated module. + + ASTERISK-29598 + + Change-Id: I8ef17023f55bf01f2e309b06f4778a8ca7252c91 + +2021-08-16 13:22 +0000 [20b2741232] Joshua C. Colp + + * chan_vpb: Remove deprecated module. + + ASTERISK-29597 + + Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1 + +2021-08-16 13:18 +0000 [1eb2d85c99] Joshua C. Colp + + * chan_misdn: Remove deprecated module. + + ASTERISK-29596 + + Change-Id: Ibae9490c1b35cadbf7028d24610f745277c8535e + +2021-08-16 13:13 +0000 [6ecc48086c] Joshua C. Colp + + * chan_nbs: Remove deprecated module. + + ASTERISK-29595 + + Change-Id: Ib5c7d43a780f2fb94cee90738e4c1af211ae4a33 + +2021-08-16 13:10 +0000 [6cc948f94e] Joshua C. Colp + + * chan_phone: Remove deprecated module. + + ASTERISK-29594 + + Change-Id: I79a9961cb5062fadbccb0ea93f087bdd32685316 + +2021-08-16 13:06 +0000 [95f3a4a9ad] Joshua C. Colp + + * chan_oss: Remove deprecated module. + + ASTERISK-29593 + + Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99 + +2021-08-16 13:04 +0000 [30d5264409] Joshua C. Colp + + * cdr_syslog: Remove deprecated module. + + ASTERISK-29592 + + Change-Id: Ic8eb6a2100ad5bc3b48338a6d0a6cfa70ecbc50f + +2021-08-16 12:56 +0000 [9e5269c7ae] Joshua C. Colp + + * app_dahdiras: Remove deprecated module. + + ASTERISK-29591 + + Change-Id: I021d37b729631d40f84e35bb21e2893777be1858 + +2021-08-16 12:55 +0000 [98e0745a14] Joshua C. Colp + + * app_nbscat: Remove deprecated module. + + ASTERISK-29590 + + Change-Id: I87cf0f536b77d222c8eda003376ac47fae86ed43 + +2021-08-16 12:52 +0000 [13963e643b] Joshua C. Colp + + * app_image: Remove deprecated module. + + ASTERISK-29589 + + Change-Id: I8057eb2ca1ca4c3b27ed2fe04bea10e9cb551cdd + +2021-08-16 12:50 +0000 [7c642c55b8] Joshua C. Colp + + * app_url: Remove deprecated module. + + ASTERISK-29588 + + Change-Id: If846d40b37c5b646bcd7326111db280529a5971b + +2021-08-16 12:48 +0000 [24e21e59af] Joshua C. Colp + + * app_fax: Remove deprecated module. + + ASTERISK-29587 + + Change-Id: I038237bbb56b1161d7d5e20cda11ed32e13d3ca2 + +2021-08-16 12:46 +0000 [1f1a87a97b] Joshua C. Colp + + * app_ices: Remove deprecated module. + + ASTERISK-29586 + + Change-Id: I1e0a4535135b00938b609fe0ccba9bbddbac93ad + +2021-08-16 12:43 +0000 [2f510d7a88] Joshua C. Colp + + * app_mysql: Remove deprecated module. + + ASTERISK-29585 + + Change-Id: I262930d0387d043f2a3345e8a977b314528059bf + +2021-08-16 12:39 +0000 [2a0e383e4f] Joshua C. Colp + + * cdr_mysql: Remove deprecated module. + + ASTERISK-29584 + + Change-Id: I4bd3695d089121f810d692a82361d39d2f97ae39 + +2021-08-10 12:41 +0000 [743e057bb4] Sean Bright + + * mgcp: Remove dead debug code + + ASTERISK-20339 #close + + Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5 + +2021-08-11 06:15 +0000 [93870e7bb4] Joshua C. Colp + + * policy: Deprecate modules and add versions to others. + + app_meetme is deprecated in 19, to be removed in 21. + app_osplookup is deprecated in 19, to be removed in 21. + chan_alsa is deprecated in 19, to be removed in 21. + chan_mgcp is deprecated in 19, to be removed in 21. + chan_skinny is deprecated in 19, to be removed in 21. + res_pktccops is deprecated in 19, to be removed in 21. + app_macro was deprecated in 16, to be removed in 21. + chan_sip was deprecated in 17, to be removed in 21. + res_monitor was deprecated in 16, to be removed in 21. + + ASTERISK-29548 + ASTERISK-29549 + ASTERISK-29550 + ASTERISK-29551 + ASTERISK-29552 + ASTERISK-29553 + ASTERISK-29558 + ASTERISK-29567 + ASTERISK-29572 + + Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131 + +2021-06-16 15:30 +0000 [6a89266b5b] Naveen Albert + + * func_frame_drop: New function + + Adds function to selectively drop specified frames + in the TX or RX direction on a channel, including + control frames. + + ASTERISK-29478 + + Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec + +2021-08-02 12:33 +0000 [8a6c9c3a76] Alexander Traud + + * aelparse: Accept an included context with timings. + + With Asterisk 1.6.0, in the main parser for the configuration file + extensions.conf, the separator was changed from vertical bar to comma. + However, the first separator was not changed in aelparse; it still had + to be a vertical bar, and no comma was allowed. + + Additionally, this change allows the vertical bar for the first and + last parameter again, even in the main parser, because the vertical bar + was still accepted for the other parameters. + + ASTERISK-29540 + + Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81 + +2021-08-03 11:30 +0000 [049c7c1361] Kevin Harwell + + * format_ogg_speex: Implement a "not supported" write handler + + This format did not specify a "write" handler, so when attempting to write + to it (ast_writestream) a crash would occur. + + This patch adds a default handler that simply issues a "not supported" + warning, thus no longer crashing. + + ASTERISK-29539 + + Change-Id: I8f6ddc7cc3b15da30803be3b1cf68e2ba0fbce91 + +2021-06-28 08:48 +0000 [b5709e610e] Naveen Albert + + * cdr_adaptive_odbc: Prevent filter warnings + + Previously, if CDR filters were used so that + not all CDR records used all sections defined + in cdr_adaptive_odbc.conf, then warnings will + always be emitted (if each CDR record is unique + to a particular section, n-1 warnings to be + specific). + + This turns the offending warning log into + a verbose message like the other one, since + this behavior is intentional and not + indicative of anything wrong. + + ASTERISK-29494 + + Change-Id: Ifd314fa9298722bc99494d5ca2658a5caa94a5f8 + +2021-07-25 16:53 +0000 [0e023e6cf1] Naveen Albert + + * app_queue: Allow streaming multiple announcement files + + Allows multiple files comprising an agent announcement + to be played by separating on the ampersand, similar + to the multi-file support in other Asterisk applications. + + ASTERISK-29528 + + Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a + +2021-04-13 02:36 +0000 [4f437ea1f4] Igor Goncharovsky + + * res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern + + PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request. + It may be used to get all X- headers in case the actual set and names of headers unknown. + + ASTERISK-29389 + + Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b + +2021-07-08 07:34 +0000 [728a52fb61] Rijnhard Hessel + + * res_statsd: handle non-standard meter type safely + + Meter types are not well supported, + lacking support in telegraf, datadog and the official statsd servers. + We deprecate meters and provide a compliant fallback for any existing usages. + + A flag has been introduced to allow meters to fallback to counters. + + + ASTERISK-29513 + + Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7 + +2021-06-16 15:26 +0000 [fa7d147e1b] Naveen Albert + + * app_dtmfstore: New application to store digits + + Adds application to asynchronously collect digits + dialed on a channel in the TX or RX direction + using a framehook and stores them in a specified + variable, up to a configurable number of digits. + + ASTERISK-29477 + + Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f + +2021-07-22 11:39 +0000 [de3f5350de] under + + * codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother + + If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps. + This makes the audio stream not-playable at the receiver side. + Linphone isn't able to play such an audio - lots of disruptions are heard. + Also I had complains of bad audio from users which use other types of phones. + + After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c). + + Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c). + + However, this flag is never set in Asterisk-12 and newer. + Previously it has been set (see Asterisk-11). + + ASTERISK-29526 #close + + Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d + +2021-07-23 11:00 +0000 [6428124b06] Sean Bright + + * res_http_media_cache: Cleanup audio format lookup in HTTP requests + + Asterisk first looks at the end of the URL to determine the file + extension of the returned audio, which in many cases will not work + because the URL may end with a query string or a URL fragment. If that + fails, Asterisk then looks at the Content-Type header and then finally + parses the URL to get the extension. + + The order has been changed such that we look at the Content-Type + header first, followed by looking for the extension of the parsed + URL. We no longer look at the end of the URL, which was error prone. + + ASTERISK-29527 #close + + Change-Id: I1e3f83b339ef2b80661704717c23568536511032 + +2021-07-27 07:53 +0000 [d0f189a5c9] Joshua C. Colp + + * docs: Remove embedded macro in WaitForCond XML documentation. + + Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c + +2021-07-21 10:14 +0000 [db7b025532] Ben Ford + + * Update AMI and ARI versions for Asterisk 20. + + Bumped AMI and ARI versions for the next major Asterisk version (20). + + Change-Id: I2e65794f206d443178ab6895767fb53f04cc3e6a + +2021-06-14 13:28 +0000 [e8cda4b32c] Kevin Harwell + + * AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS + + If an SSL socket parent/listener was destroyed during the handshake, + depending on timing, it was possible for the handling callback to + attempt access of it after the fact thus causing a crash. + + ASTERISK-29415 #close + + Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d + +2021-05-10 17:59 +0000 [1b62831f2c] Kevin Harwell + + * AST-2021-008 - chan_iax2: remote crash on unsupported media format + + If chan_iax2 received a packet with an unsupported media format, for + example vp9, then it would set the frame's format to NULL. This could + then result in a crash later when an attempt was made to access the + format. + + This patch makes it so chan_iax2 now ignores/drops frames received + with unsupported media format types. + + ASTERISK-29392 #close + + Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1 + +2021-04-28 07:36 +0000 [ec16d2ecbd] Joshua C. Colp + + * AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. + + If a re-INVITE is received after we have sent a BYE request then it + is possible for no channel to be present on the session. If this + occurs we allow PJSIP to produce the offer instead. Since the call + is being hung up if it produces an incorrect offer it doesn't + actually matter. This also ensures that code which produces SDP + does not need to handle if a channel is not present. + + ASTERISK-29381 + + Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042 + +2021-07-21 09:59 +0000 [e6ddbe0922] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 19.0.0 +2021-10-13 10:44 +0000 Asterisk Development Team + + * asterisk 19.0.0-rc1 Released. + +2021-10-13 05:21 +0000 [9ff955f4d1] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 19.0.0 +2021-10-07 12:50 +0000 [9175012a12] Sean Bright + + * Makefile: Use basename in a POSIX-compliant way. + + If you aren't using GNU coreutils, chances are that your basename + doesn't know about the -s argument. Luckily for us, basename does what + we need it do even without the -s argument. + + Change-Id: I8b81a429bb037b997ee6640ff8a2b5e860962bb7 + +2021-10-05 19:59 +0000 [1f5ac24fa3] Mark Murawski + + * pbx_ael: Fix crash and lockup issue regarding 'ael reload' + + Avoid infinite recursion and crash + + Change-Id: I8ed05ec3aa2806c50c77edc5dd0cd4e4fa08b3f4 + +2021-05-24 13:04 +0000 [32ea7c7ca5] Naveen Albert + + * chan_iax2: Add encryption for RSA authentication + + Adds support for encryption to RSA-authenticated + calls. Also prevents crashes if an RSA IAX2 call + is initiated to a switch requiring encryption + but no secret is provided. + + ASTERISK-20219 + + Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40 + +2021-07-19 11:34 +0000 [9d04535bbd] Matthew Kern + + * res_pjsip_t38: bind UDPTL sessions like RTP + + In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the + fallback use of the transport's bind address solve problems sending + media on systems that cannot send ipv4 packets on ipv6 sockets, and + certain other situations. This change extends both of these behaviors + to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific + problems on these systems, introducing a new option + endpoint/t38_bind_udptl_to_media_address. + + ASTERISK-29402 + + Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557 + +2021-09-29 12:58 +0000 [60bbfe4572] Naveen Albert + + * app_read: Fix null pointer crash + + If the terminator character is not explicitly specified + and an indications tone is used for reading a digit, + there is no null pointer check so Asterisk crashes. + This prevents null usage from occuring. + + ASTERISK-29673 #close + + Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac + +2021-09-29 04:32 +0000 [576119e076] Jean Aunis + + * res_rtp_asterisk: fix memory leak + + Add missing reference decrement in rtp_deallocate_transport() + + ASTERISK-29671 + + Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9 + +2021-09-19 15:08 +0000 [f3ff893310] Shloime Rosenblum + + * main/say.c: Support future dates with Q and q format params + + The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today + + ASTERISK-29637 + + Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02 + +2021-07-21 16:36 +0000 [6a04c43035] Joseph Nadiv + + * res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts + + The behavior of max_contacts and remove_existing are connected. If + remove_existing is enabled, the soonest expiring contacts are removed. + This may occur when there is an unavailable contact. Similarly, + when remove_existing is not enabled, registrations from good + endpoints are rejected in favor of retaining unavailable contacts. + + This commit adds a new AOR option remove_unavailable, and the effect + of this setting will depend on remove_existing. If remove_existing + is set to no, we will still remove unavailable contacts when they + exceed max_contacts, if there are any. If remove_existing is set to + yes, we will prioritize the removal of unavailable contacts before + those that are expiring soonest. + + ASTERISK-29525 + + Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784 + +2021-09-23 09:13 +0000 [35a94ec708] Joshua C. Colp + + * ari: Ignore invisible bridges when listing bridges. + + When listing bridges we go through the ones present in + ARI, get their snapshot, turn it into JSON, and add it + to the payload we ultimately return. + + An invisible "dial bridge" exists within ARI that would + also try to be added to this payload if the channel + "create" and "dial" routes were used. This would ultimately + fail due to invisible bridges having no snapshot + resulting in the listing of bridges failing. + + This change makes it so that the listing of bridges + ignores invisible ones. + + ASTERISK-29668 + + Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a + +2021-09-19 06:14 +0000 [13ec117595] Naveen Albert + + * func_vmcount: Add support for multiple mailboxes + + Allows multiple mailboxes to be specified for VMCOUNT + instead of just one. + + ASTERISK-29661 #close + + Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813 + +2021-09-21 09:58 +0000 [52b5821694] Sean Bright + + * message.c: Support 'To' header override with AMI's MessageSend. + + The MessageSend AMI action has been updated to allow the Destination + and the To addresses to be provided separately. This brings the + MessageSend manager command in line with the capabilities of the + MessageSend dialplan application. + + ASTERISK-29663 #close + + Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c + +2021-09-15 13:21 +0000 [f38c7d67d3] Naveen Albert + + * func_channel: Add CHANNEL_EXISTS function. + + Adds a function to check for the existence of a channel by + name or by UNIQUEID. + + ASTERISK-29656 #close + + Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb + +2021-09-05 13:11 +0000 [eff78c8549] Naveen Albert + + * app_queue: Fix hint updates for included contexts + + Previously, if custom hints were used with the hint: + format in app_queue, when device state changes occured, + app_queue would only do a literal string comparison of + the context used for the hint in app_queue and the context + of the hint which just changed state. This caused hints + to not update and become stale if the context associated + with the agent included the context which actually changes + state, essentially completely breaking device state for + any such agents defined in this manner. + + This fix adds an additional check to ensure that included + contexts are also compared against the context which changed + state, so that the behavior is correct no matter whether the + context is specified to app_queue directly or indirectly. + + ASTERISK-29578 #close + + Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78 + +2021-09-10 09:40 +0000 [ff493d6f7d] Sean Bright + + * res_http_media_cache.c: Compare unaltered MIME types. + + Rather than stripping parameters from Content-Type headers before + comparison, first try to compare the whole string. If no match is + found, strip the parameters and try that way. + + ASTERISK-29275 #close + + Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f + +2021-07-25 17:19 +0000 [eb874f92db] Naveen Albert + + * logger: Add custom logging capabilities + + Adds the ability for users to log to custom log levels + by providing custom log level names in logger.conf. Also + adds a logger show levels CLI command. + + ASTERISK-29529 + + Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702 + +2021-09-17 10:57 +0000 [245778a756] Sean Bright + + * app_externalivr.c: Fix mixed leading whitespace in source code. + + No functional changes. + + Change-Id: I46514152c0af67f395526374aaa847ccd6a85378 + +2021-09-17 14:58 +0000 [675adbf0f5] Guido Falsi + + * res_rtp_asterisk.c: Fix build failure when not building with pjproject. + + Some code has been added referencing symbols defined in a block + protected by #ifdef HAVE_PJPROJECT. Protect those code parts in + ifdef blocks too. + + ASTERISK-29660 + + Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f + +2021-09-14 12:02 +0000 [3d6e133ccf] George Joseph + + * pjproject: Add patch to fix trailing whitespace issue in rtpmap + + An issue was found where a particular manufacturer's phones add a + trailing space to the end of the rtpmap attribute when specifying + a payload type that has a "param" after the format name and clock + rate. For example: + + a=rtpmap:120 opus/48000/2 \r\n + + Because pjmedia_sdp_attr_get_rtpmap currently takes everything after + the second '/' up to the line end as the param, the space is + included in future comparisons, which then fail if the param being + compared to doesn't also have the space. + + We now use pj_scan_get() to parse the param part of rtpmap so + trailing whitespace is automatically stripped. + + ASTERISK-29654 + + Change-Id: Ibd0a4e243a69cde7ba9312275b13ab62ab86bc1b + +2021-09-13 10:18 +0000 [ad1f7fae70] Carlos Oliva + + * app_mp3: Force output to 16 bits in mpg123 + + In new mpg123 versions (since 1.26) the default output is 32 bits + Asterisk expects the output in 16 bits, so we force the output to be on 16 bits. + It will work wit new and old versions of mpg123. + Thanks Thomas Orgis for giving the key! + + ASTERISK-29635 #close + + Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816 + +2021-06-28 10:37 +0000 [203e73f5af] Naveen Albert + + * app_mf: Add channel agnostic MF sender + + Adds a SendMF application and PlayMF manager + event to send arbitrary R1 MF tones on the + current or specified channel. + + ASTERISK-29496 + + Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4 + +2021-06-08 15:44 +0000 [f8bf5e7b47] Naveen Albert + + * res_pjsip_caller_id: Add ANI2/OLI parsing + + Adds parsing of ANI II digits (Originating + Line Information) to PJSIP, on par with + what currently exists in chan_sip. + + ASTERISK-29472 + + Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847 + +2021-09-02 18:20 +0000 [5fe3a745e4] Naveen Albert + + * app_stack: Include current location if branch fails + + Previously, the error emitted when app_stack tries + to branch to a dialplan location that doesn't exist + has included only the information about the attempted + branch in the error log. This adds the current location + as well so users can see where the branch failed in + the logs. + + ASTERISK-29626 + + Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce + +2021-09-10 09:56 +0000 [f26505d615] Sean Bright + + * test_http_media_cache.c: Fix copy/paste error during test deregistration. + + Change-Id: I9a3a978b2f818be464e062d97b93831b127ef28c + +2021-09-02 18:57 +0000 [d5a53efb4f] Naveen Albert + + * func_strings: Add STRBETWEEN function + + Adds the STRBETWEEN function, which can be used to insert a + substring between each character in a string. For instance, + this can be used to insert pauses between DTMF tones in a + string of digits. + + ASTERISK-29627 + + Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258 + +2021-09-03 13:27 +0000 [4d9ba65c53] Sungtae Kim + + * resource_channels.c: Fix external media data option + + Fixed the external media creation handle to handle the 'data' option correctly. + + ASTERISK-29629 + + Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129 + +2021-09-08 14:29 +0000 [085cc94f16] Sean Bright + + * test_abstract_jb.c: Fix put and put_out_of_order memory leaks. + + We can't rely on RAII_VAR(...) to properly clean up data that is + allocated within a loop. + + ASTERISK-27176 #close + + Change-Id: Ib575616101230c4f603519114ec62ebf3936882c + +2021-09-02 19:00 +0000 [71b021433f] Naveen Albert + + * func_env: Add DIRNAME and BASENAME functions + + Adds the DIRNAME and BASENAME functions, which are + wrappers around the corresponding C library functions. + These can be used to safely and conveniently work with + file paths and names in the dialplan. + + ASTERISK-29628 #close + + Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3 + +2021-07-26 12:46 +0000 [0b8ae58e67] Naveen Albert + + * func_sayfiles: Retrieve say file names + + Up until now, all of the logic used to translate + arguments to the Say applications has been + directly coupled to playback, preventing other + modules from using this logic. + + This refactors code in say.c and adds a SAYFILES + function that can be used to retrieve the file + names that would be played. These can then be + used in other applications or for other purposes. + + Additionally, a SayMoney application and a SayOrdinal + application are added. Both SayOrdinal and SayNumber + are also expanded to support integers greater than + one billion. + + ASTERISK-29531 + + Change-Id: If9718c89353b8e153d84add3cc4637b79585db19 + +2021-08-09 12:41 +0000 [a94b51ee60] Naveen Albert + + * res_tonedetect: Tone detection module + + dsp.c contains arbitrary tone detection functionality + which is currently only used for fax tone recognition. + This change makes this functionality publicly + accessible so that other modules can take advantage + of this. + + Additionally, a WaitForTone and TONE_DETECT app and + function are included to allow users to do their + own tone detection operations in the dialplan. + + ASTERISK-29546 + + Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26 + +2021-09-08 09:36 +0000 [df63a99337] George Joseph + + * res_snmp: Add -fPIC to _ASTCFLAGS + + With gcc 11, res/res_snmp.c and res/snmp/agent.c need the + -fPIC option added to its _ASTCFLAGS. + + ASTERISK-29634 + + Change-Id: I34649c85e075fd954e578378fabf798c3f038f50 + +2021-09-04 12:07 +0000 [61136fd297] Sean Bright + + * term.c: Add support for extended number format terminfo files. + + ncurses 6.1 introduced an extended number format for terminfo files + which the terminfo parsing in Asterisk is not able to parse. This + results in some TERM values that do support color (screen-256color on + Ubuntu 20.04 for example) to not get a color console. + + ASTERISK-29630 #close + + Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3 + +2021-09-07 12:32 +0000 [f67b72093e] Sean Bright + + * app_voicemail.c: Ability to silence instructions if greeting is present. + + There is an option to silence voicemail instructions but it does not + take into consideration if a recorded greeting exists or not. Add a + new 'S' option that does that. + + ASTERISK-29632 #close + + Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4 + +2021-09-03 00:30 +0000 [f1e1f9f37f] Jasper Hafkenscheid + + * res_srtp: Disable parsing of not enabled cryptos + + When compiled without extended srtp crypto suites also disable parsing + these from received SDP. This prevents using these, as some client + implementations are not stable. + + ASTERISK-29625 + + Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a + +2021-09-06 11:37 +0000 [5a5ea06ffc] Sean Bright + + * dns.c: Load IPv6 DNS resolvers if configured. + + IPv6 nameserver addresses are stored in different part of the + __res_state structure, so look there if we appear to have support for + it. + + ASTERISK-28004 #close + + Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d + +2021-09-08 07:52 +0000 [0070b9184c] George Joseph + + * bridge_softmix: Suppress error on topology change failure + + There are conditions under which a failure to change topology + is expected so there's no need to print an ERROR message. + + ASTERISK-29618 + Reported by: Alexander + + Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481 + +2021-08-31 02:50 +0000 [3c31b6aaa2] sungtae kim + + * resource_channels.c: Fix wrong external media parameter parse + + Fixed ARI external media handler to accept body parameters. + + ASTERISK-29622 + + Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d + +2021-08-25 10:21 +0000 [16b0f460f6] Sean Bright + + * config_options: Handle ACO arrays correctly in generated XML docs. + + There are 3 separate changes here but they are all closely related: + + * Only try to set matchfield attributes on 'field' nodes + + * We need to adjust how we treat the category pointer based on the + value of the category_match, to avoid memory corruption. We now + generate a regex-like string when match types other than + ACO_WHITELIST and ACO_BLACKLIST are used. + + * Switch app_agent_pool from ACO_BLACKLIST_ARRAY to + ACO_BLACKLIST_EXACT since we only have one category we need to + ignore, not two. + + ASTERISK-29614 #close + + Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e + +2021-08-18 14:44 +0000 [29770520b3] Naveen Albert + + * chan_iax2: Add ANI2/OLI information element + + Adds an information element for ANI2 so that + Originating Line Information can be transmitted + over IAX2 channels. + + ASTERISK-29605 #close + + Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2 + +2021-08-31 15:03 +0000 [185321066f] Mark Murawski + + * pbx_ael: Fix crash and lockup issue regarding 'ael reload' + + Currently pbx_ael does not check if a reload is currently pending + before proceeding with a reload. This can cause multiple threads to + operate at the same time on what should be mutex protected data. This + change adds protection to reloading to ensure only one ael reload is + executing at a time. + + ASTERISK-29609 #close + + Change-Id: I5ed392ad226f6e4e7696ad742076d3e45c57af35 + +2021-08-25 06:49 +0000 [0e4a1c5079] Naveen Albert + + * app_read: Allow reading # as a digit + + Allows for the digit # to be read as a digit, + just like any other DTMF digit, as opposed to + forcing it to be used as an end of input + indicator. The default behavior remains + unchanged. + + ASTERISK-18454 #close + + Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b + +2021-04-05 14:06 +0000 [18189ff594] Sebastien Duthil + + * res_rtp_asterisk: Automatically refresh stunaddr from DNS + + This allows the STUN server to change its IP address without having to + reload the res_rtp_asterisk module. + + The refresh of the name resolution occurs first when the module is + loaded, then recurringly, slightly after the previous DNS answer TTL + expires. + + ASTERISK-29508 #close + + Change-Id: I7955a046293f913ba121bbd82153b04439e3465f + +2021-08-24 20:04 +0000 [4301fe20d1] Naveen Albert + + * bridge_basic: Change warning to verbose if transfer cancelled + + The attended transfer feature will emit a warning if the user + cancels the transfer or the attended transfer doesn't complete + for any reason. Changes the warning to a verbose message, + since nothing is actually wrong here. + + ASTERISK-29612 #close + + Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d + +2021-08-20 15:35 +0000 [9e947b0463] Naveen Albert + + * app_queue: Don't reset queue stats on reload + + Prevents reloads of app_queue from also resetting + queue statistics. + + Also preserves individual queue agent statistics + if we're just reloading members. + + ASTERISK-28701 + + Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1 + +2021-08-25 09:29 +0000 [f22b413ece] Alexander Traud + + * dialplan: Add one static and fix two whitespace errors. + + Change-Id: Ia14d515ab63e773097adc6af772ca7123a392f83 + +2021-08-25 09:23 +0000 [e65e1c5c6c] Alexander Traud + + * res_rtp_asterisk: sqrt(.) requires the header math.h. + + ASTERISK-29616 + + Change-Id: I6c01623926bf10ccac32612687a50fdab3ba0900 + +2021-06-19 23:36 +0000 [db4a3b117d] Sarah Autumn + + * sig_analog: Changes to improve electromechanical signalling compatibility + + This changeset is intended to address compatibility issues encountered + when interfacing Asterisk to electromechanical telephone switches that + implement ANI-B, ANI-C, or ANI-D. + + In particular the behaviours that this impacts include: + + - FGC-CAMA did not work at all when using MF signaling. Modified the + switch case block to send calls to the correct part of the + signaling-handling state machine. + + - For FGC-CAMA operation, the delay between called number ST and + second wink for ANI spill has been made configurable; previously + all calls were made to wait for one full second. + + - After the ANI spill, previous behavior was to require a 'ST' tone + to advance the call. This has been changed to allow 'STP' 'ST2P' + or 'ST3P' as well, for compatibility with ANI-D. + + - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. + + - For calls with an ANI failure, No. 1 Crossbar switches will send + forward a single-digit failure code, with no calling number digits + and no ST pulse to terminate the spill. I've made the ANI timeout + configurable so to reduce dead air time on calls with ANI fail. + + - ANI info digits configurable. Modern digital switches will send 2 + digits, but ANI-B sends only a single info digit. This caused the + ANI reported by Asterisk to be misaligned. + + - Changed a confusing log message to be more informative. + + ASTERISK-29518 + + Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256 + +2021-08-16 08:25 +0000 [a662d75556] George Joseph + + * res_pjproject: Allow mapping to Asterisk TRACE level + + Allow mapping pjproject log messages to the Asterisk TRACE + log level. The defaults were also changes to log pjproject + levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6 + all went to DEBUG. + + ASTERISK-29582 + + Change-Id: I859a37a8dec263ed68099709cfbd3e665324c72d + +2021-08-05 11:55 +0000 [2451dfd89f] Andre Barbosa + + * media_cache: Don't lock when curl the remote file + + When playing a remote sound file, which is not in cache, first we need + to download it with ast_bucket_file_retrieve. + + This can take a while if the remote host is slow. The current CURL + timeout is 180secs, so in extreme situations, it can take 3 minutes to + return. + + Because ast_media_cache_retrieve has a lock on all function, while we + are waiting for the delayed download, Asterisk is not able to play any + more files, even the files already cached locally. + + ASTERISK-29544 #close + + Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408 + +2021-08-04 14:16 +0000 [e01a6c026d] Naveen Albert + + * func_scramble: Audio scrambler function + + Adds a function to scramble audio on a channel using + whole spectrum frequency inversion. This can be used + as a privacy enhancement with applications like + ChanSpy or other potentially sensitive audio. + + ASTERISK-29542 + + Change-Id: I01020769d91060a1f56a708eb405f87648d1a67e + +2021-06-28 09:25 +0000 [d6034df64a] Naveen Albert + + * func_math: Return integer instead of float if possible + + The MIN, MAX, and ABS functions all support float + arguments, but currently return floats even if the + arguments are all integers and the response is + a whole number, in which case the user is likely + expecting an integer. This casts the float to an integer + before printing into the response buffer if possible. + + ASTERISK-29495 + + Change-Id: I902d29eacf3ecd0f8a6a5e433c97f0421d205488 + +2021-08-04 09:46 +0000 [b5044586f7] Naveen Albert + + * app_morsecode: Add American Morse code + + Previously, the Morsecode application only supported international + Morse code. This adds support for American Morse code and adds an + option to configure the frequency used in off intervals. + + Additionally, the application checks for hangup between tones + to prevent application execution from continuing after hangup. + + ASTERISK-29541 + + Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4 + +2021-08-12 16:02 +0000 [3f9ef427b5] Naveen Albert + + * app_milliwatt: Timing fix + + The Milliwatt application uses incorrect tone timings + that cause it to play the 1004 Hz tone constantly. + + This adds an option to enable the correct timing + behavior, so that the Milliwatt application can + be used for milliwatt test lines. The default behavior + remains unchanged for compatability reasons, even + though it is incorrect. + + ASTERISK-29575 #close + + Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c + +2021-08-04 19:28 +0000 [2394757e55] Naveen Albert + + * app_originate: Add ability to set codecs + + A list of codecs to use for dialplan-originated calls can + now be specified in Originate, similar to the ability + in call files and the manager action. + + Additionally, we now default to just using the slin codec + for originated calls, rather than all the slin* codecs up + through slin192, which has been known to cause issues + and inconsistencies from AMI and call file behavior. + + ASTERISK-29543 + + Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883 + +2021-08-16 11:11 +0000 [73e2288db7] Alexander Traud + + * BuildSystem: Remove two dead exceptions for compiler Clang. + + Commit 305ce3d added -Wno-parentheses-equality to Makefile.rules, + turning the previous two warning suppressions from commit e9520db + redundant. Let us remove the latter. + + Change-Id: I0b471254b31e6e05902062761dded4b3e626c7ac + +2021-08-16 14:31 +0000 [432fe9dc2a] Naveen Albert + + * chan_alsa, chan_sip: Add replacement to moduleinfo + + Adds replacement modules to the moduleinfo for + chan_alsa and chan_sip. + + ASTERISK-29601 #close + + Change-Id: I7a4877b0d5c0c17e088e8fa8ebbfa9a195223cbc + +2021-08-17 08:11 +0000 [ecf699c325] Joshua C. Colp + + * res_monitor: Disable building by default. + + ASTERISK-29602 + + Change-Id: I6f0af0a959409cdbc6b185b1604301bafc872a5a + +2021-08-16 13:47 +0000 [daca793ad4] Joshua C. Colp + + * muted: Remove deprecated application. + + ASTERISK-29600 + + Change-Id: I0ae1c6a2996da43217126f094de90761314dcf82 + +2021-08-16 13:39 +0000 [650cf0b444] Joshua C. Colp + + * conf2ael: Remove deprecated application. + + ASTERISK-29599 + + Change-Id: I75dc77162926fb17e7c6caf8f04e3aabd792fb0c + +2021-08-16 13:26 +0000 [368aa47962] Joshua C. Colp + + * res_config_sqlite: Remove deprecated module. + + ASTERISK-29598 + + Change-Id: I8ef17023f55bf01f2e309b06f4778a8ca7252c91 + +2021-08-16 13:22 +0000 [9d5f55a5f3] Joshua C. Colp + + * chan_vpb: Remove deprecated module. + + ASTERISK-29597 + + Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1 + +2021-08-16 13:18 +0000 [72a2140a50] Joshua C. Colp + + * chan_misdn: Remove deprecated module. + + ASTERISK-29596 + + Change-Id: Ibae9490c1b35cadbf7028d24610f745277c8535e + +2021-08-16 13:13 +0000 [7b0d3d3550] Joshua C. Colp + + * chan_nbs: Remove deprecated module. + + ASTERISK-29595 + + Change-Id: Ib5c7d43a780f2fb94cee90738e4c1af211ae4a33 + +2021-08-16 13:10 +0000 [7361a52820] Joshua C. Colp + + * chan_phone: Remove deprecated module. + + ASTERISK-29594 + + Change-Id: I79a9961cb5062fadbccb0ea93f087bdd32685316 + +2021-08-16 13:06 +0000 [d0ad32c7cf] Joshua C. Colp + + * chan_oss: Remove deprecated module. + + ASTERISK-29593 + + Change-Id: Ib53a42ad974c63871344b95078c61c188e43da99 + +2021-08-16 13:04 +0000 [e4b6f24a1d] Joshua C. Colp + + * cdr_syslog: Remove deprecated module. + + ASTERISK-29592 + + Change-Id: Ic8eb6a2100ad5bc3b48338a6d0a6cfa70ecbc50f + +2021-08-16 12:56 +0000 [f18107f191] Joshua C. Colp + + * app_dahdiras: Remove deprecated module. + + ASTERISK-29591 + + Change-Id: I021d37b729631d40f84e35bb21e2893777be1858 + +2021-08-16 12:55 +0000 [b1e5b1874c] Joshua C. Colp + + * app_nbscat: Remove deprecated module. + + ASTERISK-29590 + + Change-Id: I87cf0f536b77d222c8eda003376ac47fae86ed43 + +2021-08-16 12:52 +0000 [7ee6fb0372] Joshua C. Colp + + * app_image: Remove deprecated module. + + ASTERISK-29589 + + Change-Id: I8057eb2ca1ca4c3b27ed2fe04bea10e9cb551cdd + +2021-08-16 12:50 +0000 [0b3a149001] Joshua C. Colp + + * app_url: Remove deprecated module. + + ASTERISK-29588 + + Change-Id: If846d40b37c5b646bcd7326111db280529a5971b + +2021-08-16 12:48 +0000 [41afcb9422] Joshua C. Colp + + * app_fax: Remove deprecated module. + + ASTERISK-29587 + + Change-Id: I038237bbb56b1161d7d5e20cda11ed32e13d3ca2 + +2021-08-16 12:46 +0000 [83cad340fc] Joshua C. Colp + + * app_ices: Remove deprecated module. + + ASTERISK-29586 + + Change-Id: I1e0a4535135b00938b609fe0ccba9bbddbac93ad + +2021-08-16 12:43 +0000 [1961a1b83e] Joshua C. Colp + + * app_mysql: Remove deprecated module. + + ASTERISK-29585 + + Change-Id: I262930d0387d043f2a3345e8a977b314528059bf + +2021-08-16 12:39 +0000 [3e07b1ff62] Joshua C. Colp + + * cdr_mysql: Remove deprecated module. + + ASTERISK-29584 + + Change-Id: I4bd3695d089121f810d692a82361d39d2f97ae39 + +2021-08-10 12:41 +0000 [41ed46f474] Sean Bright + + * mgcp: Remove dead debug code + + ASTERISK-20339 #close + + Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5 + +2021-08-11 06:15 +0000 [141dc519b0] Joshua C. Colp + + * policy: Deprecate modules and add versions to others. + + app_meetme is deprecated in 19, to be removed in 21. + app_osplookup is deprecated in 19, to be removed in 21. + chan_alsa is deprecated in 19, to be removed in 21. + chan_mgcp is deprecated in 19, to be removed in 21. + chan_skinny is deprecated in 19, to be removed in 21. + res_pktccops is deprecated in 19, to be removed in 21. + app_macro was deprecated in 16, to be removed in 21. + chan_sip was deprecated in 17, to be removed in 21. + res_monitor was deprecated in 16, to be removed in 21. + + ASTERISK-29548 + ASTERISK-29549 + ASTERISK-29550 + ASTERISK-29551 + ASTERISK-29552 + ASTERISK-29553 + ASTERISK-29558 + ASTERISK-29567 + ASTERISK-29572 + + Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131 + +2021-06-16 15:30 +0000 [7383f74dfc] Naveen Albert + + * func_frame_drop: New function + + Adds function to selectively drop specified frames + in the TX or RX direction on a channel, including + control frames. + + ASTERISK-29478 + + Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec + +2021-08-02 12:33 +0000 [835ab50724] Alexander Traud + + * aelparse: Accept an included context with timings. + + With Asterisk 1.6.0, in the main parser for the configuration file + extensions.conf, the separator was changed from vertical bar to comma. + However, the first separator was not changed in aelparse; it still had + to be a vertical bar, and no comma was allowed. + + Additionally, this change allows the vertical bar for the first and + last parameter again, even in the main parser, because the vertical bar + was still accepted for the other parameters. + + ASTERISK-29540 + + Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81 + +2021-08-03 11:30 +0000 [37f7d19c8c] Kevin Harwell + + * format_ogg_speex: Implement a "not supported" write handler + + This format did not specify a "write" handler, so when attempting to write + to it (ast_writestream) a crash would occur. + + This patch adds a default handler that simply issues a "not supported" + warning, thus no longer crashing. + + ASTERISK-29539 + + Change-Id: I8f6ddc7cc3b15da30803be3b1cf68e2ba0fbce91 + +2021-06-28 08:48 +0000 [4c49c84dee] Naveen Albert + + * cdr_adaptive_odbc: Prevent filter warnings + + Previously, if CDR filters were used so that + not all CDR records used all sections defined + in cdr_adaptive_odbc.conf, then warnings will + always be emitted (if each CDR record is unique + to a particular section, n-1 warnings to be + specific). + + This turns the offending warning log into + a verbose message like the other one, since + this behavior is intentional and not + indicative of anything wrong. + + ASTERISK-29494 + + Change-Id: Ifd314fa9298722bc99494d5ca2658a5caa94a5f8 + +2021-07-25 16:53 +0000 [0975cff6c0] Naveen Albert + + * app_queue: Allow streaming multiple announcement files + + Allows multiple files comprising an agent announcement + to be played by separating on the ampersand, similar + to the multi-file support in other Asterisk applications. + + ASTERISK-29528 + + Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a + +2021-04-13 02:36 +0000 [ac958b0f50] Igor Goncharovsky + + * res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern + + PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request. + It may be used to get all X- headers in case the actual set and names of headers unknown. + + ASTERISK-29389 + + Change-Id: Ic09d395de71a0021e0d6c5c29e1e19d689079f8b + +2021-07-08 07:34 +0000 [f13eef719c] Rijnhard Hessel + + * res_statsd: handle non-standard meter type safely + + Meter types are not well supported, + lacking support in telegraf, datadog and the official statsd servers. + We deprecate meters and provide a compliant fallback for any existing usages. + + A flag has been introduced to allow meters to fallback to counters. + + + ASTERISK-29513 + + Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7 + +2021-07-23 11:00 +0000 [382143e58e] Sean Bright + + * res_http_media_cache: Cleanup audio format lookup in HTTP requests + + Asterisk first looks at the end of the URL to determine the file + extension of the returned audio, which in many cases will not work + because the URL may end with a query string or a URL fragment. If that + fails, Asterisk then looks at the Content-Type header and then finally + parses the URL to get the extension. + + The order has been changed such that we look at the Content-Type + header first, followed by looking for the extension of the parsed + URL. We no longer look at the end of the URL, which was error prone. + + ASTERISK-29527 #close + + Change-Id: I1e3f83b339ef2b80661704717c23568536511032 + +2021-07-22 11:39 +0000 [ff8ca2c9f1] under + + * codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother + + If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps. + This makes the audio stream not-playable at the receiver side. + Linphone isn't able to play such an audio - lots of disruptions are heard. + Also I had complains of bad audio from users which use other types of phones. + + After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c). + + Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c). + + However, this flag is never set in Asterisk-12 and newer. + Previously it has been set (see Asterisk-11). + + ASTERISK-29526 #close + + Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d + +2021-06-16 15:26 +0000 [6645cf8d45] Naveen Albert + + * app_dtmfstore: New application to store digits + + Adds application to asynchronously collect digits + dialed on a channel in the TX or RX direction + using a framehook and stores them in a specified + variable, up to a configurable number of digits. + + ASTERISK-29477 + + Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f + +2021-07-27 07:53 +0000 [90c9c90b11] Joshua C. Colp + + * docs: Remove embedded macro in WaitForCond XML documentation. + + Change-Id: I40c6514e1843e320f3cbe0b2c70d4a98c0e35b9c + +2021-05-10 17:59 +0000 [56f9c28a50] Kevin Harwell + + * AST-2021-008 - chan_iax2: remote crash on unsupported media format + + If chan_iax2 received a packet with an unsupported media format, for + example vp9, then it would set the frame's format to NULL. This could + then result in a crash later when an attempt was made to access the + format. + + This patch makes it so chan_iax2 now ignores/drops frames received + with unsupported media format types. + + ASTERISK-29392 #close + + Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1 + (cherry picked from commit 2a141a58b61ba0ed91061e1acc2c1955e0160f73) + +2021-04-28 07:36 +0000 [45af7e9984] Joshua C. Colp + + * AST-2021-007 - res_pjsip_session: Don't offer if no channel exists. + + If a re-INVITE is received after we have sent a BYE request then it + is possible for no channel to be present on the session. If this + occurs we allow PJSIP to produce the offer instead. Since the call + is being hung up if it produces an incorrect offer it doesn't + actually matter. This also ensures that code which produces SDP + does not need to handle if a channel is not present. + + ASTERISK-29381 + + Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042 + (cherry picked from commit 523a79528932e63c6aaad2fffb3fa08427f8f920) + +2021-06-14 13:28 +0000 [151bdbc658] Kevin Harwell + + * AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS + + If an SSL socket parent/listener was destroyed during the handshake, + depending on timing, it was possible for the handling callback to + attempt access of it after the fact thus causing a crash. + + ASTERISK-29415 #close + + Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d + (cherry picked from commit 3025ef4f6e79730d35c4514bf9c6dc4be87fa532) + +2021-07-21 10:20 +0000 [0ac346ec47] Ben Ford + + * Update default branch for Asterisk 19. + + Changed default branch to correct version for Asterisk 19. + + Change-Id: I6244c8ac14b9c0eb8bdf07fe58db24dc95cb1086 + +2021-06-29 11:07 +0000 [f4d3f021f9] Andre Barbosa + + * res_stasis_playback: Check for chan hangup on play_on_channels + + Verify `ast_check_hangup` before looping to the next sound file. + If the call is already hangup we just break the cycle. + It also ensures that the PlaybackFinished event is sent if the call was hangup. + + This is also use-full when we are playing a big list of file for a channel that is hangup. + Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played. + + With the patch we just break the playback cycle when the chan is hangup. + + ASTERISK-29501 #close + + Change-Id: Ic4e1c01b974c9a1f2d9678c9d6b380bcfc69feb8 + +2021-07-02 10:15 +0000 [d5bb27a06f] Sean Bright + + * res_http_media_cache.c: Fix merge errors from 18 -> master + + ASTERISK-27871 #close + + Change-Id: I6624f2d3a57f76a89bb372ef54a124929a0338d7 + +2021-07-15 15:04 +0000 [237285a9a8] Sean Bright + + * res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup. + + From RFC 8225 Section 5.2.1: + + The "dest" claim is a JSON object with the claim name of "dest" + and MUST have at least one identity claim object. The "dest" + claim value is an array containing one or more identity claim JSON + objects representing the destination identities of any type + (currently "tn" or "uri"). If the "dest" claim value array + contains both "tn" and "uri" claim names, the JSON object should + list the "tn" array first and the "uri" array second. Within the + "tn" and "uri" arrays, the identity strings should be put in + lexicographical order, including the scheme-specific portion of + the URI characters. + + Additionally, make it clear that there was a failure to sign the JWT + payload and not necessarily a memory allocation failure. + + Change-Id: Ia8733b861aef6edfaa9c2136e97b447a01578dc9 + +2021-07-02 10:15 +0000 [d568326807] Sean Bright + + * res_http_media_cache.c: Parse media URLs to find extensions. + + Use cURL's URL parsing API, falling back to the urlparser library, to + parse playback URLs in order to find their file extensions. + + For backwards compatibility, we first look at the full URL, then at + any Content-Type header, and finally at just the path portion of the + URL. + + ASTERISK-27871 #close + + Change-Id: I16d0682f6d794be96539261b3e48f237909139cb + +2021-07-13 10:31 +0000 [785e4afc20] Sean Bright + + * main/cdr.c: Correct Party A selection. + + This appears to just have been a copy/paste error from 6258bbe7. Fix + suggested by Ross Beer in ASTERISK~29166. + + Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37 + +2021-06-30 17:15 +0000 [8a21d466ea] Sebastien Duthil + + * stun: Emit warning message when STUN request times out + + Without this message, it is not obvious that the reason is STUN timeout. + + ASTERISK-29507 #close + + Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74 + +2021-05-26 12:09 +0000 [244491f9b2] Naveen Albert + + * app_reload: New Reload application + + Adds an application to reload modules + from within the dialplan. + + ASTERISK-29454 + + Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774 + +2021-07-08 09:32 +0000 [99d44f0c5a] Igor Goncharovsky + + * res_ari: Fix audiosocket segfault + + Add check that data parameter specified when audiosocket used for externalMedia. + + ASTERISK-29514 #close + + Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617 + +2021-06-30 08:07 +0000 [0ac9c83561] Sean Bright + + * res_pjsip_config_wizard.c: Add port matching support. + + In f8b0c2c9 we added support for port numbers in 'match' statements + but neglected to include that support in the PJSIP config wizard. + + The removed code would have also prevented IPv6 addresses from being + successfully used in the config wizard as well. + + ASTERISK-29503 #close + + Change-Id: Idd5bbfd48009e7a741757743dbaea68e2835a34d + +2021-05-22 09:31 +0000 [c01b4e0d4b] Naveen Albert + + * app_waitforcond: New application + + While several applications exist to wait for + a certain event to occur, none allow waiting + for any generic expression to become true. + This application allows for waiting for a condition + to become true, with configurable timeout and + checking interval. + + ASTERISK-29444 + + Change-Id: I08adf2824b8bc63405778cf355963b5005612f41 + +2021-06-04 06:11 +0000 [a47308ccb2] Andre Barbosa + + * res_stasis_playback: Send PlaybackFinish event only once for errors + + When we try to play a list of sound files in the same Play command, + we get only one PlaybackFinish event, after all sounds are played. + + But in the case where the Play fails (because channel is destroyed + for example), Asterisk will send one PlaybackFinish event for each + sound file still to be played. If the list is big, Asterisk is + sending many events. + + This patch adds a failed state so we can understand that the play + failed. On that case we don't send the event, if we still have a + list of sounds to be played. + + When we reach the last sound, we send the PlaybackFinish with + the failed state. + + ASTERISK-29464 #close + + Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322 + +2021-06-17 07:57 +0000 [bc973bd719] George Joseph + + * jitterbuffer: Correct signed/unsigned mismatch causing assert + + If the system time has stepped backwards because of a time + adjustment between the time a frame is timestamped and the + time we check the timestamps in abstract_jb:hook_event_cb(), + we get a negative interval, but we don't check for that there. + abstract_jb:hook_event_cb() then calls + fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed) + and the first thing that does is assert(interval >= 0). + + There are several issues with this... + + * abstract_jb:hook_event_cb() saves the interval in a variable + named "now" which is confusing in itself. + + * "now" is defined as an unsigned int which converts the negative + value returned from ast_tvdiff_ms() to a large positive value. + + * fixed_jb_get()'s parameter is defined as a signed int so the + interval gets converted back to a negative value. + + * fixed_jb_get()'s assert is NOT an ast_assert but a direct define + that points to the system assert() so it triggers even in + production mode. + + So... + + * hook_event_cb()'s "now" was renamed to "relative_frame_start" and + changed to an int64_t. + * hook_event_cb() now checks for a negative value right after + retrieving both the current and framedata timestamps and just + returns the frame if the difference is negative. + * fixed_jb_get()'s local define of ASSERT() was changed to call + ast_assert() instead of the system assert(). + + ASTERISK-29480 + Reported by: Dan Cropp + + Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9 + +2021-05-21 19:08 +0000 [1e5a2cfe30] Naveen Albert + + * app_dial: Expanded A option to add caller announcement + + Hitherto, the A option has made it possible to play + audio upon answer to the called party only. This option + is expanded to allow for playback of an audio file to + the caller instead of or in addition to the audio + played to the answerer. + + ASTERISK-29442 + + Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e + +2021-06-21 06:31 +0000 [5382b9dbb8] Joshua C. Colp + + * core: Don't play silence for Busy() and Congestion() applications. + + When using the Busy() and Congestion() applications the + function ast_safe_sleep is used by wait_for_hangup to safely + wait on the channel. This function may send silence if Asterisk + is configured to do so using the transmit_silence option. + + In a scenario where an answered channel dials a Local channel + either directly or through call forwarding and the Busy() + or Congestion() dialplan applications were executed with the + transmit_silence option enabled the busy or congestion + tone would not be heard. + + This is because inband generation of tones (such as busy + and congestion) is stopped when other audio is sent to + the channel they are being played to. In the given + scenario the transmit_silence option would result in + silence being sent to the channel, thus stopping the + inband generation. + + This change adds a variant of ast_safe_sleep which can be + used when silence should not be played to the channel. The + wait_for_hangup function has been updated to use this + resulting in the tones being generated as expected. + + ASTERISK-29485 + + Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133 + +2021-05-07 01:18 +0000 [c30f68a57b] Bernd Zobl + + * res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress + + With the fix for ASTERISK_28754 channels are no longer put on hold if an + outbound INVITE is answered with a "Session Progress" containing + "inactive" audio. + + The previous change moved the evaluation of the media attributes to + `negotiate_incoming_sdp_stream()` to have the `remotely_held` status + available when building the SDP in `create_outgoing_sdp_stream()`. + This however means that an answer to an outbound INVITE, which does not + traverse `negotiate_incoming_sdp_stream()`, cannot set the + `remotely_held` status anymore. + + This change moves the check so that both, `negotiate_incoming_sdp_stream()` and + `apply_negotiated_sdp_stream()` can do the checks. + + ASTERISK-29479 + + Change-Id: Icde805a819399d5123b688e1ed1d2bcd9d5b0f75 + +2021-06-16 08:50 +0000 [b7027de195] George Joseph + + * res_pjsip_messaging: Overwrite user in existing contact URI + + When the MessageSend destination is in the form + PJSIP/@ and the endpoint's contact + URI already has a user component, that user component + will now be replaced with when creating the + request URI. + + ASTERISK_29404 + + Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5 + +2021-03-16 11:45 +0000 [f160725fc4] Bernd Zobl + + * res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter + + Set preferred transport when querying the local address to use in + filter_on_tx_messages(). This prevents the module to erroneously select + the wrong transport if more than one transports of the same type (TCP or + TLS) are configured. + + ASTERISK-29241 + + Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6 + +2021-06-10 09:34 +0000 [f812c57477] Naveen Albert + + * pbx_builtins: Corrects SayNumber warning + + Previously, SayNumber always emitted a warning if the caller hung up + during execution. Usually this isn't correct, so check if the channel + hung up and, if so, don't emit a warning. + + ASTERISK-29475 + + Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594 + +2021-05-22 07:29 +0000 [56c2cc474b] Jaco Kroon + + * func_lock: Add "dialplan locks show" cli command. + + For example: + + arthur*CLI> dialplan locks show + func_lock locks: + Name Requesters Owner + uls-autoref 0 (unlocked) + 1 total locks listed. + + Obviously other potentially useful stats could be added (eg, how many + times there was contention, how many times it failed etc ... but that + would require keeping the stats and I'm not convinced that's worth the + effort. This was useful to troubleshoot some other issues so submitting + it. + + Change-Id: Ib875e56feb49d523300aec5f36c635ed74843a9f + Signed-off-by: Jaco Kroon + +2021-05-22 07:53 +0000 [19a8383a1f] Jaco Kroon + + * func_lock: Prevent module unloading in-use module. + + The scenario where a channel still has an associated datastore we + cannot unload since there is a function pointer to the destroy and fixup + functions in play. Thus increase the module ref count whenever we + allocate a datastore, and decrease it during destroy. + + In order to tighten the race that still exists in spite of this (below) + add some extra failure cases to prevent allocations in these cases. + + Race: + + If module ref is zero, an LOCK or TRYLOCK is invoked (near) + simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if + in such a case the datastore is created *prior* to unloading being set + to true (first step in module unload) then it's possible that the module + will unload with the destructor being called (and segfault) post the + module being unloaded. The module will however wait for such locks to + release prior to unloading. + + If post that we can recheck the module ref before returning the we can + (in theory, I think) eliminate the last of the race. This race is + mostly theoretical in nature. + + Change-Id: I21a514a0b56755c578a687f4867eacb8b59e23cf + Signed-off-by: Jaco Kroon + +2021-05-22 07:42 +0000 [e8875d5ca1] Jaco Kroon + + * func_lock: Fix memory corruption during unload. + + AST_TRAVERSE accessess current as current = current->(field).next ... + and since we free current (and ast_free poisons the memory) we either + end up on a ast_mutex_lock to a non-existing lock that can never be + obtained, or a segfault. + + Incidentally add logging in the "we have to wait for a lock to release" + case, and remove an ineffective statement that sets memory that was just + cleared by ast_calloc to zero. + + Change-Id: Id19ba3d9867b23d0e6783b97e6ecd8e62698b8c3 + Signed-off-by: Jaco Kroon + +2021-05-22 07:48 +0000 [caceba7988] Jaco Kroon + + * func_lock: Fix requesters counter in error paths. + + In two places we bail out with failure after we've already incremented + the requesters counter, if this occured then it would effectively result + in unload to wait indefinitely, thus preventing clean shutdown. + + Change-Id: I362a6c0dc424f736d4a9c733d818e72d19675283 + Signed-off-by: Jaco Kroon + +2021-05-25 10:36 +0000 [b742514553] Naveen Albert + + * app_originate: Allow setting Caller ID and variables + + Caller ID can now be set on the called channel and + Variables can now be set on the destination + using the Originate application, just as + they can be currently using call files + or the Manager Action. + + ASTERISK-29450 + + Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66 + +2021-06-10 16:24 +0000 [c0fc8adbb6] Sean Bright + + * menuselect: Fix description of several modules. + + The text description needs to be the last thing on the AST_MODULE_INFO + line to be pulled in properly by menuselect. + + Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832 + +2021-05-23 19:20 +0000 [35437879e5] Naveen Albert + + * app_confbridge: New ConfKick() application + + Adds a new ConfKick() application, which may + be used to kick a specific channel, all channels, + or all non-admin channels from a specified + conference bridge, similar to existing CLI and + AMI commands. + + ASTERISK-29446 + + Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b + +2021-06-02 08:25 +0000 [1b38e89734] Naveen Albert + + * res_pjsip_dtmf_info: Hook flash + + Adds hook flash recognition support + for application/hook-flash. + + ASTERISK-29460 + + Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea + +2021-05-20 09:51 +0000 [5f8cabc232] Naveen Albert + + * app_confbridge: New option to prevent answer supervision + + A new user option, answer_channel, adds the capability to + prevent answering the channel if it hasn't already been + answered yet. + + ASTERISK-29440 + + Change-Id: I26642729d0345f178c7b8045506605c8402de54b + +2021-06-02 08:11 +0000 [c8bf8a54c2] Naveen Albert + + * sip_to_pjsip: Fix missing cases + + Adds the "auto" case which is valid with + both chan_sip dtmfmode and chan_pjsip's + dtmf_mode, adds subscribecontext to + subscribe_context conversion, and accounts + for cipher = ALL being invalid. + + ASTERISK-29459 + + Change-Id: Ie27d6606efad3591038000e5f3c34fa94730f6f2 + +2021-04-22 13:07 +0000 [c3654a9959] George Joseph + + * res_pjsip_messaging: Refactor outgoing URI processing + + * Implemented the new "to" parameter of the MessageSend() + dialplan application. This allows a user to specify + a complete SIP "To" header separate from the Request URI. + + * Completely refactored the get_outbound_endpoint() function + to actually handle all the destination combinations that + we advertized as supporting. + + * We now also accept a destination in the same format + as Dial()... PJSIP/number@endpoint + + * Added lots of debugging. + + ASTERISK-29404 + Reported by Brian J. Murrell + + Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce + +2021-05-16 10:21 +0000 [eeffad1b62] Naveen Albert + + * func_math: Three new dialplan functions + + Introduces three new dialplan functions, MIN and MAX, + which can be used to calculate the minimum or + maximum of up to two numbers, and ABS, an absolute + value function. + + ASTERISK-29431 + + Change-Id: I2bda9269d18f9d54833c85e48e41fce0e0ce4d8d + +2021-05-19 13:45 +0000 [12e8600849] Ben Ford + + * STIR/SHAKEN: Add Date header, dest->tn, and URL checking. + + STIR/SHAKEN requires a Date header alongside the Identity header, so + that has been added. Still on the outgoing side, we were missing the + dest->tn section of the JSON payload, so that has been added as well. + Moving to the incoming side, URL checking has been added to the public + cert URL to ensure that it starts with http. + + https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 + + Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c + +2021-05-24 13:38 +0000 [44fde9f428] Joshua C. Colp + + * res_pjsip: On partial transport reload also move factories. + + For connection oriented transports PJSIP uses factories to + produce transports. When doing a partial transport reload + we need to also move the factory of the transport over so + that anything referencing the transport (such as an endpoint) + has the factory available. + + ASTERISK-29441 + + Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161 + +2021-05-20 08:18 +0000 [19b5097d87] Naveen Albert + + * func_volume: Add read capability to function. + + Up until now, the VOLUME function has been write + only, so that TX/RX values can be set but not + read afterwards. Now, previously set TX/RX values + can be read later. + + ASTERISK-29439 + + Change-Id: Ia23e92fa2e755c36e9c8e69f2940d2703ccccb5f + +2021-04-13 02:57 +0000 [2193cf1b26] Evgenios_Greek + + * stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing + + When unsubscribing from an endpoint technology a FRACK + would occur due to incorrect reference counting. This fixes + that issue, along with some other issues. + + Fixed a typo in get_subscription when calling ao2_find as it + needed to pass the endpoint ID and not the entire object. + + Fixed scenario where a subscription would get returned when + it shouldn't have been when searching based on endpoint + technology. + + A doulbe unreference has also been resolved by only explicitly + releasing the reference held by tech_subscriptions. + + ASTERISK-28237 #close + Reported by: Lucas Tardioli Silveira + + Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729 + +2021-05-20 02:15 +0000 [98e4119642] Joseph Nadiv + + * res_pjsip.c: Support endpoints with domain info in username + + In multidomain environments, it is desirable to create + PJSIP endpoints with the domain info in the endpoint name + in pjsip_endpoint.conf. This resulted in an error with + registrations, NOTIFY, and OPTIONS packet generation. + + This commit will detect if there is an @ in the endpoint + identifier and generate the URI accordingly so NOTIFY and + OPTIONS From headers will generate correctly. + + ASTERISK-28393 + + Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619 + +2021-05-20 07:51 +0000 [a985e5069c] Joshua C. Colp + + * res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates. + + RTCP ICE candidates use a base address derived from the RTP + candidate. The port on the base address was not being updated to + the RTCP port. + + This change sets the base port to the RTCP port and all is well. + + ASTERISK-29433 + + Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040 + +2021-05-25 05:38 +0000 [987f5eb0ad] Joshua C. Colp + + * asterisk: We've moved to Libera Chat! + + Change-Id: I48c1933dd79b50ddc0a6793acec4754b4e95c575 + +2021-05-19 13:13 +0000 [d162789c4d] Jeremy Lainé + + * res_rtp_asterisk: make it possible to remove SOFTWARE attribute + + By default Asterisk reports the PJSIP version in a SOFTWARE attribute + of every STUN packet it sends. This may not be desired in a production + environment, and RFC5389 recommends making the use of the SOFTWARE + attribute a configurable option: + + https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2 + + This patch adds a `stun_software_attribute` yes/no option to make it + possible to omit the SOFTWARE attribute from STUN packets. + + ASTERISK-29434 + + Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b + +2021-04-15 10:43 +0000 [9cc1d6fc22] George Joseph + + * res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs + + RFC7616 and RFC8760 allow more than one WWW-Authenticate or + Proxy-Authenticate header per realm, each with different digest + algorithms (including new ones like SHA-256 and SHA-512-256). + Thankfully however a UAS can NOT send back multiple Authenticate + headers for the same realm with the same digest algorithm. The + UAS is also supposed to send the headers in order of preference + with the first one being the most preferred. We're supposed to + send an Authorization header for the first one we encounter for a + realm that we can support. + + The UAS can also send multiple realms, especially when it's a + proxy that has forked the request in which case the proxy will + aggregate all of the Authenticate headers and then send them all + back to the UAC. + + It doesn't stop there though... Each realm can require a + different username from the others. There's also nothing + preventing each digest algorithm from having a unique password + although I'm not sure if that adds any benefit. + + So now... For each Authenticate header we encounter, we have to + determine if we support the digest algorithm and, if not, just + skip the header. We then have to find an auth object that + matches the realm AND the digest algorithm or find a wildcard + object that matches the digest algorithm. If we find one, we add + it to the results vector and read the next Authenticate header. + If the next header is for the same realm AND we already added an + auth object for that realm, we skip the header. Otherwise we + repeat the process for the next header. + + In the end, we'll have accumulated a list of credentials we can + pass to pjproject that it can use to add Authentication headers + to a request. + + NOTE: Neither we nor pjproject can currently handle digest + algorithms other than MD5. We don't even have a place for it in + the ast_sip_auth object. For this reason, we just skip processing + any Authenticate header that's not MD5. When we support the + others, we'll move the check into the loop that searches the + objects. + + Changes: + + * Added a new API ast_sip_retrieve_auths_vector() that takes in + a vector of auth ids (usually supplied on a call to + ast_sip_create_request_with_auth()) and populates another + vector with the actual objects. + + * Refactored res_pjsip_outbound_authenticator_digest to handle + multiple Authenticate headers and set the stage for handling + additional digest algorithms. + + * Added a pjproject patch that allows them to ignore digest + algorithms they don't support. This patch has already been + merged upstream. + + * Updated documentation for auth objects in the XML and + in pjsip.conf.sample. + + * Although res_pjsip_authenticator_digest isn't affected + by this change, some debugging and a testsuite AMI event + was added to facilitate testing. + + Discovered during OpenSIPit 2021. + + ASTERISK-29397 + + Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281 + +2021-04-14 09:44 +0000 [3cccdf6d98] Joseph Nadiv + + * res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml + + RFC 4235 Section 4.1.6 describes XML elements that should be + sent to subscribed endpoints to identify the local and remote + participants in the dialog. + + This patch adds this functionality to PJSIP by iterating through the + ringing channels causing the NOTIFY, and inserts the channel info + into the dialog so that information is properly passed to the endpoint + in dialog-info+xml. + + ASTERISK-24601 + Patch submitted: Joshua Elson + Modified by: Joseph Nadiv and Sean Bright + Tested by: Joseph Nadiv + + Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b + +2021-05-13 10:32 +0000 [04454fc238] Naveen Albert + + * AMI: Add AMI event to expose hook flash events + + Although Asterisk can receive and propogate flash events, it currently + provides no mechanism for doing anything with them itself. + + This AMI event allows flash events to be processed by Asterisk. + Additionally, AST_CONTROL_FLASH is included in a switch statement + in channel.c to avoid throwing a warning when we shouldn't. + + ASTERISK-29380 + + Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81 + +2021-05-13 09:47 +0000 [567ea5abf8] Naveen Albert + + * app_voicemail: Configurable voicemail beep + + Hitherto, VoiceMail() played a non-customizable beep tone to indicate + the caller could leave a message. In some cases, the beep may not + be desired, or a different tone may be desired. + + To increase flexibility, a new option allows customization of the tone. + If the t option is specified, the default beep will be overridden. + Supplying an argument will cause it to use the specified file for the tone, + and omitting it will cause it to skip the beep altogether. If the option + is not used, the default behavior persists. + + ASTERISK-29349 + + Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280 + +2021-05-13 10:13 +0000 [0026aeada3] Naveen Albert + + * main/file.c: Don't throw error on flash event. + + AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c + where it should be ignored. Adding this to the switch ensures a + warning isn't thrown on RFC2833 flash events, since nothing's amiss. + + ASTERISK-29372 + + Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc + +2021-05-13 08:50 +0000 [fd40752954] Naveen Albert + + * chan_sip: Expand hook flash recognition. + + Some ATAs send hook flash events as application/hook-flash, rather than a DTMF + event. Now, we also recognize hook-flash as a flash event. + + ASTERISK-29370 + + Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725 + +2021-05-11 12:00 +0000 [49c2e7e307] Joshua C. Colp + + * pjsip: Add patch for resolving STUN packet lifetime issues. + + In some cases it was possible for a STUN packet to be destroyed + prematurely or even destroyed partially multiple times. + + This patch provided by Teluu fixes the lifetime of these + packets and ensures they aren't partially destroyed multiple + times. + + https://github.com/pjsip/pjproject/pull/2709 + + ASTERISK-29377 + + Change-Id: Ie842ad24ddf345e01c69a4d333023f05f787abca + +2021-05-12 21:20 +0000 [1b41629447] Sean Bright + + * chan_pjsip: Correct misleading trace message + + ASTERISK-29358 #close + + Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647 + +2021-04-26 17:00 +0000 [0564d12280] Ben Ford + + * STIR/SHAKEN: Switch to base64 URL encoding. + + STIR/SHAKEN encodes using base64 URL format. Currently, we just use + base64. New functions have been added that convert to and from base64 + encoding. + + The origid field should also be an UUID. This means there's no reason to + have it as an option in stir_shaken.conf, as we can simply generate one + when creating the Identity header. + + https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 + + Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c + +2021-05-11 12:26 +0000 [05f7bc9c66] Ben Ford + + * STIR/SHAKEN: OPENSSL_free serial hex from openssl. + + We're getting the serial number of the certificate from openssl and + freeing it with ast_free(), but it needs to be freed with OPENSSL_free() + instead. Now we duplicate the string and free the one from openssl with + OPENSSL_free(), which means we can still use ast_free() on the returned + string. + + https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 + + Change-Id: Ia6e1a4028c1933a0e1d204b769ebb9f5a11f00ab + +2021-04-21 11:12 +0000 [259ecfa289] Ben Ford + + * STIR/SHAKEN: Fix certificate type and storage. + + During OpenSIPit, we found out that the public certificates must be of + type X.509. When reading in public keys, we use the corresponding X.509 + functions now. + + We also discovered that we needed a better naming scheme for the + certificates since certificates with the same name would cause issues + (overwriting certs, etc.). Now when we download a public certificate, we + get the serial number from it and use that as the name of the cached + certificate. + + The configuration option public_key_url in stir_shaken.conf has also + been renamed to public_cert_url, which better describes what the option + is for. + + https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 + + Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d + +2021-04-22 13:07 +0000 [09303e8e22] George Joseph + + * Updates for the MessageSend Dialplan App + + Enhancements: + + * The MessageSend dialplan application now takes an optional + third argument that can set the message's "To" field on + outgoing messages. It's an alternative to using the + MESSAGE(to) dialplan function. + + NOTE: No channel driver currently implements this field. A + follow-on commit for res_pjsip_messaging will implement it for + the chan_pjsip channel driver. + + * To prevent confusion with the first argument, currently named + "to", it's been renamed to "destination". Its function, + creating the request URI, hasn't changed. + + * The documentation for MessageSend was updated to be + more clear about the parameters and how they interact + the MESSAGE() dialplan function. + + * With the rename of MessageSend's first parameter, and the fact + that message.c references elements in chan_sip.c, + res_pjsip_messaging.c and res_xmpp, they each needed + documentation updates to use MessageDestinationInfo instead of + MessageToInfo. + + * appdocsxml.dtd was updated to include a missing element + declaration for "dataType". This was showing up as an error + in Eclipse's dtd editor. + + * Despite the changes in this commit, there should be + no impact to current users of MessageSend. + + Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a + +2021-04-30 15:21 +0000 [e39efabd97] Sean Bright + + * translate.c: Avoid refleak when checking for a translation path + + Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6 + +2021-04-27 12:31 +0000 [b1807d440e] Sean Bright + + * res_rtp_asterisk: More robust timestamp checking + + We assume that a timestamp value of 0 represents an 'uninitialized' + timestamp, but 0 is a valid value. Add a simple wrapper to be able to + differentiate between whether the value is set or not. + + This also removes the fix for ASTERISK~28812 which should not be + needed if we are checking the last timestamp appropriately. + + ASTERISK-29030 #close + + Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7 + +2021-04-28 07:17 +0000 [f142ca254e] Joshua C. Colp + + * chan_local: Skip filtering audio formats on removed streams. + + When a stream topology is provided to chan_local when dialing + it filters the audio formats down. This operation did not skip + streams which were removed (that have no formats) resulting in + calling being aborted. + + This change causes such streams to be skipped. + + ASTERISK-29407 + + Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6 + +2021-04-23 12:37 +0000 [4a843e00ef] Sean Bright + + * res_pjsip.c: OPTIONS processing can now optionally skip authentication + + ASTERISK-27477 #close + + Change-Id: I68f6715bba92a525149e35d142a49377a34a1193 + +2021-04-21 06:42 +0000 [55279bfd9c] Jean Aunis + + * translate.c: Take sampling rate into account when checking codec's buffer size + + Up/down sampling changes the number of samples produced by a translation. + This must be taken into account when checking the codec's buffer size. + + ASTERISK-29328 + + Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055 + +2021-04-25 04:45 +0000 [531eb65cf3] Joshua C. Colp + + * svn: Switch to https scheme. + + Some versions of SVN seemingly don't follow the redirect + to https. + + Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253 + +2021-04-20 08:42 +0000 [512d38868c] George Joseph + + * res_pjsip: Update documentation for the auth object + + Change-Id: I2f76867ce02ec611964925159be099de83346e38 + +2021-03-29 12:28 +0000 [45a1977de4] Ben Ford + + * res_aeap: Add basic config skeleton and CLI commands. + + Added support for a basic AEAP configuration read from aeap.conf. + Also added 2 CLI commands for showing individual configurations as + well as all of them: aeap show server and aeap show servers. + + Only one configuration option is required at the moment, and that one is + server_url. It must be a websocket URL. The other option, codecs, is + optional and will be used over the codecs specified on the endpoint if + provided. + + https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453 + + Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3 + +2021-04-02 07:21 +0000 [44aef0449a] George Joseph + + * bridge_channel_write_frame: Check for NULL channel + + There is a possibility, when bridge_channel_write_frame() is + called, that the bridge_channel->chan will be NULL. The first + thing bridge_channel_write_frame() does though is call + ast_channel_is_multistream() which had no check for a NULL + channel and therefore caused a segfault. Since it's still + possible for bridge_channel_write_frame() to write the frame to + the other channels in the bridge, we don't want to bail before we + call ast_channel_is_multistream() but we can just skip the + multi-channel stuff. So... + + bridge_channel_write_frame() only calls ast_channel_is_multistream() + if bridge_channel->chan is not NULL. + + As a safety measure, ast_channel_is_multistream() now returns + false if the supplied channel is NULL. + + ASTERISK-29379 + Reported-by: Vyrva Igor + Reported-by: Ross Beer + + Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce + +2021-04-01 10:38 +0000 [5a13e95c56] Sean Bright + + * loader.c: Speed up deprecation metadata lookup + + Only use an XPath query once per module, then just navigate the DOM for + everything else. + + Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92 + +2021-04-01 08:39 +0000 [53c702e1cc] George Joseph + + * res_prometheus: Clone containers before iterating + + The channels, bridges and endpoints scrape functions were + grabbing their respective global containers, getting the + count of entries, allocating metric arrays based on + that count, then iterating over the container. If the + global container had new objects added after the count + was taken and the metric arrays were allocated, we'd run + out of metric entries and attempt to write past the end + of the arrays. + + Now each of the scape functions clone their respective + global containers and all operations are done on the + clone. Since the clone is stable between getting the + count and iterating over it, we can't run past the end + of the metrics array. + + ASTERISK-29130 + Reported-By: Francisco Correia + Reported-By: BJ Weschke + Reported-By: Sébastien Duthil + + Change-Id: If0c8e40853bc0e9429f2ba9c7f5f358d90c311af + +2021-03-10 09:03 +0000 [46ed6af9c2] Joshua C. Colp + + * loader: Output warnings for deprecated modules. + + Using the information from the MODULEINFO XML we can + now output useful information at the end of module + loading for deprecated modules. This includes the + version it was deprecated in, the version it will be + removed in, and the replacement if available. + + ASTERISK-29339 + + Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a + +2021-03-22 15:22 +0000 [0fc906a5e1] Kevin Harwell + + * res_rtp_asterisk: Fix standard deviation calculation + + For some input to the standard deviation algorithm extremely large, + and wrong numbers were being calculated. + + This patch uses a new formula for correctly calculating both the + running mean and standard deviation for the given inputs. + + ASTERISK-29364 #close + + Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f + +2021-03-29 17:40 +0000 [c4a376aac2] Kevin Harwell + + * res_rtp_asterisk: Don't count 0 as a minimum lost packets + + The calculated minimum lost packets represents the lowest number of + lost packets missed during an RTCP report interval. Zero of course + is the lowest, but the idea is that this value contain the lowest + number of lost packets once some have been missed. + + This patch checks to make sure the number of lost packets over an + interval is not zero before checking and setting the minimum value. + + Also, this patch updates the rtp lost packet test to check for + packet loss over several reports vs one. + + Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008 + +2021-03-31 12:17 +0000 [65b68fd060] Kevin Harwell + + * res_rtp_asterisk: Statically declare rtp_drop_packets_data object + + This patch makes the drop_packets_data object static. + + Change-Id: If4f9b21fa0c47d41a35b6b05941d978efb4da87b + +2021-03-29 17:52 +0000 [8bd13a995a] Joshua C. Colp + + * res_rtp_asterisk: Only raise flash control frame on end. + + Flash in RTP is conveyed the same as DTMF, just with a + specific digit. In Asterisk however we do flash as a + single control frame. + + This change makes it so that only on end do we provide + the flash control frame to the core. Previously we would + provide a flash control frame on both begin and end, + causing flash to work improperly. + + ASTERISK-29373 + + Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226 + +2021-03-05 12:53 +0000 [b86f1ef54c] Kevin Harwell + + * res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command + + This patch makes it so when Asterisk is compiled in DEVMODE a CLI + command is available that allows someone to drop incoming RTP + packets. The command allows for dropping of packets once, or on a + timed interval (e.g. drop 10 packets every 5 seconds). A user can + also specify to drop packets by IP address. + + Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024 + +2021-03-30 06:59 +0000 [623abc2b6a] Joshua C. Colp + + * res_pjsip: Give error when TLS transport configured but not supported. + + Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b + +2021-03-05 12:47 +0000 [eb92fb7298] Kevin Harwell + + * time: Add timeval create and unit conversion functions + + Added a TIME_UNIT enumeration, and a function that converts a + string to one of the enumerated values. Also, added functions + that create and initialize a timeval object using a specified + value, and unit type. + + Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392 + +2021-03-24 08:38 +0000 [8db2a34065] Sean Bright + + * app_queue: Add alembic migration to add ringinuse to queue_members. + + ASTERISK-28356 #close + + Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39 + +2021-03-28 10:47 +0000 [c2dbfb9a8e] Sean Bright + + * modules.conf: Fix more differing usages of assignment operators. + + I missed the changes in 18 and master in the previous review. + + ASTERISK-24434 #close + + Change-Id: Ieb132b2a998ce96daa9c9acf26535a974b895876 + +2021-03-24 10:52 +0000 [25758670b8] Ben Ford + + * logger.conf.sample: Add more debug documentation. + + Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff + +2021-03-24 10:59 +0000 [55c53de022] Ben Ford + + * logging: Add .log to samples and update asterisk.logrotate. + + Added .log extension to the sample logs in logger.conf.sample so that + they will be able to be opened in the browser when attached to JIRA + tickets. Because of this, asterisk.logrotate has also been updated to + look for .log extensions instead of no extension for log files such as + full and messages. + + Change-Id: I5de743c03f08047d6c6cc80cac5019ae0c4c200f + +2021-03-23 15:15 +0000 [aac442eecd] Sean Bright + + * app_queue.c: Remove dead 'updatecdr' code. + + Also removed the sample documentation, and some oddly-placed + documentation about the timeout argument to the Queue() application + itself. There is a large section on the timeout behavior below. + + ASTERISK-26614 #close + + Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217 + +2021-03-23 17:24 +0000 [cad843fe07] Sean Bright + + * queues.conf.sample: Correct 'context' documentation. + + ASTERISK-24631 #close + + Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f + +2021-03-19 09:11 +0000 [b4347c4861] Mark Murawski + + * logger: Console sessions will now respect logger.conf dateformat= option + + The 'core' console (ie: asterisk -c) does read logger.conf and does + use the dateformat= option. + + Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf + and uses a hard coded dateformat option for printing received verbose messages: + main/logger.c: static char dateformat[256] = "%b %e %T" + + This change will load logger.conf for each remote console session and + use the dateformat= option to set the per-line timestamp for verbose messages + + Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1 + ASTERISK-25358: #close + Reported-by: Igor Liferenko + +2021-03-19 15:57 +0000 [8d3d7bdb82] Sean Bright + + * app_queue.c: Don't crash when realtime queue name is empty. + + ASTERISK-27542 #close + + Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a + +2021-03-18 11:14 +0000 [a03a05195a] George Joseph + + * res_pjsip_session: Make reschedule_reinvite check for NULL topologies + + When the check for equal topologies was added to reschedule_reinvite() + it was assumed that both the pending and active media states would + actually have non-NULL topologies. We since discovered this isn't + the case. + + We now only test for equal topologies if both media states have + non-NULL topologies. The logic had to be rearranged a bit to make + sure that we cloned the media states if their topologies were + non-NULL but weren't equal. + + ASTERISK-29215 + + Change-Id: I61313cca7fc571144338aac826091791b87b6e17 + +2021-03-19 04:56 +0000 [a8a08bcd1e] Joshua C. Colp + + * app_queue: Only send QueueMemberStatus if status changes. + + If a queue member was updated with the same status multiple + times each time a QueueMemberStatus event would be sent + which would be a duplicate of the previous. + + This change makes it so that the QueueMemberStatus event is + only sent if the status actually changes. + + ASTERISK-29355 + + Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116 + +2021-03-19 08:52 +0000 [970b84946e] Joshua C. Colp + + * core_unreal: Fix deadlock with T.38 control frames. + + When using the ast_unreal_lock_all function no channel + locks can be held before calling it. + + This change unlocks the channel that indicate was + called on before doing so and then relocks it afterwards. + + ASTERISK-29035 + + Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9 + +2021-03-01 17:32 +0000 [71dfbdc7b9] Joshua C. Colp + + * res_pjsip: Add support for partial transport reload. + + Some configuration items for a transport do not result in + the underlying transport changing, but instead are just + state we keep ourselves and use. It is perfectly reasonable + to change these items. + + These include local_net and external_* information. + + ASTERISK-29354 + + Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a + +2021-03-13 05:01 +0000 [fc03116d9b] Jaco Kroon + + * menuselect: exit non-zero in case of failure on --enable|disable options. + + ASTERISK-29348 + + Change-Id: I77e3466435f5a51a57538b29addb68d811af238d + Signed-off-by: Jaco Kroon + +2021-03-17 10:28 +0000 [cce5ee5b7a] Joshua C. Colp + + * res_rtp_asterisk: Force resync on SSRC change. + + When an SSRC change occurs the timestamps are likely + to change as well. As a result we need to reset the + timestamp mapping done in the calc_rxstamp function + so that they map properly from timestamp to real + time. + + This previously occurred but due to packet + retransmission support the explicit setting + of the marker bit was not effective. + + ASTERISK-29352 + + Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a + +2021-03-10 08:05 +0000 [efc61a96f0] Joshua C. Colp + + * menuselect: Add ability to set deprecated and removed versions. + + The "deprecated_in" and "removed_in" information can now be + set in MODULEINFO for a module and is then displayed in + menuselect so users can be aware of when a module is slated + to be deprecated and then removed. + + ASTERISK-29337 + + Change-Id: I6952889cf08e0e9e99cf8b43f99b3cef4688087a + +2021-03-10 08:18 +0000 [3330fb41f4] Joshua C. Colp + + * xml: Allow deprecated_in and removed_in for MODULEINFO. + + ASTERISK-29337 + + Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b + +2021-03-09 08:54 +0000 [149e5e5b86] Joshua C. Colp + + * xml: Embed module information into core XML documentation. + + This change embeds the MODULEINFO block of modules + into the core XML documentation. This provides a shared + mechanism for use by both menuselect and Asterisk for + information and a definitive source of truth. + + ASTERISK-29335 + + Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90 + +2021-03-10 04:47 +0000 [7438586d8e] Joshua C. Colp + + * documentation: Fix non-matching module support levels. + + Some modules have a different support level documented in their + MODULEINFO XML and Asterisk module definition. This change + brings the two in sync for the modules which were not matching. + + ASTERISK-29336 + + Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35 + +2021-03-09 18:35 +0000 [cc127a999c] Joshua C. Colp + + * channel: Fix crash in suppress API. + + There exists an inconsistency with framehook usage + such that it is only on reads that the frame should + be freed, not on writes as well. + + ASTERISK-29071 + + Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472 + +2021-02-24 12:00 +0000 [41389bfdbd] Jaco Kroon + + * func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds + + Change-Id: I75152cece8a00b7523d542e5ac22796f9595692b + Signed-off-by: Jaco Kroon + +2021-02-24 12:34 +0000 [8acb4fbd1e] Jaco Kroon + + * app.h: Fix -Werror=zero-length-bounds compile errors in dev mode. + + Change-Id: I5c104dc1f8417ccd3d01faf86e84ccbf89bc3b31 + Signed-off-by: Jaco Kroon + +2021-03-06 16:57 +0000 [8987de270f] Sean Bright + + * app_dial.c: Only send DTMF on first progress event. + + ASTERISK-29329 #close + + Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601 + +2021-03-05 11:16 +0000 [1ae40e502d] Alexander Traud + + * res_format_attr_*: Parameter Names are Case-Insensitive. + + see RFC 4855: + parameter names are case-insensitive both in media type strings and + in the default mapping to the SDP a=fmtp attribute. + + This change is required for H.263+ because some implementations are + known to use even mixed-case. This does not fix ASTERISK~29268 because + H.264 was not fixed. This approach here lowers/uppers both parameter + names and parameter values. H.264 needs a different approach because + one of its parameter values is not case-insensitive: + sprop-parameter-sets is Base64. + + Change-Id: Idf2a73457be231647aed3c87b1da197afba86892 + +2021-03-05 11:45 +0000 [8c461845c8] Alexander Traud + + * chan_iax2: System Header strings is included via asterisk.h/compat.h. + + The system header strings was included mistakenly with commit 3de0204. + That header is included via asterisk.h and there via the compat.h. + + Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a + +2021-03-08 15:43 +0000 [55bd104589] Sean Bright + + * modules.conf: Fix differing usage of assignment operators. + + ASTERISK-24434 #close + + Change-Id: I0144e8d65d878128da59dcf3df12ca8cee47d6db + +2021-03-08 14:06 +0000 [30e509c2f9] Sean Bright + + * strings.h: ast_str_to_upper() and _to_lower() are not pure. + + Because they modify their argument they are not pure functions and + should not be marked as such, otherwise the compiler may optimize + them away. + + ASTERISK-29306 #close + + Change-Id: Ibec03a08522dd39e8a137ece9bc6a3059dfaad5f + +2021-03-08 17:16 +0000 [df37b8181c] Sean Bright + + * res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse. + + ao2_replace() bumps the reference count of the object that is doing the + replacing, which is not what we want. We just want to drop the old ref + on the old object and update the pointer to point to the new object. + + Pointed out by George Joseph in #asterisk-dev + + Change-Id: Ie8167ed3d4b52b9d1ea2d785f885e8c27206743d + +2021-02-19 05:50 +0000 [8c247e2a94] Torrey Searle + + * res/res_rtp_asterisk: generate new SSRC on native bridge end + + For RTCP to work, we update the ssrc to be the one corresponding to + the native bridge while active. However when the bridge ends we + should generate a new SSRC as the sequence numbers will not continue + from the native bridge left off. + + ASTERISK-29300 #close + + Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10 + +2021-03-01 15:35 +0000 [304f8ddfb2] Joshua C. Colp + + * sorcery: Add support for more intelligent reloading. + + Some sorcery objects actually contain dynamic content + that can change despite the underlying configuration + itself not changing. A good example of this is the + res_pjsip_endpoint_identifier_ip module which allows + specifying hostnames. While the configuration may not + change between reloads the DNS information of the + hostnames can. + + This change adds the ability for a sorcery object to be + marked as having dynamic contents which is then taken + into account when reloading by the sorcery file based + config module. If there is an object with dynamic content + then a reload will be forced while if there are none + then the existing behavior of not reloading occurs. + + ASTERISK-29321 + + Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1 + +2021-03-02 12:55 +0000 [607603cf89] George Joseph + + * res_pjsip_refer: Move the progress dlg release to a serializer + + Although the dlg session count was incremented in a pjsip servant + thread, there's no guarantee that the last thread to unref this + progress object was one. Before we decrement, we need to make + sure that this is either a servant thread or that we push the + decrement to a serializer that is one. + + Because pjsip_dlg_dec_session requires the dialog lock, we don't + want to wait on the task to complete if we had to push it to a + serializer. + + Change-Id: I8ff2d5d94be3ff04298394070434e22a7d3cbc41 + +2021-03-03 12:31 +0000 [6f67f24afd] Joshua C. Colp + + * res_pjsip_registrar: Include source IP and port in log messages. + + When registering it can be useful to see the source IP address and + port in cases where multiple devices are using the same endpoint + or when anonymous is in use. + + ASTERISK-29325 + + Change-Id: Ie178a6f55f53f8473035854c411bc3d056e0a2e0 + +2021-03-03 12:44 +0000 [f8d1758792] Joshua C. Colp + + * asterisk: Update copyright. + + ASTERISK-29326 + + Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a + +2021-02-25 13:50 +0000 [fd560ad9fa] Ben Ford + + * AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. + + When Asterisk sends a reinvite negotiating T38 faxing, it's possible a + crash can occur if the response contains a m=image and zero port. The + reinvite callback code now checks session_media to see if it is null or + not before trying to access the udptl variable on it. + + ASTERISK-29305 + + Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9 + +2021-01-28 08:39 +0000 [a34e7de61c] Alexander Traud + + * res_format_attr_h263: Generate valid SDP fmtp for H.263+. + + Fixed: + * RFC 4629 does not allow the value "0" for MPI, K, and N. + * Allow value "0" for PAR. + * BPP is printed only when specified because "0" has a meaning. + + New: + * Added CPCF and MaxBR. + * Some implementations provide CIF without MPI: a=fmtp:xx CIF;F=1 + Although a violation of RFC 3555 section 3, we can support that. + + Changed: + * Resorts the CIFs from large to small which partly fixes ASTERISK~29267. + + Change-Id: I95a650c715007b8dde11a77cb37d9c6c123a441e + +2021-02-24 07:04 +0000 [2c1b6b7b15] Joshua C. Colp + + * res_pjsip_nat: Don't rewrite Contact on REGISTER responses. + + When sending a SIP response to an incoming REGISTER request + we don't want to change the Contact header as it will + contain the Contacts registered to the AOR and not our own + Contact URI. + + ASTERISK-29235 + + Change-Id: I35a0723545281dd01fcd5cae497baab58720478c + +2021-03-03 07:32 +0000 [3e5b9e3952] Joshua C. Colp + + * channel: Fix memory leak in suppress API. + + A frame suppression API exists as part of channels + which allows audio frames to or from a channel to + be dropped. The MuteAudio AMI action uses this + API to perform its job. + + This API uses a framehook to intercept flowing + audio and drop it when appropriate. It is the + responsibility of the framehook to free the + frame it is given if it changes the frame. The + suppression API failed to do this resulting in + a leak of audio frames. + + This change adds the freeing of these frames. + + ASTERISK-29071 + + Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b + +2021-01-27 14:01 +0000 [5d42dd2e6a] Salah Ahmed + + * res_rtp_asterisk: Check remote ICE reset and reset local ice attrb + + This change will check is the remote ICE session got reset or not by + checking the offered ufrag and password with session. If the remote ICE + reset session then Asterisk reset its local ufrag and password to reject + binding request with Old ufrag and Password. + + ASTERISK-29266 + + Change-Id: I9c55e79a7af98a8fbb497d336b828ba41bc34eeb + +2021-01-07 08:25 +0000 [48ed4f670f] Holger Hans Peter Freyther + + * pjsip: Generate progress (once) when receiving a 180 with a SDP + + ASTERISK-29105 + + Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38 + +2021-02-28 03:24 +0000 [2ea75ed3d5] Nico Kooijman + + * main: With Dutch language year after 2020 is not spoken in say.c + + Implemented the english way of saying the year in ast_say_date_with_format_nl. + Currently the numbers are spoken correctly until 2020 and stopped working + this year. + + ASTERISK-29297 #close + Reported-by: Jacek Konieczny + + Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4 + +2021-02-24 20:51 +0000 [8f6e0f9367] Nick French + + * res_pjsip: dont return early from registration if init auth fails + + If set_outbound_initial_authentication_credentials() fails, + handle_client_registration() bails early without creating or + sending a register message. + + [set_outbound_initial_authentication_credentials() failures + can occur during the process of retrieving an oauth access + token.] + + The return from handle_client_registration is ignored, so + returning an error doesn't do any good. + + This is a real problem when the registration request is a + re-register, because then the registration will still be + marked 'active' despite the re-register never being sent at all. + + So instead, log a warning but let the registration be created + and sent (and probably fail) and follow the normal registration + failed retry/abort logic. + + ASTERISK-29315 #close + + Change-Id: I2e03b1ea7fba1fa1a8279086aa4b17679e7fa7fa + +2021-02-23 10:14 +0000 [d2f623bae2] Alexei Gradinari + + * res_fax: validate the remote/local Station ID for UTF-8 format + + If the remote Station ID contains invalid UTF-8 characters + the asterisk fails to publish the Stasis and ReceiveFax status messages. + + json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. + 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28] + 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc] + 2: /usr/sbin/asterisk(ast_channel_publish_varset+0x2b) [0x57aa0b] + 3: /usr/sbin/asterisk(pbx_builtin_setvar_helper+0x121) [0x530641] + 4: /usr/lib64/asterisk/modules/res_fax.so(+0x44fe) [0x7f27f4bff4fe] + ... + stasis_channels.c: Error creating message + + json.c: Error building JSON from '{s: s, s: s, s: s, s: s, s: s, s: s, s: o}': Invalid UTF-8 string. + 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28] + 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc] + 2: /usr/lib64/asterisk/modules/res_fax.so(+0x5acd) [0x7f27f4c00acd] + ... + res_fax.c: Error publishing ReceiveFax status message + + This patch replaces the invalid UTF-8 Station IDs with an empty string. + + ASTERISK-29312 #close + + Change-Id: Ieb00b6ecf67db3bfca787649caa8517f29d987db + +2021-02-25 13:55 +0000 [932eae69ab] Sean Bright + + * app_page.c: Don't fail to Page if beep sound file is missing + + ASTERISK-16799 #close + + Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4 + +2021-02-19 13:25 +0000 [4c9c5c985b] George Joseph + + * res_pjsip_refer: Refactor progress locking and serialization + + Although refer_progress_notify() always runs in the progress + serializer, the pjproject evsub module itself can cause the + subscription to be destroyed which then triggers + refer_progress_on_evsub_state() to clean it up. In this case, + it's possible that refer_progress_notify() could get the + subscription pulled out from under it while it's trying to use + it. + + At one point we tried to have refer_progress_on_evsub_state() + push the cleanup to the serializer and wait for its return before + returning to pjproject but since pjproject calls its state + callbacks with the dialog locked, this required us to unlock the + dialog while waiting for the serialized cleanup, then lock it + again before returning to pjproject. There were also still some + cases where other callers of refer_progress_notify() weren't + using the serializer and crashes were resulting. + + Although all callers of refer_progress_notify() now use the + progress serializer, we decided to simplify the locking so we + didn't have to unlock and relock the dialog in + refer_progress_on_evsub_state(). + + Now, refer_progress_notify() holds the dialog lock for its + duration and since pjproject also holds the dialog lock while + calling refer_progress_on_evsub_state() (which does the cleanup), + there should be no more chances for the subscription to be + cleaned up while still being used to send NOTIFYs. + + To be extra safe, we also now increment the session count on + the dialog when we create a progress object and decrement + the count when the progress is destroyed. + + ASTERISK-29313 + + Change-Id: I97a8bb01771a3c85345649b8124507f7622a8480 + +2021-02-24 16:05 +0000 [e5e49d7ecd] Kevin Harwell + + * res_rtp_asterisk: Add packet subtype during RTCP debug when relevant + + For some RTCP packet types the report count is actually the packet's subtype. + This was not being reflected in the packet debug output. + + This patch makes it so for some RTCP packet types a "Packet Subtype" is + now output in the debug replacing the "Reception reports" (i.e count). + + Change-Id: Id4f4b77bb37077a4c4f039abd6a069287bfefcb8 + +2021-02-16 12:33 +0000 [a81d07ea56] Joshua C. Colp + + * res_pjsip_session: Always produce offer on re-INVITE without SDP. + + When PJSIP receives a re-INVITE without an SDP offer the INVITE + session library will first call the on_create_offer callback and + if unavailable then use the active negotiated SDP as the offer. + + In some cases this would result in a different SDP then was + previously used without an incremented SDP version number. The two + known cases are: + + 1. Sending an initial INVITE with a set of codecs and having the + remote side answer with a subset. The active negotiated SDP would + have the pruned list but would not have an incremented SDP version + number. + + 2. Using re-INVITE for unhold. We would modify the active negotiated + SDP but would not increment the SDP version. + + To solve these, and potential other unknown cases, the on_create_offer + callback has now been implemented which produces a fresh offer with + incremented SDP version number. This better fits within the model + provided by the INVITE session library. + + ASTERISK-28452 + + Change-Id: I2d81048d54edcb80fe38fdbb954a86f0a58281a1 + +2021-02-23 05:28 +0000 [6d2614be68] Jaco Kroon + + * res_odbc_transaction: correctly initialise forcecommit value from DSN. + + Also improve the in-process documentation to clarify that the value is + initialised from the DSN and not default false, but that the DSN's value + is default false if unset. + + ASTERISK-29311 #close + + Change-Id: I46e2379f7b0656034442bce77cb37ccd4e61098d + Signed-off-by: Jaco Kroon + +2021-02-15 12:24 +0000 [e1126ffc10] Ben Ford + + * res_pjsip_session.c: Check topology on re-invite. + + Removes an unnecessary check for the conditional that compares the + stream topologies to see if they are equal to suppress re-invites. This + was a problem when a Digium phone received an INVITE that offered codecs + different than what it supported, causing Asterisk to send the + re-invite. + + ASTERISK-29303 + + Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63 + +2021-02-15 13:02 +0000 [b046e960af] Boris P. Korzun + + * res_config_pgsql: Limit realtime_pgsql() to return one (no more) record. + + Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function. + + ASTERISK-29293 #close + + Change-Id: If5a6d4b1072ea2e6e89059b21139d554a74b34f5 + +2019-09-13 08:02 +0000 [4d8fc97e4a] Ivan Poddubnyi + + * app_queue: Fix conversion of complex extension states into device states + + Queue members using dialplan hints as a state interface must handle + INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE. + + ASTERISK-28369 + + Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f + +2021-02-10 11:59 +0000 [725eca3bfa] Jaco Kroon + + * app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS + + This partially reverts commit 3d1bf3c537bba0416f691f48165fdd0a32554e8a, + specifically for app.h. + + This works with both gcc 9.3.0 and 10.2.0 now, both for C and C++ (as + tested with external modules). + + ASTERISK-29287 + + Change-Id: I5b9f02a9b290675682a1d13f1788fdda597c9fca + Signed-off-by: Jaco Kroon + +2021-02-05 06:29 +0000 [5894535fed] Alexander Traud + + * chan_sip: Filter pass-through audio/video formats away, again. + + Instead of looking for pass-through formats in the list of transcodable + formats (which is going to find nothing), go through the result which + is going to be the jointcaps of the tech_pvt of the channel. Finally, + only with that list, ast_format_cap_remove(.) is going to succeed. + + This restores the behaviour of Asterisk 1.8. However, it does not fix + ASTERISK_29282 because that issue report is about chan_sip and PJSIP. + Here, only chan_sip is fixed because PJSIP does not even call + ast_rtp_instance_available_formats -> ast_translate_available_format. + + Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34 + +2021-02-17 14:51 +0000 [b0f349a330] Jaco Kroon + + * func_odbc: Introduce minargs config and expose ARGC in addition to ARGn. + + minargs enables enforcing of minimum count of arguments to pass to + func_odbc, so if you're unconditionally using ARG1 through ARG4 then + this should be set to 4. func_odbc will generate an error in this case, + so for example + + [FOO] + minargs = 4 + + and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a + potentially leaked ARG4 from Gosub(). + + ARGC is needed if you're using optional argument, to verify whether or + not an argument has been passed, else it's possible to use a leaked ARGn + from Gosub (app_stack). So now you can safely do + ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. + + Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24 + Signed-off-by: Jaco Kroon + +2021-01-13 14:05 +0000 [6e695c867f] Sebastien Duthil + + * app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. + + ASTERISK-29244 + + Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5 + +2021-02-01 15:24 +0000 [5e998d8bd3] Kevin Harwell + + * AST-2021-002: Remote crash possible when negotiating T.38 + + When an endpoint requests to re-negotiate for fax and the incoming + re-invite is received prior to Asterisk sending out the 200 OK for + the initial invite the re-invite gets delayed. When Asterisk does + finally send the re-inivite the SDP includes streams for both audio + and T.38. + + This happens because when the pending topology and active topologies + differ (pending stream is not in the active) in the delayed scenario + the pending stream is appended to the active topology. However, in + the fax case the pending stream should replace the active. + + This patch makes it so when a delay occurs during fax negotiation, + to or from, the audio stream is replaced by the T.38 stream, or vice + versa instead of being appended. + + Further when Asterisk sent the re-invite with both audio and T.38, + and the endpoint responded with a declined T.38 stream then Asterisk + would crash when attempting to change the T.38 state. + + This patch also puts in a check that ensures the media state has a + valid fax session (associated udptl object) before changing the + T.38 state internally. + + ASTERISK-29203 #close + + Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09 + +2021-01-26 11:09 +0000 [389b8b0774] Alexander Traud + + * rtp: Enable srtp replay protection + + Add option "srtpreplayprotection" rtp.conf to enable srtp + replay protection. + + ASTERISK-29260 + Reported by: Alexander Traud + + Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458 + +2020-12-28 06:43 +0000 [7d15655f9d] Ivan Poddubnyi + + * res_pjsip_diversion: Fix adding more than one histinfo to Supported + + New responses sent within a PJSIP sessions are based on those that were + sent before. Therefore, adding/modifying a header once causes it to be + sent on all responses that follow. + + Sending 181 Call Is Being Forwarded many times first adds "histinfo" + duplicated more and more, and eventually overflows past the array + boundary. + + This commit adds a check preventing adding "histinfo" more than once, + and skipping it if there is no more space in the header. + + Similar overflow situations can also occur in res_pjsip_path and + res_pjsip_outbound_registration so those were also modified to + check the bounds and suppress duplicate Supported values. + + ASTERISK-29227 + Reported by: Ivan Poddubny + + Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322 + +2020-12-11 14:49 +0000 [e7b13df394] Sean Bright + + * res_rtp_asterisk.c: Fix signed mismatch that leads to overflow + + ASTERISK-29205 #close + + Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea + +2021-02-05 05:26 +0000 [492945ac60] Joshua C. Colp + + * pjsip: Make modify_local_offer2 tolerate previous failed SDP. + + If a remote side is broken and sends an SDP that can not be + negotiated the call will be torn down but there is a window + where a second 183 Session Progress or 200 OK that is forked + can be received that also attempts to negotiate SDP. Since + the code marked the SDP negotiation as being done and complete + prior to this it assumes that there is an active local and remote + SDP which it can modify, while in fact there is not as the SDP + did not successfully negotiate. Since there is no local or remote + SDP a crash occurs. + + This patch changes the pjmedia_sdp_neg_modify_local_offer2 + function to no longer assume that a previous SDP negotiation + was successful. + + ASTERISK-29196 + + Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e + +2021-02-09 11:25 +0000 [15b4080679] George Joseph + + * res_pjsip_refer: Always serialize calls to refer_progress_notify + + refer_progress_notify wasn't always being called from the progress + serializer. This could allow clearing notification->progress->sub + in one thread while another was trying to use it. + + * Instances where refer_progress_notify was being called in-line, + have been changed to use ast_sip_push_task(). + + Change-Id: Idcf1934c4e873f2c82e2d106f8d9f040caf9fa1e + +2021-01-11 14:20 +0000 [00b229c69c] Ben Ford + + * core_unreal: Fix T.38 faxing when using local channels. + + After some changes to streams and topologies, receiving fax through + local channels stopped working. This change adds a stream topology with + a stream of type IMAGE to the local channel pair and allows fax to be + received. + + ASTERISK-29035 #close + + Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb + +2021-02-02 02:33 +0000 [a96eb6de6c] Boris P. Korzun + + * format_wav: Support of MIME-type for wav16 + + Provided a support of a MIME-type for wav16. Added new MIME-type + for classic wav. + + ASTERISK-29275 #close + + Change-Id: I749bda287ba1ab20c1e0af5e4c0153817d47873b + +2021-02-05 02:33 +0000 [1f77c33c02] Alexander Traud + + * chan_sip: Allow [peer] without audio (text+video). + + Two previous commits, 620d9f4 and 6d980de, allow to set up a call + without audio, again. That was introduced originally with commit f04d5fb + but changed and broke over time. The original commit missed one + scenario: A [peer] section in sip.conf, which does not allow audio at + all. In that case, chan_sip rejected the call, although even when the + requester offered no audio. Now, chan_sip does not check whether there + is no audio format but checks whether there is no format in general. In + other words, if there is at least one format to offer, the call succeeds. + + However, to prevent calls with no-audio, chan_sip still rejects calls + when both call parties (caller = requester of the call *and* callee = + [peer] section in sip.conf) included audio. In such a case, it is + expected that the call should have audio. + + ASTERISK-29280 + + Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e + +2021-01-28 12:02 +0000 [91b0778791] George Joseph + + * chan_iax2.c: Require secret and auth method if encryption is enabled + + If there's no secret specified for an iax2 peer and there's no secret + specified in the dial string, Asterisk will crash if the auth method + requested by the peer is MD5 or plaintext. You also couldn't specify + a default auth method in the [general] section of iax.conf so if you + don't have static peers defined and just use the dial string, Asterisk + will still crash even if you have a secret specified in the dial string. + + * Added logic to iax2_call() and authenticate_reply() to print + a warning and hanhup the call if encryption is requested and + there's no secret or auth method. This prevents the crash. + + * Added the ability to specify a default "auth" in the [general] + section of iax.conf. + + ASTERISK-29624 + Reported by: N A + + Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254 + +2021-02-03 12:53 +0000 [4a71b08091] Sean Bright + + * app_read: Release tone zone reference on early return. + + Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a + +2021-01-27 11:42 +0000 [620d9f4782] Alexander Traud + + * chan_sip: Set up calls without audio (text+video), again. + + The previous commit 6d980de fixed this issue in the core of Asterisk. + With that, each channel technology can be used without audio + theoretically. Practically, the channel-technology driver chan_sip + turned out to have an invalid check preventing that. chan_sip tested + whether there is at least one audio format. However, chan_sip has to + test whether there is at least one format. More cannot be tested while + requesting chan_sip because only the [general] capabilities but not the + [peer] caps are known yet. And the [peer] caps might not be a subset or + show any intersection with the [general] caps. This change here fixes + this. + + The original commit f04d5fb, thirteen years ago, contained a software + bug as it passed ANY audio capability to the channel-technology driver. + Instead, it should have passed NO audio format. Therefore, this + addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8. + Then, Asterisk 10 changed that from ANY to NO, but nobody reported since + then. + + ASTERISK-29265 + + Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f + +2021-01-22 09:12 +0000 [55891227e8] Dan Cropp + + * chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable + + When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is + 0 when no protocl specific error + SIP example of failure, 3xx-6xx for the SIP error code received + + This allows applications to perform actions based on the failure + reason. + + ASTERISK-29252 #close + Reported-by: Dan Cropp + + Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4 + +2021-01-22 02:54 +0000 [6d980de282] Alexander Traud + + * channel: Set up calls without audio (text+video), again. + + ASTERISK-29259 + + Change-Id: Ib6a6550e0e08355745d66da8e60ef49e81f9c6c5 + +2021-01-22 07:12 +0000 [9b5d20e3d5] Mark Petersen + + * res/res_pjsip.c: allow user=phone when number contain *# + + if From number contain * or # asterisk will not add user=phone + + Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY + this is a problem when you want to send call to ISUP + as they will disregard the From header and either replace From with anonymous or with p-asserted-identity + + ASTERISK-29261 + Reported by: Mark Petersen + Tested by: Mark Petersen + + Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4 + +2021-01-21 13:28 +0000 [4aff42b274] Alexander Traud + + * chan_sip: SDP: Reject audio streams correctly. + + This completes the fix for ASTERISK_24543. Only when the call is an + outgoing call, consult and append the configured format capabilities + (p->caps). When all audio formats got rejected the negotiated format + capabilities (p->jointcaps) contain no audio formats for incoming + calls. This is required when there are other accepted media streams. + + ASTERISK-29258 + + Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9 + +2021-01-22 11:17 +0000 [05472da92b] Ivan Poddubnyi + + * main/frame: Add missing control frame names to ast_frame_subclass2str + + Log proper control frame names instead of "Unknown control '14'", etc. + + Change-Id: I1724f2f4d1b064b25a5c93a7da0cb03be5143935 + +2021-01-23 07:15 +0000 [92f5cf7f2d] Boris P. Korzun + + * res_musiconhold: Add support of various URL-schemes by MoH. + + Provided a support of variuos URL-schemes for res_musiconhold, + registered by ast_bucket_scheme_register(). + + ASTERISK-29262 #close + + Change-Id: If0ea8697587353dce358a70035d82649fd4632b6 + +2021-01-08 10:02 +0000 [060ce10163] Jaco Kroon + + * AC_HEADER_STDC causes a compile failure with autoconf 2.70 + + From https://www.mail-archive.com/bug-autoconf@gnu.org/msg04408.html + + > ... the long-obsolete AC_HEADER_STDC, previously used internally by + > AC_INCLUDES_DEFAULT, used AC_EGREP_HEADER. The AC_HEADER_STDC macro + > is now a no-op (and is not used at all within Autoconf anymore), so + > that change is likely what made the first use of AC_EGREP_HEADER the + > one inside the if condition, causing the observed results. + + The implication is that the test does nothing anyway, and due to it + being a no-op from 2.70 onwards, results in the required not being set + to yes, resulting in ./configure to fail. + + Change-Id: Ic1ff38d87f791fbf1f2a80512f81bb7110392460 + Signed-off-by: Jaco Kroon + +2021-01-15 03:33 +0000 [10a0a0c59b] Alexander Traud + + * pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang. + + Otherwise, Clang 10 warned because of logical-not-parentheses. + + Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79 + +2021-01-15 05:09 +0000 [df6afadf26] Alexander Traud + + * res_pjsip_session: Avoid sometimes-uninitialized warning with Clang. + + ASTERISK-29248 + + Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556 + +2021-01-14 08:47 +0000 [6d2bec7028] Sean Bright + + * res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet + + The last argument to ast_copy_string() is the buffer size, not the + number of characters, so we add 1 to avoid stamping out the final \n + in the persisted SUBSCRIBE message. + + Change-Id: I019b78942836f57965299af15d173911fcead5b2 + +2021-01-11 14:25 +0000 [948ceb1228] Ben Ford + + * chan_pjsip.c: Add parameters to frame in indicate. + + There are a couple of parameters (datalen and data) that do not get set + in chan_pjsip_indicate which could cause an Invalid message to pop up + for things such as fax. This patch adds them to the frame. + + Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8 + +2020-12-22 04:42 +0000 [24e678b9bb] Robert Cripps + + * res/res_pjsip_session.c: Check that media type matches in + function ast_sip_session_media_state_add. + + Check ast_media_type matches when a ast_sip_session_media is found + otherwise when transitioning from say image to audio, the wrong + session is returned in the first if statement. + + ASTERISK-29220 #close + + Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d + +2020-12-30 07:56 +0000 [c559667868] Jean Aunis + + * Stasis/messaging: tech subscriptions conflict with endpoint subscriptions. + + When both a tech subscription and an endpoint subscription exist for a given + endpoint, TextMessageReceived events are dispatched to the tech subscription + only. + + ASTERISK-29229 + + Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8 + +2020-12-23 08:44 +0000 [1c05667cfc] Alexander Traud + + * chan_sip: SDP: Sidestep stream parsing when its media is disabled. + + Previously, chan_sip parsed all known media streams in an SDP offer + like video (and text) even when videosupport=no (and textsupport=no). + This wasted processor power. Furthermore, chan_sip accepted SDP offers, + including no audio but just video (or text) streams although + videosupport=no (or textsupport=no). Finally, chan_sip denied the whole + offer instead of individual streams when they had encryption (SDES-sRTP) + unexpectedly enabled. + + ASTERISK-29238 + ASTERISK-29237 + ASTERISK-29222 + + Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755 + +2020-12-29 12:16 +0000 [f2aa6c7017] Ivan Poddubnyi + + * chan_pjsip: Assign SIPDOMAIN after creating a channel + + session->channel doesn't exist until chan_pjsip creates it, so intead of + setting a channel variable every new incoming call sets one and the same + global variable. + + This patch moves the code to chan_pjsip so that SIPDOMAIN is set on + a newly created channel, it also removes a misleading reference to + channel->session used to fetch call pickup configuraion. + + ASTERISK-29240 + + Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755 + +2020-12-31 05:53 +0000 [134d2e729d] Ivan Poddubnyi + + * chan_pjsip: Stop queueing control frames twice on outgoing channels + + The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice. + This resulted in extra noise in logs (for example, "is making progress" + and "is ringing" get logged twice by app_dial), as well as in noise in + signalling: one incoming 183 Session Progress results in 2 outgoing 183-s. + + This change splits the response handler into 2 functions: + - one for updating HANGUPCAUSE, which is still called twice, + - another that does the rest, which is called only once as before. + + ASTERISK-28016 + Reported-by: Alex Hermann + + ASTERISK-28549 + Reported-by: Gant Liu + + ASTERISK-28185 + Reported-by: Julien + + Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940 + +2020-12-18 13:06 +0000 [2d3441772b] Jaco Kroon + + * contrib/systemd: Added note on common issues with systemd and asterisk + + With newer version of linux /var/run/ is a symlink to /run/ that has + been turned into tmpfs. + + Added note that if asterisk has to bind to a specific IP that + systemd has to wait until the network is up. + + Added note on how to make sure that the environment variable + HOSTNAME is included. + + ASTERISK-29216 + Reported by: Mark Petersen + Tested by: Mark Petersen + + Change-Id: Ib3e560655befd3e99eec743687144f5569533379 + +2021-01-07 08:40 +0000 [9a4486e9fb] George Joseph + + * Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff" + + This reverts commit 2fe76dd816706f045ecbc44bf8ad6498977415b3. + + Reason for revert: Too many issues reported. Need to research and correct. + + ASTERISK-29230 + ASTERISK-29231 + Reported by: Michael Maier + + Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb + +2020-12-18 13:06 +0000 [c797500956] Jaco Kroon + + * func_lock: fix multiple-channel-grant problems. + + Under contention it becomes possible that multiple channels will be told + they successfully obtained the lock, which is a bug. Please refer + + ASTERISK-29217 + + This introduces a couple of changes. + + 1. Replaces requesters ao2 container with simple counter (we don't + really care who is waiting for the lock, only how many). This is + updated undex ->mutex to prevent memory access races. + 2. Correct semantics for ast_cond_timedwait() as described in + pthread_cond_broadcast(3P) is used (multiple threads can be released + on a single _signal()). + 3. Module unload races are taken care of and memory properly cleaned + up. + + Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29 + Signed-off-by: Jaco Kroon + +2020-12-23 11:41 +0000 [4e038c1eaa] Jaco Kroon + + * pbx_lua: Add LUA_VERSIONS environment variable to ./configure. + + On Gentoo it's possible to have multiple lua versions installed, all + with a path of /usr, so it's not possible to use the current --with-lua + option to determisticly pin to a specific version as is required by the + Gentoo PMS standards. + + This environment variable allows to lock to specific versions, + unversioned check will be skipped if this variable is supplied. + + Change-Id: I8c403eda05df25ee0193960262ce849c7d2fd088 + Signed-off-by: Jaco Kroon + +2020-12-23 13:06 +0000 [3bcf483373] Kevin Harwell + + * app_mixmonitor: cleanup datastore when monitor thread fails to launch + + launch_monitor_thread is responsible for creating and initializing + the mixmonitor, and dependent data structures. There was one off + nominal path after the datastore gets created that triggers when + the channel being monitored is hung up prior to monitor starting + itself. + + If this happened the monitor thread would not "launch", and the + mixmonitor object and associated objects are freed, including the + underlying datastore data object. However, the datastore itself was + not removed from the channel, so when the channel eventually gets + destroyed it tries to access the previously freed datastore data + and crashes. + + This patch removes and frees datastore object itself from the channel + before freeing the mixmonitor object thus ensuring the channel does + not call it when destroyed. + + ASTERISK-28947 #close + + Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8 + +2020-12-24 09:03 +0000 [44d68bd56b] Sean Bright + + * app_voicemail: Prevent deadlocks when out of ODBC database connections + + ASTERISK-28992 #close + + Change-Id: Ia7d608924036139ee2520b840d077762d02668d0 + +2020-12-07 16:59 +0000 [ffa87ecade] Dan Cropp + + * chan_pjsip: Incorporate channel reference count into transfer_refer(). + + Add channel reference count for PJSIP REFER. The call could be terminated + prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY + occurred several minutes later, it would attempt to access a session which was + no longer valid. Terminate event subscription if pjsip_xfer_initiate() or + pjsip_xfer_send_request() fails in transfer_refer(). + + ASTERISK-29201 #close + Reported-by: Dan Cropp + + Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435 + +2020-12-22 17:40 +0000 [4274a4a7dd] Kevin Harwell + + * pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type + + A prior patch segmented channel snapshots, and changed the underlying + data object type associated with ast_channel_snapshot_type stasis + messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but + now it type ast_channel_snapshot_update. + + When publishing ast_channel_snapshot_type in pbx_realtime the + ast_channel_snapshot was being passed in as the message data + object. When a handler, expecting a data object type of + ast_channel_snapshot_update, dereferenced this value a crash + would occur. + + This patch makes it so pbx_realtime now uses the expected type, and + channel snapshot publish method when publishing. + + ASTERISK-29168 #close + + Change-Id: I9a2cfa0ec285169317f4b9146e4027da8a4fe896 + +2020-12-18 09:16 +0000 [1b74555fcf] Sean Bright + + * asterisk: Export additional manager functions + + Rename check_manager_enabled() and check_webmanager_enabled() to begin + with ast_ so that the symbols are automatically exported by the + linker. + + ASTERISK~29184 + + Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9 + +2020-12-19 11:54 +0000 [505939c9ed] Nick French + + * res_pjsip: Prevent segfault in UDP registration with flow transports + + Segfault occurs during outbound UDP registration when all + transport states are being iterated over. The transport object + in the transport is accessed, but flow transports have a NULL + transport object. + + Modify to not iterate over any flow transport + + ASTERISK-29210 #close + + Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a + +2020-12-01 08:11 +0000 [80c14f74bc] Alexander Traud + + * codecs: Remove test-law. + + This was dead code, test code introduced with Asterisk 13. This was + found while analyzing ASTERISK_28416 and ASTERISK_29185. This change + partly fixes, not closes those two issues. + + Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940 + +2020-12-22 02:58 +0000 [51e2187a14] Torrey Searle + + * res/res_pjsip_diversion: prevent crash on tel: uri in History-Info + + Add a check to see if the URI is a Tel URI and prevent crashing on + trying to retrieve the reason parameter. + + ASTERISK-29191 + ASTERISK-29219 + + Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37 + (cherry picked from commit a7aea71e60d513af82c6e3825e2308e063139b63) + +2020-12-26 12:14 +0000 [058bc0d593] Richard Mudgett + + * chan_vpb.cc: Fix compile errors. + + Fix the usual compile problem when someone adds a new callback to struct + ast_channel_tech. + + Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32 + +2020-12-26 11:42 +0000 [6d7af72559] Richard Mudgett + + * res_pjsip_session.c: Fix compiler warnings. + + AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an + unsigned long on all machines. + + Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338 + +2020-12-13 06:03 +0000 [02c4b2ac60] Sungtae Kim + + * res_pjsip_session: Fixed NULL active media topology handle + + Added NULL pointer check to prevent Asterisk crash. + + ASTERISK-29215 + + Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95 + +2020-12-11 13:27 +0000 [357510cec3] Sean Bright + + * app_chanspy: Spyee information missing in ChanSpyStop AMI Event + + The documentation in the wiki says there should be spyee-channel + information elements in the ChanSpyStop AMI event. + + https://wiki.asterisk.org/wiki/x/Xc5uAg + + However, this is not the case in Asterisk <= 16.10.0 Version. We're + using these Spyee* arguments since Asterisk 11.x, so these arguments + vanished in Asterisk 12 or higher. + + For maximum compatibility, we still send the ChanSpyStop event even if + we are not able to find any 'Spyee' information. + + ASTERISK-28883 #close + + Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f + +2020-11-30 19:27 +0000 [91fc57f56b] Sungtae Kim + + * res_ari: Fix wrong media uri handle for channel play + + Fixed wrong null object handle in + /channels//play request handler. + + ASTERISK-29188 + + Change-Id: I6691c640247a51ad15f23e4a203ca8430809bafe + +2020-12-10 09:09 +0000 [7d4ae7dc18] George Joseph + + * logger.c: Automatically add a newline to formats that don't have one + + Scope tracing allows you to not specify a format string or variable, + in which case it just prints the indent, file, function, and line + number. The trace output automatically adds a newline to the end + in this case. If you also have debugging turned on for the module, + a debug message is also printed but the standard log functionality + which prints it doesn't add the newline so you have messages + that don't break correctly. + + * format_log_message_ap(), which is the common log + message formatter for all channels, now adds a + newline to the end of format strings that don't + already have a newline. + + ASTERISK-29209 + Reported by: Alexander Traud + + Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da + +2020-12-08 11:37 +0000 [0b10995811] Pirmin Walthert + + * res_pjsip_nat.c: Create deep copies of strings when appropriate + + In rewrite_uri asterisk was not making deep copies of strings when + changing the uri. This was in some cases causing garbage in the route + header and in other cases even crashing asterisk when receiving a + message with a record-route header set. Thanks to Ralf Kubis for + pointing out why this happens. A similar problem was found in + res_pjsip_transport_websocket.c. Pjproject needs as well to be patched + to avoid garbage in CANCEL messages. + + ASTERISK-29024 #close + + Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b + +2020-12-10 17:06 +0000 [5e426987c2] Nathan Bruning + + * res_musiconhold: Don't crash when real-time doesn't return any entries + + ASTERISK-29211 #close + + Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877 + +2020-12-16 06:17 +0000 [9ee1f7154f] Joshua C. Colp + + * res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent. + + This adds support for both Digium and Sangoma user agent strings + for the Sangoma specific body supplement. + + Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482 + +2020-10-29 12:21 +0000 [6475fe3dd7] Joshua C. Colp + + * pjsip: Match lifetime of INVITE session to our session. + + In some circumstances it was possible for an INVITE + session to be destroyed while we were still using it. + This occurred due to the reference on the INVITE session + being released internally as a result of its state + changing to DISCONNECTED. + + This change adds a reference to the INVITE session + which is released when our own session is destroyed, + ensuring that the INVITE session remains valid for + the lifetime of our session. + + ASTERISK-29022 + + Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9 + +2020-11-21 11:51 +0000 [90fd1fd96a] Sean Bright + + * res_http_media_cache.c: Set reasonable number of redirects + + By default libcurl does not follow redirects, so we explicitly enable + it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl + will follow up to CURLOPT_MAXREDIRS redirects, which by default is + configured to be unlimited. + + This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If + we determine at some point that this needs to be increased on + configurable it is a trivial change. + + ASTERISK-29173 #close + + Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30 + +2020-10-29 06:25 +0000 [b08427134f] laszlovl + + * Introduce astcachedir, to be used for temporary bucket files + + As described in the issue, /tmp is not a suitable location for a + large amount of cached media files, since most distributions make + /tmp a RAM-based tmpfs mount with limited capacity. + + I opted for a location that can be configured separately, as opposed + to using a subdirectory of spooldir, given the different storage + profile (transient files vs files that might stay there indefinitely). + + This commit just makes the cache directory configurable, and changes + the default location from /tmp to /var/cache/asterisk. + + ASTERISK-29143 + + Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d + +2020-11-23 14:56 +0000 [c8b6340023] Sean Bright + + * media_cache: Fix reference leak with bucket file metadata + + Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6 + +2020-11-24 00:55 +0000 [ab7a08b4ef] Stanislav + + * res_pjsip_stir_shaken: Fix module description + + the 'J' is missing in module description. + "PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk" + + ASTERISK-29175 #close + + Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a + +2020-10-12 05:30 +0000 [eda3679c1c] Joshua C. Colp + + * voicemail: add option 'e' to play greetings as early media + + When using this option, answering the channel is deferred until + all prompts/greetings have been played and the caller is about + to leave their message. + + ASTERISK-29118 #close + + Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb + +2020-11-02 01:24 +0000 [b91fb3c396] Alexander Traud + + * loader: Sync load- and build-time deps. + + In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO. + + ASTERISK-29148 + + Change-Id: I254dd33194ae38d2877b8021c57c2a5deb6bbcd2 + +2020-11-18 13:11 +0000 [d04b5903d1] Sean Bright + + * CHANGES: Remove already applied CHANGES update + + Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a + +2020-11-17 14:19 +0000 [fba10fb54c] Alexander Greiner-Baer + + * res_pjsip: set Accept-Encoding to identity in OPTIONS response + + + + RFC 3261 says that the Accept-Encoding header should be present + in an options response. Permitted values according to RFC 2616 + are only compression algorithms like gzip or the default identity + encoding. Therefore "text/plain" is not a correct value here. + As long as the header is hard coded, it should be set to "identity". + + Without this fix an Alcatel OmniPCX periodically logs warnings like + "[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed" + on a SIP Trunk. + + ASTERISK-29165 #close + + Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840 + +2020-11-04 07:39 +0000 [103d7da3bb] Alexander Traud + + * chan_sip: Remove unused sip_socket->port. + + 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port + vanished. However, the struct member itself and all seven set/uses + remained as dead code. + + ASTERISK-28798 + + Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59 + +2020-11-13 06:19 +0000 [8cb439f7e4] Boris P. Korzun + + * bridge_basic: Fixed setup of recall channels + + Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641. + common_recall_channel_setup() setups common things on the recalled transfer + target, but used same target as source instead trasfered. + + ASTERISK-29161 #close + + Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f + +2020-11-03 02:27 +0000 [7c355d78cb] Alexander Traud + + * modules.conf: Align the comments for more conclusiveness. + + Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306 + +2020-11-11 08:55 +0000 [73f458b1e0] George Joseph + + * app_queue: Fix deadlock between update and show queues + + Operations that update queues when shared_lastcall is set lock the + queue in question, then have to lock the queues container to find the + other queues with the same member. On the other hand, __queues_show + (which is called by both the CLI and AMI) does the reverse. It locks + the queues container, then iterates over the queues locking each in + turn to display them. This creates a deadlock. + + * Moved queue print logic from __queues_show to a separate function + that can be called for a single queue. + + * Updated __queues_show so it doesn't need to lock or traverse + the queues container to show a single queue. + + * Updated __queues_show to snap a copy of the queues container and iterate + over that instead of locking the queues container and iterating over + it while locked. This prevents us from having to hold both the + container lock and the queue locks at the same time. This also + allows us to sort the queue entries. + + ASTERISK-29155 + + Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41 + +2020-11-02 13:53 +0000 [2fe76dd816] George Joseph + + * res_pjsip_outbound_registration.c: Use our own scheduler and other stuff + + * Instead of using the pjproject timer heap, we now use our own + pjsip_scheduler. This allows us to more easily debug and allows us to + see times in "pjsip show/list registrations" as well as being able to + see the registrations in "pjsip show scheduled_tasks". + + * Added the last registration time, registration interval, and the next + registration time to the CLI output. + + * Removed calls to pjsip_regc_info() except where absolutely necessary. + Most of the calls were just to get the server and client URIs for log + messages so we now just save them on the client_state object when we + create it. + + * Added log messages where needed and updated most of the existong ones + to include the registration object name at the start of the message. + + Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2 + +2020-11-02 13:53 +0000 [5a4640d208] George Joseph + + * pjsip_scheduler.c: Add type ONESHOT and enhance cli show command + + * Added a ONESHOT type that never reschedules. + + * Added "like" capability to "pjsip show scheduled_tasks" so you can do + the following: + + CLI> pjsip show scheduled_tasks like outreg + PJSIP Scheduled Tasks: + + Task Name Interval Times Run ... + ============================================= ========= ========= ... + pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ... + pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ... + + * Fixed incorrect display of "Next Start". + + * Compacted the displays of times in the CLI. + + * Added two new functions (ast_sip_sched_task_get_times2, + ast_sip_sched_task_get_times_by_name2) that retrieve the interval, + next start time, and next run time in addition to the times already + returned by ast_sip_sched_task_get_times(). + + Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3 + +2020-10-02 14:32 +0000 [cc7eb72f65] Alexei Gradinari + + * sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data + + The data can be freed if the old object '_data' is the same object as + new 'data'. Because at first the object is unreferenced which can lead + to destroying it. + + This could happened in res_pjsip_pubsub when the publication is updated + which could lead to segfault in function publish_expire. + + Change-Id: I0164f57c387243510bdbd2f8dcf33377b6c202da + +2020-10-30 11:43 +0000 [b52acb87b0] Alexander Traud + + * res_pjsip/config_transport: Load and run without OpenSSL. + + ASTERISK-28933 + Reported-by: Walter Doekes + + Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f + +2020-10-30 05:53 +0000 [64d2de19ee] Alexander Traud + + * res_stir_shaken: Include OpenSSL headers where used actually. + + This avoids the inclusion of the OpenSSL headers in the public header, + which avoids one external library dependency in res_pjsip_stir_shaken. + + Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4 + +2020-10-18 13:40 +0000 [bc58e84f47] Dovid Bender + + * func_curl.c: Allow user to set what return codes constitute a failure. + + Currently any response from res_curl where we get an answer from the + web server, regardless of what the response is (404, 403 etc.) Asterisk + currently treats it as a success. This patch allows you to set which + codes should be considered as a failure by Asterisk. If say we set + failurecodes=404,403 then when using curl in realtime if a server gives + a 404 error Asterisk will try to failover to the next option set in + extconfig.conf + + ASTERISK-28825 + + Reported by: Dovid Bender + Code by: Gobinda Paul + + Change-Id: I94443e508343e0a3e535e51ea6e0562767639987 + +2020-11-04 15:08 +0000 [b82f880647] Kevin Harwell + + * AST-2020-001 - res_pjsip: Return dialog locked and referenced + + pjproject returns the dialog locked and with a reference. However, + in Asterisk the method that handles this decrements the reference + and removes the lock prior to returning. This makes it possible, + under some circumstances, for another thread to free said dialog + before the thread that created it attempts to use it again. Of + course when the thread that created it tries to use a freed dialog + a crash can occur. + + This patch makes it so Asterisk now returns the newly created + dialog both locked, and with an added reference. This allows the + caller to de-reference, and unlock the dialog when it is safe to + do so. + + In the case of a new SIP Invite the lock, and reference are now + held for the entirety of the new invite handling process. + Otherwise it's possible for the dialog, or its dependent objects, + like the transaction, to disappear. For example if there is a TCP + transport error. + + ASTERISK-29057 #close + + Change-Id: I5ef645a47829596f402cf383dc02c629c618969e + (cherry picked from commit 6baa4b53bef5d9c53692f22cf146215b42de1e89) + +2020-11-03 10:38 +0000 [cd8f8b94f8] Ben Ford + + * AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. + + If Asterisk sends out and INVITE and receives a challenge with a + different nonce value each time, it will continually send out INVITEs, + even if the call is hung up. The endpoint must be configured for + outbound authentication in order for this to occur. A limit has been set + on outbound INVITEs so that, once reached, Asterisk will stop sending + INVITEs and the transaction will terminate. + + ASTERISK-29013 + + Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7 + +2020-10-29 10:21 +0000 [a5d55fc9e1] Sean Bright + + * sip_to_pjsip.py: Handle #include globs and other fixes + + * Wildcards in #includes are now properly expanded + + * Implement operators for Section class to allow sorting + + ASTERISK-29142 #close + + Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df + +2020-10-29 03:55 +0000 [57ee79a563] Alexander Traud + + * Compiler fixes for GCC with -Og + + ASTERISK-29144 + + Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62 + +2020-10-30 03:46 +0000 [28faafd1c4] Alexander Traud + + * Compiler fixes for GCC when printf %s is NULL + + ASTERISK-29146 + + Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41 + +2020-10-29 08:59 +0000 [914aecb8d8] Alexander Traud + + * Compiler fixes for GCC with -Os + + ASTERISK-29145 + + Change-Id: I9af705f2b9725c53141aef5d0ff512a1800f073c + +2020-10-23 10:26 +0000 [cd32317691] Alexander Traud + + * chan_sip: On authentication, pick MD5 for sure. + + RFC 8760 added new digest-access-authentication schemes. Testing + revealed that chan_sip does not pick MD5 if several schemes are offered + by the User Agent Server (UAS). This change does not implement any of + the new schemes like SHA-256. This change makes sure, MD5 is picked so + UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can + still be used. This should have worked since day one because SIP/2.0 + already envisioned several schemes (see RFC 3261 and its augmented BNF + for 'algorithm' which includes 'token' as third alternative; note: if + 'algorithm' was not present, MD5 is still assumed even in RFC 7616). + + Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd + +2020-06-04 09:23 +0000 [1650d50e91] Walter Doekes + + * main/say: Work around gcc 9 format-truncation false positive + + Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0 + Warning: + say.c:2371:24: error: ‘%d’ directive output may be truncated writing + between 1 and 11 bytes into a region of size 10 + [-Werror=format-truncation=] + 2371 | snprintf(buf, 10, "%d", num); + say.c:2371:23: note: directive argument in the range [-2147483648, 9] + + That's not possible though, as the if() starts out checking for (num < 0), + making this Warning a false positive. + + (Also replaced some elseif with elseif while in the vicinity.) + + Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a + +2020-10-19 15:31 +0000 [c62193c5de] Kevin Harwell + + * res_pjsip, res_pjsip_session: initialize local variables + + This patch initializes a couple of local variables to some default values. + Interestingly, in the 'pj_status_t dlg_status' case the value not being + initialized caused memory to grow, and not be recovered, in the off nominal + path (at least on my machine). + + Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7 + +2020-10-23 09:55 +0000 [f3452c85e5] Alexander Traud + + * install_prereq: Add GMime 3.0. + + Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not + come with GMime 3.0. aptitude ignores any missing package. Therefore, + it installs the correct package(s). However, in Ubuntu 18.04 LTS and + Ubuntu 20.04 LTS, both versions are installed alongside although only + one is really needed. + + Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7 + +2020-10-23 09:49 +0000 [db4320a6a0] Alexander Traud + + * BuildSystem: Enable Lua 5.4. + + Note to maintainers: Lua 5.4, Lua 5.3, and Lua 5.2 have not been tested + at runtime with pbx_lua. Until then, use the lowest available version + of Lua, if you enabled the module pbx_lua at all. + + Change-Id: Ie5270448b11fcb4e2a53d899e4fe7fea793ce7e0 + +2020-10-13 12:15 +0000 [bd98e153d1] Nick French + + * res_pjsip_session: Restore calls to ast_sip_message_apply_transport() + + Commit 44bb0858cb3ea6a8db8b8d1c7fedcfec341ddf66 ("debugging: Add enough + to choke a mule") accidentally removed calls to + ast_sip_message_apply_transport when it was attempting to just add + debugging code. + + The kiss of death was saying that there were no functional changes in + the commit comment. + + This makes outbound calls that use the 'flow' transport mechanism fail, + since this call is used to relay headers into the outbound INVITE + requests. + + ASTERISK-29124 #close + + Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a + +2020-10-22 11:21 +0000 [8f33e23dfb] Sean Bright + + * features.conf.sample: Sample sound files incorrectly quoted + + ASTERISK-29136 #close + + Change-Id: I3186536d65a50014c8da4780c9224919caa81440 + +2020-10-12 00:45 +0000 [0190e706b8] Andrew Siplas + + * logger.conf.sample: add missing comment mark + + Add missing comment mark from stock configuration. + + ASTERISK-29123 #close + + Change-Id: I4f94eb4544166bca8af4c17fd11edee3c6980620 + +2020-10-06 10:32 +0000 [dcd2ed69a3] Joshua C. Colp + + * res_pjsip: Adjust outgoing offer call pref. + + This changes the outgoing offer call preference + default option to match the behavior of previous + versions of Asterisk. + + The additional advanced codec negotiation options + have also been removed from the sample configuration + and marked as reserved for future functionality in + XML documentation. + + The codec preference options have also been fixed to + enforce local codec configuration. + + ASTERISK-29109 + + Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2 + +2020-09-30 15:00 +0000 [fa023cbfa0] Sean Bright + + * tcptls.c: Don't close TCP client file descriptors more than once + + ASTERISK-28430 #close + + Change-Id: Ib556b0a0c95cca939e956886214ec8d828d89606 + +2020-10-05 10:44 +0000 [61116d5dbc] Jean Aunis + + * resource_endpoints.c: memory leak when providing a 404 response + + When handling a send_message request to a non-existing endpoint, the response's + body is overriden and not properly freed. + + ASTERISK-29108 + + Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8 + +2020-08-28 16:32 +0000 [56028426de] Kevin Harwell + + * Logging: Add debug logging categories + + Added debug logging categories that allow a user to output debug + information based on a specified category. This lets the user limit, + and filter debug output to data relevant to a particular context, + or topic. For instance the following categories are now available for + debug logging purposes: + + dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, + stun, stun_packet + + These debug categories can be enable/disable via an Asterisk CLI command. + + While this overrides, and outputs debug data, core system debugging is + not affected by this patch. Statements still output at their appropriate + debug level. As well backwards compatibility has been maintained with + past debug groups that could be enabled using the CLI (e.g. rtpdebug, + stundebug, etc.). + + ASTERISK-29054 #close + + Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849 + +2020-09-29 13:04 +0000 [51cba591e3] Sean Bright + + * pbx.c: On error, ast_add_extension2_lockopt should always free 'data' + + In the event that the desired extension already exists, + ast_add_extension2_lockopt() will free the 'data' it is passed before + returning an error, so we should not be freeing it ourselves. + + Additionally, there were two places where ast_add_extension2_lockopt() + could return an error without also freeing the 'data' pointer, so we + add that. + + ASTERISK-29097 #close + + Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae + +2020-09-24 13:46 +0000 [773f424c7f] George Joseph + + * app_confbridge/bridge_softmix: Add ability to force estimated bitrate + + app_confbridge now has the ability to set the estimated bitrate on an + SFU bridge. To use it, set a bridge profile's remb_behavior to "force" + and set remb_estimated_bitrate to a rate in bits per second. The + remb_estimated_bitrate parameter is ignored if remb_behavior is something + other than "force". + + Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a + +2020-09-29 19:57 +0000 [4b5ed817bd] Sean Bright + + * app_voicemail.c: Document VMSayName interruption behavior + + ASTERISK-26424 #close + + Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0 + +2020-09-22 22:39 +0000 [9c0ded6e76] Holger Hans Peter Freyther + + * res_pjsip_sdp_rtp: Fix accidentally native bridging calls + + Stop advertising RFC2833 support on the rtp_engine when DTMF mode is + auto but no tel_event was found inside SDP file. + + On an incoming call create_rtp will be called and when session->dtmf is + set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without + looking at the SDP file. + + Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND + but continued to advertise RFC2833 support. + + This meant the native_rtp bridge would falsely consider the two channels + as compatible. In addition to changing the DTMF mode we now set or + remove the AST_RTP_PROPERTY_DTMF. + + The property is checked in ast_rtp_dtmf_compatible and called by + native_rtp_bridge_compatible. + + ASTERISK-29051 #close + + Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287 + +2020-09-28 07:42 +0000 [990c72bbcf] laszlovl + + * res_musiconhold: Load all realtime entries, not just the first + + ASTERISK-29099 + + Change-Id: I45636679c0fb5a5f59114c8741626631a604e8a6 + +2020-09-23 04:05 +0000 [e831952eba] Jasper van der Neut + + * channels: Don't dereference NULL pointer + + Check result of ast_translator_build_path against NULL before dereferencing. + + ASTERISK-29091 + + Change-Id: Ia3538ea190bd371f70c9dd49984b021765691b29 + +2020-09-24 09:54 +0000 [e7bd97e2e5] Torrey Searle + + * res_pjsip_diversion: fix double 181 + + Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and + AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice, + resulting in to 181 being generated. + + Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab + +2020-09-24 11:47 +0000 [505211551a] Sean Bright + + * res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs + + Change-Id: I41e77a04e4a523f4ed61a7a20b738ffd42be441e + +2020-09-23 15:20 +0000 [16dfe8f03f] Sean Bright + + * dsp.c: Update calls to ast_format_cmp to check result properly + + ASTERISK-28311 #close + + Change-Id: Ib1ce8fc1a8752751f5bf3615c59245532dfd9aa2 + +2020-09-22 05:05 +0000 [23e427bbd2] Joshua C. Colp + + * res_pjsip_session: Fix stream name memory leak. + + When constructing a stream name based on the media type + and position the allocated name was not being freed + causing a leak. + + Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de + +2020-09-18 08:09 +0000 [b11b49945b] Sean Bright + + * func_curl.c: Prevent crash when using CURLOPT(httpheader) + + Because we use shared thread-local cURL instances, we need to ensure + that the state of the cURL instance is correct before each invocation. + + In the case of custom headers, we were not resetting cURL's internal + HTTP header pointer which could result in a crash if subsequent + requests do not configure custom headers. + + ASTERISK-29085 #close + + Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d + +2020-09-18 15:02 +0000 [0aaf9aa6de] Sean Bright + + * res_musiconhold: Start playlist after initial announcement + + Only track our sample offset if we are playing a non-announcement file, + otherwise we will skip that number of samples when we start playing the + first MoH file. + + ASTERISK-24329 #close + + Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc + +2020-09-22 05:13 +0000 [f67f5676b7] Joshua C. Colp + + * res_pjsip_session: Fix session reference leak. + + The ast_sip_dialog_get_session function returns the session + with reference count increased. This was not taken into + account and was causing sessions to remain around when they + should not be. + + ASTERISK-29089 + + Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8 + +2020-09-16 08:01 +0000 [b4ab0dd41a] Michal Hajek + + * res_stasis.c: Add compare function for bridges moh container + + Sometimes not play MOH on bridge. + + ASTERISK-29081 + Reported-by: Michal Hajek + + Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232 + +2020-09-17 11:40 +0000 [923d95cc84] George Joseph + + * logger.h: Fix ast_trace to respect scope_level + + ast_trace() was always emitting messages when it's level was set to -1 + because it was ignoring scope_level. + + Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506 + +2020-09-15 10:48 +0000 [52ca2323aa] Sean Bright + + * chan_sip.c: Don't build by default + + ASTERISK-29083 #close + + Change-Id: I9ea25fba3ba8f63a47886894bd966e0f08d5e003 + +2020-09-15 15:44 +0000 [5a0e1d256d] Sean Bright + + * audiosocket: Fix module menuselect descriptions + + The module description needs to be on the same line as the + AST_MODULE_INFO or it is not parsed correctly. + + Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21 + +2020-09-17 13:01 +0000 [39bb45cdfc] George Joseph + + * bridge_softmix/sfu_topologies_on_join: Ignore topology change failures + + When a channel joins a bridge, we do topology change requests on all + existing channels to add the new participant to them. However the + announcer channel will return an error because it doesn't support + topology in the first place. Unfortunately, there doesn't seem to be a + reliable way to tell if the error is expected or not so the error is + ignored for all channels. If the request fails on a "real" channel, + that channel just won't get the new participant's video. + + Change-Id: Ic95db4683f27d224c1869fe887795d6b9fdea4f0 + +2020-09-15 16:16 +0000 [bc038e6191] Sean Bright + + * res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined + + Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276 + +2020-08-13 03:34 +0000 [888090ab18] Torrey Searle + + * res_pjsip_diversion: implement support for History-Info + + Implemention of History-Info capable of interworking with Diversion + Header following RFC7544 + + ASTERISK-29027 #close + + Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6 + +2020-09-14 13:23 +0000 [30e08ce1bb] Sean Bright + + * format_cap: Perform codec lookups by pointer instead of name + + ASTERISK-28416 #close + + Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4 + +2020-09-11 11:09 +0000 [53910b1f25] George Joseph + + * res_pjsip_session: Fix issue with COLP and 491 + + The recent 491 changes introduced a check to determine if the active + and pending topologies were equal and to suppress the re-invite if they + were. When a re-invite is sent for a COLP-only change, the pending + topology is NULL so that check doesn't happen and the re-invite is + correctly sent. Of course, sending the re-invite sets the pending + topology. If a 491 is received, when we resend the re-invite, the + pending topology is set and since we didn't request a change to the + topology in the first place, pending and active topologies are equal so + the topologies-equal check causes the re-invite to be erroneously + suppressed. + + This change checks if the topologies are equal before we run the media + state resolver (which recreates the pending topology) so that when we + do the final topologies-equal check we know if this was a topology + change request. If it wasn't a change request, we don't suppress + the re-invite even though the topologies are equal. + + ASTERISK-29014 + + Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314 + +2020-08-20 15:09 +0000 [44bb0858cb] George Joseph + + * debugging: Add enough to choke a mule + + Added to: + * bridges/bridge_softmix.c + * channels/chan_pjsip.c + * include/asterisk/res_pjsip_session.h + * main/channel.c + * res/res_pjsip_session.c + + There NO functional changes in this commit. + + Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3 + +2020-08-20 11:21 +0000 [86f1bce186] George Joseph + + * res_pjsip_session: Handle multi-stream re-invites better + + When both Asterisk and a UA send re-invites at the same time, both + send 491 "Transaction in progress" responses to each other and back + off a specified amount of time before retrying. When Asterisk + prepares to send its re-invite, it sets up the session's pending + media state with the new topology it wants, then sends the + re-invite. Unfortunately, when it received the re-invite from the + UA, it partially processed the media in the re-invite and reset + the pending media state before sending the 491 losing the state it + set in its own re-invite. + + Asterisk also was not tracking re-invites received while an existing + re-invite was queued resulting in sending stale SDP with missing + or duplicated streams, or no re-invite at all because we erroneously + determined that a re-invite wasn't needed. + + There was also an issue in bridge_softmix where we were using a stream + from the wrong topology to determine if a stream was added. This also + caused us to erroneously determine that a re-invite wasn't needed. + + Regardless of how the delayed re-invite was triggered, we need to + reconcile the topology that was active at the time the delayed + request was queued, the pending topology of the queued request, + and the topology currently active on the session. To do this we + need a topology resolver AND we need to make stream named unique + so we can accurately tell what a stream has been added or removed + and if we can re-use a slot in the topology. + + Summary of changes: + + * bridge_softmix: + * We no longer reset the stream name to "removed" in + remove_all_original_streams(). That was causing multiple streams + to have the same name and wrecked the checks for duplicate streams. + + * softmix_bridge_stream_sources_update() was checking the old_stream + to see if it had the softmix prefix and not considering the stream + as "new" if it did. If the stream in that slot has something in it + because another re-invite happened, then that slot in old might + have a softmix stream but the same stream in new might actually + be a new one. Now we check the new_stream's name instead of + the old_stream's. + + * stream: + * Instead of using plain media type name ("audio", "video", etc) as + the default stream name, we now append the stream position to it + to make it unique. We need to do this so we can distinguish multiple + streams of the same type from each other. + + * When we set a stream's state to REMOVED, we no longer reset its + name to "removed" or destroy its metadata. Again, we need to + do this so we can distinguish multiple streams of the same + type from each other. + + * res_pjsip_session: + * Added resolve_refresh_media_states() that takes in 3 media states + and creates an up-to-date pending media state that includes the changes + that might have happened while a delayed session refresh was in the + delayed queue. + + * Added is_media_state_valid() that checks the consistency of + a media state and returns a true/false value. A valid state has: + * The same number of stream entries as media session entries. + Some media session entries can be NULL however. + * No duplicate streams. + * A valid stream for each non-NULL media session. + * A stream that matches each media session's stream_num + and media type. + + * Updated handle_incoming_sdp() to set the stream name to include the + stream position number in the name to make it unique. + + * Updated the ast_sip_session_delayed_request structure to include both + the pending and active media states and updated the associated delay + functions to process them. + + * Updated sip_session_refresh() to accept both the pending and active + media states that were in effect when the request was originally queued + and to pass them on should the request need to be delayed again. + + * Updated sip_session_refresh() to call resolve_refresh_media_states() + and substitute its results for the pending state passed in. + + * Updated sip_session_refresh() with additional debugging. + + * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE + to pjproject if a transaction is in progress. This stops us from + creating a partial pending media state that would be invalid later on. + + * Updated reschedule_reinvite() to clone both the current pending and + active media states and pass them to delay_request() so the resolver + can tell what the original intention of the re-invite was. + + * Added a large unit test for the resolver. + + ASTERISK-29014 + + Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb + +2020-08-31 07:21 +0000 [9052e448ec] Sungtae Kim + + * realtime: Increased reg_server character size + + Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20). + This is fine for normal cases, but if the hostname is longer than 20, it returns error and then + failed to register the contact address of the peer. + + Normally, 20 characters limitation for the hostname is fine, but with the cloud env. + So, increased the size to 255. + + ASTERISK-29056 + + Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d + +2020-08-30 15:42 +0000 [aae0904c7d] Sungtae Kim + + * res_stasis.c: Added video_single option for bridge creation + + Currently, it was not possible to create bridge with video_mode single. + This made hard to put the bridge in a vidoe_single mode. + So, added video_single option for Bridge creation using the ARI. + This allows create a bridge with video_mode single. + + ASTERISK-29055 + + Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae + +2020-08-31 11:14 +0000 [80a609fcce] Ben Ford + + * Bridging: Use a ref to bridge_channel's channel to prevent crash. + + There's a race condition with bridging where a bridge can be torn down + causing the bridge_channel's ast_channel to become NULL when it's still + needed. This particular case happened with attended transfers, but the + crash occurred when trying to publish a stasis message. Now, the + bridge_channel is locked, a ref to the ast_channel is obtained, and that + ref is passed down the chain. + + Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814 + +2020-09-01 08:43 +0000 [f8fe20eb9f] Patrick Verzele + + * res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly + + Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again. + + Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6 + +2020-08-28 16:31 +0000 [1a5597741f] Kevin Harwell + + * conversions: Add string to signed integer conversion functions + + Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9 + +2020-08-26 04:58 +0000 [c3a3ab8628] Kfir Itzhak + + * app_queue: Fix leave-empty not recording a call as abandoned + + This fixes a bug introduced mistakenly in ASTERISK-25665: + If leave-empty is enabled, a call may sometimes be removed from + a queue without recording it as abandoned. + This causes Asterisk to not generate an abandon event for that + call, and for the queue abandoned counter to be incorrect. + + ASTERISK-29043 #close + + Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7 + +2020-08-28 09:34 +0000 [5989e0de0f] George Joseph + + * ast_coredumper: Fix issues with naming + + If you run ast_coredumper --tarball-coredumps in the same directory + as the actual coredump, tar can fail because the link to the + actual coredump becomes recursive. The resulting tarball will + have everything _except_ the coredump (which is usually what + you need) + + There's also an issue that the directory name in the tarball + is the same as the coredump so if you extract the tarball the + directory it creates will overwrite the coredump. + + So: + + * Made the link to the coredump use the absolute path to the + file instead of a relative one. This prevents the recursive + link and allows tar to add the coredump. + + * The tarballed directory is now named .output instead + of just so if you expand the tarball it won't + overwrite the coredump. + + Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea + +2020-08-28 04:29 +0000 [c4bed96742] Joshua C. Colp + + * parking: Copy parker UUID as well. + + When fixing issues uncovered by GCC10 a copy of the parker UUID + was removed accidentally. This change restores it so that the + subscription has the data it needs. + + ASTERISK-29042 + + Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a + +2020-08-26 10:43 +0000 [f225e9bf35] Alexander Traud + + * sip_nat_settings: Update script for latest Linux. + + With the latest Linux, 'ifconfig' is not installed on default anymore. + Furthermore, the output of the current net-tools 'ifconfig' changed. + Therefore, parsing failed. This update uses 'ip addr show' instead. + Finally, the service for the external IP changed. + + Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0 + +2020-08-26 10:19 +0000 [8907a9f0b9] Alexander Traud + + * samples: Fix keep_alive_interval default in pjsip.conf. + + Since ASTERISK_27978 the default is not off but 90 seconds. That change + happened because ASTERISK_27347 disabled the keep-alives in the bundled + PJProject and Asterisk should behave the same as before. + + Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46 + +2020-08-24 16:26 +0000 [3c4a1722b6] Kevin Harwell + + * chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution + + This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function + is called on a channel prior to answering a warning is issued and the + function returns unsuccessful. + + ASTERISK-28878 #close + + Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb + +2020-08-27 05:31 +0000 [28bae5e901] Joshua C. Colp + + * pbx: Fix hints deadlock between reload and ExtensionState. + + When the ExtensionState AMI action is executed on a pattern matched + hint it can end up adding a new hint if one does not already exist. + This results in a locking order of contexts -> hints -> contexts. + + If at the same time a reload is occurring and adding its own hint + it will have a locking order of hints -> contexts. + + This results in a deadlock as one thread wants a lock on contexts + that the other has, and the other thread wants a lock on hints + that the other has. + + This change enforces a hints -> contexts locking order by explicitly + locking hints in the places where a hint is added when queried for. + This matches the order seen through normal adding of hints. + + ASTERISK-29046 + + Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504 + +2020-08-14 11:13 +0000 [54ddf19141] George Joseph + + * logger.c: Added a new log formatter called "plain" + + Added a new log formatter called "plain" that always prints + file, function and line number if available (even for verbose + messages) and never prints color control characters. It also + doesn't apply any special formatting for verbose messages. + Most suitable for file output but can be used for other channels + as well. + + You use it in logger.conf like so: + debug => [plain]debug + console => [plain]error,warning,debug,notice,pjsip_history + messages => [plain]warning,error,verbose + + Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d + +2020-08-21 16:53 +0000 [5b9ac90531] Nickolay Shmyrev + + * res_speech: Bump reference on format object + + Properly bump reference on format object to avoid memory corruption on double free + + ASTERISK-29040 #close + + Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3 + +2020-07-22 03:45 +0000 [04051b324b] Torrey Searle + + * res_pjsip_diversion: handle 181 + + Adapt the response handler so it also called when 181 is received. + In the case 181 is received, also generate the 181 response. + + ASTERISK-29001 #close + + Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df + +2020-08-21 09:17 +0000 [c925ed0eb9] Sean Bright + + * app_voicemail: Process urgent messages with mailcmd + + Rather than putting messages into INBOX and then moving them to Urgent + later, put them directly in to the Urgent folder. This prevents + mailcmd from being skipped. + + ASTERISK-27273 #close + + Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5 + +2020-08-21 00:10 +0000 [b2bd38a4f0] Evandro César Arruda + + * app_queue: Member lastpause time reseting + + This fixes the reseting members lastpause problem when realtime members is being used, + the function rt_handle_member_record was forcing the reset members lastpause because it + does not exist in realtime + + ASTERISK-29034 #close + + Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5 + +2020-08-18 04:36 +0000 [71ceefa75d] Joshua C. Colp + + * res_pjsip_session: Don't aggressively terminate on failed re-INVITE. + + Per the RFC when an outgoing re-INVITE is done we should + only terminate the dialog if a 481 or 408 is received. + + ASTERISK-29033 + + Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503 + +2020-08-19 12:29 +0000 [3553192900] Sean Bright + + * bridge_channel: Ensure text messages are zero terminated + + T.140 data in RTP is not zero terminated, so when we are queuing a text + frame on a bridge we need to ensure that we are passing a zero + terminated string. + + ASTERISK-28974 #close + + Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3 + +2020-08-07 09:31 +0000 [057fda460b] Sean Bright + + * res_musiconhold.c: Use ast_file_read_dir to scan MoH directory + + Two changes of note in this patch: + + * Use ast_file_read_dir instead of opendir/readdir/closedir + + * If the files list should be sorted, do that at the end rather than as + we go which improves performance for large lists + + Change-Id: Ic7e9c913c0f85754c99c74c9cf6dd3514b1b941f + +2020-08-19 07:37 +0000 [64ca2d48da] George Joseph + + * scope_trace: Added debug messages and added additional macros + + The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages + at the same level as the scope level. This allows the same + messages to be printed to the debug log when AST_DEVMODE + isn't enabled. + + Also added a few variants of the SCOPE_EXIT macros that will + also call ast_log instead of ast_debug to make it easier to + use scope tracing and still print error messages. + + Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21 + +2020-08-20 08:32 +0000 [118cb3f0dd] George Joseph + + * stream.c: Added 2 more debugging utils and added pos to stream string + + * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros + which are shortcuts for + ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP)) + + * Added the stream position to the string representation of the + stream. + + * Fixed some formatting in ast_stream_to_str(). + + Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b + +2020-02-18 06:30 +0000 [aab666bb9d] Dennis Buteyn + + * chan_sip: Clear ToHost property on peer when changing to dynamic host + + The ToHost parameter was not cleared when a peer's host value was + changed to dynamic. This causes invites to be sent to the original host. + + ASTERISK-29011 #close + + Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c + +2020-07-20 14:39 +0000 [647c53c41f] George Joseph + + * ACN: Changes specific to the core + + Allow passing a topology from the called channel back to the + calling channel. + + * Added a new function ast_queue_answer() that accepts a stream + topology and queues an ANSWER CONTROL frame with it as the + data. This allows the called channel to indicate its resolved + topology. + + * Added a new virtual function to the channel tech structure + answer_with_stream_topology() that allows the calling channel + to receive the called channel's topology. Added + ast_raw_answer_with_stream_topology() that invokes that virtual + function. + + * Modified app_dial.c and features.c to grab the topology from the + ANSWER frame queued by the answering channel and send it to + the calling channel with ast_raw_answer_with_stream_topology(). + + * Modified frame.c to automatically cleanup the reference + to the topology on ANSWER frames. + + Added a few debugging messages to stream.c. + + Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c + +2020-08-06 12:51 +0000 [3040edcbb1] cmaj + + * Makefile: Fix certified version numbers + + Adds sed before awk to produce reasonable ASTERISKVERSIONNUM + on certified versions of Asterisk eg. 16.8-cert3 is 160803 + instead of the previous 00800. + + ASTERISK-29021 #close + + Change-Id: Icf241df0ff6db09011b8c936a317a84b0b634e16 + +2020-08-06 11:41 +0000 [b7c2205402] Sean Bright + + * res_musiconhold.c: Prevent crash with realtime MoH + + The MoH class internal file vector is potentially being manipulated by + multiple threads at the same time without sufficient locking. Switch to + a reference counted list and operate on copies where necessary. + + ASTERISK-28927 #close + + Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217 + +2020-08-06 13:10 +0000 [447f6cc37a] Joshua C. Colp + + * res_pjsip: Fix codec preference defaults. + + When reading in a codec preference configuration option + the value would be set on the respective option before + applying any default adjustments, resulting in the + configuration not being as expected. + + This was exposed by the REST API push configuration as + it used the configuration returned by Asterisk to then do + a modification. In the case of codec preferences one of + the options had a transcode value of "unspecified" when the + defaults should have ensured it would be "allow" instead. + + This also renames the options in other places that were + missed. + + Change-Id: I4ad42e74fdf181be2e17bc75901c62591d403964 + +2020-08-04 10:51 +0000 [048b12b59d] Sean Bright + + * vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors + + The assumed behavior of realloc() - that it was effectively a free() if + its second argument was 0 - is Linux specific behavior and is not + guaranteed by either POSIX or the C specification. + + Instead, if we want to resize a vector to 0, do it explicitly. + + Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f + +2020-06-30 10:40 +0000 [e8c2ce2873] Michael Neuhauser + + * pjproject: clone sdp to protect against (nat) modifications + + PJSIP, UDP transport with external_media_address and session timers + enabled. Connected to SIP server that is not in local net. Asterisk + initiated the connection and is refreshing the session after 150s + (timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has + a malformed IP address in its SDP (garbage string). This only happens + when the SDP is modified by the nat-code to replace the local IP address + with the configured external_media_address. + Analysis: the code to modify the SDP (in + res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?) + in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses + the tdata->pool to allocate the replacement string. But the same + pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also + used for the 2nd refresh-INVITE (because it is stored in pjmedia's + pjmedia_sdp_neg structure). The problem is, that at that moment, the + tdata->pool that holds the stringified external_media_address from the + 1. refresh-INVITE has long been reused for something else. + Fix by Sauw Ming of pjproject (see + https://github.com/pjsip/pjproject/pull/2476): the local, potentially + modified pjmedia_sdp_stream is cloned in + pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the + clone is stored, thereby detaching from the tdata->pool (which is only + released *after* process_answer()) + + ASTERISK-28973 + Reported-by: Michael Neuhauser + + Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e + +2020-08-04 14:36 +0000 [9ed6387c14] Ben Ford + + * utils.c: NULL terminate ast_base64decode_string. + + With the addition of STIR/SHAKEN, the function ast_base64decode_string + was added for convenience since there is a lot of converting done during + the STIR/SHAKEN process. This function returned the decoded string for + you, but did not NULL terminate it, causing some issues (specifically + with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the + documentation has been updated to reflect this. + + Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5 + +2020-07-21 09:17 +0000 [a15e64aaf5] George Joseph + + * ACN: Configuration renaming for pjsip endpoint + + This change renames the codec preference endpoint options. + incoming_offer_codec_prefs becomes codec_prefs_incoming_offer + to keep the options together when showing an endpoint. + + Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d + +2020-07-20 13:05 +0000 [deaa3742dc] Ben Ford + + * res_stir_shaken: Fix memory allocation error in curl.c + + Fixed a memory allocation that was not passing in the correct size for + the struct in curl.c. + + Change-Id: I5fb92fbbe84b075fa6aefa2423786df80e114c3a + +2020-07-23 14:47 +0000 [1f78ee9d0f] George Joseph + + * res_pjsip_session: Ensure reused streams have correct bundle group + + When a bundled stream is removed, its bundle_group is reset to -1. + If that stream is later reused, the bundle parameters on session + media need to be reset correctly it could mistakenly be rebundled + with a stream that was removed and never reused. Since the removed + stream has no rtp instance, a crash will result. + + Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7 + +2020-07-22 04:41 +0000 [921b1a02c4] Joshua C. Colp + + * res_pjsip_registrar: Don't specify an expiration for static contacts. + + Statically configured contacts on an AOR don't have an expiration + time so when adding them to the resulting 200 OK if an endpoint + registers ensure they are marked as such. + + ASTERISK-28995 + + Change-Id: I9f0e45eb2ccdedc9a0df5358634a19ccab0ad596 + +2020-07-13 15:06 +0000 [7d96b3e437] Sean Bright + + * utf8.c: Add UTF-8 validation and utility functions + + There are various places in Asterisk - specifically in regards to + database integration - where having some kind of UTF-8 validation would + be beneficial. This patch adds: + + * Functions to validate that a given string contains only valid UTF-8 + sequences. + + * A function to copy a string (similar to ast_copy_string) stopping when + an invalid UTF-8 sequence is encountered. + + * A UTF-8 validator that allows for progressive validation. + + All of this is based on the excellent UTF-8 decoder by Björn Höhrmann. + More information is available here: + + https://bjoern.hoehrmann.de/utf-8/decoder/dfa/ + + The API was written in such a way that should allow us to replace the + implementation later should we determine that we need something more + comprehensive. + + Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9 + +2020-07-10 18:14 +0000 [c10ed8d4d6] sungtae kim + + * stasis_bridge.c: Fixed wrong video_mode shown + + Currently, if the bridge has created by the ARI, the video_mode + parameter was + not shown in the BridgeCreated event correctly. + + Fixed it and added video_mode shown in the 'bridge show ' + cli. + + ASTERISK-28987 + + Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295 + +2020-07-20 13:17 +0000 [b5bb4a7a0d] Sean Bright + + * vector.h: Add AST_VECTOR_SORT() + + Allows a vector to be sorted in-place, rather than only during + insertion. + + Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780 + +2020-07-16 08:41 +0000 [e1d30f3e6c] George Joseph + + * CI: Force publishAsteriskDocs to use python2 + + Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23 + +2020-07-22 12:57 +0000 [9f641483e6] Joshua C. Colp + + * websocket / pjsip: Increase maximum packet size. + + When dealing with a lot of video streams on WebRTC + the resulting SDPs can grow to be quite large. This + effectively doubles the maximum size to allow more + streams to exist. + + The res_http_websocket module has also been changed + to use a buffer on the session for reading in packets + to ensure that the stack space usage is not excessive. + + Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01 + +2020-07-15 09:05 +0000 [9c3b57822a] George Joseph + + * Prepare master for the next Asterisk version + + * Updated AMI version to 8.0.0 + * Updated ARI version to 7.0.0 + * Update make_ari_stubs.py to "Asterisk 19" + + Change-Id: I51fb38c2e29f2db785f64a8bbd5565d56bea5af5 + +2020-07-13 15:42 +0000 [c3588d9c0b] Sean Bright + + * acl.c: Coerce a NULL pointer into the empty string + + If an ACL is misconfigured in the realtime database (for instance, the + "rule" is blank) and Asterisk attempts to read the ACL, Asterisk will + crash. + + ASTERISK-28978 #close + + Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610 + +2020-07-13 04:41 +0000 [f1d7de121f] Joshua C. Colp + + * pjsip: Include timer patch to prevent cancelling timer 0. + + I noticed this while looking at another issue and brought + it up with Teluu. It was possible for an uninitialized timer + to be cancelled, resulting in the invalid timer id of 0 + being placed into the timer heap causing issues. + + This change is a backport from the pjproject repository + preventing this from happening. + + Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7 + +2020-09-09 15:43 +0000 Asterisk Development Team + + * asterisk 18.0.0-rc1 Released. + +2020-09-09 09:08 +0000 [f589985840] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 18.0.0 +2020-09-01 08:43 +0000 [5a49757e40] Patrick Verzele + + * res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly + + Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again. + + Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6 + +2020-08-28 16:31 +0000 [ec03909831] Kevin Harwell + + * conversions: Add string to signed integer conversion functions + + Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9 + +2020-08-26 04:58 +0000 [c83e4821e5] Kfir Itzhak + + * app_queue: Fix leave-empty not recording a call as abandoned + + This fixes a bug introduced mistakenly in ASTERISK-25665: + If leave-empty is enabled, a call may sometimes be removed from + a queue without recording it as abandoned. + This causes Asterisk to not generate an abandon event for that + call, and for the queue abandoned counter to be incorrect. + + ASTERISK-29043 #close + + Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7 + +2020-08-28 09:34 +0000 [e32815dddb] George Joseph + + * ast_coredumper: Fix issues with naming + + If you run ast_coredumper --tarball-coredumps in the same directory + as the actual coredump, tar can fail because the link to the + actual coredump becomes recursive. The resulting tarball will + have everything _except_ the coredump (which is usually what + you need) + + There's also an issue that the directory name in the tarball + is the same as the coredump so if you extract the tarball the + directory it creates will overwrite the coredump. + + So: + + * Made the link to the coredump use the absolute path to the + file instead of a relative one. This prevents the recursive + link and allows tar to add the coredump. + + * The tarballed directory is now named .output instead + of just so if you expand the tarball it won't + overwrite the coredump. + + Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea + +2020-08-28 04:29 +0000 [4f0766dcda] Joshua C. Colp + + * parking: Copy parker UUID as well. + + When fixing issues uncovered by GCC10 a copy of the parker UUID + was removed accidentally. This change restores it so that the + subscription has the data it needs. + + ASTERISK-29042 + + Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a + +2020-08-26 10:43 +0000 [9ed1b1452d] Alexander Traud + + * sip_nat_settings: Update script for latest Linux. + + With the latest Linux, 'ifconfig' is not installed on default anymore. + Furthermore, the output of the current net-tools 'ifconfig' changed. + Therefore, parsing failed. This update uses 'ip addr show' instead. + Finally, the service for the external IP changed. + + Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0 + +2020-08-26 10:19 +0000 [217449a1e5] Alexander Traud + + * samples: Fix keep_alive_interval default in pjsip.conf. + + Since ASTERISK_27978 the default is not off but 90 seconds. That change + happened because ASTERISK_27347 disabled the keep-alives in the bundled + PJProject and Asterisk should behave the same as before. + + Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46 + +2020-08-24 16:26 +0000 [31fbfc5e95] Kevin Harwell + + * chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution + + This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function + is called on a channel prior to answering a warning is issued and the + function returns unsuccessful. + + ASTERISK-28878 #close + + Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb + +2020-08-27 05:31 +0000 [6d50d152d8] Joshua C. Colp + + * pbx: Fix hints deadlock between reload and ExtensionState. + + When the ExtensionState AMI action is executed on a pattern matched + hint it can end up adding a new hint if one does not already exist. + This results in a locking order of contexts -> hints -> contexts. + + If at the same time a reload is occurring and adding its own hint + it will have a locking order of hints -> contexts. + + This results in a deadlock as one thread wants a lock on contexts + that the other has, and the other thread wants a lock on hints + that the other has. + + This change enforces a hints -> contexts locking order by explicitly + locking hints in the places where a hint is added when queried for. + This matches the order seen through normal adding of hints. + + ASTERISK-29046 + + Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504 + +2020-08-14 11:13 +0000 [5a8cacb93d] George Joseph + + * logger.c: Added a new log formatter called "plain" + + Added a new log formatter called "plain" that always prints + file, function and line number if available (even for verbose + messages) and never prints color control characters. It also + doesn't apply any special formatting for verbose messages. + Most suitable for file output but can be used for other channels + as well. + + You use it in logger.conf like so: + debug => [plain]debug + console => [plain]error,warning,debug,notice,pjsip_history + messages => [plain]warning,error,verbose + + Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d + +2020-08-21 16:53 +0000 [0319e0b07f] Nickolay Shmyrev + + * res_speech: Bump reference on format object + + Properly bump reference on format object to avoid memory corruption on double free + + ASTERISK-29040 #close + + Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3 + +2020-07-22 03:45 +0000 [addd295cda] Torrey Searle + + * res_pjsip_diversion: handle 181 + + Adapt the response handler so it also called when 181 is received. + In the case 181 is received, also generate the 181 response. + + ASTERISK-29001 #close + + Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df + +2020-08-21 00:09 +0000 [36dd15c659] Evandro César Arruda + + * app_queue: Member lastpause time reseting + + This fixes the reseting members lastpause problem when realtime members is being used, + the function rt_handle_member_record was forcing the reset members lastpause because it + does not exist in realtime + + ASTERISK-29034 #close + + Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5 + +2020-08-21 09:17 +0000 [b575868000] Sean Bright + + * app_voicemail: Process urgent messages with mailcmd + + Rather than putting messages into INBOX and then moving them to Urgent + later, put them directly in to the Urgent folder. This prevents + mailcmd from being skipped. + + ASTERISK-27273 #close + + Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5 + +2020-08-18 04:36 +0000 [3c074038fe] Joshua C. Colp + + * res_pjsip_session: Don't aggressively terminate on failed re-INVITE. + + Per the RFC when an outgoing re-INVITE is done we should + only terminate the dialog if a 481 or 408 is received. + + ASTERISK-29033 + + Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503 + +2020-08-19 12:29 +0000 [5ec7099312] Sean Bright + + * bridge_channel: Ensure text messages are zero terminated + + T.140 data in RTP is not zero terminated, so when we are queuing a text + frame on a bridge we need to ensure that we are passing a zero + terminated string. + + ASTERISK-28974 #close + + Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3 + +2020-08-07 09:31 +0000 [5dfeeba623] Sean Bright + + * res_musiconhold.c: Use ast_file_read_dir to scan MoH directory + + Two changes of note in this patch: + + * Use ast_file_read_dir instead of opendir/readdir/closedir + + * If the files list should be sorted, do that at the end rather than as + we go which improves performance for large lists + + Change-Id: Ic7e9c913c0f85754c99c74c9cf6dd3514b1b941f + +2020-08-19 07:37 +0000 [c4c72d55a2] George Joseph + + * scope_trace: Added debug messages and added additional macros + + The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages + at the same level as the scope level. This allows the same + messages to be printed to the debug log when AST_DEVMODE + isn't enabled. + + Also added a few variants of the SCOPE_EXIT macros that will + also call ast_log instead of ast_debug to make it easier to + use scope tracing and still print error messages. + + Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21 + +2020-08-20 08:32 +0000 [d26ab7f8f9] George Joseph + + * stream.c: Added 2 more debugging utils and added pos to stream string + + * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros + which are shortcuts for + ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP)) + + * Added the stream position to the string representation of the + stream. + + * Fixed some formatting in ast_stream_to_str(). + + Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b + +2020-02-18 06:30 +0000 [9058d9e591] Dennis Buteyn + + * chan_sip: Clear ToHost property on peer when changing to dynamic host + + The ToHost parameter was not cleared when a peer's host value was + changed to dynamic. This causes invites to be sent to the original host. + + ASTERISK-29011 #close + + Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c + +2020-07-20 14:39 +0000 [6faf76308d] George Joseph + + * ACN: Changes specific to the core + + Allow passing a topology from the called channel back to the + calling channel. + + * Added a new function ast_queue_answer() that accepts a stream + topology and queues an ANSWER CONTROL frame with it as the + data. This allows the called channel to indicate its resolved + topology. + + * Added a new virtual function to the channel tech structure + answer_with_stream_topology() that allows the calling channel + to receive the called channel's topology. Added + ast_raw_answer_with_stream_topology() that invokes that virtual + function. + + * Modified app_dial.c and features.c to grab the topology from the + ANSWER frame queued by the answering channel and send it to + the calling channel with ast_raw_answer_with_stream_topology(). + + * Modified frame.c to automatically cleanup the reference + to the topology on ANSWER frames. + + Added a few debugging messages to stream.c. + + Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c + +2020-08-06 12:51 +0000 [543f936147] cmaj + + * Makefile: Fix certified version numbers + + Adds sed before awk to produce reasonable ASTERISKVERSIONNUM + on certified versions of Asterisk eg. 16.8-cert3 is 160803 + instead of the previous 00800. + + ASTERISK-29021 #close + + Change-Id: Icf241df0ff6db09011b8c936a317a84b0b634e16 + +2020-08-06 11:41 +0000 [57554c2834] Sean Bright + + * res_musiconhold.c: Prevent crash with realtime MoH + + The MoH class internal file vector is potentially being manipulated by + multiple threads at the same time without sufficient locking. Switch to + a reference counted list and operate on copies where necessary. + + ASTERISK-28927 #close + + Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217 + +2020-08-06 13:10 +0000 [a3d87f78ed] Joshua C. Colp + + * res_pjsip: Fix codec preference defaults. + + When reading in a codec preference configuration option + the value would be set on the respective option before + applying any default adjustments, resulting in the + configuration not being as expected. + + This was exposed by the REST API push configuration as + it used the configuration returned by Asterisk to then do + a modification. In the case of codec preferences one of + the options had a transcode value of "unspecified" when the + defaults should have ensured it would be "allow" instead. + + This also renames the options in other places that were + missed. + + Change-Id: I4ad42e74fdf181be2e17bc75901c62591d403964 + +2020-08-04 10:51 +0000 [da8a617dc9] Sean Bright + + * vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors + + The assumed behavior of realloc() - that it was effectively a free() if + its second argument was 0 - is Linux specific behavior and is not + guaranteed by either POSIX or the C specification. + + Instead, if we want to resize a vector to 0, do it explicitly. + + Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f + +2020-06-30 10:40 +0000 [6482ab5bea] Michael Neuhauser + + * pjproject: clone sdp to protect against (nat) modifications + + PJSIP, UDP transport with external_media_address and session timers + enabled. Connected to SIP server that is not in local net. Asterisk + initiated the connection and is refreshing the session after 150s + (timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has + a malformed IP address in its SDP (garbage string). This only happens + when the SDP is modified by the nat-code to replace the local IP address + with the configured external_media_address. + Analysis: the code to modify the SDP (in + res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?) + in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses + the tdata->pool to allocate the replacement string. But the same + pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also + used for the 2nd refresh-INVITE (because it is stored in pjmedia's + pjmedia_sdp_neg structure). The problem is, that at that moment, the + tdata->pool that holds the stringified external_media_address from the + 1. refresh-INVITE has long been reused for something else. + Fix by Sauw Ming of pjproject (see + https://github.com/pjsip/pjproject/pull/2476): the local, potentially + modified pjmedia_sdp_stream is cloned in + pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the + clone is stored, thereby detaching from the tdata->pool (which is only + released *after* process_answer()) + + ASTERISK-28973 + Reported-by: Michael Neuhauser + + Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e + +2020-08-04 14:36 +0000 [769a9611e7] Ben Ford + + * utils.c: NULL terminate ast_base64decode_string. + + With the addition of STIR/SHAKEN, the function ast_base64decode_string + was added for convenience since there is a lot of converting done during + the STIR/SHAKEN process. This function returned the decoded string for + you, but did not NULL terminate it, causing some issues (specifically + with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the + documentation has been updated to reflect this. + + Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5 + +2020-07-21 09:17 +0000 [802aa97fa0] George Joseph + + * ACN: Configuration renaming for pjsip endpoint + + This change renames the codec preference endpoint options. + incoming_offer_codec_prefs becomes codec_prefs_incoming_offer + to keep the options together when showing an endpoint. + + Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d + +2020-07-20 13:05 +0000 [de23cb4002] Ben Ford + + * res_stir_shaken: Fix memory allocation error in curl.c + + Fixed a memory allocation that was not passing in the correct size for + the struct in curl.c. + + Change-Id: I5fb92fbbe84b075fa6aefa2423786df80e114c3a + (cherry picked from commit deaa3742dc998e38369d34bfc308d84e9036dcba) + +2020-07-23 14:47 +0000 [71446b68fc] George Joseph + + * res_pjsip_session: Ensure reused streams have correct bundle group + + When a bundled stream is removed, its bundle_group is reset to -1. + If that stream is later reused, the bundle parameters on session + media need to be reset correctly it could mistakenly be rebundled + with a stream that was removed and never reused. Since the removed + stream has no rtp instance, a crash will result. + + Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7 + +2020-07-22 04:41 +0000 [99eafe5771] Joshua C. Colp + + * res_pjsip_registrar: Don't specify an expiration for static contacts. + + Statically configured contacts on an AOR don't have an expiration + time so when adding them to the resulting 200 OK if an endpoint + registers ensure they are marked as such. + + ASTERISK-28995 + + Change-Id: I9f0e45eb2ccdedc9a0df5358634a19ccab0ad596 + +2020-07-13 15:06 +0000 [d9ae902f52] Sean Bright + + * utf8.c: Add UTF-8 validation and utility functions + + There are various places in Asterisk - specifically in regards to + database integration - where having some kind of UTF-8 validation would + be beneficial. This patch adds: + + * Functions to validate that a given string contains only valid UTF-8 + sequences. + + * A function to copy a string (similar to ast_copy_string) stopping when + an invalid UTF-8 sequence is encountered. + + * A UTF-8 validator that allows for progressive validation. + + All of this is based on the excellent UTF-8 decoder by Björn Höhrmann. + More information is available here: + + https://bjoern.hoehrmann.de/utf-8/decoder/dfa/ + + The API was written in such a way that should allow us to replace the + implementation later should we determine that we need something more + comprehensive. + + Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9 + +2020-07-10 18:14 +0000 [2e32b56bdb] sungtae kim + + * stasis_bridge.c: Fixed wrong video_mode shown + + Currently, if the bridge has created by the ARI, the video_mode + parameter was + not shown in the BridgeCreated event correctly. + + Fixed it and added video_mode shown in the 'bridge show ' + cli. + + ASTERISK-28987 + + Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295 + +2020-07-20 13:17 +0000 [9022f35f09] Sean Bright + + * vector.h: Add AST_VECTOR_SORT() + + Allows a vector to be sorted in-place, rather than only during + insertion. + + Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780 + +2020-07-16 08:41 +0000 [a678dafac8] George Joseph + + * CI: Force publishAsteriskDocs to use python2 + + Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23 + +2020-07-22 12:57 +0000 [af70bbb13a] Joshua C. Colp + + * websocket / pjsip: Increase maximum packet size. + + When dealing with a lot of video streams on WebRTC + the resulting SDPs can grow to be quite large. This + effectively doubles the maximum size to allow more + streams to exist. + + The res_http_websocket module has also been changed + to use a buffer on the session for reading in packets + to ensure that the stack space usage is not excessive. + + Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01 + +2020-07-13 15:42 +0000 [7a43bedd72] Sean Bright + + * acl.c: Coerce a NULL pointer into the empty string + + If an ACL is misconfigured in the realtime database (for instance, the + "rule" is blank) and Asterisk attempts to read the ACL, Asterisk will + crash. + + ASTERISK-28978 #close + + Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610 + +2020-07-13 04:41 +0000 [8d15f72721] Joshua C. Colp + + * pjsip: Include timer patch to prevent cancelling timer 0. + + I noticed this while looking at another issue and brought + it up with Teluu. It was possible for an uninitialized timer + to be cancelled, resulting in the invalid timer id of 0 + being placed into the timer heap causing issues. + + This change is a backport from the pjproject repository + preventing this from happening. + + Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7 + +2020-07-15 09:14 +0000 [3330764213] George Joseph + + * Update .gitreview defaultbranch to 18 + + Change-Id: Ib2c42fc2d46563e2fbadbd5513cb029b4042791e + +2020-07-15 08:59 +0000 [1f5e6805bf] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 18.0.0 +2020-07-02 17:19 +0000 [e4d24f5137] Nickolay Shmyrev + + * res_http_websocket: Avoid reading past end of string + + We read beyond the end of the buffer when copying the string out of the + buffer when we used ast_copy_string() because the original string was + not null terminated. Instead switch to ast_strndup() which does not + exhibit the same behavior. + + ASTERISK-28975 #close + + Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a + +2020-06-24 11:49 +0000 [5fbed5af24] Ben Ford + + * res_stir_shaken: Add stir_shaken option and general improvements. + + Added a new configuration option for PJSIP endpoints - stir_shaken. If + set to yes, then STIR/SHAKEN support will be added to inbound and + outbound INVITEs. The default is no. Alembic has been updated to include + this option. + + Previously the dialplan function was not trimming the whitespace from + the parameters it recieved. Now it does. + + Also added a conditional that, when TEST_FRAMEWORK is enabled, the + timestamp in the identity header will be overlooked. This is just for + testing, since the testsuite will rely on a SIPp scenario with a preset + identity header to trigger the MISMATCH result. + + Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1 + +2020-07-09 09:56 +0000 [e88beedd08] George Joseph + + * res_pjsip_session: Fix segv in session_on_rx_response + + session_on_rx_response wasn't checking for a NULL dialog before + attempting to get the invite session from it. + + Change-Id: Id13534375966cc2eb7f2b55717c9813c63c10065 + +2020-06-23 02:34 +0000 [312c23b0e1] Walter Doekes + + * app_queue: (Breaking change) shared_lastcall and autofill default to no + + If your queues.conf had _no_ [general] section, they would default to + 'yes'. Now, they always default to 'no'. + + (Actually, commit ed615afb7e0d630a58feba569c657eadc6ddc0a9 already + partially fixed it for shared_lastcall.) + + ASTERISK-28951 + + Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6 + +2020-07-06 14:23 +0000 [9bd1d686a1] George Joseph + + * ACN: Add tracing to existing code + + Prior to making any modifications to the pjsip infrastructure + for ACN, I've added the tracing functions to the existing code. + This should make the final commit easier to review, but we can also + now run a "before and after" trace. + + No functional changes were made with this commit. + + Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c + +2020-07-06 09:56 +0000 [2d22e34206] George Joseph + + * ACN: res_pjsip endpoint options + + This commit adds the endpoint options required to control + Advanced Codec Negotiation. + + incoming_offer_codec_prefs + outgoing_offer_codec_prefs + incoming_answer_codec_prefs + outgoing_answer_codec_prefs + + The documentation may need tweaking and some additional edits + added, especially for the "answer" prefs. That'll be handled + when things finalize. + + This commit is safe to merge as it doens't alter any existing + functionality nor does it alter the previous codec negotiation + work which may now be obsolete. + + Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11 + +2020-06-23 18:27 +0000 [81b5e4a73f] sungtae kim + + * res_pjsip.c: Added disable_rport option for pjsip.conf + + Currently when the pjsip making an outgoing request, it keep adding the + rport parameter in a request message as a default. + + This causes unexpected rport handle at the other end. + + Added option for disable this behaviour in the pjsip.conf. + + This is a system option, but working as a gloabl option. + + ASTERISK-28959 + + Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc + +2020-07-06 10:57 +0000 [d093e44b1e] George Joseph + + * frame.c: Make debugging easier + + * ast_frame_subclass2str() and ast_frame_type2str() now return + a pointer to the buffer that was passed in instead of void. + This makes it easier to use these functions inline in + printf-style debugging statements. + + * Added many missing control frame entries in + ast_frame_subclass2str. + + Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc + +2020-07-05 18:51 +0000 [955b7b4fdb] George Joseph + + * Scope Trace: Make it easier to trace through synchronous tasks + + Tracing through synchronous tasks was a little troublesome because + the new thread's stack counter reset to 0. This change allows + a synchronous task to set its trace level to be the same as the + thread that pushed the task. For now, the task's level has to be + passed in the task's data structure but a future enhancement to the + taskprocessor subsystem could automatically set the trace level + of the servant to be that of the caller. + + This doesn't really make sense for async tasks because you never + know when they're going to run anyway. + + Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825 + +2020-06-22 12:16 +0000 [7163efd934] Nickolay Shmyrev + + * res_http_websocket.c: Continue reading after ping/pong + + Do not return error if the client received ping frame + while looking for a string and just wait for another frame. + + ASTERISK-28958 #close + + Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922 + +2020-06-30 11:08 +0000 [4eba6b9eb2] Kevin Harwell + + * PJSIP_MEDIA_OFFER: override configuration on refresh + + When using the PSJIP_MEDIA_OFFER dialplan function it was not + overriding an endpoint's configured codecs on refresh unless + they had a shared codec between the two. + + This patch makes it so whatever is set using PJSIP_MEDIA_OFFER + is used when creating the SDP for a refresh no matter what. + + ASTERISK-28878 #close + + Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6 + +2020-06-10 17:02 +0000 [cfed0ea033] Kevin Harwell + + * manager - Add Content-Type parameter to the SendText action + + This patch allows a user of AMI to now specify the type of message + content contained within by setting the 'Content-Type' parameter. + + Note, the AMI version has been bumped for this change. + + ASTERISK-28945 #close + + Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb + +2020-06-26 11:14 +0000 [8d1064eaaf] George Joseph + + * Streams: Add features for Advanced Codec Negotiation + + The Streams API becomes the home for the core ACN capabilities. + These include... + + * Parsing and formatting of codec negotation preferences. + * Resolving pending streams and topologies with those configured + using configured preferences. + * Utility functions for creating string representations of + streams, topologies, and negotiation preferences. + + For codec negotiation preferences: + * Added ast_stream_codec_prefs_parse() which takes a string + representation of codec negotiation preferences, which + may come from a pjsip endpoint for example, and populates + a ast_stream_codec_negotiation_prefs structure. + * Added ast_stream_codec_prefs_to_str() which does the reverse. + * Added many functions to parse individual parameter name + and value strings to their respectrive enum values, and the + reverse. + + For streams: + * Added ast_stream_create_resolved() which takes a "live" stream + and resolves it with a configured stream and the negotiation + preferences to create a new stream. + * Added ast_stream_to_str() which create a string representation + of a stream suitable for debug or display purposes. + + For topology: + * Added ast_stream_topology_create_resolved() which takes a "live" + topology and resolves it, stream by stream, with a configured + topology stream and the negotiation preferences to create a new + topology. + * Added ast_stream_topology_to_str() which create a string + representation of a topology suitable for debug or display + purposes. + * Renamed ast_format_caps_from_topology() to + ast_stream_topology_get_formats() to be more consistent with + the existing ast_stream_get_formats(). + + Additional changes: + * A new function ast_format_cap_append_names() appends the results + to the ast_str buffer instead of replacing buffer contents. + + Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56 + +2020-06-30 08:56 +0000 [7440fd0397] George Joseph + + * Scope Trace: Add some new tracing macros and an ast_str helper + + Created new SCOPE_ functions that don't depend on RAII_VAR. Besides + generating less code, the use of the explicit SCOPE_EXIT macros + capture the line number where the scope exited. The RAII_VAR + versions can't do that. + + * SCOPE_ENTER(level, ...): Like SCOPE_TRACE but doesn't use + RAII_VAR and therefore needs needs one of... + + * SCOPE_EXIT(...): Decrements the trace stack counter and optionally + prints a message. + + * SCOPE_EXIT_EXPR(__expr, ...): Decrements the trace stack counter, + optionally prints a message, then executes the expression. + SCOPE_EXIT_EXPR(break, "My while got broken\n"); + + * SCOPE_EXIT_RTN(, ...): Decrements the trace stack counter, + optionally prints a message, then returns without a value. + SCOPE_EXIT_RTN("Bye\n"); + + * SCOPE_EXIT_RTN_VALUE(__return_value, ...): Decrements the trace + stack counter, optionally prints a message, then returns the value + specified. + SCOPE_EXIT_RTN_VALUE(rc, "Returning with RC: %d\n", rc); + + Create an ast_str helper ast_str_tmp() that allocates a temporary + ast_str that can be passed to a function that needs it, then frees + it. This makes using the above macros easier. Example: + + SCOPE_ENTER(1, Format Caps 1: %s Format Caps 2: %s\n", + ast_str_tmp(32, ast_format_cap_get_names(cap1, &STR_TMP), + ast_str_tmp(32, ast_format_cap_get_names(cap2, &STR_TMP)); + + The calls to ast_str_tmp create an ast_str of the specified initial + length which can be referenced as STR_TMP. It then calls the + expression, which must return a char *, ast_strdupa's it, frees + STR_TMP, then returns the ast_strdupa'd string. That string is + freed when the function returns. + + Change-Id: I44059b20d55a889aa91440d2f8a590865998be51 + +2020-06-26 05:18 +0000 [4f86118bd8] Joshua C. Colp + + * res_pjsip: Apply AOR outbound proxy to static contacts. + + The outbound proxy for an AOR was not being applied to + any statically configured Contacts. This resulted in the + OPTIONS requests being sent to the wrong target. + + This change sets the outbound proxy on statically configured + contacts once the AOR configuration is done being + applied. + + ASTERISK-28965 + + Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0 + +2020-06-24 05:25 +0000 [9b5042433b] Joshua C. Colp + + * menuselect: Resolve infinite loop in dependency scenario. + + Given a scenario where a module has a dependency on both + an external library and a module if the external library was + available and the module was not an infinite loop would + occur. This happened due to the code changing the dependecy + status to no failure on each dependency checking loop + iteration, resulting in the code thinking that it had + gone from no failure to failure each time triggering another + dependency check. + + This change makes it so that the old dependency status is + preserved throughout the dependency checking allowing it to + determine that after the first iteration the dependency + status does not transition from no failure to failure. + + ASTERISK-28930 + + Change-Id: Iea06d45d9fd6d8bfd068882a0bb7e23a53ec3e84 + +2020-06-22 04:08 +0000 [a423f935c9] Frederic LE FOLL + + * chan_sip: chan_sip does not process 400 response to an INVITE. + + chan_sip handle_response() function, for a 400 response to an INVITE, + calls handle_response_invite() and does not generate ACK. + handle_response_invite() does not recognize 400 response and has no + default response processing for unexpected responses, thus it does not + generate ACK either. + The ACK on response repetition comes from handle_response() mechanism + "We must re-send ACKs to re-transmitted final responses". + + According to code history, 400 response specific processing was + introduced with commit + "channels/chan_sip: Add improved support for 4xx error codes" + This commit added support for : + - 400/414/493 in handle_response_subscribe() handle_response_register() + and handle_response(). + - 414/493 only in handle_response_invite(). + + This fix adds 400 response support in handle_response_invite(). + + ASTERISK-28957 + + Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad + +2020-06-22 15:27 +0000 [8b925fbda3] Kevin Harwell + + * chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+ + + A patch made a reference to the PJSIP_SC_NULL enumeration value, which + was added to pjproject 2.8 and above thus making it so Asterisk would + fail to compile with prior versions of pjproject. + + This patch removes the reference, and instead initializes the value + to '0'. + + ASTERISK-28886 #close + + Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7 + +2020-06-03 05:05 +0000 [0c1c386634] Università di Bologna - CESIA VoIP + + * res_corosync: Fix crash in huge distributed environment. + + 1) Fix memory-leaks + Added code to release ast_events extracted from corosync and stasis messages + + 2) Clean stasis cache when a member of the corosync cluster leaves the group + Added code to remove from the stasis cache of the members remained on the + group all the messages with the EID of the left member. + If the device states of the left member remain in the stasis cache of other + members, they will not be updated anymore and high priority cached values, + like BUSY, will take precedence over current device states. + + 3) Stop corosync event propagation when node is not joined to the group + Updated dispatch_thread_handler code to detect when asterisk is not joined + to the corosync group and added some condition in publish_event_to_corosync + code to send corosync messages only when joined. + When a node is not joined its corosync daemon can't send messages: + the cpg_mcast_joined function append new messages to the FIFO buffer until + it's full and then it blocks indefinitely. + In this scenario if the stasis_message_cb callback, registered by + res_corosync to handle stasis messages, try to send a corosync messages, + the thread of the stasis thread-pool will be blocked until the node join + the corosync cluster. + + ASTERISK-28888 + Reported by: Università di Bologna - CESIA VoIP + + Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2 + +2020-06-13 11:29 +0000 [9445dac43b] Moises Silva + + * res_http_websocket: Add payload masking to the websocket client + + ASTERISK-28949 + + Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca + +2020-06-18 03:49 +0000 [00a52b4752] Joshua C. Colp + + * app_stream_echo: Fix state of added streams. + + When stream support was added to Asterisk the stream state + was used inconsistently, resulting in odd behavior. This + was then standardized to be the state of a stream from the + perspective of Asterisk. + + This change updates the StreamEcho dialplan application + to use the correct state, send only, since we are only + sending to the endpoint and not expecting them to send us + multiple video streams. + + ASTERISK-28954 + + Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56 + +2020-06-18 05:14 +0000 [d88e230037] Guido Falsi + + * chan_dadhi: Fix setvar in dahdi channels + + The change to how setvar works for various channels performed in + ASTERISK~23756 missed some required change in the dahdi channel, + where the variables are actually set while reading configuration. + This change should fix the issue. + + ASTERISK-28955 + + Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274 + +2020-06-17 03:58 +0000 [ee8ea9275f] Joshua C. Colp + + * res_pjsip_session: Preserve label on incoming re-INVITE. + + When a re-INVITE is received we create a new set of + streams that are then swapped in as the active streams. + We did not preserve the SDP label from the previous + streams, resulting in the label getting lost. + + This change ensures that if an SDP label is present + on the previous stream then it is set on the new stream. + + ASTERISK-28953 + + Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445 + +2020-06-10 04:35 +0000 [a143c3a7b7] Joshua C. Colp + + * res_sorcery_memory_cache: Disallow per-object expire with full backend. + + The AMI action and CLI command did not take into account the properties + of full backend caching. This resulted in an expired object remaining + removed until a full backend update occurred, instead of having the + object updated when needed. + + This change makes it so that the AMI action and CLI command for object + expire will now fail instead of putting the cache into an undesired + state. If full backend caching is enabled then only operations + which act on the entire cache are available. + + ASTERISK-28942 + + Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4 + +2020-06-02 09:04 +0000 [1274117102] Ben Ford + + * res_stir_shaken: Add outbound INVITE support. + + Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is + sent, the caller ID will be checked to see if there is a certificate + that corresponds to it. If so, that information will be retrieved and an + Identity header will be added to the SIP message. The format is: + + header.payload.signature;info=alg=ES256;ppt=shaken + + Header, payload, and signature are all BASE64 encoded. The public key + URL is retrieved from the certificate. Currently the algorithm and ppt + are ES256 and shaken, respectively. This message is signed and can be + used for verification on the receiving end. + + Two new configuration options have been added to the certificate object: + attestation and origid. The attestation is required and must be A, B, or + C. origid is the origination identifier. + + A new utility function has been added as well that takes a string, + allocates space, BASE64 encodes it, then returns it, eliminating the + need to calculate the size yourself. + + Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4 + +2020-06-15 06:53 +0000 [db012e8cc6] Walter Doekes + + * app_queue: Remove stale code in try_calling + + Because ring_entry() is not called, outgoing->chan is not touched here + either. + + ASTERISK-28950 + ASTERISK-28644 + + Change-Id: I564613715dfaf45af868251eb75a451f512af90f + +2020-06-15 07:09 +0000 [f1cfd54976] Walter Doekes + + * res_pjsip: Include instead of internal "pjsua-lib/pjsua.h" + + Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78 + +2020-06-16 08:18 +0000 [0fb6738314] Walter Doekes + + * app_queue: Read latest wrapuptime instead of (possibly stale) copy + + Before this changeset, it was possible that a queue member (agent) was + called even though they just got out of a call, and wrapuptime seconds + hadn't passed yet. + + This could happen if a member ended a call _between_ a new call attempt + and asterisk trying that particular member for a new call. + + In that case, Asterisk would check the hangup time of the + call-before-the-last-call instead of the hangup time of the-last-call. + + ASTERISK-28952 + + Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3 + +2020-05-15 16:08 +0000 [415b55af5a] Kevin Harwell + + * pjproject: Upgrade bundled version to pjproject 2.10 + + This patch makes the usual necessary changes when upgrading to a new + version pjproject. For instance, version number bump, patches removed + from third-party, new *.md5 file added, etc.. + + This patch also includes a change to the Asterisk pjproject Makefile to + explicitly create the 'source/pjsip-apps/lib' directory. This directory + is no longer there by default so needs to be added so the Asterisk + malloc debug can be built. + + This patch also includes some minor changes to Asterisk that were a result + of the upgrade. Specifically, there was a backward incompatibility change + made in 2.10 that modified the "expires header" variable field from a + signed to an unsigned value. This potentially effects comparison. Namely, + those check for a value less than zero. This patch modified a few locations + in the Asterisk code that may have been affected. + + Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to + check a minimum version of pjproject at compile time. + + ASTERISK-28899 #close + + Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81 + +2020-06-03 11:47 +0000 [de2813cf23] Joshua C. Colp + + * core_unreal / core_local: Add multistream and re-negotiation. + + When requesting a Local channel the requested stream topology + or a converted stream topology will now be placed onto the + resulting channels. + + Frames written in on streams will now also preserve the stream + identifier as they are queued on the opposite channel. + + Finally when a stream topology change is requested it is + immediately accepted and reflected on both channels. Each + channel also receives a queued frame to indicate that the + topology has changed. + + ASTERISK-28938 + + Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186 + +2020-06-12 05:16 +0000 [bbe0f2230d] sungtae kim + + * res_ari: Fix create channel request channelId parameter parsing + + If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. + + Fixed it to parse the channelId, other_channel_id parameter correclty. + + ASTERISK-28948 + + Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157 + +2020-06-08 06:27 +0000 [c84d962eae] Joshua C. Colp + + * res_rtp_asterisk: Don't assume setting retrans props means to enable. + + The "value" passed in when setting an RTP property determines + whether it should be enabled or disabled. The RTP send and + receive retrans props did not examine this to know if the + buffers should be enabled. They assumed they always should be. + + This change makes it so that the "value" passed in is + respected. + + ASTERISK-28939 + + Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc + +2020-06-10 12:11 +0000 [8ad06394c4] Joshua C. Colp + + * bridge_softmix: Add additional old states for adding new source. + + There are three states that an old stream can be in to allow + becoming a source stream in a new stream: + + 1. Removed + 2. Inactive + 3. Sendonly + + This change adds the two missing ones, inactive and sendonly, + so if a stream transitions from those to a state where they are + providing video to Asterisk we properly re-negotiate the other + participants. + + ASTERISK-28944 + + Change-Id: Id8256b9b254b403411586284bbaedbf50452de01 + +2020-06-03 11:23 +0000 [41f3a7da4d] George Joseph + + * res_fax: Don't start a gateway if either channel is hung up + + When fax_gateway_framehook is called and a gateway hasn't already + been started, the framehook gets the t38 state for both the current + channel and the peer. That call trickles down to the channel + driver which determines the state. If either channel is hung up + (or in the process of being hung up), the channel driver's tech_pvt + is going to be NULL which, in the case of chan_pjsip, will cause a + segfault. + + * Added a hangup check for both the channel and peer channel + before starting a fax gateway. + + * Added a check for NULL tech_pvt to chan_pjsip_queryoption + so we don't attempt to reference a tech_pvt that's already + gone. + + ASTERISK-28923 + Reported by: Yury Kirsanov + + Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c + +2020-06-07 19:02 +0000 [b9f42a717e] George Joseph + + * app_confbridge: Plug ref leak of bridge channel with send_events + + When send_events is enabled for a user, we were leaking a reference + to the bridge channel in confbridge_manager.c:send_message(). This + also caused the bridge snapshot to not be destroyed. + + Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97 + +2020-06-01 18:25 +0000 [3d1bf3c537] Kevin Harwell + + * Compiler fixes for gcc 10 + + This patch fixes a few compile warnings/errors that now occur when using gcc + 10+. + + Also, the Makefile.rules check to turn off partial inlining in gcc versions + greater or equal to 8.2.1 had a bug where it only it only checked against + versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures + any version above the specified version is correctly compared. + + Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9 + +2020-06-08 14:34 +0000 [559fa0e89c] Ben Ford + + * cli.c: Fix compiler error. + + Added default variable value to fix a compiler error. + + Change-Id: I7b592adbb1274dc5464dea1c5e5de0685c928553 + +2020-06-09 06:57 +0000 [fa7c69f40f] sungtae kim + + * res_ari: Fix create request body parameter parsing. + + If parameters were passed in the body as JSON to the + create route they were not being parsed before checking + to ensure that required fields were set. + + This change moves the parsing so it occurs before + checking. + + ASTERISK-28940 + + Change-Id: I898b4c3c7ae1cde19a6840e59f498822701cf5cf + +2020-06-05 04:30 +0000 [e74dde5100] Walter Doekes + + * pjsip: Prevent invalid memory access when attempting to contact a non-sip URI + + You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri, + without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S). + + ASTERISK-28936 + + Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a + +2020-05-19 14:46 +0000 [3927f79cb5] Ben Ford + + * res_stir_shaken: Add inbound INVITE support. + + Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an + INVITE, the Identity header is retrieved, parsing the message to verify + the signature. If any of the parsing fails, + AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this + caller ID. If verification itself fails, + AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in + the payload does not line up with the SIP signaling, + AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps + pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the + verification process. + + A new config option has been added to the general section for + stir_shaken.conf. "signature_timeout" is the amount of time a signature + will be considered valid. If an INVITE is received and the amount of + time between when it was received and when it was signed is greater than + signature_timeout, verification will fail. + + Some changes were also made to signing and verification. There was an + error where the whole JSON string was being signed rather than the + header combined with the payload. This has been changed to sign the + correct thing. Verification has been changed to do this as well, and the + unit tests have been updated to reflect these changes. + + A couple of utility functions have also been added. One decodes a BASE64 + string and returns the decoded string, doing all the length calculations + for you. The other retrieves a string value from a header in a rdata + object. + + Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913 + +2020-06-05 04:45 +0000 [1fcb6b1b21] Joshua C. Colp + + * bridge_channel: Don't queue unmapped frames. + + If a frame is written to a channel in a bridge we + would normally queue this frame up and the channel + thread would then act upon it. If this frame had no + stream mapping on the channel it would then be + discarded. + + This change adds a check before the queueing occurs + to determine if a mapping exists. If it does not + exist then the frame is not even queued at all. This + stops a frame duplication from happening and from + the channel thread having to wake up and deal with + it. + + Change-Id: I17189b9b1dec45fc7e4490e8081d444a25a00bda + +2020-05-27 03:47 +0000 [d2500c6273] Joshua C. Colp + + * res_fax: Don't consume frames given to fax gateway on write. + + In a particular fax gateway scenario whereby it would + have to translate using the read translation path on a + channel the frame being translated would be consumed. + When the frame is in the write path it is not permitted + to free the frame as the caller expects it to continue + to exist. + + This change makes it so that the frame is only consumed + on the read path where it is acceptable to free it. + + ASTERISK-28900 + + Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d + +2020-06-02 06:24 +0000 [0a4dffe6f8] Alexander Traud + + * pjproject_bundled: Honor --without-pjproject. + + The previous change missed that 'make' uses 'PJPROJECT_BUNDLED' anyway. + + ASTERISK-28929 + + Change-Id: I7ef0e78a06ea391b59d95b99d46bbed3fec4fed9 + +2020-06-04 01:50 +0000 [e8c6e9ae5d] Pirmin Walthert + + * res_pjsip_logger: use the correct pointer when logging tx_messages to pcap + + When writing tx messages to pcap files, Asterisk is using the wrong + pointer resulting in lots of wasted space. This patch fixes it to use + the correct pointer. + + ASTERISK-28932 #close + + Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23 + +2020-05-28 20:03 +0000 [25ae412f75] sungtae kim + + * bridge.c: Fixed null pointer exception + + If the bridge show all command could not get the bridge snapshot, it causes null pointer exception. + Fixed it to check the snapshot is null. + + ASTERISK-28920 + + Change-Id: I3521fc1b832bfc69644d0833f2c78177e1e51f58 + +2020-05-14 13:24 +0000 [ca3c22c5f1] George Joseph + + * Scope Tracing: A new facility for tracing scope enter/exit + + What's wrong with ast_debug? + + ast_debug is fine for general purpose debug output but it's not + really geared for scope tracing since it doesn't present its + output in a way that makes capturing and analyzing flow through + Asterisk easy. + + How is scope tracing better? + + Scope tracing uses the same "cleanup" attribute that RAII_VAR + uses to print messages to a separate "trace" log level. Even + better, the messages are indented and unindented based on a + thread-local call depth counter. When output to a separate log + file, the output is uncluttered and easy to follow. + + Here's an example of the output. The leading timestamps and + thread ids are removed and the output cut off at 68 columns for + commit message restrictions but you get the idea. + + --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 + --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 + --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ + --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after + --> chan_pjsip.c:3194 chan_pjsip_incoming_response P + chan_pjsip.c:3245 chan_pjsip_incoming_respon + <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P + <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after + <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ + <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 + <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 + + The messages with the "-->" or "<--" were produced by including + the following at the top of each function: + + SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); + + Scope isn't limited to functions any more than RAII_VAR is. You + can also see entry and exit from "if", "for", "while", etc blocks. + + There is also an ast_trace() macro that doesn't track entry or + exit but simply outputs a message to the trace log using the + current indent level. The deepest message in the sample + (chan_pjsip.c:3245) was used to indicate which "case" in a + "select" was executed. + + How do you use it? + + More documentation is available in logger.h but here's an overview: + + * Configure with --enable-dev-mode. Like debug, scope tracing + is #ifdef'd out if devmode isn't enabled. + + * Add a SCOPE_TRACE() call to the top of your function. + + * Set a logger channel in logger.conf to output the "trace" level. + + * Use the CLI (or cli.conf) to set a trace level similar to setting + debug level... CLI> core set trace 2 res_pjsip.so + + Summary Of Changes: + + * Added LOG_TRACE logger level. Actually it occupies the slot + formerly occupied by the now defunct "event" level. + + * Added core asterisk option "trace" similar to debug. Includes + ability to specify global trace level in asterisk.conf and CLI + commands to turn on/off and set levels. Levels can be set + globally (probably not a good idea), or by module/source file. + + * Updated sample asterisk.conf and logger.conf. Tracing is + disabled by default in both. + + * Added __ast_trace() to logger.c which keeps track of the indent + level using TLS. It's #ifdef'd out if devmode isn't enabled. + + * Added ast_trace() and SCOPE_TRACE() macros to logger.h. + These are all #ifdef'd out if devmode isn't enabled. + + Why not use gcc's -finstrument-functions capability? + + gcc's facility doesn't allow access to local data and doesn't + operate on non-function scopes. + + Known Issues: + + The only know issue is that we currently don't know the line + number where the scope exited. It's reported as the same place + the scope was entered. There's probably a way to get around it + but it might involve looking at the stack and doing an 'addr2line' + to get the line number. Kind of like ast_backtrace() does. + Not sure if it's worth it. + + Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027 + +2020-05-29 04:28 +0000 [c16937cdbe] Pirmin Walthert + + * res_pjsip_logger.c: correct the return value checks when writing to pcap + files + + fwrite() does return the number of elements written and not the + number of bytes. However asterisk is currently comparing the return + value to the size of the written element what means that asterisk logs + five WARNING messages on every packet written to the pcap file. + + This patch changes the code to check for the correct value, which will + always be 1. + + ASTERISK-28921 #close + + Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2 + +2020-05-27 09:35 +0000 [9c2871edf4] Joshua C. Colp + + * res_pjsip: Use correct pool for storing the contact_user value. + + When replacing the user portion of the Contact URI the code + was using the ephemeral pool instead of the tdata pool. This + could cause the Contact user value to become invalid after a + period of time. + + The code will now use the tdata pool which persists for the + lifetime of the message instead. + + ASTERISK-28794 + + Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5 + +2020-05-13 07:06 +0000 [1399f8b4fe] Pirmin Walthert + + * res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header + + While asterisk is filtering out the x-ast-orig-host parameter from the + contact on response messages, it is not filtering it out from the + request URI and the to header on SIP requests (for example INVITE). + + ASTERISK-28884 #close + + Change-Id: Id032b33098a1befea9b243ca994184baecccc59e + +2020-05-18 09:05 +0000 [afa2c9a868] Joshua C. Colp + + * bridge: Don't try to match audio formats. + + When bridging channels we were trying to match the audio + formats of both sides in combination with the configured + formats. While this is allowed in SDP in practice this + causes extra reinvites and problems. This change ensures + that audio streams use the formats of the first existing + active audio stream. It is only when other stream types + (like video) exist that this will result in re-negotiation + occurring for those streams only. + + ASTERISK-28871 + + Change-Id: I22f5a3e7db29e00c165e74d05d10856f6086fe47 + +2020-05-19 07:55 +0000 [ec7890d7c6] Joshua C. Colp + + * res_sorcery_config: Always reload configuration on errors. + + When a configuration file in Asterisk is loaded + information about it is stored such that on a + reload it is not reloaded if nothing has changed. + This can be problematic when an error exists in + a configuration file in PJSIP since the error + will be output at start and not subsequently on + reload if the file is unchanged. + + This change makes it so that if an error is + encountered when res_sorcery_config is loading + a configuration file a reload will always read + in the configuration file, allowing the error + to be seen easier. + + Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed + +2020-05-18 10:10 +0000 [4de0e50c32] Alexander Traud + + * res_srtp: Set all possible flags while selecting the Crypto Suite. + + The flags of a previous selection could have been set within the + object 'srtp', for example, when the previous selection returned + failure after setting just 'some' flags. Now, not to clutter the + code, all possible flags are cleared first, and then the selected + flags are set as before. + + ASTERISK-28903 + + Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee + +2020-05-19 04:18 +0000 [e8c8d69d47] Joshua C. Colp + + * bridge_softmix: Always remove audio from mixed frame. + + When receiving audio from a channel we determine if it + is talking or silence based on a threshold value. If + this threshold is met we always mix the audio into the + conference bridge. If this threshold is not met we also + mix the audio into the conference bridge UNLESS the + drop silence option is enabled. + + The code that removed the audio from the mixed frame + assumed that it was always not present if it did not + meet the threshold to be considered talking. This is + incorrect. If it has been stated that the audio was + mixed into the mixed frame then it has been mixed into + the mixed frame. By not removing audio that was + considered non-talking it was possible for a channel + to receive a slight echo of audio of itself at times. + + This change ensures that the audio is always removed + from the mixed frame going back to the channel so it + no longer receives the slight echo. + + ASTERISK-28898 + + Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb + +2020-05-13 16:37 +0000 [f506cc4896] Ben Ford + + * res_stir_shaken: Add unit tests for signing and verification. + + Added two unit tests, one for signing and another for verifying. + stir_shaken_sign checks to make sure that all the required parameters + are passed in and then signs the actual payload. If a signature is + produced and a payload returned as a result, the test passes. + stir_shaken_verify takes the signature from a signed payload to verify. + This unit test also verifies that all the required information is passed + in, and then attempts to verify the signature. If verification is + successful and a payload is returned, the test passes. + + Change-Id: I9fa43380f861ccf710cd0f6b6c102a517c86ea13 + +2020-04-30 17:57 +0000 [a7aaee70c6] Joshua C. Colp + + * res_pjsip_logger: Expand functionality to improve logging. + + The PJSIP packet logger now has the following CLI commands: + + pjsip set logger pcap + + When used this will create a pcap file containing the incoming + and outgoing SIP packets, in unencrypted form. + + pjsip set logger verbose + + This allows you to toggle logging to verbose on and off. + + pjsip set logger host add + + This allows you to add an additional IP address or subnet + mask to logging, allowing you to log multiple instead of + just a single IP address or all traffic. + + The normal "pjsip set logger host" CLI command has also been + expanded to allow subnet masks as well. + + ASTERISK-28895 + + Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e + +2020-05-13 13:32 +0000 [fef97a9a72] Nicholas John Koch + + * res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings + + A warning was triggered that there may be a problem regarding file + extension (which is correct and should not be set anyway). The warning + also appeared if there was dot within the path itself. + + E.g. + [sales-queue-hold] + mode=playlist + entry=/var/www/domain.tld/moh/funky_music + + The music played correctly but you get a warning message. + + Now there will be a check if the position of a potential dot character + is after the last position of a slash character. This dot charachter + will be treated as a extension naming. Dots within the path then ignored. + + ASTERISK-28892 + Reported-By: Nicholas John Koch + + Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e + +2020-05-18 11:31 +0000 [c8c94b6cf1] sungtae kim + + * res_rtp_asterisk.c: Fixed memory leak + + Added freeifaddrs() for memory releasing. + + ASTERISK-28904 + + Change-Id: I109403866e85a30659351946903a679de9727a8f + +2020-05-12 18:15 +0000 [15cbff9d54] Joshua C. Colp + + * ari: Allow variables to be set on channel create. + + This change adds the same variable functionality that + is available for originating a channel to the create + call. Now when creating a channel you can specify + dialplan variables to set instead of having to do another + API call. + + ASTERISK-28896 + + Change-Id: If13997ba818136d7c070585504fc4164378aa992 + +2020-05-10 05:01 +0000 [c8dec423d2] Peter Sokolov (License #7070) + + * pjsip_resolver.c: Ensure AAAA dns requests are made. + + 1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups + when an IPV6 transport type has been requested. + + 2. Rename all occurrences of pjsip_transport_get_type_name to + pjsip_transport_get_type_desc. This ensures that the log/debug info + shows whether the transport is IPv6 or IPv4. + + 3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport + types. This results in invalid values. Use a bitwise or instead. + + ASTERISK-26780 + Patches: + pjsip_resolver.c uploaded by Peter Sokolov (License #7070) + + Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58 + +2020-05-04 16:11 +0000 [e29df34de0] Ben Ford + + * res_stir_shaken: Added dialplan function and API call. + + Adds the "STIR_SHAKEN" dialplan function and an API call to add a + STIR_SHAKEN verification result to a channel. This information will be + held in a datastore on the channel that can later be queried through the + "STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results + including identity, attestation, and verify_result. Here are some + examples: + + STIR_SHAKEN(count) + STIR_SHAKEN(0, identity) + STIR_SHAKEN(1, attestation) + STIR_SHAKEN(2, verify_result) + + Getting the count can be used to iterate through the results and pull + information by specifying the index and the field you want to retrieve. + + Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82 + +2020-05-08 06:11 +0000 [801d570f6e] Guido Falsi + + * pjproject: Fix race condition when building with parallel make + + Pjproject makefiles miss some dependencies which can cause race + conditions when building with parallel make processes. This patch + adds such dependencies correctly. + + ASTERISK-28879 #close + Reported-by: Dmitry Wagin + + Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22 + +2020-05-09 02:46 +0000 [4a072c4890] Roger James + + * res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered. + + Changed source and destination address fields in struct + pjsip_history_entry so that they are long enough to hold an IPv6 + address. + + ASTERISK-28854 + + Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e + +2020-04-01 08:38 +0000 [f9ea75d117] Alexander Traud + + * tcptls: Fix notice when TLS is enabled but not supported. + + ASTERISK-28797 + + Change-Id: Iab364a2c2519fd9d11d1c28293fda43d61b64c28 + +2020-04-04 04:28 +0000 [527e4f6542] Alexander Traud + + * app_osplookup: Avoid a format truncation. + + Ensure that output buffers for the osp_convert_inout + function have sufficient space for additional data + such as brackets and ports. + + ASTERISK-28804 + + Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b + +2020-04-14 11:02 +0000 [6b2d945174] Pirmin Walthert + + * app.c: make sure that no non-async-signal-safe syscalls are used after + fork before exec + + Posix does only allow async-signal-safe syscalls after fork before exec. + As asterisk ignores this, functions like TrySystem or System sometimes + end up in a deadlocked child process. The patch prevents the use of + non-async-signal-safe syscalls. + + ASTERISK-28776 + + Change-Id: Idc76365c0592ee3f3b3bd72a4f48f7a098978e8e + +2020-05-04 11:31 +0000 [7fbfbe7da0] George Joseph + + * streams: Fix one memory leak and one formats ref issue + + ast_stream_topology_create_from_format_cap() was setting the + stream->formats directly but not freeing the default formats. This + causes a memory leak. + + * ast_stream_topology_create_from_format_cap() now calls + ast_stream_set_formats() which properly cleans up the existing + stream formats. + + When cloning a stream, the source stream's format caps _pointer_ is + copied to the new stream and it's reference count bumped. If + either stream is set to "removed", this will cause _both_ streams + to have their format caps cleared. + + * ast_stream_clone() now creates a new format caps object and copies + the formats from the source stream instead of just copying the + pointer. + + ASTERISK-28870 + + Change-Id: If697d81c3658eb7baeea6dab413b13423938fb53 + +2020-04-08 18:41 +0000 [f217fcdc62] Nathan Bruning + + * app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions + + Add a new "masquarade" channel event, and use it in app_queue to track unique id's. + + Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 + + ASTERISK-28829 #close + ASTERISK-25844 #close + + Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6 + +2020-05-04 03:29 +0000 [44e5dd288b] Jaco Kroon + + * Remove #include + + These are not provided by standards, and as a result causes failure to + compile on musl. + + https://wiki.musl-libc.org/faq.html#Q:-When-compiling-something-against-musl,-I-get-error-messages-about-%3Ccode%3Esys/cdefs.h%3C/code%3E + + Change-Id: I6a357cefd106c72cfecafd898638f6b5692c2e05 + +2020-05-03 05:30 +0000 [c831f03273] Guido Falsi + + * pjproject: Remove bashism from configure.m4 script + + The configure.m4 script for pjproject contains some += syntax, which + is specific to bash, replacing it with string substitutions makes + the script compatible with traditional Bourne shells. + + ASTERISK-28866 #close + Reported-by: Christoph Moench-Tegeder + + Change-Id: I382a78160e028044598b7da83ec7e1ff42b91c05 + +2020-05-01 07:29 +0000 [1cfd30bd8a] Joshua C. Colp + + * res_stir_shaken: Use ast_asprintf for creating file path. + + Change-Id: Ice5d92ecea2f1101c80487484f48ef98be2f1824 + +2020-04-15 13:15 +0000 [9acf840f7c] Ben Ford + + * res_stir_shaken: Implemented signature verification. + + There are a lot of moving parts in this patch, but the focus of it is on + the verification of the signature using a public key located at the + public key URL provided in the JSON payload. First, we check the + database to see if we have already downloaded the key. If so, check to + see if it has expired. If it has, redownload from the URL. If we don't + have an entry in the database, just go ahead and download the public + key. The expiration is tested each time we download the file. After + that, read the public key from the file and use it to verify the + signature. All sanity checking is done when the payload is first + received, so the verification is complete once this point is reached. + + The XML has also been added since a new config option was added to + general (curl_timeout). The maximum amount of time to wait for a + download can be configured through this option, with a low value by + default. + + Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c + +2020-04-30 10:56 +0000 [7baf2c4bf1] George Joseph + + * app_voicemail: Add workaround for a gcc 10 issue with -Wrestrict + + The gcc 10 -Wrestrict option was causing "overlap" errors when + snprintf was copying one char[256] structure member to another + char[256] member in the same structure. + + Using ast_alloca instead of declaring the structure inline + solves the issue. + + Here's a link to the "enhancement": + https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html + + We may follow up with a gcc bug report. + + Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534 + +2020-04-28 10:31 +0000 [3078a00a6d] Joshua C. Colp + + * pjsip: Increase maximum ICE candidate count. + + In practice it has been seen that some users come + close to our maximum ICE candidate count of 32. + In case people have gone over this increases the + count to 64, giving ample room. + + ASTERISK-28859 + + Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf + +2020-04-27 10:28 +0000 [29070b61f7] Alexander Traud + + * core_local: Local calls are always secure. + + In a Dialplan, the channel drivers 'chan_sip' and 'chan_iax2' support + the channel items 'secure_bridge_media' and 'secure_bridge_signaling'. + That way, a channel can be forced to use encryption even if not + specified in its configuration. + + However, when the Local Proxy kicks in, for example, in case of a + forwarding (SIP status 302), Local Proxy stated it does not know those + items. Consequently, such a call could not be proxied how clever your + Dialplan was. Because local calls within Asterisk are always secure, + Local Proxy accepts such a request now. + + ASTERISK-22920 + + Change-Id: I4c143bb70f686790953cc04c5a4b810bbb03636c + +2020-04-26 05:56 +0000 [e4366308e1] Guido Falsi + + * res_rtp_asterisk: Protect access to nochecksums with #ifdef + + Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined. + + ASTERISK-28852 #close + + Change-Id: I381718893b80599ab8635f2b594a10c1000d595e + +2020-04-26 06:08 +0000 [97494d8984] Guido Falsi + + * core/dns: Add system include required on FreeBSD + + While testing the latest RC on FreeBSD I noticed this new file fails to build. On FreeBSD inlcuding resolv.h requires sockaddr_in to be defined, and it's defined in netinet/in.h. So I added this include. + + ASTERISK-28853 #close + + Change-Id: I6997daf3956e6eb70ab6cb358628d162fad80079 + +2020-04-17 02:39 +0000 [3303defd3f] Peter Turczak + + * chan_mobile: Add smoother to make SIP/RTP endpoints happy. + + In contrast to RFC 3551, section 4.2, several SIP/RTP clients misbehave + severly (up to crashing). This patch adds another smoother for the audio + received via bt. Therefore the audio frames sent to the core will be + CHANNEL_FRAME_SIZE. + + ASTERISK-28832 #close + + Change-Id: Ic5f9e2f35868ae59cc9356afbd1388b779a1267f + +2020-04-22 12:38 +0000 [26b8c99963] Alexander Traud + + * app_fax: SpanDSP headers do not use ast_malloc; ignore that. + + Since Asterisk 14, app_fax did not compile at all because Asterisk + requires that not malloc but ast_malloc is used everywhere. However, + the system headers of SpanDSP use malloc. Because we cannot (and do + not need to) change system headers, let us ignore this. + + ASTERISK-28848 + + Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546 + +2020-04-21 04:52 +0000 [1c5e68580a] Joshua C. Colp + + * stream: Enforce formats immutability and ensure formats exist. + + Some places in Asterisk did not treat the formats on a stream + as immutable when they are. + + The ast_stream_get_formats function is now const to enforce this + and parts of Asterisk have been updated to take this into account. + Some violations of this were also fixed along the way. + + An additional minor tweak is that streams are now allocated with + an empty format capabilities structure removing the need in various + places to check that one is present on the stream. + + ASTERISK-28846 + + Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe + +2020-04-21 10:40 +0000 [9ad3d2829c] sungtae kim + + * res_ari_channels: Fixed endpoint 80 characters limit + + Fixed it to copy the entire string from the requested endpoint body except tech-prefix. + + ASTERISK-28847 + + Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25 + +2020-04-20 10:18 +0000 [e56f4de7e6] Joshua C. Colp + + * fax: Fix crashes in PJSIP re-negotiation scenarios. + + This change fixes a few re-negotiation issues + uncovered with fax. + + 1. The fax support uses its own mechanism for + re-negotiation by conveying T.38 information in + its own frames. The new support for re-negotiating + when adding/removing/changing streams was also + being triggered for this causing multiple re-INVITEs. + The new support will no longer trigger when + transitioning between fax. + + 2. In off-nominal re-negotiation cases it was + possible for some state information to be left + over and used by the next re-negotiation. This + is now cleared. + + ASTERISK-28811 + ASTERISK-28839 + + Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2 + +2020-04-15 15:13 +0000 [9f117ac9ef] Daniel Heckl + + * res_pjsip: Fixed format of IPv6 addresses for external media addresses + + ASTERISK-28835 + + Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26 + +2020-04-20 13:11 +0000 [52f07176b6] Alexander Traud + + * chan_sip: externhost/externaddr with non-default TCP/TLS ports. + + ASTERISK-28372 + Reported by: Anton Satskiy + + ASTERISK-24428 + Reported by: sstream + + Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c + +2020-04-17 05:41 +0000 [abf4d74384] Alexander Traud + + * cdr_odbc: Sync load- and build-time deps. + + MODULEINFO is checked while buidling/linking the module. + AST_MODULE_INFO is checked while loading/running the module. + + ASTERISK-28838 + + Change-Id: I55dc05ce19552d0415c9045021b42bd82ef44e52 + +2020-04-16 08:15 +0000 [6cfc6ff53c] Joshua C. Colp + + * confbridge: Add support for disabling text messaging. + + When in a conference bridge it may be necessary to have + text messages disabled for specific participants or for + all. This change adds a configuration option, "text_messaging", + which can be used to enable or disable this on the + user profile. By default existing behavior is preserved + as it defaults to "yes". + + ASTERISK-28841 + + Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13 + +2020-04-17 04:18 +0000 [191f136260] Alexander Traud + + * res_pjsip_refer: Add build-time dependency. + + ASTERISK-28838 + + Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25 + +2020-04-17 05:17 +0000 [5c2b8fdeca] Alexander Traud + + * app_getcpeid: Add build-time dependency. + + ASTERISK-28838 + + Change-Id: I68b78e7e4718be82507247433127ce3992a5ba96 + +2020-04-17 04:47 +0000 [008f46bf1e] Alexander Traud + + * res_pjsip: Sync load- and build-time deps. + + MODULEINFO is checked while buidling/linking the module. + AST_MODULE_INFO is checked while loading/running the module. + + ASTERISK-28838 + + Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042 + +2020-04-17 06:51 +0000 [e2affa3b0a] Alexander Traud + + * curl: Add build-time dependency. + + ASTERISK-28838 + + Change-Id: I34724e799e1ffaf723eb2c358abe8934dbadcd52 + +2020-04-17 04:55 +0000 [f1135b453b] Alexander Traud + + * res_pjsip: Add build-time dependency. + + ASTERISK-28838 + + Change-Id: Icb08304744ae3f34dce6ccb76f94379b8382a074 + +2020-04-15 13:01 +0000 [966acc6251] Alexander Traud + + * pjproject_bundled: Honor --without-pjproject. + + ASTERISK-28837 + + Change-Id: Id057324912a3cfe6f50af372675626bb515907d9 + +2020-04-14 10:48 +0000 [d50fd0acc0] Pirmin Walthert + + * res_rtp_asterisk: Resolve loop when receive buffer is flushed + + When the receive buffer was flushed by a received packet while it + already contained a packet with the same sequence number, Asterisk + never left the while loop which tried to order the packets. + + This change makes it so if the packet is in the receive buffer it + is retrieved and freed allowing the buffer to empty. + + ASTERISK-28827 + + Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3 + +2020-04-15 07:16 +0000 [a107e85b2e] Alexander Traud + + * install_prereq: Add libcap for high bits in DiffServ/ToS. + + works automatically; see Mantis 7047 (now ASTERISK-6863) + + Change-Id: I27d2c109180bd857b6757fd532de48eddb78aee6 + +2020-04-15 01:20 +0000 [4d0ab620be] Alexander Traud + + * chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets. + + ASTERISK-27195 + Reported by: Joshua Roys + + Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de + +2020-04-15 06:09 +0000 [4ef5ba58f5] Alexander Traud + + * BuildSystem: Only if found LibPRI, check its optional parts. + + Change-Id: If8445f899ee4ce0c606c484943d4ce0c8e43b5da + +2020-04-14 10:31 +0000 [ca032d1e2e] Pirmin Walthert + + * res_rtp_asterisk: Free payload when error on insertion to data buffer + + When the ast_data_buffer_put rejects to add a packet, for example because + the buffer already contains a packet with the same sequence number, the + payload will never be freed, resulting in a memory leak. + + The data buffer will now return an error if this situation occurs + allowing the caller to free the payload. The res_rtp_asterisk module + has also been updated to do this. + + ASTERISK-28826 + + Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1 + +2020-04-14 06:26 +0000 [ef580f96e7] Alexander Traud + + * BuildSystem: Only if found external PJProject, check its optional parts. + + Change-Id: I11d5693d25c166c99d8cebffc16184d58f6362de + +2020-04-08 05:29 +0000 [7db03e12a7] Bernard Merindol + + * res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0 + + When the first DTMF receive in RF2833 codec have TimeStamp at 0 + is not processed. + + ASTERISK-28812 + + Change-Id: I3196803a062dd2daee4938c9a778c3810cb7e504 + +2020-04-13 11:47 +0000 [611529fa52] Alexander Traud + + * res_stir_shaken: Do not build without OpenSSL. + + Change-Id: Idba5151a3079f9dcc0076d635422c5df5845114f + +2020-04-13 11:38 +0000 [27de0c9700] Alexander Traud + + * res_audiosocket: Avoid Sometimes-uninitialized Warning with Clang. + + Change-Id: I40c014c2cb88e943cf6f1b99a08c7c885e855b3a + +2020-04-07 07:05 +0000 [de66713fd5] Jean Aunis + + * func_volume: Accept decimal number as argument + + Allow voice volume to be multiplied or divided by a floating point number. + + ASTERISK-28813 + + Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c + +2019-12-03 12:35 +0000 [2b80e5f5da] Jaco Kroon + + * res_rtp_asterisk: iterate all local addresses looking to populate ICE. + + By using pjproject to give us a list of candidates, and then filtering, + if the host has >32 addresses configured, then it is possible that we + end up filtering out all 32 of those, and ending up with no candidates + at all. Instead, get getifaddrs (which pjsip is using underlying + anyway) to retrieve all local addresses, and iterate those, adding the + first 32 addresses not excluded by the ICE ACL. + + In our setup at any point in time We've got between 6 and 328 addresses + on any given system. The lower limit is the lower limit but the upper + limit is growing on a near daily basis currently. + + Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb + Signed-off-by: Jaco Kroon + +2020-04-10 08:13 +0000 [3431949a52] Alexander Traud + + * pjproject_bundled: Repair ./configure --with-ssl without ARG. + + ASTERISK-28758 + Reported by: Patrick Wakano + Reported by: Dmitriy Serov + + Change-Id: Ifb6b85c559d116739af00bc48d1f547caa85efac + +2020-04-11 14:03 +0000 [1cf569ba2b] Jaco Kroon + + * res_pjsip: document legal dtls_verify endpoint options. + + Change-Id: I7fa7c5c8a7ddb0bd525982f58bff3264ebbd9a1b + +2020-04-12 09:53 +0000 [610e058189] Alexander Traud + + * BuildSystem: Search for Python/C API when possibly needed only. + + The Python/C API is used only if the Test Framework was enabled in Asterisk + 'make menuselect'. The Test Framework is available only if the Developer Mode + was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API + is used only if the PJProject was found and not disabled in Asterisk; the user + did not go for './configure --without-pjproject'. + + Furthermore, because version 2 of that Python/C API is required (currently) and + because some platforms do not offer a generic version 2, the script searches + for 2.7 explicitly as well. + + To avoid version mismatch between the Python/C API and the Python environment, + the script searches for the latter in the same versions, in the same the order + as well. Because this Python/C API is just for (some) Asterisk contributors, + the script also goes for the Python 3 environment as a last resort for all + other Asterisk users. This allows 'make full' even on minimal installations of + Ubuntu 18.04 LTS and newer. + + Because the Python/C API is Asterisk contributor specific, the Python packages + are removed from the script './contrib/scripts/install_prereq' as this script + is intended for Asterisk users. Asterisk contributors have to install much more + packages in any case, like: + sudo apt install autoconf automake git git-review python2.7-dev + + ASTERISK-28824 + ASTERISK-27717 + + Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876 + +2020-04-01 11:52 +0000 [da9554d925] Alexander Traud + + * chan_sip: TCP/TLS client without server. + + It is possible to configure a TCP/TLS client without having a TCP/TLS + server. In that case, no error or warning was printed but the headers + Contact and Via in SIP REGISTER were "(null)". + + ASTERISK-28798 + + Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2 + +2020-04-13 12:05 +0000 [52ecbbd014] Alexander Traud + + * _pjsua: Build even with Clang. + + Change-Id: Iebf7687613aa0295ea3c82256460b337f1595be2 + +2020-04-13 11:27 +0000 [ee1c7f465b] Alexander Traud + + * res_rtp_asterisk: Build without PJProject. + + Change-Id: Ifc5059cd867e77b9c92ed9f4b895a9a91200d3ec + +2020-04-08 14:33 +0000 [fa3c8f94e0] Kevin Harwell + + * chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet + + If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a + digit begin before media, or rtp has been setup then it's possible the + outgoing channel will hear a constant DTMF tone upon answering. + + This happens because when there is no media, or rtp chan_pjsip notifies + the core to initiate inband DTMF. However, upon digit end if media, and + rtp become available then chan_pjsip does not notify the core to stop + inband DTMF. Thus the tone continues playing. + + This patch makes it so chan_pjsip only notifies the core to start + inband DTMF in only the required cases. Now if there is no media, or + rtp availabe upon digit begin chan_pjsip does nothing, but tells the + core it handled it. + + ASTERISK-28817 #close + + Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5 + +2020-04-09 08:25 +0000 [7febd22304] Alexander Traud + + * bridge_softmix_binaural: Show state in menuselect. + + ASTERISK-28819 + + Change-Id: Iba7ee7bc7936d7a156171c8fc0f1670e864e7600 + +2020-04-07 12:44 +0000 [7cdb493a1e] Alexander Traud + + * BuildSystem: Remove doc/tex and doc/pdf leftovers. + + Furthermore, the nowhere used compress is removed. + + ASTERISK-28816 + + Change-Id: I77daab80cfabb56d51c3ea6b1d14bd9b9fbc577c + +2020-04-09 07:05 +0000 [7a04947abd] Alexander Traud + + * BuildSystem: Allow space in path. + + ASTERISK-28818 + + Change-Id: Ib7f246896457d9e3b14d7f5199136d6545ce0b6f + +2020-04-06 08:00 +0000 [1ef1b1b0c2] Alexander Traud + + * res_rtp_asterisk: Avoid absolute value on unsigned subtraction. + + ASTERISK-28809 + + Change-Id: I269731715347c8e5ef7db1b6ffd3f8d15fc04be4 + +2020-03-31 15:14 +0000 [d40e343710] Sebastien Duthil + + * func_channel: allow reading 4 fields from dialplan + + The following fields return an error when read from dialplan: + + - exten + - context + - userfield + - channame + + ASTERISK-28796 #close + + Change-Id: Ieacaac629490f8710fdacc9de80ed5916c5f6ee2 + +2020-04-03 12:25 +0000 [b38f664250] Alexander Traud + + * chan_unistim: Avoid tautological warnings with clang. + + ASTERISK-28803 + + Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af + +2020-04-06 06:56 +0000 [bb28ed0d1b] Alexander Traud + + * test_stasis: Avoid always true warning with clang. + + ASTERISK-28808 + + Change-Id: I5e76831373532d7b8065d024e66cd1fb75dedd80 + +2020-04-06 09:29 +0000 [60925c68e8] Sean Bright + + * Revert "res_config_odbc: Preserve empty strings returned by the database" + + This reverts commit a3a2fbaec685d931d56f669f2d4171220e9977ac. + + Reason for revert: There is a lot of code that relies on the broken + behavior that this fixes. + + Change-Id: I410c395a0168acbdaf89e616e3cb5e1312d190cb + +2020-04-01 04:00 +0000 [c5f3836bcc] Jaco Kroon + + * main/backtrace: binutils-2.34 fix. + + My tester missed this one previously, have confirmed a positive build + this time round. + + Change-Id: Id06853375954a200f03f9a1b9c97fe0b10d31fbf + +2020-03-26 17:42 +0000 [d845464c76] Jason Hord (license 6978) + + * res_pjsip: Don't set endpoint to unavailable in all cases. + + When an AOR is modified endpoints are updated that reference + the AOR so they can start receiving updates and reflect the + correct state. If this is the case then we shouldn't change + the endpoint to be offline if it does not reference the AOR + but instead only when the endpoint is completely updated for + all its AORs. + + ASTERISK-28056 + patches: + pjsip_options-aor.diff submitted by jhord (license 6978) + + Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164 + +2020-03-25 12:51 +0000 [7ba6d43083] George Joseph + + * test_res_pjsip_session_caps: Create unit test + + This unit test runs through combinations of... + * Local codecs + * Remote Codecs + * Codec Preference + * Incoming/Outgoing + + A few new APIs were created to make it easier to test + the functionality but didn't result in any actual + functional change. + + ASTERISK_28777 + + Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164 + +2020-03-13 14:40 +0000 [2ee455958e] George Joseph + + * codec_negotiation: Implement outgoing_call_offer_pref + + Based on this new endpoint setting, a joint list of preferred codecs + between those received from the Asterisk core (remote), and those + specified in the endpoint's "allow" parameter (local) is created and + is used to create the outgoing SDP offer. + + * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) + + * Add "call_direction" to res_pjsip_session. + + * Update pjsip_session_caps.c to make the functions more generic + so they could be used for both incoming and outgoing. + + * Update ast_sip_session_create_outgoing to create the + pending_media_state->topology with the results of + ast_sip_session_create_joint_call_stream(). + + * The endpoint "preferred_codec_only" option now automatically sets + AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. + + * A helper function ast_stream_get_format_count() was added to + streams to return the current count of formats. + + ASTERISK-28777 + + Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437 + +2020-03-26 13:34 +0000 [57a457c26c] Ben Ford + + * res_stir_shaken: Implemented signing of JSON payload. + + This change provides functions that take in a JSON payload, verify that + the contents contain all the mandatory fields and required values (if + any), and signs the payload with the private key. Four fields are added + to the payload: x5u, attest, iat, and origid. As of now, these are just + placeholder values that will be set to actual values once the logic is + implemented for what to do when an actual payload is received, but the + functions to add these values have all been implemented and are ready to + use. Upon successful signing and the addition of those four values, a + ast_stir_shaken_payload is returned, containing other useful information + such as the algorithm and signature. + + Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393 + +2020-03-31 12:52 +0000 [3c345ec56d] Kevin Harwell + + * channel: write to a stream on multi-frame writes + + If a frame handling routine returns a list of frames (vs. a single frame) + those frames are never passed to a tech's write_stream handler even if one is + available. For instance, if a codec translation occurred and that codec + returned multiple frames then those particular frames were always only sent + to the tech's "write" handler. If that tech (pjsip for example) was stream + capable then those frames were essentially ignored. Thus resulting in bad + audio. + + This patch makes it so the "write_stream" handler is appropriately called + for all cases, and for all frames if available. + + ASTERISK-28795 #close + + Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d + +2020-03-24 06:43 +0000 [fc07eeaba1] Alexander Traud + + * test_utils: Avoid incorrect error message on load. + + In case of no OpenSSL headers, the module was built but did not load. + + ASTERISK-28789 + + Change-Id: Ie007e84296bcf2bd4237f19d68ba5f932b84cd02 + +2020-03-23 12:25 +0000 [cd8cbf7384] Alexander Traud + + * func_aes: Avoid incorrect error message on load. + + In case of no OpenSSL headers, the module func_aes was built but did not load. + + ASTERISK-28788 + + Change-Id: I0b99b8468cbeb3b0eab23069cbd64062ef885ffc + +2020-03-26 17:18 +0000 [dbddb6725d] sungtae kim + + * dial.c: Removed dial string 80 character limitation + + The dial application had 80 characters of destination length + limitation. But this limitation causes unexpected dial string + cut if the dial string is long. + + Removed unnecessary limited buffer to support longer dial + destination. + + ASTERISK-27946 + + Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330 + +2020-03-19 04:34 +0000 [e12244153a] Torrey Searle + + * res_pjsip_session: implement processing of Content-Disposition + + RFC5621 requires any content type with a Content-Disposition + with handling=required to be rejected with a 415 response + + ASTERISK-28782 #close + + Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4 + +2020-03-18 08:49 +0000 [d32e559e8a] Jaco Kroon + + * acl: implement a centralized ACL output mechanism for HAs and ACLs. + + named_acl.c (which is really a named_ha) now uses ast_ha_output. + + I've also updated main/manager.c to output the actual ACL on "manager + show user " if one is set. If this works then we can add + similar to other modules as required. + + Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f + +2020-03-26 08:49 +0000 [1b6c58896f] Joshua C. Colp + + * chan_sip: Send 403 when ACL fails. + + Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2 + +2020-03-26 11:42 +0000 [3ed80fc57b] Joshua C. Colp + + * CHANGES: Change md file extension to txt. + + Change-Id: I168e2d3a65d444fb0961bd228257441fe718f6a7 + (cherry picked from commit c9cd68126152bae26d42f5b9ce8811ddf1eda4d8) + +2020-03-23 05:49 +0000 [21e9051461] Joshua C. Colp + + * res_pjsip_session: Apply intention behind requested formats. + + When an outgoing channel is created a list of formats may + optionally be provided which is used as a request that the + formats be used if possible. If an endpoint is not configured + for any of the formats we ignore this request and use what is + configured. This has the side effect of also including other + stream types (such as video) that were not present in the + requested formats. + + This change makes it so that the intention of the request is + preserved - that is if only an audio format is requested then + even if there is no joint audio format between the request and + the configuration we will still only place an audio stream in + the outgoing call. + + ASTERISK-28787 + + Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f + +2020-03-25 04:38 +0000 [96e8d411e1] Joshua C. Colp + + * res_rtp_asterisk: Ensure sufficient space for worst case NACK. + + ASTERISK-28790 + + Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99 + +2020-03-17 15:54 +0000 [26713dc88b] Kevin Harwell + + * ast_coredumper: add Asterisk information dump + + This patch makes it so ast_coredumper now outputs the following information to + a *-info.txt file when processing a core file: + + asterisk version and "built by" string + BUILD_OPTS + system start, and last reloaded date/time + taskprocessor list + equivalent of "bridge show all" + equivalent of "core show channels verbose" + + Also a slight modification was made when trying to obtain the pid(s) of a + running Asterisk. If it fails to retrieve any it now reports an error. + + Change-Id: I54f35c19ab69b8f8dc78cc933c3fb7c99cef346b + +2020-03-20 09:12 +0000 [6f731f153b] Jaco Kroon + + * netsock2: compile fixes. + + This fixes ast_addressfamily_to_sockaddrsize to reference the + provided argument, and ast_sockaddr_from_sockaddr to not use the name of + a structure as argument. + + Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7 + +2020-03-23 15:00 +0000 [211bb8a79c] Ben Ford + + * res_stir_shaken: Initial commit and reading private key. + + This commit sets up some of the initial framework for the module and + adds a way to read the private key from the specified file, which will + then be appended to the certificate object. This works fine for now, but + eventually some other structure will likely need to be used to store all + this information. Similarly, the caller_id_number is specified on the + certificate config object, but in the end we will want that information + to be tied to the certificate itself and read it from there. + + A method has been added that will retrieve the private key associated + with the caller_id_number passed in. Tab completion for certificates and + stores has also been added. + + Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f + +2020-03-18 04:21 +0000 [4f92dcd66b] Jaco Kroon + + * dahdiras: Only set plugin dahdi.so to pppd if we're running as root. + + Users of this should set plugin dahdi.so in their options file. + + ASTERISK-16676 + + Change-Id: I6d01ad0a10e9fea477876d0941c3f38aac357e91 + +2020-03-18 04:38 +0000 [40e93b0240] Jaco Kroon + + * dundi: fix NULL dereference. + + If a negative (error) return is received from dundi_lookup_internal, + this is not handled correctly when assigning the result to the buffer. + As such, use a signed integer in the assignment and do a proper + comparison. + + ASTERISK-21205 + + Change-Id: I5214ebb6491e2bd14f90c7d3ce229da86888f739 + +2020-03-19 13:34 +0000 [34750d2068] Joshua C. Colp + + * res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream. + + When examining a stream to determine hold/unhold information we + only care about the default audio stream. Other streams aren't + used for hold/unhold. + + ASTERISK-28784 + + Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4 + +2020-02-14 02:45 +0000 [8147f43756] Sungtae Kim + + * res_pjsip_session: Fixed wrong session termination + + When the Asterisk receives 200 OK with invalid SDP, + the Asterisk/PJPROJECT terminating the session. + But if the channel was in the Bridge, Asterisk tries send + the Re-Invite before terminating the session. + And when the Asterisk sending the Re-Invite, it doesn't check + the SDP is NULL or not. This crashes the Asterisk. + + Fixed it to close the session correctly if the UAS sends the + 200 OK with wrong SDP. + + ASTERISK-28743 + + Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da + +2020-03-18 04:49 +0000 [a699e016dd] Jaco Kroon + + * build: enable building with uClibc + + This patch has been included in Gentoo distribution for at least since + asterisk 1.8, but there are references in the logs going back as far as + 1.0.0 - not sure if this is still required in any way, it does apply, + and it doesn't (as far as we can determine) cause build failures. + + Change-Id: I46d8845e30200205e80580680bf060aa3012ba54 + +2020-03-18 04:43 +0000 [f824cd6a13] Jaco Kroon + + * build: search from newest to oldest for gmime. + + We (Gentoo distribution) reckon that in the case of multiple versions of + gmime installed we should prefer the newest one. + + Change-Id: Idf7be613230232eb1d573d93c4a5a8297f4ecd2d + +2020-03-19 08:48 +0000 [9620ecbf80] Joshua C. Colp + + * res_pjsip_session: Don't restrict non-audio default streams to sendrecv. + + The state of the default audio stream is used for hold/unhold so we + restrict it to sendrecv as the core does not handle when it changes as + a result of hold/unhold. + + This restriction does not apply to other media types though so we now + only restrict it to audio. This allows the other default streams to + store their state at all values, and not just sendrecv and removed. + + ASTERISK-28783 + + Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625 + +2020-03-06 10:50 +0000 [5562fb2ea0] Michael Neuhauser + + * chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active + + Do not hang up a PJSIP channel on RTP timeout if that channel is in + a direct-media bridge. Also reset the time of the last received RTP packet when + direct-media ends (wait full rtp_timeout period before checking first time after + audio came back to Asterisk). + + ASTERISK-28774 + Reported-by: Michael Neuhauser + + Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1 + +2019-11-27 07:54 +0000 [82c3939c38] Jaco Kroon + + * res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. + + A pure blacklist is not good enough, we need a whitelist mechanism as + well, and the simplest way to do that is to re-use existing ACL + infrastructure. + + This makes it simpler to blacklist say an entire block (/24) except a + smaller block (eg, a /29 or even a /32). Normally you'd need to + recursively split the block, so if you want to blacklist a /24 except + for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I + feel that having an ACL instead of a blacklist only is clearer. + + Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 + Signed-off-by: Jaco Kroon + +2020-03-16 05:11 +0000 [2ad64e97c0] Jaco Kroon + + * Update main/backtrace.c to deal with changes in binutils 2.34. + + binutils 2.34 merged this commit: + + https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\ + h=fd3619828e94a24a92cddec42cbc0ab33352eeb4 + + Which effectively does things like: + + -#define bfd_section_size(bfd, ptr) ((ptr)->size) + -#define bfd_get_section_size(ptr) ((ptr)->size) + + +#define bfd_section_size(sec) ((sec)->size) + + So in order to remain backwards compatible we need to detect this API + change, and adjust accordingly. The simplest is to notice that the + bfd_get_section_size and bfd_get_section_vma MACROs are no longer + defined, and define then onto the new API. The alternative is to litter + the code with a number of #ifdef #else #endif splatters right through + the code. + + Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f + +2020-03-15 09:07 +0000 [c4e0983742] Sean Bright + + * func_odbc.conf.sample: Clarify sample documentation + + ASTERISK-20325 #close + + Change-Id: I06cb9b467b0fd06c8af2a5aee049f872c09cc4b6 + +2020-03-13 13:43 +0000 [49cf84578e] Sean Bright + + * chan_vpb: Fix 'catching polymorphic type ... by value' error + + Fixes the following compile error: + + chan_vpb.cc:2688:26: error: catching polymorphic type + ‘class std::exception’ by value + + Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649 + +2020-03-09 19:07 +0000 [d68f940f6e] Sean Bright + + * dns_txt: Add TXT record parsing support + + Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5 + +2020-03-12 09:22 +0000 [98d10d0a16] Joshua C. Colp + + * audiohook: Don't allow audiohooks to attach to hung up channels. + + Given a scenario where MixMonitor was initiated over AMI it + was possible for the channel and MixMonitor thread to remain + alive past hang up of the channel. This scenario required + the AMI initiated MixMonitor to retrieve the channel, a + hangup to occur on the channel in another thread, and then + for MixMonitor to actually start. If this occurred the + MixMonitor thread would remain alive indefinitely and + the channel reference would remain. + + This change ensures that audiohooks are never able to + be attached to channels that have been hung up. An + additional fix has also been done in app_mixmonitor to + properly release the channel reference if this occurs. + + ASTERISK-28780 + + Change-Id: I8044c06daa06f0f16607788c596f55623be26f58 + +2020-03-04 15:45 +0000 [00a7e4b51d] George Joseph + + * CI: Create generic jenkinsfile + + This is a generic jenkinsfile to build Asterisk and optionally + perform one or more of the following: + * Publish the API docs to the wiki + * Run the Unit tests + * Run Testsuite Tests + + This job can be triggered manually from Jenkins or be triggered + automatically on a schedule based on a cron string. + + Change-Id: Id9d22a778a1916b666e0e700af2b9f1bacda0852 + +2020-03-06 10:13 +0000 [a1dba820cf] Torrey Searle + + * res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use + + bridge_p2p_rtp_write will forward rtp to the bridged rtp instance + without modifying the ssrc. However, it is not updating the SSRC + in the bridged rtp. Thus, when SSRC packets are generated, they + have the correct SSRC for the sender. + + ASTERISK-28773 #close + + Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478 + +2020-03-05 03:08 +0000 [14ba1806f3] Torrey Searle + + * res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated + + If ICE support is enabled but not negotiated, the rtp->ice structure is + not being destroyed. This leads to Asterisk waiting for ICE to complete + instead of immediately starting the DTLS handshake, resulting in the + call leg having no RTP. + + ASTERISK-28769 #close + + Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27 + +2020-02-25 18:30 +0000 [ed2a7e3eaf] Paulo Vicentini + + * chan_pjsip: Check audio frame when remote SSRC changes. + + If the SSRC of a received RTP packet differed from the previous SSRC + an SSRC change control frame would be queued ahead of the media + frame. In the case of audio this would result in the format of the + audio frame not being checked, and if it differed or was not allowed + then it could cause the call to drop due to failure to set up a + translation path. + + The chan_pjsip module will now no longer assume the first frame + will be the audio frame and instead goes through the complete list + to find it. + + ASTERISK-28759 + + Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec + +2020-03-06 14:59 +0000 [517224ce85] Sean Bright + + * enum.c: Add support for regular expression flag in NAPTR record + + A regular expression in a NAPTR response record can have a trailing + 'i' flag to indicate that the expression should be evaluated in a + case-insensitive way. We were not checking for that flag which caused + the record parsing to fail on otherwise valid input. + + Although this change will initially go into Asterisk 13, 16, and 17, + it is my intention to replace the majority of this code in 16 and up - + including this fix - by changing enum.c to consume the new DNS API + which duplicates most of this logic already. Asterisk 13 doesn't have + the DNS API, so this fix will be as good as it gets. + + ASTERISK-26711 #close + Reported by: Vitold + + Change-Id: I33943a5b3e7539c6dca3a5079982ee15a08186f0 + +2020-03-06 06:10 +0000 [0a7fe3097f] Jared Smith + + * indications.conf.sample: Add indication tones for Indonesia + + These tones come from http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf + + ASTERISK-23407 + + Change-Id: I48e2285f1e5bb29b3335f762006f66c423d0fcb8 + +2020-03-03 08:42 +0000 [e089779908] Rodrigo Ramírez Norambuena + + * res_rtp_asterisk: Add 'rtp show settings' cli command + + This change introduce a CLI command for the RTP to display the general + configuration. + + In the first step add the follow fields of the configurations: + - rtpstart + - rtpend + - dtmftimeout + - rtpchecksum + - strictrtp + - learning_min_sequential + - icesupport + + Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f + +2020-03-04 16:53 +0000 [ab63f0cd0f] Sean Bright + + * enum.c: Make ast_get_txt() actually do something. + + The ast_get_txt() API function (and by extension, the TXTCIDNAME + dialplan function) were broken in + 65b8381550a9f46fdce84de79960073e9d51b05d such that we would never + actually make a DNS TXT query as described. + + This patch restores the documented behavior. + + ASTERISK-19460 #close + Reported by: George Joseph + + Change-Id: I1b19aea711488cb1ecd63843cddce05010e39376 + +2020-03-03 10:57 +0000 [d1a2ff0aaf] lvl + + * res_pjsip_refer: ensure refer progress is still sent after Proceeding() + + ASTERISK-28766 #close + + Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1 + +2020-02-24 12:47 +0000 [06dada3f01] Kevin Harwell + + * codec negotiation: add incoming_call_offer_prefs option + + Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that + specifies the preferred order of codecs after receiving an offer. + + This patch does the following: + + Adds a new enumeration, ast_sip_call_codec_pref, used by the the new + configuration option that's added to the endpoint media structure. + + Adds a new ast_sip_session_caps structure that's set for each session media + object. + + Creates a new file, res_pjsip_session_caps that "implements" the new + structure and option, and is compiled into the res_pjsip_session library. + + ASTERISK-28756 #close + + Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f + +2020-02-20 11:33 +0000 [87fda066ea] Joshua C. Colp + + * res_rtp_asterisk: Improve video performance in certain networks. + + The receive buffer will now grow if we end up flushing the + receive queue after not receiving the expected packet in time. + This is done in hopes that if this is encountered again the + extra buffer size will allow more time to pass and any missing + packets to be received. + + The send buffer will now grow if we are asked for packets and + can't find them. This is done in hopes that the packets are + from the past and have simply been expired. If so then in + the future with the extra buffer space the packets should be + available. + + Sequence number cycling has been handled so that the + correct sequence number is calculated and used in + various places, including for sorting packets and + for determining if a packet is old or not. + + NACK sending is now more aggressive. If a substantial number + of missing sequence numbers are added a NACK will be sent + immediately. Afterwards once the receive buffer reaches 25% + a single NACK is sent. If the buffer continues to grow and + reaches 50% or greater a NACK will be sent for each received + future packet to aggressively ask the remote endpoint to + retransmit. + + ASTERISK-28764 + + Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41 + +2020-02-28 12:55 +0000 [a715cf5aaa] Kevin Harwell + + * message & stasis/messaging: make text message variables work in ARI + + When a text message was received any associated variable was not written to + the ARI TextMessageReceived event. This occurred because Asterisk only wrote + out "send" variables. However, even those "send" variables would fail ARI + validation due to a TextMessageVariable formatting bug. + + Since it seems the TextMessageReceived event has never been able to include + actual variables it was decided to remove the TextMessageVariable object type + from ARI, and simply return a JSON object of key/value pairs for variables. + This aligns more with how the ARI sendMessage handles variables, and other + places in ARI. + + ASTERISK-28755 #close + + Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f + +2020-01-12 05:37 +0000 [b7fbb9c41f] Sebastian Kemper + + * check_expr2: fix cross-compile/hardening issues + + When building check_expr2 with ASLR PIE hardening enabled the linker + fails. This is resolved by adding the regular compiler flags when + building the object files from ast_expr2f.c and ast_expr2.c. + + Note: The STANDALONE define is removed because it is already defined in + _ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives + '--enable-dev-mode'. + + Also, a Makefile variable "CROSS_COMPILING" is added so that the + build system doesn't try to run check_expr2 when cross-compiling, + because that will fail the build as will. + + ASTERISK-28685 #close + + Signed-off-by: Sebastian Kemper + Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915 + +2020-02-24 09:00 +0000 [77c9ba8e63] Torrey Searle + + * res/res_pjsip_sdp_rtp: Fix MOH transitions + + Update the state of remote_hold immediately on receipt of remote + SDP so that the information is available when building the SDP + answer + + ASTERISK-28754 #close + + Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f + +2020-02-25 03:51 +0000 [680e6b9774] Walter Doekes + + * app_queue: Refactor odd placement of if's around say_position + + Change-Id: Icba97905e331812f129e5966e91a59b104c7a748 + +2020-02-24 12:44 +0000 [1e1651b4f4] Kevin Harwell + + * format_cap: make function parameters 'const' + + There were a couple places where the format cap function parameter was not + 'const' when it should have been. This patch makes them 'const'. + + Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e + +2020-02-24 08:39 +0000 [0b5c6fddf1] Walter Doekes + + * say: Remove unused "plural" option from main/say + + There are exceptions for plural objects, but they are detected using the + supplied NUMBER, not using an extra option. + + Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113 + +2020-02-20 06:52 +0000 [5cd7230f3c] Jaco Kroon + + * addons/res_config_mysql: silense warnings about printf format errors. + + Warnings without this: + + res_config_mysql.c: In function 'update2_mysql': + res_config_mysql.c:741:15: warning: format '%llu' expects argument of type + 'long long unsigned int', but argument 6 has type 'my_ulonglong' + {aka 'long unsigned int'} [-Wformat=] + ast_debug(1, "MySQL RealTime: Updated %llu rows on table: %s\n", + numrows, tablename); + + (reformatted for readability within line-wrap) + + Change-Id: I2af4d419a37c1a7eeee750cf9ae4a9a2b3a37fd3 + +2020-02-18 07:10 +0000 [d6712790cd] Joshua C. Colp + + * pjsip: Update ACLs on named ACL changes. + + This change extends the Sorcery API to allow a wizard to be + told to explicitly reload objects or a specific object type + even if the wizard believes that nothing has changed. + + This has been leveraged by res_pjsip and res_pjsip_acl to + reload endpoints and PJSIP ACLs when a named ACL changes. + + ASTERISK-28697 + + Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b + +2020-02-19 13:20 +0000 [7f2d56fc8c] Sean Bright + + * tcptls.c: Log more informative OpenSSL errors + + Dump OpenSSL's error stack to the error log when things fail. + + ASTERISK-28750 #close + Reported by: Martin Zeh + + Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a + +2020-02-19 08:38 +0000 [de6919f339] Sean Bright + + * ast_tls_cert: Allow private key size to be set on command line + + The default size in release branches will be 1024 but we'll use 2048 in master. + + ASTERISK~28750 + + Change-Id: I435cea18bdd58824ed2b55259575c7ec7133842a + +2020-02-13 13:39 +0000 [78b01f41ae] George Joseph + + * res_pjsip_outbound_registration: Fix SRV failover on timeout + + In order to retry outbound registrations for some situations, we + need access to the tdata from the original request. For instance, + for 401/407 responses we need it to properly construct the + subsequent request with the authentication. We also need it if + we're iterating over a DNS SRV response record set so we can skip + entries we've already tried. + + We've been getting the tdata from the server response rdata and + transaction but that only works for the failures where there was + actually a response (4XX, 5XX, etc). For timeouts there's no + response and therefore no rdata or transaction from which to get + the tdata. When processing a single A/AAAA record for a server, + this wasn't an issue as we just retried that same server after the + retry timer expired. If we got an SRV record set for the server + though, without the state from the tdata, we just kept trying the + first entry in the set repeatedly instead of skipping to the next + one in the list. + + * Added a "last_tdata" member to the client state structure to keep + track of the sent tdata. + + * Updated registration_client_send() to save the tdata it used into + the client_state. + + * Updated sip_outbound_registration_response_cb() to use the tdata + saved in client_state when we don't get a response from the + server. We still use the tdata from the transaction when we DO + get a response from the server so we can properly handle 4XX + responses where our new request depends on it. + + General note on timeouts: + + Although res_pjsip_outbound_registration skips to the next record + immediately when a timeout occurs during SRV set traversal, it's + pjproject that determines how long to wait before a timeout is + declared. As with other SIP message types, pjproject will continue + trying the same server at an interval specified by "timer_t1" until + "timer_b" expires. Both of those timers are set in the pjsip.conf + "system" section. + + ASTERISK-28746 + + Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06 + +2020-01-04 18:11 +0000 [5a5be92b79] Joshua C. Colp + + * bridging: Add better support for adding/removing streams. + + This change adds support to bridge_softmix to allow the addition + and removal of additional video source streams. When such a change + occurs each participant is renegotiated as needed to reflect the + update. If another video source is added then each participant + gets another source. If a video source is removed then it is + removed from each participant. This functionality allows you to + have both your webcam and screenshare providing video if you + desire, or even more streams. Mapping has been changed to use + the topology index on the source channel as a unique identifier + for outgoing participant streams, this will never change and + provides an easy way to establish the mapping. + + The bridge_simple and bridge_native_rtp modules have also been + updated to renegotiate when the stream topology of a party changes + allowing the same behavior to occur as added to bridge_softmix. + If a screen share is added then the opposite party is renegotiated. + If that screen share is removed then the opposite party is + renegotiated again. + + Some additional fixes are also included in here. Stream state is + now conveyed in SDP so sendonly/recvonly/inactive streams can + be requested. Removed streams now also remove previous state + from themselves so consumers don't get confused. + + ASTERISK-28733 + + Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5 + +2020-01-23 13:17 +0000 [168637cc0c] Ben Ford + + * RTP/ICE: Send on first valid pair. + + When handling ICE negotiations, it's possible that there can be a delay + between STUN binding requests which in turn will cause a delay in ICE + completion, preventing media from flowing. It should be possible to send + media when there is at least one valid pair, preventing this scenario + from occurring. + + A change was added to PJPROJECT that adds an optional callback + (on_valid_pair) that will be called when the first valid pair is found + during ICE negotiation. Asterisk uses this to start the DTLS handshake, + allowing media to flow. It will only be called once, either on the first + valid pair, or when ICE negotiation is complete. + + ASTERISK-28716 + + Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867 + +2020-02-18 08:33 +0000 [8dcdce42a9] Sean Bright + + * app_mixmonitor: Turn on synchronization by default + + The optional synchronization behavior created in + 64906c4c9ba63e18f2c71310fdbf14450dac7b62 is now the default for + MixMonitor. + + * Add a new flag 'n' that allows for this behavior to be turned off + + * Add a notice when the 'S' option is used indicating that it is no + longer necessary + + Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4 + +2020-02-17 08:05 +0000 [ddfb60ac2c] Sean Bright + + * app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used + + When opening a file for writing, Asterisk silently converts filenames + ending with 'wav49' to 'WAV.' We aren't taking that in to account when + setting the MIXMONITOR_FILENAME variable in MixMonitor. + + * If the user wants to write to a wav49 file, make sure that it is + reflected properly in MIXMONITOR_FILENAME. + + * Add a note to the documentation describing this behavior. + + * Add a note in main/file.c indicating that app_mixmonitor needs to be + changed if the logic in build_filename was changed. + + ASTERISK-24798 #close + Reported by: xrobau + + Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c + +2020-02-12 10:05 +0000 [bf4340f0ec] Torrey Searle + + * res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough + + When moh_passthrough is used, asterisk is only generating invites + of type sendonly and sendrecv instead of taking fully into account + the on hold state of the local and remote parties + + ASTERISK-28738 #close + + Change-Id: Iaaad9fbc033cb14803d433b8a4071bc337047761 + +2020-02-15 08:01 +0000 [0f6ee98c3f] Joshua C. Colp + + * stasis: Use format specifier for size_t. + + Change-Id: Ic9b4afcc5398e7f46314419fc3c90433d818e35c + +2020-02-13 15:08 +0000 [3865b3fd6a] Kevin Harwell + + * res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup + + There was a race condition between client initiated DTLS setup, and handling + of server side ice completion that caused the underlying SSL object to get + cleared during DTLS initialization. If this happened Asterisk would be left + in a partial DTLS setup state. RTP packets were sent and received, but were + not being encrypted and decrypted. This resulted in no audio, or static. + + Specifically, this occurred when '__rtp_recvfrom' was processing the handshake + sequence from the client to the server, and then 'ast_rtp_on_ice_complete' + gets called from another thread and clears the SSL object when calling the + 'dtls_perform_setup' function. The timing had to be just right in the sense + that from the external SSL library perspective SSL initialization completed + (rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks + to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks + it is not finished, thus not completing 'dtls_srtp_setup'. + + This patch removes calls to 'dtls_perform_setup', which clears the SSL object, + in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear + the underlying SSL object. If an ice candidate changes a full protocol level + renegotiation occurs. Also, in the case of bundled ICE candidates are reused + when a stream is added. So no real reason to have to clear, and reset in this + instance. + + Also, this patch adds a bit of extra logging to aid in diagnosis of any future + problems. + + ASTERISK-28742 #close + + Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f + +2020-02-11 07:46 +0000 [aeff1f2c53] Sean Bright + + * res_musiconhold: Avoid spurious warning when 'format' is the empty string + + The change to res_config_odbc that allowed empty strings to be + returned to realtime consumers¹ causes a warning to be emitted when + loading MoH classes. So we need to treat an empty 'format' as if it + was not specified to avoid the warning. + + ASTERISK-28735 #close + Reported by: Ross Beer + + [1] https://gerrit.asterisk.org/c/asterisk/+/13722 + + Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5 + +2020-02-10 15:40 +0000 [1e037ebb97] Sean Bright + + * func_odbc: Prevent snprintf() truncation warning + + For reasons that are not clear to me - this only appears for me when + _not_ building in dev-mode. + + Change-Id: Ib45c54daaea8e0d571cb470cab1daaae2edba968 + +2020-02-10 05:04 +0000 [ac155decae] Joshua C. Colp + + * res_pjsip_session: Fix off-nominal session refreshes. + + Given a scenario where session refreshes occur close to + each other while another is finishing it was possible for + the session refreshes to occur out of order. It was + also possible for session refreshes to be delayed for + quite some time if a session refresh did not result in + a topology change. + + For the out of order session refreshes the first session + refresh would be queued due to a transaction in progress. + This transaction would then finish. When finished a + separate task to process the delayed requests queue + would be queued for handling. A second refresh would + be requested internally before this delayed request + queued task was processed. As no transaction was in + progress this session refresh would be immediately + handled before the queued session refresh. + + The code will now check if any delayed requests exist + before allowing a session refresh to immediately occur. + If any exist then the session refresh is queued. + + For the delayed session refreshes if a session refresh + did not result in a topology change the attempt would + be immediately stopped and no other delayed requests would + be processed. + + The code will now go through the entire delayed requests + queue until a delayed request results in a request + actually being sent. + + ASTERISK-28730 + + Change-Id: Ied640280133871f77d3f332be62265e754605088 + +2020-02-07 13:44 +0000 [a72caa041f] George Joseph + + * doc: Fix CHANGES entries to have .txt suffix and update READMEs + + Although the wiki page for the new CHANGES and UPGRADE scheme + states that the files must have the ".txt" suffix, the READMEs + didn't. + + Change-Id: I490306aa2cc24d6f014738e9ebbc78592efe0f05 + (cherry picked from commit 7416703f04f12eb583a3427a3f64d06951c18c6e) + +2020-01-16 09:50 +0000 [9d9bde76a9] Sean Bright + + * pjproject_bundled: Allow brackets in via parameters + + ASTERISK-26955 #close + Reported by: Peter Sokolov + + Change-Id: Ib2803640905a77b65d0cee2d0ed2c7b310d470ac + +2020-02-05 02:26 +0000 [0c02d0a450] Sylvain Afchain + + * install_prereq: Install aptitude non-interactively + + Currently aptitude is installed using interactive mode. This patch + changes this to use the non-interactive mode as it can block + automatic dependencies installation, ex: CI, Docker build. + + ASTERISK-28726 #close + + Change-Id: I271ee00d230513a6f044810351a32d83b2181133 + +2020-02-04 08:18 +0000 [1b53d329ac] Joshua C. Colp + + * res_rtp_asterisk: Don't produce transport-cc if no packets. + + The code assumed that when the transport-cc feedback + function was called at least one packet will have been + received. In practice this isn't always true, so now + we just reschedule the sending and do nothing. + + Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a + +2020-02-03 10:24 +0000 [b76ab5e5c9] George Joseph + + * message.c: Add option to suppress the Message channel AMI and ARI events + + In order to reduce the amount of AMI and ARI events generated, + the global "Message/ast_msg_queue" channel can be set to suppress + it's normal channel housekeeping events such as "Newexten", + "VarSet", etc. This can greatly reduce load on the manager + and ARI applications when the Digium Phone Module for Asterisk + is in use. To enable, set "hide_messaging_ami_events" in + asterisk.conf to "yes" In Asterisk versions <18, the default + is "no" preserving existing behavior. Beginning with + Asterisk 18, the option will default to "yes". + + NOTE: This change does not affect UserEvents or the ARI + TextMessageReceived events. + + * Added the "hide_messaging_ami_events" option to asterisk.conf. + + * Changed message.c to set the AST_CHAN_TP_INTERNAL property on + the "Message/ast_msg_queue" channel if the option is set in + asterisk.conf. This suppresses the reporting of the events. + + Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b + +2020-01-31 06:58 +0000 [43620cbf6c] Walter Doekes + + * chan_sip: Return 503 if we're out of RTP ports + + If you're for some reason out of RTP ports, chan_sip would previously + responde to an INVITE with a 403, which will fail the call. + + Now, it returns a 503, allowing the device/proxy to retry the call on a + different machine. + + ASTERISK-28718 + + Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90 + +2020-01-29 08:57 +0000 [eb9252ea27] Sean Bright + + * res_config_odbc: Preserve empty strings returned by the database + + When res_config_odbc (and perhaps other realtime backends) reads a SQL + NULL from the database, it coalesces the value to the empty string + which prevents it from being returned to the realtime core. + + However, if it instead reads the empty string from the database, it + needs a way to encode that fact without having the value omitted + entirely. It does this by changing the value to a string with a single + space. The realtime code in main/config.c recognizes this special case + and _turns the string back into the empty string_ before passing it to + realtime API consumers. + + For all of this to work, we need to ensure that we actually pass the + single-space-string back to the realtime core, which is currently + failing because we are trimming the value before checking its + content. So instead we now special case the single-space-string case + so that empty values are returned properly. + + ASTERISK-28719 #close + Reported by: EDV O-TON + + Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de + +2020-01-28 13:23 +0000 [31dc904380] Sean Bright + + * res_stasis_playback: Prevent media_index from going out of bounds + + Incrementing stasis_app_playback.media_index directly in our playback + loop means that when we reach the end of our playlist the index into + the vector will be outside of the bounds of the vector. + + Instead use a temporary variable and only assign when we're sure that + we are in bounds. + + ASTERISK-28713 #close + Reported by: Sébastien Duthil + + Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928 + +2020-01-28 09:18 +0000 [a1f0c833ab] Joshua C. Colp + + * res_pjsip_pubsub: Increment persistence data ref when recreating. + + Each subscription needs to have a reference to the persisted data + for it, as well as the main JSON contained within the tree. When + recreating a subscription this did not occur and they both shared + the same reference. + + ASTERISK-28714 + + Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a + +2020-01-27 19:58 +0000 [03d24ca4c1] Sean Bright + + * res_pjsip_messaging: Allow Content-Type to be overridden + + ASTERISK-26082 #close + Reported by: Alex + + Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb + +2020-01-28 02:34 +0000 [113d05e504] Walter Doekes + + * chan_sip: Clarify in sample docs how directmediapermit/-acl should be used + + It said "restrict [...] which peers should be able to pass [audio] + to each other". + + However, these settings are not global (for which you would expect + signaling IPs to be checked). These settings are available per peer + only, and the IPs being checked, are the RTP IPs. + + Change-Id: I2a6c6cd7c2f5f30d1df4844e3e0308a077021660 + +2020-01-27 12:01 +0000 [cce2b0da95] Kevin Harwell + + * stasis/app: don't lock an app before a call to send + + Calling 'app_send' eventually calls the app's message handler. It's possible + for a handler to obtain a lock on another object, and then need/want to lock + the app object. If the caller of 'app_send' locks the app object prior to + calling then there's a potential for a deadlock, if another thread calls + 'app_send' without locking. + + This patch makes it so 'app_send' is not called with the app object locked in + the section of code doing such. + + ASTERISK-28423 #close + + Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27 + +2020-01-27 11:44 +0000 [4206830a52] Kevin Harwell + + * res_stasis: trigger cleanup after update + + The cleanup code in stasis shuts down applications if they are in a deactivated + state, and no longer have explicit subscriptions. When registering an app the + cleanup code was running before calling 'update'. When it should be executed + after 'update' since a call to register may re-activate the app. We don't want + it to shutdown before the 'update' otherwise the app won't be re-activated, + or registered. + + This patch makes it so the cleanup code is executed post 'update'. + + ASTERISK-28679 #close + + Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b + +2020-01-27 08:03 +0000 [b1ca2c5d71] Sean Bright + + * res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly + + We need to wait for the message sending callback to finish to know if + we succeeded or failed. + + ASTERISK-25421 #close + Reported by: Dmitriy Serov + + Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059 + +2020-01-13 04:13 +0000 [711a3fed56] Walter Doekes + + * chan_sip: Always process updated SDP on media source change + + Fixes no-audio issues when the media source is changed and + strictrtp is enabled (default). + + If the peer media source changes, the SDP session version also changes. + If it is lower than the one we had stored, chan_sip would ignore it. + + This changeset keeps track of the remote media origin identifier, + comparing that as well. If it changes, the session version needn't be + higher for us to accept the SDP. + + Common scenario where this would've caused problems: a separate media + gateway that informs the caller about premium rates before handing off + the call to the final destination. + + (An alternative fix would be to set ignoresdpversion=yes on the peer.) + + ASTERISK-28686 + + Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee + +2020-01-23 09:06 +0000 [313189aae2] Sean Bright + + * chan_pjsip: Ignore RTP that we haven't negotiated + + If chan_pjsip receives an RTP packet whose payload differs from the + channel's native format, and asymmetric_rtp_codec is disabled (the + default), Asterisk will switch the channel's native format to match + that of the incoming packet without regard to the negotiated payloads. + + We now check that the received frame is in a format we have negotiated + before switching payloads which results in these packets being dropped + instead of causing the session to terminate. + + ASTERISK-28139 #close + Reported by: Paul Brooks + + Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3 + +2020-01-22 12:56 +0000 [6818c3d1d2] George Joseph + + * cdr.c: Set event time on party b when leaving a parking bridge + + When Alice calls Bob and Bob does a blind transfer to Charlie, + Bob's bridge leave event generates a finalize on both the party_a + and party_b CDRs but while the party_a CDR has the correct end time + set from the event time, party_b's leg did not. This caused that + CDR's end time to be equal to the answered time and resulted in a + billsec of 0. + + * We now pass the bridge leave message event time to + cdr_object_party_b_left_bridge_cb() and set it on that CDR before + calling cdr_object_finalize() on it. + + NOTE: This issue affected transfers using chan_sip most of the + time but also occasionally affected chan_pjsip probably due to + message timing. + + ASTERISK-28677 + Reported by: Maciej Michno + + Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca + +2020-01-22 09:39 +0000 [0dce6f746b] Sean Bright + + * http: Add ability to disable /httpstatus URI + + Add a new configuration option 'enable_status' which allows the + /httpstatus URI handler to be administratively disabled. + + We also no longer unconditionally register the /static and /httpstatus + URI handlers, but instead do it based upon configuration. + + Behavior change: If enable_static was turned off, the URI handler was + still installed but returned a 403 when it was accessed. Because we + now register/unregister the URI handlers as appropriate, if the + /static URI is disabled we will return a 404 instead. + + Additionally: + + * Change 'enablestatic' to 'enable_static' but keep the former for + backwards compatibility. + * Improve some internal variable names + + ASTERISK-28710 #close + + Change-Id: I647510f796473793b1d3ce1beb32659813be69e1 + +2020-01-18 15:54 +0000 [5bd7281442] Andrew Siplas + + * chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout" + + The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX + to match behavior of "no timeout" defined in comment. + + ASTERISK-28702 #close + + Change-Id: I4ea015986e061374385dba247b272f7aac60bf11 + +2020-01-20 11:18 +0000 [c376e9f8a8] Sean Bright + + * res_statsd: Document that res_statsd does nothing on its own + + ASTERISK-24484 #close + Reported by: Dan Jenkins + + Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec + +2020-01-20 13:53 +0000 [dfad69ce7c] Sean Bright + + * translate.c: Fix silk 24kHz truncation in 'core show translation' + + SILK @ 24kHz is not shown in the 'core show translation' output because of an + off-by-one-error. Discovered while looking into ASTERISK~19871. + + ASTERISK-28706 + Reported by: Sean Bright + + Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510 + +2020-01-20 15:26 +0000 [262221f4d9] Sean Bright + + * func_odbc.conf.sample: Add example lookup + + Change-Id: Ia05aab1f579597963d2ea23920d2210cfcb97c84 + +2020-01-16 15:29 +0000 [f09cf4da44] Sean Bright + + * app_voicemail: Remove MessageExists and MESSAGE_EXISTS() + + * The MailboxExists dialplan application was deprecated on 2006-09-26 + in Asterisk 1.6.0 (commit ec83b111831fe865753f5b1c382ab73750685e50) + + * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in + Asterisk 11.0.0 (commit fd64bb66f94f1a7bb47e17319512b14e522353ec) + + Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f + +2020-01-16 13:47 +0000 [5cbf47714a] Sean Bright + + * app_voicemail, say: Fix various leading whitespace problems + + In af90afd90c64c5183c2207d061f9aa15138081b2, Japanese language support + was added to app_voicemail and main/say.c, but the leading whitespace + is not consistent with Asterisk coding guidelines. This patch fixes + that. + + Whitespace only, no functional change. + + ASTERISK~23324 + Reported by: Kevin McCoy + + Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87 + +2020-01-16 07:32 +0000 [50d02d6194] Sean Bright + + * pbx.c: Include filesystem cache in free memory calculation + + ASTERISK-28695 #close + Reported by: Kevin Flyn + + Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8 + +2020-01-16 09:09 +0000 [f309b86e36] Sean Bright + + * chan_sip.c: Stop handling continuation lines after reading headers + + lws2sws() does not stop trying to handle header continuation lines + even after all headers have been found. This is problematic if the + first character of a SIP message body is a space or tab character, so + we update to recognize the end of the message header. + + ASTERISK-28693 #close + Reported by: Frank Matano + + Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df + +2020-01-15 14:29 +0000 [ba8ccb9132] Sean Bright + + * app_voicemail: Prevent crash when saving message with realtime voicemail + + ast_store_realtime() is not NULL tolerant, so we need to initialize + the field values we pass to it to the empty string to avoid a crash. + + ASTERISK-23739 #close + Reported by: Stas Kobzar + + Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c + +2020-01-14 16:20 +0000 [9be89d9913] Sean Bright + + * app_voicemail: Set globals to default values when voicemail.conf missing + + If voicemail.conf exists but is empty, the config parsing process will + default a number of global variables to non-zero values. On the other + hand, if voicemail.conf is missing (arguably semantically equivalent + to an empty file), this process is skipped and the globals are + defaulted to 0. + + Set the globals to the same values they would be set to if a + configuration were present. This allows voicemail configuration to be + done completely by Realtime without the need to create an empty + voicemail.conf file. + + ASTERISK-27622 #close + Reported by: Jim Van Meggelen + + Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409 + +2020-01-13 16:37 +0000 [094e87b0dc] Sean Bright + + * res_realtime: Fix 'realtime update2' argument handling + + The change in 9b99ef50b5d01ee8111d26efa7b926bdfaf3f980 updated the + syntax of the 'realtime update2' CLI command but did not correctly + update the calls to ast_update2_realtime(). + + The issue this addresses was originally opened because we aren't + allowing a SQL NULL to be set as part of the update, but this is a + limitation of the Realtime API and is not a bug. + + Additionally, this patch: + + * Corrects the example in the command documentation to reflect + 'update2' instead of 'update.' + + * Fixes the leading spacing of the command documentation. + + * Checks that the required 'NULL' literal argument is present where we + expect it to be. + + ASTERISK-21794 #close + Reported by: Cédric Bassaget + + Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd + +2019-07-17 19:47 +0000 [163efbd724] Seán C McCord + + * feat: AudioSocket channel, application, and ARI support. + + This commit adds support for + [AudioSocket]( + https://wiki.asterisk.org/wiki/display/AST/AudioSocket), + a very simple bidirectional audio streaming protocol. There are both + channel and application interfaces. + + A description of the protocol can be found on the above referenced + GitHub page. A short talk about the reasons and implementation can be + found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from + CommCon 2019. + + ARI support has also been added via the existing "externalMedia" ARI + functionality. The UUID is specified using the arbitrary "data" field. + + ASTERISK-28484 #close + + Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5 + +2020-01-10 13:30 +0000 [0c2bf1664c] Sean Bright + + * func_curl: Add 'followlocation' option to CURLOPT() + + We allow for 'maxredirs' to be set, but this value is ignored when + followlocation is not enabled which, by default, it is not. + + ASTERISK-17491 #close + Reported by: candrews + + Change-Id: I96a4ab0142f2fb7d2e96ff976f6cf7b2982c761a + +2020-01-11 07:29 +0000 [9522390a69] Sean Bright + + * app_queue: Deprecate the QueueMemberPause.Reason field + + The QueueMemberPause AMI event includes two fields that return the + reason a member was paused. + + * In release branches, deprecate Reason in favor of PausedReason. + * In master, remove the Reason field entirely. + + ASTERISK-28349 #close + Reported by: Niksa Baldun + + Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296 + +2020-01-10 15:13 +0000 [29d867ed67] Sean Bright + + * res_pjsip_endpoint_identifier_ip: Document support for hostnames + + ASTERISK-25429 #close + Reported by: Joshua C. Colp + + Change-Id: I7cdfc6026821636acc2465094b7fcde8471a3824 + +2020-01-10 14:43 +0000 [90af050fa4] Sean Bright + + * res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY + + ASTERISK-27775 #close + Reported by: AvayaXAsterisk + + Change-Id: Iad158e908e34675ad98f74d09c5e73024e50c257 + +2019-12-03 12:27 +0000 [3bc8b36537] Jaco Kroon + + * netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr. + + ast_addressfamily_to_sockaddrize will determine the size that's + required, and ast_sockaddr_from_sockaddr then wraps this new function + and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without + knowing the address family) into the ast_sockaddr structure. + + Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86 + Signed-off-by: Jaco Kroon + +2020-01-09 04:37 +0000 [2f8b20b949] Corey Farrell + + * app_record: Do not hang up if beep audio is missing + + Additionally alter the warning to mention that it was "beep" which could + not be streamed to give admins a better clue about what the warning + means. + + ASTERISK-28682 + + Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e + +2020-01-08 13:54 +0000 [00a7432156] Kevin Harwell + + * app_agent_pool: Update XML docs for AgentLogin + + This patch fixes some wrongly formatted documentation for the AgentLogin + application. A couple of "see also" links should contain only the function + name, and no parameters. + + Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661 + +2020-01-08 12:11 +0000 [d5f3ec92d0] George Joseph + + * CI: Update buildAsterisk.sh to do a "make full" + + If you do a "make all" when building Asterisk the xml documentation + produced will be missing certain AMI events where their + documentation is located not at the top of the c source file but + embedded further down next to the event's manager_event() + registration call. See main/manager_mwi.c for an example. + + "make full" does produce the correct documentation so we're changing + it in the build script. A separate commit/issue will address the + problem with "make all". + + ASTERISK-28507 + Reported by: David Lee + + Change-Id: I4a22635d6eef99eacecc0efb69e28360eebdb86c + +2020-01-06 09:02 +0000 [4e7adbd8f4] Joshua C. Colp + + * res_pjsip_pubsub: Add ability to persist generator state information. + + Some body generators, such as dialog-info+xml, require storing state + information which is then conveyed in the NOTIFY request itself. Up + until now there was no way for such body generators to persist this + information. + + Two new API calls have been added to allow body generators to set and + get persisted data. This data is persisted out alongside the normal + persistence information and allows the body generator to restore + state information or to simply use this for normal storage of state. + State is stored in the form of JSON and it is up to the body + generator to interpret this as needed. + + The dialog-info+xml body generator has been updated to take advantage + of this to persist the version number. + + ASTERISK-27759 + + Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de + +2019-12-24 09:16 +0000 [312abaa1fe] Sean Bright + + * res_pjsip_endpoint_identifier_ip.c: Add port matching support + + Adds source port matching support when IP matching is used: + + [example] + type = identify + match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 + + If the IP matches but the source port does not, we reject and search for + alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), + unless the configured FQDN includes a port number in which case just a host + lookup is performed. + + ASTERISK-28639 #close + Reported by: Mitch Claborn + + Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92 + +2019-12-30 11:04 +0000 [ee7d72eb72] George Joseph + + * sig_pri: Fix deadlock caused by sig_pri_queue_hangup + + The change to add setting hangupsource to sig_pri_queue_hangup() + made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued + deadlocks when a hangup request was received from the core at the + same time a hanguprequest was received from the remote end via the + D channel. + + Although the PRI's channel private structure was being unlocked + before setting the hangupsource, the PRI's own lock was still being + held during the process. If channel actions were also coming from + the core, a deadlock on the PRI could result. This deadlock could + then escalate to the entire DAHDI subsystem via DAHDI's global + interface list lock, especially if someone used the PRI CLI commands. + + Fix: + + * We now unlock the PRI as well as the PRI's channel private + structure before setting the hangupsource, then relock both + afterwards. + + ASTERISK-28605 + Reported by: Dirk Wendland + + Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9 + +2019-12-30 13:13 +0000 [fe3cce816c] Richard Mudgett + + * app_chanisavail.c: Simplify dialplan using ChanIsAvail. + + Dialplan has to be careful about passing an empty device list or empty + positions in the list. As a result, dialplan has to check for these + conditions before using ChanIsAvail. Simplify dialplan by making + ChanIsAvail handle these conditions gracefully. + + * Made tolerate empty positions in the device list. + + * Simplified the code and eliminated some unnecessary indention. + + ASTERISK-28638 + + Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3 + +2020-01-02 14:25 +0000 [1c9ddad4db] George Joseph + + * stasis.c: Use correct topic name in stasis_topic_pool_delete_topic + + When a topic is created for an object, its name is only + : + For example: + bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 + + When a topic is added to a pool, its name has the pool's topic + name prepended. For example: + bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 + + The topic_pool_entry's name however, is only what was passed + in to stasis_topic_pool_get_topic which is + bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 + That's actually correct because the entry is qualified by the + pool that's in. + + When you're ready to delete the entry from the pool, you retrieve + the tropic name from the object but since it now has the pool's + topic name prepended, it won't be found in the pool container. + + Fix: + + * Modified stasis_topic_pool_delete_topic() to skip past the + pool topic's name, if it was prepended to the topic name, + before searching the container for a pool entry. + + ASTERISK-28633 + Reported by: Joeran Vinzens + + Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6 + +2019-12-30 15:05 +0000 [19069f7db7] Richard Mudgett + + * app_bridgeaddchan.c: Make BridgeAdd be more like Bridge + + * Made BridgeAdd not hangup the call if there is a problem. + * Reduced message level from warning to verbose for normal exception + cases. + * Added a loop safety check to BridgeAdd. + * Made BridgeAdd set BRIDGERESULT with the status when dialplan is + resumed. + + Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426 + +2019-12-29 22:38 +0000 [abcb4ab321] Richard Mudgett + + * app_dial.c: Simplify dialplan using Dial. + + Dialplan has to be careful about passing an empty destination list or + empty positions in the list. As a result, dialplan has to check for + these conditions before using Dial. Simplify dialplan by making Dial + handle these conditions gracefully. + + * Made tolerate empty positions in the dialed device list. + + * Reduced some message log levels from notice to verbose. + + ASTERISK-28638 + + Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9 + +2019-12-29 20:41 +0000 [d86a6ac5ce] Richard Mudgett + + * app_page.c: Simplify dialplan using Page. + + Dialplan has to be careful about passing an empty destination list or + empty positions in the list. As a result, dialplan has to check for + these conditions before using Page. Simplify dialplan by making Page + handle these conditions gracefully. + + * Made tolerate empty positions in the paged device list. + + * Reduced some warnings associated with the 's' option to verbose + messages. The warning level for those messages really serves no purpose + as that is why the 's' option exists. + + ASTERISK-28638 + + Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3 + +2019-12-29 18:36 +0000 [0376f2bba9] Richard Mudgett + + * features.c: Make Bridge application tolerate unspecified channel. + + The Bridge application was inconsistent if the channel to bridge with is + not specified. If no parameters are given then a warning is issued and + the current channel is hung up. If options are given but no channel is + specified then a warning is issued and the current channel is not hung up. + + * Made the Bridge application give a verbose message instead of a warning + if the channel to bridge with is not specified and made not hang up the + current channel. As a result dialplan no longer needs to check if a + channel name is passed before calling Bridge and simply needs to check the + BRIDGERESULT channel variable instead. This is something you likely want + your dialplan to do anyway. + + * Fixed up L() option warning message. It is up to the caller to + determine if the channel is hung up because of the warning. Dial() hangs + up the current channel while Bridge() does not. + + Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73 + +2019-12-29 17:48 +0000 [0d1f3d9bf3] Richard Mudgett + + * app_chanspy.c: Reduce log message level from notice to verbose. + + Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a + +2019-12-29 17:31 +0000 [a457947198] Richard Mudgett + + * app_softhangup.c: Reduce unnecessary warning to verbose message. + + Why log a warning for something your dialplan explicitly asked for? + + Change-Id: I167b90daf4c7d75dd4b7ef94849f6cef05aa43a7 + +2020-01-05 10:00 +0000 [b40dd11afe] Sean Bright + + * res_pjsip_config_wizard: Fix change detection for wizard settings + + ast_sorcery_changeset_create() is not commutative and will fail to detect + differences between two variable lists depending on what changed, so switch to + ast_variable_lists_match(). + + ASTERISK-28492 #close + Reported by: Jean-Denis Girard + + Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1 + +2020-01-03 10:20 +0000 [7d94bdde9d] Sean Bright + + * res_agi: Improve GET FULL VARIABLE documentation + + ASTERISK-28673 #close + Reported by: Jonathan Harris + + Change-Id: I591afdec669622bfa19243aabec31b579652c92f + +2019-11-26 13:24 +0000 [87110c1bdf] Sean Bright + + * websocket: Consider pending SSL data when waiting for socket input + + When TLS is in use, checking the readiness of the underlying FD is insufficient + for determining if there is data available to be read. So before polling the + FD, check if there is any buffered data in the TLS layer and use that first. + + ASTERISK-28562 #close + Reported by: Robert Sutton + + Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d + +2019-11-22 08:32 +0000 [034ac357ad] Jean Aunis + + * ARI: Ability to inhibit COLP frames when adding channels to a bridge + + This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' + operation in the Bridges REST API. When set, this flag avoids generating COLP + frames when the specified channels enter the bridge. + + ASTERISK-28629 + + Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc + +2019-12-27 17:29 +0000 [fc99ac8c9a] Sean Bright + + * db: Initialize condition primitive before use + + The db_init() function ultimately calls db_sync() which signals the + condition before it is initialized. + + Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab + +2019-12-18 09:13 +0000 [40b5cf8f52] Sean Bright + + * config.c: Skip UTF-8 BOMs if present when reading config files + + ASTERISK-28667 #close + + Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2 + +2019-11-21 12:48 +0000 [c626ccec12] Kevin Reeves + + * main/file.c: Limit media cache usage to remote files. + + When testing for the existance of a file, the media cache is searched even if + the file has no chance of being in it. This can cause performance issues + as the media cache size increases. + + As a result, calls to applications like Read and Playback using local files + must scan through the media cache before playing. Under load and with a + large cache, this can delay the playback of those files. + + This patch updates the function that checks for the existance of a file to + only consult the media cache database if the requested file is a remote path. + It introduces a new is_remote_path() function in main/file.c. + + ASTERISK-28625 #close + Reported-by: kevin@phoneburner.com + + Change-Id: If91137493732d9034dafa381c081c69274a7dcc9 + +2019-12-17 18:20 +0000 [095c204fe0] snuffy + + * contrib/valgrind: Fix use of frame-level suppression + + Fix use of frame-level wildcard usage in suppression file. + + ASTERISK-27243 #close + Reported-by: Richard Kenner + + Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51 + +2019-12-17 07:38 +0000 [e494d5fd76] Pascal Cadotte Michaud + + * sip_to_pjsip.py: Fix trustrpid typo + + ASTERISK-28664 #close + + Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6 + +2019-11-27 11:34 +0000 [a83625b366] Frederic LE FOLL + + * app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR. + + Temporary channel lifespan is very short and CDR deactivation request + through ast_cdr_set_property() may happen when CDR is not available + yet. Use CDR_PROP() dialplan function instead, it will first wait + for pending CDR insertion requests to be processed. + + ASTERISK-28636 + + Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074 + +2019-12-16 06:35 +0000 [ed394ce5b1] Joshua C. Colp + + * configure: Add check for MySQL client bool and my_bool type usage. + + Instead of trying to use the defined MySQL client version from the + header use a configure check to determine whether the bool or my_bool + type should be used for defining a boolean. + + ASTERISK-28604 + + Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7 + +2019-12-11 18:03 +0000 [89b7144fbd] Joshua C. Colp + + * confbridge: Add support for specifying maximum sample rate. + + ConfBridge has the ability to move between different sample + rates for mixing the conference bridge. Up until now there has + only been the ability to set the conference bridge to mix at + a specific sample rate, or to let it move between sample rates + as necessary. This change adds the ability to configure a + conference bridge with a maximum sample rate so it can move + between sample rates but only up to the configured maximum. + + ASTERISK-28658 + + Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee + +2019-12-16 05:23 +0000 [a603d7d324] Joshua C. Colp + + * res_pjsip_session: Set stream state on created streams for incoming SDP. + + A previous review, 13174, made a change whereby on an incoming offer SDP + the pending topology was initialized to the configured. This caused a problem + for bundle with WebRTC where bundle could reference a stream that did not + actually exist if the configuration had both audio and video but the + offer SDP only contained audio. + + This change undoes that review and instead fixes the original problem it + sought to solve by setting the state of created streams based on the + contents of the offer SDP. This way the stream state is not inactive + until negotiation later completes. + + ASTERISK-28659 + + Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355 + +2019-12-13 13:46 +0000 [b6f5607359] Kevin Harwell + + * res_fax: wrap v21 detected Asterisk initiated negotiation with config option + + A previous patch: + + Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 + + made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 + negotiation request to both endpoints supported T.38 versus the previous + behavior of forwarding negotiation to the "other" channel once a preamble + was detected. + + This had the unfortunate side effect of breaking some setups. Specifically + ones that set the max datagram option on an endpoint configuration (configured + max datagram was not propagated since Asterisk now initiates negotiations). + + This patch adds a configuration option, "negotiate_both", that when enabled + makes it so Asterisk initiates the negotiation requests to both endpoints vs. + the previous behavior of waiting, and forwarding the request. + + The default is disabled keeping with the old behavior. + + ASTERISK-28660 + + Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a + +2019-12-04 02:35 +0000 [32160cb456] Jaco Kroon + + * ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. + + Due to use in res_rtp_asterisk there is a need to be able to apply an + ACL without logging any invalid/denies. It's probably sensible to at + least validate the ACL once directly after load and report invalid ACLs. + + Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 + Signed-off-by: Jaco Kroon + +2019-12-11 10:52 +0000 [bf4dd3d837] Pascal Cadotte Michaud + + * PJSIP_CONTACT: add missing argument documentation + + add missing argument "rtt" and "status" to the documentation + + The change to the dtd file allow an enumlist to contain one or many + configOptionToEnum or enum. + + This is different from the previous patch I submitted when you could have a + configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or + (one or many enums) + + ASTERISK-28626 + + Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d + +2019-12-11 07:01 +0000 [d0b198b330] Joshua Colp + + * Revert "PJSIP_CONTACT: add missing argument documentation" + + This reverts commit 7e3015d77913accad9b1dcd200ceec30e52bf445. + + Reason for revert: Regression in XML validation. + + validity error : Content model of enumlist is not determinist: + (configOptionToEnum | (configOptionToEnum , enum+) | enum+) + + As we are preparing to do releases and this is not critical + I am reverting this for now until resolved. + + Change-Id: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f + +2019-12-04 15:01 +0000 [39c920ac78] George Joseph + + * res_rtp_asterisk: Add frame list cleanups to ast_rtp_read + + In Asterisk 16+, there are a few places in ast_rtp_read where we've + allocated a frame list but return a null frame instead of the list. + In these cases, any frames left in the list won't be freed. In the + vast majority of the cases, the list is empty when we return so + there's nothing to free but there have been leaks reported in the + wild that can be traced back to frames left in the list before + returning. + + The escape paths now all have logic to free frames left in the + list. + + ASTERISK-28609 + Reported by: Ted G + + Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a + +2019-12-04 08:35 +0000 [365d007eb6] Jaco Kroon + + * chan_sip: in case of tcp/tls, be less annoying about tx errors. + + chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to + 213.150.203.60:1492 returned -2: Interrupted system call + + returned -2 implies this wasn't actually an OS error, so errno makes no + sense either. Internal error was already logged higher up, and -2 + generally means that either there isn't a valid connection available, or + the pipe notification failed, and that is already correctly logged. + + ASTERISK-28651 #close + + Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2 + Signed-off-by: Jaco Kroon + +2019-08-25 21:20 +0000 [cbc1136704] George Joseph + + * res_pjsip_nat: Restore original contact for REGISTER responses + + RFC3261 Section 10 "Registrations", specifically paragraph + "10.2.4: Refreshing Bindings", states that a user agent compares + each contact address (in a 200 REGISTER response) to see if it + created the contact. If the Asterisk endpoint has the + rewrite_contact option set however, the contact host and port sent + back in the 200 response will be the rewritten one and not the + one sent by the user agent. This prevents the user agent from + matching its own contact. Some user agents get very upset when + this happens and will not consider the registration successful. + While this is rare, it is acceptable behavior especially if more + than 1 user agent is allowed to register to a single endpoint/aor. + + This commit updates res_pjsip_nat (where rewrite_contact is + implemented) to store the original incoming Contact header in + a new "x-ast-orig-host" URI parameter before rewriting it, and to + restore the original host and port to the Contact headers in the + outgoing response. + + This is only done if the request is a REGISTER and rewrite_contact + is enabled. + + pjsip_message_filter was also updated to ensure that if a request + comes in with any existing x-ast-* URI parameters, we remove them + so they don't conflict. Asterisk will never send a request + with those headers in it but someone might just decide to add them + to a request they craft and send to Asterisk. + + NOTE: If a device changes its contact address and registers again, + it's a NEW registration. If the device didn't unregister the + original registration then all existing behavior based + on aor/remove_existing and aor/max_contacts apply. + + ASTERISK-28502 + Reported-by: Ross Beer + + Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e + +2019-12-04 15:26 +0000 [b1be06df8d] Sean Bright + + * res_pjsip_registrar.c: Prevent potential double free if AOR is not found + + The simple fix here is simply to NULL out username and password after we call + ast_free on them. Unfortunately, I noticed that we weren't checking for + allocation failures for username and password, and adding those checks made + things noisy and cumbersome. + + So instead we partially rollback the recent LGTM patch, and move the alloca + calls into find_aor_name(). + + ASTERISK-28641 #close + Reported by: Ross Beer + + Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2 + +2019-12-04 15:12 +0000 [0183e2bc67] Sean Bright + + * res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases + + We're appropriately sizing the id_domain_alias buffer, but then copying the data + into the id_domain one. We were then using the uninitialized id_domain_alias + buffer we just allocated. + + This is ASTERISK~28641 adjacent, but significant enough to warrant its own + patch. + + Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9 + +2019-12-03 05:58 +0000 [9c9296c635] Jean Aunis + + * chan_sip: voice frames are no longer transmitted after emitting a COLP + + The SIP transaction state was reset when emitting an UPDATE or a re-INVITE + related to a COLP, preventing RTP packets to be emitted. + + ASTERISK-28647 + + Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b + +2019-11-27 12:11 +0000 [7624cbb155] Frederic LE FOLL + + * chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. + + During capabilities selection (joint capabilities of us and peer, + configured capability for this peer, or general configured + capabilities), if sip_new() does not keep framing information, + then directmedia activation will fail for any framing different + from default framing. + + ASTERISK-28637 + + Change-Id: I99257502788653c2816fc991cac7946453082466 + +2019-12-04 03:33 +0000 [0e750cdd10] Walter Doekes + + * app_queue: Fix old confusing comment about when the members are called + + ASTERISK-28644 + + Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b + +2019-12-03 15:42 +0000 [6ee1f1f507] Sean Bright + + * res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled + + We need to copy the endpoint name before we call ao2_cleanup() on it, + otherwise we might try to access memory that has been reclaimed. + + ASTERISK-28445 #close + Reported by: Bernhard Schmidt + + Change-Id: I404b952608aa606e0babd3c4108346721fb726b3 + +2019-11-22 10:39 +0000 [fd823225a6] Thomas Arimont (license 5525) + + * channel.c: Resolve issue with receiving SIP INFO packets for DTMF + + The problem is essentially the same as in ASTERISK~28245. Besides + the direct media scenario we have an additional scenario where a + special client is involved. This device mutes audio by default in + transmit direction (no rtp frames) and activates audio only by a + foot switch. In this situation dtmf input (pin for conferences, + transfer features codes , etc) using SIP INFO mode is not + understood properly especially when SIP INFO messages are sent + quickly. + + This patch ensures that SIP INFO frames are properly queued and + processed in the above scenario. The patch also corrects situations + where successive dtmf events are received quicker than the + signalled event duration (plus minimum gap/pause) allows, i.e. DTMF + events have to be buffered in the ast channel read queue and + emulation has to be processed asynchronously at slower speed. + + Reported by: Thomas Arimont + patches: + trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525) + + Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2 + +2019-12-02 06:48 +0000 [366da90f74] George Joseph + + * CI: Turn off shallow cloning altogether + + Change-Id: I73ed4aef33a92f20080128aafc34e19fd4457196 + +2019-11-25 06:55 +0000 [811ae88da4] Joshua Colp + + * parking: Fall back to parker channel name even if it matches parkee. + + ASTERISK-28631 + + Change-Id: Ia74d084799fbb9bee3403e30d2391aacd46243cc + +2019-11-22 15:31 +0000 [91c3b5b09d] Sean Bright + + * media_cache.c: Various CLI improvements + + * Use ast_cli_completion_add() to improve performance when large number of + cached items are present. + + * Only complete one URI for commands that only accept a single URI. + + * Change command documentation to wrap at 80 characters to improve + readability. + + Change-Id: Iedb0a2c3541e49561bc231dca2dcc0ebd8612902 + +2016-09-30 21:56 +0000 [48161dfc71] Rodrigo Ramírez Norambuena + + * queue_log: Add alembic script for generate db table for queue_log + + Change-Id: I35b928a6251f9da9a1742b2cd14c63a00c3d0f0c + +2019-11-15 11:34 +0000 [330ffa2bce] Salah Ahmed + + * res_pjsip_t38: T.38 error correction mode selection at 200 ok received + + if asterisk offer T38 SDP with none error correction scheme and + the endpoint respond with redundancy EC scheme, asterisk switch + to that mode. Since we configure the endpoint as none EC mode + we should not switch to any other mode except none. + following logic implemented in code. + + 1. If asterisk offer none, and anything except none in answer + will be ignored. + 2. If asterisk offer fec, answer with fec, redundancy and none will + be accepted. + 3. If asterisk offer redundancy, answer with redundancy and none + will be accepted. + + ASTERISK-28621 + + Change-Id: I343c62253ea4c8b7ee17abbfb377a4d484a14b19 + +2019-10-21 14:55 +0000 [4a1cadeadb] Ben Ford + + * chan_sip.c: Prevent address change on unauthenticated SIP request. + + If the name of a peer is known and a SIP request is sent using that + peer's name, the address of the peer will change even if the request + fails the authentication challenge. This means that an endpoint can + be altered and even rendered unusuable, even if it was in a working + state previously. This can only occur when the nat option is set to the + default, or auto_force_rport. + + This change checks the result of authentication first to ensure it is + successful before setting the address and the nat option. + + ASTERISK-28589 #close + + Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df + +2019-10-24 12:41 +0000 [7e3a6e158f] George Joseph + + * manager.c: Prevent the Originate action from running the Originate app + + If an AMI user without the "system" authorization calls the + Originate AMI command with the Originate application, + the second Originate could run the "System" command. + + Action: Originate + Channel: Local/1111 + Application: Originate + Data: Local/2222,app,System,touch /tmp/owned + + If the "system" authorization isn't set, we now block the + Originate app as well as the System, Exec, etc. apps. + + ASTERISK-28580 + Reported by: Eliel Sardañons + + Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa + +2019-11-21 07:24 +0000 [7e3015d779] Pascal Cadotte Michaud + + * PJSIP_CONTACT: add missing argument documentation + + add missing argument "rtt" and "status" to the documentation + + ASTERISK-28626 + Change-Id: I8419e4c8203e411b87d93dc395acdbcf7526dedf + +2019-11-20 12:56 +0000 [d5d41409e2] Kevin Harwell + + * res_pjsip_outbound_registration: add support for SRV failover + + ASTERISK-28624 + + Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561 + +2019-11-19 12:11 +0000 [2a6a2800e7] George Joseph + + * CI: Fix missing script block in jenkinsfiles + + Change-Id: I9f44a3d5085ea7880fad1a3883a4820907e29ea3 + (cherry picked from commit 95213b01d2d5e72e38b40c30fa5d0c8cf4b37b16) + +2019-11-19 11:40 +0000 [4abb54b2e4] George Joseph + + * CI: Fix missing script block in jenkinsfiles + + Change-Id: Ib4b6e4887695f230ea7a5b0c879b29fc5a13be4f + (cherry picked from commit d60f23ecbdb748b188da424c92335152941c7673) + (cherry picked from commit ce8a23fdf966dc6824678f3cb722753db06baa7a) + (cherry picked from commit f0d1ce50afd25a1269e680b90c8bb612bd543565) + +2019-11-19 08:51 +0000 [e8e1314fcb] George Joseph + + * CI: Increase clone depth and do better cleanup + + The original clone depth of 10 was causing the need to rebase + changes whose parent was older than the 10 commits. The clone + depth has been increased to 100. + + Workspace cleanup was only happening for successful builds which + wasn't enough to keep the 8G workspace in-memory drives on the + docker slaves from filling up. Now the workspaces are cleaned up + after every build regardless of success/failure. If you need to + preserve builds temporarily, you can log into Jenkins/Manage + Jenkins/Configure System and change the CLEANUP_WS_* environment + variable for the job type you're troubleshooting to "FALSE". + + Change-Id: I0d7366e87cea714e5dbc9488caf718802fce75ca + +2019-11-19 09:31 +0000 [a5fa0d662e] Sean Bright + + * res_pjsip_registrar: Fix uninitlized variable warning + + Fixes: error: ‘domain_name’ may be used uninitialized in this function + + Found with gcc (Ubuntu 9.2.1-9ubuntu2) 9.2.1 20191008 + + Change-Id: I44413b49ea1205aa25538142161deb73883c79e8 + +2019-11-05 12:16 +0000 [5bda460300] Michael Cargile + + * app_amd: Fixed timeout issue + + ASTERISK_28143 attempted to fix an issue where calls with no audio would never + timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its + calculations. Unfortunately these frames seem to show up at irregular time + intervals. This resulted in app_amd returning prematurely most of the time. + + * Removed AST_FRAME_NULL from the calculations + * Added a check to see how much time has actually passed since app_amd began + + ASTERISK-28608 + + Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42 + +2019-11-07 11:54 +0000 [a68299f508] Frederic LE FOLL + + * chan_dahdi: PRI span status may stay "Down, Active" after a short alarm + + Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and + may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events. + If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection + and no Q.921 reconnection sequence occurs, chan_dahdi will keep + seeing span status "Down" at the end of alarm. + + This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit + unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects + a disconnection of Q.921 layer and this will clear DCHAN_UP if required. + + ASTERISK-28615 + + Change-Id: Ibe27df4971fd4c82cc6850020bce4a8b2692c996 + +2019-11-07 11:05 +0000 [772b59034f] lvl + + * app_senddtmf: Add receive mode to AMI Action PlayDTMF + + ASTERISK-28614 + + Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664 + +2019-11-07 10:56 +0000 [f2d5ed54ea] Alexei Gradinari + + * serializer: set high/low alert levels on whole pool + + The current code sets alert levels starting from index 1. + Need to set on whole pool starting from index 0. + + Change-Id: I5decbb43160954fb9a512f04302637fc666b6f5d + +2019-11-14 04:19 +0000 [02129ad4d0] Joshua Colp + + * res_rtp_asterisk: Always return provided DTLS packet length. + + OpenSSL can not tolerate if the packet sent out does not + match the length that it provided to the sender. This change + lies and says that each time the full packet was sent. If + a problem does occur then a retransmission will occur as + appropriate. + + ASTERISK-28576 + + Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449 + +2019-11-13 14:25 +0000 [bf7c808604] Sean Bright + + * func_env: Prevent FILE() from reading garbage at end-of-file + + If the last line of a file does not have a terminating EOL sequence, we + potentially add garbage to the value returned from the FILE() function. + + There is no overflow potential here as we are reading from a buffer of a + known size, we are just reading too much of it. + + ASTERISK-26481 #close + + Change-Id: I50dd4fcf416fb3c83150040a1a79a59d9eb1ae01 + +2019-11-13 17:24 +0000 [e77cb32583] Kevin Harwell + + * bridge_softmix: clear hold when joining a softmix bridge + + MOH continues to play to a channel if that channel was on hold prior to + entering a softmix bridge. MOH will not stop even if the original "holder" + attempts an unhold. + + For the most part a softmix bridge ignores holds, so a participating channel + shouldn't join while on hold. This patch checks to see if the channel joining + the softmix bridge is currently on hold. If so then it indicates an unhold. + + ASTERISK-28618 + + Change-Id: I66ccd4efc80f5b4c3dd68186b379eb442916392b + +2019-10-23 12:36 +0000 [bdd785d31c] Kevin Harwell + + * various files - fix some alerts raised by lgtm code analysis + + This patch fixes several issues reported by the lgtm code analysis tool: + + https://lgtm.com/projects/g/asterisk/asterisk + + Not all reported issues were addressed in this patch. This patch mostly fixes + confirmed reported errors, potential problematic code points, and a few other + "low hanging" warnings or recommendations found in core supported modules. + These include, but are not limited to the following: + + * innapropriate stack allocation in loops + * buffer overflows + * variable declaration "hiding" another variable declaration + * comparisons results that are always the same + * ambiguously signed bit-field members + * missing header guards + + Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25 + +2019-11-07 11:54 +0000 [d257a0898e] Martin Tomec + + * func_curl.c: Support custom http headers + + When user wants to send json data, the default Content-Type header + is incorect (application/x-www-form-urlencoded). This patch allows + to set any custom headers so the Content-Type header can be + overriden. User can set multiple headers by multiple calls of + curlopt(). This approach is not consistent with other parameters, + but is more readable in dialplan than one call with multiple + headers. + + ASTERISK-28613 + + Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9 + +2019-11-15 04:46 +0000 [807a70b7ae] Joshua Colp + + * parking: Fix case where we can't get the parker. + + ASTERISK-28616 + + Change-Id: Iabe31ae38d01604284fcc5c2438d44e29a32ea4d + +2019-11-06 05:47 +0000 [990a91b44a] George Joseph + + * stasis: Don't hold app_registry and session locks unnecessarily + + resource_events:stasis_app_message_handler() was locking the session, + then attempting to determine if the app had debug enabled which + locked the app_registry container. res_stasis:__stasis_app_register + was locking the app_registry container then calling app_update + which caused app_handler (which locks the session) to run. + The result was a deadlock. + + * Updated resource_events:stasis_app_message_handler() to determine + if debug was set (which locks the app_registry) before obtaining the + session lock. + + * Updated res_stasis:__stasis_app_register to release the app_registry + container lock before calling app_update (which locks the sesison). + + ASTERISK-28423 + Reported by Ross Beer + + Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4 + +2019-11-12 05:00 +0000 [e924c5107c] Joshua Colp + + * parking: Use channel snapshot instead of channel. + + There exists a scenario where a thread can hold a lock on the + channels container while trying to lock a bridge. At the same + time another thread can hold the lock for said bridge while + attempting to retrieve a channel. This causes a deadlock. + + This change fixes this scenario by retrieving a channel snapshot + instead of a channel, as information present in the snapshot + is all that is needed. + + ASTERISK-28616 + + Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2 + +2019-11-12 12:36 +0000 [0e3b397812] Kevin Harwell + + * res_pjsip_session: initialize pending's topology to endpoint's + + Found during some testing, there is a race condition between selecting an + appropriate bridge type for a call versus the applying of media on the callee's + session. In some instances a native bridge type would have been chosen, but + due to the callee's media not yet being established at bridge compatibility + check time the simple bridge type is picked instead. + + When using chan_pjsip this initiates a topology change event. The topologies + are then compared for the two sessions. However, when the topology was created + for the caller its streams are initialized to "inactive". This topology is then + used as a base when creating the callee's topology, and streams. Soon after + the caller's topology's stream(s) get updated based on the sdp (get set to + sendrecv in the failing scenario). + + Now when the topology change event is raised, and the two topologies are + compared, the comparison fails due to a stream state mismatch (sendrecv vs + inactive). And since they differ a reinvite is sent out (to the caller in + this case). + + This patch makes it such that when the caller's topology is initially created + it gets created based on its configured endpoint's media topology. When the + endpoint's topology is created its stream's state(s) are initialized to + sendrecv instead of inactive. Subsequently, now when the callee's topology is + created its topology streams are now initialized to sendrecv. Thus when the + topology change event occurs due to the mentioned scenario the stream states + match for the given sessions, and the reinvite is not sent unless due to some + other valid mismatch. + + Note, this patch only changes one pending media state's creation point. It's + possible other places *could* be changed, however for now it was deemed best + to only alter what's here. + + Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7 + +2019-11-08 09:20 +0000 [8a1f30af04] Corey Farrell + + * core: Improve MALLOC_DEBUG for frames. + + * Pass caller information to frame allocation functions. + * Disable caching as it interfers with MALLOC_DEBUG reporting. + * Stop using ast_calloc_cache. + + Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260 + +2019-10-29 08:23 +0000 [a47cb71bb1] George Joseph + + * Build: Fix compile issues with seldom used modules + + The following modules needed tweaks for API changes. + + addons/cdr_mysql.c + addons/chan_ooh323.c + apps/app_meetme.c + + ASTERISK-28604 + + Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d + +2019-10-25 06:46 +0000 [e73eba85c1] Joshua Colp + + * res_pjsip_outbound_registration: Extend documentation for "max_retries". + + If the "max_retries" option is set to 0 then upon failure no + further attemps are made, so explicitly document the behavior. + + ASTERISK-28602 + + Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf + +2019-10-24 09:15 +0000 [16e668c7dd] Sean Bright + + * res_calendar: Resolve memory leak on calendar destruction + + Calling ne_uri_parse allocates memory that needs to be freed with a + corresponding call to ne_uri_free. + + ASTERISK-28572 #close + + Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61 + +2019-10-24 05:21 +0000 [360936ead5] Joshua Colp + + * res_ari_events: Add module reference when a WebSocket is open. + + This change ensures that the module isn't unloaded when a + WebSocket is open. Previously it was possible to unload the + module manually or during shutdown which could cause a crash + when any active WebSockets were terminated. + + ASTERISK-28585 + + Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c + +2019-10-18 13:47 +0000 [a4222614c4] Sean Bright + + * utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN + + ASTERISK-28590 #close + + Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c + +2019-10-18 06:36 +0000 [d71d0f9489] George Joseph + + * ExternalMedia: Change return object from ExternalMedia to Channel + + When we created the External Media addition to ARI we created an + ExternalMedia object to be returned from the channels/externalMedia + REST endpoint. This object contained the channel object that was + created plus local_address and local_port attributes (which are + also in the Channel variables). At the time, we thought that + creating an ExternalMedia object would give us more flexibility + in the future but as we created the sample speech to text + application, we discovered that it doesn't work so well with ARI + client libraries that a) don't have the ExternalMedia object + defined and/or b) can't promote the embedded channel structure + to a first-class Channel object. + + This change causes the channels/externalMedia REST endpoint to + return a Channel object (like channels/create and channels/originate) + instead of the ExternalMedia object. + + Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9 + +2019-10-18 04:22 +0000 [ddb0091da5] Salah Ahmed + + * Crash during "pjsip show channelstats" execution + + During execution "pjsip show channelstats" cli command by an + external module asterisk crashed. It seems this is a separate + thread running to fetch and print rtp stats. The crash happened on + the ao2_lock method, just before it going to read the rtp stats on + a rtp instance. According to gdb backtrace log, it seems the + session media was already cleaned up at that moment. + + ASTERISK-28578 + + Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f + +2019-10-17 05:50 +0000 [6e907ae5d4] Joshua Colp + + * res_rtp_asterisk: Remove a log message that slipped in. + + This was only supposed to be for testing, so now it can be + removed. + + Change-Id: I3dfc2e776e70b3196aeed5688372ea80c0214b59 + +2019-10-16 16:06 +0000 [0dc7e29dd8] Sean Bright + + * README-SERIOUSLY.bestpractices.md: Speling correetions. + + ASTERISK-28586 #close + + Change-Id: I43dc4e8bd9dc685b17695b215a5360314074734f + +2019-09-26 19:24 +0000 [2d67dbfef5] cmaj + + * app_voicemail.c: Support multiple file formats for forwarded messages. + + If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav" + and are using realtime ODBC backend, only the first format gets stored + in the database. So when you forward a message later on, there is a bug + generating the email, related to the stored format (GSM) being different + than the desired email format (WAV) specified for the user. Sox can + handle this, but Asterisk needs to tell sox exactly what to do. + + ASTERISK-22192 + + Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd + +2019-10-14 06:19 +0000 [a60d2e905c] Joshua Colp + + * test_res_rtp: Enable FIR and REMB nominal tests. + + Now that both FIR and REMB are being sent in compound packets + these tests can be enabled. + + This also extends the REMB nominal test to cover the REMB + contents itself. + + Change-Id: Ibfee526ad780eefcce5dd787f53785382210024a + +2019-10-08 13:40 +0000 [52ade18420] Christoph Moench-Tegeder + + * cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12 + + PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef + (column default values), which has been deprecated since version 8.0. + Since then, the official/correct/supported way to retrieve the column + default value from the catalog is function pg_catalog.pg_get_expr(). + + This change breaks compatibility with pre-8.0 PostgreSQL servers, + but has reached end-of-support more than a decade ago. + cdr_pgsql and res_config_pgsql still have support for pre-7.3 + servers, but cleaning that up is perhaps a topic for a major release, + not this bugfix. + + ASTERISK-28571 + + Change-Id: I834cb3addf1937e19e87ede140bdd16cea531ebe + +2019-10-10 15:30 +0000 [5dae803eea] Kevin Harwell + + * res_pjsip_mwi: potential double unref, and potential unwanted double link + + When creating an unsolicited MWI aggregate subscription it was possible for + the subscription object to be double unref'ed. This patch removes the explicit + unref as it is not needed since the RAII_VAR will handle it at function end. + + Less concerning there was also a bug that could potentially allow the aggregate + subscription object to be added to the unsolicited container twice. This patch + ensures it is added only once. + + ASTERISK-28575 + + Change-Id: I9ccfdb5ea788bc0c3618db183aae235e53c12763 + +2019-10-09 16:00 +0000 [b27a5183da] Chris Savinovich + + * test_taskprocessor.c: Fix test failure on Ubuntu + + Fixes a failure in /main/taskprocesor unit test, only occurring in Ubuntu. + Newer versions of GCC require variable initialization. + + Change-Id: I2994d8aab9307a8c2c7330584f287a27144a580c + +2019-10-09 09:32 +0000 [5d9f9f4871] George Joseph + + * pjproject_bundled: Replace earlier reverts with official fixes. + + Issues in pjproject 2.9 caused us to revert some of their changes + as a work around. This introduced another issue where pjproject + wouldn't build with older gcc versions such as that found on + CentOS 6. This commit replaces the reverts with the official + fixes for the original issues and allows pjproject to be built + on CentOS 6 again. + + ASTERISK-28574 + Reported-by: Niklas Larsson + + Change-Id: I06f8507bea553d1a01b0b8874197d35b9d47ec4c + +2019-10-09 15:17 +0000 [bf6f27388d] Joshua Colp (license 5000) + + * pbx: deadlock when outgoing dialed channel hangs up too quickly + + Here's the basic scenario that occurred when executing an AMI fast originate + while at the same time something else locks the channels container, and also + wants a lock on the dialed channel: + + 1. pbx_outgoing_attempt obtains a lock on a dialed channel + 2. concurrently another thread obtains a lock on the channels container, and + subsequently requests a lock on the dialed channel. It waits on #1. For + instance, "core show channel + + * Revert "app_voicemail: Cleanup stale lock files on module load" + + This reverts commit fd2e8d0da7ba539470ed73d463d8bc641f7843af. + + Reason for revert: Problematic for users who store their voicemail + on network storage devices, or share voicemail storage between + multiple Asterisk instances. + + ASTERISK-28567 #close + + Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41 + +2019-10-01 06:29 +0000 [c03f50c1c8] lvl + + * chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel + + ASTERISK-28086 #close + + Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea + +2019-10-02 12:56 +0000 [12dbeb69b0] Kevin Harwell + + * res_pjsip_mwi: use an ao2_global object for mwi containers + + On shutdown it's possible for the unsolicited mwi container to be freed before + other dependent threads are done using it. This patch ensures this can no + longer happen by wrapping the container in an ao2_global object. The solicited + container was also changed too. + + ASTERISK-28552 + + Change-Id: I8f812286dc19a34916acacd71ce2ec26e1042047 + +2019-10-02 12:55 +0000 [c0efe19cec] Kevin Harwell + + * serializer: move/add asterisk serializer pool functionality + + Serializer pools have previously existed in Asterisk. However, for the most + part the code has been duplicated across modules. This patch abstracts the + code into an 'ast_serializer_pool' object. As well the code is now centralized + in serializer.c/h. + + In addition serializer pools can now optionally be monitored by a shutdown + group. This will prevent the pool from being destroyed until all serializers + have completed. + + Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971 + +2019-10-02 12:56 +0000 [2970a13fb8] Kevin Harwell + + * res_pjsip/res_pjsip_mwi: use centralized serializer pools + + Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they + both implemented their own serializer pool functionality that was pretty much + identical in each of the source files. This patch removes the duplicated code, + and uses the new 'ast_serializer_pool' object instead. + + Additionally res_pjsip_mwi enables a shutdown group on the pool since if the + timing was right the module could be unloaded while taskprocessor threads still + needed to execute, thus causing a crash. + + Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d + +2019-10-04 15:31 +0000 [51850a79ef] Sean Bright + + * cdr_mysql: Don't clean up on unload unless we can unregister from CDRs + + ASTERISK-28566 #close + + Change-Id: I6daa4e5128e9406d04d3aed670c3bae98d38d40c + +2019-10-01 09:01 +0000 [729b286d59] Joshua Colp + + * stasis: Pass bumped topic_all reference to proxy_dtor. + + This avoids use of the global variable and ensures topic_all remains + active until all topics are freed. + + ASTERISK-28553 + patches: + ASTERISK-28553.patch by coreyfarrell (license 5909) + + Change-Id: I9a8cd8977f3c3a6aa00783f8336d2cfb9c2820f1 + +2019-09-19 03:56 +0000 [b43cdc7f1e] Torrey Searle + + * channel/chan_pjsip: add dialplan function for music on hold + + Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows + the on-hold behavior to be controlled on a per-call basis + + ASTERISK-28542 #close + + Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8 + +2019-09-24 14:18 +0000 [068ed2c626] Alexei Gradinari + + * res_pjsip_pubsub: add endpoint to some warning + + There are some warning messages which are not informative without endpoint: + "No registered subscribe handler for event presence.winfo" + "No registered publish handler for event presence" + + This patch adds an endpoint name to these messages. + + Change-Id: Ia2811ec226d8a12659b4f9d4d224b48289650827 + +2019-09-27 09:54 +0000 [377d7bdab6] Sean Bright + + * res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6 + + ASTERISK-28544 #close + + Change-Id: I8e62c444d107674c298f472e3545661de8a80dce + +2015-03-27 17:34 +0000 [ba64d68273] Jonathan Rose + + * basic-pbx: Bring forward queue configuration from 13 + + Original commit: cfbf5fbe918bc34f3d600760fc0b6f13a3a9a0ed + + Change-Id: I34a841d73c429ca8d944481f8dccb756ee231c9c + +2019-09-25 11:01 +0000 [702019fc80] Sean Bright + + * pbx: Prevent Realtime switch crash on invalid priority + + pbx_extension_helper takes two 'context' arguments. One (con) is a + pointer directly to a 'struct ast_context' and the other (context) is + the name of the context. In all cases, one of these arguments is NULL + and the other is non-NULL. + + Functions that are ultimately called by pbx_extension_helper expect that + 'context' will be non-NULL, so we set it unconditionally on entry into + this function. + + ASTERISK-28534 #close + + Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d + +2019-09-24 15:44 +0000 [4c3655ecfd] Ben Ford + + * taskprocessor.c: Added "like" support to 'core show taskprocessors' + + Added "like" support for 'core show taskprocessors'. Now you + can specify a specific set of taskprocessors (or just one) by + adding the keyword "like" to the above command, followed by + your search criteria. + + Change-Id: I021e740201e9ba487204b5451e46feb0e3222464 + +2019-09-18 06:56 +0000 [966488ab52] Sean Bright + + * res_musiconhold: Add new 'playlist' mode + + Allow the list of files to be played to be provided explicitly in the + music class's configuration. The primary driver for this change is to + allow URLs to be used for MoH. + + Change-Id: I9f43b80b43880980b18b2bee26ec09429d0b92fa + +2019-09-24 17:43 +0000 [982a5025b3] Sean Bright + + * res_pjsip_registrar: Validate Contact URI before adding to responses + + If a permanent contact URI associated with an AOR is invalid, we add a + Contact header to REGISTER responses with a NULL URI, causing a crash. + + ASTERISK-28463 #close + + Change-Id: Id2b643e58b975bc560aab1c111e6669d54db9102 + +2019-09-20 09:08 +0000 [f7045cefd9] Corey Farrell + + * stasis_state: Create internal stasis_state_proxy object. + + This improves the way which stasis_state reference counting works. + Since manager->states holds onto the proxy object instead of the real + object this allows stasis_state objects to be freed when appropriate + without use of a special state_remove function. Additionally each + distinct eid associated with the state holds a reference to the state to + prevent early release and potentially allow easier debug of leaks. + + Change-Id: I400e0db4b9afa3d5cb4ac7dad60907897e73f9a9 + +2019-09-24 11:21 +0000 [67ba62f4e6] Kevin Harwell + + * res_pjsip_pubsub: change warning to debug + + The following message: + + "Subscription request from endpoint rejected. Expiration of 0 is invalid" + + Would sometimes spam the log with warnings if Asterisk restarted and a bunch + of clients sent unsubscribes. This patch changes it from a warning to a debug + message. + + Change-Id: I841ec42f65559f3135e037df0e55f89b6447a467 + +2019-09-24 09:40 +0000 [4de1e6d0e6] Ben Ford + + * taskprocessor.c: Add CLI commands to reset taskprocessor stats. + + Added two new CLI commands to reset stats for taskprocessors. You can + reset stats for a single, specific taskprocessor ('core reset + taskprocessor '), or you can reset all taskprocessors + ('core reset taskprocessors'). These commands will reset the counter for + the number of tasks processed as well as the max queue size. + + Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d + +2019-09-19 09:50 +0000 [cc83e76aa5] George Joseph + + * pjproject_bundled: Revert pjproject 2.9 commits causing leaks + + We've found a connection re-use regression in pjproject 2.9 + introduced by commit + "Close #1019: Support for multiple listeners." + https://trac.pjsip.org/repos/changeset/6002 + https://trac.pjsip.org/repos/ticket/1019 + + Normally, multiple SSL requests should reuse the same connection + if one already exists to the remote server. When a transport + error occurs, the next request should establish a new connection + and any following requests should use that same one. With this + patch, when a transport error occurs, every new request creates + a new connection so you can wind up with thousands of open tcp + sockets, possibly exhausting file handles, and increasing memory + usage. + + Reverting pjproject commit 6002 (and related 6021) restores the + expected behavior. + + We also found a memory leak in SSL processing that was introduced by + commit + "Fixed #2204: Add OpenSSL remote certificate chain info" + https://trac.pjsip.org/repos/changeset/6014 + https://trac.pjsip.org/repos/ticket/2204 + + Apparently the remote certificate chain is continually recreated + causing the leak. + + Reverting pjproject commit 6014 (and related 6022) restores the + expected behavior. + + Both of these issues have been acknowledged by Teluu. + + ASTERISK-28521 + + Change-Id: I8ae7233c3ac4ec29a3b991f738e655dabcaba9f1 + +2019-09-22 16:59 +0000 [725e991faf] Corey Farrell + + * core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option. + + Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based + functions would result in the reference not being logged under + REF_DEBUG. This could sometimes cause inaccurate logging if NULL was + accidentally passed to a reference action. Now reference logging is + only disabled by option passed to the allocation method. + + Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54 + +2019-09-23 11:01 +0000 [a4caaef64c] Kevin Harwell + + * res_sorcery_memory_cache: stale item update leak + + When a stale item was being updated the object was being retrieved, but its + reference was not being decremented after the update. This patch makes it so + the object is now appropriately de-referenced. + + ASTERISK-28523 + + Change-Id: I9d8173d3a0416a242f4eba92fa0853279c500ec7 + +2019-09-23 07:09 +0000 [e82f2f6e82] George Joseph + + * astmm.c: Display backtrace with memory show allocations + + You can currently capture backtraces of memory allocations but they + only get displayed when you stop asterisk and the atexit hooks + are enabled. Now, if memory backtrace is on and you issue a + "memory show allocations" CLI command for a specific file, then + a backtrace will show for each allocation that occurred after + you turned "memory backtrace on". The backtrace display is shown + only when a specific file's allocations are displayed to prevent + a massive CLI dump of every file's allocations. + + Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f + +2019-09-22 21:04 +0000 [a4142c8437] Corey Farrell + + * core: Fix ABI mismatch of ao2_global_obj. + + astobj2.c declares DEBUG_THREADS_LOOSE_ABI to avoid overhead of debug + threads tracking information in the internal structures of astobj2. + Unfortunately this means that ao2_global_obj contains the statically + allocated debug threads tracking fields which are used by initialization + and cleanup but main/astobj2.c believed those fields and associated + space did not exist. + + Change-Id: Icef41ad97d88a8c1d1515e034ec8133cab3b1527 + +2019-09-20 08:29 +0000 [ca608d2575] Corey Farrell + + * stasis: refcounter.py can incorrectly report skewed objects. + + It is possible for topic->name to be NULL, this causes the allocation + reference to not be logged. Use the name variable instead which has + been verified to be a non-empty. + + Change-Id: I3d0031d03c8356e4808f00cdf2d5428712575883 + +2019-09-19 17:32 +0000 [3dfbc05c53] Corey Farrell + + * stasis: Fix leaks + + * Release reference returned by cache_remove + * state_alloc unconditionally bumped state_topic even when it was + locally allocated. + + Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7 + +2019-09-19 10:53 +0000 [863fe2225f] Corey Farrell + + * app_voicemail: Fix module unload leak. + + Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a + +2019-09-06 08:18 +0000 [7298a785ad] Joshua Colp + + * func_jitterbuffer: Add audio/video sync support. + + This change adds support to the JITTERBUFFER dialplan function + for audio and video synchronization. When enabled the RTCP SR + report is used to produce an NTP timestamp for both the audio and + video streams. Using this information the video frames are queued + until their NTP timestamp is equal to or behind the NTP timestamp + of the audio. The audio jitterbuffer acts as the leader deciding + when to shrink/grow the jitterbuffer when adaptive is in use. For + both adaptive and fixed the video buffer follows the size of the + audio jitterbuffer. + + ASTERISK-28533 + + Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492 + +2019-08-22 07:44 +0000 [c18983207d] Florian Floimair + + * core: Add H.265/HEVC passthrough support + + This change adds H.265/HEVC as a known codec and creates a cached + "h265" media format for use. + + Note that RFC 7798 section 7.2 also describes additional SDP + parameters. Handling of these is not yet supported. + + ASTERISK-28512 + + Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2 + +2019-09-14 10:05 +0000 [4072e219f7] Guido Falsi + + * chan_dahdi: Fix build with clang/llvm + + On FreeBSD using the clang/llvm compiler build fails to build due + to the switch statement argument being a non integer type expression. + Switch to an if/else if/else construct to sidestep the issue. + + ASTERISK-28536 #close + + Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8 + +2019-09-15 14:35 +0000 [c358da472e] Joshua Colp + + * chan_pjsip: Relock correct channel during "fax" redirect. + + When fax detection occurs on an outbound PJSIP channel the + redirect operation will result in a masquerade occurring and + the underlying channel on the session changing. The code + incorrectly relocked the new channel instead of the old + channel when returning. This resulted in the new channel + being locked indefinitely. The code now always acts on the + expected channel. + + ASTERISK-28538 + + Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3 + +2019-08-28 05:07 +0000 [8979921da9] Boris P. Korzun + + * func_odbc: acf_odbc_read() and cli_odbc_read() unicode support + + Added ast_odbc_ast_str_SQLGetData() considers SQL_DESC_OCTET_LENGTH + column attribute for correct allocating the buffer. + + ASTERISK-28497 #close + + Change-Id: I50e86c8a277996f13d4a4b9b318ece0d60b279bf + +2019-09-03 12:20 +0000 [723b695ce5] Ben Ford + + * res_rtp_asterisk.c: Send RTCP as compound packets. + + According to RFC3550, ALL RTCP packets must be sent in a compond packet + of at least two individual packets, including SR/RR and SDES. REMB, + FIR, and NACK were not following this format, and as a result, would + fail the packet check in ast_rtcp_interpret. This was found from writing + unit tests for RTCP. The browser would accept the way we were + constructing these RTCP packets, but when sending directly from one + Asterisk instance to another, the above mentioned problem would occur. + + Change-Id: Ieb140e9c22568a251a564cd953dd22cd33244605 + +2019-09-11 15:58 +0000 [32ce6e9a06] Michael Goryainov + + * channels: Allow updating variable value + + When modifying an already defined variable in some channel drivers they + add a new variable with the same name to the list, but that value is + never used, only the first one found. + + Introduce ast_variable_list_replace() and use it where appropriate. + + ASTERISK-23756 #close + Patches: + setvar-multiplie.patch submitted by Michael Goryainov + + Change-Id: Ie1897a96c82b8945e752733612ee963686f32839 + +2019-08-27 17:44 +0000 [cf364cd007] sungtae kim + + * res_musiconhold: Added unregister realtime moh class + + This fix allows a realtime moh class to be unregistered from the command + line. This is useful when the contents of a directory referenced by a + realtime moh class have changed. + The realtime moh class is then reloaded on the next request and uses the + new directory contents. + + ASTERISK-17808 + + Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce + +2019-08-28 14:25 +0000 [0e56643d9f] Ben Ford + + * res_rtp: Add unit tests for RTCP stats. + + Added unit tests for RTCP video stats. These tests include NACK, REMB, + FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR + tests are currently disabled due to a bug. We expect to receive a + compound packet, but the code sends this out as a single packet, which + the browser accepts, but makes Asterisk upset. + + While writing these tests, I noticed an issue with NACK as well. Where + it is handling a received NACK request, it was reading in only the first + 8 bits of following packets that were also lost. This has been changed + to the correct value of 16 bits. + + Also made a minor fix to the data buffer unit test. + + Change-Id: I56107c7411003a247589bbb6086d25c54719901b + +2019-09-05 11:09 +0000 [2d0eee5418] Frederic LE FOLL + + * ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf. + + ChanIsAvail() creates a temporary channel with ast_request() to test + resource availability. It should not generate a CDR when it hangs up + this temporary channel. + + This patch disables CDR generation for the temporary channel with + ast_cdr_set_property(). + + ASTERISK-28527 + + Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1 + +2019-09-05 10:52 +0000 [41b67f150e] Frederic LE FOLL + + * chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up + + When the remote ISDN party ends an ISDN call on a PRI link + (DISCONNECT), CHANNEL(hangupsource) information is not available. + + chan_dahdi already contains an ast_set_hangupsource() in + __dahdi_exception() function but it seems that ISDN message processing + does not use this part of code. + + Two other channel modules associate ast_queue_hangup() and + ast_set_hangupsource() functions calls: + - chan_pjsip in chan_pjsip_session_end() function, + - chan_sip in sip_queue_hangup_cause() function. + chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and + set_hangup_source_and_cause(). + + Thus, I propose to add ast_set_hangupsource() beside + ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and + chan_sip already do. + + ASTERISK-28525 + + Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c + +2019-08-05 06:59 +0000 [2ae1a22e0e] George Joseph + + * ARI: External Media + + The Channel resource has a new sub-resource "externalMedia". + This allows an application to create a channel for the sole purpose + of exchanging media with an external server. Once created, this + channel could be placed into a bridge with existing channels to + allow the external server to inject audio into the bridge or + receive audio from the bridge. + See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI + for more information. + + Change-Id: I9618899198880b4c650354581b50c0401b58bc46 + +2019-09-10 07:32 +0000 [5fb9b23105] George Joseph + + * chan_sip: Update links referenced in deprecation notice + + The links in the deprecation notice were the shortened + variety but it makes better sense to show the unshortened + links as they're more descriptive. + + I.E. + wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip + rather than + wiki.asterisk.org/wiki/x/tAHOAQ + + Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9 + +2019-09-08 10:38 +0000 [e4289b9e56] Sean Bright + + * codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary + + ASTERISK-28511 + + Change-Id: If0d58598ce14aad3c786a1c0127b5f7b200b737d + +2019-08-26 07:53 +0000 [1e9714a050] Joshua Colp + + * AST-2019-005 - translate: Don't assume all frames will have a src. + + This change removes the assumption that a frame will always have + a src set on it. This assumption is incorrect. + + Given a scenario where an RTP packet is received with no payload + the resulting audio frame will have no samples. If this frame goes + through a signed linear translation path an interpolated frame can + be created (if generic packet loss concealment is enabled) that has + minimal data on it, including no src. If this frame is given to a + translation path a crash will occur due to the lack of src. + + ASTERISK-28499 + + Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8 + +2019-08-20 15:05 +0000 [18f5f5fc99] Alexei Gradinari (license 5691) + + * AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media + + After receiving a 200 OK with a declined stream in response to a T.38 + initiated re-invite Asterisk would crash when attempting to dereference + a NULL session media object. + + This patch checks to make sure the session media object is not NULL before + attempting to use it. + + ASTERISK-28495 + patches: + ast-2019-004.patch submitted by Alexei Gradinari (license 5691) + + Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572 + +2019-09-04 16:19 +0000 [ed757cc7bb] Chris-Savinovich + + * test_utils.c: Skip test adsi_loaded_test if module not loaded. + + Module res_adsi.so is deprecated, therefore it does not load by default. + Module not loaded causes it to yield a FAIL when tested by tests/test_utils.c. + This fix checks if the corresponding module is loaded at the start of the test, + and if not, it passes the test and exits with a message. + + This fix is applied to all versions where the module is marked deprecated. + + Change-Id: I52be64c8f6af222e15148a856d1f10cb113e1e94 + +2019-08-27 06:10 +0000 [3863ab9af9] Igor Goncharovsky + + * chan_unistim: Fix clang warning: variable sized type not at end of a struct + + On reading information about initial client packet unistim use dirty + implementation of destination ip address retrieval. This fix uses + CMSG_*(..) to get ip address and make clang compile without warning. + + ASTERISK-25592 #close + Reported-by: Alexander Traud + + Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1 + +2019-08-23 17:03 +0000 [172e183b9d] Kevin Harwell + + * res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions + + res_pjsip_mwi allows both solicited and unsolicited MWI subscription types. + While both can be set in the configuration for a given endpoint/aor, only + one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor + is configured to allow both types then the solicited subscription is rejected + when it comes in. However, there is a configuration option to override that + behavior: + + mwi_subscribe_replaces_unsolicited + + When set to "yes" then when a solicited subscription comes in instead of + rejecting it Asterisk is suppose to replace the unsolicited one if it exists. + Prior to this patch there was a bug in Asterisk that allowed the solicted one + to be added, but did not remove the unsolicited. As a matter of fact a new + unsolicited subscription got added everytime a SIP register was received. + Over time this eventually could "flood" a phone with SIP notifies. + + This patch fixes that behavior to now make it work as expected. If configured + to do so a solicited subscription now properly replaces the unsolicited one. + As well when an unsubscribe is received the unsolicited subscription is + restored. Logic was also put in to handle reloads, and any configuration changes + that might result from that. For instance, if a solicited subscription had + previously replaced an unsolicited one, but after reload it was configured to + not allow that then the solicited one needs to be shutdown, and the unsolicited + one added. + + ASTERISK-28488 + + Change-Id: Iec2ec12d9431097e97ed5f37119963aee41af7b1 + +2019-08-27 00:49 +0000 [1d06a1efb3] Igor Goncharovsky + + * chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk + + Current implementation of ast_channel_tech send_digit_begin hook uses + same function for tone playback as key press handler. This cause every + incoming dtmf send back to asterisk. In case of two unistim phones + connected to each other, it'll cause indefinite DTMF loop. Fix add + separate function for dtmf tone phone play. + + Change-Id: I5795db468df552f0c89c7576b6b3858b26c4eab4 + +2019-08-16 06:01 +0000 [649003821e] Igor Goncharovsky + + * chan_unistim: Fix RTP port byte order for big-endian arch + + This patch fixes one-way oudio that users expirienced on + big-endian architechtires. RTP port number bytes was stored + in improper order and phone sent RTP to wrong RTP port. + + Reported-by: Andrey Ionov + Change-Id: I9a9ca7f26e31a67bbbceff12923baa10dfb8a3be + +2019-08-23 15:14 +0000 [b096389660] Sean Bright + + * codec_resample: Upgrade speex_resample to fix up-sampling bug + + ASTERISK-28511 #close + + Change-Id: Idd07bf341e89ac999c7f5701d9b72b8a9cb11e82 + +2019-08-22 13:19 +0000 [3ef52b0b17] Alexei Gradinari + + * Fix misname 'res_external_mwi' to 'res_mwi_external' in comments. + + Change-Id: Ic784be8500e5cb75dcb34bae9f03cfd93b6b34fb + +2019-08-21 13:29 +0000 [19045db392] George Joseph + + * chan_rtp: Accept hostname as well as ip address as destination + + The UnicastRTP channel driver provided by chan_rtp now accepts + ":" as an alternative to ":" + in the destination. The first AAAA (preferred) or A record resolved + will be used as the destination. The lookup is synchronous so beware + of possible dialplan delays if you specify a hostname. + + Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677 + +2019-08-21 12:03 +0000 [9e015713cc] George Joseph + + * dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4 + + The new function takes in a pointer to an ast_sockaddr structure, + a hostname and an optional port and then dispatches parallel + "AAAA" and "A" record queries. If an "AAAA" record is returned, + it's parsed into the ast_sockaddr structure along with the port + if it was supplied. If no "AAAA" record was returned, the + first "A" record returned (if any) is parsed instead. + + This is a synchronous call. If you need asynchronous lookups, + use ast_dns_query_set_resolve_async and roll your own. + + Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95 + +2019-08-21 10:58 +0000 [0844d6b127] Dan Cropp + + * pjproject: Configurable setting for cnonce to include hyphens or not + + NEC SIP Station interface with authenticated registration only supports cnonce + up to 32 characters. In Linux, PJSIP would generate 36 character cnonce + which included hyphens. Teluu developed this patch adding a compile time + setting to default to not include the hyphens. They felt it best to still + generate the UUID and strip the hyphens. + They have indicated it will be part of PJSIP 2.10. + + ASTERISK-28509 + Reported-by: Dan Cropp + + Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470 + +2019-08-20 13:04 +0000 [8da4e28a81] George Joseph + + * res_ari.c: Prefer exact handler match over wildcard + + Given the following request path and 2 handler paths... + Request: /channels/externalMedia + Handler: /channels/{channelId} "wildcard" + Handler: /channels/externalmedia "non-wildcard" + + ...if /channels/externalMedia was registered as a handler after + /channels/{channelId} as shown above, the request would automatically + match the wildcard handler and attempt to parse "externalMedia" into + the channelId variable which isn't what was intended. It'd work + if the non-wildard entry was defined in rest-api/api-docs/channels.json + before the wildcard entry but that makes the json files + order-dependent which isn't a good thing. + + To combat this issue, the search loop saves any wildcard match but + continues looking for exact matches at the same level. If it finds + one, it's used. If it hasn't found an exact match at the end of + the current level, the wildcard is used. Regardless, after + searching the current level, the wildcard is cleared so it won't + accidentally match for a different object or a higher level. + + BTW, it's currently not possible for more than 1 wildcard entry + to be defined for a level. For instance, there couldn't be: + Handler: /channels/{channelId} + Handler: /channels/{channelName} + We wouldn't know which one to match. + + Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925 + +2019-08-09 15:53 +0000 [64906c4c9b] Sean Bright + + * audiohook.c: Substitute silence for unavailable audio frames + + There are 4 scenarios to consider when capturing audio from a channel + with an audiohook: + + 1. There is no rx and no tx audio, so return nothing. + 2. There is rx but no tx audio, so return rx. + 3. There is tx but no rx audio, so return tx. + 4. There is rx and tx audio, so mix them and return. + + The file passed as the primary argument to MixMonitor will be written to + in scenarios 2, 3, and 4. However, if you pass the r() and t() options + to MixMonitor, a frame will only be written to the r() file if there was + rx audio and a frame will only be written to the t() file if there was + tx audio. + + If you subsequently take the r() and t() files and try to mix them, the + sides of the conversation will 'drift' and be non-representative of the + user experience. + + This patch adds a new 'S' option to MixMonitor that injects a frame of + silence on either the r() side or the t() side of the channel so that + when later mixed, there is no such drift. + + Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e + +2019-07-30 12:08 +0000 [c7270dca81] Stas Kobzar + + * res_pjsip: Channel variable SIPFROMDOMAIN + + In chan_sip, there was variable SIPFROMDOMAIN that allows to set + From header URI domain per channel. This patch introduces res_pjsip + variable SIPFROMDOMAIN for backward compatibility with chan_sip. + + ASTERISK-28489 + + Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e + +2019-08-14 14:52 +0000 [15624d9a7a] Alexei Gradinari + + * app_voicemail/IMAP: check mailstream not NULL in leave_voicemail + + The function leave_voicemail checks if expungeonhangup is set, + but does not check if IMAP stream is closed, + so it could call imap function with NULL stream. + This leads to segfault. + + ASTERISK-28505 #close + + Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c + +2019-08-09 05:51 +0000 [e40f248fac] Sean Bright + + * menuselect: Fix curses build on Gentoo Linux + + Because keypad() is exported by libtinfo, it needs to be explicitly + added to the linker options. + + ASTERISK-28487 #close + + Change-Id: I6c2ad5b95f422c263d078b5c0e84c111807dffc6 + +2019-08-08 12:10 +0000 [446bac733d] George Joseph + + * CI: Escape backslashes in printenv/sort/tr + + Change-Id: I52be64c8f6af2bbe15148a856d1f10cb113e1e94 + (cherry picked from commit c6558e09af3ac15b31377de735cc96d8df0275a7) + +2019-08-07 17:54 +0000 [b805e1237d] Kevin Harwell + + * srtp: Fix possible race condition, and add NULL checks + + Somehow it's possible for the srtp session object to be NULL even though the + Asterisk srtp object itself is valid. When this happened it would cause a + crash down in the srtp code when attempting to protect or unprotect data. + + After looking at the code there is at least one spot that makes this situation + possible. If Asterisk fails to unprotect the data, and after several retries + it still can't then the srtp->session gets freed, and set to NULL while still + leaving the Asterisk srtp object around. However, according to the original + issue reporter this does not appear to be their situation since they found + no errors logged stating the above happened (which Asterisk does for that + situation). + + An issue was found however, where a possible race condition could occur between + the pjsip incoming negotiation, and the receiving of RTP packets. Both places + could attempt to create/setup srtp for the same rtp instance at the same time. + This potentially could be the cause of the problem as well. + + Given the above this patch adds locking around srtp setup for a given rtp, or + rtcp instance. NULL checks for the session have also been added within the + protect and unprotect functions as a precaution. These checks should at least + stop Asterisk from crashing if it gets in this situation again. + + This patch also fixes one other issue noticed during investigation. When doing + a replace the old object was freed before creating the replacement. If the new + replacement object failed to create then the rtp/rtcp instance would now point + to freed srtp data which could potentially cause a crash as well when the next + attempt to reference it was made. This is now fixed so the old srtp object is + kept upon replacement failure. + + Lastly, more logging has been added to help diagnose future issues. + + ASTERISK-28472 + + Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc + +2019-08-08 07:12 +0000 [be6130607d] George Joseph + + * CI: Add "throttle" label and "skip_gate" capability + + To make throttling by label fully active, the "throttle" option + has to be specified with a specific label. + + You can now specify "skip_gate" in the Gerrit comments when you + do a +2 code review to tell Jenkins not to actually run the + gate. You'd do this if you plan to manually merge the change. + + Also updated the "printenv" debug output to better sort multi-line + comments. + + Change-Id: I4c0b1085acec4805f2ca207eebac50aad81f27e2 + +2019-08-05 07:23 +0000 [261646c1c4] Joshua Colp + + * cdr / cel: Use event time at event creation instead of processing. + + When updating times on CDR or CEL records using the time at which + it is done can result in times being incorrect if the system is + heavily loaded and stasis message processing is delayed. + + This change instead makes it so CDR and CEL use the time at which + the stasis messages that drive the systems are created. This allows + them to be backed up while still producing correct records. + + ASTERISK-28498 + + Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a + +2019-08-06 10:40 +0000 [c01dd2a41a] George Joseph + + * CI: Make node labels job-specific + + Originally, the eligible nodes for a job were labelled only by + "swdev-docker". So basically any node could run any job. We had + found that allowing a node to run more than 1 gate at a time was + problematic so we limited the nodes to processing 1 job at a time. + With the creation of the Asterisk 17 branches however, we now have + so many active branches that getting checks and gates through in + a timely manner is problematic when a node can run only 1 job + at a time. + + Now the nodes are also labelled by the job type they can run. + For instance: "asterisk-check", "asterisk-gate", etc. With the + "Throttle Concurrent Builds" plugin, we can now allow a node to + run more than 1 job BUT throttle by job type. For instance: + Allow 2 jobs but only 1 asterisk-gate at a time. + Now a node can run 2 checks or 1 check and 1 gate or 1 gate but + not 2 gates at a time. + + Change-Id: I2032bf6afbcec5c341d9b852214c0c812d3d6db5 + +2019-08-06 08:20 +0000 [9d07d5a6d6] Sean Bright + + * app_voicemail: Remove extra menuselect build options + + You now select voicemail backends like normal dialplan applications, so + there is no longer a need for their own menuselect category. + + Reported by snuff-work in #asterisk-dev + + Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005 + +2019-08-01 16:22 +0000 [3656c42cb0] Kevin Harwell + + * various modules: json integer overflow + + There were still a few places in the code that could overflow when "packing" + a json object with a value outside the base type integer's range. For instance: + + unsigned int value = INT_MAX + 1 + ast_json_pack("{s: i}", value); + + would result in a negative number being "packed". In those situations this patch + alters those values to a ast_json_int_t, which widens the value up to a long or + long long. + + ASTERISK-28480 + + Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1 + +2019-07-29 10:15 +0000 [1f8ae708a0] Sean Bright + + * res_musiconhold: Use a vector instead of custom array allocation + + Change-Id: Ic476a56608b1820ca93dcf68d10cd76fc0b94141 + +2019-08-01 05:07 +0000 [86452c9fa4] Joshua Colp + + * res_pjsip: Fix multiple of the same contact in "pjsip show contacts". + + The code for gathering contacts could result in the same contact + being retrieved and added to the list multiple times. The container + which stores the contacts to display will now only allow a contact + to be added to it once instead of multiple times. + + ASTERISK-28228 + + Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df + +2019-07-17 07:35 +0000 [084901d548] Torrey Searle + + * main/udptl.c: correctly handle udptl sequence wrap around + + incorrect handling of UDPTL squence number wrap arounds causes + loss of packets every time the wrap around occurs + + ASTERISK-28483 #close + + Change-Id: I33caeb2bf13c574a1ebb81714b58907091d64234 + +2019-07-24 15:12 +0000 [5f66fb5139] Sean Bright + + * manager: Send fewer packets + + The functions that build manager message headers do so in a way that + results in a single messages being split across multiple packets. While + this doesn't matter to the remote end, it makes network captures noisier + and harder to follow, and also means additional system calls. + + With this patch, we build up more of the message content into the TLS + buffer before flushing to the network. This change is completely + internal to the manager code and does not affect any of the existing + API's consumers. + + Change-Id: I50128b0769060ca5272dbbb5e60242d131eaddf9 + +2019-07-29 11:38 +0000 [5e6e1175d5] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 17.0.0 +2019-07-26 13:03 +0000 [8d10028b98] George Joseph + + * Update master for Asterisk 18 + + Change-Id: I8b8ed97001446fab0c14d7c89391ee572fb29dd6 + +2019-07-29 10:04 +0000 [7ce9ee7f2e] Sean Bright + + * res_musiconhold: Use ast_pipe_nonblock() wrapper + + Change-Id: Ib0a4b41e5ececbe633079e2d8c2b66c031d2d1f2 + +2019-07-29 08:31 +0000 [8e44d823c1] George Joseph + + * loader.c: Fix possible SEGV when a module fails to register + + When a module fails to register itself (usually a coding error + in the module), dlerror() can return NULL. We weren't checking + for that in load_dlopen() before trying to strdup the error message + so a SEGV was thrown. dlerror() is now surrounded with an S_OR + so we don't SEGV. + + Change-Id: Ie0fb9316f08a321434f3f85aecf3c7d2ede8b956 + +2019-08-28 15:58 +0000 Asterisk Development Team + + * asterisk 17.0.0-rc1 Released. + +2019-08-22 13:19 +0000 [c961d3d9ad] Alexei Gradinari + + * Fix misname 'res_external_mwi' to 'res_mwi_external' in comments. + + Change-Id: Ic784be8500e5cb75dcb34bae9f03cfd93b6b34fb + +2019-08-21 10:58 +0000 [64a2eeef89] Dan Cropp + + * pjproject: Configurable setting for cnonce to include hyphens or not + + NEC SIP Station interface with authenticated registration only supports cnonce + up to 32 characters. In Linux, PJSIP would generate 36 character cnonce + which included hyphens. Teluu developed this patch adding a compile time + setting to default to not include the hyphens. They felt it best to still + generate the UUID and strip the hyphens. + They have indicated it will be part of PJSIP 2.10. + + ASTERISK-28509 + Reported-by: Dan Cropp + + Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470 + +2019-08-20 13:04 +0000 [fe6551f69b] George Joseph + + * res_ari.c: Prefer exact handler match over wildcard + + Given the following request path and 2 handler paths... + Request: /channels/externalMedia + Handler: /channels/{channelId} "wildcard" + Handler: /channels/externalmedia "non-wildcard" + + ...if /channels/externalMedia was registered as a handler after + /channels/{channelId} as shown above, the request would automatically + match the wildcard handler and attempt to parse "externalMedia" into + the channelId variable which isn't what was intended. It'd work + if the non-wildard entry was defined in rest-api/api-docs/channels.json + before the wildcard entry but that makes the json files + order-dependent which isn't a good thing. + + To combat this issue, the search loop saves any wildcard match but + continues looking for exact matches at the same level. If it finds + one, it's used. If it hasn't found an exact match at the end of + the current level, the wildcard is used. Regardless, after + searching the current level, the wildcard is cleared so it won't + accidentally match for a different object or a higher level. + + BTW, it's currently not possible for more than 1 wildcard entry + to be defined for a level. For instance, there couldn't be: + Handler: /channels/{channelId} + Handler: /channels/{channelName} + We wouldn't know which one to match. + + Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925 + +2019-08-14 14:52 +0000 [7591e0f3a4] Alexei Gradinari + + * app_voicemail/IMAP: check mailstream not NULL in leave_voicemail + + The function leave_voicemail checks if expungeonhangup is set, + but does not check if IMAP stream is closed, + so it could call imap function with NULL stream. + This leads to segfault. + + ASTERISK-28505 #close + + Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c + +2019-08-09 05:51 +0000 [fa7883c492] Sean Bright + + * menuselect: Fix curses build on Gentoo Linux + + Because keypad() is exported by libtinfo, it needs to be explicitly + added to the linker options. + + ASTERISK-28487 #close + + Change-Id: I6c2ad5b95f422c263d078b5c0e84c111807dffc6 + +2019-08-07 17:54 +0000 [a92f9f595b] Kevin Harwell + + * srtp: Fix possible race condition, and add NULL checks + + Somehow it's possible for the srtp session object to be NULL even though the + Asterisk srtp object itself is valid. When this happened it would cause a + crash down in the srtp code when attempting to protect or unprotect data. + + After looking at the code there is at least one spot that makes this situation + possible. If Asterisk fails to unprotect the data, and after several retries + it still can't then the srtp->session gets freed, and set to NULL while still + leaving the Asterisk srtp object around. However, according to the original + issue reporter this does not appear to be their situation since they found + no errors logged stating the above happened (which Asterisk does for that + situation). + + An issue was found however, where a possible race condition could occur between + the pjsip incoming negotiation, and the receiving of RTP packets. Both places + could attempt to create/setup srtp for the same rtp instance at the same time. + This potentially could be the cause of the problem as well. + + Given the above this patch adds locking around srtp setup for a given rtp, or + rtcp instance. NULL checks for the session have also been added within the + protect and unprotect functions as a precaution. These checks should at least + stop Asterisk from crashing if it gets in this situation again. + + This patch also fixes one other issue noticed during investigation. When doing + a replace the old object was freed before creating the replacement. If the new + replacement object failed to create then the rtp/rtcp instance would now point + to freed srtp data which could potentially cause a crash as well when the next + attempt to reference it was made. This is now fixed so the old srtp object is + kept upon replacement failure. + + Lastly, more logging has been added to help diagnose future issues. + + ASTERISK-28472 + + Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc + +2019-08-08 12:10 +0000 [b083537d84] George Joseph + + * CI: Escape backslashes in printenv/sort/tr + + Change-Id: I52be64c8f6af2bbe15148a856d1f10cb113e1e94 + (cherry picked from commit c6558e09af3ac15b31377de735cc96d8df0275a7) + +2019-08-08 07:12 +0000 [c4b6e3c1af] George Joseph + + * CI: Add "throttle" label and "skip_gate" capability + + To make throttling by label fully active, the "throttle" option + has to be specified with a specific label. + + You can now specify "skip_gate" in the Gerrit comments when you + do a +2 code review to tell Jenkins not to actually run the + gate. You'd do this if you plan to manually merge the change. + + Also updated the "printenv" debug output to better sort multi-line + comments. + + Change-Id: I4c0b1085acec4805f2ca207eebac50aad81f27e2 + +2019-08-05 07:23 +0000 [37a49cc6d3] Joshua Colp + + * cdr / cel: Use event time at event creation instead of processing. + + When updating times on CDR or CEL records using the time at which + it is done can result in times being incorrect if the system is + heavily loaded and stasis message processing is delayed. + + This change instead makes it so CDR and CEL use the time at which + the stasis messages that drive the systems are created. This allows + them to be backed up while still producing correct records. + + ASTERISK-28498 + + Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a + +2019-08-06 10:40 +0000 [6d610a6b56] George Joseph + + * CI: Make node labels job-specific + + Originally, the eligible nodes for a job were labelled only by + "swdev-docker". So basically any node could run any job. We had + found that allowing a node to run more than 1 gate at a time was + problematic so we limited the nodes to processing 1 job at a time. + With the creation of the Asterisk 17 branches however, we now have + so many active branches that getting checks and gates through in + a timely manner is problematic when a node can run only 1 job + at a time. + + Now the nodes are also labelled by the job type they can run. + For instance: "asterisk-check", "asterisk-gate", etc. With the + "Throttle Concurrent Builds" plugin, we can now allow a node to + run more than 1 job BUT throttle by job type. For instance: + Allow 2 jobs but only 1 asterisk-gate at a time. + Now a node can run 2 checks or 1 check and 1 gate or 1 gate but + not 2 gates at a time. + + Change-Id: I2032bf6afbcec5c341d9b852214c0c812d3d6db5 + +2019-08-01 16:22 +0000 [66b607db88] Kevin Harwell + + * various modules: json integer overflow + + There were still a few places in the code that could overflow when "packing" + a json object with a value outside the base type integer's range. For instance: + + unsigned int value = INT_MAX + 1 + ast_json_pack("{s: i}", value); + + would result in a negative number being "packed". In those situations this patch + alters those values to a ast_json_int_t, which widens the value up to a long or + long long. + + ASTERISK-28480 + + Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1 + +2019-08-06 08:20 +0000 [40e3bdc50c] Sean Bright + + * app_voicemail: Remove extra menuselect build options + + You now select voicemail backends like normal dialplan applications, so + there is no longer a need for their own menuselect category. + + Reported by snuff-work in #asterisk-dev + + Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005 + +2019-08-01 05:07 +0000 [02826c20f5] Joshua Colp + + * res_pjsip: Fix multiple of the same contact in "pjsip show contacts". + + The code for gathering contacts could result in the same contact + being retrieved and added to the list multiple times. The container + which stores the contacts to display will now only allow a contact + to be added to it once instead of multiple times. + + ASTERISK-28228 + + Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df + +2019-07-17 07:35 +0000 [6af55244a7] Torrey Searle + + * main/udptl.c: correctly handle udptl sequence wrap around + + incorrect handling of UDPTL squence number wrap arounds causes + loss of packets every time the wrap around occurs + + ASTERISK-28483 #close + + Change-Id: I33caeb2bf13c574a1ebb81714b58907091d64234 + +2019-07-29 11:46 +0000 [8b3fd0f564] Asterisk Development Team + + * Update CHANGES and UPGRADE.txt for 17.0.0 + +2019-07-29 11:10 +0000 [7b3a612d69] George Joseph + + * doc: Add "master-only" flag back to the CHANGES and UPGRADE files + + In order to run the documentation scripts the flags needs to be + added back to the staging files. + + Change-Id: Ia10a153c50c970cfa1e85815208dfaddb3f2ccd4 + +2019-07-29 08:31 +0000 [2938679ff2] George Joseph + + * loader.c: Fix possible SEGV when a module fails to register + + When a module fails to register itself (usually a coding error + in the module), dlerror() can return NULL. We weren't checking + for that in load_dlopen() before trying to strdup the error message + so a SEGV was thrown. dlerror() is now surrounded with an S_OR + so we don't SEGV. + + Change-Id: Ie0fb9316f08a321434f3f85aecf3c7d2ede8b956 + +2019-07-26 13:06 +0000 [80d8dce6af] George Joseph + + * Prepare Asterisk 17 Branch + + Change-Id: Idb79a69646d2511e7bf1573b9b0322cc22ea54e8 + +2019-07-24 15:15 +0000 [03813e51f0] George Joseph + + * CI: Don't enable non-core modules in Certified branches + + We don't support non-core modules for Certified releases but we + were enabling them for CI builds which was causing lots of test + failures. Now we don't. + + Change-Id: I0b3254c08a2479f3d39151690350cce5ce5ad766 + +2019-07-23 12:58 +0000 [2424ecaf66] Sean Bright + + * res_config_sqlite3: Only join threads that we started + + ASTERISK-28477 #close + Reported by: Dennis + + ASTERISK-28478 #close + Reported by: Dennis + + Change-Id: I77347ad46a86dc5b35ed68270cee56acefb4f475 + +2019-05-12 13:29 +0000 [098797628e] Leonid Fainshtein + + * openr2(6/6): Set hangup cause + + Change-Id: I94dc38920e6e77cc73062648f62fdd613d0d1452 + Signed-off-by: Oron Peled + +2019-04-22 14:14 +0000 [f67094503d] Tzafrir Cohen + + * openr2(5/6): added cli command -- mfcr2 destroy link + + Change-Id: I452d6a853bcd8c6e194455b19e5e017713e9c0fe + Signed-off-by: Oron Peled + +2019-04-22 10:27 +0000 [64bf3e3e82] Tzafrir Cohen + + * openr2(4/6): added new cli command -- mfcr2 show links + + * This command show the MFC/R2 links + + Change-Id: I213822e1b7ef9c05bd89a2ba62df8e0856ce9f84 + Signed-off-by: Oron Peled + +2019-04-22 07:27 +0000 [f61adf2cf5] Tzafrir Cohen + + * openr2(3/6): Convert r2links to standard Asterisk AST_LIST* + + Change-Id: Ibcb2401515a58782a1488c0b9efbed201c3f3a17 + Signed-off-by: Oron Peled + +2019-04-22 07:33 +0000 [97d2549bb1] Tzafrir Cohen + + * openr2(2/6): Stop polling channels when DAHDI returns -ENODEV (e.g: plug-out) + + Otherwise, OpenR2 threads go crazy and consume almost all CPU resources + + Change-Id: I10a41f617613fe7399c5bdced5c64a2751173f28 + Signed-off-by: Oron Peled + +2019-04-22 10:02 +0000 [2f0a8e12f9] Tzafrir Cohen + + * openr2(1/6): bugfix in configuration saving + + Details: + - The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2" + - As a result, the memcmp() in dahdi_r2_get_link() always fails + - This cause dahdi_r2_get_link() to create new link for every channel + (instead of a new link for every ~30 channels) + - With the fix, far less links are generated -- so we use far less threads + + Change-Id: I7259dd6272f5e46e8a6c7f5bf3e8c2ec01b8c132 + Signed-off-by: Oron Peled + +2019-07-22 10:43 +0000 [4304c6534a] Walter Doekes + + * contrib/scripts: Make spandspflow2pcap.py Python 2.7+/3.3+ compatible + + Change-Id: Ica182a891743017ff3cda16de3d95335fffd9a91 + +2019-07-19 11:20 +0000 [be8d41bd24] George Joseph + + * CI: Add cleanWs to cleanup steps in jenkinsfiles + + We're at the point where there are enough Jenkins jobs for + Asterisk branches than even cleaned checkouts of Asterisk + will add up to more disk space than is available on the + in-memory workspace mount. Since we archive all relevent + artifacts anyway, there's no need to keep the workspace + around after the job finishes, whether it succeeds or fails. + + Change-Id: I1cd3b73ebb045a987df0f62526d152a510210c39 + +2019-07-19 08:38 +0000 [8b88994b18] George Joseph + + * CI: Add install-headers to the install make targets + + The testsuite actually needs the headers installed to run + it's self_test. + + Change-Id: Ice41d331131b876ad4a9c056085fe6aac34b32b2 + +2019-07-17 08:06 +0000 [3c6f11992b] Walter Doekes + + * sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread + + When fixing ASTERISK~24212, a change was done so a scheduled callback could not + be removed while it was running. The caller of ast_sched_del would have to wait. + + However, when the caller of ast_sched_del is the callback itself (however wrong + this might be), this new check would cause a deadlock: it would wait forever + for itself. + + This changeset introduces an additional check: if ast_sched_del is called + by the callback itself, it is immediately rejected (along with an ERROR log and + a backtrace). Additionally, the AST_SCHED_DEL_UNREF macro is adjusted so the + after-ast_sched_del-refcall function is only run if ast_sched_del returned + success. + + This should fix the following spurious race condition found in chan_sip: + - thread 1: schedule sip_poke_peer_now (using AST_SCHED_REPLACE) + - thread 2: run sip_poke_peer_now + - thread 2: blank out sched-ID (too soon!) + - thread 1: set sched-ID (too late!) + - thread 2: try to delete the currently running sched-ID + + After this fix, an ERROR would be logged, but no deadlocks (in do_monitor) nor + excess calls to sip_unref_peer(peer) (causing double frees of rtp_instances and + other madness) should occur. + + (Thanks Richard Mudgett for reviewing/improving this "scary" change.) + + Note that this change does not fix the observed race condition: unlocked + access to peer->pokeexpire (and potentially other scheduled items in chan_sip), + causing AST_SCHED_DEL_UNREF to look at a changing id. But it will make the + deadlock go away. And in the observed case, it will not have adverse affects + (like memory leaks) because the scheduled item is removed through a different + path. + + ASTERISK-28282 + + Change-Id: Ic26777fa0732725e6ca7010df17af77a012aa856 + +2019-07-16 07:55 +0000 [c781806e26] George Joseph + + * Build: Separate header install/uninstall + + Asterisk headers are no longer installed and uninstalled + automatically when performing a "make install" or a + "make uninstall". To install/uninstall the headers, use + "make install-headers" and "make uninstall-headers". + The headers also continue to be uninstalled when performing a + "make uninstall-all". + + Also corrects an issue where /usr/include/asterisk.h was never + being removed at all. + + Change-Id: Ia7399f3a0203a4825fc4a9f43b9034dae9a2b643 + +2019-07-09 14:42 +0000 [ba25038fd5] Kevin Harwell + + * manager: Log AMI actions + + When manager debugging is turned on, this patch makes it so incoming AMI actions + are now also logged. + + Change-Id: I8047524510e7ac97d99482b2448f8e368f29cd47 + +2019-07-14 13:26 +0000 [2feac1d361] Joshua Colp + + * res_rtp_asterisk: Move where DTLS MTU variable is defined. + + The DTLS MTU variable is not dependent on pjproject and should + not exist in its block. + + Change-Id: I7e97d64dc192f2ac81bfe2b72b8229d321c7d026 + +2019-06-12 13:03 +0000 [3c520147e1] George Joseph + + * res_pjsip_messaging: Check for body in in-dialog message + + We now check that a body exists and it has a length > 0 before + attempting to process it. + + ASTERISK-28447 + Reported-by: Gil Richard + + Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f + +2019-06-28 11:15 +0000 [8438d19b81] Francesco Castellano + + * chan_sip: Handle invalid SDP answer to T.38 re-invite + + The chan_sip module performs a T.38 re-invite using a single media + stream of udptl, and expects the SDP answer to be the same. + + If an SDP answer is received instead that contains an additional + media stream with no joint codec a crash will occur as the code + assumes that at least one joint codec will exist in this + scenario. + + This change removes this assumption. + + ASTERISK-28465 + + Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87 + +2019-06-12 13:49 +0000 [c93c579190] Kevin Harwell + + * app_voicemail: Remove dependency on the stasis cache + + app_voicemail utilized the stasis cache when polling mailboxes for MWI. This + caused a memory leak (items were not being appropriately removed from the + cache), and subsequent slowdown in system processing. This patch removes the + stasis cache dependency, thus alleviating the memory leak. It does this by + utilizing the new MWI API that better manages state lifetime. + + ASTERISK-28443 + ASTERISK-27121 + + Change-Id: Ie89fedaca81ea1fd03d150d9d3a1ef3d53740e46 + +2019-06-12 13:11 +0000 [9637e1dfdc] Kevin Harwell + + * MWI: Update modules that subscribe to MWI to use new API calls + + The MWI core recently got some new API calls that make tracking MWI state + lifetime more reliable. This patch updates those modules that subscribe to + specific MWI topics to use the new API. Specifically, these modules now + subscribe to both MWI topics and MWI state. + + ASTERISK-28442 + + Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd + +2019-06-11 14:12 +0000 [b31ac83900] Kevin Harwell + + * mwi: Update the MWI core to use stasis_state API + + ** Note ** + + This patch is meant to be the minimum needed in order for the MWI core to use + the now underlying stasis_state module. As such it does not completely remove + its reliance on the stasis_cache. Doing so has allowed current consumers to + not have to change, and update those code paths for this patch. When time + allows, subsequent patches can/will be made to those consumers to take advantage + of some of the new MWI API included here. Thus, eventually and ultimately + removing MWI dependency on the stasis_cache. + + ** End Note ** + + This patch makes it so the MWI core now takes advantage of the new stasis_state + API. Consumers of MWI should no longer need to depend upon stasis topic pooling, + and the stasis cache directly. Similar functionality and implementation details + have now been pushed into the stasis_state module. However, all MWI state should + be accessed via the MWI API itself. + + As such a few new methods, and constructs have been added to the MWI core that + facilitate consumer publishing, subscribing, and iterating over MWI state data. + + * ast_mwi_subscriber * + + Created via ast_mwi_add_subscriber, a subscriber subscribes to a given mailbox + in order to receive updates about the given mailbox. Adding a subscriber will + create the underlying topic, and associated state data if those do not already + exist for it. The topic, and last known state data is guaranteed to exist for + the lifetime of the subscriber. + + * ast_mwi_publisher * + + Before publishing to a particular topic a publisher should be created. This can + be achieved by using ast_mwi_add_publisher. Publishing to a mailbox should then + be done using one of the MWI publish functions. This ensures the message is + published to the appropriate topic, and the last known state is maintained. + + * ast_mwi_observer * + + Add an observer in order to watch for particular MWI module related events. For + instance if a submodule needs to know when a subscription is added to any + mailbox an observer can be added to watch for that. + + * other * + + Urgent message count is now part of the published MWI state object. Also state + can be iterated over using defined callbacks. + + ASTERISK-28442 + + Change-Id: I93f935f9090cd5ddff6d4bc80ff90703c05cf776 + +2019-07-08 18:10 +0000 [83c6ebbae8] Kevin Harwell + + * stasis_state: Make unsubscribes NULL tolerant + + Regular stasis unsubscribes can handle NULL subscription objects. This patch + makes it so stasis state unsubscribes handles NULL's as well. + + ASTERISK-28442 + + Change-Id: Ic3648e8df043a85b77cff085e9ff10356028e479 + +2019-07-04 19:46 +0000 [64a908f897] Rodrigo Ramírez Norambuena + + * README.md: Update year + + Change-Id: I746fb94d112c7d797e206bca0fd1e13fcd26bae3 + +2019-07-01 16:57 +0000 [0e669712e2] Chris-Savinovich + + * chan_dahdi.c: crash in chan_dahdi + + Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous + patch introduced a variable of type unassigned long long which is 64-bits. + Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work + with 32-bit systems. + + ASTERISK-28457 + + Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe + +2019-07-01 10:49 +0000 [93936e367d] Kevin Harwell + + * res_pjsip_sdp_rtp: Remove unused variable + + The variable 'endpoint_caps' in function 'set_caps' is not used, so remove. + + ASTERISK-28458 + + Change-Id: Ia8766d05a0738aecb29dd018302c2dafca5cab34 + +2019-06-11 12:30 +0000 [363bafc29e] Kevin Harwell + + * stasis_state: Add new stasis_state module + + This new module describes an API that can be thought of as a combination of + stasis topic pools, and caching. Except, hopefully done in a more efficient + and less memory "leaky" manner. + + The API defines methods, and data structures for managing, and tracking + published message state through stasis. By adding a subscriber or publisher, + consumers can more easily track the lifetime of the contained state. For + instance, when no more publishers and/or subscribers have need of the topic, + and associated state its data is removed from the managed container. + + * stasis_state_manager * + + The manager stores and well, manages state data. Each state is an association + of a unique stasis topic, and the last known published stasis message on that + topic. There is only ever one managed state object per topic. For each topic + all messages are forwarded to an "all" topic also maintained by the manager. + + * stasis_state_subscriber * + + Topic and state can be created, or referenced within the manager by adding a + stasis_state_subscriber. When adding a subscriber if no state currently exists + new managed state is immediately created. If managed state already exists then + a new subscriber is created referencing that state. The managed state is + guaranteed to live throughout the subscriber's lifetime. State is only removed + from the manager when no other entities require it. + + * stasis_state_publisher * + + Topic and state can be created, or referenced within the manager by also adding + a stasis_state_publisher. When adding a publisher if no state currently exists + new managed state is created. If managed state already exists then a new + publisher is created referencing that state. The managed state is guaranteed to + live throughout the publisher's lifetime. State is only removed from the + manager when no other entities require it. + + * stasis_state_observer * + + Some modules may wish to watch for, and react to managed state events. By + registering a state observer, and implementing handlers for the desired + callbacks those modules can do so. + + * other * + + Callbacks also exist that allow consumers to iterate over all, or some of the + managed state. + + ASTERISK-28442 + + Change-Id: I7a4a06685a96e511da9f5bd23f9601642d7bd8e5 + +2019-06-27 13:50 +0000 [6b1f6ea2c4] Chris-Savinovich + + * app_voicemail.c: Build all three variants for app_voicemail at the same time + + Changes made to apps/Makefile to optionally build all three app_voicemail + variations at the same time: 1) file (default), 2) odbc, and 3) imap. + This functionality was requested by users. modules.conf.sample warns the + user to make sure only one voicemail is loaded at a time. + + Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7 + +2019-06-27 15:04 +0000 [c2ffb004aa] George Joseph + + * tcptls.c: Add peer hostname and port to some error messages + + Where possble, hostname and port has been added to error + messages, mostly on the server side. + + ASTERISK-26006 + Reported by: Oleksandr Natalenko + + Change-Id: Iff4f897277bc36ce8c5b493b71d0a4a7b74e62f0 + +2019-06-27 12:46 +0000 [8b3ee7fe61] George Joseph + + * pjproject_bundled: Add peer information to most SSL/TLS errors + + Most SSL/TLS error messages coming from pjproject now have either + the peer address:port or peer hostname, depending on what was + available at the time and code location where the error was + generated. + + ASTERISK-28444 + Reported by: Bernhard Schmidt + + Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e + +2019-04-15 18:26 +0000 [613a335de5] sungtae kim + + * res/ari/resource_channels.c: Added hangup reason code for channels + + Currently, DELETE /ari/channels/ supports only few hangup reasons. + It's good enough for simple use, but when it needs to set the detail reason, + it comes challenges. + Added reason_code query parameter for that. + + ASTERISK-28385 + + Change-Id: I1cf1d991ffd759d0591b347445a55f416ddc3ff2 + +2019-04-02 14:42 +0000 [e52fbae00f] Dan Cropp + + * chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS + + Previously, when a Transfer (REFER) was performed, chan_pjsip would set + the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not + reflect a successful/unsuccessful transfer the way chan_sip did. + Added a callback module to process the refer subscription information. + + Now depends on res_pjsip_pubsub so call transfer progress can be monitored + and reported + + ASTERISK-26968 #close + Reported-by: Dan Cropp + + Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc + +2019-06-24 08:30 +0000 [13e89d372b] George Joseph + + * sig_pri: Address gcc9 issues + + A few more format truncation issues addressed. + + Change-Id: I047f373169caaca0eec4889d3c0e5e10f130017a + +2019-05-21 01:38 +0000 [29bc7cf6b3] Nasir Iqbal + + * app_amd: issue with silence suppression fixed + + Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual + wait time instead of `framelength` is enough to fix timeout / TOOLONG issue. + + ASTERISK-28419 #close + + Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb + +2019-05-29 17:54 +0000 [f414ca069c] Alexei Gradinari + + * res_fax: gateway sends T.38 request to both endpoints if V.21 detected + + According T.38 Gateway 'Use case 3' + https://wiki.asterisk.org/wiki/display/AST/T.38+Gateway + T.38 Gateway should send T.38 negotiation request to called endpoint + if FAX preamble (using V.21 detector) generated by called endpoint. + But it does not, because fax_gateway_detect_v21 constructs T.38 + negotiation request, but forwards it only to other channel, + not to the channel on which FAX preamble is detected. + + Some SIP endpoints could be improperly configured to rely on the other side + to initiate T.38 re-INVITEs. + + With this patch the T.38 Gateway tries to negotiate with both sides + by sending T.38 negotiation request to both endpoints supported T.38. + + Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 + +2019-06-19 11:58 +0000 [0ba52ce3cf] George Joseph + + * CI: New way to determnine libdir + + We were using the presence of /usr/lib64 to determine where + shared libraries should be installed. This only existed on + Redhat based systems and was safe. If it existed, use it, + otherwise use /usr/lib. + + Unfortunately, Ubuntu 19 decided to create a /usr/lib64 BUT + NOT INCLUDE IT IN THE DEFAULT ld.so.conf. So if anything is + installed there, it won't work. + + The new method, just looks for $ID in /etc/os-release and if it's + centos or fedora, uses /usr/lib64 and if ubuntu, uses /usr/lib. + + NOTE: This applies only to the CI scripts. Normal asterisk + build and install is not affected. + + Change-Id: Iad66374b550fd89349bedbbf2b93f8edd195a7c3 + +2019-06-14 15:45 +0000 [e3866cb714] Alexei Gradinari + + * translate.c do not log WARNING on empty audio frame + + There is WARNING "no samples for ..." on each Playtones. + The function ast_playtones_start calls ast_activate_generator, + which calls ast_prod. + The function ast_prod calls ast_write with empty audio frame. + In this case it's spam log. + + Change-Id: Id4ac309489d9ff281bad02abdef341cecdede660 + +2019-06-17 12:11 +0000 [92d4ec2906] George Joseph + + * chan_dahdi: Address gcc9 issues + + Fixed format-truncation issues in chan_dahdi.c and + sig_analog.c. Since they're related to fields provided + by dahdi-tools we can't change the buffer sizes so we're just + checking the return from snprintf and printing an errior if we + overflow. + + Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5 + +2019-06-10 16:58 +0000 [f3e5419d41] George Joseph + + * app_confbridge: Attended transfer event fixup + + When a channel already in a conference bridge is attended transfered + to another extension, or when an existing call is attended + transferred into a conference bridge, we now generate ConfbridgeJoin + and ConfbridgeLeave events for the entering and departing channels. + + Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1 + +2019-06-13 10:11 +0000 [c70d874f7d] Sean Bright + + * pjproject: Update to 2.9 release + + Relies on https://github.com/asterisk/third-party/pull/4 + + Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3 + +2019-06-11 07:26 +0000 [a8e5cf557d] Joshua Colp + + * res_rtp_asterisk: Add support for DTLS packet fragmentation. + + This change adds support for larger TLS certificates by allowing + OpenSSL to fragment the DTLS packets according to the configured + MTU. By default this is set to 1200. + + This is accomplished by implementing our own BIO method that + supports MTU querying. The configured MTU is returned to OpenSSL + which fragments the packet accordingly. When a packet is to be + sent it is done directly out the RTP instance. + + ASTERISK-28018 + + Change-Id: If2d5032019a28ffd48f43e9e93ed71dbdbf39c06 + +2019-05-21 14:12 +0000 [3eaeb3e6c4] Alexei Gradinari + + * app_attended_transfer: new application AttendedTransfer + + AttendedTransfer queues up attended transfer to the given extension. + + This application can be useful with Custom Dynamic Features. + For example to make attended transfer to a predefined number. + + features.conf + ;;; + [applicationmap] + my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default + ;;; + + extensions.conf + ;;; + [globals] + DYNAMIC_FEATURES=my_atxfer + TRANSFER_CONTEXT=my_transfer + + [my_atxfer] + exten => s,1,AttendedTransfer(1234567890) + same => n,Return() + + [my_transfer] + include => default + ;;; + + This application also can be used to completly redefine Attended transfer + feature using dialplan. For example: + + features.conf + ;;; + [featuremap] + atxfer => *7 + + [applicationmap] + custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default + ;;; + + extensions.conf + ;;; + [globals] + DYNAMIC_FEATURES=custom_atxfer + TRANSFER_CONTEXT=my_transfer + + [custom_atxfer] + exten => s,1, + same => n,Playback(pbx-transfer) + same => n,Read(dest,dial,10,i,3,3) + same => n,AttendedTransfer(${dest}) + same => n,Return() + + [my_transfer] + include => default + ;;; + + Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b + +2019-06-06 07:48 +0000 [d2f7b22640] Abhay Gupta + + * chan_pjsip.c: Check for channel and session to not be NULL in hangup + + We have seen some rare case of segmentation fault in hangup function + and we could notice that channel pointer was NULL. Debug log shows + that there is a 200 OK answer and SIP timeout at the same time. It + looks that while the SIP session was being destroyed due to timeout + call hangup due to answer event lead to race condition and channel + is being destroyed from two different places. The check ensures we + check it not to be NULL before freeing it. + + ASTERISK-25371 + + Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778 + +2019-05-21 14:53 +0000 [745cbab501] Alexei Gradinari + + * app_blind_transfer: new application BlindTransfer + + BlindTransfer redirects all channels currently bridged to the + caller channel to the specified destination. + + This application can be useful with Custom Dynamic Features. + For example to make blind transfer to a predefined number. + + features.conf + ;;; + [applicationmap] + my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default + ;;; + + extensions.conf + ;;; + [globals] + DYNAMIC_FEATURES=my_blindxfer + + [my_blindxfer] + exten => s,1,BlindTransfer(1234567890,default) + same => n,Return() + ;;; + + This application also can be used to completly redefine Blind transfer + feature using dialplan. For example: + + features.conf + ;;; + [featuremap] + blindxfer => + + [applicationmap] + custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default + ;;; + + extensions.conf + ;;; + [globals] + DYNAMIC_FEATURES=custom_blindxfer + + [custom_blindxfer] + exten => s,1, + same => n,Playback(pbx-transfer) + same => n,Read(dest,dial,10,i,3,3) + same => n,BlindTransfer(${dest},default) + same => n,Return() + ;;; + + Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a + +2019-01-08 00:14 +0000 [bcaa01b024] Kirsty Tyerman + + * pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi + + ASTERISK-28234 + Reported-by: Kirsty Tyerman + + Change-Id: I5d6e6b52dbe51415046bb3953fd16f5b421bc2e1 + +2019-06-04 12:41 +0000 [e61f2af89d] Chris-Savinovich + + * cdr_pgsql: fix error in connection string + + Fixes an error occurring in function pgsql_reconnect() caused when value of + hostname is blank. Which in turn will cause the connection string to look + like this: "host= port=xx", which creates a sintax error. This fix now checks + if the corresponding values for host, port, dbname, and user are blank. Note + that since this is a reconnect function the database library will replace any + missing value pairs with default ones. + + ASTERISK-28435 + + Change-Id: I0a921f99bbd265768be08cd492f04b30855b8423 + +2019-05-28 15:35 +0000 [1b62781be0] Alexei Gradinari + + * res_fax: fix segfault on inactive "reserved" fax session + + The change #10017 "Handle fax gateway being started more than once" + introdiced a bug which leads to segfault in res_fax_spandsp. + + The res_fax_spandsp module does not support reserving sessions, so + fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE. + + The fax_gateway_start does not create a real fax session if the fax session + is already present and the state is not AST_FAX_STATE_RESERVED. + But the "reserved" session created for res_fax_spandsp has state + AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting. + + Then when fax_gateway_framehook is called and gateway T.38 state is + NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to + segfault, because session tech_pvt is not set, i.e. the tech session + was not initialized/started. + + This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved" + session created for res_fax_spandsp will start. + + This patch also adds extra check and log ERROR if tech_pvt is not set + before call tech->write. + + ASTERISK-27981 #close + + Change-Id: Ife3e65e5f18c902db2ff0538fccf7d28f88fa803 + +2019-05-28 17:15 +0000 [bfd93995d9] Alexei Gradinari + + * res_fax: add channel name to CLI 'fax show session' + + This patch adds a channel name to output of CLI 'fax show session' + and also expands the channel name field up to 30 characters on + CLI 'fax show sessions' + + Change-Id: Id059c43ff41811f5e76712b83fb63b8f246da953 + +2019-05-24 09:01 +0000 [9969c77bc2] Ben Ford + + * build: Fix file format in CHANGES-staging. + + One of the change files doesn't conform to the format that the release + scripts need in order to parse it. + + Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c + +2019-05-23 09:44 +0000 [db535439f2] Guido Falsi + + * chan_dahdi: add missing include. + + After some definitions have been moved to asterisk/mwi.h the files + channels/chan_dahdi.h channels/sig_pri.c are missing this new + include. + + ASTERISK-28427 #close + + Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91 + +2019-05-17 17:45 +0000 [408210bd4c] Alexei Gradinari + + * app_readexten: new option 'p' to stop reading on '#' key + + This patch adds the 'p' option. + The extension entered will be considered complete when a # is entered. + + Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1 + +2019-05-10 09:36 +0000 [0bb38796b7] Matt Jordan + + * res_prometheus: Add metrics for PJSIP outbound registrations + + When monitoring Asterisk instances, it's often useful to know when an + outbound registration fails, as this often maps to the notion of a trunk + and having a trunk fail is usually a "bad thing". As such, this patch + adds monitoring metrics that track the state of PJSIP outbound registrations. + It does this by looking for the Registry events coming across the Stasis + system topic, and publishing those as metrics to Prometheus. Note that + while this may support other outbound registration types (IAX2, SIP, etc.) + those haven't been tested. Your mileage may vary. + + (And why are you still using IAX2 and SIP? It's 2019 folks. Get with the + program.) + + This patch also adds Sorcery observers to handle modifications to the + underlying PJSIP outbound registration objects. This is useful when a + reload is triggered that modifies the properties of an outbound registration, + or when ARI push configuration is used and an object is updated or + deleted. Because we rely on properties of the registration object to + define the metric (label key/value pairs), we delete the relevant metric when + we notice that something has changed and wait for a new Stasis message to + arrive to re-create the metric. + + ASTERISK-28403 + + Change-Id: If01420e38530fc20b6dd4aa15cd281d94cd2b87e + +2019-01-03 10:28 +0000 [a2648b22eb] Matt Jordan + + * res_prometheus: Add CLI commands + + This patch adds a few CLI commands to the res_prometheus module to aid + system administrators setting up and configuring the module. This includes: + + * prometheus show status: Display basic statistics about the Prometheus + module, including its essential configuration, when it was last scraped, + and how long the scrape took. The last two bits of information are useful + when Prometheus isn't generating metrics appropriately, as it will at + least tell you if Asterisk has had its HTTP route hit by the remote + server. + + * prometheus show metrics: Dump the current metrics to the CLI. Useful for + system administrators to see what metrics are currently available without + having to cURL or go to Prometheus itself. + + ASTERISK-28403 + + Change-Id: Ic09813e5e14b901571c5c96ebeae2a02566c5172 + +2019-05-09 09:41 +0000 [066280f0cc] Matt Jordan + + * res_prometheus: Add Asterisk bridge metrics + + This patch adds basic Asterisk bridge statistics to the res_prometheus + module. This includes: + + * asterisk_bridges_count: The current number of bridges active on the + system. + + * asterisk_bridges_channels_count: The number of channels active in a + bridge. + + In all cases, enough information is provided with each bridge metric + to determine a unique instance of Asterisk that provided the data, along + with the technology, subclass, and creator of the bridge. + + ASTERISK-28403 + + Change-Id: Ie27417dd72c5bc7624eb2a7a6a8829d7551788dc + +2019-05-09 09:41 +0000 [ed6cd13b5b] Matt Jordan + + * res_prometheus: Add Asterisk endpoint metrics + + This patch adds basic Asterisk endpoint statistics to the res_prometheus + module. This includes: + + * asterisk_endpoints_state: The current state (unknown, online, offline) + for each defined endpoint. + + * asterisk_endpoints_channels_count: The current number of channels + associated with a given endpoint. + + * asterisk_endpoints_count: The current number of defined endpoints. + + In all cases, enough information is provided with each endpoint metric + to determine a unique instance of Asterisk that provided the data, as well + as the underlying technology and resource definition. + + ASTERISK-28403 + + Change-Id: I46443963330c206a7d12722d08dcaabef672310e + +2019-05-21 11:29 +0000 [3224ac07c9] Morten Tryfoss + + * res_rtp_asterisk: timestamp should be unsigned instead of signed int + + Using timestamp with signed int will cause timestamps exceeding max value + to be negative. + This causes the jitterbuffer to do passthrough of the packet. + + ASTERISK-28421 + + Change-Id: I9dabd0718180f2978856c50f43aac4e52dc3cde9 + +2019-05-02 19:45 +0000 [0760af71ad] Matt Jordan + + * res_prometheus: Add Asterisk channel metrics + + This patch adds basic Asterisk channel statistics to the res_prometheus + module. This includes: + + * asterisk_calls_sum: A running sum of the total number of + processed calls + + * asterisk_calls_count: The current number of calls + + * asterisk_channels_count: The current number of channels + + * asterisk_channels_state: The state of any particular channel + + * asterisk_channels_duration_seconds: How long a channel has existed, + in seconds + + In all cases, enough information is provided with each channel metric + to determine a unique instance of Asterisk that provided the data, as + well as the name, type, unique ID, and - if present - linked ID of each + channel. + + ASTERISK-28403 + + Change-Id: I0db306ec94205d4f58d1e7fbabfe04b185869f59 + +2019-04-29 10:10 +0000 [54f7f7dc20] Matt Jordan + + * pjproject/Makefile: Updates for Darwin compatible builds + + This patch fixes three compatibility issues for Darwin compatible builds: + + (1) Use BSD compatible command line option for sed + + For some versions of BSD sed, the -r command line option is unknown. + Both GNU and BSD sed support the -E command line option for enabling + extended regular expressions; as such, this patch replaces the -r + option with -E. + + (2) Look for '_' in pjproject generated symbols + + In Darwin comaptible systems, the symbols generated for pjproject may be + prefixed with an '_'. When exporting these to a symbol file, the invocation + to sed has to optionally look for a prefix of said '_' character. + + (3) Use -all_load/-noall_load when linking + + The flags -whole-archive/-no-whole-archive are not supported by the + linker, and must instead be replaced with -all_load/-noall_load. + + Change-Id: I58121756de6a0560a6e49ca9d6bf9566a333cde3 + +2019-01-03 10:28 +0000 [c50f29dfad] Matt Jordan + + * Add core Prometheus support to Asterisk + + Prometheus is the defacto monitoring tool for containerized applications. + This patch adds native support to Asterisk for serving up Prometheus + compatible metrics, such that a Prometheus server can scrape an Asterisk + instance in the same fashion as it does other HTTP services. + + The core module in this patch provides an API that future work can build + on top of. The API manages metrics in one of two ways: + (1) Registered metrics. In this particular case, the API assumes that + the metric (either allocated on the stack or on the heap) will have + its value updated by the module registering it at will, and not + just when Prometheus scrapes Asterisk. When a scrape does occur, + the metrics are locked so that the current value can be retrieved. + (2) Scrape callbacks. In this case, the API allows consumers to be + called via a callback function when a Prometheus initiated scrape + occurs. The consumers of the API are responsible for populating + the response to Prometheus themselves, typically using stack + allocated metrics that are then formatted properly into strings + via this module's convenience functions. + + These two mechanisms balance the different ways in which information is + generated within Asterisk: some information is generated in a fashion + that makes it appropriate to update the relevant metrics immediately; + some information is better to defer until a Prometheus server asks for + it. + + Note that some care has been taken in how metrics are defined to + minimize the impact on performance. Prometheus's metric definition + and its support for nesting metrics based on labels - which are + effectively key/value pairs - can make storage and managing of metrics + somewhat tricky. While a naive approach, where we allow for any number + of labels and perform a lot of heap allocations to manage the information, + would absolutely have worked, this patch instead opts to try to place + as much information in length limited arrays, stack allocations, and + vectors to minimize the performance impacts of scrapes. The author of + this patch has worked on enough systems that were driven to their knees + by poor monitoring implementations to be a bit cautious. + + Additionally, this patch only adds support for gauges and counters. + Additional work to add summaries, histograms, and other Prometheus + metric types may add value in the future. This would be of particular + interest if someone wanted to track SIP response types. + + Finally, this patch includes unit tests for the core APIs. + + ASTERISK-28403 + + Change-Id: I891433a272c92fd11c705a2c36d65479a415ec42 + +2019-05-20 12:45 +0000 [3853fab3f5] Joshua Colp + + * pjproject-bundled: Add upstream timer fixes + + Fixed #2191: + - Stricter double timer entry scheduling prevention. + - Integrate group lock in SIP transport, e.g: for add/dec ref, + for timer scheduling. + + ASTERISK-28161 + Reported-by: Ross Beer + + Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5 + +2019-05-17 18:44 +0000 [be83591f99] George Joseph + + * res_rtp_asterisk: Add ability to propose local address in ICE + + You can now add the "include_local_address" flag to an entry in + rtp.conf "[ice_host_candidates]" to include both the advertized + address and the local address in ICE negotiation: + + [ice_host_candidates] + 192.168.1.1 = 1.2.3.4,include_local_address + + This causes both 192.168.1.1 and 1.2.3.4 to be advertized. + + Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db + +2019-05-13 15:37 +0000 [466a17964f] Alexei Gradinari + + * pjsip: replace 180 by 183 if SDP negotiation has completed + + The caller endpoint hears dead silence if a callee replies 180 (without SDP) + and the caller already received 183 (with SDP). + It happens because Asterisk sends 180 (WITH SDP) to the caller, + there are not incoming RTP packets from the callee + and Asterisk does not generate inband ringing, + so there are not any outgoing RTP packets to the caller. + + This patch replaces 180 by 183 if SDP negotiation has completed, + as if the caller endpoint is configured with "inband_progress=yes". + + In this case Asterisk will generate inband ringing untill Asterisk receive + incoming RTP packets from the callee. + + ASTERISK-27994 #close + + Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73 + +2019-05-10 10:48 +0000 [c5c953c1f1] George Joseph + + * Fixes for GCC 9 + + Various fixes for issues caught by gcc 9. Mostly snprintf + trying to copy to a buffer potentially too small. + + ASTERISK-28412 + + Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e + +2019-05-08 10:41 +0000 [7a6fd83aca] Joshua Colp + + * res_rtp_asterisk: Fix sequence number cycling and packet loss count. + + This change fixes two bugs which both resulted in the packet loss + count exceeding 65,000. + + The first issue is that the sequence number check to determine if + cycling had occurred was using the wrong variable resulting in the + check never seeing that cycling has occurred, throwing off the + packet loss calculation. It now uses the correct variable. + + The second issue is that the packet loss calculation assumed that + the received number of packets in an interval could never exceed + the expected number. In practice this isn't true due to delayed + or retransmitted packets. The expected will now be updated to + the received number if the received exceeds it. + + ASTERISK-28379 + + Change-Id: If888ebc194ab69ac3194113a808c414b014ce0f6 + +2019-05-07 11:08 +0000 [86836e0442] Ben Ford + + * pjsip_options.c: Allow immediate qualifies for new contacts. + + When multiple endpoints try to register close together using the same + AOR with qualify_frequency set, one contact would qualify immediately + while the other contacts would have to wait out the duration of the + timer before being able to qualify. Changing the conditional to check + the contact container count for a non-zero value allows all contacts to + qualify immediately. + + Change-Id: I79478118ee7e0d6e76af7c354d66684220db9415 + +2019-05-06 16:26 +0000 [def6bbc96b] Kevin Harwell + + * conversions.c: Add conversions for largest max sized integer + + Added a conversion for umax (largest maximum sized integer allowed). Adjusted + the other current conversion functions (uint and ulong) to be derivatives of + the umax conversion since they are simply subsets of umax. + + Also made the negative check move the pointer on spaces since strtoumax does it + anyways. + + Change-Id: I56c2ef2629d49b524c8df58af12951c181f81f08 + +2019-05-03 10:49 +0000 [85242a9bb9] Abhay Gupta + + * stasis: Hangup channel for Local channel No such extension error + + When we use early bridge with create and dial from stasis using Local channel + and the dialplan does not any entry the it is returned from core_local.c with + No such extension . + + In such case asterisk locks up till the channel is not hangup with the error + Exceptionally long voice queue length + + * Found that in such case app_control_dial fails on ast_call method and + return -1 + * Since it is called from stasis_app_send_command_async and return -1 does + not cause resources to be freed and since no PBX exist it is not able to + read from channel causing exceptionally long queue + * After putting this code found that the channel was releasing immediately + and resources were freed. + + ASTERISK-28399 + Reported by: Abhay Gupta + Tested by: Abhay Gupta + + Change-Id: I0a55c923fc6995559f808d63b9488762b4489318 + +2019-05-03 13:31 +0000 [089581f20a] George Joseph + + * build: Pass --fno-partial-inlining to third-party when appropriate + + When the gcc version is >= 8.2.1, we were already setting the + --fno-partial-inlining flag for Asterisk source files to get around + a gcc bug but we weren't passing the flag down to the bundled + builds of pjproject and jansson. + + ASTERISK-28392 + + Change-Id: I99ede9bc35408ecd096f7d5369e8192d3dc75704 + +2019-05-02 13:29 +0000 [ef92c69fa8] George Joseph + + * res_pjsip: Check return from pjsip_parse_uri calls + + Updated ast_sip_create_rdata_with_contact and registrar_find_contact + to check the return from pjsip_parse_uri before attempting to + use the uri returned. + + ASTERISK-28402 + Reported-by: Ross Beer + + Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7 + +2019-04-30 09:21 +0000 [71040078a3] Abhay Gupta + + * stasis: Only place stasis created and dialed channels into dial bridge. + + The dial bridge is meant to hold channels which have been created + and dialed in stasis. It handles the frames coming from them and raises + the appropriate events. + + It was possible for the code to mistakenly place calls which came + from the dialplan into the dial bridge if they were not in an + answered state. These channels are not outgoing channels and + should not be placed into the dial bridge. + + The code now checks to ensure that only stasis created channels are + placed into the dial bridge by checking that a PBX does not exist + on the channel. + + ASTERISK-27756 + + Change-Id: Ideee69ff06c9a0b31f7ed61165f5c055f51d21b6 + +2019-04-09 23:30 +0000 [3087c82eb6] Holger Hans Peter Freyther + + * stasis: Call callbacks when imparting fails + + After a bridge has been deleted the stasis control will depart + the channel and might attempt to re-add it to the dial bridge. + + The later can fail and this can lead to a situation that the stasis + control is unlinked but the after_bridge_cb_failed cb is executed trying + to access a dangling control object. + + Fix it by calling the after_cb's before bridge_channel_impart_signal. + + ASTERISK-26718 + + Change-Id: Ib4e8f70d7a21bd54afe3cb51cc6717ef7c355496 + +2019-04-30 06:22 +0000 [80dba268ea] Joshua Colp + + * app_confbridge: Add "all" variants of REMB behavior. + + When producing a combined REMB value the normal behavior + is to have a REMB value which is unique for each sender + based on all of their receivers. This can result in one + sender having low bitrate while all the rest are high. + + This change adds "all" variants which produces a bridge + level REMB value instead. All REMB reports are combined + together into a single REMB value that is the same for + each sender. + + ASTERISK-28401 + + Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c + +2019-04-23 05:00 +0000 [6bb70c93f1] Joshua Colp + + * rtp: Add support for transport-cc in receiver direction. + + The transport-cc draft is a mechanism by which additional information + about packet reception can be provided to the sender of packets so + they can do sender side bandwidth estimation. This is accomplished + by having a transport specific sequence number and an RTCP feedback + message. This change implements this in the receiver direction. + + For each received RTP packet where transport-cc is negotiated we store + the time at which the RTP packet was received and its sequence number. + At a 1 second interval we go through all packets in that period of time + and use the stored time of each in comparison to its preceding packet to + calculate its delta. This delta information is placed in the RTCP + feedback message, along with indicators for any packets which were not + received. + + The browser then uses this information to better estimate available + bandwidth and adjust accordingly. This may result in it lowering the + available send bandwidth or adjusting how "bursty" it can be. + + ASTERISK-28400 + + Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc + +2018-12-04 02:10 +0000 [7ce6d960d4] Abhay Gupta + + * app_amd: Fix infinite loop on silent calls + + The total time logic will now be executed on calls which + do not pass any media. + + ASTERISK-28143 + + Change-Id: I24726bd29d7e467fc721ca265363417234b22855 + +2019-04-29 11:13 +0000 [ed615afb7e] Rodrigo Ramírez Norambuena + + * app_queue: Set correct value by default for shared_lastcall + + There a long history here: + + In commit dd1e62c095c has introduce by default shared_lastcall = true by + default but this now only happen is there not [general] directive in + queues.conf + + After that, the commit 4b50e3f1ee84ae29da6d9eb3cfd9896a49d2394b fix the + sample file. + + We'll need to keep the same setting if there a general or not section in + configuration file since the shared_lastcall is by a long time in + sample files as default value to 'no'. + + Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c + +2019-04-23 09:47 +0000 [dc02d0d9f2] Ben Ford + + * stasis: Fix crash at shutdown. + + When compiling in dev mode, stasis statistics are enabled and can cause + a crash at shutdown due to the following: + - Containers are freed + - Topics and subscriptions remain + - When those topics and subscriptions are deallocated, they go to do + things with the container + + This changes the containers to global ao2 objects, and whenever needed + in the code, a reference must be obtained and checked before any + operations can be done. + + ASTERISK-28353 #close + + Change-Id: Ie7d5e907fcfcb4d65bd36d5e4eb923126fde8d33 + +2019-03-29 09:04 +0000 [8e21c25ce5] Antoni Goldstein + + * app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings + + Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled + at the earliest received PROGRESS or RINGING. + Added millisecond versions of DIALEDTIME and ANSWEREDTIME. + + Added millisecond versions of ast_channel_get_up_time and + ast_channel_get_duration in channel.c. + + ASTERISK-28363 + + Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1 + +2019-04-09 14:48 +0000 [ff0d0ac23a] Kevin Harwell + + * mwi core: Move core MWI functionality into its own files + + There is enough MWI functionality to warrant it having its own 'c' and header + files. This patch moves all current core MWI data structures, and functions + into the following files: + + main/mwi.h + main/mwi.c + + Note, code was simply moved, and not modified. However, this patch is also in + preparation for core MWI changes, and additions to come. + + Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0 + +2019-04-07 11:36 +0000 [8b7324ed3f] Guido Falsi + + * core/buildsystem: check the actual compiler being version + + Make compiler check use the output of the actual compiler being + used as reported by the CC variable, instead of unconditionally + running the "gcc" binary. Also only run the check if the compiler + is gcc or a cross-compile gcc. + + ASTERISK-28374 + + Change-Id: Icaacf6d93686ad21076878aa1504a23b4fc9d0f4 + +2019-04-19 09:33 +0000 [4f69ea928a] Lucas Mendes + + * res_indications: Fix indications remove command autocomplete + + We changed the validation of autocomplete parameter in the "indications + remove" command to avoid continue the execution of the command after + asking for autocomplete out of range parameters. + + ASTERISK-28391 + Reported by: lmendes86 + + Change-Id: I92b24131fd02f2e3c7fec966eea6f7a663310d40 + +2019-04-17 14:45 +0000 [d4e25710f7] George Joseph + + * res_remb_modifier: Propertly initialize bitrate to 0.0 + + ...and return the frame unaltered if bitrate can't be determined. + + Change-Id: Ib2175ab84f85a3d7060d31625f5a2c7fbcc2ba4c + +2019-04-08 17:04 +0000 [cffa2a74cb] Dan Cropp + + * res_pjsip: Added a norefersub configuration setting + + Added a new PJSIP global setting called norefersub. + Default is true to keep support working as before. + + res_pjsip_refer: Configures PJSIP norefersub capability accordingly. + + Checks the PJSIP global setting value. + If it is true (default) it adds the norefersub capability to PJSIP. + If it is false (disabled) it does not add the norefersub capability + to PJSIP. + + This is useful for Cisco switches that do not follow RFC4488. + + ASTERISK-28375 #close + Reported-by: Dan Cropp + + Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9 + +2019-04-09 19:09 +0000 [1d3272d4ed] sungtae kim + + * main/stasis.c: Added detail info for stasis show app cli + + Currently, the "stasis show app" cli doesn't give detail + of subscription/subscriber information. + Added more printings to show details. + + ASTERISK-28378 + + Change-Id: If25a6f14fe4f622bfb37462e891333da1fdf875f + +2019-04-16 10:58 +0000 [e69fcdfd83] Sean Bright + + * res_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority + + Suggested by abelbeck on the issue tracker. + + ASTERISK~28384 + Reported by: abelbeck + + Change-Id: Icee0fff2b58dbfaa80f2b68270fe69dfb0463fc0 + +2019-04-12 11:32 +0000 [8a32b68038] George Joseph + + * CI: Move test group config files to Jenkins + + One of the downaides of having things like test configuration + in the git repo is that it can't be changed at runtime. You have + to create a review for the changes and merge it mefore it will + take effect. + + This review moves the data currently held in + tests/CI/periodic-dailyTestGroups.json and + tests/CI/gateTestGroups.json into a Jenkins Config File attached + to the job definitions. This allows us to alter it from the + Jenkins UI at runtime. The original files stay in the repo + as documentation. + + Change-Id: I14b9702f6285ce1fb2420287ba0e7d3b59109763 + +2019-04-13 13:36 +0000 [d58d7d4500] Sean Bright + + * app_voicemail: Don't split mailbox options on comma + + Because the per-mailbox options are the last thing on a line, don't look + for or stomp on any subsequent commas. + + ASTERISK-27935 #close + Reported by: Sébastien Duthil + + Change-Id: I07b2eb4a33c303d0c7114d5b906f8c067c60a153 + +2019-04-12 09:33 +0000 [7e5709d726] Sean Bright + + * pbx.c: Ignore dashes in extensions when using extenpatternmatchnew + + Because hyphens are not matched literally in Asterisk dialplan, we need + to ignore them in our candidate extensions as well. + + ASTERISK-17695 #close + Reported by: test011 + + Change-Id: I227f02301577b1633e8a55b9fe9dc149935c03f0 + +2019-04-09 10:10 +0000 [63f86cac09] Sean Bright + + * app_voicemail: Cleanup stale lock files on module load + + If Asterisk crashes while a VM directory is locked, lock files in the VM + spool directory will not get properly cleaned up. We now clear them on + module load. + + ASTERISK-20207 #close + Reported by: Steven Wheeler + + Change-Id: If40ccd508e2f6e5ade94dde2f0bcef99056d0aaf + +2019-04-12 07:33 +0000 [26cdf042f4] George Joseph + + * ARI: Run 'make ari-stubs' + + An earlier contributor apparently forgot to run 'make ari-stubs' + before committing after making ARI model changes. + + Change-Id: I7813e5638e2821d11f4b968dc2aeab4f725190a6 + +2019-04-11 15:48 +0000 [f827193424] Sean Bright + + * res_ael: Create consistent label names across reloads + + Reset the internal counter that the AEL2 compiler uses for unique label + names before compiling. This keeps dialplan labels consistent across + reloads assuming the AEL2 has not changed. + + ASTERISK-17799 #close + Reported by: Kirill Katsnelson + + Change-Id: I30b3cc887d1ee0644d3f341e2fef16f525d7fae5 + +2019-04-11 15:29 +0000 [f7f1a2cbb7] Sean Bright + + * res_ael: Use Gosub in for loop expressions + + In AEL2, if a 'for' statement contains macro* calls, like: + + for (&iterator(${TRY},A); "${A}" != ""; &iterate(A)) { + + The AEL2 parser will translate these into calls to the deprecated Macro + dialplan application and use the antiquated pipe delimiter. + + Instead, convert these into calls to the Gosub dialplan application and + use commas as argument separators. + + ASTERISK-18593 #close + Reported by: Luke-Jr + + * 'macro' in this context means AEL2 macros, not the 'Macro' application + + Change-Id: I3d73716033b8e3e42e0209d355bf5f10c97045fc + +2019-04-11 11:03 +0000 [395c7ed5b7] Sean Bright + + * res_ael: Fix pattern matching against literal '+' + + When generating the regular expression that matches against existing + extensions, we need to escape literal characters that can also be + regular expression metacharacters. This was already being done for '*' + but we need to do the same for '+'. + + In passing, remove some unreachable code - strcmp() is already run + immediately when entering extension_matches(). + + ASTERISK-14939 #close + Reported by: klaus3000 + + Change-Id: I8d2cccb3479168fba1b0a6704c52198b396468f1 + +2019-04-11 12:49 +0000 [2cf4e8bff9] Sean Bright + + * pbx.c: Properly parse labels with leading digits + + If the target of a Goto is a label that starts with a number, we + erroneously treat the leading digits as a priority. + + ASTERISK-20182 #close + Reported by: Janu + + Change-Id: Ia78408c0805a729103917247ecfc802f6fafc94b + +2019-04-10 18:07 +0000 [a8f1e26d34] Alexander Anikin + + * chan_ooh323: fix h323 log file path + + Change h323 log path relative to AST_LOG_DIR instead of + /var/log/asterisk hardcoded + Add return back error message from OOH323EP initialize + + ASTERISK-28348 #close + + Reported by: Dmitry Shubin + + Change-Id: Ib102dd36bbe6c2a7a4ce6870ae9110d9000d7e98 + +2019-04-09 16:47 +0000 [fe58bc7bdf] Alexei Gradinari + + * res_pjsip: Fix transport_states ref leak + + Add missing ao2_ref(transport_state, -1) while iterate on a transport_states + container. + + Change-Id: I40e35b5a339121300c80075c30db47201a6c374e + +2019-04-01 15:38 +0000 [a4ab7f5f80] Ben Ford + + * build: Revise CHANGES and UPGRADE.txt handling. + + This changes the way that we handle adding changes to CHANGES and + UPGRADE.txt. The reason for this is because whenever someone needed to + make a change to one of these files and someone else had already done + so, you would run into merge conflicts. With this new setup, there will + never be merge conflicts since all changes will be documented in the + doc/-staging directory. The release script is now responsible for + merging all of these changes into the appropriate files. + + There is a special format that these files have to follow in order to be + parsed. The files do not need to have a meaningful name, but it is + strongly recommended. For example, if you made a change to pjsip, you + may have something like this "res_pjsip_relative_title", where + "relative_title" is something more descriptive than that. Inside each + file, you will need a subject line for your change, followed by a + description. There can be multiple subject lines. The file may look + something like this: + + Subject: res_pjsip + Subject: Core + + A description that explains the changes made and why. The release + script will handle the bulleting and section separators! + + You can still separate with new lines within your + description. + + The headers ("Subject" and "Master-Only") are case sensative, but the + value for "Master-Only" ("true" or "True") is not. + + For more information, check out the wiki page: + https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt + + ASTERISK-28111 #close + + Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47 + +2019-04-04 16:02 +0000 [391112d89a] Chris-Savinovich + + * config.c: Fix a crash in extconfig parsing + + When extconfig.conf file is parsed, the code previously searched for + character comma without verifying if error (null or blank). This caused + a segmentation error. + + Change-Id: Id76b452d8f330d11c2742c37232761ad71472a8b + +2019-04-03 10:55 +0000 [5009d6d97a] Salah Ahmed + + * chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info + + When the dtmf_mode on an endpoint is configured as "auto_info" + Asterisk will produce an inband DTMF tone alongside an INFO + message when sending DTMF. + + ASTERISK-28371 + + Change-Id: I1380b82f006e110a1b83fbb50c9873edd13a5d9a + +2019-04-02 15:49 +0000 [ccac55b894] Sebastian Kemper + + * loader: support for permanent dlopen() + + Asterisk assumes that dlopen() will always run the constructor of a + shared library and every dlclose() will run its destructor. But dlopen() + may be permanent, meaning the constructor will only be run once, as is + the case with musl libc. + + With a permanent dlopen() the Asterisk module loader does not work + correctly, because it's expectations regarding when the constructors and + destructors are run are not met. In fact a segmentation fault will occur + when the first module is "re-opened" that has AST_MODFLAG_GLOBAL_SYMBOLS + set (the dlopen() does not call the constructor, resource_being_loaded + is not set to NULL, then strlen is called with NULL instead of a string, + see issue ASTERISK-28319). + + This commit adds code to the loader that will manually run the + constructors/destructors of the (non-builtin) modules where needed. To + achieve this a new ao2 container (linked list) is started and filled + with objects that contain the names of the modules and the pointers to + their respective info structs. + + This behavior can be activated when configuring Asterisk + (--enable-permanent-dlopen). By default this is disabled, of course. + + ASTERISK-28319 #close + + Signed-off-by: Sebastian Kemper + Change-Id: I86693a0ecf25d5ba81c73773a03df4abc3426875 + +2019-04-03 17:55 +0000 [8ae9339f71] George Joseph + + * CI: Add --no-dev-mode option to buildAsterisk.sh + + The new option disables dev mode, TEST_FRAMEWORK and + MALLOC_DEBUG making the build more production-like. + + Change-Id: Ieb72497d4d91d5416684aaed702cc3f532099738 + +2019-04-03 10:24 +0000 [dd1cc7791c] Ben Ford + + * build: Fix compiler warnings/errors. + + The compiler complained about a couple of variables that weren't + initialized but were being used. Initializing them to NULL resolves the + warnings/errors. + + ASTERISK-28362 #close + + Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b + +2019-03-27 12:59 +0000 [d1d0692858] Kevin Harwell + + * bridge_softmix: use a float type to store the internal REMB bitrate + + REMB's exponent is 6-bits (0..63) and has a mantissa of 18-bits. We were using + an unsigned integer to represent the bitrate. However, that type is not large + enough to hold all potential bitrate values. If the bitrate is large enough + bits were being shifted off the "front" of the mantissa, which caused the + wrong value to be sent to the browser. + + This patch makes it so it now uses a float type to hold the bitrate. Using a + float allows for all bitrate values to be correctly represented. + + ASTERISK-28255 + + Change-Id: Ice00fdd16693b16b41230664be5d9f0e465b239e + +2019-03-29 08:07 +0000 [e8cf3693f6] Sean Bright + + * app_queue: Fix a few member pause bugs + + * Always set member->lastpause when setting member->paused + + * Fixed typo (using member->lastcall instead of member->lastpause) in + 'queue show' output. + + * Use a constant 'now' in 'queue show' output for a better point-in-time + view of time based stats. + + ASTERISK-27541 #close + Reported by: César Benjamín García Martínez + + Change-Id: Ib41ced90cfdb66f9bb1e7b263d0f6fc1ac6e18fa + +2019-03-26 14:56 +0000 [4edd24841d] Ben Ford + + * alembic: Fix errors during upgrade head. + + When trying to upgrade using alembic, a couple different errors kept + popping up that prevented the upgrade. An additional parameter was + needed when changing the schema for mwi_subscribe_replaces_unsolicited + from an integer to an enum. When changing from a string to an enum, the + type needed to be cast for postgresql. The other issue was a parameter + being used during column creation that did not exist. + + After fixing the upgrade process, it revealed errors with the downgrade + process. One was a variable not being defined in the downgrade function, + and the other was tables not existing when using MySQL. This was due to + a context check that should have encompassed MySQL, but in the end was + not doing so. + + Change-Id: Ib4d70cf3ce5080023a50be496272a777b55d6c8e + +2019-01-26 15:51 +0000 [30d568ddec] sungtae kim + + * stasis.c: Added topic_all container + + Added topic_all container for centralizing the topic. This makes more + easier to managing the topics. + + Added cli commands. + stasis show topics : It shows all registered topics. + stasis show topic : It shows speicifed topic's detail info. + + ASTERISK-28264 + + Change-Id: Ie86d125d2966f93de74ee00f47ae6fbc8c081c5f + +2019-03-27 14:30 +0000 [f78306470b] Matthew Fredrickson + + * res/res_rtp_asterisk: Enable rxjitter calculation for video + + It looks like we're not properly calculating jitter values on received + video streams. This patch enables the code that does jitter calculations + for those streams. + + Change-Id: Iaac985808829c8f034db8c57318789c4c8c11392 + +2019-03-27 11:03 +0000 [d5d8448ce5] Ben Ford + + * build: Add staging directories for future changes. + + This is the first step in changing the release process so that changes + made to the CHANGES and UPGRADE.txt files do not result in merge + conflicts every time multiple people modify these files. The changes + made will go in these new directories: doc/CHANGES-staging and + doc/UPGRADE-staging. The README.md files explain how things will work, + but here's a little overview. When you make a change that would go in + either CHANGES or UPGRADE.txt, this should instead be documented in a + new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory, + respectively. The format will look like this: + + Subject: res_pjsip + + A description that explains the changes made and why. The release + script will handle the bulleting and section separators! The + 'Subject:' header is case-sensitive. + + You can still separate with new lines within your description. + + Subject: res_ari + Master-Only: true + + You can have more than one subject, and they don't have to be the + same! Also, the 'Master-Only' header should always be true and is + also case-sensitive (but the value is not - you can have 'true' or + 'True'). This header will only ever be present in the master branch. + + For more information, check out the wiki page: + https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt + + This is an initial change for ASTERISK_28111. Functionally, this will + make no difference, but it will prep the directories for when the + changes from CHANGES and UPGRADE.txt are extracted. + + Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6 + +2019-03-25 18:05 +0000 [f236377ce9] Alexei Gradinari + + * pjsip: restrict function PJSIP_PARSE_URI to parse only SIP/SIPS URIs + + The next usage of PJSIP_PARSE_URI will crash asterisk + ${PJSIP_PARSE_URI(tel:+1234567890,host)} + or + ${PJSIP_PARSE_URI(192.168.1.1:5060,host)} + + The function pjsip_parse_uri successfully parses then, but returns + struct pjsip_other_uri *. + + This patch restricts parsing only SIP/SIPS URIs. + + Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e + +2019-03-26 13:07 +0000 [7043ed6ac9] Sean Bright + + * pjproject: Add timer patch from pjproject r5934 + + ASTERISK-28161 #close + Reported by: Ross Beer + + Change-Id: I65331d554695753005eaa66c1d5d4807fe9009c8 + +2019-03-26 16:55 +0000 [834d022da5] Sean Bright + + * app_queue: Fix documentation for QUEUE_MEMBER function. + + It was a copy/paste of the QUEUE_MEMBER_COUNT function's synopsis. + + ASTERISK-20986 #close + Reported by: Olivier Krief + + Change-Id: If51ec481feb35824a4e78ab5600b197b819b10be + +2019-03-21 18:09 +0000 [76768ad6ce] sungtae kim + + * main/json.c: Added app_name, app_data to channel type + + It was difficult to check the channel's current application and + parameters using ARI for current channels. Added app_name, app_data + items to show the current application information. + + ASTERISK-28343 + + Change-Id: Ia48972b3850e5099deab0faeaaf51223a1f2f38c + +2019-03-25 06:34 +0000 [d480f5eab2] Joshua Colp + + * manager: Use separate lock for session event notification. + + When notifying a manager session that new events were available + the same lock was used that was also held when doing things within + the session (such as sending events out). If the manager session + blocked for a period of time this would cause a back up of messages + in Stasis and would also block any other sessions from receiving + events. + + This change adds a separate lock to the manager session which is + strictly used for notifying it that new events are available. + + ASTERISK-28350 + + Change-Id: Ifbcac007faca9ad0231640f5e82a6ca9228f261b + +2019-03-25 14:31 +0000 [1499640da9] Sean Bright + + * chan_sip: Ensure 'qualifygap' isn't negative + + Passing negative intervals to the scheduler rips a hole in the + space-time continuum. + + ASTERISK-25792 #close + Reported by: Paul Sandys + + Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7 + +2019-03-25 11:42 +0000 [e5d990d01d] Alexei Gradinari + + * res_config_odbc: set empty extended field as a single whitespace + + If Realtime @ variable value is NULL or empty or contains only whitespaces + then when we try to retrieve it using PJSIP_ENDPOINT we get WARNING + pjsip_endpoint_function_read: Unknown property @my_var for PJSIP endpoint. + And the variable is missing in the result of CLI pjsip show endpoint. + + This patch keeps empty sorcery extended field. + + ASTERISK-28341 #close + + Change-Id: I221fccc04cbfa2be17ce971f64ae0e74e465eea0 + +2019-03-22 14:46 +0000 [41a2662e16] Matthew Fredrickson + + * main/taskprocessor: Increase max name length of taskprocessors + + Since the new names went in, the maximum taskprocessor name is too + short. This patch increases the name field to a length to better + handle the new names. + + Change-Id: I32f32d6926f25c8ef5a91303fd2988d2c2858877 + +2019-03-14 11:46 +0000 [7e77815ad1] George Joseph + + * sorcery.c: Sorcery enhancements for wizard management + + Added ability to specifiy a wizard is read-only when applying + it to a specific object type. This allows you to specify + create, update and delete callbacks for the wizard but limit + which object types can use them. + + Added the ability to allow an object type to have multiple + wizards of the same type. This is indicated when a wizard + is added to a specific object type. + + Added 3 new sorcery wizard functions: + + * ast_sorcery_object_type_insert_wizard which does the same thing + as the existing ast_sorcery_insert_wizard_mapping function but + accepts the new read-only and allot-duplicates flags and also + returns the ast_sorcery_wizard structure used and it's internal + data structure. This allows immediate use of the wizard's + callbacks without having to register a "wizard mapped" observer. + + * ast_sorcery_object_type_apply_wizard which does the same + thing as the existing ast_sorcery_apply_wizard_mapping function + but has the added capabilities of + ast_sorcery_object_type_insert_wizard. + + * ast_sorcery_object_type_remove_wizard which removes a wizard + matching both its name and its original argument string. + + * The original logic in __ast_sorcery_insert_wizard_mapping was moved + to __ast_sorcery_object_type_insert_wizard and enhanced for the + new capabilities, then __ast_sorcery_insert_wizard_mapping was + refactored to just call __ast_sorcery_insert_wizard_mapping. + + * Added a unit test to test_sorcery.c to test the read-only + capability. + + Change-Id: I40f35840252e4313d99e11dbd80e270a3aa10605 + +2019-03-10 17:53 +0000 [629962d1f7] sungtae kim + + * res/res_stasis: Fixed wrong StasisEnd timestamp + + Because StasisEnd's timestamp created it's own timestamp, it makes + wrong timestamp, compare to other channel event(ChannelDestroyed). + Fixed to getting a timestamp from the Channel's timestamp. + + ASTERISK-28333 + + Change-Id: I5eb8380fc472f1100535a6bc4983c64767026116 + +2019-03-14 09:55 +0000 [0fac5bcbe5] Sean Bright + + * vector: Add AST_VECTOR_COMPACT() to reclaim wasted space + + This might be useful in situations where you are loading an undetermined number + of items into a vector and don't want to keep (potentially) 2x the necessary + memory around indefinitely. + + Change-Id: I9711daa0fe01783fc6f04c5710eba84f2676d7b9 + +2019-03-14 11:53 +0000 [45a8892e67] Richard Mudgett + + * taskprocessor.c: Fix printf type mismatch + + A size_t is not always an unsigned long. + + * Use the %zu format specifier in the ast_cli() printf format string since + AST_VECTOR_SIZE() returns a size_t value. + + Change-Id: Ib102dd36bbe6c2a7a4ce6870ae9110d978dd7e98 + +2019-03-08 09:40 +0000 [63d90c38eb] George Joseph + + * app.c: Remove deletion of pool topic on mwi state delete + + As part of an earlier voicemail refactor, ast_delete_mwi_state_full + was modified to remove the pool topic for a mailbox when the state + was deleted. This was an attempt to prevent stale topics from + accumulating when app_voicemail was reloaded and a mailbox went + away. Unfortunately because of the fact that when app_voicemail + reloads, ALL mailboxes are deleted then only current ones recreated, + topics were being removed from the pool that still had subscribers + on them, then recreated as new topics of the same name. So now + modules like res_pjsip_mwi are listening on a topic that will + never receive any messages because app_voicemail is publishing on + a different topic that happens to have the same name. The solutiuon + to this is not easy and given that accumulating topics for + deleted mailboxes is less evil that not sending NOTIFYs... + + * Removed the call to stasis_topic_pool_delete_topic in + ast_delete_mwi_state_full. + + Also: + + * Fixed a topic reference leak in res_pjsip_mwi + mwi_stasis_subscription_alloc. + + * Added some debugging to mwi_stasis_subscription_alloc, + stasis_topic_create, and topic_dtor. + + * Fixed a topic reference leak in an error path in + internal_stasis_subscribe. + + ASTERISK-28306 + Reported-by: Jared Hull + + Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27 + +2019-03-02 05:37 +0000 [71c0c7f631] sungtae kim + + * res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics + + Added ARI resource for channel statistics. + GET /ari/channels/{channelId}/rtp_statistics : It returns given + channel's rtp statistics detail. + + ASTERISK-28320 + + Change-Id: I4343eec070438cec13f2a4f22e7fd9e574381376 + +2019-03-09 08:39 +0000 [7d5409912f] cirillor + + * Variable ALTCONF ignored when service is used in Debian + + When variable ALTCONF is defined, the command start prints the message + "Unable to open specified master config file '"/etc/asterisk/asteris..." + and use default configurations. + + ASTERISK-28332 + + Change-Id: I7595e582a0ee2c1051ea35435e247e27906957ef + +2019-03-13 06:05 +0000 [1d074debfb] Joshua Colp + + * stasis: Allow empty application arguments to move. + + Change-Id: I1e4d37415f3034abe36496dc30209c2303e6af5c + +2019-03-12 20:39 +0000 [a40198a4d4] Corey Farrell + + * Revert "Test_cel: Fails when DONT_OPTIMIZE is off" + + This reverts commit 1c8378bbc9639739c079df37897ff02f94af0f07. + + Change-Id: I1b9227b263c3dc4246a50aebf52a7640a0f7ea07 + +2019-03-06 07:20 +0000 [48e407e506] Dömsödi Gergely + + * app_queue: fix ring_entry to access nativeformats with a channel lock + + Fixes an intermittent segmentation fault which occured when accessing + nativeformats of a channel which entered into a queue. + + ASTERISK-27964 + Reported by: Francisco Seratti + + Change-Id: Ic87fa7a363f3b487c24ce07032f4b2201c22db9e + +2019-03-12 13:25 +0000 [6f158d27fc] George Joseph + + * Makefile.moddir_rules: Pass PJPROJECT_BUNDLED to download_externals + + The download_externals script wasn't getting the PJPROJECT_BUNDLED + environment variable passed down to it so it wasn't downloading + the appropriate variant of res_digium_phone. This could cause + crashes in the DPMA. + + Change-Id: I5daa9369c7af1fd556d892e89a85f279a2533425 + +2019-03-06 16:21 +0000 [e2eb19b363] sungtae kim + + * res/res_ari: Added timestamp as a requirement for all ARI events + + Changed to requirement to having timestamp for all of ARI events. + The below ARI events were changed to having timestamp. + PlaybackStarted, PlaybackContinuing, PlaybackFinished, + RecordingStarted, RecordingFinished, RecordingFailed, + ApplicationReplaced, ApplicationMoveFailed + + ASTERISK-28326 + + Change-Id: I382c2fef58f5fe107e1074869a6d05310accb41f + +2019-03-07 13:48 +0000 [449dff997c] Chris-Savinovich + + * partial-inlining: disable partial-inlining if gcc>=8.2.1 + + Apply flag -fno-partial-inlining on default optimization if and only if + gcc version >= 8.2.1 (this is the current ver on Fedora and Ubuntu). + This is done to avoid a bug that causes arithmetic calculations to fail + if the following conditions are met: + 1. TEST_FRAMEWORK on + 2. DONT_OPTIMIZE off + 3. Fedora and Ubuntu + 4. GCC 8.2.1 + 5. There must exist a certain combination of multithreading. + 6. Optimization level -O2 and -O3 + 7. Flag -fpartial-inline activated (default when optimization level>=2) + The following link points to a similar gcc bug reported in 2015. This leads me + to believe the bug has regressed. Note I am not able to replicate this bug + in an environment other than Asterisk + Test Framework + Test_cel because the + multithreading combination that causes it seems to be unique. Therefore I + am temporarily abandoning any thoughts of reporting the new occurrence of this + bug to gcc.gnu.org. https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65307 + + Change-Id: Ibd1afe60e0a38b88e85fdcd9b051004601c2f102 + +2019-03-07 06:28 +0000 [0231dd6ae7] Joshua Colp + + * stasis: Improve topic/subscription names and statistics. + + Topic names now follow: :[/] + + This ensures that they are all unique, and also provides better + insight in to what each topic is for. + + Subscriber ids now also use the main topic name they are + subscribed to and an incrementing integer as their identifier to + make it easier to understand what the subscription is primarily + responsible for. + + Both the CLI commands for listing topic and subscription statistics + now sort to make it a bit easier to see what is going on. + + Subscriptions will now show all topics that they are receiving messages + from, not just the main topic they were subscribed to. + + ASTERISK-28335 + + Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d + +2019-03-05 08:15 +0000 [0d6d51b175] cirillor + + * chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function + + Add logical group at DAHDIChannel event + and create "dahdi_group" at CHANNEL function. + + ASTERISK-28317 + + Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a + +2019-03-03 09:20 +0000 [8641fd9700] sungtae kim + + * res/res_rtp_asterisk.c: Fixing possible divide by zero + + Currently, when the Asterisk calculates rtp statistics, it uses + sample_count as a unsigned integer parameter. This would be fine + for most of cases, but in case of large enough number of sample_count, + this might be causing the divide by zero error. + + ASTERISK-28321 + + Change-Id: If7e0629abaceddd2166eb012456c53033ea26249 + +2019-03-08 14:12 +0000 [825ea9ddb9] Sean Bright + + * res_musiconhold: Remove redundant option parsing + + Change-Id: I481fabd8eaf2e4e7ffb5c8285b294742826e7d12 + +2019-03-04 01:50 +0000 [4661c08549] Torrey Searle + + * chan_pjsip: add a flag to ignore 183 responses if no SDP present + + chan_sip will always ignore 183 responses that do not contain SDP + however, chan_pjsip will currently always translate it into a + 183 with SDP. This new flag allows chan_pjsip to have the same + behavior as chan_sip. + + ASTERISK-28322 #close + + Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a + +2019-03-07 17:17 +0000 [9b7b8cb155] Corey Farrell + + * jansson: json_pack with new format to verify required runtime version. + + Add a json_pack at startup that will fail if runtime links against a + library older than jansson-2.11. + + Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a + +2019-03-07 17:15 +0000 [57850c7861] Sean Bright + + * app_meetme: Don't mute joining admins if conference is muted + + ASTERISK-28328 #close + + Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b + +2019-03-06 15:04 +0000 [2473b791b9] Sean Bright + + * Replace calls to strtok() with strtok_r() + + strtok() uses a static buffer, making it not thread safe. + + Also add a #define to cause a compile failure if strtok is used. + + Change-Id: Icce265153e1e65adafa8849334438ab6d190e541 + +2019-03-07 16:05 +0000 [7b02a9617c] Sean Bright + + * samples: Fix comment typo in pjsip.conf.sample + + Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857 + +2019-03-07 07:52 +0000 [6626df586e] Ben Ford + + * res_stasis: Add ability to switch applications. + + Added the ability to move between Stasis applications within Stasis. + This can be done by calling 'move' in an application, providing (at + minimum) the channel's id and the application to switch to. If the + application is not registered or active, nothing will happen and the + channel will remain in the current application, and an event will be + triggered to let the application know that the move failed. The event + name is "ApplicationMoveFailed", and provides the "destination" that the + channel was attempting to move to, as well as the usual channel + information. Optionally, a list of arguments can be passed to the + function call for the receiving application. A full example of a 'move' + call would look like this: + + client.channels.move(channelId, app, appArgs) + + The control object used to control the channel in Stasis can now switch + which application it belongs to, rather than belonging to one Stasis + application for its lifetime. This allows us to use the same control + object instead of having to tear down the current one and create + another. + + ASTERISK-28267 #close + + Change-Id: I43d12b10045a98a8d42541889b85695be26f288a + +2019-03-04 16:05 +0000 [f6b5b7208c] Sean Bright + + * app_queue: Handle empty 'interface' in queue member config + + While the 'interface' column is a NOT NULL, the empty string is still + allowed. res_config_odbc treats the empty string as a NULL and we crash + when trying to dereference. + + Also cleaned up an adjacent error message for consistency. + + ASTERISK-28168 #close + + Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202 + +2019-03-04 12:36 +0000 [f098d4a325] Sean Bright + + * sip_to_pjsip: Make multiline comment parsing consistent with Asterisk + + In Asterisk configuration, a multiline comment starts with ;-- as long as it is + not followed by another dash (i.e. ;--- is not a multiline comment). + + ASTERISK-28323 #close + + Change-Id: I32dc38e0fac01d3c0805d27d35d2365a7c37ca72 + +2019-02-28 06:24 +0000 [2980622d2b] Joshua Colp + + * basic-pbx: Update configuration to work with current modules. + + The res_pjsip_websocket module requires the res_http_websocket + module so ensure it is loaded. As well the res_pjsip_notify + module needs the pjsip_notify.conf configuration file so + ensure it is installed. + + ASTERISK-28272 + + Change-Id: I261659b84e7a6ac4cb49990d9badb4b2ad01bacd + +2019-02-08 15:32 +0000 [3638c433ac] sungtae kim + + * bridging: Add creation timestamps + + This small feature will help to checking the bridge's status to + figure out which bridge is in old/zombie or not. Also added + detail items for the 'bridge show *' cli to provide more detail + info. And added creation item to the ARI as well. + + ASTERISK-28279 + + Change-Id: I460238c488eca4d216b9176576211cb03286e040 + +2019-02-28 10:01 +0000 [106a8ff05c] Sean Bright + + * res_pjsip_diversion: Use static pj_str_t for Diversion header names + + PJSIP assumes that these header names are not allocated, and does not + clone the name strings when reusing headers. + + Block unload of res_pjsip_diversion until shutdown to ensure static + memory stays valid. + + ASTERISK-28312 #close + + Change-Id: Ibd6ea55ec4a604bbd43ac07f8d0b54da2c39b8b9 + +2019-03-01 15:17 +0000 [f8295e0771] Rodrigo Ramírez Norambuena + + * CHANGES: Document addition of 'wrapuptime' argument to AddQueueMember() + + Change-Id: Ieb332d018ae3f2fc82b9465381fde0f299af1611 + +2019-02-28 15:36 +0000 [8dc5f86095] Sean Bright + + * menuselect: Add license header to menuselect_gtk.c + + This file was added to the Subversion repository on 2007-03-15 by + Russell Bryant, a Digium employee at the time. + + ASTERISK-24173 #close + + Change-Id: Ie866fa9d31d550467613d362b35b03c031ee594d + +2019-02-27 19:09 +0000 [719a4643ab] Sean Bright + + * res_config_odbc: Avoid deadlock when max_connections = 1 + + Rather than calling ast_odbc_find_table() in the prepare callback, call + it beforehand and pass it in to the callback to avoid the need for a + second connection. + + ASTERISK-28166 #close + + Change-Id: I6f8a0b9990d636fd6bc1a92ed70f7050d2436202 + +2019-01-30 13:25 +0000 [8f9ffe5905] George Joseph + + * res_pjsip_sdp_rtp: Fix return code from apply_negotiated_sdp_stream + + apply_negotiated_sdp_stream was returning a "1" when no joint + capabilities were found on an outgoing call instead of a "-1". + This indicated to res_pjsip_session that the handler DID handle + the sdp when in fact it didn't. Without the appropriate setup, + a subsequent media frame coming in would have an invalid stream_num + and cause a seg fault when the stream was attempted to be retrieved. + + apply_negotiated_sdp_stream now returns the correct "-1" and any + media is now discarded before it reaches the core stream processing. + + ASTERISK-28260 + Reported by: Sotiris Ganouris + + Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f + +2019-02-28 06:51 +0000 [101272d0dc] Sean Bright + + * Revert "pjsip_message_filter: Only do interface lookup for wildcard addresses." + + This reverts commit d524ad523d0d32babba309810b5bccd09cb7f467. + + Reason for revert: This causes Contact and Via headers to have the wrong + transport address. + + ASTERISK-28309 #close + + Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8 + +2019-02-27 19:52 +0000 [82a43394ed] Sean Bright + + * res_pjsip_config_wizard: Don't crash if misconfigured + + If both send_registrations and send_auth are both set to yes, + outbound_auth/username must be set or we crash. + + ASTERISK-27992 #close + + Change-Id: I6418d56de1ae53f80393b314c2584048fbf7f11d + +2019-02-20 11:03 +0000 [930a7fe910] Kevin Harwell + + * res_pjsip_registrar: blocked threads on reliable transport shutdown take 3 + + When a contact was removed by the registrar it did not always check to see if + the circumstances involved a monitored reliable transport. For instance, if the + 'remove_existing' option was set to 'true' then when existing contacts were + removed due to 'max_contacts' being reached, those existing contacts being + removed did not unregister the transport monitor. + + Also, it was possible to add more than one monitor on a reliable transport for + a given aor and contact. + + This patch makes it so all contact removals done by the registrar also remove + any associated transport monitors if necessary. It also makes it so duplicate + monitors cannot be added for a given transport. + + ASTERISK-28213 + + Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a + +2019-02-27 10:37 +0000 [e0fc663295] George Joseph + + * CI: Update jenkinsfiles with new Gerrit URLs + + The recent upgrade of Gerrit to 2.16 elimiated referencing a + repository in a way the jenkinsfiles were relying on so + the URL references were changed to a more consistent and supported + format. + + Change-Id: I2e8e3f213b9a96bb1b27665eca4a9a24bc49820e + (cherry picked from commit 5ce084579f897096163b4e0c2ed4e8e1a8558cca) + +2019-02-20 13:15 +0000 [9ee76cf070] George Joseph + + * res_mwi_devstate.c: New module to allow presence subs to VM boxes + + This module allows presence subscriptions to voicemail boxes. This + allows common BLF keys to act as voicemail waiting indicators. + + ASTERISK-28301 + + Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd + +2019-02-25 09:41 +0000 [360f543677] Torrey Searle + + * res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen + + Delivery timeval in the smoother object will fall behind while a DTMF is + being generated. This can eventually lead to invalid rtp timestamps. + To prevent this from happening the smoother needs to be reset after every + DTMF to keep the timing up to date. + + ASTERISK-28303 #close + + Change-Id: Iaba3f7b428ebd72a4caa90e13b829ab4f088310f + +2019-02-25 15:32 +0000 [574128dec6] Kevin Harwell + + * rest-api-templates/asterisk_processor - replace http line breaks with line feed + + Including line breaks (
,
,
) in certain parts of the rest-api + json definition (e.g. summary, notes) displays them correctly in swagger. + However, when the field gets converted to the wiki format those breaks get + escaped and show up in the text as the actual string literal "
" etc... + + This patch makes it so when converting to the wiki format it replaces all line + break values (
, etc...) with line feeds ('\n'). + + Change-Id: Ie1c9faa0d1c5d622804cc0a21ce769095b08aa3d + +2019-02-25 06:10 +0000 [e687cf214d] Joshua C. Colp + + * res_ari_applications: Fix incorrect call to ao2_lock. + + When listing the applications the apps lock was incorrectly + locked twice instead of being locked and then unlocked. + + ASTERISK-28302 + + Change-Id: If7d064592a9e88c0f1049214c50e02be6dabf79e + +2019-02-21 12:06 +0000 [e6b67b2a5d] Joshua Colp + + * res_pjsip_sdp_rtp: Allow only single ssrc attribute. + + When processing SSRC attributes we were iterating through + all of them, even though we only need to know the remote + SSRC once. This was problematic because some browsers group + SSRCs together on a stream, and due to our negotiation only + end up using the first one. Since we set the second one as + the remote SSRC we would drop the received media from them + instead of allowing it through. + + In the future this may be extended to allow SSRC groups + and to use information from the attributes. + + Change-Id: I4dc87087dbe56a83aa65f0f897bbd4ca75ec1270 + +2019-01-09 04:27 +0000 [b4ccaad671] Sungtae Kim + + * http.c: Support separated HTTP request + + Currently, the Asterisk does not support seperated HTTP request. + This patch make the Asterisk enables to wait lest part of HTTP request. + Also increases acceptable HTTP body length to 40k to support more + larger request. + + ASTERISK-28236 + + Change-Id: I48a401aa64a21c3b37bf3cb4e0486d64b7dd8aa1 + +2019-02-20 12:48 +0000 [bc8dead610] George Joseph + + * Core: Increase AST_PBX_MAX_STACK to 512 if not LOW_MEMORY + + The current settings AST_PBX_MAX_STACK is 128 entries which is + too low for some FreePBX installations with complex parking + arrangements. Increased to 512 if LOW_MEMORY is not defined. + + ASTERISK-28300 + + Change-Id: I7c4b540bc92e6642df0f3da639b003f7da8b1299 + +2019-02-20 12:22 +0000 [a286f546f1] Joshua C. Colp + + * stasis: Store subscriber uniqueids with topic statistics. + + This change provides an easier mechanism to determine which + subscribers are subscribed to a topic. Using this you can + inspect the specific subscribers for further details. + + Change-Id: I8deea21703cd5c5357b85593b46c3eaf24e18c0c + +2019-02-15 12:53 +0000 [c2adeb9dc2] George Joseph + + * taskprocessor: Enable subsystems and overload by subsystem + + To prevent one subsystem's taskprocessors from causing others + to stall, new capabilities have been added to taskprocessors. + + * Any taskprocessor name that has a '/' will have the part + before the '/' saved as its "subsystem". + Examples: + "sorcery/acl-0000006a" and "sorcery/aor-00000019" + will be grouped to subsystem "sorcery". + "pjsip/distributor-00000025" and "pjsip/distributor-00000026" + will bn grouped to subsystem "pjsip". + Taskprocessors with no '/' have an empty subsystem. + + * When a taskprocessor enters high-water alert status and it + has a non-empty subsystem, the subsystem alert count will + be incremented. + + * When a taskprocessor leaves high-water alert status and it + has a non-empty subsystem, the subsystem alert count will be + decremented. + + * A new api ast_taskprocessor_get_subsystem_alert() has been + added that returns the number of taskprocessors in alert for + the subsystem. + + * A new CLI command "core show taskprocessor alerted subsystems" + has been added. + + * A new unit test was addded. + + REMINDER: The taskprocessor code itself doesn't take any action + based on high-water alerts or overloading. It's up to taskprocessor + users to check and take action themselves. Currently only the pjsip + distributor does this. + + * A new pjsip/global option "taskprocessor_overload_trigger" + has been added that allows the user to select the trigger + mechanism the distributor uses to pause accepting new requests. + "none": Don't pause on any overload condition. + "global": Pause on ANY taskprocessor overload (the default and + current behavior) + "pjsip_only": Pause only on pjsip taskprocessor overloads. + + * The core pjsip pool was renamed from "SIP" to "pjsip" so it can + be properly grouped into the "pjsip" subsystem. + + * stasis taskprocessor names were changed to "stasis" as the + subsystem. + + * Sorcery core taskprocessor names were changed to "sorcery" to + match the object taskprocessors. + + Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56 + +2019-02-08 14:48 +0000 [8681fc9db7] Kevin Harwell + + * ARI event type filtering + + Event type filtering is now enabled, and configurable per application. An app is + now able to specify which events are sent to the application by configuring an + allowed and/or disallowed list(s). This can be done by issuing the following: + + PUT /applications/{applicationName}/eventFilter + + And then enumerating the allowed/disallowed event types as a body parameter. + + ASTERISK-28106 + + Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b + +2019-02-19 10:06 +0000 [f4c9a351d8] Joshua Colp + + * CI: Use tmpfs option to Docker instead of mount. + + Some tests require Asterisk to execute scripts which + are stored in /tmp. When mount is used for tmpfs there + is no ability to allow scripts to be executed from + that location. + + This change switches to using tmpfs which can be told + to allow executables to be run from /tmp. + + Change-Id: I0e598ca2b76af1f7f2d29f0da7b1731a214a291a + +2019-02-08 14:47 +0000 [8f1b3edde8] Kevin Harwell + + * json.c/strings.c - Add a couple of utility functions + + Added 'ast_json_object_string_get' to the JSON wrapper in order to make it a + little easier to retrieve a string field from the JSON object. + + Also added an 'ast_strings_equal' function that safely checks (checks for NULLs) + for equality between two strings. + + Change-Id: I26f0a16d61537505eb41b4b05ef2e6d67fc2541b + +2018-12-11 08:15 +0000 [ce0523a57e] Rodrigo Ramírez Norambuena + + * app_queue: Enable set the wrapuptime from AddQueueMember application + + This change add ability to set the wrapuptime per-member using the + AddQueueMember application. + + The feature to set wrapuptime per member was include in the issue + ASTERISK-27483 for static member by configuration file and was not + added to set from AddQueueMember. + + ASTERISK-28055 #close + + Change-Id: I7c7ee4a6f804922cd7c42cb02eea26eb3806c6cf + +2019-02-12 03:50 +0000 [8ea9608efb] Torrey Searle + + * res/res_rtp_asterisk: clear smoother when local bridging + + p2p_write updates txformat but doesn't require a smoother. If a smoother + was created by another bridge type the smoother could fall out of date causing + one way audio issues. To prevent this the smoother is now destroyed on the + start of native bridge. + + ASTERISK-28284 #close + + Change-Id: I84e67f144963787fff9b4d79ac500514fb40cdc6 + +2019-02-14 17:09 +0000 [fb651756c7] sungtae kim + + * chan_pjsip: Changed to continued after invalid media for pjsip show channelstats + + Currently, the pjsip show channelstats cli does not show channel's + stats after hits the invalid channel info. This makes hard to use + this cli. Changed to keep iterate after hits the invalid channel + info. + + ASTERISK-28292 + + Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c + +2019-01-22 06:02 +0000 [7e1d881d89] Sungtae Kim + + * res_pjsip_session Added rtcp stats result vector into the session + + Currently, the Asterisk's pjsip_session module does not keeping the + rtcp's stats info after it was removed. But by adding the results + vector and keeping it until session is destroying, it can give more + useful information for other modules. + + ASTERISK-28253 + + Change-Id: Ib25c2d3fc4da084aecfde2a82c1b1d733bd64fa5 + +2019-02-07 09:52 +0000 [c2ea9c90a2] Joshua Colp + + * ci: Rerun unit tests when non-code changes occur. + + This change makes it so that even if non-code changes + occur (such as commit message changing) unit tests + will still be run and result in a verification. + + ASTERISK-28251 + + Change-Id: I6491fff7c93e5d5cd8e41054486968bf66c4f608 + +2019-02-07 09:23 +0000 [61a8f79a29] Kevin Harwell + + * res_pjsip_registrar: lock transport monitor when setting 'removing' flag + + A previous patch attempt to mitigate blocked threads on transport shutdown for + a given contact. It was thought that a second lock could be avoided by checking + the 'removing' flag on the transport monitor twice (once before and once after + the normal named aor locking). However as with usual threading issues if the + timing was right the original problem still occured. + + This patch adds locking around the first 'removing' flag check and set, thus + nullifying the secondary check, so it was removed. + + ASTERISK-28213 + + Change-Id: Iaa8e36e5311789549b76d8de42dfcea96013b2ed + +2019-02-06 06:16 +0000 [54a912b26d] Joshua Colp + + * res_odbc: Add basic query logging. + + When Asterisk is connected and used with a database the response + time of the database can cause problems in Asterisk if it is long. + Normally the only way to see this problem would be to retrieve a + backtrace from Asterisk and examine where things are blocked, or + examine the database to see if there is any indication of a + problem. + + This change adds some basic query logging to make it easier to + investigate such a problem. When logging is enabled res_odbc will + now keep track of the number of queries executed, as well as the + query that has taken the longest time to execute. There is also + an option which will cause a WARNING message to be output if a + query takes longer than a configurable amount of time to execute. + + This makes it easier and clearer for users that their database may + be experiencing a problem that could impact Asterisk. + + ASTERISK-28277 + + Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6 + +2018-12-05 16:09 +0000 [5a2a7d65b5] Sungtae Kim + + * main/cdr: Fixed cdr start overwriting + + The CDR was overwriting the start time when the call continued the + dialplan from the ARI stasis or a Local channel was originated. + + This change fixes this by no longer reinitializing the CDR when + transitioning out of the dialed pending state to the single state. + + ASTERISK-28181 + + Change-Id: I921bc04064b6cff1deb2eea56a94d86489561cdc + +2018-11-19 18:44 +0000 [e2bbab17b3] Giuseppe Sucameli + + * Fix deadlock handling subscribe req during res_parking reload + + Split destroy_hint method to separate hint removal and extension hint + state changed callback, the latter now called via stasis. + This avoids deadlock between res_parking reload that is removing the + parking lot and the related hint and subscribe requests coming for the + same parking lot. + + ASTERISK-28173 + + Change-Id: I5b03c3455b3b12b6f83cea4cc34f4b4b20444f7e + +2019-02-04 13:55 +0000 [f174eb4ac1] Sean Bright + + * sounds: Sort 'core show sounds' output + + Change-Id: Ib39052a745040f75eb635f15a042da15b20e22ab + +2019-01-29 10:48 +0000 [3f9c5fba95] Ben Ford + + * res_stasis: Auto-create context and extens on Stasis app launch. + + At AstriCon, there was a strong desire for the ability to completely + bypass dialplan when using ARI. This is possible through the automatic + creation of a context and a couple of extensions whenever an application + is started. + + For example, if you have an application named 'ari-example', a context + named 'stasis-ari-example' will be automatically created whenever this + application is started as long as one does not already exist. Two + extensions (a match-all extension for Stasis and a 'h' extension) are + created within this context. Any endpoint that registers to Asterisk + within this context will send all calls to the corresponding Stasis + application. When the application is destroyed, the context is removed. + + ASTERISK-28104 #close + + Change-Id: Ie35bd93075e05b05e3ae129a83c9426931b7ebac + +2019-02-04 07:09 +0000 [ac2d302c2c] George Joseph + + * bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64 + + On OpenSuse Leap, libjansson.a is installed in + third-party/jansson/dest/lib64 instead of lib (which is where + the top-level makeopts looks). This causes a link failure. + + * Updated jansson/Makefile to add an explicit --libdir to force + the installation to third-party/jansson/dest/lib. + + ASTERISK-28271 + Reported by: David Wilcox + + Change-Id: Ibf8af75e5da13562105fcc39ed898c6ef0b5a5f3 + +2019-01-28 17:21 +0000 [ac90968afd] sungtae kim + + * Added ARI resource /ari/asterisk/ping + + Added ARI resource. + GET /ari/asterisk/ping : It returns "pong" message with timestamp + and asterisk id. It would be useful for simple heath check. + + Change-Id: I8d24e1dcc96f60f73437c68d9463ed746f688b29 + +2019-01-15 17:20 +0000 [f668db9ba0] Kevin Harwell + + * pjsip/config_global: regcontext context not created + + The context specified by 'regcontext' was not being created, so when Asterisk + attempted to later dynamically add an extension it would fail. This patch now + creates the context if a 'regcontext' is specified. + + ASTERISK-28238 + + Change-Id: I0f36cf4ab0a93ff4b1cc5548d617ecfd45e09265 + +2019-01-22 09:02 +0000 [7071e9d64c] George Joseph + + * media_index.c: Refactored so it doesn't cache the index + + Testing revealed that the cache added no benefit but that it could + consume excessive memory. + + Two new index related functions were created: + ast_sounds_get_index_for_file() and ast_media_index_update_for_file() + which restrict index updating to specific sound files. + + The original ast_sounds_get_index() and ast_media_index_update() + calls are still available but since they no longer cache the results + internally, developers should re-use an index they may already have + instead of calling ast_sounds_get_index() repeatedly. If information + for only a single file is needed, ast_sounds_get_index_for_file() + should be called instead of ast_sounds_get_index(). + + The media_index directory scan code was elimininated in favor of + using the existing ast_file_read_dirs() function. + + Since there's no more cache, ast_sounds_index_init now only + registers the sounds cli commands instead of generating the + initial index and subscribing to stasis format register/unregister + messages. + + "sounds" is no longer a valid target for the "module reload" + command. + + Both the sounds cli commands and the sounds ari resources were + refactored to only call ast_sounds_get_index() once per invocation + and to use ast_sounds_get_index_for_file() when a specific sound + file is requested. + + Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d + +2019-01-25 12:27 +0000 [0bcaadc037] Kevin Harwell + + * codecs.conf.sample: update codec opus docs + + The option value "sdp" for some of the settings was removed a while back, + however the sample conf was not updated. + + This patch removes any wording with regards to the old "sdp" option value, + and adjusts the defaults to what they are now. + + ASTERISK-28263 + + Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445 + +2019-01-22 09:24 +0000 [aede739778] eyalhasson + + * format_g726: add support for seeking + + Added support for the seek function in format_g726 + so playback can start from anywhere. + Before the fix, playback of g726 files + always started from the beginning. + + ASTERISK-28246 + + Change-Id: I626235bc4642df1479050d3d06828412603a9b40 + +2019-01-23 04:45 +0000 [69e9fd63e1] Jeremy Lainé + + * res_http_websocket: ensure control frames do not interfere with data + + Control frames (PING / PONG / CLOSE) can be received in the middle of a + fragmented message. In order to ensure they do not interfere with the + reassembly buffer, we exit early and do not return the payload to the + caller. + + ASTERISK-28257 #close + + Change-Id: Ia5367144fe08ac6141bba3309517a48ec7f013bc + +2019-01-23 07:59 +0000 [d9fae4a824] Jean Aunis + + * build : Fix cross-compilation errors + + Bundled pjproject and jansson must be configured with the host and build + parameters provided to the configure script. + Autotools do not permit to check for the existence of local header files, so + the control of hrirs.h must not be done when cross-compiling. + + ASTERISK-28250 + + Change-Id: If0a76e52a87d4ab82b7d4c72d27d8759ca931880 + +2019-01-22 15:03 +0000 [f9ca0afb39] Gerald Schnabel + + * manager_channels: Fix throwing of HangupHandler manager events + + The type value extracted from stasis message data in channel_hangup_handler_cb + isn't compared against the valid values "run", "pop" and "push". Thus the + manager events HangupHandlerPush, HangupHandlerPop and HangupHandlerRun are + never thrown. + + This regression was introduced by ASTERISK_21462. + + ASTERISK-28252 + + Change-Id: I9956e35e18da1873113644df1ddc3c7cd37bf524 + +2019-01-19 15:55 +0000 [1c8378bbc9] Chris-Savinovich + + * Test_cel: Fails when DONT_OPTIMIZE is off + + A bug in GCC causes TEST_CEL to return failure under the following + conditions: + 1. TEST_FRAMEWORK on + 2. DONT_OPTIMIZE off + 3. Fedora and Ubuntu + 4. GCC 8.2.1 + 5. Test name: test_cel_dial_pickup + 6. There must exist a certain combination of multithreading. + The bug affects arithmetic calculations when the optimization level + is bigger than O1 and the -fpartial-inline flag is on. Provided these + conditions, function ast_str_to_lower() fails to convert to lower case + due to said function being of type force_inline. The solution is to + remove the "force_inline" type declaration from function ast_str_to_lower() + + Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7 + +2018-12-10 07:20 +0000 [c6980e32ae] George Joseph + + * app_voicemail: Add Mailbox Aliases + + You can now define an "aliases" context in voicemail.conf + whose entries point to actual mailboxes. These can be used anywhere + the mailbox is specified. + + Example: + [general] + aliasescontext = myaliases + + [default] + 1234 = yadayada + + [myaliases] + 4321@devices = 1234@default + + Now you can use 4321@devices to refer to the 1234@default mailbox. + + This can be useful to provide channel drivers with constant + mailbox specifications such as @devices leaving + app_voicemail to control exactly which mailbox the alias points to. + Now, only voicemail has to be reloaded to make changes instead of + individual channel drivers which are usually more expensive to + reload. + + Change-Id: I395b9205c91523a334fe971be0d1de4522067b04 + +2019-01-22 12:07 +0000 [b82d2856b4] Kevin Harwell + + * res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown + + When a reliable transport is shutdown it's possible for the pjsip registrar + resource shutdown handler to get called multiple times. If this happens and one + of the threads is taking "too long" (slow database call for instance) then the + others get blocked waiting to delete. + + Since it only takes one to delete the contact then the other threads should be + able to continue on if one of the threads is currently "deleting". This patch + makes it so now when a thread enters the shutdown handler it checks to see if a + thread is currently already "deleting". If so, then the thread does not attempt + to get the lock, and instead continues on thus avoiding the blockage. + + ASTERISK-28213 #close + + Change-Id: I7563ca596312b1dff4f3ab41483e89fe2862328a + +2019-01-22 09:02 +0000 [deffb8a6e0] George Joseph + + * pjproject_bundled: Add patch for double free issue in timer heap + + Fixed #2172: Avoid double reference counter decrements in + timer in the scenario of race condition between + pj_timer_heap_cancel() and pj_timer_heap_poll(). + + Change-Id: If000e9438c83ac5084b678eb811e902c035bd2d8 + +2018-12-16 06:43 +0000 [a526676836] Xiemin Chen + + * bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix + + To avoid the stream name collide if there're more than one video track + in one client. If client has multi video tracks, the name of ast_stream + which represents each video track may be the same. Use the MSID:LABEL + here because it's identifiable. + + ASTERISK-28196 #close + Reported-by: xiemchen + + Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b + +2019-01-08 01:38 +0000 [0b8867f7d6] Jeremy Lainé + + * res_http_websocket: respond to CLOSE opcode + + This ensures that Asterisk responds properly to frames received from a + client with opcode 8 (CLOSE) by echoing back the status code in its own + CLOSE frame. + + Handling of the CLOSE opcode is moved up with the rest of the opcodes so + that unmasking gets applied. The payload is no longer returned to the + caller, but neither ARI nor the chan_sip nor pjsip made use of the + payload, which is a good thing since it was masked. + + ASTERISK-28231 #close + + Change-Id: Icb1b60205fc77ee970ddc91d1f545671781344cf + +2019-01-18 16:11 +0000 [20f672539e] Sean Bright + + * pjsip_transport_management: Shutdown transport immediately on disconnect + + The transport management code that checks for idle connections keeps a + reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by + default). Because of this, if the transport is closed before this + timeout, the idle checking code will keep the transport from actually + being shutdown until the timeout expires. + + Rather than passing the AO2 object to the scheduler task, we just pass + its key and look it up when it is time to potentially close the idle + connection. The other transport management code handles cleaning up + everything else for us. + + Additionally, because we use the address of the transport when + generating its name, we concatenate an incrementing ID to the end of the + name to guarantee uniqueness. + + Related to ASTERISK~28231 + + Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb + +2019-01-20 12:15 +0000 [17f76d27cc] Valentin Vidic + + * channel.c: Fix segfault with Monitor(wav,file,i) + + If the Monitor is started with the i option the read_stream will be + NULL. One code path in channel.c checks if write_stream is set but than + uses read_stream instead causing a segfault. + + ASTERISK-28249 + + Change-Id: I1bae9126537be54895c7fea2d08dd9488d8cc525 + +2019-01-10 13:34 +0000 [1323730f6c] Joshua C. Colp + + * stasis / manager / ari: Better filter messages. + + Previously both AMI and ARI used a default route on + their stasis message router to handle some of the + messages for publishing out their respective + connection. This caused messages to be given to + their subscription that could not be formatted + into AMI or JSON. + + This change adds an API call to the stasis message + router which allows a default route to be set as well + as formatters that the default route is expecting. + This allows both AMI and ARI to specify that their + default route only wants messages of their given + formatter. By doing so stasis can more intelligently + filter at publishing time so that they do not receive + messages which will not be turned into AMI or JSON. + + ASTERISK-28244 + + Change-Id: I65272819a53ce99f869181d1d370da559a7d1703 + +2019-01-17 09:56 +0000 [58b55f2a30] Sean Bright + + * sched: Make sched_settime() return void because it cannot fail + + Change-Id: I66b8b2b2778f186919d73ae9bf592104b8fb1cd5 + +2019-01-04 17:14 +0000 [2b8602e8cf] Sean Bright + + * res_pjsip_transport_websocket: Don't assert on 0 length payloads + + When --enable-dev-mode is used, pjsip_tpmgr_receive_packet() will assert + if passed a payload length of 0, so treat empty frames as if we didn't + receive them. + + Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48 + +2019-01-12 02:29 +0000 [d60ee2eeae] Mohit Dhiman + + * stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure. + + During Bridging of two channels if masquerade operation is performed on a + channel (clone channel) which was created with endpoint details + (ast_channel_alloc_with_endpoint()) and the original channel which is created + without endpoint details (ast_channel_alloc()) then both the channels must + exchange their endpoint details or else after masquerade when clone channel + is being destroyed the endpoint cleanup callbacks will be destroyed too and + after call completion unique_id of original channel will still be there in + ast_endpoint structure's channel_ids container. + + ASTERISK-28197 + + Change-Id: I97ce73da390af20fd082fb09d722a6fe9cb2f39d + +2019-01-11 09:48 +0000 [f0546d1d87] Alexei Gradinari + + * res_pjsip: add option to enable ContactStatus event when contact is updated + + The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out + the ContactStatus AMI event when a contact is updated. + Thist change broke things which rely on old behavior. + + This patch adds a new PJSIP global configuration option + 'send_contact_status_on_update_registration' to be able to preserve old + ContactStatus behavior. + By default new behavior, i.e. the ContactStatus event will not be sent when a + device refreshes its registration. + + Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46 + +2019-01-07 08:06 +0000 [18e206381a] Joshua Colp + + * res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled. + + For video streams it was possible for the abs-send-time information + to be placed into RTP streams even if not negotiated. Depending on + the endpoint in use this could cause video to not flow. + + We now only enable abs-send-time for negotiation if WebRTC is enabled. + + ASTERISK-28230 + + Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c + +2019-01-05 11:14 +0000 [7bd30905fd] Diederik de Groot + + * RAII: Change order or variables in clang version + + This prevents use-after-scope issues when unwinding the stack, + which happens in reverse order. The varname variable needs to + remain alive for the destruction to be able to access it. + Issue was found using clang + address-sanitizer. + + ASTERISK-28232 #close + + Change-Id: I00811c34ae910836a5fb6d22304528aef92624db + +2019-01-04 09:57 +0000 [f662a26ea0] Alexei Gradinari + + * RTP: reset DTMF last seqno/timestamp on RTP renegotiation + + The remote side may start a new stream when renegotiating RTP. + Need to reset the DTMF last sequence number and the timestamp + of the last END packet on RTP renegotiation. + + If the new time stamp is lower then the timestamp of the last DTMF END packet + the asterisk drops all DTMF frames as out of order. + + This bug was caught using Cisco ip-phone SPA5XX and codec g722. + On SIP session update the SPA50X resets stream and a new timestamp is twice + smaller then the previous. + + ASTERISK-28162 #close + + Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254 + +2019-01-02 11:44 +0000 [2c48b5d9bf] Bryan Boatright + + * app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail + + If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is + not updated correctly since urgent messages are in a different directory. The + fix is to update the channel variable when the path to the urgent message is + created. + + ASTERISK-28225 + + Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca + +2019-01-02 11:33 +0000 [b7b080a0aa] Joshua Colp + + * app_queue: Fix crash when using 'b' option on non-ringall queue. + + When using the 'b' option to Queue with a queue that was not configured + for ring all a crash would occur as the wrong pointer would be used. + + ASTERISK-28218 + + Change-Id: If1390f64e321047dff24fd2410c95dde74904980 + +2018-12-19 13:02 +0000 [7c08ff51d7] Richard Mudgett + + * stasic.c: Fix printf format type mismatches with arguments. + + An int64_t is not likely the same size as a long. + + * Changed the int64_t values in the statistics structs to longs so casting + is not necessary when generating the formatted CLI output. The offending + members did not need to be int64_t anyway as they were only set by an int + type variable which was already truncating bits. + + * Reordered the statistics structs to reduce potential padding bytes. + + Change-Id: Ic090a070e9dc4ca650ebdb9c01ed50a581289962 + +2018-12-26 11:49 +0000 [110934706f] Corey Farrell + + * stasis: Fix ABI between DEVMODE and non-DEVMODE. + + Eliminate differences with DEVMODE prototypes for public functions. + + ASTERISK-28212 #close + + Change-Id: I872c04842ab6b61e9dd6d37e4166bc619aa20626 + +2018-12-26 10:26 +0000 [4c084c6b1b] George Joseph + + * Revert "stasis_cache: Stop caching stasis subscription change messages" + + This reverts commit 5ec6d2c33e3b02755e0b2ea3fc94f048af5c741f. + + This commit caused issues with polling when combined with + the revert commit "Revert "app_voicemail: Remove need to subscribe to stasis" + + ASTERISK-28222 + Reported by: abelbeck + + Change-Id: I1e83a433e4202574181bc128dce876ef24936a52 + +2018-12-24 11:42 +0000 [809e836265] George Joseph + + * ast_coredumper: Refactor the pid determination process + + In order to get a dump of the running process, we need to find the + pid of the main asterisk process. This can be tricky if there are + also instances of "asterisk -r" running or if an alternate location + for asterisk.conf was specified on the command line with the -C + option that also specified an alternation location for the pid file. + + So now... + + 1. We find the asterisk executable with "which" or the --asterisk-bin + command line option. + 2. If there's only 1 process with an executable path that matches, + we use that pid. If not... + 3. We try " -rx 'core show settings'" and parse the + output to find the pidfile, then read that for the pid. If that + didn't work... + 4. We get a list of all the pids matching and look + in /proc//cmdline for a -C argument and retry the "core show + settings" using the same -C option. We can't parse the output + of "ps" to get the -C path because it may contain spaces. The + contents of /proc//cmdline are delimited by NULLs. For BSDs + we may have to mount /proc first. :( + + ASTERISK-28221 + Reported by: Andrew Nagy + + Change-Id: I8aa1f3f912f949df2b5348908803c636bde1d57c + +2018-12-19 12:39 +0000 [314782e874] Richard Mudgett + + * backtrace.c: Fix casting pointer to/from integral type. + + The backtrace library bfd.h include file does not get the sizes of + pointers and ints right on some platforms. On my old test box the size + of bfd_vma is 8 while the size of a pointer is 4. gcc on the box + complains of the integer casting to/from pointers size mismatch. + + * uintptr_t to the rescue by doing an appropriate two stage cast. + + Change-Id: Icb2621583f50c8728de08a3c824d95fe53cc45d0 + +2018-12-18 10:33 +0000 [c23c8d92d5] George Joseph + + * app_voicemail: Don't delete mailbox state unless mailbox is deleted + + The free_user function was automatically deleting the stasis mailbox + state but this only makes sense when the mailbox is actually + deleted, not just the structure freed. This was causing issues + where leave_voicemail would publish the mwi message to stasis and + delete the state before the message could be processed by + res_pjsip_mwi. + + * Removed the delete of state from free_user(). + + * Created a new free_user_final() function that both frees the data + structure and deletes the state. This function is only called + during module load/unload where it's appropriate to delete the + state. + + ASTERISK-28215 + + Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd + +2018-12-14 11:52 +0000 [357219dfb3] Sean Bright + + * res_rtp_asterisk: Remove some unused structure fields. + + All of the fields that were removed were no longer referenced except for + 'lastrxts' and 'rxseqno' which were only ever written to. + + Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c + +2018-12-13 15:56 +0000 [5b12dfa6dd] Sean Bright + + * res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set + + The profile-iop octet (the 2nd) of profile-level-id can be zero + according to RFC 6184 Section 8.1. So we ignore its value when deciding + to include profile-level-id in the outgoing SDP. + + ASTERISK-27959 #close + Reported by: David Kuehling + + Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73 + +2018-12-11 14:49 +0000 [3db1df301e] Sean Bright + + * bridge_builtin_features.c: Set auto(mix)mon variables on both channels + + This is how features behaved up through Asterisk 11, but was changed + when the new bridging framework was implemented in Asterisk 12. + + Reported by rrittgarn in #asterisk. + + Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b + +2018-12-07 14:22 +0000 [cb1a08bdcb] Alexei Gradinari + + * confbridge: announce to the marked users when they join an empty conference + + Currently the file sound_only_person is not played when a marked + user (with announce_only_user=yes) joins an empty conference. + + This patch fixes it. + + ASTERISK-28201 #close + + Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4 + +2018-11-30 05:40 +0000 [fe07093660] Joshua C. Colp + + * stasis: Add statistics gathering in developer mode. + + This change adds statistics gathering to Stasis topics, + subscriptions, and message types. These can be viewed using + CLI commands and provide insight into how Stasis is used + and how long certain operations take to execute. + + These are only available when Asterisk is compiled in + developer mode and do not have any impact under normal + operation. + + ASTERISK-28117 + + Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f + +2018-12-11 08:54 +0000 [42ff856216] Sean Bright + + * Use non-blocking socket() and pipe() wrappers + + Change-Id: I050ceffe5a133d5add2dab46687209813d58f597 + +2018-12-11 09:06 +0000 [bedf16b041] Sean Bright + + * utils: Don't set or clear flags that don't need setting or clearing + + Change-Id: I0e7fb507ac09b15e45e1ff8501ecfca67afa5217 + +2018-12-11 06:55 +0000 [00b36bb045] Sean Bright + + * build: Update config.guess and config.sub + + Pulled from the authoritative respository at: + + https://git.savannah.gnu.org/cgit/config.git/tree/ + + Change-Id: I748708ce24d4d47ff1f395075d0b08d3da3355e0 + +2018-12-11 08:28 +0000 [d1598dbc7d] George Joseph + + * Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit" + + This reverts commit 3f53041267234b21aedd522c1197ec57cca90845. + + Pending resolution of ASTERISK_28200 + + Change-Id: Iad4f3614cac95b00fdbb2b799aab8ae6285ec988 + +2018-12-06 11:23 +0000 [a24bb1c4b6] Sebastian Damm + + * res/res_ari: Add additional hangup reasons + + The ARI DELETE /channels command takes a "reason" parameter + Previously, there were only five reasons implemented + This patch adds more reasons to choose from for more + complex setups + + ASTERISK-28198 #close + + Change-Id: I85996f1076c9946d65c778413f040a845a90fecc + +2018-12-07 06:57 +0000 [6d69fb3cc2] Sean Bright + + * utils: Wrap socket() and pipe() to reduce syscalls + + Some platforms provide an implementation of socket() and pipe2() that allow the + caller to specify that the resulting file descriptors should be non-blocking. + + Using these allows us to potentially elide 3 calls into 1 by avoiding extraneous + calls to fcntl() to set the O_NONBLOCK flag afterwards. + + In passing, change ast_alertpipe_init() to use pipe2() directly instead of the + wrapper if it is available. + + Change-Id: I3ebe654fb549587537161506c6c950f4ab298bb0 + +2018-11-29 09:53 +0000 [3f3dd992a2] George Joseph + + * stasis: Allow filtering by formatter + + A subscriber can now indicate that it only wants messages + that have formatters of a specific type. For instance, + manager can indicate that it only wants messages that have a + "to_ami" formatter. You can combine this with the existing + filter for message type to get only messages with specific + formatters or messages of specific types. + + ASTERISK-28186 + + Change-Id: Ifdb7a222a73b6b56c6bb9e4ee93dc8a394a5494c + +2018-12-05 15:28 +0000 [b899119a5d] David M. Lee + + * Removing registrar_expire from basic-pbx config + + The module has been removed, so it shouldn't be in the default config any more. + + Change-Id: Ie7e09f00f9c9a885574e29478250de4c2cefd9f1 + +2018-12-04 18:00 +0000 [0bde3751a0] Giuseppe Sucameli + + * chan_sip: Fix leak using contact ACL + + Free old peer's contactacl before overwrite it within build_peer. + + ASTERISK-28194 + + Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c + +2018-12-05 09:37 +0000 [19c4e0f592] George Joseph + + * CI: Various updates to buildAsterisk.sh + + * Added ---no-configure, --no-menuselect, --no-make and --no-alembic + options that prevent those actions from being performed. Useful + for testing and re-running portions of the build after fixing + earlier failures. + + * Added "set -e" to abort the script on command failure. + Not sure why this wasn't there in the first place. + + * Fixed a few echos that were redirecting to stderr when they shouldn't + have been. + + * Catch more alembic failures by actually trying to generate the SQL. + + Change-Id: I9f395fa4e9254be7299e7c1014f1a13db78faffb + +2018-12-03 17:45 +0000 [cbb7633ad3] Kevin Harwell + + * pjsip_add_use_callerid_contact: fixed alembic script + + Change-Id: I413f1583c797fb79651786cd8d0b003599f8ed10 + +2018-12-03 16:41 +0000 [8f5df046f6] Sean Bright + + * core: Add some documentation to the malloc_trim code + + This adds documentation to handle_cli_malloc_trim() indicating how it + can be useful when debugging OOM conditions. + + Change-Id: I1936185e78035bf123cd5e097b793a55eeebdc78 + +2018-12-03 14:01 +0000 [58e50e56cb] Chris-Savinovich + + * core: Merge malloc_trim patch + + We've had multiple opportunities where Richard Mudgett's + malloc_trim patch has been useful. Let's get it + pushed up to gerrit and merged. + + Since malloc_trim is only available in libc, an entry is + added to configure.ac to create a definition for + HAVE_MALLOC_TRIM. + + Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c + +2018-11-11 10:29 +0000 [8644511cbf] Sungtae Kim + + * res_pjsip: Patch for res_pjsip_* module load/reload crash + + The session_supplements for the pjsip makes crashes when the module + load/unload. + + ASTERISK-28157 + + Change-Id: I5b82be3a75d702cf1933d8d1417f44aa10ad1029 + +2018-10-22 07:47 +0000 [140702ba2d] lvl + + * app_queue: Revert broken queue channel reference patch + + Revert commit 6409e7b11a2310196a9978b30a6b79e2760be592, and add + NULL checks for all app_queue event handling code. + + Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844 + + ASTERISK-28125 + + Change-Id: I37334ea184ebb56e54471496b82937d4927815a0 + +2018-11-30 14:00 +0000 [6c13b20803] Chris-Savinovich + + * test_websocket_client.c: Disable websocket_client_create_and_connect test. + + This test was occasionally failing, with: + + WARNING[5812]: http.c:1939 httpd_helper_thread: Failed to set + TCP_NODELAY on HTTP connection: Bad file descriptor + ERROR[5812]: iostream.c:91 ast_iostream_nonblock: Failed to get + fcntl() flags for file descriptor: Bad file descriptor + ERROR[5812]: iostream.c:569 ast_iostream_close: close() failed: Bad + file descriptor + + Disabled for now by making the test explicit only. + + Change-Id: I778f6cbb6104c6b4e89737a2eaf1a9540888d351 + +2018-11-28 01:14 +0000 [ecb9ed0958] Pirmin Walthert + + * pjproject_bundled: check whether UPDATE is supported on outgoing calls + + In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not + trying to send UPDATE messages when connected_line_method was set to invite. + However this only solved the issue for incoming INVITES. For outgoing INVITES + (important when transferring calls) the options variable needs to be updated + at a different place. + + ASTERISK-28182 #close + Reported-by: nappsoft + + Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29 + +2018-11-29 13:26 +0000 [4f0bf0270e] George Joseph + + * Revert "app_voicemail: Remove need to subscribe to stasis" + + This reverts commit 29115e23848cceee0e2763bc70e87cb311919cdd. + + That commit closed a long standing hole which allowed subscriptions + to mailboxes that weren't configured in voicemail.conf. This + caused an issue with FreePBX which depdended on that behavior. + The commit is being reverted until FreePBX can handle the new + behavior. + + ASTERISK-28151 + Reported by: Ronald Raikes + + Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15 + +2018-11-26 16:18 +0000 [f4924d40db] George Joseph + + * test_cel: Plug a few ref leaks + + These are only a few of the leaks. The large number of macros + and return paths in this file would make a weeks worth of work + to plug them all. + + Change-Id: Ie2369fa944023d44767871c5c30974cb077ffb56 + +2018-09-19 14:34 +0000 [3667c5e1d2] George Joseph + + * bridges: Remove reliance on stasis caching + + * The bridging core no longer uses the stasis cache for bridge + snapshots. The latest bridge snapshot is now stored on the + ast_bridge structure itself. + + * The following APIs are no longer available since the stasis cache + is no longer used: + ast_bridge_topic_cached() + ast_bridge_topic_all_cached() + + * A topic pool is now used for individual bridge topics. + + * The ast_bridge_cache() function was removed since there's no + longer a separate container of snapshots. + + * A new function "ast_bridges()" was created to retrieve the + container of all bridges. Users formerly calling + ast_bridge_cache() can use the new function to iterate over + bridges and retrieve the latest snapshot directly from the + bridge. + + * The ast_bridge_snapshot_get_latest() function was renamed to + ast_bridge_get_snapshot_by_uniqueid(). + + * A new function "ast_bridge_get_snapshot()" was created to retrieve + the bridge snapshot directly from the bridge structure. + + * The ast_bridge_topic_all() function now returns a normal topic + not a cached one so you can't use stasis cache functions on it + either. + + * The ast_bridge_snapshot_type() stasis message now has the + ast_bridge_snapshot_update structure as it's data. It contains + the last snapshot and the new one. + + * cdr, cel, manager and ari have been updated to use the new + arrangement. + + Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369 + +2018-11-07 11:18 +0000 [50ac85cb40] Joshua Colp + + * stasis: Segment channel snapshot to reduce creation cost. + + When a channel snapshot was created it used to be done + from scratch, copying all data (many strings). This incurs + a cost when doing so. + + This change segments the channel snapshot into different + components which can be reused if unchanged from the + previous snapshot creation, reducing the cost. In normal + cases this results in some pointers being copied with + reference count being bumped, some integers being set, + and a string or two copied. The other benefit is that it + is now possible to determine if a channel snapshot update + is redundant and thus stop it before a message is published + to stasis. + + The specific segments in the channel snapshot were split up + based on whether they are changed together, how often they + are changed, and their general grouping. In practice only + 1 (or 0) of the segments actually get changed in normal + operation. + + Invalidation is done by setting a flag on the channel when + the segment source is changed, forcing creation of a new + segment when the channel snapshot is created. + + ASTERISK-28119 + + Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423 + +2018-10-10 09:28 +0000 [d0ccbb3377] Joshua Colp + + * stasis: Use an implementation specific channel snapshot cache. + + Channels no longer use the Stasis cache for channel snapshots. Instead + they are stored in a hash table in stasis_channels which reduces the + number of Stasis messages created and allows better storage. + + As a result the following APIs are no longer available since the stasis + cache is no longer used: + ast_channel_topic_cached() + ast_channel_topic_all_cached() + + The ast_channel_cache_all() and ast_channel_cache_by_name() functions + now return an ao2_container of ast_channel_snapshots rather than + a container of stasis_messages therefore you can't (and don't need + to) call stasis_cache functions on it. + + The ast_channel_topic_all() function now returns a normal topic not + a cached one so you can't use stasis cache functions on it either. + + The ast_channel_snapshot_type() stasis message now has the + ast_channel_snapshot_update structure as it's data. It contains the + last snapshot and the new one. + + ast_channel_snapshot_get_latest() still returns the latest snapshot. + + The latest snapshot is now stored on the channel itself to eliminate + cache hits when Stasis messages that have the snapshot as a payload + are created. + + ASTERISK-28102 + + Change-Id: I9334febff60a82d7c39703e49059fa3a68825786 + +2018-11-26 06:09 +0000 [8e1ab4f11c] Corey Farrell + + * jansson: Upgrade to 2.12. + + This brings in jansson-2.12, removes all patches that were merged + upstream. README is created in third-party/jansson/patches to explain + how to add patches but also because the patches folder must exist for + the build process to succeed. + + Change-Id: If0f2d541c50997690660c21fb7b03d625a5cdadd + +2018-11-23 09:40 +0000 [3f53041267] Alexei Gradinari + + * RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit + + The marker bit set on the voice packet indicates the start + of a new stream and a new time stamp. + Need to reset the DTMF last sequence number and the timestamp + of the last END packet. + + If the new time stamp is lower then the timestamp of the last DTMF END packet + the asterisk drops all DTMF frames as out of order. + + This bug was caught using Cisco ip-phone SPA50X and codec g722. + On SIP session update the SPA50X resets stream indicating it with market bit + and a new timestamp is twice smaller then the previous. + + ASTERISK-28162 #close + + Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620 + +2018-11-19 14:10 +0000 [021ce938ca] Corey Farrell + + * astobj2: Remove legacy ao2_container_alloc routine. + + Replace usage of ao2_container_alloc with ao2_container_alloc_hash or + ao2_container_alloc_list. Remove ao2_container_alloc macro. + + Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088 + +2018-11-14 05:02 +0000 [bc7f4f4db3] Corey Farrell + + * astobj2: Create function to copy weak proxied objects from container. + + Create ao2_container_dup_weakproxy_objs to perform a similar function to + ao2_container_dup. This function expects the source container to have + weakproxy objects, inserts the associated non-weak objects into the + destination container. Orphaned weakproxy objects are ignored. + + Create test for this new function and for ao2_weakproxy_find. + + Change-Id: I898387f058057e08696fe9070f8cd94ef3a27482 + +2018-11-16 14:45 +0000 [4b5d11ec17] Michael Walton (license 6502) + + * func_strings: HASHKEY - negative array index can cause corruption + + This patch makes it so only matching non-empty key names, and keys created by + the HASH function are eligible for inclusion in the comma separated string. It + also fixes a bug where it was possible to write to a negative index if the + result buffer was empty. + + ASTERISK-28159 + patches: + ASTERISK-28159.diff submitted by Michael Walton (license 6502) + + Change-Id: I6e57fe7307dfd856271753aed5ba64c59b511487 + +2018-11-19 11:59 +0000 [bcdfb90362] George Joseph + + * CI: Get job timeouts from environment + + The job timeouts were hard coded in the jenkinsfiles which + means changes had to go through gerrit. Now they are taken + from the following environment variables (and their defaults) that + can be set in Jenkins configuration... + + TIMEOUT_GATES = "60 MINUTES" + TIMEOUT_DAILIES = "3 HOURS" + TIMEOUT_REF_DEBUG = "24 HOURS" + TIMEOUT_UNITTESTS = "30 MINUTES" + + Change-Id: I673a551c1780bf665a3bc160b245da574aa4bbab + +2018-11-19 07:00 +0000 [64e21c9ea9] Corey Farrell + + * app_queue: Cleanup queue_ref / queue_unref routines. + + This replaces the inline functions with macros. This removes the need + to directly use __ao2_ref, opts instead for standard ao2_bump and + ao2_cleanup macros. + + Change-Id: If4e04e9bab2e3c883188437cb9f487b3e498a21b + +2018-11-08 09:53 +0000 [ece5f8015f] George Joseph + + * backtrace: Refactor ast_bt_get_symbols so it doesn't crash + + We've been seeing crashes in libbfd when we attempt to generate + a stack trace from multiple threads. It turns out that libbfd + is NOT thread-safe. It can cache the bfd structure and give it to + multiple threads without protecting itself. To get around this, + we've added a global mutex around the bfd functions and also have + refactored the use of those functions to be more efficient and + to provide more information about inlined functions. + + Also added a few more tests to test_pbx.c. One just calls + ast_assert() and the other calls ast_log_backtrace(). Neither are + run by default. + + WARNING: This change necessitated changing the return value of + ast_bt_get_symbols() from an array of strings to a VECTOR of + strings. However, the use of this function outside Asterisk is not + likely. + + ASTERISK-28140 + + Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621 + +2018-11-18 17:53 +0000 [56eb18f395] Joshua C. Colp + + * stasis: Remove stringfields and lock from change message. + + When a subscribe or unsubscribe occurs a message is published + containing this information. This change makes it so that the + message no longer uses stringfields or a lock, as both are not + really needed for the message. + + Change-Id: I3f4831931d79f94fd979baf48048738df5dc1632 + +2018-11-13 09:28 +0000 [fa048183aa] Alexei Gradinari + + * pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI + + New dialplan function PJSIP_PARSE_URI added to parse an URI and return + a specified part of the URI. + + This is useful when need to get part of the URI instead of cutting it + using a CUT function. + + For example to get 'user' part of Remote URI + ${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)} + + ASTERISK-28144 #close + + Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a + +2018-09-23 15:50 +0000 [3077ad0c24] Joshua Colp + + * stasis: Add internal filtering of messages. + + This change adds the ability for subscriptions to indicate + which message types they are interested in accepting. By + doing so the filtering is done before being dispatched + to the subscriber, reducing the amount of work that has + to be done. + + This is optional and if a subscriber does not add + message types they wish to accept and set the subscription + to selective filtering the previous behavior is preserved + and they receive all messages. + + There is also the ability to explicitly force the reception + of all messages for cases such as AMI or ARI where a large + number of messages are expected that are then generically + converted into a different format. + + ASTERISK-28103 + + Change-Id: I99bee23895baa0a117985d51683f7963b77aa190 + +2018-11-18 10:38 +0000 [915b80709d] George Joseph + + * CI: Add tmpfs to all jenkinsfiles + + Change-Id: Ida29d70d48d5f39aabf0b25c66b51f79324a8cba + +2018-11-17 15:40 +0000 [f5e3832dff] George Joseph + + * CI: Mount a tmpfs on /tmp for testsuite docker containers + + Change-Id: I0566d81b0852f22066cd76d58eae5f1fda5602aa + (cherry picked from commit 73efe86436427e5f43c532e5d42505ab4ec104d9) + +2018-11-17 13:07 +0000 [be87185f6d] George Joseph + + * CI: Pass work directory to runTestsuite + + The testsuite can now use a user-specified work directory for + all it's temp files. This allows the docker containers to use + a tmpfs backed directory for the temp files instead of it's + own write-layer image. + + * runTestsuite.sh now accepts a --work-dir command line argument + that gets exported as AST_WORK_DIR before running the testsuite. + + * gates.jenkinsfile now specifies --work-dir to be + /astroot. + + Since the Asterisk CI docker hosts now mount /srv/jenkins/workspace + on a tmpfs, asterisk should be compiled and the testsuite run all in + memory. + + Change-Id: If5ee905a15821296c355bb84cda38950ad8edc45 + (cherry picked from commit a335f4c9adb0a00211345634f61917bdf5b412c2) + +2018-11-16 20:33 +0000 [1dea497454] Sungtae Kim + + * res/res_ari: Fix null endpoint handle + + The res_ari(POST /channels/create handler) deos not check the endpoint + parameter length. And it causes core + dump. + Fixed it to check the parameter length. Also fixed memory leak. + + ASTERISK-28169 + + Change-Id: Ibf10a9eb8a2e3a9ee1e13fbe748b2ecf955c3993 + +2018-11-15 11:41 +0000 [8ff3435c8a] George Joseph + + * CI: Allow runUnittests to use 'expect' to run the tests + + There seems to be a race condition between starting the asterisk + daemon and attempting to use 'asterisk -r' that can cause the + control socket file to not be created. Since all of the Jenkins + slaves have 'expect' installed, the runUnittests script can use + it to start asterisk in the forground and issue the commands + interactively. This is much more reliable and it can also make + startup errors more visible since they'll be in the Jenkins console + output. + + If 'expect' isn't installed, the original daemon/asterisk -r + process is used. + + Also added a "core show settings" before running the tests + and added "notice,warning,error" to the console log. + + Change-Id: Idd656085f854afede813ac241b9e312b31358160 + +2018-11-12 12:23 +0000 [9abd5e1004] Corey Farrell + + * taskprocessor: Prevent race creating new taskprocessor. + + Task processors are retrieved using a 'get or create' pattern. The + singleton container was unlocked between the get and create steps so + it's possible that two threads could create task processors with the + same name at the same time. + + Change-Id: Id64fae94a6a1e940ddf38fde622dcd4391635382 + +2018-11-16 06:20 +0000 [752fd06d12] Corey Farrell + + * pjproject-bundled: Use AST_DEVMODE for conditional compilation. + + We previously allowed resample and g711 codecs to be built when + TEST_FRAMEWORK was enabled. This could cause errors if the testsuite + was run without this option enabled. Switch the build system to allow + those codecs to be built when --enable-dev-mode is used. This removes a + chance for strange testsuite errors from use of an inadequate pjsua + binary. + + Change-Id: Iee8a3613cdb711fa7e7d217c5a775a575907ae22 + +2018-11-15 14:47 +0000 [02c7a061ea] Corey Farrell + + * res_pjsip_caller_id: Use static pj_str_t for fromto header names. + + PJSIP assumes that these header names are not allocated, does not clone + the name strings when reusing headers. + + Block unload of res_pjsip_caller_id until shutdown to ensure static + memory stays valid. It was previously unsafe to unload while any + sessions are active. + + Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f + +2018-10-24 07:38 +0000 [d0554783e2] Torrey Searle + + * res/res_pjsip_nat: Fix logic for REINVITES + + The presence of Record-Route in re-invites is optional, thus it is + important to make sure the dialog doesn't have a routset before + rewriting the contact header. + + ASTERISK-28129 #close + + Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc + +2018-11-15 05:33 +0000 [c3d7b19cdd] Corey Farrell + + * core: Fix handling of restart from remote console. + + We cannot use need_el_end and SIGURG when restarting. Instead we need + to run el_end within the SIGHUP restartnow handler. + + ASTERISK-28158 + + Change-Id: Ia852276363c81bdcf1aa29eb4558c5c2fa1218a0 + +2018-10-25 10:25 +0000 [eb5b83b8ea] Jan Hoffmann (license 6986) + + * AST-2018-010: Fix length of buffer needed for SRV and NAPTR results + + When dn_expand was being called on SRV and NAPTR results, the + return value was being used to calculate the size of the buffer + needed to store the host names. Since dn_expand returns the + length of the COMPRESSED name the buffer could be too short + to hold the EXPANDED name. The expanded name is NULL terminated + so using strlen() is the correct way to determine the length + actually needed for the buffer. + + ASTERISK-28127 + Reported by: Jan Hoffmann + + patches: + patch.diff submitted by janhoffmann (license 6986) + + Change-Id: I4d35d6c431c6c6836cb61d37b1378cc47f0b414d + +2018-11-13 10:51 +0000 [4b24731640] Corey Farrell + + * test_res_pjsip_scheduler: Fix possible write after free in scheduler_policy. + + It's possible for a 4th task to be spawned before we cancel. This + results in a write to the already freed test_data1. Wait long enough to + verify success of the cancelation before freeing test_data1. + + Change-Id: I057e2fcbe97f8a175e50890be89c28c20490a20f + +2018-10-17 08:48 +0000 [da562eb82d] Robert Cripps + + * bridge_native_rtp.c: Fail native bridge if no framing match. + + ASTERISK-28110 #close + + Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607 + +2018-11-11 18:32 +0000 [944d90a7ea] Corey Farrell + + * taskprocessor: Do not use separate allocation for stats or name. + + Merge storage for the stats object and name string into the main + allocation for struct ast_taskprocessor. + + Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1 + +2018-11-11 07:34 +0000 [194e40122a] Corey Farrell + + * core: Ensure that el_end is always run when needed. + + * Ignore console=yes configuration option in remote console processes. + * Use new flag to tell consolethread to run el_end and exit when needed. + + ASTERISK-28158 + + Change-Id: I9e23b31d4211417ddc88c6bbfd83ea4c9f3e5438 + +2018-11-08 15:37 +0000 [d9add7e086] Corey Farrell + + * jansson-bundled: Patch for off-nominal crash. + + pack_string crashed on non-NULL strings returned when s->has_error was + true if the string was the result of 's' format without '#', '%' or '+'. + + Change-Id: Ic125df691d81ba2cbc413e37bdae657b304d20d0 + +2018-11-02 06:38 +0000 [8e34cb302e] Corey Farrell + + * pbx_config: Only the first [globals] section is seen. + + If multiple [globals] sections are used (for example via separate + included files), only the first one is processed. This can result in + lost global variables when using a modular extensions.conf. + + ASTERISK-28146 #close + + Change-Id: Iaac810c0a7c4d9b1bf8989fcc041cdb910ef08a0 + +2018-11-06 16:44 +0000 [a3fc97aa13] Chris-Savinovich + + * res_pjsip: Send a 503 response when overload state if reliable transport. + + When Asterisk's taskprocessors get overloaded we need to reduce the work + load. res_pjsip currently ignores new SIP requests and relies on SIP + retransmissions in the hope that the overload condition will clear soon + enough to handle the retransmitted SIP request. + This change adds the following code after ast_taskprocessor_alert_get() + has returned TRUE: + 1- identifies transport type. If non-udp then send a 503 response + 2- if transport type is udp/udp6 then ignore, as before. + + Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836 + +2018-11-06 16:35 +0000 [fdca9cb64f] Kevin Harwell + + * res_pjsip: formatting error in documentation + + The use of a '|' in the "global/debug" synopsis documentation caused the + generated html table on the wiki to add an extra column that included the + text after the pipe. + + This patch replaces the pipe with a comma. + + ASTERISK-28150 + + Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719 + +2018-11-05 12:44 +0000 [5f3f707793] Alexei Gradinari + + * res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue + + The current round-robin method does not take the current taskprocessor + load into consideration when distributing requests. Using the least-size + method the request goes to the taskprocessor that is servicing the least + number of active tasks at the current time. + + Longer running tasks with the round-robin method can delay processing + tasks. + + * Change the algorithm from round-robin to least-size for picking the + PJSIP taskprocessor from the default serializer pool. + + Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd + +2018-11-05 08:30 +0000 [bf579222c4] Joshua Colp + + * stasis: Clarify lifetime of topics. + + As mentioned in the comment I've added in the code there is no + ability to unsubscribe all subscribers from a topic and explicitly + destroy it. This is not currently a problem as we have two types of + topics: + + Long lived topics which exist for the lifetime of the system. + Ephemeral topics which feed a long lived topic. + + In the case of the ephemeral topics there is no subscriber which does + not have its lifetime managed by the same entity that has created + the topic. This ensures that when the topic is being unreferenced the + subscribers are also unsubscribed and destroyed, allowing the topic + to ultimately be destroyed as well. + + Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a + +2018-10-09 07:44 +0000 [2cf5079205] Jasper Hafkenscheid + + * chan_sip: Attempt ast_do_pickup in handle_invite_replaces + + When a call pickup is performed using and invite with replaces header + the ast_do_pickup method is attempted and a PICKUP stasis message is sent. + + ASTERISK-28081 #close + Reported-by: Luit van Drongelen + + Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a + +2018-10-26 10:53 +0000 [ebff81e3a0] Pascal Cadotte Michaud + + * contrib/sip_to_pjsip: add a --quiet option to avoid prints + + Using the --quiet or -q option in conjonction with /dev/stdout as the output + file allow the output to be used as a valid configuration. + + Given a script that generates a valid sip.conf I can pipe the output of that + script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use + that piped command in my pjsip.conf using the `exec` command. + + ASTERISK-28136 + + Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d + +2018-10-31 07:53 +0000 [0c9e217c81] Joshua Colp + + * res_pjsip: Add XML documentation for "use_callerid_contact" + + ASTERISK-28087 + + Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4 + +2018-10-30 10:52 +0000 [c7528f16e6] Richard Mudgett + + * alembic: Fix use_callerid_contact option add script. + + ASTERISK-28087 + + Change-Id: I046d018015427d0916fab571b5a4f5367476f729 + +2018-10-22 11:49 +0000 [eee935983b] Alexei Gradinari + + * pjsip: new endpoint's options to control Connected Line updates + + This patch adds new options 'trust_connected_line' and 'send_connected_line' + to the endpoint. + + The option 'trust_connected_line' is to control if connected line updates + are accepted from this endpoint. + + The option 'send_connected_line' is to control if connected line updates + can be sent to this endpoint. + + The default value is 'yes' for both options. + + Change-Id: I16af967815efd904597ec2f033337e4333d097cd + +2018-10-27 09:59 +0000 [b0155f7e58] Pascal Cadotte Michaud + + * contrib/sip_to_pjsip: handle setvar in conversion + + Given a sip.conf with the following content: + + setvar FOO=1 + setvar BAR=42 + + I want my generated pjsip.conf to containt the following set_vars + + set_var FOO=1 + set_var BAR=42 + + in the matching endpoint section. + + Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26 + +2018-10-26 16:18 +0000 [e407b8af21] Alexei Gradinari + + * res_pjsip_notify: improve realtime performance on CLI completion on the endpoint + + The module 'res_pjsip_notify' inefficiently makes a lot of DB requests + on CLI completion on the endpoint. + + For example if there are 10k endpoints the module makes 10k requests + of these 10k records. + + Even if a part of the endpoint entered + the module makes the same 10k requests and then filtered them by itself. + + This patch gathers endpoints container by prefix + and adds all gathered endpoints to completion at once. + + ASTERISK-28137 #close + + Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b + +2018-10-02 07:31 +0000 [cac4ccef25] Torrey Searle + + * res_pjsip_session: add new flag use_callerid_contact + + Add a new global flag to res_pjsip to allow the callerid to be used + as the username in the contact header. This allows chan_pjsip to have + the same behavour as chan_sip + + ASTERISK-28087 #close + + Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95 + +2018-10-10 07:09 +0000 [90a11c4ae7] Corey Farrell + + * chan_sip deprecation. + + This officially deprecates chan_sip in Asterisk 17+. A warning is + printed upon startup or module load to tell users that they should + consider migrating. chan_sip is still built by default but the default + modules.conf skips loading it at startup. + + Very important to note we are not scheduling a time where chan_sip will + be removed. The goal of this change is to accurately inform end users + of the current state of chan_sip and encourage movement to the fully + supported chan_pjsip. + + Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93 + +2018-10-25 07:54 +0000 [e81d33e78f] Corey Farrell + + * UPDATE.txt: Fix formatting to match previous files. + + Add 'Section:' headings and use '-' for bullet points. + + Change-Id: I7e2be35601ac8fea53b90d926da564512b6716e4 + +2018-10-18 14:51 +0000 [79c2b4fddd] Sean Bright + + * res_parking: Stop setting the deprecated PARKINGSLOT channel variable. + + Change-Id: Ia155ce2a53d61556aa4685524d1b48cfacfa3a8b + +2018-10-17 19:34 +0000 [1b397ebd00] Richard Mudgett + + * logger.c: Fix default console logging when no logger.conf available. + + Default logging was not setup correctly when there was no logger.conf. + This resulted in many expected log messages not actually getting out to + the console. + + Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c + +2018-09-26 15:05 +0000 [4a567cee3a] Alexei Gradinari + + * app_dial/queue/followme: 'I' options to block initial updates in both directions + + The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates + from the called parties to the caller. + + This patch also blocks updates in the other direction before call is + answered. + + ASTERISK-27980 + + Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01 + +2018-10-22 14:31 +0000 [96d5e444f0] Richard Mudgett + + * modules.conf.sample: Update preload usage documentation. + + Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90 + +2018-10-16 07:02 +0000 [8d1c6bb6e6] George Joseph + + * bridge_softmix: Add SDP "label" attribute to streams + + Adding the "label" attribute used for participant info correlation + was previously done in app_confbridge but it wasn't working + correctly because it didn't have knowledge about which video + streams belonged to which channel. Only bridge_softmix has that + data so now it's set when the bridge topology is changed. + + ASTERISK-28107 + + Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499 + +2018-10-18 14:24 +0000 [056ca07449] Sean Bright + + * func_callerid: Remove deprecated CALLERPRES() function. + + Change-Id: Ia1b2b386505b3102136dab02c45eaaf09f0f89c5 + +2018-07-18 07:45 +0000 [37b2e68628] Nick French + + * res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability + + This change implements a few different generic things which were brought + on by Google Voice SIP. + + 1. The concept of flow transports have been introduced. These are + configurable transports in pjsip.conf which can be used to reference a + flow of signaling to a target. These have runtime configuration that can + be changed by the signaling itself (such as Service-Routes and + P-Preferred-Identity). When used these guarantee an individual connection + (in the case of TCP or TLS) even if multiple flow transports exist to the + same target. + + 2. Service-Routes (RFC 3608) support has been added to the outbound + registration module which when received will be stored on the flow + transport and used for requests referencing it. + + 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been + added to the outbound registration module. If a P-Associated-URI header + is received it will be used on requests as the P-Preferred-Identity. + + 4. Configurable outbound extension support has been added to the outbound + registration module. When set the extension will be placed in the + Supported header. + + 5. Header parameters can now be configured on an outbound registration + which will be placed in the Contact header. + + 6. Google specific OAuth / Bearer token authentication + (draft-ietf-sipcore-sip-authn-02) has been added to the outbound + registration module. + + All functionality changes are controlled by pjsip.conf configuration + options and do not affect non-configured pjsip endpoints otherwise. + + ASTERISK-27971 #close + + Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58 + +2018-10-23 07:37 +0000 [f940b7b63d] Sean Bright + + * say: Remove legacy language deprecation logic + + These language codes (tw, ge, mx, and cz) were deprecated in 1.6.2. + + Change-Id: I18e4d2af2e83556fa91e39a7338030583ef05d50 + +2018-10-18 14:39 +0000 [9e8d671658] Sean Bright + + * res_xmpp: Remove deprecated JabberStatus application. + + Change-Id: I1a00ca22d59d6b6d2166aa56f0e9338a33e5ac60 + +2018-10-16 14:11 +0000 [687ab7aeee] Corey Farrell + + * astobj2: Eliminate legacy container allocation macros. + + These macros have been documented as legacy for a long time but are + still used in new code because they exist. Remove all references to: + * ao2_container_alloc_options + * ao2_t_container_alloc_options + * ao2_t_container_alloc + + These macro's are also removed. Only ao2_container_alloc remains due to + it's use in over 100 places. + + Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a + +2018-09-28 13:31 +0000 [4c19b94968] Corey Farrell + + * lock: Replace __ast_mutex_logger with private log_mutex_error. + + __ast_mutex_logger used the variable `canlog` without accepting it as a + argument. Replace with internal macro `log_mutex_error` which takes + canlog as the first arguement. This will prevent confusion when working + with lock.c code, many of the function declare the canlog variable and + in some cases it previously appeared to be unused. + + Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853 + +2018-10-18 14:36 +0000 [9838a5e57a] Richard Mudgett + + * app_dial/app_queue: Update application option documentation + + * Update the post-answer documentation and example. The Dial example was + incorrect and misleading for the post-answer subroutine useage. + + * Fix note and warning paragraphs in option descriptions. They don't show + up in the wiki. + + Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14 + +2018-10-18 14:56 +0000 [90bd8371f2] Sean Bright + + * samples: PARKINGSLOT -> PARKING_SPACE in parking sample config + + PARKINGSLOT was deprecated in Asterisk 12 but the sample config was not + updated to reflect that. + + Change-Id: I3e087c19d9ee587094fa5304102d8084a79c2b3c + +2018-10-18 14:17 +0000 [be04a64c49] Sean Bright + + * options.c: Remove 'internal_timing' notice + + Change-Id: I9882394617724a497df1d6f529a87965191be3ce + +2018-10-18 12:32 +0000 [467f7c6724] Richard Mudgett + + * Fix 'statement' typo throughout code. + + Most were in comments. A couple were in warning messages. + + Pointed out by Jonathan H on the Asterisk users mailing list. + + Change-Id: I6286939dff5d0a27a2758140570106f1cb351855 + +2018-10-17 16:08 +0000 [7ab4befc2b] Richard Mudgett + + * res_rtp_asterisk.c: Add conditional module dependency to res_pjproject + + * The dependency ensures that res_pjproject cannot be manually unloaded + before res_rtp_asterisk. + * The dependency allows startup loading errors to report that + res_rtp_asterisk depends upon res_pjproject. + + Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377 + +2018-10-17 14:34 +0000 [1fad6b9079] Richard Mudgett + + * modules: Add missing run time module support levels. + + Change-Id: I29b9dbfa4bbfc49f21eba356858e38b1d3041824 + +2018-10-14 07:58 +0000 [5ab94d2a3e] Corey Farrell + + * taskprocessor: Warn on unused result from pushing task. + + Add attribute_warn_unused_result to ast_taskprocessor_push, + ast_taskprocessor_push_local and ast_threadpool_push. This will help + ensure we perform the necessary cleanup upon failure. + + Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d + +2018-10-16 12:28 +0000 [915861b431] Richard Mudgett + + * bundled pjproject: Remove timer cleanup usage patch. + + This patch is not in the upstream pjproject and does unsafe things with + the timer->_timer_id and timer->_grp_lock values in pj_timer_entry_reset() + outside of the timer heap lock. pj_timer_entry_reset() is also called for + timers that are not about to be rescheduled in a few places. + + Change-Id: I4fe0b4bc648f7be5903cf4531b94fc87275713c1 + +2018-10-10 04:37 +0000 [79677ead28] Corey Farrell + + * refdebug: Create refstats.py script. + + This allows us to process AO2 statistics for total objects, memory + usage, memory overhead and lock usage. + + * Install refstats.py and reflocks.py into the Asterisk scripts folder. + * Enable support for reflocks.py without DEBUG_THREADS. + + Steal a bit from the ao2 magic to flag when an object lock is used. + Remove 'lockobj' from reflocks.py since we can now record 'used' or + 'unused' for those objects. + + Add comments to explain thread safety of the 'struct __priv_data' + bitfields. + + Change-Id: I84e9d679cc86d772cc97c888d9d856a17e0d3a4a + +2018-10-12 12:14 +0000 [aae5bdc22e] Alexei Gradinari + + * res_pjsip: set callerid_tag to empty string + + This patch sets the callerid_tag to empty string by default. + + If the callerid_tag is set to NULL then the tag does not + become part of a connected line update. + For example: + Alice's tag is "Alice". + Bob's tag is empty. + Charlie's tag is "Charlie". + Alice calls Bob and then does attended transfer to Charlie. + When Alice hangs up the CONNECTEDLINE(tag) is "Alice" + on the interception routine on the Charlie's channel, but should be empty. + + Ths patch also fix memory leaks if there are more then one options + "callerid", "callerid_tag", "voicemail_extension" and "contact_user" + in the pjsip.conf endpoint definition. + + Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4 + +2018-10-11 06:24 +0000 [f06de6900e] Corey Farrell + + * threadpool: Eliminate pointless AO2 usage. + + thread_worker_pair, set_size_data and task_pushed_data structures are + allocated with AO2 objects, passed to a taskprocessor, then released. + They never have multiple owners or use locking so AO2 only adds + overhead. + + Change-Id: I2204d2615d9d952670fcb48e0a9c0dd1a6ba5036 + +2018-10-12 12:21 +0000 [675d8a46b4] Corey Farrell + + * main/astfd: Fix GCC8 format-truncation warning. + + The field used to store call arguments was not large enough to hold the + arguments string that can be constructed for 'open'. Expand it to + prevent this warning/error. + + Change-Id: I514927f256481bc84df10a51b19d5b5fb1bc387e + +2018-10-09 16:18 +0000 [682f96cb5c] Richard Mudgett + + * res_statsd.c: Fix returned reload status. + + The return status when there was no change in statsd.conf was incorrect. + This resulted in the wrong status message on the CLI when reloading the + module. + + * Fixed cleanup on initial load if initializing statsd failed. + + Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481 + +2018-10-03 16:51 +0000 [17f4e6ad4d] Emmanuel BUU + + * core/frame: generate correct T.140 payload in ast_sendtext_data() + + ast_sendtext_data() would create an incorrect T.140 text frame which + length include the null terminator byte. It causes ultimately RTP + packets to be send with this trailing 0. The proposed fix just set the + correct length to the text frame + + ASTERISK-28089 + Reported by: Emmanuel BUU + Tested by: Emmanuel BUU + + Change-Id: I7ab1b9ed1e21683b2b667ea0a59d9aba3c77dd96 + +2018-10-04 18:33 +0000 [c8ee1a183f] Corey Farrell + + * loader: Flag module as declined in all cases where it fails to load. + + This has no effect on startup since AST_MODULE_LOAD_FAILURE aborts + startup, but it's possible for this code to be returned on manual load + of a module after startup. + + It is an error for a module to not have a load callback but this is not + a fatal system error. In this case flag the module as declined, return + AST_MODULE_LOAD_FAILURE only if a required module is broken. + + Expand doxygen documentation for AST_MODULE_LOAD_*. + + Change-Id: I3c030bb917f6e5a0dfd9d91491a4661b348cabf8 + +2018-10-04 13:13 +0000 [c6c3a63696] Richard Mudgett + + * func_periodic_hook.c: Cleanup module resources on failure. + + * Make load_module() cleanup if it failed to setup the module. + + * Make unload_module() always return 0. It is silly to fail unloading if + the hook function we try to unregister was not even registered. + + Change-Id: I280fc6e8ba2a7ee2588ca01d870eebaf74b4ffe6 + +2018-10-04 11:49 +0000 [9f02861d22] Richard Mudgett + + * codec_speex.c: Cleanup module loading to DECLINE and not FAILURE. + + If codec_speex fails to register a translator it would cause Asterisk to + exit instead of continue as a DECLINED module. + + * Make unload_module() always return 0. It is silly to fail unloading if + any translators we try to unregister were not even registered. + + Change-Id: Ia262591f68333dad17673ba7104d11c88096f51a + +2018-10-04 13:03 +0000 [30717bafbf] George Joseph + + * CI: Fix missing () in gates.jenkinsfile + + Change-Id: I2f252e0f8c7f1a6328438fbd2be5d6574b7dfa5b + +2018-10-04 10:13 +0000 [58622a87f4] George Joseph + + * CI: Add timestamps and timeouts to jenkinsfiles + + Change-Id: Ide83574dc957bc1df28e30a69079140050dfc35f + +2018-10-03 17:02 +0000 [b2ed667712] Sean Bright + + * ast_coredumper: Remove .gdbinit file on exit + + Change-Id: I1297de78628773ca368e687c6f148bf74857cae9 + +2018-10-03 09:33 +0000 [e19f27a667] Sean Bright + + * CI: Look up configured kernel.core_pattern sysctl + + Change-Id: I8246a0147df8d821fbbcabc1db1887104b8bedc4 + +2018-10-03 15:51 +0000 [42880fab50] Corey Farrell + + * jenkins: Fix cleanup command redirection. + + Fix redirection to /dev/null of cleanup commands. The '2' was being + interpreted as part of the command instead of part of the redirect. + + Change-Id: I2e3a591b165e0288c4b82b9ef475fdfd5392a90a + +2018-10-03 15:29 +0000 [a29cefe5b2] George Joseph + + * ast_coredumper: Don't use "declare -n" + + Change-Id: I7ddfed4cd6549a0cd458e4d5cf9ac95d784de6cb + +2018-10-02 16:15 +0000 [3601329c5a] Richard Mudgett + + * res_smdi.c: Fix module ref counting and inverted test. + + I think this module is so screwed up that it doesn't work anymore. Even + with these attempts to fix things it still won't gracefully shut down. + The module refs will not go to zero to allow unloading the module. + + * Fix module ref counting dealing with the SMDI interface object. There + were several off-nominal paths that unbalanced the module ref count. Also + the destructor freed the ao2 object itself which is bad. Made the + smdi_read thread not hold its own ref to the SMDI interface object so when + all refs go away the destructor will stop the listener thread. + + * Fixed the smdi_load() return code of 1 concerning the number of + listeners. The test was inverted. + + Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784 + +2018-10-02 16:23 +0000 [305d08f112] Richard Mudgett + + * res_smdi.c: Made use defaults if the smdi.conf file does not exist. + + This module is an optional dependency of a couple of other modules. If it + declines to load, it then forces other modules that can optionally use + this module to also decline. + + * Made use the default configuration if the config file does not exist and + simplified some of the logic. + + Change-Id: Ib93191f1fe28c0dd9ebe3d84c7762b32f83c4eb9 + +2018-10-02 17:15 +0000 [932d0a40cf] Corey Farrell + + * astobj2: Comment on OBJ_NOLOCK in ao2_container_clone. + + The test for OBJ_NOLOCK looks wrong but it isn't. Add comments to + prevent confusion. + + Change-Id: I9662b82eb39e7627a1f1944fd18f967a2b987344 + +2018-10-03 09:05 +0000 [f608b73a29] Sean Bright + + * CI: Use brace expansion instead of calling out to seq + + Also make the shebang in publishAsteriskDocs.sh the first line. + + Change-Id: I3fdd6f22e652e4fb5b5fe85df46fa34eb6d0cf08 + +2018-10-03 08:59 +0000 [9c9f060b3a] Sean Bright + + * CI: Use bindport instead of port in test http.conf + + Change-Id: Ife9a6879da63a56e5b8348a2024eeed4e7b1615b + +2018-10-03 07:56 +0000 [286339aa34] Sean Bright + + * http.c: Reload TLS even if http.conf hasn't changed + + There is currently no way to indicate to Asterisk that TLS certificates + and/or keys have been updated other than by modifying http.conf or + restarting Asterisk. + + There is already code in main/tcptls.c that determines if a reload is + actually necessary based on the hashes of the certicate and dependent + files, so this change merely gives us a way to request a reload without + explicitly modifying http.conf. + + Change-Id: Ie795420dcc7eb3d91336820688a29adbcc321276 + +2018-10-02 13:29 +0000 [a69a50b6ec] Richard Mudgett + + * res_statsd.c: Made use defaults if the statsd.conf file does not exist. + + This module is an optional dependency of many modules. If it declines to + load it then forces other modules that can optionally use this module to + also decline. + + * Made use default configuration if there is a config error or the config + file does not exist. + + Change-Id: If1068a582ec54ab7fb437265cb5370a97a825737 + +2018-10-01 22:12 +0000 [cacbe32534] Corey Farrell + + * core: Disable astobj2 locking for some common objects. + + * ACO options + * Indications + * Module loader ref_debug object + * Media index info and variants + * xmldoc items + + These allocation locations were identified using reflocks.py on the + master branch. + + Change-Id: Ie999b9941760be3d1946cdb6e30cb85fd97504d8 + +2018-09-13 13:03 +0000 [639718211a] Corey Farrell + + * Resolve warning about duplicate 'dialplan' CLI. + + Change-Id: I029db1b4a32ccfb38374d6fe944dc430866f4b30 + +2018-10-02 01:33 +0000 [b25a261aa5] Corey Farrell + + * loader: Fix result of module reload error. + + When a module reload fails we never set AST_MODULE_RELOAD_ERROR. This + caused reload failures to incorrectly report 'No module found'. + + Change-Id: I5f3953e0f7d135e53ec797f24c97ee3f73f232e7 + +2018-09-28 10:13 +0000 [e4cf513f81] Corey Farrell + + * loader: Improve error handling. + + * Display list of unavailable dependencies when they cause another + module to fail loading. + * When a module declines to load find all modules which depend on it so + they can be declined and listed together. + * Prevent retry of declined modules during startup. + * When a module fails to dlopen try loading it with RTLD_LAZY so we can + attempt to display the list of missing dependencies. + + These changes are meant to reduce logger spam that is caused when a + module has many dependencies and declines to load. This also fixes some + error paths which failed to recognize required modules. + + Module load/start errors are delayed until the end of loader startup. + + Change-Id: I046052c71331c556c09d39f47a3b92975f3e1758 + +2018-09-25 16:19 +0000 [24cece660b] Emmanuel BUU + + * core/frame: Fix ast_frdup() and ast_frisolate() for empty text frames + + If a channel creates an AST_TEXT_FRAME with datalen == 0, the ast_frdup() + and ast_frisolate() functions could create a clone frame with an invalid + data.ptr which would cause a crash. The proposed fix is to make sure that + for such empty text frames, ast_frdup() and ast_frisolate() return cloned + text frames with a valid data.ptr. + + ASTERISK-28076 + Reported by: Emmanuel BUU + Tested by: Emmanuel BUU + + Change-Id: Ib882dd028598f13c4c233edbfdd7e54ad44a68e9 + +2018-09-30 23:11 +0000 [13df745278] Corey Farrell + + * astobj2: Record lock usage to refs log when DEBUG_THREADS is enabled. + + When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock + was ever used, so it can be recorded in the REF_DEBUG destructor entry. + + Create contrib/scripts/reflocks.py to process locking used by + allocator. This can be used to identify places where + AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage. + + Change-Id: I2e3cd23336a97df2692b545f548fd79b14b53bf4 + +2018-10-01 12:11 +0000 [52b530503f] Corey Farrell + + * app_page: Add dependency against app_confbridge. + + Change-Id: I1946509f518961d23fb21229d91676ee3e441921 + +2018-09-28 13:55 +0000 [b68b3012ea] Richard Mudgett + + * app_queue.c: Fix json ref leak + + Declining the queue_member_status_type stasis message in stasis.conf + causes these messages to leak json objects. + + * Add missing ast_json_unref() if the type is NULL in + queue_publish_member_blob(). + + ASTERISK-28084 + + Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc + +2018-10-01 03:24 +0000 [497973c8a2] Corey Farrell + + * Append CHANGES/UPGRADE.txt for module loader changes. + + Change-Id: Ib8db4e14187f5c11ecbff532df17d30c5d36fa3e + +2018-09-25 17:33 +0000 [8bb031abc7] Alexei Gradinari + + * res_pjsip: improve realtime performance on CLI 'pjsip show contacts' + + CLI command 'pjsip show contacts' inefficiently make a lot of DB requests. + + For example if there are 10k aors then asterisk requests these 10k records + of aor and then does 10k requests of contact - one request per aor. + + Even if use 'like ' the asterisk requests all aor's and contact's + records and then filters them by itself. + + This patch gathers contact's container by + - retrieving all dynamic contacts by regex (filtered by reg_server) + - retrieving all aors with permanent contacts + - finally filters container by regex + + ASTERISK-28077 #close + + Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618 + +2018-09-28 14:45 +0000 [24b92291d5] Corey Farrell + + * jansson-bundled: Add patches to improve json_pack error reporting. + + Change-Id: I045e420d5e73e60639079246e810da6ae21ae22b + +2018-09-27 19:32 +0000 [205c6be895] Corey Farrell + + * lock: Improve performance of DEBUG_THREADS. + + Add a volatile flag to lock tracking structures so we only need to use + the global lock when first initializing tracking. + + Additionally add support for DEBUG_THREADS_LOOSE_ABI. This is used by + astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is + not defined. + + Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b + +2018-09-27 13:19 +0000 [f10c7b6eeb] George Joseph + + * app_confbridge: Use bridge join hook to send join and leave events + + The first attempt at publishing confbridge events to participants + involved publishing them at the same time stasis events were + created. This caused issues with bridge and channel locks. The + second attempt involved publishing them when the stasis events + were received by the code that published the confbridge AMI events. + This caused timing issues because, depending on resources available, + the event could be received before channels actually joined the + bridge and would therefore fail to send messages to the participant. + + This attempt reverts to the original mechanism with one exception. + The join and leave events are published via bridge join and leave + hooks. This guarantees the states of the channels and bridge and + provides deterministic timing for event publishing. + + Change-Id: I2660074f8a30a5224cb953d5e047ee84484a9036 + +2018-09-27 04:51 +0000 [62a0db2df1] Corey Farrell + + * astobj2: Reduce memory overhead. + + Reduce options to 2-bit field, magic to 30 bit field. Move ref_counter + next to options and explicitly use int32_t so the fields will pack. + + This reduces memory overhead for every ao2 object by 8 bytes on x86_64. + + Change-Id: Idc1baabb35ec3b3d8de463c4fa3011eaf7fcafb5 + +2018-09-27 15:01 +0000 [ac23e5ad48] Sean Bright + + * config.c: Cleanup AST_INCLUDE_GLOB + + * In main/config.c, AST_INCLUDE_GLOB is fixed to '1' making the #ifdefs + pointless. + + * In utils/extconf.c, AST_INCLUDE_GLOB is never defined so there is a + lot of dead code. + + Change-Id: I1bad1a46d7466ddf90d52cc724e997195495226c + +2018-09-27 05:33 +0000 [39bf9881e0] Corey Farrell + + * astobj2: Fix shutdown order. + + When REF_DEBUG and AO2_DEBUG are both enabled we closed the refs log + before we shutdown astobj2_container. This caused the AO2_DEBUG + container registration container to be reported as a leak. + + Change-Id: If9111c4c21c68064b22c546d5d7a41fac430430e + +2018-09-05 21:14 +0000 [f23a12244d] Cao Minh Hiep + + * app_queue: Fix Attended transfer hangup with removing pending member. + + This issue related to setting of holdtime, announcements, member delays. + It works well if we set the member delays to "0" and no announcements + and no holdtime.This issue will happen if we set member delays to "1", + "2"... or announcements or holdtime and hangs up the call during + processing it. + + And here is the reason: + (At the step of answering a phone.) + It takes care any holdtime, announcements, member delays, + or other options after a call has been answered if it exists. + + Normally, After the call has been aswered, + and we wait for the processing one of the cases of the member delays + or hold time or announcements finished, "if (ast_check_hangup(peer))" + will be not executed, then queue will be updated at update_queue(). + Here, pending member will be removed. + + However, after the call has been aswered, + if we hangs up the call during one of the cases of the member delays + or hold time or announcements, "if (ast_check_hangup(peer))" + will be executed. + outgoing = NULL and at hangupcalls, pending members will not be removed. + + * This fixed patch will remove the pending member from container + before hanging up the call with outgoing is NULL. + + ASTERISK-27920 + + Reported by: Cao Minh Hiep + Tested by: Cao Minh Hiep + + Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855 + +2018-06-26 09:17 +0000 [f3422312ea] Moritz Fain + + * res_stasis: Fix stale data in ARI bridges + + Fixed an issue that resulted in "Allocation failed" each time an ARI + request was made to start playing MOH on a bridge. + + In bridge_moh_create() we were attaching the after bridge callbacks to + chan which is the ;1 channel of the unreal channel pair. We should have + attached them to the ;2 channel which is pushed into the bridge by + ast_unreal_channel_push_to_bridge(). The callbacks are called when the + specific channel leaves the bridging system. Since the ;1 channel is + never put into a bridge the callbacks never get called. The callbacks + then never remove the moh_wrapper from the app_bridges_moh container. As + a result we cannot find the channel associated with the wrapper to start + MOH because it has hungup. This is the reason causing the reported issue. + + * Rather than using after bridge callbacks to cleanup, we now have + moh_channel_thread() doing the cleanup when the channel hangs up. + + * Fixed moh_channel_thread() accumulating control frames on the stasis + bridge MOH channel until MOH is stopped. Control frames are no longer + accumulated while MOH is playing. + + * Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may + or may not return a channel ref. As a result ast_ari_bridges_start_moh() + wouldn't know it may have a channel ref to release. + stasis_app_bridge_moh_channel() will now return a ref with the channel it + returns. + + * Eliminated RAII_VAR in bridge_moh_create(). + + ASTERISK-26094 #close + + Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c + +2018-09-10 11:28 +0000 [b11a6643cf] Ben Ford + + * res_rtp_asterisk.c: Add "seqno" strictrtp option + + When networks experience disruptions, there can be large gaps of time + between receiving packets. When strictrtp is enabled, this created + issues where a flood of packets could come in and be seen as an attack. + Another option - seqno - has been added to the strictrtp option that + ignores the time interval and goes strictly by sequence number for + validity. + + Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71 + +2018-09-20 13:59 +0000 [e6a69ea2cf] Alexei Gradinari + + * res_odbc: fix missing SQL error diagnostic + + On SQL error there is not diagnostic information about this error. + There is only + WARNING res_odbc.c: SQL Execute error -1! + + The function ast_odbc_print_errors calls a SQLGetDiagField to get the number + of available diagnostic records, but the SQLGetDiagField returns 0. + However SQLGetDiagRec could return one diagnostic records in this case. + + Looking at many example of getting diagnostics error information + I found out that the best way it's to use only SQLGetDiagRec + while it returns SQL_SUCCESS. + + Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR + to res_config_odbc. + + ASTERISK-28065 #close + + Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1 + +2018-09-26 08:12 +0000 [950d0b65e5] George Joseph + + * CI: Add --test-timeout option to runTestsuite.sh + + The default is 600 seconds. + Also added timeouts to the *TestGroups.json files. + + Change-Id: I8ab6a69e704b6a10f06a0e52ede02312a2b72fe0 + +2018-09-18 08:01 +0000 [6627c56b3d] Peter Katzmann + + * chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI + + With tls and udp enabled asterisk generates a warning about sending + message via udp instead of tls. + sip notify command via cli works as expected and without warning. + + asterisk has to set the connection information accordingly to connection + and not on presumption + + ASTERISK-28057 #close + + Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e + +2018-09-24 17:56 +0000 [1ba51b00cc] George Joseph + + * configure.ac: Check for unbound version >= 1.5 + + In order to do this and provide good feedback, a new macro was + created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and + path setups for the library then compiles, links and runs a supplied + code fragment to do the final determination. In this case, the + final code fragment compares UNBOUND_VERSION_MAJOR + and UNBOUND_VERSION_MINOR to determine if they're greater than or + equal to 1.5. + + Since we require version 1.5, some code in res_resolver_unbound + was also simplified. + + ASTERISK-28045 + Reported by: Samuel Galarneau + + Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72 + +2018-09-24 12:43 +0000 [8bb264841a] Joshua Colp + + * res_rtp_asterisk: Raise event when RTP port is allocated + + This change raises a testsuite event to provide what port + Asterisk has actually allocated for RTP. This ensures that + testsuite tests can remove any assumption of ports and instead + use the actual port in use. + + ASTERISK-28070 + + Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044 + +2018-07-16 22:55 +0000 [adf539b2f0] Corey Farrell + + * jansson: Backport fixes to bundled, use json_vsprintf if available. + + Use json_vsprintf from versions which contain fix for va_copy leak. + + Apply fixes from jansson master: + * va_copy leak fix. + * Avoid potential invalid memory read in json_pack. + * Rename variable that shadowed another. + + Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539 + +2018-07-16 22:55 +0000 [93777faf36] Corey Farrell + + * json: Take advantage of new API's. + + * Use "o*" format specifier for optional fields in ast_json_party_id. + * Stop using ast_json_deep_copy on immutable objects, it is now thread + safe to just use ast_json_ref. + + Additional changes to ast_json_pack calls in the vicinity: + * Use "O" when an object needs to be bumped. This was previously + avoided as it was not thread safe. + * Use "o?" and "O?" to replace NULL with ast_json_null(). The + "?" is a new feature of ast_json_pack starting with Asterisk 16. + + Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48 + +2018-09-20 10:15 +0000 [06c0676da0] George Joseph + + * app_voicemail: Cleanup mailbox topic and cache + + app_voicemail wasn't properly cleaning up the stasis cache or the + mwi topic pool when the module was unloaded or when a user was + deleted as a result of a reload. This resulted in leaks in both + areas. + + * app_voicemail now calls ast_delete_mwi_state_full when it frees + a user structure and ast_delete_mwi_state_full in turn now calls + the new stasis_topic_pool_delete_topic function to clear the topic + from the pool. + + Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8 + +2018-09-17 15:35 +0000 [31fba4e869] Kevin Harwell + + * rtp_engine: rtcp_report_to_json can overflow the ssrc integer value + + When writing an RTCP report to json the code attempts to pack the "ssrc" and + "source_ssrc" unsigned integer values as a signed int value type. This of course + means if the ssrc's unsigned value is greater than that which can fit into a + signed integer value it gets converted to a negative number. Subsequently, the + negative value goes out in the json report. + + This patch now packs the value as a json_int_t, which is the widest integer type + available on a given system. This should make it so the value no longer + overflows. + + Note, this was caught by two failing tests hep/rtcp-receiver/ and + hep/rtcp-sender. + + Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0 + +2018-09-21 14:32 +0000 [22cf065ec9] George Joseph + + * app_voicemail: Fix stack overrun in append_mailbox + + The append_mailbox function wasn't calculating the correct length + to pass to ast_alloca and it wasn't handling the case where context + might be empty. + + Found by the Address Sanitizer. + + Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161 + +2018-09-21 15:23 +0000 [4d51a8e05b] George Joseph + + * channel.c: Address stack overflow in does_id_conflict() + + does_id_conflict() was passing a pointer to an int to a callback + that expected a pointer to a size_t. + + Found by the Address Sanitizer. + + Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503 + +2018-09-21 10:19 +0000 [bdc8159799] Corey Farrell + + * res_rtp_asterisk: Fix crash on ast_rtp_new failure. + + ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call + the destroy callback. Remove call to ast_free from ast_rtp_new, leave + it to rtp_engine.c to initiate the full cleanup. Add error detection + for the ssrc_mapping vector initialization. In rtp_allocate_transport + set rtp->s = -1 in the failure path where we close that FD to ensure we + don't try closing it twice. + + ASTERISK-27854 #close + + Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451 + +2018-09-20 15:26 +0000 [ad4a6bc27a] Sean Bright + + * res_rtp_asterisk: Reset all settings on module reload + + 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults + if they are not present in the updated configuration file. + + Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670 + +2018-09-20 09:41 +0000 [d277db4a38] George Joseph + + * stasis: Add function to delete topic from pool + + There's been a long standing leak when using topic pools. The + topics in the pool get cleaned up when the last pool reference is + released but you can't remove a topic specifically. If you reloaded + app_voicemail for instance, and mailboxes went away, their topics + were left in the pool. + + * Added stasis_topic_pool_delete_topic() so modules can clean up + topics from pools. + * Registered the topic pool containers so it can be examined from + the CLI when AO2_DEBUG is enabled. They'll be named + "-pool". + + Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25 + +2018-08-16 10:45 +0000 [a801543f79] Sean Bright + + * AST-2018-009: Fix crash processing websocket HTTP Upgrade requests + + The HTTP request processing in res_http_websocket allocates additional + space on the stack for various headers received during an Upgrade request. + An attacker could send a specially crafted request that causes this code + to overflow the stack, resulting in a crash. + + * No longer allocate memory from the stack in a loop to parse the header + values. NOTE: There is a slight API change when using the passed in + strings as is. We now require the passed in strings to no longer have + leading or trailing whitespace. This isn't a problem as the only callers + have already done this before passing the strings to the affected + function. + + ASTERISK-28013 #close + + Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a + +2018-09-03 09:55 +0000 [406be41f21] David Hajek + + * chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads + + Fixes random asterisk crash on start or reload with TLS phones. + + ASTERISK-28034 #close + Reported-by: David Hajek + + Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac + +2018-09-20 04:48 +0000 [b9874da790] Joshua Colp + + * res_remb_modifier: Add module for controlling REMB from CLI. + + This adds a module which registers a CLI command that can set the + REMB bitrate value for REMB as it enters or exits Asterisk. This + allows you to ignore what Asterisk or a client produces and is + useful for demonstrations. + + This does not generate REMB frames, however, but just modifies + them as they flow to or from a channel. + + Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec + +2018-09-14 15:51 +0000 [c99a9b228b] Richard Mudgett + + * stasis: No need to keep a stasis type ref in a stasis msg or cache object. + + Stasis message types are global ao2 objects and we make stasis messages + and cache entries hold references to them. Since there are currently + situations where cache objects are never deleted, the reference count on + the types can exceed 100000 and generate a FRACK assertion message. The + stasis message cache could conceivably also have that many messages + legitimately on large systems. + + The only down side to not holding the message type ref in the stasis + message is it only makes a crash either at shutdown or when manually + unloading a busy module slightly more likely. However, this is more + exposing a pre-existing stasis shutdown ordering issue than a problem with + not holding a message type ref in stasis messages. + + * Made stasis messages and cache entries no longer hold a ref to the + message type. + + Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707 + +2018-09-18 13:59 +0000 [58035702cb] Richard Mudgett + + * pjproject: Update initial 2.8 patches to apply cleanly. + + ASTERISK-28059 + + Change-Id: I027472f2753391646dde594a709a75f14422db93 + +2018-09-14 15:48 +0000 [79e3becc5d] Richard Mudgett + + * stasis_message.c: Don't create immutable stasis objects with locks. + + * Create the stasis message object without a lock as it is immutable. + * Create the stasis message type object without a lock as it is immutable. + * Creating the stasis message type could crash if the passed in type name + is NULL and REF_DEBUG is enabled. Added missing NULL check when passing + the ao2 object tag string. + + Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32 + +2018-09-17 11:38 +0000 [ce9a980be6] Joshua Colp + + * pjproject: Upgrade to 2.8. + + This change brings in PJSIP 2.8, removes all the patches + that were merged upstream, and makes a minor change to + support a breaking change that was done. + + ASTERISK-28059 + + Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189 + +2018-09-18 09:39 +0000 [6a1c313fac] Florian Floimair + + * alembic: fix suppress_q850_reason_headers column name + + In the original commit introducing the feature the column in the alembic + script was called 'suppress_q850_reason_header'. + In the code however the option is called 'suppress_q850_reason_headers' + (trailing 's'). This leads to errors when ARI push configuration is used. + + Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f + +2018-09-13 07:55 +0000 [cdece3b637] George Joseph + + * app_voicemail: Remove need to subscribe to stasis + + app_voicemail was using the stasis cache to build and maintain a + list of mailboxes that had subscribers. It then used this list + to determine if a mailbox should be polled for new messages if + polling was enabled. For this to work, stasis had to cache every + subscription and unsubscription to the mailbox which caused a lot of + overhead, both cpu and memory related. + + Since polling is only required when changes are being made to + mailboxes outside of app_voicemail and since the number of mailboxes + that don't have any subscribers is likely to be very low, all + mailboxes are now polled instead of just the ones with subscribers. + + This paves the way for disabling the caching of stasis subscription + change messages. + + Also fixed cleanup in some of the unit tests that not only left + test users in the users list but also caused segfaults if the tests + were run more than once. + + ASTERISK-27121 + + Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee + +2018-09-18 06:08 +0000 [32a7b9f4b3] Joshua Colp + + * res_pjsip_session: Don't add declined stream if one does not exist. + + Given a scenario where a session refresh was done with a removed + stream we would always add a removed stream to the outgoing SDP + even if one did not already exist. + + This change makes it so that a removed stream is only placed into + the SDP if one already exists. + + ASTERISK-28047 + + Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442 + +2018-09-10 10:12 +0000 [246c39e46c] Corey Farrell + + * install_prereq: Remove unpackaged version of jansson. + + This is removed in favor of ./configure --with-jansson-bundled. The + install-unpackaged command would only install jansson once, so once + installed it would never update, where the bundled copy will be kept up + to date. + + Change-Id: Ideab1f65419608d3795aa608e9da873823cc42d3 + +2018-09-17 10:38 +0000 [3d9deb35f0] Sean Bright + + * autoconf: Check for srtp_get_version_string() before using it + + Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df + +2018-09-17 07:10 +0000 [ceafac3d7f] George Joseph + + * CI: Fix typo in testsuite git checkout + + Change-Id: I30024515e5b00a5044fd39fbff27d818f016b719 + +2018-09-16 06:08 +0000 [b68617ac2c] Sean Bright + + * res_srtp.c: Show linked version of libsrtp on module init + + Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342 + +2018-09-07 09:40 +0000 [07cb13f75f] Sean Bright + + * res_pjsip: Log IPv6 addresses correctly + + Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name + store IPv6 addresses without enclosing brackets. This causes some log + output to be confusing because it is difficult to separate the IPv6 + address from a port specification. + + * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and + pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 + output. + + * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses + in brackets. + + * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure + to also set pjsip_rx_data.pkt_info.src_addr. + + Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8 + +2018-09-14 12:31 +0000 [8be6998f8d] George Joseph + + * CI: Use proper credentials for Security testsuite checkout + + Can't do anonymous http checkout from Security-testsuite. + Need to use same credentials as the gerrit review checkout. + + Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05 + +2018-09-13 11:06 +0000 [5ec6d2c33e] George Joseph + + * stasis_cache: Stop caching stasis subscription change messages + + Since app_voicemail no longer uses the cache to maintain its state + there is no longer a need to cache these messages. + + ASTERISK-27121 + + Change-Id: I321c708505f5ad8d00e1b0afc4c27dc2ac12ecb4 + +2018-09-12 12:39 +0000 [2ba2ff050d] Corey Farrell + + * CI: Use .gitreview to default BRANCH_NAME. + + This ensures that binary modules are avoided in the master branch even + if BRANCH_NAME is not set. + + Change-Id: I79162d2063f22fa9d6b31fde4827ace2dd5bf0da + +2018-09-11 07:22 +0000 [bc8cdcefa8] Walter Doekes + + * optional_api: Remove unused nonoptreq fields + + As they're not actively used, they only grow stale. The moduleinfo field itself + is kept in Asterisk 13/15 for ABI compatibility. + + ASTERISK-28046 #close + + Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc + +2018-09-03 06:50 +0000 [012272a114] lvl + + * manager: Set AMI event "Newexten" to the EVENT_FLAG_DIALPLAN class + + The documentation already specified EVENT_FLAG_DIALPLAN for this + event, but the implementation was using EVENT_FLAG_CALL. + + Using EVENT_FLAG_DIALPLAN allows AMI clients to opt out of receiving + this highly verbose event. + + ASTERISK-28033 + + Change-Id: I45b3119f30e4dbc17b49831f2b1a4f2c1beadafe + +2018-09-12 07:18 +0000 [65e0eb8fc6] Sean Bright + + * res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP + + The bundled version of pjproject has a patch for Solaris compatability + that changes the definition of various socket structures which we need + to account for when compiling against a non-bundled version. + + ASTERISK-28049 #close + + Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0 + +2018-09-10 22:28 +0000 [28b32fbd44] Corey Farrell + + * Build System: Resolve conflict between DESTDIR and bundled jansson. + + If Asterisk is built using a DESTDIR this will cause the bundled jansson + to be installed to an unexpected location and we will fail to find it. + + Change-Id: Id033e2813261e0d45232383d44c6391122169548 + +2018-08-30 03:42 +0000 [35e02d6f17] Frederic LE FOLL + + * res_musiconhold.c: Restart MOH if previous hold just reached end-of-file + + On MOH activation, moh_files_readframe() is called while the current + stream attached to the channel is NULL and it calls ast_moh_files_next() + immediately. However, it won't call ast_moh_files_next() again if sample + reading fails. The failure may occur because res_musiconhold retains the + last sample reading position in the channel data and MOH during the + previous hold/retrieve just reached EOF. Obviously, a bit of bad luck is + required here. + + * Restructured moh_files_readframe() to try a second time to start MOH if + there was no stream setup and the saved position was at EOF. Also added + comments describing what is going on for each step. + + ASTERISK-28029 + + Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860 + +2018-09-05 06:39 +0000 [f97d92bd0a] Joshua Colp + + * core: Don't stop generators when writing RTCP frames. + + Generators provide such functionality as tone generation or + silence generation. RTCP frames provide RTCP information and + should not stop generators from operating. + + ASTERISK-28005 + + Change-Id: Ieadada07b068a7aa426e8763f1b73a18e1ac34a9 + +2018-09-03 06:28 +0000 [1174759f0c] lvl + + * app_queue: Update realtime queuemembers after wait_a_bit(), not before + + This ensures the most up-to-date information is used for the next + call attempt. + + ASTERISK-28032 + + Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce + +2018-08-28 08:42 +0000 [600c5d79fd] Sean Bright + + * res_pjproject: Add utility functions to convert between socket structures + + Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address + needs to be rendered to a string and then parsed into the correct + structure. This also involves a call to getaddrinfo(3). The same is true + for the inverse operation. + + Instead, because we know the internal structure of both ast_sockaddr and + pj_sockaddr, we can translate directly between the two without the + need for an intermediate string. + + Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761 + +2018-08-30 13:08 +0000 [0dd8ab3532] George Joseph + + * stasis_cache: Prune stasis_subscription_change messages + + The stasis cache provides a way to reconstruct the current state + of topic subscribers. Unfortunately, since every subscribe and + unsubscribe is cached, the cache continues to grow unabated while + asterisk is running. This patch removes subscribe messages from + the cache when the corresponding unsubscribe is received. + + This patch also registers the cache containers with ao2 so that if + AO2_DEBUG is turned on, you can list the container and get its + stats from the CLI. + + ASTERISK-27121 + + Change-Id: I3d18905e477f3721815da91f30da8d3fbb2d4f56 + +2018-09-03 09:27 +0000 [1a3115d1c5] Rodrigo Ramírez Norambuena + + * app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done + + Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8 + +2018-08-15 14:27 +0000 [b779a93d8d] Chris-Savinovich + + * pbx_config.c: Fix reloading module if initially declined to load + + Added decline if extensions.conf file not available + when loading pbx_config, and also made sure everything + gets properly unregistered and/or destroyed on unload. + + Change-Id: Ib00665106043b1be5148ffa7a477396038915854 + +2018-08-30 14:42 +0000 [e387750104] Richard Mudgett + + * http.c: Give HTTP error response when received lines are too long. + + Added a check when we receive a HTTP request line or header line that is + too long. We now return an error response to the sender because we are + not able to process the request. + + Change-Id: I6df2705435fd7dde4d5d3bdf7acec859cfb7c12d + +2018-08-29 16:14 +0000 [f657793ee4] Richard Mudgett + + * iostream.c: Fix ast_iostream_gets() needlessly returning failure. + + Providing a buffer larger than the internal buffer of ast_iostream_gets() + fails to get lines longer than the internal buffer. + + * Made ast_iostream_gets() fill the supplied buffer with read data until + either a '\n' is found or the supplied buffer is filled just like fgets(). + + Change-Id: If18b3f6ee500e22f0633a68779ed09f7e0f305ed + +2018-08-06 15:37 +0000 [d60411a2b4] Richard Mudgett + + * res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch + + ASTERISK-27988 + + Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843 + +2018-08-29 05:18 +0000 [40def05949] Joshua Colp + + * res_fax: Handle fax gateway being started more than once. + + The T.38 fax gateway state machine can cause the fax gateway + to be started more than once on a channel depending on the + responses of the remote endpoint. This would previously leak + the channel name, channel unique id, and underlying fax engine + state. This change instead makes it so that if the fax gateway + session is already present and not reserved the fax gateway + is not started again. + + ASTERISK-27981 + + Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e + +2018-08-28 08:01 +0000 [39459b1ee4] Sean Bright + + * res_pjsip_transport_websocket: Properly set src_name for IPv6 + + SIP responses over WebSockets when the client is using IPv6 have invalid + Via headers according to RFC 3261. The 'received' header parameter + should not be wrapped in brackets if it is an IPv6 address. + + When src_name is populated by the built-in PJSIP transports, the code + uses pj_sockaddr_print() with 'flags' set to 0, meaning that the + brackets are not rendered around IPv6 addresses. + + This may be related to ASTERISK~27101. + + See also: https://github.com/onsip/SIP.js/pull/594 + + ASTERISK-28020 #close + + Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77 + +2018-08-26 13:18 +0000 [a2001c00e6] Corey Farrell + + * Create --disable-binary-modules option. + + This new option can be passed for ./configure or + ./tests/CI/buildAsterisk.sh to prevent download/install of binary + modules. + + Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES + will result in binary modules being enabled even if the build target is + incompatible with those modules. This includes CI scripts which enable + categories before disabling specific modules. + + If more binary modules are offered in the future this will help avoid + accidentally downloading them if unwanted or incompatible. Adding a + binary module will only require creating a new menuselect entry similar + to the existing ones, it will not be necessary to modify the CI scripts. + + Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166 + +2018-08-21 07:59 +0000 [289016239d] Emmanuel BUU + + * res/res_rtp_asterisk: remove debug traces generated by an empty frame + + The realtime text timer pops regularly and sends text frames even if + the buffer is empty. This causes a lot of unecessary debug logging. + + * Made red_write() test if we need to send a frame before calling + ast_rtp_write() + + ASTERISK-28002 + Reported by: Emmanuel BUU + Tested by: Emmanuel BUU + + Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd + +2018-08-13 08:12 +0000 [9680790531] Jaco Kroon + + * chan_sip: improved ip:port finding of peers for non-UDP transports. + + Also remove function peer_ipcmp_cb since it's not used (according to + rmudgett). + + Prior to b2c4e8660a9c89d07041271371151779b7ec75f6 (ASTERISK_27457) + insecure=port was the defacto standard. That commit also prevented + insecure=port from being applied for sip/tcp or sip/tls. + + Into consideration there are three sets of behaviour: + + 1. "previous" - before the above commit. + 2. "current" - post above commit, pre this one. + 3. "new" - post this commit. + + The problem that the above commit tried to address was guests over TCP. + It succeeded in doing that but broke transport!=udp with host!=dynamic. + + This commit attempts to restore sane behaviour with respect to + transport!=udp for host!=dynamic whilst still retaining the guest users + over tcp. + + It should be noted that when looking for a peer, two passes are made, the + first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer, + thus looking for full matches (IP + Port), the second pass sets + SIP_INSECURE_PORT, thus expecting matches on IP only where the matched + peer allows for that (in the author's opinion: UDP with insecure=port, + or any TCP based, non-dynamic host). + + In previous behaviour there was special handling for transport=tcp|tls + whereby a peer would match during the first pass if the utilized + transport was TCP|TLS (and the peer allowed that specific transport). + + This behaviour was wrong, or dubious at best. Consider two dynamic tcp + peers, both registering from the same IP (NAT), in this case either peer + could match for connections from an IP. It's also this behaviour that + prevented SIP guests over tcp. + + The above referenced commit removed this behaviour, but kept applying + the SIP_INSECURE_PORT only to WS|WSS|UDP. Since WS and WSS is also TCP + based, the logic here should fall into the TCP category. + + This patch updates things such that the previously non-explicit (TCP + behaviour) transport test gets performed explicitly (ie, matched peer + must allow for the used transport), as well as the indeterministic + source-port nature of the TCP protocol is taken into account. The new + match algorithm now looks like: + + 1. As per previous behaviour, IP address is matched first. + + 2. Explicit filter with respect to transport protocol, previous + behaviour was semi-implied in the test for TCP pure IP match - this now + made explicit. + + 3. During first pass (without SIP_INSECURE_PORT), always match on port. + + 4. If doing UDP, match if matched against peer also has + SIP_INSECURE_PORT, else don't match. + + 5. Match if not a dynamic host (for non-UDP protocols) + + 6. Don't match if this is WS|WSS, or we can't trust the Contact address + (presumably due to NAT) + + 7. Match (we have a valid Contact thus if the IP matches we have no + choice, this will likely only apply to non-NAT). + + To logic-test this we need a few different scenarios. Towards this end, + I work with a set number of peers defined in sip.conf: + + [peer1] + host=1.1.1.1 + transport=tcp + + [peer2] + host=1.1.1.1 + transport=udp + + [peer3] + host=1.1.1.1 + port=5061 + insecure=port + transport=udp + + [peer4] + host=1.1.1.2 + transport=udp,tcp + + [peer5] + host=dynamic + transport=udp,tcp + + Test cases for UDP: + + 1 - incoming UDP request from 1.1.1.1: + - previous: + - pass 1: + * peer1 or peer2 if from port 5060 (indeterminate, depends on peer + ordering) + * peer3 if from port 5061 + * peer5 if registered from 1.1.1.1 and source port matches + - pass 2: + * peer3 + - current: as per previous. + - new: + - pass 1: + * peer2 if from port 5060 + * peer3 if from port 5061 + * peer5 if registered from 1.1.1.1 and source port matches + - pass 2: + * peer3 + + 2 - incoming UDP request from 1.1.1.2: + - previous: + - pass 1: + * peer5 if registered from 1.1.1.2 and port matches + * peer4 if source port is 5060 + - pass 2: + * no match (guest) + - current: as previous. + - new as previous (with the variation that if peer5 didn't have udp as + allowed transport it would not match peer5 whereas previous + and current code could). + + 3 - incoming UDP request from anywhere else: + - previous: + - pass 1: + * peer5 if registered from that address and source port matches. + - pass 2: + * peer5 if insecure=port is additionally set. + * no match (guest) + - current - as per previous + - new - as per previous + + Test cases for TCP based transports: + + 4 - incoming TCP request from 1.1.1.1 + - previous: + - pass 1 (indeterministic, depends on ordering of peers in memory): + * peer1; or + * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or + * peer2 if the source port happens to be 5060; or + * peer3 if the source port happens to be 5061. + - pass 2: cannot happen since pass 1 will always find a peer. + - current: + - pass 1: + * peer1 or peer2 if from source port 5060 + * peer3 if from source port 5060 + * peer5 if registered as 1.1.1.1 and source port matches + - pass 2: + * no match (guest) + - new: + - pass 1: + * peer 1 if from port 5060 + * peer 5 if registered and source port matches + - pass 2: + * peer 1 + + 5 - incoming TCP request from 1.1.1.2 + - previous (indeterminate, depends on ordering): + - pass 1: + * peer4; or + * peer5 if peer5 registered from 1.1.1.2 + - pass 2: cannot happen since pass 1 will always find a peer. + - current: + - pass 1: + * peer4 if source port is 5060 + * peer5 if peer5 registered as 1.1.1.2 and source port matches + - pass 2: + * no match (guest). + - new: + - pass 1: + * peer4 if source port is 5060 + * peer5 if peer5 registered as 1.1.1.2 and source port matches + - pass 2: + * peer4 + + 6 - incoming TCP request from anywhere else: + - previous: + - pass 1: + * peer5 if registered from that address + - pass 2: cannot happen since pass 1 will always find a peer. + - current: + - pass 1: + * peer5 if registered from that address and port matches. + - pass 2: + * no match (guest) + - new: as per current. + + It should be noted the test cases don't make explicit mention of TLS, WS + or WSS. WS and WSS previously followed UDP semantics, they will now + enforce source port matching. TLS follow TCP semantics. + + The previous commit specifically tried to address test-case 6, but broke + test-cases 4 and 5 in the process. + + ASTERISK-27881 #close + + Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2 + +2018-08-20 07:23 +0000 [a74f8e51a6] Jaco Kroon + + * AMI: be less verbose when adding HTTP headers to AMI/HTTP messages. + + All HTTP/AMI message headers are being sent to the verbose channel. + There are multiple places this is happening. Consolidate the loop into + a function. Drop the debug/verbose message. + + Convert to using ast_asprintf to perform the length calculation, memory + allocation and snprintf all in one step. + + Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1 + +2018-08-23 06:57 +0000 [3bdbbb7637] Florian Floimair + + * alembic: increase uri column size + + When mobile SIP clients register with Asterisk that use some sort of + push notifications, the URI can get quite lengthy due to the + additional push-service annotations (things like tokens, pn-type, etc.) + contained in it. + + ASTERISK-28022 #close + + Change-Id: I4c7ceadc3bb405f3daf722641c8cd5ca4188cc37 + +2018-08-22 10:50 +0000 [c8bacd45f1] Matthew Fredrickson + + * sample_configs: noload res_hep.so by default + + Change disables loading of res_hep.so in default installation. Loading + res_hep has a performance impact whether it's used or not. This disables + loading of it in sample config files. + + Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0 + +2018-08-21 13:50 +0000 [14c6f8be9d] Sean Bright + + * app_queue: Silence GCC 8 compiler warning + + I'm only seeing an error in 14+, so I assume it is due to different + compiler options: + + app_queue.c: In function ‘handle_queue_add_member’: + app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11 + bytes into a region of size 3 [-Werror=format-overflow=] + sprintf(num, "%d", state); + ^~ + app_queue.c:10234:18: note: directive argument in the range + [-2147483648, 99] + sprintf(num, "%d", state); + ^~~~ + + Compiler: gcc version 8.0.1 20180414 (experimental) + [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) + + Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10 + +2018-08-20 11:23 +0000 [5ec27d5206] Richard Mudgett + + * AMI: Remove docs for nonexistent AMI ContactStatus event headers + + Change-Id: I5736965c64c44338f7330e85a24bb46818607f19 + +2018-08-06 06:22 +0000 [457ba355aa] Joshua Colp + + * res_pjsip: Reduce processing when a Contact is updated. + + When a Contact is updated the only material change that qualify + support cares about is the underlying configuration for the AOR. + In this case we will update things with the new AOR information but + otherwise the callback to indicate the Contact has changed can be + ignored. + + This is because it is only when a Contact is added or deleted that + material changes occur within the qualify support. An update can't + change the URI since it would result in a new Contact so it can be + ignored. + + Change-Id: I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d + +2018-08-10 19:28 +0000 [40f1604e2f] Richard Mudgett + + * res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response. + + We were still getting crashes after the first fix. Somehow we receive a + non-2xx final response before we get a 200 final response. With the + failure response we had already cleaned up and destroyed some data + structures. When the unexpected 200 response comes in we crash. + + * Add protection code to prevent processing another final T.38 reINVITE + response. + + ASTERISK-27944 + + Change-Id: I8b5baba8d07fe4d63f0d7d05d3eb9a3d27d40a74 + +2018-08-09 18:46 +0000 [8cd36ab9b6] Richard Mudgett + + * res_sorcery_realtime.c: Fix unqualified fetch warning. + + The allow_unqualified_fetch option for the sorcery realtime backend + blocked actually fetching all rows when the option is set to warn. + + * Made issue a warning and actually do the request when + allow_unqualified_fetch=warn is set. + + Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312 + +2018-06-11 00:07 +0000 [328f772d3b] Kirsty Tyerman + + * pbx_dundi: Added IPv6 support for dundi + + Change includes move to netsock2 library. + + ASTERISK-27164 + Reported-by: Adam Secombe + + Change-Id: Ia9e8dc3d153de7a291dbda4bd87fc827dd2bb846 + +2018-08-15 21:31 +0000 [273e2802aa] Richard Mudgett + + * pbx_dundi.c: Misc memory management fixes when destroying peers + + * In destroy_peer(), fixed memory leaks of lookup history strings and + qualify transactions when destroying peers. + + * In destroy_peer(), fixed leaving the registerexpire scheduled callback + active when a peer is destroyed on a reload. The reload marks and sweeps + peers so any peers not explicitly configured get destroyed. Peers created + dynamically from the '*' peer will not exist until they re-register after + the reload. These destroyed peers caused memory corruption when the + registerexpire timer expired. + + * Made build_peer() not schedule any callbacks on the '*' peer + (empty_eid). It is a special peer that is cloned to dynamically created + peers so it doesn't actually get involved in any message transactions. + + * Made do_register_expire() remove the dundi/dpeers AstDB entry when a + peer registration expires. + + * Fix deep_copy_peer() to not copy some things that cannot be copied to + the cloned peer structure. Timers, message transactions, and lookup + history are specific to a peer instance. + + * Made set_config() lock around processing the mappings configuration. + + * Reordered unload_module() to handle load_module() declining the load due + to error. + + Change-Id: Ib846b2b60d027f3a2c2b3b563d9a83a357dce1d6 + +2018-08-15 23:49 +0000 [d4e72ee296] Richard Mudgett + + * pbx_dundi.c: Handle thread shutdown better. + + Change-Id: Id52f99bd6a948fe6dd82acc0a28b2447a224fe87 + +2018-08-15 18:14 +0000 [916abe7cdc] Richard Mudgett + + * pbx_dundi: Fix debug frame decode string. + + * Fixed a typo in the name of the REGREQ frame decode string array. + * Fixed off by one range check indexing into the frame decode string + array. + * Removed some unneeded casts associated with the decode string array. + + Change-Id: I77435e81cd284bab6209d545919bf236ad7933c2 + +2018-08-16 16:21 +0000 [c035d0afe0] Richard Mudgett + + * pbx_dundi: Update sample config documentation. + + Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849 + +2018-08-15 14:44 +0000 [aee5f7c1b6] Richard Mudgett + + * res_rtp_asterisk.c: Fix unused variable warnings + + Compiling without SRTP support installed resulted in some unused variable + warnings. These warnings also showed that the srtp variable was obtained + and passed around some functions but not really used even when a system + has SRTP installed. + + Change-Id: I6daad34be3e89b19adef6e2fbe738018975155fc + +2018-08-16 13:51 +0000 [00563ce21a] George Joseph + + * CI: Fixup for non-13 branches + + Change-Id: I5e1d4a09e58b92b541bc8ed6f9e10e54c4e5101f + +2018-08-16 13:28 +0000 [e5f30eba79] George Joseph + + * CI: Final version of setting correct gerrit creds + + Change-Id: I7729ecceedceb12f52bf18dae259846aa1d993b3 + +2018-08-16 12:08 +0000 [8e1c541acf] George Joseph + + * CI: Add https credentials to gerrit checkouts + + If the review to be tested is in a project with restricted access, + we need to use the jenkins user's gerrit https credentials when we + do the checkout or the checkout will fail. + + Change-Id: I9dc9994763c5ebfeb9f1cff60fb53f6902b7fd5f + +2018-08-16 09:04 +0000 [01c90fefb3] Rodrigo Ramírez Norambuena + + * make config: os-release output error. + + Fix not show the error + "/bin/sh: /etc/os-release: No such file or directory" when the command + 'make config' is run in a System without systemv. + + The instruction 'make config' pre execute the syntax + "$(shell . /etc/os-release && echo $$ID)" to identified if system is a + Slackware and Opensuse. + + This change prevent show the message and is send to the /dev/null + + Change-Id: I7f43e281a8d9405b2519fc653de82d9b8b645fdf + +2018-08-09 02:34 +0000 [926d647def] Torrey Searle + + * res/res_pjsip_sdp_rtp: put rtcp-mux in answer only if offered + + If in the initial sdp the caller doesn't include the line + a=rtcp-mux + + Then asterisk shoud not include rtcp-mux in the response regardless + of rtcp-mux being enabled on the endpoint + + ASTERISK-28007 #close + + Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7 + +2018-08-15 14:49 +0000 [a83c464d9d] Corey Farrell + + * res_resolver_unbound: Fix leak of config nameserver strings. + + Change-Id: I3f396316bb40d1ae6e91f5f688042420f1a540ed + +2018-08-15 13:51 +0000 [24302bda21] Corey Farrell + + * res_pjsip: Resolve transport management leak at shutdown. + + Cleanup idle check scheduled events at shutdown. + + Change-Id: I61bfbb56bac69fe840c3242927d31ff3593be461 + +2018-08-15 11:31 +0000 [eb34b881a4] Corey Farrell + + * res_odbc: Allow unload at shutdown. + + This makes it possible for REF_DEBUG to report no leaks when loading + res_odbc. + + Change-Id: I1a3dea786bd6e7f4820a6dd5cbaa197fa783ce93 + +2018-08-15 11:12 +0000 [52fe5fe2c8] Corey Farrell + + * res_pjsip: Fix leak in pjsip_options. + + sip_options_get_endpoint_state_compositor_state leaked a reference to + the first available endpoint state compositor that was found. + + Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7 + +2018-08-14 11:55 +0000 [58c3677581] Richard Mudgett + + * contrib/scripts: Make astgenkey executable + + Change-Id: I11641d65592536dea9cbca5aa94a24c25d24dd5f + +2018-08-14 07:29 +0000 [fca3d4fe5f] Joshua Colp + + * res_pjsip_caller_id: Add "party" parameter to RPID header. + + This change adds the "party" parameter to the Remote-Party-ID header + which indicates which party the header information is applicable + to. In Asterisk this is determined on whether we are the calling + or called party. This is added to improve interoperability with some + implementations. + + ASTERISK-28006 + + Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca + +2018-08-07 10:57 +0000 [c31a01bd75] Ben Ford + + * res_pjsip/rtp: No joint capabilities between streams. + + When a conference contained a mixture of audio/video and audio-only + users, a NOTICE message would pop up stating there are no joint + capabilities between streams. This happens because streams can never be + removed, but they can be in a REMOVED state. If we have the scenario + where user A joins with audio/video, user B joins with audio-only, and + user C joins with audio/video, then user A leaves, the message would + be triggered. That removed stream is still in the SDP, but Asterisk + would pass it through, causing it to be seen as a ulaw stream. A check + has been added for removed streams, setting their status to REMOVED when + handling negotiated SDPs. + + Also addressed an issue where user A joins, then user B joins but does + not receive video until much later. Full frames were not being sent, + causing some PLI from the browser. Because the video was flowing in one + direction, the browser sets the SSRC to 1, but Asterisk was dropping the + PLI because of that. Added a check to see if the SSRC is 1 or not, which + sends full frames and allows video to flow between user A and user B. + This should only happen when dealing with PSFB or FUR, and in the case + of PSFB, only for PLI. + + ASTERISK-27398 + + Change-Id: I26e7c6f101bc119549eeca406b5bcd25ad8ebc5e + +2018-08-12 11:04 +0000 [2ce061091e] Ivan Poddubny + + * app_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE + + When a call leaves a queue on leaveempty condition, QUEUESTATUS + must be set to LEAVEEMPTY, no matter whether Queue was executed with or + without the "c" (continue) option. + + The regression was introduced in the fix for ASTERISK_25665. + The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was + overwritten in case when "c" is set, regardless of what was the cause + for leaving the queue. + + ASTERISK-27973 #close + Reported-by: Valentin Safonov + + Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c + +2018-08-09 15:25 +0000 [63ca367ab9] Corey Farrell + + * Sample configs: Fix pjsip.conf syntax error. + + It is valid for a config file to be empty or contain only comments, but + not valid for a config value to be set when no uncommented context + exists. This caused an error to be loged numerous times during start + when loading the default pjsip.conf. + + Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6 + +2018-07-19 22:28 +0000 [addfc93815] Corey Farrell + + * CI: Add support for coverage processing. + + Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`. This + will cause Asterisk to be compiled with coverage support. It also + initializes 'before' coverage data for all sources. Accept + --tested-only to disable modules which are not run by any test. + Enabling coverage also sets tested-only true by default. To build + everything with coverage enabled use `--coverage --tested-only=0`. + + ./tests/CI/processCoverage.sh is used to process the coverage and + generate HTML reports. + + Fix utils/check_expr2 which failed to compiled with coverage enabled. + + Add status output 5 times per stage of astobj2_test_perf to ensure + remote CLI does not timeout when compiled with coverage. Remote CLI + disconnects if no output is received for 60 seconds. When coverage is + enabled it takes about 70 seconds for my laptop to run the stages of + this test, so with the change a message is printed every 14 seconds. + + Change-Id: I890f7d5665087426ad7d3e363187691b9afc2222 + +2018-08-06 12:19 +0000 [c6ad25dcb7] Richard Mudgett + + * res_pjsip.h: Fix doxygen comments. + + Change-Id: I9cf97bdc756012d1f552ab007f4aa85e0ddb4e62 + +2018-08-06 06:36 +0000 [455ca1095e] Joshua Colp + + * stasis: Reduce calculation of stasis message type hash. + + When the stasis cache is used a hash is calculated for + retrieving or inserting messages. This change calculates + a hash when the message type is initialized that is then + used each time needed. This ensures that the hash is + calculated only once for the message type. + + Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37 + +2018-07-30 07:49 +0000 [603d1e8d4b] Alexander Traud + + * pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. + + The authors of PJProject undef s_addr because of some issue in Microsoft + Windows. However in Oracle Solaris, s_addr is not a structure member, but + defined to map to the real structure member. + + Updates the patch from ASTERISK_20366 + + ASTERISK-27997 + + Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11 + +2018-08-03 15:59 +0000 [acbb9f52b2] Richard Mudgett + + * res_pjsip: Make pjlib.h consistently included. + + * Don't include pjlib.h twice in res_pjsip.h + * Consistently use #include <> form for pjproject includes. + (pjsip.h and pjlib.h) + + Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7 + +2018-08-02 14:37 +0000 [a90177cd63] Salah Ahmed + + * dialplan_functions: wrong srtp use status report of a dialplan function + + If asterisk offer an endpoint with SRTP and that endpoint respond + with non srtp, in that case channel(rtp,secure,audio) reply wrong + status. + + Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key: + Currently this flag has being set redundantly. In either case identical + or different remote_key this flag has being set. So if we + don't set it while we receive identical remote_key or non SRTP SDP + response then we can take decision of srtp use by using that flag. + + ASTERISK-27999 + + Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7 + +2018-07-30 06:05 +0000 [1c7c867ce0] Alexander Traud + + * pjproject_bundled: Find shared libraries in root --with-ssl=PATH. + + The script configure from Teluu expects shared libraries (.so) in a subfolder + called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the + default location is the root of the source folder = PATH. Furthermore, Asterisk + supports both, 'lib' and root. For consistency and because Asterisk is using + (only) OpenSSL in PJProject, it is enhanced to support both locations, just + like Asterisk. + + ASTERISK-27995 + + Change-Id: I8eb916a88b6b8c22e29bb40bee8faaca6c73406f + +2018-08-01 09:45 +0000 [cbf082ed53] Joshua Colp + + * res_pjsip_registrar: Improve performance on inbound handling. + + This change removes a sorcery lookup for retrieving all + contacts at the end of the registration process by keeping + track of the contacts that are added/updated/deleted. + + This ensures at the end of the process the container of + contacts we have is the current state. + + Pool usage has also been reduced by allocating one for + usage throughout the handling of a REGISTER and resetting + it to a clean state. This ensures that in most cases + we allocate once and just reuse it. + + ASTERISK-28001 + + Change-Id: I1a78b2d46f9a2045dbbff1a3fd6dba84b612b3cb + +2018-07-17 07:13 +0000 [3424795f3a] Torrey Searle + + * thirdparty/pjproject: fix deadlock in response retransmissions + + The tdata containing the response can be shared by both the dialog + object and the tsx object. In order to prevent the race condition + between the tsx retransmission and the dialog sending a response, + clone the tdata before modifying it for the dialog send response. + + ASTERISK-27966 #close + + Change-Id: Ic381004a3a212fe1d8eca0e707fe09dba4a6ab4e + +2018-07-31 23:54 +0000 [a10a3aff6a] Corey Farrell + + * Build System: Improve ccache matching for different menuselect options. + + Changing any Menuselect option in the `Compiler Flags` section causes a + full rebuild of the Asterisk source tree. Every enabled option causes + a #define to be added to buildopts.h, thus breaking ccache caching for + every source file that includes "asterisk.h". In most cases each option + only applies to one or two files. Now we only define those options for + the specific sources which use them, this causes much better cache + matching when working with multiple builds. For example testing code + with an without MALLOC_DEBUG will now use just over half the ccache + size, only main/astmm.o will have two builds cached instead of every + file. + + Reorder main/Makefile so _ASTCFLAGS set on specific object files are all + together, sorted by filename. Stop adding -DMALLOC_DEBUG to CFLAGS of + bundled pjproject, this define is no longer used by any header so only + serves to break cache. + + The only code change is a slight adjustment to how main/astmm.c is + initialized. Initialization functions always exist so main/asterisk.c + can call them unconditionally. Additionally rename the astmm + initialization functions so they are not exported. + + Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027 + +2018-07-31 11:24 +0000 [68a3d39a99] Richard Mudgett + + * pjsip_wizard.conf.sample: Update remote_hosts description. + + Remove the note that SRV records are not supported as that is no longer + true. + + ASTERISK-27993 + + Change-Id: Id0dd6ef40e52702be9727a2b6122216cb00bb4ca + +2018-07-27 13:23 +0000 [a354599ecc] George Joseph + + * CI: Add optional uninstall step before installing asterisk + + Change-Id: I7dedf1e925eafc3a0adf01dd9dfbe44eb642aab7 + +2018-07-28 11:49 +0000 [7418dfa2c7] Alexander Traud + + * BuildSystem: Enable ncurses for menuselect in Solaris 11. + + The check for the library ncurses should use not use the header but + , because on some platforms is not a drop-in replacement + for : For example in Solaris, the symbol initscr is a typedef in + to a symbol which does not exist in the library ncurses (initscr32). + Simply use when you link to ncurses. + + Furthermore in Solaris, the header is in a subdirectory + /usr/include/ncurses and not available via pkg-config. + + ASTERISK-15331 + ASTERISK-14935 + ASTERISK-12382 + ASTERISK-9107 + + Change-Id: Ife367776b0ccf17d3fefed868245376bfb93745d + +2018-07-28 08:00 +0000 [3aa6be6b51] Joshua Colp + + * res_pjsip_pubsub: Use ast_true for "prune_on_boot". + + Change-Id: Iedec4e7390b3e821987681da24d0298632b9873d + +2018-07-28 07:39 +0000 [0a4d58735f] Alexander Traud + + * BuildSystem: Enable Jansson in Solaris 11. + + In Solaris, the header is in /usr/include/jansson. To find + Jansson even in such a subdirectory, the tool pkg-config is queried via + AST_PKG_CONFIG_CHECK. For those platforms, which do not list Jansson via + pkg-config, the previous check remains and is executed thereafter. + + Because the check for the NetBSD Editline library uses the tool pkg-config + the code of PKG_PROG_PKG_CONFIG must be used. Because that check happens + earlier than Jansson, it must be placed in front of that. + + ASTERISK-27991 + + Change-Id: I69ea0f379f87a50049654b2487c76ee1c04fa53a + +2018-07-24 13:44 +0000 [e5ae04b48b] Richard Mudgett + + * res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header + + This patch adds regular expression support to make the identify section's + match_header option more useful when attempting to match complex headers + like the 'To' or 'From' headers. The 'From' header has variable + components such as the tag parameter that you cannot predict. To specify + a regular expression put slashes around the regular expression in place of + the header value. + + [identify-alice] + type=identify + endpoint=alice + match_header=From: // + + * Added regex support to match_header so you could match a 'To' header + among other complex headers. + + Fixed reported crashes when trying to match special headers like 'Contact'. + The identify section's match_header method used code that assumed you were + matching a generic header. Any other type of header could cause a crash + if the header structure variant did not match the generic header enough. + + * Made use code that will work for any header type instead of code + specific to generic headers. + + Other fixes while in the area: + + * Made check all headers of the requested name. + * Added some more sanity checks to the configured identify matching + options when applying the configuration. + + ASTERISK-27548 + + Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1 + +2018-07-27 10:46 +0000 [4265391859] Joshua Colp + + * res_pjsip_pubsub: Treat "prune_on_boot" as a yes / no. + + The alembic for the PJSIP subscription persistence table has the + "prune_on_boot" field as a boolean. While in Asterisk we are + tolerant of many different definitions of true and false in the + database we only accept "yes" and "no". This change makes the + field treated as a yes/no instead of an integer, thus storing + "yes" and "no" instead of "1" and "0". + + Change-Id: Ic8b9211b36babefe78f70def6828a135a6ae7ab6 + +2018-07-27 08:26 +0000 [870fe7f60c] Alexander Traud + + * res_rtp_asterisk: In Developer Mode, do not require OpenSSL. + + OpenSSL is an optional external library and should stay optional even when + Developer Mode is configured. + + ASTERISK-27990 + + Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b + +2018-07-26 18:54 +0000 [116a599b7e] George Joseph + + * CI: Fix placement of job summary statments + + Change-Id: Iace19e718f4e8fb48eb7dc9f98af53b115cc45f3 + +2018-07-26 12:52 +0000 [709f4b81e7] Corey Farrell + + * loader: Process dependencies for built-in modules. + + With the new module loader it was missed that built-in modules never + parsed dependencies from mod->info into vectors of mod. This caused + manager to be initialized before acl (named_acl). If manager.conf + used any named ACL's they would not be found and result in no ACL being + applied to the AMI user. + + In addition to the manager ACL fix this adds "extconfig" to all builtin + modules which support realtime configuration. This only matters if one + of the builtin modules is configured with 'preload', depending on + "extconfig" will cause config.c to automatically be initialize during + the preload stage. + + Change-Id: I482ed6bca6c1064b05bb538d7861cd7a4f02d9fc + +2018-07-18 09:32 +0000 [cb276b5085] Emmanuel BUU + + * res_rtp_asterisk: Avoid merging command and regular T.140 text packets + + When realtime text packets are to be sent, the text is accumulated in a + buffer and sent regularly by a timer. It can happen that commands such as + a backspace, CR, or LF get merged with regular text. This breaks some + UAs. + + The proposed change: + * We test if the current packet contains a command. If so we send the + buffer immediately. + * We test if the buffer contained a command. If so we send the buffer + immediately. + * We accumulate the text (or the command) in the buffer. + + ASTERISK-27970 + + Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462 + +2018-07-26 11:34 +0000 [e55cad967e] George Joseph + + * CI: Add docker info to job summary + + Change-Id: I45d52005a9b692ad303c11792f226ace1e449901 + +2018-07-23 13:49 +0000 [852e157b19] Corey Farrell + + * Build System: Create 'make install-configs' target. + + This target requires specifying CONFIG_SRC=path_to_configs. This can be + used to install custom configs for the Asterisk build while still + performing directory replacements on asterisk.conf. + + Modify internal INSTALL_CONFIGS so first argument requires full path to + the config sources relative to Asterisk source root. + + Change-Id: Idcd841df3c8d5bfe23d566bb9e2e448e9df4f8ab + +2018-07-25 15:33 +0000 [783bff0637] Kevin Harwell + + * json.c: improve ast_json_to_ast_variables performance + + When converting from a json object to an ast variables list the conversion + algorithm was doing a complete traversal of the entire variables list for + every item appended from the json structure. + + This patch makes it so the list is no longer traversed for each new ast + variable being appended. + + Change-Id: I8bf496a1fc449485150d6db36bfc0354934a3977 + +2018-07-25 05:32 +0000 [66f581313f] Joshua Colp + + * devicestate: Don't create topic when change isn't cached. + + When publishing a device state the change can be marked as being + cachable or not. If it is not cached the change is just published + to all interested and not stored away for later query. This was not + fully taken into account when publishing in stasis. The act of + publishing would create a topic for the device even if it may be + ephemeral. + + This change makes it so messages which are not cached won't create + a topic for the device. If a topic does already exist it will be + published to but otherwise the change will only be published to + the device state all topic. + + ASTERISK-27591 + + Change-Id: I18da0e8cbb18e79602e731020c46ba4101e59f0a + +2018-07-25 10:20 +0000 [3dcf26cb94] George Joseph + + * CI: Explicitly pass BRANCH_NAME to buildAsterisk and installAsterisk + + Change-Id: I652f4a0ea5107c778e27a78bccb67b18b0c4e087 + +2018-07-24 13:29 +0000 [797835c5b9] George Joseph + + * CI: Add options to initialize and cleanup database to runTestsuite.sh + + Change-Id: I352333233bab5377723bf37d490ba84fc55bc853 + +2018-07-25 09:07 +0000 [05a4b448af] Corey Farrell + + * CI: Do not `mkdir 2`. + + Change-Id: Ib7377d26a6c98b38bad463f47c84f1875ac84eb7 + +2018-07-25 07:34 +0000 [2f275f8472] Corey Farrell + + * Build System: Silence build of bundled jansson. + + Change-Id: I7392c79c0173057f5378010bf1fe65e300e8fc56 + +2018-07-25 07:13 +0000 [ceb199e19f] George Joseph + + * CI: RefDebug: Fix reference to testsuite URL + + Change-Id: I0ee41d95a87f0d97b01f2757012b846bcfe6443d + +2018-07-24 14:28 +0000 [af5984d694] Corey Farrell + + * Build System: Fix bundled jansson install. + + Update the bundled jansson Makefile to do nothing during Asterisk + install, use a target that is not phony to initiate the jansson make and + install. + + Change-Id: I7643cc3d39af9feba8fc0da676b646efc5f8b3bb + +2018-07-24 10:43 +0000 [cdb725526e] Corey Farrell + + * CI: Use bundled jansson if needed. + + Use pkg-config to determine if jansson is at least 2.11, enabled bundled + version otherwise. + + Change-Id: Ib555a8b72ff6f6925f9280ef035caa0b91ca4bd2 + +2018-07-24 04:57 +0000 [c5bac9ed90] Florian Floimair + + * res_pjsip: Change log message from error to warning for valid use cases + + If a SIP MESSAGE is triggered for an endpoint that is currently not registered + - and therefore has no valid contact associated - an error message was logged. + Since this is a valid request in a valid use cases this is now changed to a + warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list. + + Change-Id: I55eb62d2712818a58c7532119dec288bd98cf0c0 + +2018-07-24 05:39 +0000 [f827f36ff3] George Joseph + + * CI: Add --privileged flag to docker options + + Change-Id: If92d55f15306e55dd7091ac3c47b13ebbbb03488 + +2018-07-24 05:22 +0000 [eed429c811] George Joseph + + * CI: Set correct user:group when publishing docs + + Change-Id: Ibabeb9ac730d9755cf54318d0da74771c939b86b + +2018-07-23 12:21 +0000 [0504594a3e] Richard Mudgett + + * core: AST_DEVMODE no longer affects ABI. + + Remove AST_DEVMODE from the AST_BUILDOPTS list and the AST_BUILDOPTS_SUM + calculation as it no longer affects API/ABI compatibility. + + Change-Id: Id5bd6dfade173a53b3a49f715586b86e3fb24acb + +2018-07-20 16:21 +0000 [0f8657aae9] Richard Mudgett + + * asterisk.c: Make displayed copyright always consistent + + Change-Id: I4f5499486e8ec90d7c7ffeebc659ceda1db6d5b5 + +2018-07-23 10:23 +0000 [3b78651c3c] Corey Farrell + + * CI: Split --test-command argument. + + The --test-command argument has now been split, unit tests now use + `--unittest-command` and the testsuite uses --testsuite-command. + + This will make it easier to create a script which run everything by + forwarding the same arguments to all CI scripts. + + Change-Id: Ia54aa4848eaffbdf13175fcda40fc0b23080ad71 + +2018-07-20 06:20 +0000 [ba8f2c401c] George Joseph + + * xmldoc.c: Fix dump of xml document + + The "xmldoc dump" cli command was simply concatenating xml documents + into the output file. The resulting file had multiple "xml" + processing instructions and multiple root elements which is illegal. + Normally this isn't an issue because Asterisk has only 1 main xml + documentation file but codec_opus has its own file so if it's + downloaded and you do "xmldoc dump", the result is invalid. + + * Added 2 new functions to xml.c: + ast_xml_copy_node_list creates a copy of a list of children. + ast_xml_add_child_list adds a list to an existing list. + + * Modified handle_dump_docs to create a new output document and + add to it the children from each input file. It then dumps the + new document to the output file. + + Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07 + +2018-07-21 11:58 +0000 [0ee061326a] Corey Farrell + + * CI: Fix mkdir CACHE_DIR. + + Change-Id: Ic9f9a61e230047836c836206731f8ff7eb3538c9 + +2018-07-21 10:48 +0000 [747b65f675] Corey Farrell + + * build_tools/make_version: Get MAINLINE_BRANCH from .gitreview. + + Use .gitreview defaultbranch setting to determine the mainline branch. + This allows the script to be used against other directories which might + not be on the same defaultbranch. This can be used by CI scripts to + report the testsuite version being used: + ./build_tools/make_version ${TESTSUITE_DIR} + + Change-Id: Ifdad4a9d8a26138c41bc6b630ecc3e34ea1c2758 + +2018-07-22 10:41 +0000 [33f855bb69] Joshua Colp + + * sched: Make ABI compatible between dev mode and non-dev mode. + + In the past there was an assertion in the ast_sched_del function + and in order to ensure it was useful the calling function name, + line number, and filename had to be passed in. This cause the ABI + to be different between dev mode and non-dev mode. + + This assertion is no longer present so the special logic can be + removed to make it the same between them both. + + Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8 + +2018-07-20 15:52 +0000 [09c4be9433] Richard Mudgett + + * asterisk.c: Update displayed copyright year for v16 release. + + Change-Id: I60622731d928ee9506b1d28934095f0dc3e5306e + +2018-07-16 15:08 +0000 [ee154464d7] Corey Farrell + + * Enable bundling of jansson, require 2.11. + + Change-Id: Ib3111b151d37cbda40768cf2a8a9c6cf6c5c7cbd + +2018-07-20 09:25 +0000 [fa6d5db229] Corey Farrell + + * CI: Fix logger.conf for unit tests. + + Change-Id: Idea59d60eab20105de50b34f0f0d506e6ef55d5c + +2018-07-19 10:34 +0000 [739cfe128d] George Joseph + + * CI: Add wiki doc publish to periodics + + Change-Id: I29ba26134e5083bc6788ede235f1a5d4383c148a + +2018-07-20 06:54 +0000 [2c9757bc90] Joshua Colp + + * res_pjsip: Update default keepalive interval to 90 seconds. + + A change recently went in which disabled the built-in PJSIP + keepalive. This defaulted to 90 seconds and kept TCP/TLS + connections alive. Disabling this functionality has resulted + in a behavior change of not doing keepalives by default resulting + in TCP/TLS connections dropping for some people. + + This change makes our default keepalive interval 90 seconds + to match the previous behavior and preserve it. + + ASTERISK-27978 + + Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6 + +2018-07-19 16:17 +0000 [e6bb2efaab] Richard Mudgett + + * res_pjsip: Update endpoint transport option documentation. + + Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52 + +2018-07-19 13:27 +0000 [8a100ca52b] Richard Mudgett + + * pjsip_resolver.c: Use replacement function + + * Use the replacement function ast_sip_push_task_wait_servant() instead of + the deprecated ast_sip_push_task_synchronous(). + + Change-Id: I145b550ba7054640c7faa3b644e63137f505c612 + +2018-07-18 17:13 +0000 [d7db9f2152] Corey Farrell + + * contrib: Update systemd README.txt. + + Mention need to compile Asterisk with systemd development package + installed. + + ASTERISK-27968 + + Change-Id: Ib3a973be403c61cbe09572b0f912fb1aa1bff026 + +2018-07-18 14:20 +0000 [e01e636959] Joshua Colp + + * Update UPDATE.txt for 16 and update ARI stubs. + + Copied UPGRADE.txt -> UPGRADE-16.txt + Created new UPGRADE.txt + + Updated ARI stubs version to 17. + + Change-Id: I4210e53f8022a2a68c7653595bdd13fbebac41ee + +2018-08-08 16:02 +0000 Asterisk Development Team + + * asterisk 16.0.0-rc1 Released. + +2018-07-27 13:23 +0000 [d3789cc420] George Joseph + + * CI: Add optional uninstall step before installing asterisk + + Change-Id: I7dedf1e925eafc3a0adf01dd9dfbe44eb642aab7 + +2018-07-28 08:00 +0000 [89b669a227] Joshua Colp + + * res_pjsip_pubsub: Use ast_true for "prune_on_boot". + + Change-Id: Iedec4e7390b3e821987681da24d0298632b9873d + +2018-07-27 10:46 +0000 [0028db48cc] Joshua Colp + + * res_pjsip_pubsub: Treat "prune_on_boot" as a yes / no. + + The alembic for the PJSIP subscription persistence table has the + "prune_on_boot" field as a boolean. While in Asterisk we are + tolerant of many different definitions of true and false in the + database we only accept "yes" and "no". This change makes the + field treated as a yes/no instead of an integer, thus storing + "yes" and "no" instead of "1" and "0". + + Change-Id: Ic8b9211b36babefe78f70def6828a135a6ae7ab6 + +2018-07-26 18:54 +0000 [24e4e45177] George Joseph + + * CI: Fix placement of job summary statments + + Change-Id: Iace19e718f4e8fb48eb7dc9f98af53b115cc45f3 + +2018-07-26 12:52 +0000 [c384a4cdcd] Corey Farrell + + * loader: Process dependencies for built-in modules. + + With the new module loader it was missed that built-in modules never + parsed dependencies from mod->info into vectors of mod. This caused + manager to be initialized before acl (named_acl). If manager.conf + used any named ACL's they would not be found and result in no ACL being + applied to the AMI user. + + In addition to the manager ACL fix this adds "extconfig" to all builtin + modules which support realtime configuration. This only matters if one + of the builtin modules is configured with 'preload', depending on + "extconfig" will cause config.c to automatically be initialize during + the preload stage. + + Change-Id: I482ed6bca6c1064b05bb538d7861cd7a4f02d9fc + +2018-07-26 11:34 +0000 [9f1041c4d0] George Joseph + + * CI: Add docker info to job summary + + Change-Id: I45d52005a9b692ad303c11792f226ace1e449901 + +2018-07-25 15:33 +0000 [c5761ee58e] Kevin Harwell + + * json.c: improve ast_json_to_ast_variables performance + + When converting from a json object to an ast variables list the conversion + algorithm was doing a complete traversal of the entire variables list for + every item appended from the json structure. + + This patch makes it so the list is no longer traversed for each new ast + variable being appended. + + Change-Id: I8bf496a1fc449485150d6db36bfc0354934a3977 + +2018-07-25 10:20 +0000 [cfd61ba237] George Joseph + + * CI: Explicitly pass BRANCH_NAME to buildAsterisk and installAsterisk + + Change-Id: I652f4a0ea5107c778e27a78bccb67b18b0c4e087 + +2018-07-24 13:29 +0000 [a81870110a] George Joseph + + * CI: Add options to initialize and cleanup database to runTestsuite.sh + + Change-Id: I352333233bab5377723bf37d490ba84fc55bc853 + +2018-07-25 09:07 +0000 [4a01be5c80] Corey Farrell + + * CI: Do not `mkdir 2`. + + Change-Id: Ib7377d26a6c98b38bad463f47c84f1875ac84eb7 + +2018-07-25 07:34 +0000 [e6f2bae0cc] Corey Farrell + + * Build System: Silence build of bundled jansson. + + Change-Id: I7392c79c0173057f5378010bf1fe65e300e8fc56 + +2018-07-25 07:13 +0000 [f1156f0cfd] George Joseph + + * CI: RefDebug: Fix reference to testsuite URL + + Change-Id: I0ee41d95a87f0d97b01f2757012b846bcfe6443d + +2018-07-24 14:28 +0000 [7e99090c9d] Corey Farrell + + * Build System: Fix bundled jansson install. + + Update the bundled jansson Makefile to do nothing during Asterisk + install, use a target that is not phony to initiate the jansson make and + install. + + Change-Id: I7643cc3d39af9feba8fc0da676b646efc5f8b3bb + +2018-07-24 10:43 +0000 [b32adca9b4] Corey Farrell + + * CI: Use bundled jansson if needed. + + Use pkg-config to determine if jansson is at least 2.11, enabled bundled + version otherwise. + + Change-Id: Ib555a8b72ff6f6925f9280ef035caa0b91ca4bd2 + +2018-07-24 05:39 +0000 [e22cbe7c17] George Joseph + + * CI: Add --privileged flag to docker options + + Change-Id: If92d55f15306e55dd7091ac3c47b13ebbbb03488 + +2018-07-24 05:22 +0000 [3509ada06f] George Joseph + + * CI: Set correct user:group when publishing docs + + Change-Id: Ibabeb9ac730d9755cf54318d0da74771c939b86b + +2018-07-23 12:21 +0000 [008d304be2] Richard Mudgett + + * core: AST_DEVMODE no longer affects ABI. + + Remove AST_DEVMODE from the AST_BUILDOPTS list and the AST_BUILDOPTS_SUM + calculation as it no longer affects API/ABI compatibility. + + Change-Id: Id5bd6dfade173a53b3a49f715586b86e3fb24acb + +2018-07-23 10:23 +0000 [5dbbc68311] Corey Farrell + + * CI: Split --test-command argument. + + The --test-command argument has now been split, unit tests now use + `--unittest-command` and the testsuite uses --testsuite-command. + + This will make it easier to create a script which run everything by + forwarding the same arguments to all CI scripts. + + Change-Id: Ia54aa4848eaffbdf13175fcda40fc0b23080ad71 + +2018-07-21 11:58 +0000 [2a13a4344e] Corey Farrell + + * CI: Fix mkdir CACHE_DIR. + + Change-Id: Ic9f9a61e230047836c836206731f8ff7eb3538c9 + +2018-07-22 10:41 +0000 [9742fb07c9] Joshua Colp + + * sched: Make ABI compatible between dev mode and non-dev mode. + + In the past there was an assertion in the ast_sched_del function + and in order to ensure it was useful the calling function name, + line number, and filename had to be passed in. This cause the ABI + to be different between dev mode and non-dev mode. + + This assertion is no longer present so the special logic can be + removed to make it the same between them both. + + Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8 + +2018-07-20 15:52 +0000 [2c51079d05] Richard Mudgett + + * asterisk.c: Update displayed copyright year for v16 release. + + Change-Id: I60622731d928ee9506b1d28934095f0dc3e5306e + +2018-07-16 15:08 +0000 [3cdffa1342] Corey Farrell + + * Enable bundling of jansson, require 2.11. + + Change-Id: Ib3111b151d37cbda40768cf2a8a9c6cf6c5c7cbd + +2018-07-20 09:25 +0000 [136d855f69] Corey Farrell + + * CI: Fix logger.conf for unit tests. + + Change-Id: Idea59d60eab20105de50b34f0f0d506e6ef55d5c + +2018-07-19 10:34 +0000 [0c1513d8a0] George Joseph + + * CI: Add wiki doc publish to periodics + + Change-Id: I29ba26134e5083bc6788ede235f1a5d4383c148a + +2018-07-20 06:20 +0000 [61a974ed4e] George Joseph + + * xmldoc.c: Fix dump of xml document + + The "xmldoc dump" cli command was simply concatenating xml documents + into the output file. The resulting file had multiple "xml" + processing instructions and multiple root elements which is illegal. + Normally this isn't an issue because Asterisk has only 1 main xml + documentation file but codec_opus has its own file so if it's + downloaded and you do "xmldoc dump", the result is invalid. + + * Added 2 new functions to xml.c: + ast_xml_copy_node_list creates a copy of a list of children. + ast_xml_add_child_list adds a list to an existing list. + + * Modified handle_dump_docs to create a new output document and + add to it the children from each input file. It then dumps the + new document to the output file. + + Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07 + +2018-07-20 06:54 +0000 [50a26b15a3] Joshua Colp + + * res_pjsip: Update default keepalive interval to 90 seconds. + + A change recently went in which disabled the built-in PJSIP + keepalive. This defaulted to 90 seconds and kept TCP/TLS + connections alive. Disabling this functionality has resulted + in a behavior change of not doing keepalives by default resulting + in TCP/TLS connections dropping for some people. + + This change makes our default keepalive interval 90 seconds + to match the previous behavior and preserve it. + + ASTERISK-27978 + + Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6 + +2018-07-18 14:19 +0000 [958f76205b] Joshua Colp + + * Update mainline version for the 16 branch. + + Change-Id: I4d36277d10335349d83ae218fa10fee99c3e4c14 + +2018-07-18 14:18 +0000 [e7a76ffee1] Joshua Colp + + * Update ARI version for master/16. + + ARI goes from 3.0.0 to 4.0.0 + + Change-Id: I0649fa34926dc4fc89a166f1d2e3bbd965ef9ebe + +2018-05-29 09:31 +0000 [fe78d374b0] Alexander Traud + + * pjproject_bundled: Repair ./configure --with-ssl=PATH. + + Previously, Asterisk did not tell its bundled PJProject about this configure + parameter. Therefore, PJProject used the platform provided OpenSSL always. + + ASTERISK-27880 + + Change-Id: Iea545aec854dd0e2c061c69bb118a76ce56c5dc6 + +2018-05-10 13:11 +0000 [5bacde37a2] Ben Ford + + * res_rtp_asterisk: Add support for sending NACK requests. + + Support has been added for receiving a NACK request and handling it. + Now, Asterisk can detect when a NACK request should be sent and knows + how to construct one based on the packets we've received from the remote + end. A buffer has been added that will store out of order packets until + we receive the packet we are expecting. Then, these packets are handled + like normal and frames are queued to the core like normal. Asterisk + knows which packets to request in the NACK request using a vector + which stores the sequence numbers of the packets we are currently missing. + + If a missing packet is received, cycle through the buffer until we reach + another packet we have not received yet. If the buffer reaches a certain + size, send a NACK request. If the buffer reaches its max size, queue all + frames to the core and wipe the buffer and vector. + + According to RFC3711, the NACK request must be sent out in a compound + packet. All compound packets must start with a sender or receiver + report, so some work was done to refactor the current sender / receiver + code to allow it to be used without having to also include sdes + information and automatically send the report. + + Also added additional functionality to ast_data_buffer, along with some + testing. + + For more information, refer to the wiki page: + https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements + + ASTERISK-27810 #close + + Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d + +2018-07-18 11:12 +0000 [59323121f3] Joshua Colp + + * res_sorcery_config: Allow configuration section to be used based on name. + + A problem I've seen countless times is a global or system section + for PJSIP not getting applied. This is inevitably the result of + the "type=" line missing. This change alleviates that problem. + + The ability to specify an explicit section name has been + added to res_sorcery_config. If the configured section + name matches this and there are no unknown things configured + the section is taken as being for the given type. + + Both the PJSIP "global" and "system" types now support this + so you can just name your section "global" or "system" and it + will be matched and used, even without a "type=" line. + + ASTERISK-27972 + + Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893 + +2018-07-17 05:24 +0000 [134e2f0ddc] Joshua Colp + + * module: Remove deprecated modules and update support levels. + + I have removed the STATIC_BUILD option immediately as it has not + been maintained in many years and is non-functional. + + ASTERISK-27965 + + Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7 + +2018-07-18 11:34 +0000 [94dd0544e5] Chris-Savinovich + + * stasis: Improve message type "Use of before/init after destruction" + + Fixes issue where error msg + "Use of before/init after destruction" + was being printed on disabled messages + in dev mode. With this + fix if message is disabled + a warning will print. + + ASTERISK-25548 + Change-Id: Ie0d866d1cbc60c16dbef08bc65e99505c3c1adfa + +2018-07-17 14:12 +0000 [993ba84cd3] Nick French + + * SRTP: Lower SDES key lifetime minimum to 2^20 + + SRTP SDES key lifetime support was added in ASTERISK_17899. + + In that addition, the minimum key lifetime to be accepted was + set at the 10 hours @ 20ms/packet = 1800000 packets. + + The firmware in the obi1xx ATA uses a hardcoded lifetime of + 2^20 packets. + + Lower the limit to 2^20 to support a wider field of clients. + + ASTERISK-27967 #close + + Change-Id: I81a0703c595a0c9101dfdf02300149a3cc39bf94 + +2018-07-17 11:09 +0000 [fcc0a6fe8a] George Joseph + + * CI: Fix merge strategy + + Change-Id: I5e3fb6adfa6cbf694c0deecf02e3879297b0c12e + +2018-07-17 10:41 +0000 [3e5a6a6cfc] George Joseph + + * CI: Fix regex in daily and ref_debug jobs + + Change-Id: Icf2e67818b2155a158d2390b138613e1f653ea92 + +2018-07-17 09:09 +0000 [0e8976116f] Nick French + + * res_pjsip: Remove spurious error logging when printing silent headers + + Asterisk patched the pjproject source to avoid crashing when pjproject + sip_msg headers are encountered with NULL vptr's, but the patch also + output error messages for some valid headers which simply did not need + to be added to the message itself, such as hidden route headers. + + pjproject has since applied a similar patch to their baseline to avoid + crashes, but their version also avoids the spurious error logging. + + Lets use their patch instead. + + ASTERISK-27961 #close + + Change-Id: I2ddbd82c8da10e0dcc9807a48089d1f3c2d6e389 + +2018-07-17 10:15 +0000 [fa333dedd0] George Joseph + + * CI: Add pre-build merge back in as RECURSIVE + + Change-Id: I0ff1730ef4a4f0ac9f18ccc9bc0dfe7a782f57a8 + +2018-07-17 09:01 +0000 [2553255ace] George Joseph + + * CI: Remove pre-build merge from gates and checks + + Change-Id: Ibc151f63dcec4db847915c2f3cbe5b467dd59574 + +2018-07-17 07:13 +0000 [524f900382] George Joseph + + * CI: Fix logic inversion in runTestsuite + + Change-Id: I56399aa384468f45494c2c3650420563a0b6efe1 + +2018-07-17 04:03 +0000 [0af4a558da] George Joseph + + * CI: Add teardownRealtime + + Change-Id: I2fe55c38607eaec2fbf69ef23a5019e0c443a64b + +2018-07-15 13:58 +0000 [49f83a7490] Corey Farrell + + * loader: Fix startup issues. + + * Merge the preload and load stages, use load ordering to try preload's + first. This fixes an issue where `preload=res_config_curl` would fail + unless res_curl and func_curl were also preloaded. Now it is only + required that those modules be loaded during startup: autoload or + regular load is good enough. + * The configuration option `require` and `preload-require` were only + effective if the modules failed to load. These options will now abort + Asterisk startup if required modules fail to reach the 'Running' + state. + * Missing or invalid 'module.conf' did not prevent startup. Asterisk + doesn't do anything without modules so this a fatal error. + + Change-Id: Ie4176699133f0e3a823b43f90c3348677e43a5f3 + +2018-07-16 13:30 +0000 [a9cef123d9] George Joseph + + * CI: Prevent Jenkins from triggering jobs back to itself + + Change-Id: I9cae8bb3d1a2cea335d3ccd88d471832549666fd + +2018-07-13 18:26 +0000 [5febc995df] Richard Mudgett + + * Build: Fix modules getting their optimization setting overridden. + + Asterisk modules that use PJPROJECT services have their compiler + optimization and possibly their symbolic debug options overridden by the + PJPROJECT configure script selected settings. + + * We need to filter-out any -O and -g options in PJ_CFLAGS before echoing + out the result so the PJPROJECT_INCLUDE variable does not override the + Asterisk module settings when using bundled PJPROJECT. + + NOTE: This patch only has an effect when using bundled PJPROJECT. + + ASTERISK-27563 + + Change-Id: If124169735ecf572ad1535cd43bff94cb44d5b30 + +2018-07-16 11:08 +0000 [d15ef68892] George Joseph + + * CI: runUnittests: loop a few times on waitfullybooted + + Change-Id: Icebc0d013896f3b2a7214945cac60647435c1651 + +2018-07-16 10:49 +0000 [252c4284df] George Joseph + + * CI: Add realtime checks to dailies + + Change-Id: I6dc8ab1679b3505c6dde1d47e1b9276df47814f8 + +2018-07-16 09:13 +0000 [1a52ab70c7] George Joseph + + * CI: Add weekly REF_DEBUG testsuite run + + Change-Id: I5b581d0a0d1d1bb9b38961d40b112fb448355037 + +2018-07-16 08:44 +0000 [9633e9dfd7] George Joseph + + * CI: Fix bad reporting of status by the verification pub + + Change-Id: I6f31a130b3ba0187149aaaa2ce94195a79e0f6a6 + +2018-07-16 07:16 +0000 [b8d75bbb37] George Joseph + + * CI: Make build tag an acceptable docker name + + Change-Id: I3a4b8a4a9c488ddabf9daf651dc1334222056f38 + +2018-07-13 22:44 +0000 [0885ab8afc] Corey Farrell + + * Fix declaration of PBX_CURL for ./configure --without-libcurl + + When `--without-libcurl` is used PBX_CURL is never set. Set default + value 0 so the proper value is passed to menuselect. + + Change-Id: I03e2842a00899cbca2dbde52bb1f6636d54bae1e + +2018-07-10 13:28 +0000 [34f3fe9552] George Joseph + + * app_confbridge: Use the SDP 'label' attribute to correlate users + + Previously, the msid "label" attribute was used to correlate + participant info but because streams could be reused, the msid + wasn't being updated correctly when someone left the bridge and + another joined. + + Now, instead of looking for the msid attribute on a channel's streams, + app_confbridge sets an "SDP:LABEL" attribute on the stream which + res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label" + attribute to the current sdp. + + Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5 + +2018-07-13 06:56 +0000 [e8727fcfa8] George Joseph + + * CI: Add daily periodics to CI + + Change-Id: I26933e73928e091ae72e838c02f4f2ec7c3983d6 + +2018-07-11 11:57 +0000 [e19080a184] Alexander Traud (License 6520) + + * Bundled PJPROJECT: Disable internal connection oriented keep-alive. + + Turn off the periodic sending of CRLNCRLN. Default is on (90 seconds), + which conflicts with the global section's keep_alive_interval option in + pjsip.conf. + + patches: + pjsip_keep_not_alive.patch submitted by Alexander Traud (License 6520) + + ASTERISK-27347 + + Change-Id: I6a197f56e1830d3b7e5ec70f17025840a290b057 + +2018-07-09 04:42 +0000 [1445384699] Torrey Searle + + * res_pjsip_sdp_rtp: include ice in ANSWER only if offered + + Keep track if ICE candidates were in the SDP offer & only put them + in the corresponding SDP answer if the offer condaind ICE candidates + + ASTERISK-27957 #close + + Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92 + +2018-07-12 16:34 +0000 [33a84745d0] George Joseph + + * CI: Add Asterisk Gates + + Change-Id: I7e2467f9120812551238d8005deb97f965279205 + +2018-07-11 15:55 +0000 [65b002ab8f] George Joseph + + * CI: Remove duplicate checkout + + Change-Id: If5f925b4c4ed7000b153f3ed8386ce2140c886f8 + +2018-07-11 15:09 +0000 [ba8f8a2813] George Joseph + + * CI: Update cleanup steps and permissions + + Change-Id: I7ca92935979d94845af8e1caf4468cbd6209b7de + +2018-07-11 14:54 +0000 [ad36c4ba9b] George Joseph + + * CI: Fix log artifact paths + + Change-Id: I55136de8f4d9c3b56bd4d054306a187bb04a4b7d + +2018-07-11 14:45 +0000 [4842af6364] George Joseph + + * CI: Remove CleanBeforeCheckout option for testsuite + + Change-Id: I510231c9087f7be5272b8ef3f3223eadaaffb754 + +2018-07-11 14:00 +0000 [3dfc37c60a] George Joseph + + * CI: Move gates into source repo + + Change-Id: If028ede5f3b127fa274c63ce166bc04ad7c1e5db + +2018-07-11 06:14 +0000 [b302ee6bd5] George Joseph + + * CI: Initial commit for moving CI into source repo + + Create tests/CI directory and add files used by Jenkins to + build and test Asterisk. + + With this commit, Jenkins will run the Asterisk Unit Tests using + the Jenkinsfile at tests/CI/unittests.jenkinsfile. Bash scripts + to do the actual building and testing are also in the same directory. + Output is placed in tests/CI/output so that directory has been + added to .gitignore. + + Change-Id: I9448065465e6de2b878634510ace8fd1ef378608 + +2018-07-06 17:00 +0000 [f7137e1230] Joshua Elson + + * res_parking: Add dialplan function for lot channel + + This commit adds a new function to res_parking. + + This function, PARK_GET_CHANNEL allows the retrieval + of the channel name of the channel occupying the parking slot. + + ASTERISK-22825 #close + + Change-Id: Idba6ae55b8a53f734238cb3d995cedb95c0e7b74 + +2018-06-23 01:33 +0000 [10de9fcbf1] Alexander Traud + + * chan_ooh323: IPTOS_MINCOST is not defined on Solaris. + + Furthermore, is required for SIOCGIF*. + + ASTERISK-27938 + + Change-Id: Idc9153ece769944765b66122efb11728d8d8ebde + +2018-07-06 15:05 +0000 [5bb874ee09] Kevin Harwell + + * res_pjsip_session: sdp group:BUNDLE attribute being truncated + + When setting/appending the media id's to the bundle group attribute a '-1' was + being passed to the 'ast_str_set/append' function for the 'max_len' parameter. + This essentially capped the length of the string to what it was originally + allocated with. In this case 64 bytes. + + This patch makes it so a '0' is passed as in for the 'max_len', which means + "no maximum length". + + ASTERISK-27955 #close + + Change-Id: Iec565df6600401d54a502854a53d19bb4cc34876 + +2018-07-05 16:02 +0000 [96abe79ddf] Alexei Gradinari + + * res_pjsip_pubsub: segfault in function publish_expire + + The function pubsub_on_rx_publish_request incorrectly uses + of AST_SCHED_REPLACE_UNREF. + + The AST_SCHED_REPLACE_UNREF should unref old '_data'. + + Because of this, there may be a double unref + of variable 'publication' when ast_sched_del is unsuccessful + that leads to use after free of the 'publication' in publish_expire. + + ASTERISK-27956 #close + + Change-Id: Ie0f0cfc7e036953d890b188656010b325a5cdc82 + +2018-07-06 09:04 +0000 [c1e49720fa] George Joseph + + * test.c: Make output jUnit compatible + + Separate "name" into "classname" and "name". + Use '.' for classname separator instead of '/'. + Prefix reserved words with '_'. + Wrap output with a top-level "testsuites" element. + + Change-Id: Iec1a985eba1c478e5c1d65d5dfd95cb708442099 + +2018-07-06 07:57 +0000 [8f42447c68] George Joseph + + * res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint + + A new option 'suppress_q850_reason_headers' has been added to the + endpoint object. Some devices can't accept multiple Reason headers and + get confused when both 'SIP' and 'Q.850' Reason headers are received. + This option allows the 'Q.850' Reason header to be suppressed. + The default value is 'no'. + + ASTERISK-27949 + Reported-by: Ross Beer + + Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1 + +2018-07-05 15:43 +0000 [c9f8e068ed] Joshua Colp + + * res_pjsip_t38: Decline T.38 stream on failure case. + + When negotiating an incoming T.38 stream the code incorrectly + returned failure instead of a decline for the stream when a + problem occurred or the configuration didn't allow it. This + resulted in SDP offers being rejected with a 488 response + in all cases, even when another valid stream was present. + + This change makes it so the stream is now declined. If no + streams are accepted a 488 response is sent while if at least + one stream is accepted all the declined streams are, well, + declined. + + ASTERISK-27763 + + Change-Id: I88bcf793788c412a9839d111a5c736bf6867807c + +2018-07-02 18:43 +0000 [d5db664d70] Richard Mudgett + + * res_pjsip_t38.c: Be smarter about how we respond when T.38 is disabled. + + We were blindly responding with AST_T38_REFUSED when ANY T.38 control + frame came accross the bridge. This causes T.38 Gateway to get confused + and the T.38 session to get in a strange state. + + * Made the T.38 framehook only respond to request frames and ignore + response frames. + + ASTERISK-27657 + ASTERISK-27080 + + Change-Id: I5fb5967c7d1efb30a7ff375f82887ca82a55b05b + +2018-07-03 12:10 +0000 [0aff1a278e] Richard Mudgett + + * res_pjsip/pjsip_transport_management.c: Fix deadlock with transport keep alive. + + Using the keep_alive_interval option can result in a deadlock between the + pjproject transport manager group lock and the monitored transports ao2 + container lock. The pjproject transport manager group lock has to be + superior in the locking order to the monitored transports ao2 container + lock because of pjproject callbacks called when already holding the group + lock. The lock inversion happens when Asterisk attempts to send a keep + alive packet over the reliable transports. + + * Made keepalive_transport_thread() iterate over the monitored transports + container rather than use the ao2_callback() method. This avoids holding + the container lock when sending the keep alive packet. + + ASTERISK-26686 + + Change-Id: I5d5392a52e698bbe41a93f7d8e92bf0e61fe3951 + +2018-07-02 18:44 +0000 [de5144e751] Joshua Colp + + * pjsip: Clarify certificate configuration for Websocket. + + The Websocket transport uses the built-in HTTP server. As a result + the TLS configuration is done in http.conf and not in pjsip.conf. + + This change adds a warning if this is configured in pjsip.conf and + also clarifies in the sample configuration file. + + Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9 + +2018-06-23 04:50 +0000 [804d931f27] Alexander Traud + + * bridge_softmix_binaural: Enable FFTW3 in Solaris 11. + + ASTERISK-27939 + + Change-Id: Ice5640e08385a64a0a6555deaccd91e86bca154f + +2018-06-29 18:28 +0000 [1aa45ffdfa] Richard Mudgett + + * res_pjsip_t38.c: Fix crash by ignoring 1xx messages. + + If we initiated a T.38 reINVITE, we would crash if we received any other + 1xx response message except 100 if it were followed by a 200 response. + + * Made ignore any 1xx response so we do not close out the T.38 negotiation + too early. For good measure we'll now accept any 2xx response as + acceptance of the reINVITE T.38 offer. + + ASTERISK-27944 + + Change-Id: I0ca88aae708d091db7335af73f41035a212adff4 + +2018-07-01 13:54 +0000 [f30ebd3823] Joshua Colp + + * res_pjsip_pubsub: Hold module reference for publications. + + Incoming publications need to ensure that the module remains + loaded for the lifetime of them. This is now done by holding + a reference to the module while the publication exists. This + mirrors that of inbound subscriptions. + + ASTERISK-27783 + + Change-Id: Ia98c95a15e11af25728d5fb3e56e12cda0cfc7c0 + +2018-05-21 07:24 +0000 [9d3f3a4b0a] Robert Mordec + + * app_confbridge: Bridge and announcers not removed if conference ends quickly + + If a conference is ended very quickly after it was created (i.e., the + first user immediately hangs up) then the conference bridge and announcer + channels are not removed. + + When a conference is created, the push_announcer() function is added to + the playback queue task processor and the conference object reference is + bumped. If a conference is ended while the push_announcer() function is + still going then the ao2_cleanup(conference) at the end of + push_announcer() will call the destructor function - + destroy_conference_bridge(). + + The destroy_conference_bridge() function will then add the + hangup_playback() task to the playback queue and will wait for it to end. + Since it is already a current task of the playback queue it will wait + forever. + + This patch makes the conference thread call push_announcer() directly. + This way the conference object reference bump is not needed. Since the + playback queue task processor is only used by the conference thread + itself, there is no danger of trying to play announcements before the + announcer is pushed to the bridge. + + ASTERISK-27870 #close + + Change-Id: I947a50fb121422d90fd1816d643a54d75185a477 + +2018-06-21 00:28 +0000 [db02218db2] Matthew Fredrickson + + * main/cdr.c: Alleviate CDR deadlock + + There is a rare case (do to the infrequent timing involved) where + CDR submission threads in batch mode can deadlock with a currently + running CDR batch process. This patch should remove the need for + holding the lock in the scheduler and should clean a few code + paths up that inconsistently submitted new work to the CDR batch + processor. + + ASTERISK-27909 + + Change-Id: I6333e865db7c593c102c2fd948cecdb96481974d + Reported-by: Denis Lebedev + +2018-06-25 22:08 +0000 [4b9bf4f5e0] Kirsty Tyerman + + * pbx_dundi: reordered unloading of module pbx_dundi + + Destroy scheduler after peers are pruned to stop dundi crashing when + unloading module. + + ASTERISK-26987 + Reported-by: Kirsty Tyerman + + Change-Id: Ic12e562cd90d8d813a9e97f302045091f59e3c05 + +2018-06-28 12:07 +0000 [7a238fe74d] Richard Mudgett + + * AMI SendText action: Fix to use correct thread to send the text. + + The AMI action was directly sending the text to the channel driver. + However, this makes two threads attempt to handle media and runs afowl of + CHECK_BLOCKING. + + * Queue a read action to make the channel's media handling thread actually + send the text message. This changes the AMI actions success/fail response + to just mean the text was queued to be sent not that the text actually got + sent. The channel driver may not even support sending text messages. + + ASTERISK-27943 + + Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379 + +2018-06-25 07:37 +0000 [e3585353f6] George Joseph + + * res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs + + In addition to text/* content types, incoming_in_dialog_request now + accepts application/* content types. + + Also fixed a length issue when copying the body text. It was one + character short. + + ASTERISK-27942 + + Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818 + +2018-06-25 15:42 +0000 [5f12e2bd07] George Joseph + + * app_confbridge: Move participant info code to confbridge_manager. + + With the participant info code in app_confbridge, we were still + in the process of adding the channel to the bridge when trying to send + an in-dialog MESSAGE. This caused 2 threads to grab the channel + blocking flag at the same time. To mitigate this, the participant + info code was moved to confbridge_manager so it runs after all + channel/bridge actions have finished. + + Change-Id: I228806ac153074f45e0b35d5236166e92e132abd + +2018-06-18 21:22 +0000 [880fbff6b7] George Joseph + + * res_pjsip_session: Add ability to accept multiple sdp answers + + pjproject by default currently will follow media forked during an INVITE + on outbound calls if the To tag is different on a subsequent response as + that on an earlier response. We handle this correctly. There have + been reported cases where the To tag is the same but we still need to + follow the media. The pjproject patch in this commit adds the + capability to sip_inv and also adds the capability to control it at + runtime. The original "different tag" behavior was always controllable + at runtime but we never did anything with it and left it to default to + TRUE. + + So, along with the pjproject patch, this commit adds options to both the + system and endpoint objects to control the two behaviors, and a small + logic change to session_inv_on_media_update in res_pjsip_session to + control the behavior at the endpoint level. + + The default behavior for "different tags" remains the same at TRUE and + the default for "same tag" is FALSE. + + Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 + ASTERISK-27936 + Reported-by: Ross Beer + +2018-06-21 11:45 +0000 [675e2ddb49] Alexander Traud + + * uuid: Enable UUID in Solaris 11. + + ASTERISK-27933 + Reported by: bautsche + + Change-Id: I9b8362824efbfb2a16981e46e85f7c8322908c49 + +2018-06-13 02:25 +0000 [184b375b41] Kristian F. Høgh + + * app_queue: Add option for predial handlers on caller and callee channels + + Add predial handler support to app_queue. app_dial (ASTERISK_19548) and + app_originate (ASTERISK_26587) have the ability to execute predial + handlers on caller and callee channels. This patch adds predial handlers + to app_queue and uses the same options as Dial and Originate (b and B). + The caller routine gets executed when the caller first enters the queue. + The callee routine gets executed for each queue member when they are about + to be called. + + ASTERISK-27912 + + Change-Id: I5acf5c32587ee008658d12e8a8049eb8fa4d0f24 + +2018-06-21 16:39 +0000 [cad50d6dbf] Richard Mudgett + + * VECTOR: Passing parameters with side effects to macros is dangerous. + + * Fix several instances where we were bumping a ref in the parameter and + then unrefing the object if it failed. The way the AST_VECTOR_APPEND() + and AST_VECTOR_REPLACE() macros are implemented means if it fails the new + value was never evaluated. + + Change-Id: I2847872a455b11ea7e5b7ce697c0a455a1d0ac9a + +2018-06-20 16:57 +0000 [aaaa6f4a4b] Richard Mudgett + + * bridge_softmix.c: Fix memory leak. + + Made release the media_types vector in + softmix_bridge_stream_topology_changed(). + + Change-Id: Ide3f47e379b614879220dfeeb13843f9f2b177b5 + +2018-06-21 11:22 +0000 [bfeded7e62] Alexander Traud + + * smsq: Remove an left-over special case for Solaris. + + Actually, this case was never needed because the check below does the same. + + Change-Id: Ia2fca4ba6c58c644a8b7cb2d9db8539728c14ffb + +2018-06-21 11:17 +0000 [bbea9cfc3b] Alexander Traud + + * res_http_post: Enable GMime in Solaris 11. + + Change-Id: Ie434541f18f894c751d2e44bcb3efb3cac626019 + +2018-06-21 05:08 +0000 [7f3882c8e9] Alexander Traud + + * codecs/ilbc: Compile in Solaris 11. + + The symbol FS is the sampling frequency. That symbol is not used in Asterisk at + all and was a copy-and-paste of the iLBC reference code from the IETF RFC. + However, in Solaris, that symbol is defined by another header already. To + compile in Solaris, that symbol has to go. + + Change-Id: I91ddbe5be7c00069c3a25abd5f58d7b2f04c51b1 + +2018-06-21 05:07 +0000 [9704c424f5] Alexander Traud + + * chan_oss: Compile in Solaris 11. + + M_READ existed already and was conflicting in name. + + Change-Id: I02108e07ae7d2dc314fe1e6c706c17731095a3e4 + +2018-06-21 05:04 +0000 [6f47b84fbd] Alexander Traud + + * func_env: Compile in Solaris 11. + + Change-Id: Idc9b36720f3d29c90a35a6a1ae79a7f9e1aaf50e + +2018-06-21 05:01 +0000 [a5c53bd323] Alexander Traud + + * utils: Avoid an unused variable in Solaris 11. + + With ./configure --enable-dev-mode[=noisy], the build fails because every + warning gets an error. Therefore, Asterisk has to be free of warnings and this + variable must go. + + Change-Id: I63dd2bc4833b9bdb04602f83422d16caf289d46a + +2018-06-21 04:59 +0000 [92109cf496] Alexander Traud + + * BuildSystem: Enable ./configure in Solaris 11. + + ASTERISK-27931 + + Change-Id: If298ce7f03be227a3687b9c20d382c9c55a72404 + +2018-06-20 13:24 +0000 [d6721e1e4c] Alexander Traud + + * BuildSystem: Enable autotools in Solaris 11. + + Because this was the last operating system which required a special case, a + version appended to the autotools, the whole version stuff is removed by this + change. This simplifies the script ./bootstrap.sh. Hopefully, this gives even + broader platform compatibility. + + ASTERISK-27929 + ASTERISK-27926 + + Change-Id: Id4cf433a1a7fa861d0210e1a2e16ca592b49fd5a + +2018-06-13 11:33 +0000 [eb8bbe660e] Richard Mudgett + + * channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. + + * Removed an unnecessary call to ast_channel_blocker_set() in + __ast_read(). + + ASTERISK-27625 + + Change-Id: I342168b999984666fb869cd519fe779583a73834 + +2018-06-13 16:41 +0000 [da54605b8a] Richard Mudgett + + * ARI POST DTMF: Make not compete with channel's media thread. + + There can be one and only one thread handling a channel's media at a time. + Otherwise, we don't know which thread is going to handle the media frames. + + ASTERISK-27625 + + Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e + +2018-06-13 13:05 +0000 [7d874c1af7] Richard Mudgett + + * AMI PlayDTMF Action: Make not compete with channel's media thread. + + There can be one and only one thread handling a channel's media at a time. + Otherwise, we don't know which thread is going to handle the media frames. + + ASTERISK-27625 + + Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905 + +2018-06-12 14:09 +0000 [080508d2eb] Richard Mudgett + + * channel.c: Fix usage of CHECK_BLOCKING() + + The CHECK_BLOCKING() macro is used to indicate if a channel's handling + thread is about to do a blocking operation (poll, read, or write) of + media. A few operations such as ast_queue_frame(), soft hangup, and + masquerades use the indication to wake up the blocked thread to reevaluate + what is going on. + + ASTERISK-27625 + + Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d + +2018-06-18 18:04 +0000 [0989b63047] Richard Mudgett + + * autoservice: Don't start channel autoservice if the thread is a user interface. + + Executing dialplan functions from either AMI or ARI by getting a variable + could place the channel into autoservice. However, these user interface + threads do not handle the channel's media so we wind up with two threads + attempting to handle the media. + + There can be one and only one thread handling a channel's media at a time. + Otherwise, we don't know which thread is going to handle the media frames. + + ASTERISK-27625 + + Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49 + +2018-06-18 16:07 +0000 [91c3ac19cb] Richard Mudgett + + * Dialplan functions: Fix some channel autoservice misuse. + + * Fix off nominal paths leaving the channel in autoservice. + * Remove unnecessary start/stop channel autoservice. + * Fix channel locking around a channel datastore search. + + Change-Id: I7ff2e42388064fe3149034ecae57604040b8b540 + +2018-06-19 10:43 +0000 [720c2d1da2] Richard Mudgett + + * Fix some doxygen and curly placement. + + Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8 + +2018-06-18 13:17 +0000 [c1686b8b3e] Richard Mudgett + + * tcptls.h: Remove redundant SSL_CTX typedef. + + It is invalid to typedef something more than once. Though not all gcc + compilers on different OS's complain about it. + + Change-Id: I5a7d4565990c985822d61ce75bde0b45f9870540 + +2018-06-12 15:13 +0000 [a470bb9e27] Richard Mudgett + + * channel: Fix some more unprotected channel flag setting. + + Change-Id: I34c3b1201b1de539945bcfdcb264fff30332d48c + +2018-06-15 15:21 +0000 [8732d62334] Matthew Fredrickson + + * menuselect/menuselect_curses: Resolves sprintf usage error + + Acccording to the man page for sprintf, using the same buffer for + output as one used as an input yields undefined behavior. + This patch should work around this problem. + + ASTERISK-27903 + Reported-by: Alexander Traud + + Change-Id: I2213dcb454aff26457e2e4cc9c6821276463ae3a + +2018-06-12 09:30 +0000 [4c7ab73468] Sam Wierema + + * app_mp3: remove 10 seconds of silence after mp3 playback + + This patch changes the way asterisk polls output from mpg123, instead + of waiting for 10 seconds(when playing an http url) it now uses a + timeout of one second and iterates 10 times using this same timeout. + + The main difference is that for every timeout asterisk receives it now + checks if mpg123 is still running before poll again. + + ASTERISK-27752 + + Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620 + +2018-06-13 04:40 +0000 [9d7958672b] Alexander Traud + + * tests/test_utils: Repair ./configure --with-ssl=PATH. + + ASTERISK-27914 + + Change-Id: Ibcab8f556ee77776f203cff8b06d776a673b7bc4 + +2018-06-04 20:31 +0000 [e1908ea484] Kirsty Tyerman + + * chan_iax2: better handling for timeout and EINTR + + The iax2 module is not handling timeout and EINTR case properly. Mainly when + there is an interupt to the kernel thread. In case of ast_io_wait recieves a + signal, or timeout it can be an error or return 0 which eventually escapes the + thread loop, so that it cant recieve any data. This then causes the modules + receive queue to build up on the kernel and stop any communications via iax in + asterisk. + + The proposed patch is for the iax module, so that timeout and EINTR does not + exit the thread. + + ASTERISK-27705 + Reported-by: Kirsty Tyerman + + Change-Id: Ib4c32562f69335869adc1783608e940c3535fbfb + +2018-05-31 16:22 +0000 [e7a7506f9c] George Joseph + + * app_confbridge: Enable sending events to participants + + ConfBridge can now send events to participants via in-dialog MESSAGEs. + All current Confbridge events are supported, such as ConfbridgeJoin, + ConfbridgeLeave, etc. In addition to those events, a new event + ConfbridgeWelcome has been added that will send a list of all + current participants to a new participant. + + For all but the ConfbridgeWelcome event, the JSON message contains + information about the bridge, such as its id and name, and information + about the channel that triggered the event such as channel name, + callerid info, mute status, and the MSID labels for their audio and + video tracks. You can use the labels to correlate callerid and mute + status to specific video elements in a webrtc client. + + To control this behavior, the following options have been added to + confbridge.conf: + + bridge_profile/enable_events: This must be enabled on any bridge where + events are desired. + + user_profile/send_events: This must be set for a user profile to send + events. Different user profiles connected to the same bridge can have + different settings. This allows admins to get events but not normal + users for instance. + + user_profile/echo_events: In some cases, you might not want the user + triggering the event to get the event sent back to them. To prevent it, + set this to false. + + A change was also made to res_pjsip_sdp_rtp to save the generated msid + to the stream so it can be re-used. This allows participant A's video + stream to appear as the same label to all other participants. + + Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e + +2018-06-13 05:06 +0000 [b01fc2ef3d] Alexander Traud + + * res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP. + + Previously, Asterisk used its script ./configure, to test whether OpenSSL was + built with no-srtp (or was simply too old). However, the header file + is the preferred way to detect the local configuration + of OpenSSL. + + As a positive side-effect the script ./configure does not interleave the + detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of + individual features of OpenSSL anymore. + + Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0 + +2018-06-05 04:36 +0000 [41175caee0] Joshua Colp + + * rtp: Don't negotiate dynamic codecs using payload. + + In Asterisk there are some dynamic codecs that have + a fixed payload number. This number was being improperly + used to negotiate the codec, instead of using the name + and sample rate. This could result in the wrong payload + number being negotiated for a codec. + + This change makes it so that only static payloads + will be negotiated using their payload number. + + ASTERISK-27848 + + Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570 + +2018-04-16 14:13 +0000 [b649682caa] Sean Bright + + * AST-2018-007: iostreams potential DoS when client connection closed prematurely + + Before Asterisk sends an HTTP response (at least in the case of errors), + it attempts to read & discard the content of the request. If the client + lies about the Content-Length, or the connection is closed from the + client side before "Content-Length" bytes are sent, the request handling + thread will busy loop. + + ASTERISK-27807 + + Change-Id: I945c5fc888ed92be625b8c35039fc6d2aa89c762 + +2018-04-30 17:38 +0000 [81ac32a85f] Richard Mudgett + + * AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses. + + When endpoint specific ACL rules block a SIP request they respond with a + 403 forbidden. However, if an endpoint is not identified then a 401 + unauthorized response is sent. This vulnerability just discloses which + requests hit a defined endpoint. The ACL rules cannot be bypassed to gain + access to the disclosed endpoints. + + * Made endpoint specific ACL rules now respond with a 401 unauthorized + which is the same as if an endpoint were not identified. The fix is + accomplished by replacing the found endpoint with the artificial endpoint + which always fails authentication. + + ASTERISK-27818 + + Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32 + +2018-06-08 15:02 +0000 [0743ad6422] Alexander Traud + + * res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. + + Furthermore, allow OpenSSL configured with no-dh. Additionally, this change + allows auto-negotiation of the elliptic curve/group for servers, not only with + OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 + (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. + + ASTERISK-27910 + + Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537 + +2018-06-08 06:01 +0000 [99aed78078] Alexander Traud + + * crypto.h: Repair ./configure --with-ssl=PATH. + + ASTERISK-27908 + + Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8 + +2018-06-08 04:03 +0000 [ca682f0030] Alexander Traud + + * res_crypto: Allow OpenSSL configured with no-deprecated. + + The header had to be included explicitly. + + ASTERISK-27906 + + Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2 + +2018-06-08 02:41 +0000 [234bf4b7ff] Alexander Traud + + * res_srtp: Repair ./configure --with-ssl=PATH. + + ASTERISK-27905 + + Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2 + +2018-05-31 10:25 +0000 [65ff2f057a] Alexei Gradinari + + * func_odbc: NODATA if SQLNumResultCols returned 0 columns on readsql + + The functions acf_odbc_read/cli_odbc_read ignore a number of columns + returned by the SQLNumResultCols. + If the number of columns is zero it means no data. + In this case, a SQLFetch function has to be not called, + because it will cause an error. + + ASTERISK-27888 #close + + Change-Id: Ie0f7bdac6c405aa5bbd38932c7b831f90729ee19 + +2018-06-07 08:46 +0000 [1725eaf8fb] George Joseph + + * chan_pjsip: Register for "BEFORE_MEDIA" responses + + chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant + it was not updating HANGUPCAUSE for 4XX responses. If the remote end + sent a "180 Ringing", then a "486 Busy", the hangup cause was left at + "180 Normal Clearing". + + * Removed chan_pjsip_incoming_response from the original session + supplement (which was handling only "AFTER MEDIA") and added it to a + new session supplement which accepts both "BEFORE_MEDIA" and + "AFTER_MEDIA". + + * Also cleaned up some cleanup code in load module. + + ASTERISK-27902 + + Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a + +2018-06-07 07:19 +0000 [9f2eb17005] Alexander Traud + + * ooh323c: GCC 8.1 warned about output truncated before terminating nul. + + ASTERISK-27901 + + Change-Id: I5a8e894f4924ef52e3094f6870656a559d67f3d7 + +2018-06-05 13:43 +0000 [7af5e86821] Alexei Gradinari + + * pjsip_options: show/reload AOR qualify options using CLI + + Currentrly pjsip_options code does not handle the situation when the + AOR qualify options were changed. + + Also there is no way to find out what qualify options are using. + + This patch add CLI commands to show and synchronize Aor qualify options: + pjsip show qualify endpoint + Show the current qualify options for all Aors on the PJSIP endpoint. + pjsip show qualify aor + Show the PJSIP Aor current qualify options. + pjsip reload qualify endpoint + Synchronize the qualify options for all Aors on the PJSIP endpoint. + pjsip reload qualify aor + Synchronize the PJSIP Aor qualify options. + + ASTERISK-27872 + + Change-Id: I1746d10ef2b7954f2293f2e606cdd7428068c38c + +2018-05-22 16:21 +0000 [e46b442e38] Alexei Gradinari + + * pjsip_options: handle modification of qualify options in realtime + + Currentrly pjsip_options code does not handle the situation when the + qualify options were changed in realtime database. + Only 'module reload res_pjsip' helps. + + This patch add a check on contact add/update observers if the contact + qualify options are different than local aor qualify options. + If the qualify options were modified then synchronize + the pjsip_options AOR local state. + + ASTERISK-27872 + + Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62 + +2018-05-30 01:12 +0000 [e078558038] Pirmin Walthert + + * bridge_channel.c: Fix Deadlock when using Local channels and fax gateway + + ast_indicate is invoked with the bridge locked. As ast_indicate locks the + other end of the bridge as well this can lead to a deadlock in some situations. + (Especially when a different thread does the same in the reverse order). + This patch calls ast_indicate after unlocking the bridge which fixes the + deadlock. Calling ast_indicate with these parameters without locking the + bridge should be safe as this is done at different places without a + bridge lock. + + ASTERISK-27094 #close + Reported-by: David Brillert + + Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f + +2018-06-04 09:50 +0000 [437ab41881] George Joseph + + * app_sendtext: Allow content types other than text/plain + + There was no real reason to limit the conteny type to text/plain other + than that's what it was limited to before. Now any text/* content + type will be allowed for channel drivers that don't support enhanced + messaging and any type will be allowed for channel drivers that do + support enhanced messaging. + + Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9 + +2018-05-28 19:17 +0000 [a7f4121238] William McCall + + * app_confbridge: Add talking indicator for ConfBridgeList AMI response + + When an AMI client connects, it cannot determine if a user was talking + prior to a transition in the user speaking state (which would generate + a ConfbridgeTalking event). This patch causes app_confbridge to track the + talking state and make this state available via ConfBridgeList. + + ASTERISK-27877 #close + + Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6 + +2018-05-29 12:28 +0000 [6bbede84fb] Richard Mudgett + + * app_meetme: Fix manager event documentation for several events. + + The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and + MeetmeTalkRequest AMI events were documented with sending out a Usernum + header when the User header was actually output. + + * Change the online documentation to match reality. + + ASTERISK-27873 + ASTERISK-25261 + + Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39 + +2018-05-28 10:29 +0000 [24503fb600] Alexander Traud + + * tcptls.h: Repair ./configure --with-ssl=PATH. + + asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those + inclusions got replaced by forward declarations. As side effect, the inclusions + got completed. + + ASTERISK-27878 + + Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7 + +2018-05-25 09:55 +0000 [d36338ce2b] Alexander Traud + + * tcptls: Allow OpenSSL configured with no-dh. + + Additionally, this change allows auto-negotiation of the elliptic curve/group + for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. + This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a + side-effect. + + ASTERISK-27876 + + Change-Id: I62c2aba4a630aefc231b71f646207e8c027d9497 + +2018-05-25 07:22 +0000 [91616f4524] Alexander Traud + + * tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. + + ASTERISK-27874 + + Change-Id: Ica65113511c7a1c13f7988e7d9e7d9e7f3f620dd + +2018-05-15 08:45 +0000 [2bf26ce5ac] George Joseph + + * ast_coredumper: Fix output directory and variable precedence + + The OUTPUTDIR variable in ast_debug_tools.conf.sample is now set + to "/tmp" instead of "/some/directory". + + Variables set on the command line or that are already in the + environment now take predecence over variables set in the config files. + + ASTERISK-27846 + Reported by: Ted G + + Change-Id: Ie8baec52d531886bf5849ec1d59bb59dc87ad387 + +2018-05-09 08:31 +0000 [c5d2bf05f4] Torrey Searle + + * res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change + + Certain race conditions between changing bridge types and DTMF can + cause the current FLAG_NEED_MARKER_BIT to send the marker bit before + the actual first packet of native bridging. + + This logic keeps track of the ssrc the bridge is currently sending + and will correctly ensure the marker bit is set if SSRC as changed + from the previous sent packet. + + ASTERISK-27845 + + Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b + +2018-04-23 09:04 +0000 [a507c73a78] Joshua Colp + + * rtp: Add support for RTP extension negotiation and abs-send-time. + + When RTP was originally created it had the ability to place a single + extension in an RTP packet. In practice people wanted to potentially + put multiple extensions in one and so RFC 5285 (obsoleted by RFC + 8285) came into existence. This allows RTP extensions to be negotiated + with a unique identifier to be used in the RTP packet, allowing + multiple extensions to be present in the packet. + + This change extends the RTP engine API to add support for this. A + user of it can enable extensions and the API provides the ability to + retrieve the information (to construct SDP for example) and to provide + negotiated information (from SDP). The end result is that the RTP + engine can then query to see if the extension has been negotiated and + what unique identifier is to be used. It is then up to the RTP engine + implementation to construct the packet appropriately. + + The first extension to use this support is abs-send-time which is + defined in the REMB draft[1] and is a second timestamp placed in an + RTP packet which is for when the packet has left the sending system. + It is used to more accurately determine the available bandwidth. + + ASTERISK-27831 + + [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 + + Change-Id: I508deac557867b1e27fc7339be890c8018171588 + +2018-05-22 17:17 +0000 [1bec0c73b3] Richard Mudgett + + * channel.c: Fix off nominal channel allocation failure path. + + __ast_channel_alloc_ap() had a failure exit path that hadn't setup the fd + descriptors to -1 yet. The destructor would then attempt to close these + fd's that had never been opened. + + Change-Id: Icf21093f36c60781e8cf6ee9d586536302af33e3 + +2018-05-18 12:46 +0000 [d402594f74] Rodrigo Ramírez Norambuena + + * app_queue: Update year Copyright and fix missing tabs in documentation + + Change-Id: Ieb8faf37dc765463ee5dbca1d1343242c756b1c7 + +2018-05-18 16:45 +0000 [39632c7e00] Alexei Gradinari + + * config.c: Fix successful DELETE treated as failure + + The config engine destroy_func callback function returns the number of + rows deleted or -1 on error. But the function + ast_destroy_realtime_fields treated non-zero return values as error. + + ASTERISK-27863 + + Change-Id: Ied02b38e8196cb03043e609a0679feebd288d17b + +2018-05-14 06:07 +0000 [9f9dce05b2] Matthew Fredrickson + + * netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API + + This function originally was used in chan_sip to enable some simplifying + assumptions and eventually was copy and pasted into res_pjsip_logger and + res_hep. Since it's replicated in three places, it's probably best to + move it into the public netsock2 API for these modules to use. + + Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04 + +2018-05-20 06:36 +0000 [1424f42d25] Alexander Traud + + * libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. + + Use CRYPTO_set_id_callback(.) only with OpenSSL 0.9.8 and older. + + ASTERISK-27867 + + Change-Id: Iadd58d5bf6f538eb224203970a4e88e26f259655 + +2018-05-19 08:23 +0000 [2228ae3f27] Alexander Traud + + * tcptls: Repair ./configure --with-ssl=PATH. + + SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 got discovered without honoring a PATH. + + ASTERISK-27865 + + Change-Id: I8cd358eed7411726d08fa7b01691bef122fbeb71 + +2018-03-27 18:53 +0000 [2ca3b6d9cc] Nic Colledge + + * app_voicemail: Fix data-type mismatch between app_voicemail and database + + Fix data-type mismatch between app_voicemail and database columns + exposed by new version of MariaDB + + ASTERISK-27760 + + Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b + +2018-05-12 06:53 +0000 [97f20fe5ed] Nic Colledge + + * app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail + + Correct the log warning message shown when ODBC voicemail + retrieve_file is called and there is a null value in the category + column. + A more meaningfull message is now written at debug level. + + ASTERISK-27853 + + Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4 + +2018-04-17 21:15 +0000 [52ed6bcc8f] Brian P. Martin + + * chan_mobile: support handling of caller-id names ("cnam"). + + Add support to handle caller-ID names ("cnam") in addition to caller-ID + numbers. The prior code ignored the caller-ID name altogether, and + used the local name for the cell phone (e.g. "my-iphone") in its place. + + Note: as of this writing, at least some Android phones don't pass cnam to + us. This can be seen by issuing "core set debug 2" in the CLI and watching + the "CLIP" record when a call comes in. If cnam isn't in the CLIP record, + there's nothing we can do to provide one. We'll provide a null cnam field, + so later Asterisk processes know to try other sources (e.g. cidname database, + OpenCNAM, etc.). + + Reported by: Brian Martin + Tested by: Brian Martin + ASTERISK-27726 + + Change-Id: I89490d85fa406c36261879c50ae5e65595538ba5 + +2018-05-17 01:58 +0000 [f10fc135d4] Alexander Traud + + * res_pjsip_endpoint_identifier_ip: Unregister the module for headers. + + Asterisk uses Reference Counting to track whether a module can be unloaded. + Every consumer who requires a module, increases the reference count. When the + consumer goes, is unloaded itself, it has to decrease the reference count on + all its used/required modules. That way + core stop gracefully + works on the command-line interface (CLI): One module after the other is + unloaded. A recent change broke this for the module res_pjsip. + + ASTERISK-27861 + + Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3 + +2018-05-11 12:49 +0000 [71d1e8d8c8] Alexander Traud + + * rtp_engine: Remove the double assigned RTP payload ID of H.263+. + + Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format + H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8 + assigned another payload ID 98 for this format in Asterisk 1.6. + + Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667 + +2018-05-11 12:26 +0000 [4722a653f4] Corey Farrell + + * cli: Display correct unit for HTTP timeout in "manager show settings". + + HTTP timeout is in seconds, not minutes. + + ASTERISK-27852 #close + + Change-Id: Ie6640835cb07307555741f9b559c2eb876d9343e + +2018-05-11 10:37 +0000 [263637a38d] Alexander Traud + + * rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code. + + Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48 + +2018-05-06 21:17 +0000 [b5914d90ac] Corey Farrell + + * Fix GCC 8 build issues. + + This fixes build warnings found by GCC 8. In some cases format + truncation is intentional so the warning is just suppressed. + + ASTERISK-27824 #close + + Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84 + +2018-05-11 07:10 +0000 [919b0eb3f2] Alexander Traud + + * rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. + + This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf + which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples + of media formats which were affected. + + ASTERISK-27850 + Reported by: Dinis Brazão, Selene Feigl + + Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496 + +2018-05-09 09:30 +0000 [2e37684913] Corey Farrell + + * git: Ignore *.orig. + + This prevents accidental commit of files created by patch. + + Change-Id: I68380db61f0f9d620046f719ccd978811d0e9964 + +2018-04-18 02:27 +0000 [2d81709ab1] Alexander Traud + + * sip_to_pjsip: Enable python3 compatibility. + + The script remains compatible with Python 2.7 but now also works with + Python 3.3 and newer; to ease the migration from chan_sip to chan_pjsip. + + ASTERISK-27811 + + Change-Id: I59cc6b52a1a89777eebcf25b3023bdf93babf835 + +2018-05-08 14:28 +0000 [cea87fe7b8] Corey Farrell + + * makeopts.in: Remove unused/undefined AST_MARCH_NATIVE. + + Change-Id: I617a96ebb83ec99f5d3176bbbee2d2a272ccb203 + +2018-05-08 04:59 +0000 [9f1e1d153a] Jaco Kroon + + * manager: fix digest auth for ami/http mechanism. + + Due to a fixed size buffer the digest authentication could be + incorrectly calculated if a large URI was provided, causing + authentication failure. The buffer is now dynamically allocated to allow + any size URI within the normal limits of the HTTP request size. + + ASTERISK-27841 + + Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3 + +2018-05-04 13:47 +0000 [d855658f23] Corey Farrell + + * app_macro: Prevent infinite loop in find_matching_priority. + + Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will + prevent crash if dialplan contains an include loop. Log an error when + this occurs, at most one message per call to Macro() so we avoid logger + spam. + + ASTERISK-26570 #close + + Change-Id: I6c71b76998c31434391b150de055ae9a531e31da + +2018-01-11 06:37 +0000 [f4c360143b] Tzafrir Cohen + + * cdr_mysql: my_connect_db(): reduce indentation + + ASTERISK-27572 + + Change-Id: I00bd5363ac94c764c56d8626a5945ed7f3934fcb + +2018-01-11 06:33 +0000 [2e44adf1c3] Tzafrir Cohen + + * cdr_mysql: split mysql init out of my_load_module + + Split out mysql connection parts to a separate my_connect_db(). + + ASTERISK-27572 + + Change-Id: If2ee676056067cc693ff08be68ee4944bf35b49f + +2018-05-04 16:07 +0000 [8f55f7c333] Matthew Fredrickson + + * res_hep: Adds hostname resolution support for capture_address + + Previously, only an IP address would be accepted for the capture_address config + setting in hep.conf. This change allows capture_address to be a resolvable + hostname or an IP address. + + ASTERISK-27796 #close + Reported-By: Sebastian Gutierrez + + Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f + +2018-04-20 18:12 +0000 [7528b86cad] Joshua Colp + + * stream: Make the topology a reference counted object. + + The stream topology has no lock of its own resulting in + another lock protecting it in some way (for example the + channel lock). If multiple channels are being juggled at + the same time this can be problematic. This change makes + the topology a reference counted object instead which + guarantees it will remain valid even without the channel + lock being held. + + Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03 + +2018-03-21 07:30 +0000 [6301531416] Tzafrir Cohen + + * chan_dahdi: Configurable dialed digit timeouts + + Analog phones dial overlap dialing and it is chan_dahdi's job to read the + numbers. It has three timeout constants that this commit converts to + channel-level configuration options: + + * firstdigit_timeout: Default time (ms) to detect first digit + + * interdigit_timeout: Default time (ms) to detect following digits + + * matchdigit_timeout: Default time (ms) to wait in case of ambiguous + match. This happens when the dialed digits match a number in the current + context but are also the prefix of another number. + + Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213 + +2018-05-03 06:34 +0000 [de3ca9bada] Joshua Colp + + * res_ari: Remove requirement that body exists when debug is on. + + The "ari set debug" code for incoming requests incorrectly assumed + that all requests would contain a body. If one did not exist the + request would be incorrectly rejected. The response that was sent + was also incomplete as an incorrect function was used to construct + the response. + + The code has now been changed to no longer require a request to have + a body and the response updated to use the correct function. + + ASTERISK-27801 + + Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8 + +2018-04-30 15:15 +0000 [069a0b7593] Sean Bright + + * iostreams: Add some documentation for the ast_iostream_* functions + + Change-Id: Id71b87637f0a484eb5a1cd26c3d1c7c15c7dcf26 + +2018-05-02 07:43 +0000 [239074c759] Sean Bright + + * pjsip: Increase maximum number of usable ciphers & other cleanups + + * Increase maximum number of ciphers from 100 to 256 (or whatever + PJ_SSL_SOCK_MAX_CIPHERS is #define'd to) + + * Simplify logic in cipher_name_to_id() + + * Make signed/unsigned comparison consistent + + Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412 + + Reported by: Ondřej Holas + + Change-Id: Iea620f03915a1b873e79743154255c3148a514e7 + +2018-04-30 17:24 +0000 [11b7de82c5] Richard Mudgett + + * res_pjsip/pjsip_distributor.c: Add missing off-nominal request response. + + Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b + +2018-04-30 17:20 +0000 [6cab3c836a] Richard Mudgett + + * res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests. + + Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112 + +2018-04-30 09:38 +0000 [afdca5c68c] Joshua Colp + + * res_rtp_asterisk: Always update SRTP on local SSRC change. + + When the local SSRC changes we need to update the SRTP information + so that the proper key is used. This is commonly done as a result + of bridging two channels together. Previously we only updated + the SRTP information if media had already flowed, but in practice + the channel driver may have already performed SRTP negotiation and + set up the previous SSRC. We now always do it on a local SSRC + change. + + ASTERISK-27795 + ASTERISK-27800 + + Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10 + +2018-02-13 12:55 +0000 [0827d5cc53] Gaurav Khurana + + * Add the ability to read the media file type from HTTP header for playback + + How it works today: + media_cache tries to parse out the extension of the media file to be played + from the URI provided to Asterisk while caching the file. + + What's expected: + Better will be to have Asterisk get extension from other ways too. One of the + common ways is to get the type of content from the CONTENT-TYPE header in the + HTTP response for fetching the media file using the URI provided. + + Steps to Reproduce: + Provide a URL of the form: http://host/media/1234 to Asterisk for media + playback. It fails to play and logs show the following error line: + + [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: + File http://host/media/1234 does not exist in any format + + Scenario this issue is blocking: + In the case where the media files are stored in some cloud object store, + following can block the media being played via Asterisk: + + Cloud storage generally needs authenticated access to the storage. The way + to do that is by using signed URIs. With the signed URIs there's no way to + preserve the name of the file. + In most cases Cloud storage returns a key to access the object and preserving + file name is also not a thing there + + ASTERISK-27286 + + Reporter: Gaurav Khurana + + Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89 + +2018-04-25 01:57 +0000 [9c9f314f64] Christof Lauber + + * pbx_lua: Support displaying lua error message if no debug table exists + + The lua_error_function assumed that lua's debug table and traceback function + are always accessible, which is not the case. This fixes the error message + 'Error in the lua error handler' triggred by switch exec() function. + If this happens lua's error message is shown without traceback. + + Change-Id: I34ba0a098f1ae06a3af7b4d1b098bd43f42f96c8 + +2017-12-11 12:34 +0000 [882e79b77e] Joshua Colp + + * pjsip: Rewrite OPTIONS support with new eyes. + + The OPTIONS support in PJSIP has organically grown, like many things in + Asterisk. It has been tweaked, changed, and adapted based on situations + run into. Unfortunately this has taken its toll. Configuration file + based objects have poor performance and even dynamic ones aren't that + great. + + This change scraps the existing code and starts fresh with new eyes. It + leverages all of the APIs made available such as sorcery observers and + serializers to provide a better implementation. + + 1. The state of contacts, AORs, and endpoints relevant to the qualify + process is maintained. This state can be updated by external forces (such + as a device registering/unregistering) and also the reload process. This + state also includes the association between endpoints and AORs. + + 2. AORs are scheduled and not contacts. This reduces the amount of work + spent juggling scheduled items. + + 3. Manipulation of which AORs are being qualified and the endpoint states + all occur within a serializer to reduce the conflict that can occur with + multiple threads attempting to modify things. + + 4. Operations regarding an AOR use a serializer specific to that AOR. + + 5. AORs and endpoint state act as state compositors. They take input + from lower level objects (contacts feed AORs, AORs feed endpoint state) + and determine if a sufficient enough change has occurred to be fed further + up the chain. + + 6. Realtime is supported by using observers to know when a contact has + been registered. If state does not exist for the associated AOR then it + is retrieved and becomes active as appropriate. + + The end result of all of this is best shown with a configuration file of + 3000 endpoints each with an AOR that has a static contact. In the old + code it would take over a minute to load and use all 8 of my cores. This + new code takes 2-3 seconds and barely touches the CPU even while dealing + with all of the OPTIONS requests. + + ASTERISK-26806 + + Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082 + +2017-12-22 13:11 +0000 [661fec4b59] Richard Mudgett + + * core: Remove unused/incomplete SDP modules. + + Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86 + +2018-04-18 15:59 +0000 [ff652711c7] Kevin Harwell + + * translate: generic plc not filled in after translation + + If during translation a codec could not handle a given frame the translation + core would return NULL, thus not passing along the "missing" frame. Due to this + there was no frame to apply generic plc to, thus rendering it useless. + + This patch makes it so the translation core produces an interpolated slin frame + in the cases where an attempt was made to translate to slin, but failed. This + interpolated frame is then passed along and can be used by the generic plc + algorithms to fill in the frame. + + ASTERISK-27814 #close + + Change-Id: I133d084da87adef913bf2ecc9c9240e3eaf4f40a + +2018-04-20 07:40 +0000 [de9c0ede4a] Joshua Colp + + * bridge_softmix: Fix sporadic incorrect video stream mapping. + + When an externally initiated renegotiation occurred it was + possible for video streams to be incorrectly remapped, + resulting in no video flowing to some receivers. + + This change ensures that only the video source sets up + mappings and also that removed streams do not have mappings + set up. + + Change-Id: Iab05f2254df3606670774844bb0935f833d3a9b0 + +2018-04-20 14:07 +0000 [c481afe873] Alexander Anikin + + * chan_ooh323: fix ooManualProgress/ooManualRingback on ooh323 debuggin on + + Call ooManualProgress/Ringback outside of ast_debug function + when ooh323 debugging is on + + ASTERISK-27812 #close + ASTERISK-26893 #close + Reported by: Dimos, Marco Giordani + + Change-Id: I5873762e4f05824e7b6e94a19dd4eb56adbbbb79 + +2018-04-19 13:44 +0000 [5712a0ae52] Joshua Colp + + * bridge_softmix: Fix some REMB bugs. + + This change fixes a bug where a REMB collector may be + freed twice, and also tweaks REMB combining such that if + there is no bitrate from anyone (or there are no sources) + we report 0 instead of using an old bitrate. + + ASTERISK-27804 + + Change-Id: Ia9dc9c150043890ee7ff85e9cdec007f1a77fcfd + +2018-04-20 07:13 +0000 [fe072f4405] Alexander Traud + + * BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. + + ASTERISK-27639 + + Change-Id: I1347f3f2f3737010d0a80a5c30b5aaf71cf3ccb0 + +2018-04-20 05:50 +0000 [efe40ff671] Alexander Traud + + * BuildSystem: Add DragonFly BSD. + + ASTERISK-27820 + + Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557 + +2018-04-20 05:40 +0000 [d54637373a] Alexander Traud + + * menuselect: Add DragonFly BSD. + + In DragonFly BSD, added libraries from ports are placed into /usr/local. + Therefore, this directory must be added for the preprocessor, compiler, and + linker. + + Beside that, the script ./configure was updated: + * OSARCH list was outdated and not used, removed. + * AC_CANONICAL_BUILD was not used. + * _REENTRANT, this feature test macro is obsolete. + + ASTERISK-27820 + + Change-Id: I186d88d99cfa4de6569888e12ac97bd2f441c422 + +2018-04-20 05:18 +0000 [6e9a612293] Alexander Traud + + * install_prereq: Add DragonFly BSD. + + ASTERISK-27820 + + Change-Id: I718ddb000fe5184b1bdc7759da67a370a7520144 + +2018-04-18 11:41 +0000 [b437656c2e] Chris-Savinovich + + * "confbridge show profile bridge" does not output "sfu" when video_mode is sfu + + Fixes a bug on the "confbridge show profile bridge" cli command + that showed "video_mode=no video" when video_mode was set + to "sfu" + + ASTERISK-27418 #close + + Change-Id: I481e3172c7f872664c7ac7809879d541c9f031e9 + +2018-04-18 15:40 +0000 [179ae87cf4] Corey Farrell + + * Build System: Add missing ASTMM_LIBC to flex output. + + Redirect libc allocation functions to use Asterisk functions for + main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors + produced by astmm.h when these files are regenerated, though other + issues still remain. + + ASTERISK~27813 + + Change-Id: I7263e9e4217a17bde4ffaa2087a8f8aeb2a8588c + +2018-04-18 13:40 +0000 [80e6952013] Sean Bright + + * format_pcm: Correct behavior of fseek and ftell for G.722 + + There are twice as many samples in the same number of bytes, so redefine + some of the G.722 format functions in terms of their PCM counterparts. + + Change-Id: I6a8c7352624b930a5f2d9e4857f75283fa5dd9f9 + +2018-04-17 05:33 +0000 [95e8450194] Alexander Anikin + + * chan_ooh323: introduce localras config parameter + + Introduce localras parameter that specify source IP + for connecting to Gatekeeper. Useful for multihome configurations. + + ASTERISK-25129 #close + Reported by: Dmitry Melekhov + Tested by: Dmitry Melekhov + + Change-Id: I0b604b01793f3e02a776502659e07cd3fc7e3097 + +2018-04-18 05:32 +0000 [446320f1d4] Alexander Anikin + + * chan_ooh323: Fix cppcheck warnings + + Fix cppcheck warnings about redundant conditions and possible + null pointer usage + + ASTERISK-27793 #close + Reported by: Ilya Shipitsin + Tested by: Ilya Shipitsin + + Change-Id: I0b31933b062a23331dbac9a82b8bcfe345f406f6 + +2018-04-04 13:12 +0000 [8de3fa2b56] Joshua Colp + + * bridge_softmix / app_confbridge: Add support for REMB combining. + + This change adds the ability for multiple REMB reports in + bridge_softmix to be combined according to a configured + behavior into a single report. This single report is sent + back to the sender of video, which adjusts the encoding bitrate + to be at or below the bitrate of the report. The available + behaviors are: lowest, highest, and average. Lowest uses the + lowest received bitrate. Highest uses the highest received + bitrate. Average goes through the received bitrates adding + them to the previous average and creates a new average. + + Other behaviors can be added in the future and the existing + average one may be adjusted, but this provides the foundation + to do so. + + Support for configuring which behavior to use has been + added to app_confbridge. + + ASTERISK-27804 + + Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66 + +2018-04-13 15:14 +0000 [f79a372941] George Joseph + + * streams: Add string metadata capability + + Replaces the never used opaque data array. + + Updated stream tests to include get/set metadata and + stream clone with metadata. + + Added stream metadata dump to "core show channel" + + Change-Id: Id7473aa4b374d7ab53046c20e321037ba9a56863 + +2018-04-13 15:17 +0000 [f7e7ce6ba2] George Joseph + + * utils: Add ast_assert_return + + Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the + following... + + If the assert passes... NoOp + + If the assert fails and AST_DEVMODE is defined, execute ast_assert() + then, if DO_CRASH isn't set, return from the calling function with + the supplied value. + + If the assert fails and AST_DEVMODE is not defined, return from the + calling function with the supplied value. + + The macro will execute a return without a value if one isn't suppled. + + Change-Id: I0003844affeab550d5ff5bca7aa7cf8a559b873e + +2018-04-10 16:09 +0000 [8135558bab] George Joseph + + * app_sendtext: Enhance SendText to support Enhanced Messaging + + SendText now accepts new channel variables that can be used + to override the To and From display names and set the Content-Type + of a message. Since you can now set Content-Type, other text/* + content types are now valid. + + Change-Id: I648b4574478119f95de09d9f08e9595831b02830 + +2017-09-27 11:44 +0000 [4fb7967c73] George Joseph + + * bridge_softmix: Forward TEXT frames + + Core bridging and, more specifically, bridge_softmix have been + enhanced to relay received frames of type TEXT or TEXT_DATA to all + participants in a softmix bridge. res_pjsip_messaging and + chan_pjsip have been enhanced to take advantage of this so when + res_pjsip_messaging receives an in-dialog MESSAGE message from a + user in a conference call, it's relayed to all other participants + in the call. + + res_pjsip_messaging already queues TEXT frames to the channel when + it receives an in-dialog MESSAGE from an endpoint and chan_pjsip + will send an MESSAGE when it gets a TEXT frame. On a normal + point-to-point call, the frames are forwarded between the two + correctly. bridge_softmix was not though so messages weren't + getting forwarded to conference bridge participants. Even if they + were, the bridging code had no way to tell the participants who + sent the message so it would look like it came from the bridge + itself. + + * The TEXT frame type doesn't allow storage of any meta data, such + as sender, on the frame so a new TEXT_DATA frame type was added that + uses the new ast_msg_data structure as its payload. A channel + driver can queue a frame of that type when it receives a message + from outside. A channel driver can use it for sending messages + by implementing the new send_text_data channel tech callback and + setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech + properties. If set, the bridging/channel core will use it instead + of the original send_text callback and it will get the ast_msg_data + structure. Channel drivers aren't required to implement this. Even + if a TEXT_DATA enabled driver uses it for incoming messages, an + outgoing channel driver that doesn't will still have it's send_text + callback called with only the message text just as before. + + * res_pjsip_messaging now creates a TEXT_DATA frame for incoming + in-dialog messages and sets the "from" to the display name in the + "From" header, or if that's empty, the caller id name from the + channel. This allows the chat client user to set a friendly name + for the chat. + + * bridge_softmix now forwards TEXT and TEXT_DATA frames to all + participants (except the sender). + + * A new function "ast_sendtext_data" was added to channel which + takes an ast_msg_data structure and calls a channel's + send_text_data callback, or if that's not defined, the original + send_text callback. + + * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame + types and ast_sendtext for TEXT frame types. + + * chan_pjsip now uses the "from" name in the ast_msg_data structure + (if it exists) to set the "From" header display name on outgoing text + messages. + + Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489 + +2018-04-17 07:06 +0000 [8a1ffb050b] Alexander Traud + + * utils/pval: Add -lBlocksRuntime for compiler clang conditionally. + + ASTERISK-27809 + + Change-Id: I930b364a33d54cc08dedfcd5bb45f7e83242f134 + +2018-04-17 05:27 +0000 [3d9345e3ae] Alexander Traud + + * chan_vpb: Avoid GNU old-style field designator extension. + + clang 6.0 warned about this. Beside that, this change removes the used variable + 'desc'. + + ASTERISK-27808 + + Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b + +2018-04-09 17:09 +0000 [f5d5083ea7] Ben Ford + + * res_rtp_asterisk: Add support for receiving and handling NACK requests. + + Adds the ability to receive and handle incoming NACK requests if + retransmissions are enabled. If retransmissions are enabled, a data + buffer is allocated that stores packets being sent. If a NACK request + is received, the packet requested for retransmission is sent if it is + still in the buffer. In the same request, if any of the following 16 + packets are marked as not received, those will be sent as well if + available, as outlined in RFC4585. + + Also changes RTCP RR and SR to use media source SSRC instead of packet + source SSRC when determining which instance to use for RTCP reports. + + For more information, refer to the wiki page: + https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements + + ASTERISK-27806 #close + + Change-Id: I7f7f124af3b9d5d2fd9cffc6ba8cb48a6fff06ec + +2018-04-16 16:38 +0000 [d50d637764] Richard Mudgett + + * stringfields: Collect extended stringfields into the stringfield section. + + Use of extended stringfields is a temporary mechanism to avoid ABI + breakage in released branches without resorting to more inconvienient + methods. + + * Collect existing extended stringfields into the parent stringfield + section of the struct. + + Change-Id: I8d46d037801b4518837c3ea4b6df95ceadc9436b + +2018-04-13 14:32 +0000 [4aeec6100f] Ben Ford + + * res_musiconhold: Don't restart MOH from beginning after announcement. + + This reverts a problem introduced by the fix for ASTERISK_24329. + Now, when an announcement is played while waiting in a queue, music on + hold will not restart from the beginning of the sound file and will + instead pick up where it left off. However, the incorrect behavior in + ASTERISK_24329 is now present again; if an announcement X seconds + long is played when music on hold starts, music on hold will start X + seconds into the file. + + ASTERISK-27774 #close + Reported by: lvl + + Change-Id: I86b2885ee7063268f9b9747eddb788336ade989b + +2018-03-28 15:13 +0000 [3bb6cf43b5] Richard Mudgett + + * pjsip_scheduler.c: Add ability to trace scheduled tasks. + + When a scheduled task is created you can pass in the + AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to + be logged. + + Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b + +2018-03-27 11:04 +0000 [237d341bbd] Richard Mudgett + + * res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. + + ast_sip_push_task_synchronous() did not necessarily execute the passed in + task under the specified serializer. If the current thread is any + registered pjsip thread then it would execute the task immediately instead + of under the specified serializer. Reentrancy issues could result if the + task does not execute with the right serializer. + + The original reason ast_sip_push_task_synchronous() checked to see if the + current thread was a registered pjsip thread was because of a deadlock + with masquerades and the channel technology's fixup callback + (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) + involving call pickups avoided the original deadlock situation entirely. + The PJSIP channel technology's fixup callback no longer needed to call + ast_sip_push_task_synchronous(). + + However, there are a few places where this unexpected behavior is still + required to avoid deadlocks. The pjsip monitor thread executes callbacks + that do calls to ast_sip_push_task_synchronous() that would deadlock if + the task were actually pushed to the specified serializer. I ran into one + dealing with the pubsub subscriptions where an ao2 destructor called + ast_sip_push_task_synchronous(). + + * Split ast_sip_push_task_synchronous() into + ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). + ast_sip_push_task_wait_servant() has the old behavior of + ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has + the new behavior where the task is always executed by the specified + serializer or a picked serializer if one is not passed in. Both functions + behave the same if the current thread is not a SIP servant. + + * Redirected ast_sip_push_task_synchronous() to + ast_sip_push_task_wait_servant() to preserve API for released branches. + + ASTERISK_26806 + + Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3 + +2018-03-21 19:43 +0000 [c2f85e881d] Richard Mudgett + + * pjsip_scheduler.c: Fix some corner cases. + + * Fix the periodic interval wander because it may take significant time + between the sched thread queueing the task in the serializer and the + serializer actually executing the task. The time it takes to actually + execute the task was already taken into account. + + * Pass a schtd ref to the serializer when we queue a scheduled task on + the serializer. We don't want it going away on us while it is in the + serializer queue. + + * Skip the scheduled task if the task was canceled between queueing the + task to the serializer and the serializer actually executing the task. + + * Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed + task_id and added next_periodic. + + * Hold a ref to the passed in serializer so the serializer cannot go away + on the scheduled task. + + ASTERISK_26806 + + Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24 + +2018-03-22 19:09 +0000 [96c4a57edf] Richard Mudgett + + * pjsip_scheduler.c: Sort "pjsip show scheduled_tasks" output. + + * A side benefit is that the scheduled tasks are not completely blocked + while the CLI command executes. + + * Adjusted the "Task Name" column width to have more room for longer + names. + + Change-Id: Iec64aa463ee8b10eef90120e00c38b1fb444087e + +2018-04-02 15:59 +0000 [429c758e48] Evandro Cesar Arruda + + * cdr_mysql: Compile error because MYSQL_PORT definition is missing + + If it is not defined, it will add MYSQL_PORT definition. After some + research on MySQL/MariaDB development tree, I couldn't find any reference + to MYSQL_PORT definition in include files. + + ASTERISK-27782 #close + + Change-Id: Ieee56c836fc2e8bd021c456145bba04c6068bb77 + +2018-04-09 20:00 +0000 [0747ac893b] Chris-Savinovich + + * res_pjsip_session: Rewrite o= with external_media_address. + + It now appends the external IP address on the + o= line of the SDP packet. The decision was made to write + the numeric IP address as opposed to the RFC that states + the FQDN should be used if and when available. We believe + the usage of literal IP address will help avoid + potential problems. + + ASTERISK-27614 #close + + Change-Id: I84f3360f3606b8c4e8d161edb228799ec0b8a302 + +2018-02-22 12:18 +0000 [1cd704de36] Nathan Bruning + + * res_pjsip_notify.c: enable in-dialog NOTIFY + + This patch adds support to send in-dialog SIP NOTIFY commands on + chan_pjsip channels, similar to the functionality recently added + for chan_sip (ASTERISK_27461). + + This extends res_pjsip_notify to allow for in-dialog messages. + + ASTERISK-27697 + + Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29 + +2018-03-22 13:35 +0000 [7157dcf83b] Richard Mudgett + + * pjsip_scheduler.c: Fix ao2 usage errors. + + * Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the + 'tasks' container being accessed without a lock in a multi-threaded + environment. A recipe for crashes. + + * Removed needlessly obtaining schtd object references. If the caller + providing you a pointer to an object doesn't have a valid reference then + you cannot safely get one from it. + + * Getting a ref to 'tasks' when you aren't copying the pointer into + another location is useless. The 'tasks' container pointer is global. + + * Removed many unnecessary uses of RAII_VAR. + + * Make ast_sip_schedule_task() name parameter const. + + ASTERISK_26806 + + Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db + +2018-03-23 06:49 +0000 [879e592baf] Corey Farrell + + * Build System: Enable python3 compatibility. + + * Consistently use spaces in rest-api-templates/asterisk_processor.py. + * Exclude third-party from docs/full-en_US.xml. + * Add docs/full-en_US.xml to .gitignore. + * Use list() to convert python3 view. + * Use python3 print function. + * Replace cmp() with equivalent equation. + * Replace reference to out of scope subtype variable with name + parameter. + * Use unescaping triple bracket notation in mustache templates where + needed. This causes behavior of Python2 to be maintained when using + Python3. + * Fix references to has_websocket / is_websocket in + res_ari_resource.c.mustache. + * Update calculation of has_websocket to use any(). + * Use unicode mode for writing output file in transform.py. + * Replace 'from swagger_model import *' with explicit import of required + symbols. + + I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the + print syntax has been fixed. + + Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33 + +2018-04-05 18:33 +0000 [0c03eab962] Richard Mudgett + + * res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge + + There is a problem when an INVITE-with-Replaces transfer targets a channel + in a ConfBridge. The transfer will unconditionally swap out the + ConfBridge channel. Unfortunately, the ConfBridge state will not be aware + of this change. Unexpected behavior will happen as a result since + ConfBridge channels currently can only be replaced by a masquerade and not + normal bridge channel moves. + + * We just need to pretend that the channel isn't in a bridge (like other + transfer methods already do) so the transfer channel will masquerade into + the ConfBridge channel. + + Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82 + +2018-03-28 07:27 +0000 [c7bd554094] Joshua Colp + + * pjsip / res_rtp_asterisk: Add support for sending REMB + + This change allows chan_pjsip to be given an AST_FRAME_RTCP + containing REMB feedback and pass it to res_rtp_asterisk. + Once res_rtp_asterisk receives the frame a REMB RTCP feedback + packet is constructed with the appropriate contents and sent + to the remote endpoint. + + ASTERISK-27776 + + Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd + +2018-04-05 20:02 +0000 [39016e3582] Joshua Colp + + * res_rtp_asterisk: Fix minimum block word length for REMB. + + The minimum block word length is actually 4, not 5. + + Change-Id: I878542218225aed72c72bdf1b856fc822cd2d649 + +2018-04-05 18:48 +0000 [8a602f18db] Joshua Colp + + * res_rtp_asterisk: Queue video update on picture loss indication. + + The previous payload specific feedback handling was very single + minded in that it just assumed everything should trigger a video + update. This was changed but the handling of picture loss indication + was not added. The result was that video may not flow. This change + adds it explicitly in. + + Change-Id: I1894be02e39ee10a0af841b5a1dca5f0ec7d60b6 + +2018-04-05 17:40 +0000 [d72a2966da] Richard Mudgett + + * chan_sip.c: Fix INVITE with replaces channel ref leak. + + Given the below call scenario: + A -> Ast1 -> B + C <- Ast2 <- B + + 1) A calls B through Ast1 + 2) B calls C through Ast2 + 3) B transfers A to C + + When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to + send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of + the channel between Ast1 and Ast2. + + Channel ref leaks are easily seen in the CLI "core show channels" output. + The leaked channels appear in the output but you can do nothing with them + and they never go away unless you restart Asterisk. + + * Properly account for the channel refs when imparting a channel into a + bridge when handling an INVITE with replaces in handle_invite_replaces(). + The ast_bridge_impart() function steals a channel ref but the code didn't + account for how many refs were held by the code at the time and which ref + was stolen. + + * Eliminated RAII_VAR in handle_invite_replaces(). + + ASTERISK-27740 + + Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4 + +2018-03-21 19:40 +0000 [71a67a98c4] Richard Mudgett + + * res_pjsip: Update authenticate_qualify documentation. + + Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4 + +2018-04-02 16:49 +0000 [6774913e82] Richard Mudgett + + * app_agent_pool.c: Fix off nominal ref leak. + + Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6 + +2018-04-04 10:02 +0000 [e40fd7a232] Corey Farrell + + * Build System: Strip '-std=c99' from CFLAGS provided by libraries. + + Asterisk requires GNU C extensions. On some systems certain libraries + may incorrectly push -std=c99 into CFLAGS, thus breaking the build. + This change causes that flag to be stripped so the Asterisk build is not + broken by those libraries. This change is made for both pkgconfig and + tool based libraries. + + ASTERISK-27629 #close + + Change-Id: I13389613b194abbac77becf90cd950dc168704db + +2018-04-03 14:39 +0000 [66f13ed694] Corey Farrell + + * Build System: Fixes for configure script. + + * Replace all 'else if' statements with 'elif'. + * Use loop to detect versioned lua headers and libraries. + + The loop for detecting lua fixes a bug where LUA_INCLUDE would be + appended with the directory of every lua version after the first one is + found. + + Change-Id: I3276f9aee955014108345be6092f51c932b43a0f + +2018-04-02 08:53 +0000 [0f6431e8e4] Joshua Colp + + * app_confbridge / bridge_softmix: Add ability to configure REMB interval. + + This change adds a configuration option to app_confbridge which can be + used to set the interval at which we will send a combined REMB (remote + estimated maximum bitrate) frame to sources of video. The bridging API + has also been extended slightly to allow setting this so bridge_softmix + can use it. + + ASTERISK-27786 + + Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82 + +2018-01-02 07:54 +0000 [f91263cf46] George Joseph + + * res_pjsip: Correct usages of pjproject's timer heap + + Fix some timer heap initializations and cancels to try and prevent + crashes and timer heap issues. + + Change-Id: I64885d190fa22097d1b55987091375541e57a7ee + +2018-03-25 13:35 +0000 [48720e7def] George Joseph + + * pjroject_bundled: Add already-destroyed check to tsx_timer_callback + + There have been cases that when the transaction timer callback is called + the tsx is already destroyed. This causes a crash. We now check the + tsx state and return if the tsx is already destroyed. + + Change-Id: If93acd5e48d9ca5bb553f2405d5afc836842fe1c + +2018-03-25 13:25 +0000 [7c03b2713e] George Joseph + + * pjproject_bundled: timer: Clean up usage of timer heap + + Added a new pj_timer_entry_reset function that resets a timer_entry + for re-use. + + Changed direct settings of timer_entry fields to use + pj_timer_entry_init and pj_timer_entry_reset. + + Fixed issues where timers were being rescheduled incorrectly. + + Change-Id: I5b624bfbc5c1429117484b9b24567293002148e6 + +2018-03-29 17:07 +0000 [97cc67b12f] Richard Mudgett + + * res_pjsip: Fix deadlock on reliable transport shutdown. + + A deadlock can happen when the PJSIP monitor thread is shutting down a + connection oriented transport (TCP/TLS) used by a subscription at the same + time as another thread tries to send something for that subscription. The + deadlock is between the pjsip monitor thread attempting to get the dialog + lock and another thread sending something for that dialog when it tries to + get the transport manager lock. + + * res_pjsip_pubsub.c: Avoid the deadlock by pushing the subscription + removal to the subscription serializer. + + * res_pjsip_registrar.c: Pushed off incoming registration contact removals + to a default serializer as a precaution. Removing the contacts involves + sorcery access which in this case will involve database access. Depending + upon the setup, the database may not be on the same machine and could take + awhile. We don't want to hold up the pjsip monitor thread with + potentially long access times. + + ASTERISK-27706 + + Change-Id: I56b647aea565f24dba33e9e5ebeed4cd3f31f8c4 + +2018-03-07 06:15 +0000 [f65488f546] Ross Beer + + * pjsip_transport_events.c: Fix crash using stale transport pointer. + + Apparently it is possible for the transport to be destroyed without + triggering the transport callback logic. As a result the transport gets + destroyed and we have a stale pointer in the active_transports container. + + * Invoke the transport monitor callback checks when the transport is + destroyed in addition to when it is disconnected and shutdown. + + ASTERISK-27688 + + Change-Id: Ia9b5469fea8f2b3f2d8476fae6b748a4d23e7261 + +2018-03-19 09:36 +0000 [879743ab8f] Ben Ford + + * test_data_buffer.c: Add unit tests for data buffer API. + + Added unit tests for the data buffer API. These tests include creating a + data buffer, putting payloads into the buffer, resizing the buffer, and + the nominal case for data buffer usage, which consists of adding + the max number of payloads to the buffer, checking to see if the correct + payloads are present, then adding more payloads and checking again to + see if the previous payloads were replaced or not. + + For more information, refer to the wiki page: + https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements + + Change-Id: Id5b599aa15a5e61d0ec080f97cd0c57bd07e6f8f + +2018-02-23 13:49 +0000 [138e0eff4e] Ben Ford + + * Add data buffer API to store packets. + + Adds a data buffer with a configurable size that can store different + kinds of packets (like RTP packets for retransmission). Given a number + it will store a data packet at that position relative to the others. + Given a number it will retrieve the given data packet if it is present. + This is purposely a storage of arbitrary things so it can be used not + just for RTP packets but also Asterisk frames in the future if needed. + The API does not internally use a lock, so it will be up to the user of + the API to properly protect the data buffer. + + For more information, refer to the wiki page: + https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements + + Change-Id: Iff13c5d4795d52356959fe2a57360cd57dfade07 + +2018-03-25 13:12 +0000 [a87141ddfd] George Joseph + + * pjproject_bundled: Add patch for pj_atomic crashes + + There have been some crashes in the past where something attempts + to use a pj_atomic after it's already been destroyed. This patch + tries to prevent it by making sure that pj_atomic_destroy sets + its mutex to NULL when it's done. The pj_mutex functions already check + for a NULL mutex and just return PJ_EINVAL. + + Teluu also added some checks to the win32 implementation as well. + + Change-Id: Id25f70b79fdedf44ead6e6e1763a4417d3b3f825 + +2018-03-21 08:52 +0000 [e14b0e960d] Joshua Colp + + * res_rtp_asterisk: Add support for raising additional RTCP messages. + + This change extends the existing AST_FRAME_RTCP frame type to be + able to contain additional RTCP message types, such as feedback + messages. The payload type is contained in the subclass which allows + knowing what is in the frame itself. + + The RTCP feedback message type is now handled and REMB[1] messages + are raised with their containing information. + + This also fixes a bug where all feedback messages were triggering + video updates instead of just FIR and FUR. + + Finally RTCP frames are now passed up through the Asterisk core to + what is handling the channel, mapped appropriately in the case of + bridging, and written to an outgoing stream. Since RTCP frames are + on a per-stream basis this is only done on multistream capable + channels. + + [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 + + ASTERISK-27758 + ASTERISK-26366 + + Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e + +2018-03-27 08:27 +0000 [455cee99ae] Florian Floimair + + * main: Update copyright notice with year 2018 + + Change-Id: I2d80bc5edf940fab914cba3d8a0fa0b5eb2a3148 + +2018-03-26 07:42 +0000 [48190c7f93] Guido Falsi + + * core: fix getopt(3) usage + + Setting optind = 0 is forced to 1 in glibc implementation, but + causes option parsing to be flawed in other implementations, for + example on FreeBSD. + + ASTERISK-27773 #close + + Change-Id: Ia548e69f8302e9754dbbedb6bc451c0700c66f61 + +2018-03-23 13:15 +0000 [07cf6b1437] Alexander Traud + + * install_prereq: Add Slackware (somehow). + + ASTERISK-27770 + + Change-Id: Ib87e0483c785542238cfe34c1e884d5a31edfaab + +2018-03-23 09:13 +0000 [307a295d00] Alexander Traud + + * install_prereq: Add Gentoo Linux. + + ASTERISK-27769 + + Change-Id: Ieb13293cd67481f3a33f58f6f7c8c3ee1e338e7a + +2018-03-17 01:02 +0000 [318bf45928] Corey Farrell + + * main/indications: Use ast_cli_completion_add for all completions. + + Change-Id: I371be01f178fb542a9fbe8d97e7ae21aa4d82c36 + +2018-03-21 14:54 +0000 [75715b95b4] Russell Bryant + + * app_originate: Add async option. + + Add an option to make app_originate not wait for the created channel + to answer. + + Change-Id: I7fc2facd77079abc6321f44e8bcd4e39298de2ae + Requested-by: Frederic Steinfels + Signed-off-by: Russell Bryant + +2018-03-22 07:27 +0000 [4f33f56a72] Alexander Traud + + * BuildSystem: pjsip_evsub_set_uas_timeout was not used (part 2). + + The previous change was not complete. + + ASTERISK-27435 + + Change-Id: I11082c14c0ef9c6af8c995084a6851337ea2a90f + +2018-03-22 05:43 +0000 [d6fda173a4] Alexander Traud + + * BuildSystem: With external editline, do not require libs for internal editline. + + ASTERISK-27761 + + Change-Id: Ib17a7415297a210cfcdbf149e4df9b6edadbfab6 + +2018-03-21 22:00 +0000 [a6d58c518a] Corey Farrell + + * core: Create main/options.c. + + This creates a separate source to 'own' symbols related to options.h and + paths.h. This significantly reduces the number of exports created by + main/asterisk.o. This change is required to eventually be able to + link unmodified Asterisk sources to utilities and/or stand-alone tests. + + ASTERISK~26245 + + Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380 + +2018-03-21 19:25 +0000 [745b5134cd] George Joseph + + * Revert "BuildSystem: In NetBSD, the Python Programming Language is python-X.Y." + + Something is causing a python2/python3 mismatch on Fedora27. + + PYTHON='/usr/bin/python2' + PYTHONDEV_CFLAGS='-I/usr/include/python3.6m ' + PYTHONDEV_INCLUDE='-I/usr/include/python3.6m ' + PYTHONDEV_LIB='-lpython3.6m ' + PYTHONDEV_LIBS='-lpython3.6m ' + + This reverts commit be0e9920b64e3b07501b299d131309b58f9b0ddf. + + Change-Id: I86dd102eb3ead199fe89178cdbadb36b4e2cfd1b + +2018-02-08 13:23 +0000 [411915af28] Corey Farrell + + * loader: Reserve space for additional pointers in ast_module_info. + + This creates 4 reserved pointers in case we need additional dependency + management fields. + + Change-Id: If991ec99b779df1b2dfbd38ce1a0cd79f9e01821 + +2018-03-20 15:28 +0000 [cf73a4203f] Kevin Harwell + + * bridge_softmix: Clear "talking" when a channel is put on hold + + This patch clears the talking flag from the channel (if already set), and + notifies listeners when that channel is put on hold. Note however, if the + endpoint continues to send audio frames and these are received by the bridge + then that channel will be put back into a "talking" state even though they + are on hold. + + ASTERISK-27755 #close + + Change-Id: I930e16c4662810f9f02043d69062f88173c5e2ef + +2018-03-20 11:53 +0000 [bfefde5b07] Alexander Traud + + * BuildSystem: For consistency, avoid extra libs to be empty. + + AST_EXT_LIB_CHECK has several optional parameters. When an optional parameter + is left empty, [] is used to indicate this. However, this is done in the script + ./configure only then, when a further parameter is not empty. For example, when + no extra libraries are needed to test the checked library, parameter 5 is not + mentioned. Except parameter 6 and higher are used, then parameter 5 must be + empty. + + However, this general rule was broken + * four times for parameter 5 (extra libs) and + * three times for parameter 4 (header) + as found via the Regular Expression \[\]\). In case of parameter 5, all cases + were changed, because that happened for no reason. In case of parameter 4, an + [] improves readability actually. Therefore for parameter 4, the only case which + did not do it was changed. All this aims to create more consistency: Only do + something different if there is a reason to do so. + + Change-Id: I037ef170cf1ad94497151a9ea5071a31c656cafe + +2018-03-20 09:58 +0000 [8bd5980e14] Ivan Poddubny + + * func_channel: Delete dead CHANNEL_TRACE code + + The functions behind the flag and the flag itself were removed + from Asterisk 12 as incompatible with the new architecture. + + Change-Id: I058493ef7a53ee290fd225bbcbb07bf46b623ccf + +2018-03-17 21:26 +0000 [040bb21771] Corey Farrell + + * core: Remove additional symbols. + + Remove symbols that are depreacated and replaced: + * ast_channel_datastore_alloc + * ast_channel_datastore_free + * ast_channel_cmpwhentohangup + * ast_channel_setwhentohangup + * config_text_file_save + * devstate2str + * ast_device_state_changed + * ast_device_state_changed_literal + * ast_verbose_get_by_module + + Remove unused symbols: + * channelreloadreason2txt (last used in Asterisk 12). + + Remove unused ast_options flags: + * AST_OPT_FLAG_END_CDR_BEFORE_H_EXTEN / ast_opt_end_cdr_before_h_exten + * AST_OPT_FLAG_VERBOSE_MODULE / ast_opt_verb_module + * AST_OPT_FLAG_INITIATED_SECONDS + + Change-Id: I841255995d195f8efc1ed47af9c7a2f131c08645 + +2018-03-17 20:03 +0000 [de77cf8698] Corey Farrell + + * core: Remove dead symbols from asterisk.exports.in. + + * dahdi_chan_name + * dahdi_chan_name_len + * dahdi_chan_mode + * __manager_event + * dialed_interface_info + + Added comment about __progname and environ being needed for FreeBSD to + prevent accidental removal in the future. + + Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5 + +2018-03-17 01:39 +0000 [201762f161] Corey Farrell + + * named_acl: Use ast_cli_completion_add. + + Change-Id: I317a82de976bbdbfe4352c243e32a7bb8f66c377 + +2018-03-17 01:58 +0000 [645203a422] Corey Farrell + + * main/sounds: Use ast_cli_completion_add. + + Change-Id: I140e1137906bbfcdb61c0c6304159be459ad873e + +2018-03-16 10:19 +0000 [5d097f8236] George Joseph + + * channel.c: Allow generic plc then channel formats are equal + + If the two formats on a channel are equal, we don't transcode and since + the generic plc needs slin to work, it doesn't get invoked. + + * A new configuration option "genericplc_on_equal_codecs" was added + to the "plc" section of codecs.conf to allow generic packet loss + concealment even if no transcoding was originally needed. + Transcoding via SLIN is forced in this case. + + ASTERISK-27743 + + Change-Id: I0577026a179dea34232e63123254b4e0508378f4 + +2018-03-17 01:09 +0000 [8d01ec572d] Corey Farrell + + * manager: Use ast_cli_completion_add for completion generators. + + Change-Id: I658141c6ec490a3e866b02d2afea757928ceaabf + +2018-03-17 02:16 +0000 [2c1ad2f510] Corey Farrell + + * main/test: Use ast_cli_completion_add. + + Change-Id: I5133ff2ba4e030f9733fb3d050c863d72a22ae6b + +2018-03-18 10:16 +0000 [115939caeb] Joshua Colp + + * rtp: Add REMB RTP property and set it on PJSIP video RTP. + + This change adds a property to RTP instances to indicate that + REMB support is enabled and that sending/receiving should be + passed through. + + This also enables it on video RTP instances in PJSIP if + WebRTC support is enabled. + + Finally the goog-remb extension is added to the SDP using + the rtcp-fb attribute to indicate our support for it. + + Details about REMB can be found on the draft document for it: + https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 + + Change-Id: I1902dda1c0882bd1a0d71b2f120684b44b97e789 + +2018-03-17 04:31 +0000 [8c25a72d57] Corey Farrell + + * main/bridge: Use ast_cli_completion_add. + + Change-Id: I3775a696d6a57139fdf09651ecb786bcf1774509 + +2018-03-17 16:41 +0000 [5b40441197] Corey Farrell + + * core: Minor cleanup of ast_el_read_char. + + * Define CHAR_T_LIBEDIT and CHAR_TO_LIBEDIT based on + HAVE_LIBEDIT_IS_UNICODE. This avoids needing to repeatedly use + conditional blocks, eliminates having multiple function prototypes. + * Remove parenthesis from return values. + * Add missing code block brackets {}. + * Reduce use of 'else' conditional statements where possible. + + Change-Id: I4315328ebea2f62641faf6881de2ac20a9f9d08e + +2018-03-17 10:49 +0000 [e61b50b67a] Alexander Traud + + * BuildSystem: Check for header file of OGG. + + Asterisk uses various symbols of the shared library libogg within the module + format_ogg_vorbis. However, the source code of that module did not include the + header file of libogg explicitly but implicitly. Because that header was not + included before Asterisk 14, the script ./configure was told not to check for + it. + + Anyway, even Asterisk 13 LTS uses symbols of libogg. Therefore, that header + should be included explicitly. Therefore, ./configure should check for that + header. + + Change-Id: I98c50d56311b68880d1084fcc62c35ab2f8692db + +2018-03-09 06:26 +0000 [f697025ae5] Alexander Traud + + * BuildSystem: When no download utility is available, display the explanation. + + ./configure --with-pjproject-bundled + did not display an explanation, when no download utility like wget, curl, or + fetch was installed beforehand, although an explanation existed in code. This + happened because the code expected the variable DOWNLOAD_TO_STDOUT to be empty. + However, the script ./configure set that variable always. + + Change-Id: I64c99b76a03525c69471e5055bf124b36a51bbd4 + +2018-03-17 05:00 +0000 [10a978829e] Alexander Traud + + * BuildSystem: Remove unused dependency on libltdl. + + Asterisk does not need the development package of libltdl, because it does not + use any symbol of -lltdl directly. Instead, it uses the runtime package via the + shared library -lodbc. On the supported platforms, that shared library declares + its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have + failed. + + ASTERISK-27745 + + Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba + +2018-03-17 02:25 +0000 [1136a22a1e] Corey Farrell + + * main/translate: Use ast_cli_completion_add. + + Change-Id: I0e2402660e54d91f74ab0804c62a5b1925577413 + +2018-03-17 02:00 +0000 [91ac95993e] Corey Farrell + + * main/taskprocessor: Use ast_cli_completion_add. + + Change-Id: Ie5f812a988ed811fd11967151932de62bc131b48 + +2018-03-15 15:06 +0000 [3ad56aa929] Corey Farrell + + * main/config: Use ast_cli_completion_add for reload completion. + + Change-Id: Ia3fa4c03f2285a1ec8814bbe7f4624ead9111ad1 + +2018-03-17 00:51 +0000 [9e335f22e7] Corey Farrell + + * aco: Use ast_cli_completion_add for 'config show help'. + + In addition this removes: + * RAII_VAR usage + * Duplicate check of pos + * Unneeded arguments. + + Change-Id: I2da8eac2670d1d8d6474c04037129804f55ebf39 + +2018-03-14 04:27 +0000 [4d1c9d8711] Corey Farrell + + * core: Stop using AST_INLINE_API for allocator functions. + + This replaces AST_INLINE_API allocators in utils.h with real functions + implemented in astmm.c. Associated macro's are also moved from utils.h + to astmm.h. + + Remove menuselect conflicts between MALLOC_DEBUG and DEBUG_CHAOS as they + can now be combined. + + This has multiple benefits: + * Simplifies asterisk/utils.h by removing inline functions and use of + the logger. + * Removal of these inline functions decreases size of Asterisk and + module binaries by 1% or more. + * Puts memory management functions together with and without + MALLOC_DEBUG enabled, simplifying management of the code. + * Enables DEBUG_CHAOS for ASTMM_REDIRECT and bundled pjproject. + + Change-Id: If9df4377f74bdbb627461b27a473123e05525887 + +2018-02-27 03:01 +0000 [ecc846b26b] Florian Floimair + + * app_dial: Enable early-media video + + Certain applications (e.g. door-phone) require that also video is transmitted + before a call is accepted. + + Change-Id: I9842e1dc2f6e1c2c49dc33fe615255007d2f821e + +2018-03-05 06:50 +0000 [be0e9920b6] Alexander Traud + + * BuildSystem: In NetBSD, the Python Programming Language is python-X.Y. + + ASTERISK-27717 + + Change-Id: If90ddf9c396c32e7402a894f42dce215c30049d1 + +2018-03-16 09:53 +0000 [02fa145a1b] Alexander Traud + + * BuildSystem: Avoid an extra case for OpenBSD. + + Nine years ago with Mantis 13639 (now ASTERISK-12841) an extra case for OpenBSD + was introduced: Vorbis required Ogg to be specified manually, because the shared + library libvorbis.so did not specify its required dependency on -logg itself. + + Today with OpenBSD 6.2, all libvorbis*.so declare their dependencies correctly. + Therefore, an extra case is not required anymore. + + Change-Id: Ifd04e0994ce9f1e4ad29c3948a0398b91d1e97bc + +2018-03-05 10:10 +0000 [00789174f6] Alexander Traud + + * BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. + + In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some + libraries do not specify all their dependencies and require additional shared + libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a + library is specified there, it must exist on the platform, because ./configure + tries to compile/link/execute a small app using those statements. For example, + the library libdl.so is Linux specific and does not exist on BSD-like platforms. + + Furthermore, no supported platform/version was found, which still (ever?) + requires those additional libraries. Therefore, they were simply removed. + + Finally, this change adds the error code ESTRPIPE to the channel driver + chan_alsa for those platforms which lack it, again for example NetBSD. + + ASTERISK-27720 + + Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b + +2018-03-16 09:02 +0000 [4d1e3fef6b] George Joseph + + * app_voicemail: Fix json blob errors + + When app_voicemail calls ast_test_suite_notify with the results of + a user keypress, it formats the keypress as '%c'. If the user hung up + or some other error occurrs, the result of the keypress is a non + printable character. This ultimately causes json_vpack_ex to think + it's being passed a non utf-8 string and return an error. + + * Keypress results passed to ast_test_suite_notify are now checked with + isprint() and a '?' is substituted if the check fails. + + Change-Id: I78ee188916bbac840f3d03f40201b692347ea865 + +2018-03-15 09:32 +0000 [ebe957c5e9] Corey Farrell + + * main/cdr: Use ast_cli_completion_add for CDR channel completion. + + Change-Id: Ie81830647a23aad61c1162583b6d50adbe6e7822 + +2018-03-12 10:20 +0000 [dbf5ff6ed0] Alexander Traud + + * install_prereq: Add Arch Linux. + + ASTERISK-27738 + + Change-Id: I7ca620e3c4dfb4b064a19382c4915aeb42a2a09f + +2018-03-15 08:19 +0000 [89ba4d4e3d] Corey Farrell + + * main/ccss: Use ast_cli_completion_add for core id. + + Change-Id: I44b25d6d24c7d9bc1bb38a50774b38883162f98f + +2018-03-15 07:29 +0000 [aa0d95c730] Corey Farrell + + * astobj2_container: Use ast_cli_completion_add for container names. + + Change-Id: I4f0fc09e820eb8d8da2354a177dbcf503c56ddd1 + +2017-12-09 04:52 +0000 [b929a7fb8d] Corey Farrell + + * main/channel: Use ast_cli_completion_add for channeltypes. + + Change-Id: Ia845fae6a84801cc7d9996767b99efb2753cbb48 + +2018-03-14 12:38 +0000 [b45bb476bb] Corey Farrell + + * cli: Enable ast_cli_completion_add on public completion generators. + + * ast_cli_complete + * ast_complete_channels + * ast_complete_applications + + These generators will now use ast_cli_completion_add if state == -1. + + Change-Id: I7ff311f0873099be0e43a3dc5415c0cd06d15756 + +2018-03-14 11:17 +0000 [92158b7f37] Ross Beer + + * res_pjsip_rfc3326.c: Account for more than one 'Reason' header + + ASTERISK-27741 + + Change-Id: I0aa59a54735c6d20b95c54db1bd095dbf93e7adf + +2018-03-12 08:05 +0000 [b0fff03bb5] Alexander Traud + + * install_prereq: Add SUSE. + + ASTERISK-27736 + + Change-Id: I4cafc8973349d50a7cb7919ddf0bb1aaef4bfc3e + +2018-02-16 21:11 +0000 [572a508ef2] Corey Farrell + + * loader: Convert reload_classes to built-in modules. + + * acl (named_acl.c) + * cdr + * cel + * ccss + * dnsmgr + * dsp + * enum + * extconfig (config.c) + * features + * http + * indications + * logger + * manager + * plc + * sounds + * udptl + + These modules are now loaded at appropriate time by the module loader. + Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so + the module loader will abort startup on failure of these modules. + + Some of these modules are still initialized or shutdown from outside the + module loader. logger.c is initialized very early and shutdown very + late, manager.c is initialized by the module loader but is shutdown by + the Asterisk core (too much uses it without holding references). + + Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f + +2018-03-13 16:39 +0000 [9e488dd482] Corey Farrell + + * core: Remove incorrect usage of attribute_malloc. + + GCC documentation states that when __attribute__((malloc)) is used it + should not return storage which contains any valid pointers. It + specifically mentions that realloc functions should not have the malloc + attribute, but this also means that complex initializers which could + contain initialized pointers should not use this attribute. + + Change-Id: If507f33ffb3ca3b83b702196eb0e8215d27fc7d2 + +2018-03-12 05:19 +0000 [d9776870e8] Alexander Traud + + * BuildSystem: Enable IMAP storage on openSUSE and Arch Linux. + + ASTERISK-27734 + + Change-Id: I8d6e6a1c08c031649764f5277fbbb85e57c3a9d4 + +2018-02-23 07:41 +0000 [ea9768ff07] Corey Farrell + + * stringfields: Remove MALLOC_DEBUG fields from struct ast_string_field_mgr. + + This causes MALLOC_DEBUG reporting to be slightly different, calls which + cause additional memory pools to be allocated now report the callers + location rather than the location which originally allocated the + string field structure. This reduces storage needed by string fields + and allows MALLOC_DEBUG to identify the source of additional allocations + rather than obscuring it by reporting the original allocation caller. + + Change-Id: Idd18e6639a87ab862079b580c114d90361412289 + +2018-03-10 03:33 +0000 [fee929c8ac] Corey Farrell + + * core: Remove non-critical cleanup from startup aborts. + + When built-in components of Asterisk fail to start they cause the + Asterisk startup to abort. In these cases only the most critical + cleanup should be performed - closing databases and terminating + proceses. These cleanups are registered using ast_register_atexit, all + other cleanups should not be run during startup abort. + + The main reason for this change is that these cleanup procedures are + untestable from the partially initialized states, if they fail it could + prevent us from ever running the critical cleanup with ast_run_atexits. + + Create separate initialization for dns_core.c to be run unconditionally + during startup instead of being initialized by the first dns resolver to + be registered. This ensures that 'sched' is initialized before it can be + potentially used. + + Replace ast_register_atexit with ast_register_cleanup in media_cache.c. + There is no reason for this cleanup to happen unconditionally. + + Change-Id: Iecc2df98008b21509925ff16740bd5fa29527db3 + +2018-03-12 06:40 +0000 [ea3b8bb080] Alexander Traud + + * install_prereq: Update FreeBSD libraries. + + Because the code review system Gerrit creates merge conflicts even when one line + apart another change happened, the previous update to the FreeBSD libraries had + to be rebased via Git. Because of a break for training of the original + contributor, this rebase was done by another contributor and the variant for + Asterisk 13 was cherry-picked to all branches. By this, dependencies for new + features added in newer Asterisk version got lost. This can be seen, when not + the original path set but a previous patch set is compared. + + This change here fixes this by adding those (optional) dependencies for + Asterisk 15 and newer (again). + + ASTERISK-27686 + + Change-Id: I6638a3d0dc37ad4ff5f94be15463e3dd8a2bfe74 + +2018-03-12 04:11 +0000 [9164be19d2] Alexander Traud + + * res_srtp: Add support for libsrtp2.x on openSUSE. + + Since ASTERISK-27253, no symbols from the header srtp2/crypto_types.h are used + anymore. Therefore, its include statement can be removed. This allows to compile + Asterisk on platforms which do not offer this private header, like openSUSE. + + ASTERISK-27733 + + Change-Id: I25c5cb8fa966043d1506ebef449e5a724412b4b6 + +2018-03-08 09:14 +0000 [5b525c9781] Alexander Traud + + * BuildSystem: Add NetBSD. + + Headers, libraries, and rpath. + + ASTERISK-27728 + ASTERISK-11015 + Reported by: Curt Sampson + + Change-Id: I50aa5fcd095937df32a2e33307caac7e79a8b5b7 + +2018-03-09 03:13 +0000 [c5f2332953] Alexander Traud + + * BuildSystem: For consistency, avoid double-checking via if clauses. + + In the script ./configure, AST_EXT_LIB_CHECK and AST_PKG_CONFIG_CHECK first test + whether parameter 1 was already found. Consequently, an if-test on PBX_ just a + line below is redundant, if exactly the same parameter 1 is used again. + + No performance gain is expected by this change. However, because this strategy + is used all over in ./configure except for two places, this change aims to + create more consistency: Only do something different if there is a reason to do + so. + + Change-Id: I4a6f48127b7af3a48168c917e888be1f70625027 + +2018-03-09 02:44 +0000 [36c8885c66] Alexander Traud + + * BuildSystem: Enable dladdr on non-Linux platforms like FreeBSD. + + ASTERISK-27641 + + Change-Id: I587e8ba0123c70fc10cfd8b0ac3299551f61d84b + +2018-03-08 13:53 +0000 [e6738b79b3] Richard Mudgett + + * Complete deprecating legacy modules. + + The menuselect comment was updated to deprecate these modules but the + AST_MODULE_INFO block at the end of file was missed. + + ASTERISK-27671 + + Change-Id: I63070b5c4d4f08af010c6034acd4793c1bcef839 + +2018-03-07 13:50 +0000 [7f4354c10f] Richard Mudgett + + * res_pjproject.c: Upgrade bundled PJPROJECT to 2.7.2 + + Update patches included in bundled PJPROJECT for the new version. + + ASTERISK-27730 + + Change-Id: Id3c8c8ad82126846bcd9768bc3d0a18d89be8944 + +2018-03-08 12:02 +0000 [9ff95e46e3] Alexander Traud + + * install_prereq: Add NetBSD. + + ASTERISK-27729 + + Change-Id: I7a706d51375d54cf5e36d32397bfe09a48670804 + +2018-03-08 09:04 +0000 [75cebc3e71] Alexander Traud + + * BuildSystem: Re-check for another UUID library only when previous check failed. + + As a side-effect, this avoids the ambiguous output: + checking for uuid_generate_random... no + which was printed always previously. + + ASTERISK-25586 + Reported by: John Nemeth + + Change-Id: I6d541dfcf453932a9856c5e251aa22e0e6c233c9 + +2018-03-08 05:28 +0000 [fc64a0e2b3] Alexander Traud + + * BuildSystem: Instead of $PJPROJECT_LIBS with s, use $PJPROJECT_LIB everywhere. + + In the script ./configure, + xyz_LIB is set by AST_PKG_CONFIG_CHECK and + xyz_LIBS is set by PKG_CHECK_MODULES within + AST_PKG_CONFIG_CHECK. Both are the same. In Asterisk normally the former and + only three times the latter was used. Let us use xyz_LIB without s, for + consistency with AST_EXT_LIB_CHECK. That eases understanding because now readers + do not have to know that xyz_LIB equals xyz_LIBS. + + Change-Id: I7359860a5d730cdc784c2c48e501a082196434d3 + +2018-03-06 06:28 +0000 [16f6e94033] Alexander Traud + + * BuildSystem: Enable PortAudio in NetBSD. + + In NetBSD, PortAudio 1 is still the default version. PortAudio 2 can be + installed side by side but gets placed in a 'portaudio2' subdirectory. To + find PortAudio 2 even in a subdirectory, the tool pkg-config is queried via + AST_PKG_CONFIG_CHECK. For those platforms, which do not list PowerAudio 2 + via pkg-config, the previous check remains and is executed thereafter. + + ASTERISK-27721 + + Change-Id: I4175500126909ad1b181fff8e11bb4a3a6ae4fa9 + +2018-03-07 14:36 +0000 [c8a521b6c8] Corey Farrell + + * Replace direct checks of option_debug with DEBUG_ATLEAST macro. + + Checking option_debug directly is incorrect as it ignores file/module + specific debug settings. This system-wide change replaces nearly all + direct checks for option_debug with the DEBUG_ATLEAST macro. + + Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0 + +2018-03-07 13:13 +0000 [1fe913f7bd] Richard Mudgett + + * BuildSystem regression: Fix errors reported by clean targets. + + Doing a 'make clean', 'make distclean', or 'make dist-clean' gets errors + about an invalid shell option: "/bin/sh: 0: Illegal option -". + + The clean targets do not include the makeopts file which defines GREP and + LDCONFIG because the file may not exist and the distclean/dist-clean + targets will delete it anyway. + + ASTERISK-27715 + + Change-Id: I33d40acdb03862bc89aeb6fb1ff497894a8ea7f5 + +2018-03-06 13:31 +0000 [88cef40f6e] Ross Beer + + * res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints + + ASTERISK-27554 + + Change-Id: If61c7faab7d2fa1031c056ed6268fe928e2391cf + +2018-03-06 08:14 +0000 [961dd9fe52] Sungtae Kim + + * voicemail: Fixed wrong voicemail message count + + Fixed wrong voicemail mailbox reference for Action: VoicemailUsersList. + + ASTERISK-27703 + + Change-Id: Ie6578ad80bba2bfaf34b84f0be978f59045ce6cd + +2018-03-07 09:32 +0000 [58f44f225a] Alexander Traud + + * utils: In Solaris, avoid a warning about an unused variable. + + When HAVE_GETHOSTBYNAME_R_5 was set by the script ./configure, GCC 7.3.0 found + an unused variable. Actually, the variable was used (set to a dummy value) but + the compiler optimization might have removed that. Instead, this change ensures + that the variable 'res' is only used when it is really required. + + Change-Id: Ic3ea23ccf84ac4bc2d501b514985b989030abab5 + +2018-03-07 01:45 +0000 [add03e207c] Corey Farrell + + * app_osplookup: Move header defines into the app. + + astosp.h is leftover from when logic was split between app_osplookup and + res_osp. All logic was moved into app_osplookup by 109737eb1c in 2006, + but astosp.h remained. This moves the remaining defines into + app_osplookup and deletes astosp.h. + + Change-Id: I0a6c4debd7c9543b608520b1765abfa4fab7b2fd + +2018-02-14 07:33 +0000 [75a35ee5e8] Jean Aunis + + * chan_sip: Fix improper RTP framing on outgoing calls + + The "ptime" SDP parameter received in a SIP response was not honoured. + Moreover, in the abscence of this "ptime" parameter, locally configured + framing was lost during response processing. + + This patch systematically stores the framing information in the + ast_rtp_codecs structure, taking it from the response or from the + configuration as appropriate. + + ASTERISK-27674 + + Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c + +2018-02-20 11:48 +0000 [3fb26df4ac] lvl + + * res_pjsip_session: properly handle SDP from a forked call with early media + + In handle_negotiated_sdp(), use session->active_media_state when + session->pending_media_state is empty. The 200's SDP should be fed into + handle_negotiated_sdp_session_media() together with the already negotiated + state, which is now in session->active_media_state instead. Only if both + the session's pending and active media are empty should + handle_negotiated_sdp() abort. + + ASTERISK-27441 + + Change-Id: If0d5150ffe6f38d8a854831fef37942258d4629c + +2018-03-05 08:01 +0000 [ef79e583ec] Alexander Traud + + * BuildSystem: Enable Lua in NetBSD. + + luaL_openlib got removed with Lua 5.2. + luaL_newstate is available in all versions. + + ASTERISK-27718 + + Change-Id: I9c8c8880315ee36ab740d7c40153306c0bfd6f71 + +2018-03-06 07:33 +0000 [162fc4fba6] Alexander Traud + + * BuildSystem: Depend not implicitly but explicitly on external libraries. + + ASTERISK-27722 + + Change-Id: Ie7b8c30d86cb00a54d6ac4e09e6f28f42d2bd52c + +2018-03-05 08:15 +0000 [99b6a14737] Alexander Traud + + * res_http_post: Enable GMime in NetBSD. + + ASTERISK-27719 + + Change-Id: I230c5f9f316b2e9465c093c13580f72ebbaf67a7 + +2018-03-05 04:16 +0000 [7e9734a858] Alexander Traud + + * BuildSystem: Enable autotools in NetBSD. + + ASTERISK-27716 + + Change-Id: I52525e35e1620341272219911d054a1e3d3ec01e + +2018-03-05 03:42 +0000 [b97905aaf2] Alexander Traud + + * BuildSystem: AC_PATH_PROG sets to colon character when not found. + + ASTERISK-27715 + Reported by: Corey Farrell + + Change-Id: I0d6d9572d1352dc7ad30c9917173f1e980d8c938 + +2018-03-03 09:06 +0000 [aabbb49e33] Alexander Traud + + * chan_unistim: NetBSD has an incompatible struct in_pktinfo. + + ASTERISK-27714 + Reported by: John Nemeth + + Change-Id: I1b84a89315a5f61222123d21bf35c59224da8990 + +2018-03-03 08:30 +0000 [5d19762b5f] Alexander Traud + + * BuildSystem: Cast any intptr_t explicitly to its proposed type. + + ASTERISK-27713 + + Change-Id: I90c769e3c7f8c26de8a3af11335862cec15a1b22 + +2018-03-03 06:56 +0000 [9749524520] Alexander Traud + + * BuildSystem: Detect whether uselocale(.) is available. + + ASTERISK-27712 + Reported by: Joerg Sonnenberger, D'Arcy Cain + + Change-Id: Idf1c9d43617a3e13028b95b313415903d80ef807 + +2018-03-03 03:53 +0000 [f7b845ff41] Alexander Traud + + * BuildSystem: Avoid re-defining of pthread_* on NetBSD. + + ASTERISK-27711 + + Change-Id: Idc9194035b2958b99f6b01eb5b438d45a074565b + +2018-03-02 07:05 +0000 [313a9fe255] Alexander Traud + + * BuildSystem: Install init scripts on openSUSE Tumbleweed. + + ASTERISK-27710 + + Change-Id: I4c777e41b31d4415bbe21cb435ad47b43ebb5467 + +2018-03-02 05:12 +0000 [a9c02e484a] Alexander Traud + + * BuildSystem: Avoid == for comparison in ./configure. + + ASTERISK-27709 + Reported by: John Nemeth + + Change-Id: I11b1ae8fd404c04066f1458f5d71f9536359d58d + +2018-02-19 19:55 +0000 [c711e4076a] Richard Mudgett + + * core: Remove ABI effects of MALLOC_DEBUG. + + This allows asterisk to be compiled with MALLOC_DEBUG to load modules + built without MALLOC_DEBUG. Now pre-compiled third-party modules will + still work regardless of MALLOC_DEBUG being enabled or not. + + Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10 + +2018-02-27 15:40 +0000 [1a36a452bd] Richard Mudgett + + * pjproject: Add cache_pools debugging option. + + The pool cache gets in the way of finding use after free errors of memory + pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a + pool is released because it gets put into the cache instead of being + freed. + + * Added the "cache_pools" option to pjproject.conf. Disabling the option + helps track down pool content mismanagement when using valgrind or + MALLOC_DEBUG. The cache gets in the way of determining if the pool + contents are used after free and who freed it. + + To disable the pool caching simply disable the cache_pools option in + pjproject.conf and restart Asterisk. + + Sample pjproject.conf setting: + [startup] + cache_pools=no + + * Made current users of the caching pool factory initialization and + destruction calls call common routines to create and destroy cached pools. + + ASTERISK-27704 + + Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828 + +2018-01-31 11:48 +0000 [eacee03f0e] Corey Farrell + + * gitreview: Reorder and add padding. + + Change-Id: I459dc320a8c9452a01eed6f403d786741587c890 + +2018-02-23 21:24 +0000 [7b01236028] Michael Cargile + + * apps/app_amd.c: Fixed total time and silence calculations + + Between Asterisk 11 and Asterisk 13 there was a significant increase + in the number of AST_FRAME_NULL frames being processed by app_amd.c's + main loop. Each AST_FRAME_NULL frame was being counted as 100ms + towards the total time and silence. This may have been accurate + when app_amd.c was orginally added, but it is not in Asterisk 13. + As such the total analysis time and silence calculations were way + off effectively breaking app_amd.c + + * Additional debug messages were added + * AST_FRAME_NULL are now ignored + + ASTERISK-27610 + + Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9 + +2018-02-23 14:58 +0000 [7e2128c8e6] George Joseph + + * ast_coredumper: Minor fixes + + * Fix --tarball-config so the option doesn't cause an error. + + * Allow for missing /etc/os-release. + + * Add a sleep between tarballing the coredump and removing the + output directory to allow the filesystem to settle. + + Change-Id: I73e03b13087978bcc7f6bc9f45753990f82d9d77 + +2018-02-22 14:27 +0000 [0be1c388e4] Ben Ford + + * Add extended properties to rtp_engine for RTP retransmission support. + + A couple of additional properties are needed in rtp_engine to enable + support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and + AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically + if an endpoint has the webrtc option enabled. While this adds no + functionality currently, it will serve as a building block for future + changes for RTP retransmission support. + + For more information, refer to the wiki page: + https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements + + Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc + +2018-02-23 10:09 +0000 [a7927471ad] Corey Farrell + + * core: Fix handling of maximum length lines in config files. + + When a line is the maximum length "\n" is found at sizeof(buf) - 2 since + the last character is actually the null terminator. In addition if a + line was exactly 8190 plus a multiple of 8192 characters long the config + parser would skip the following line. + + Additionally fix comment in voicemail.conf sample config. It previously + stated that emailbody can only contain up to 512 characters which is + always wrong. The buffer is normally 8192 characters unless LOW_MEMORY + is enabled then it is 512 characters. The updated comment states that + the line can be up to 8190 or 510 characters since the line feed and + NULL terminator each use a character. + + ASTERISK-26688 #close + + Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015 + +2018-02-22 13:53 +0000 [bb9c1938a0] Richard Mudgett + + * res_pjsip_refer.c: Fix attended transfer race condition crash. + + The transferrer's session channel was destroyed by the transferrer's + serializer thread in a race condition with the transfer target's + serializer thread during an attended transfer. The transfer target's + serializer was attempting to clean up a deferred end status on behalf of + the transferrer's channel when it should have passed the action to the + transferrer's serializer. When the transfer target's serializer lost the + race then both threads wind up trying to end the transferrer's session. + + * Push the ast_sip_session_end_if_deferred() call onto the transferrer's + serializer to avoid a race condition that results in a crash. The + session_end() function that could be called by + ast_sip_session_end_if_deferred() really must be executed by the + transferrer's serializer to avoid this kind of crash. + + ASTERISK-27568 + + Change-Id: Iacda724e7cb24d7520e49b2fd7e504aa398d7238 + +2018-02-17 03:28 +0000 [c4c5d00528] Alexander Traud + + * install_prereq: Update FreeBSD libraries. + + deleted + autoconf gcc libsamplerate sqlite + + changed + binutils to libbfd + freetds-devel to freetds + gmime2 to gmime26 + mysql55-client to mysql57-client + + added + alsa-lib bison bzip2 cclient corosync doxygen libedit flex graphviz + libhoard libical libilbc libltdl lua neon newt net-snmp + openldap-client openssl patch pkgconf portaudio postgresql10-client + python radcli speexdsp subversion uriparser xmlstarlet libzip + + ASTERISK-27686 + + Change-Id: Ibe88c9b26e59c30d26cdb313a3ef01c9f37ac80d + +2018-02-22 10:54 +0000 [50d9af101e] Sean Bright + + * func_audiohookinherit: Remove deprecated module. + + Change-Id: Id52f719078a65c4b2eee7ab99d761eba6b6aed94 + +2018-02-21 12:52 +0000 [f083edc43c] Richard Mudgett + + * manager.c: Fix lseek() parameter order. + + ASTERISK-27659 + + Change-Id: I04a2705d2cb7df250769967bc59e2b397a49b797 + +2018-02-20 13:11 +0000 [39f733406d] Richard Mudgett + + * bridge_simple.c: Fix stream topology handling. + + The handling of stream topologies was not protected by channel locks in + simple_bridge_request_stream_topology_change(). + + * Fixed topology handling to be protected by channel locks where needed in + simple_bridge_request_stream_topology_change(). + + ASTERISK-27692 + + Change-Id: Ica5d78a6c7ecf4f0b95fb16de28d3889b32c4776 + +2018-02-05 16:46 +0000 [6436137959] Sean Bright + + * AST-2018-006: Properly handle WebSocket frames with 0 length payload. + + In ast_websocket_read() we were not adequately checking that the + payload_len was non-zero before passing it to ws_safe_read(). Calling + ws_safe_read with a len argument of 0 will result in a busy loop until + the underlying socket is closed. + + ASTERISK-27658 #close + + Change-Id: I9d59f83bc563f711df1a6197c57de473f6b0663a + +2018-01-31 13:37 +0000 [880c69f00f] Kevin Harwell + + * AST-2018-003: Crash with an invalid SDP fmtp attribute + + pjproject's fmtp retrieval function failed to catch invalid fmtp attributes. + Because of this Asterisk would crash if given an SDP with an invalid fmtp + attribute. + + When retrieving the format this patch now makes sure the fmtp attribute is + available. If not available it now returns an error status. + + ASTERISK-27583 #close + + Change-Id: I5cebe000ce2d846cae3af33b6d72c416e51caf2f + +2018-01-31 13:33 +0000 [d3a398cf90] Kevin Harwell + + * AST-2018-002: Crash with an invalid SDP media format description + + pjproject's media format parsing algorithm failed to catch invalid values. + Because of this Asterisk would crash if given an SDP with a invalid media + format description. + + When parsing the media format description this patch now properly parses the + value and returns an error status if it can't successfully parse/convert the + value. + + ASTERISK-27582 #close + + Change-Id: I883b3a4ef85b6972397f7b56bf46c5779c55fdd6 + +2018-02-06 12:07 +0000 [758409de56] George Joseph + + * AST-2018-005: res_pjsip_transport_management: Move to core + + Since res_pjsip_transport_management provides several attack + mitigation features, its functionality moved to res_pjsip and + this module has been removed. This way the features will always + be available if res_pjsip is loaded. + + ASTERISK-27618 + Reported By: Sandro Gauci + + Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d + +2018-02-06 11:28 +0000 [de871515ba] George Joseph + + * AST-2018-005: Fix tdata leaks when calling pjsip_endpt_send_response(2) + + pjsip_distributor: + authenticate() creates a tdata and uses it to send a challenge or + failure response. When pjsip_endpt_send_response2() succeeds, it + automatically decrements the tdata ref count but when it fails, it + doesn't. Since we weren't checking for a return status, we weren't + decrementing the count ourselves on error and were therefore leaking + tdatas. + + res_pjsip_session: + session_reinvite_on_rx_request wasn't decrementing the ref count + if an error happened while sending a 491 response. + pre_session_setup wasn't decrementing the ref count if + while sending an error after a pjsip_inv_verify_request failure. + + res_pjsip: + ast_sip_send_response wasn't decrementing the ref count on error. + + ASTERISK-27618 + Reported By: Sandro Gauci + + Change-Id: Iab33a6c7b6fba96148ed465b690ba8534ac961bf + +2018-02-06 11:21 +0000 [c53d8dcb68] George Joseph + + * AST-2018-005: Add a check for NULL tdata in ast_sip_failover_request + + It was discovered that there are some corner cases where a pjsip tsx + might have no last_tx so calling ast_sip_failover_request with + a NULL last_tx as its tdata would cause a crash. + + ASTERISK-27618 + Reported By: Sandro Gauci + + Change-Id: Ic2b63f6d4ae617c4c19dcdec2a7a6156b54fd15b + +2018-02-07 08:09 +0000 [d424850d58] Joshua Colp + + * AST-2018-004: Restrict the number of Accept headers in a SUBSCRIBE. + + When receiving a SUBSCRIBE request the Accept headers from it are + stored locally. This operation has a fixed limit of 32 Accept headers + but this limit was not enforced. As a result it was possible for + memory outside of the allocated space to get written to resulting + in a crash. + + This change enforces the limit so only 32 Accept headers are + processed. + + ASTERISK-27640 + Reported By: Sandro Gauci + + Change-Id: I99a814b10b554b13a6021ccf41111e5bc95e7301 + +2018-01-13 08:04 +0000 [e70c4ec84d] Joshua Colp + + * AST-2018-001: rtp / channel: Don't allow an unnegotiated format to be passed up. + + When an RTP packet is received by an RTP engine it has to map the + payload into the Asterisk format. The code was incorrectly checking + our own static list for ALL payloads if it couldn't find a negotiated one. + This included dynamic payloads. If the payload mapped to a format + of a different type (for example receiving a video packet on an audio + RTP instance) then the core stream code could cause a crash if a legacy + channel driver was in use as no stream would be present. + + To provide further protection the core stream code will no longer assume + that a video or audio frame will always have a stream for legacy channel + drivers. If no stream is present the frame is dropped. + + ASTERISK-27488 + + Change-Id: I022556f524ad8379ee73f14037040af17ea3316a + +2018-02-20 13:11 +0000 [e2f98fbd63] Richard Mudgett + + * channel.c: Fix typo. + + Change-Id: I4eeedf89085697e81c354eb92d546686c67b0b5b + +2018-02-20 10:33 +0000 [259c80675e] Joshua Colp + + * chan_sip: Emit a second ringing event to ensure channel is found. + + When constructing a dialog-info+xml NOTIFY message a ringing channel + is found if the state is ringing and further information is placed into + the message. Due to the migration to the Stasis message bus this did + not always work as expected. + + This change raises a second ringing event in such a way to guarantee + that the event is received by chan_sip and another lookup is done to + find the ringing channel. + + ASTERISK-24488 + + Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c + +2018-02-20 04:31 +0000 [0ad13949c1] Corey Farrell + + * doc/lang/language-criteria.txt: Link to wiki. + + This document is out of date and is superseded by content on the + Asterisk wiki. + + ASTERISK-24386 #close + + Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822 + +2018-02-19 04:21 +0000 [4b555d7147] Thomas Guebels + + * res_rtp_asterisk: Fix ICE candidate nomination + + If the ICE role is not set right away, we might have a role conflict + that stays undetected and ICE finishing with successful tests and no + candidate nominated. This was introduced by ASTERISK-27088. + + To avoid this, we set the role as soon as before but only if the ICE + state permits it: still checking and not yet nominating candidates or + completed. + + ASTERISK-27646 + + Change-Id: I5dbc69ad63cacbb067922850fbb113d479bd729c + +2018-02-18 10:27 +0000 [8b18247af6] Sean Bright + + * res_http_websocket: Don't leak memory on read failure + + Change-Id: Ic449ea832bc81a1671c0e910c5fbe8c683e3da89 + +2018-02-19 03:57 +0000 [97c21e9cb3] Corey Farrell + + * core: Rename sounds_index.c to sounds.c. + + This will make the source filename match the 'module reload sounds' + command. This will allow conversion to a built-in module in Asterisk 16 + without needing to redefine AST_MODULE. + + Change-Id: Ifb8e489575b27eb33d8c0b6a531f266670557f6e + +2018-02-19 02:49 +0000 [e03f0f9572] Corey Farrell + + * config: Fix locking for extconfig reload. + + Expand locking to include full reload process for extconfig to ensure + nothing can read the config mappings between clearing and reloading. + + Change-Id: I378316bad04f1b599ea82d0fef62b8978a644b92 + +2018-02-15 14:09 +0000 [e4a5c9ccf4] Sean Bright + + * res_pjsip_header_funcs: Various cleanups + + * Prefer strcasecmp() over stricmp() + * Use a list with no lock since we never actually lock + * Minor cleanups to error messages + + Change-Id: I8446f44795ee8f3072e1c1f9193c6912dfc0c42b + +2018-02-17 08:49 +0000 [a70c92121d] Alexander Traud + + * rtp_engine: Load format name / mime type in uppercase again. + + This reverts a previous change partly. + + ASTERISK-27689 + + Change-Id: Ia3d2f282db6995be8c1c253b5d52f6038761e8af + +2018-02-16 17:58 +0000 [525c0251c0] Corey Farrell + + * BuildSystem: Use single bootstrap.sh for Asterisk and menuselect. + + This causes the root bootstrap.sh script to generate configure scripts + for both Asterisk and menuselect. This ensures that both configure + scripts are generated with the same version of autotools and avoids + situations where shared autoconf macros get modified without + regenerating the menuselect script. + + Change-Id: I2bfd8537bbb63b3d46b11efabbb15eaaf9ef731a + +2018-02-16 13:33 +0000 [65a4084060] Sean Bright + + * res_pjsip: Endpoint destruction does not free DTLS configuration + + ASTERISK-27679 #close + Reported by: Mak Dee + + Change-Id: I89a2783a11be0763bf123d1619ed176b6225cf42 + +2018-02-16 12:42 +0000 [a7e7302ab6] Alexander Traud + + * install_prereq: Update OpenBSD libraries. + + deleted + jack sqlite + + renamed + freetds-0.63p1-msdblib to freetds + mysql-client to mariadb-client + + added + bison bzip2 c-client doxygen e2fsprogs graphviz gsm libical jansson libltdl + lua neon net-snmp libsrtp portaudio-svn postgresql-client python speexdsp + subversion uriparser xmlstarlet + fftw3 libsndfile + + ASTERISK-27684 + + Change-Id: I26bdcb0a1d0e484a8dad1052da97f194aefd3370 + +2018-02-16 12:30 +0000 [14796f529e] Alexander Traud + + * BuildSystem: Allow newer autotools on OpenBSD. + + ASTERISK-27683 + + Change-Id: I5ec9dafbb0c16b6f2740c641980bc2eaaf995624 + +2017-10-16 07:36 +0000 [976afd26ab] Torrey Searle + + * contrib/script/sip_to_pjsip: add support for realtime + + Add a new script that can read from legacy realtime peers & generate + an sql file for populating pjsip endpoints, identify, and aor records. + + ASTERISK-27348 #close + + Change-Id: Idd3d7968a3c9c3ee7936d21acbdaf001b429bf65 + +2018-02-16 07:52 +0000 [dda73c5018] Alexander Traud + + * BuildSystem: Fix a typo related to ./configure --prefix= on OpenBSD. + + Reported by: Stuart Henderson + + Change-Id: Ieae8624f48b6ae78cf29930b9a45a3c842c7a764 + +2018-02-16 05:58 +0000 [5fd59014a5] Alexander Traud + + * res_calendar: Specialized calendars depend on symbols of general calendar. + + ASTERISK-27680 + + Change-Id: Ifb77912e424fe3710a025c18526fada673ec0b79 + +2018-02-16 06:41 +0000 [c674efa996] Alexander Traud + + * BuildSystem: Enable IMAP storage on OpenBSD. + + ASTERISK-27681 + Reported by: Stuart Henderson + + Change-Id: Ifb6b614acb251b695b9417d76510e73eb335b679 + +2018-02-16 04:50 +0000 [2c814afb86] Alexander Traud + + * BuildSystem: Enable system provided libedit on OpenBSD. + + ASTERISK-27677 + + Change-Id: I0854e3616d1361ae9b6907d3d3444a02784ac62b + +2018-02-15 14:30 +0000 [af2dd3a678] Sean Bright + + * bridge_roles: Use a non-locking linked list where appropriate + + Also explicitly initialize with the AST_LIST_HEAD_NOLOCK_INIT macro for + clarity. + + Change-Id: I4bc39ec33bc3ff77e1a971a01ace87deb965be3f + +2018-02-15 13:29 +0000 [303e43f8a6] Sean Bright + + * res_pjsip: Use pjsip_sip_uri.user_param instead of other_param + + There is a dedicated slot in the pjsip_sip_uri for the 'user' + parameter, so use that instead of adding to the list of generic URI + parameters. + + Change-Id: I0a0ce8a60ecee27489735bf56fd707719d8c2ed6 + +2018-02-12 07:37 +0000 [8ac198aff3] Alexander Traud + + * BuildSystem: Remove chan_h323 leftovers. + + ASTERISK-27670 + + Change-Id: I07a8ef8bbd6001e25711fa1bff152eb6c9efa729 + +2018-01-17 08:17 +0000 [6b6b3ffa5b] Alexander Traud + + * BuildSystem: Invoke ldconfig with previous path. + + On OpenBSD, gmake uninstall{-all} registered only libraries from /usr/lib and + lost those from /usr/local/lib. Instead, invoke ldconfig on a path. + + ASTERISK-27595 + + Change-Id: I4aa2c0b5e07119d1a556f8ff6349eaf09e986888 + +2017-12-22 15:27 +0000 [9f74afbdcf] Corey Farrell + + * Deprecate legacy modules. + + * app_fax (replaced by res_fax). + * res_config_sqlite (replaced by res_config_sqlite3). + * res_monitor (replaced by app_mixmonitor). + + This is related to ASTERISK~23657 but does not resolve that ticket. + Resolving that ticket would require complete removal of res_monitor. + + ASTERISK-27671 #close + + Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49 + +2018-01-28 03:02 +0000 [f9ba31bb21] Alexander Traud + + * BuildSystem: Do not warn when bash is not installed. + + ASTERISK-27631 + + Change-Id: Iefdf268b0b98c3e7d8089ba87cf78136ac1d785b + +2018-02-12 22:15 +0000 [9e45d3f893] Corey Farrell + + * main/asterisk.c: Remove silly usage of RAII_VAR. + + Change-Id: I7e2996397fbd3c3a6a69dd805c38448ddfc34ae9 + +2017-12-25 19:32 +0000 [02ee296f81] Corey Farrell + + * optional_api: Refactor to use vector's and standard allocators. + + * Replace ad-hoc array management with macro's from vector.h. + * Remove redundent logger messages. + * Use normal Asterisk allocators instead of directly using libc + allocators. + * Free memory when an API has no implementation or users. + + Change-Id: Ic6ecb31798d4a78e7df39ece86a68b60eac05bf5 + +2018-02-11 15:27 +0000 [8372138cce] Richard Mudgett + + * chan_sip.c: Fix crash processing CANCEL. + + Check if initreq data string exists before using it when processing a + CANCEL request. + + ASTERISK-27666 + + Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97 + +2018-02-06 14:27 +0000 [cb4cfb8c43] Sungtae Kim + + * manager: Add AMI event Load/Unload + + Add an AMI events Load and Unload for notify when the + module has been loaded and unloaded. + + ASTERISK-27661 + + Change-Id: Ib916c41eddd63651952998f2f49c57c42ef87a64 + +2018-01-30 20:31 +0000 [04490fb1d8] Corey Farrell + + * json: Add conditionals to avoid locking if Jansson is thread safe. + + Jansson is thread safe for all read-only functions and reference + counting starting v2.11. This allows simplification of our code and + removal of locking around reference counting and dumping. + + Change-Id: Id985cb3ffa6681f9ac765642e20fcd187bd4aeee + +2018-02-12 06:16 +0000 [b21915bd1c] Alexander Traud + + * pjproject_bundled: Disable G.729 from Belledonne Communications. + + When is installed, PJProject + tries to link that. Support for this bcg729 was added with PJProject 2.7. The + issue happens, because Teluu enabled that new feature on default. + + ASTERISK-27584 + Reported by: Stuart Henderson + + Change-Id: I88b6b18ad777bcfe2d8201187b4b90eec0a172a6 + +2018-02-12 05:38 +0000 [97f45d5816] Alexander Traud + + * codecs: Add support for WebRTC iLBC 2.0. + + When the latest version of that library was installed, Asterisk did not build. + + ASTERISK-27669 + Reported by: Николай Михо + + Change-Id: I27e09bb875fdd56423bd9fae1be85fddb428eb96 + +2018-02-12 01:26 +0000 [9fddc8b4dc] Corey Farrell + + * core: Remove embedded editline. + + This removes the embedded copy of editline from the Asterisk source + tree, making a system copy of libedit mandatory in Asterisk 16+. + + ASTERISK-27634 #close + + Change-Id: Iedb64ad92acb78419f3caefedaa2bb7cd2a1a33f + +2018-01-30 09:58 +0000 [32e610d9e6] Alexander Traud + + * backtrace: Avoid potential spurious output. + + clang 4.0 found this via -Wlogical-not-parentheses. + + ASTERISK-27642 + + Change-Id: I9ec3e144d425a976c02811bd23cd0c533d2eca4e + +2018-02-10 05:39 +0000 [971378bbdb] Alexander Traud + + * install_prereq: Update Debian/Ubuntu libraries. + + ASTERISK-27555 + + Change-Id: Idc36e91db30c0163c560d04c5a82bca5d6ce92a8 + +2018-02-09 12:06 +0000 [b2fcb30d38] Richard Mudgett + + * cdr.c: Fix runtime leak of CDR records. + + Need to remove all CDR's listed by a CDR object from the active_cdrs_all + container including the root/master record. + + ASTERISK-27656 + + Change-Id: I48b4970663fea98baa262593d2204ef304aaf80e + +2018-01-31 17:48 +0000 [67cd90f10d] Richard Mudgett + + * app_confbridge: ConfbridgeList event has standard channel shapshot headers. + + * Made the AMI ConfbridgeList action's ConfbridgeList events output all + the standard channel snapshot headers instead of a few hand-coded channel + snapshot headers. The benefit is that the CallerIDName gets disruptive + characters like CR, LF, Tab, and a few others escaped. However, an empty + CallerIDName is now output as "" instead of "". + + ASTERISK-27651 + + Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977 + +2018-01-31 15:45 +0000 [f4b161440b] Richard Mudgett + + * app_confbridge: Add the Muted header to ConfbridgeJoin AMI event. + + ASTERISK-27651 + + Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34 + +2017-12-19 02:52 +0000 [5b8fea93d1] Oron Peled + + * chan_console: don't read and write at the same time + + It seems that the ALSA backend of PortAudio doesn't know how to both + read and write at the same time by adding a per-device mutex. + + FIXME: currently only a draft version. Need to either auto-detect + we work with the ALSA backend or add an extra configuration option + to use this mutex. + + ASTERISK-27426 #close + + Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb + +2018-02-02 17:35 +0000 [1017db107c] Richard Mudgett + + * endpoint identifiers: Some code cleanup. + + res_pjsip_endpoint_identifier_user.c: + * Fix copy/paste error in find_endpoint(). We were using a constant + "anonymous" string instead of the passed in endpoint_name when checking + the transport domain for an endpoint match. + * Eliminate RAII_VAR in find_endpoint(). + * Remove always true check in find_transport_state_in_use(). + * Remove useless CMD_STOP in find_transport_state_in_use(). + + res_pjsip_endpoint_identifier_anonymous.c: + * Eliminate RAII_VAR in anonymous_identify(). + * Remove always true check in find_transport_state_in_use(). + * Remove useless CMD_STOP in find_transport_state_in_use(). + + Change-Id: I86924c31db5bd225ca0c1219c761b668c6f91189 + +2018-02-02 17:20 +0000 [b71e469d68] Richard Mudgett + + * res_pjsip/config_domain_aliases.c: Add check for missing domain. + + What is the point of defining an alias and not saying what is being + aliased? + + Change-Id: I98a892016ed61dcf5efeb6619fd748925103f0be + +2018-02-02 15:11 +0000 [0960de71ae] Richard Mudgett + + * res_pjsip.c: Fix documentation typos. + + Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068 + +2018-02-02 15:43 +0000 [bef49d90c1] Richard Mudgett + + * res_sorcery_realtime.c: Fix ref leak if object failed to apply. + + Change-Id: I3c7106ff77009754725cee790eadf5da44154ab6 + +2018-01-24 19:58 +0000 [7e32adf044] Sungtae Kim + + * manager.c: Fixed "(null):" header in AMI AsyncAGIEnd event + + * Changed to create ami_event string only when the given blob is not + json_null(). + * Fixed bad expression. + + ASTERISK-27621 + + Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f + +2018-02-01 13:01 +0000 [73f92c2c52] Joshua Elson + + * res_pjsip_mwi.c: Fix null pointer crash + + ASTERISK-27652 #close + + Change-Id: I78a0d38bfd8d0d82830f3d53da04872d6b67284d + +2018-02-01 15:03 +0000 [fc98843d4b] Sean Bright + + * appdocsxml.xslt: Add Language to channel snapshot transformation + + Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8 + +2018-01-31 15:40 +0000 [3419a048b9] Richard Mudgett + + * manager.c: Fix potential memory leak and corruption. + + ast_str_append_event_header() could potentially leak and corrupt memory if + the ast_str needed to expand to add the AMI event header. + + * Fixed to return error if the ast_str_append() failed. + + Change-Id: I92f36b855540743b208d76e274152ee2d758176d + +2018-01-31 17:27 +0000 [bcfe172f8d] Richard Mudgett + + * manager_channels.c: Reordered ast_manager_build_channel_state_string_prefix() + + * Made not allocate memory if the channel snapshot is an internal channel. + + * Free memory earlier when no longer needed. + + Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38 + +2018-01-31 12:44 +0000 [4e4428ef3c] Corey Farrell + + * res_pjsip_registrar_expire: Delete empty module. + + Verified nothing in the testsuite lists this module as a dependency. + + Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813 + +2018-01-30 19:22 +0000 [1ccac0be0e] Richard Mudgett + + * bridge_softmix.c: Report not talking immediately when muted. + + Currently in app_confbridge if someone mutes a channel while that channel + is talking, the talk detection code is suspended while the channel is + muted. As far an an external observer is concerned, the muted channel's + talk status is still "talking" even though the channel is not contributing + audio to the conference bridge. When the channel is later unmuted, it + takes the usual 'dsp_silence_threshold' option time to clear the talking + status even though the channel may have stopped talking while the channel + was muted. + + * In bridge_softmix.c, clear the talking status and report talking stopped + if the channel was talking when the channel is muted. When the channel is + unmuted and the channel is still talking then report the channel as + talking since it is contributing audio to the bridge again. + + ASTERISK-27647 + + Change-Id: Ie4fdbc05a0bc7343c2972bab012e2567917b3d4e + +2018-01-30 15:00 +0000 [b9024197ab] Richard Mudgett + + * app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. + + The dsp_talking_threshold does not represent time in milliseconds. It + represents the average magnitude per sample in the audio packets. This is + what the DSP uses to determine if a packet is silence or talking/noise. + + Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443 + +2018-01-31 11:00 +0000 [6c5e3226ec] Richard Mudgett + + * res_pjsip_registrar.c: Fix compiler error. + + Need to include signal.h to define pthread_kill() and SIGURG. + + Change-Id: I10ae3aa4bf8e7386ac29ade78c0f2caed8e674fa + +2018-01-30 23:05 +0000 [60701b3252] Corey Farrell + + * res_pjsip_session: Prevent crash during shutdown. + + pjproject does not have a function to reverse pjsip_inv_usage_init. + This means we need to ignore any calls to the functions once shutdown is + final. + + ASTERISK-27571 #close + + Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d + +2018-01-27 13:03 +0000 [720dbb5745] Corey Farrell + + * core: Create ast_atomic macro's. + + Create ast_atomic macro's to provide a consistent interface to the + common functionality of __atomic and __sync built-in functions. + + ASTERISK-27619 + + Change-Id: Ieba3f81832a0e25c5725ea067e5d6f742d33eb5b + +2018-01-28 10:10 +0000 [2b9aa6b5bb] George Joseph + + * res_pjsip_pubsub: Prune subs with reliable transports at startup + + In an earlier release, inbound registrations on a reliable transport + were pruned on Asterisk restart since the TCP connection would have + been torn down and become unusable when Asterisk stopped. This same + process is now also applied to inbound subscriptions. + + Also fixed issues in res_pjsip_registrar where it wasn't handling the + monitoring correctly when multiple registrations came in over the same + transport. + + To accomplish this, the pjsip_transport_event feature needed to + be refactored to allow multiple monitors (multiple subcriptions or + registrations from the same endpoint) to exist on the same transport. + Since this changed the API, any external modules that may have used the + transport monitor feature (highly unlikey) will need to be changed. + + ASTERISK-27612 + Reported by: Ross Beer + + Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36 + +2018-01-29 13:46 +0000 [81db0aca0f] George Joseph + + * res_pjsip_registrar_expire: Refactor into res_pjsip_register + + res_pjsip_registrar_expire remains as an empty module for now. + + Change-Id: Ib93698938bae548d2199cb542f3692d1a171239f + +2018-01-29 07:51 +0000 [cf21e9fc97] Corey Farrell + + * Sample modules.conf: comment out example load statement. + + The sample modules.conf explicitly loaded res_musiconhold.so. This is + redundent as autoload=yes is already set. It causes warnings if + res_musiconhold.so was not installed and results in an unexpected load + if the admin disables autoload without remembering to remove the + res_musiconhold load statement. + + Also remove reference to unknown module pbx_gtkconsole. + + Change-Id: Ib01888994d9f1364b14d3c9fb6ff96774a6e580a + +2018-01-29 10:20 +0000 [913773cd75] Alexander Traud + + * BuildSystem: Enable autotools in FreeBSD. + + In the current versions of FreeBSD, the apps of GNU autotools do not need to + be called with a version anymore. The latest version can be invoked directly. + Additionally, the script ./bootstrap.sh asked for autoconf 2.62 and + automake 1.9, versions which are not available as port anymore. + + ASTERISK-27637 + + Change-Id: Id7b94b80e78cc943a40ba79b697e3f70019820a7 + +2018-01-29 10:00 +0000 [156b12340e] Alexander Traud + + * app_voicemail: Avoid always true when using pointer address. + + clang 4.0 warned about this. + + ASTERISK-27635 + + Change-Id: I213f230607d7fbe97c0f5f2d60da9cbf5a2d8231 + +2018-01-19 05:13 +0000 [e7f8ef1935] Alexander Traud + + * install_prereq: Update RHEL/CentOS/Fedora libraries. + + deleted + automake git ncurses-devel pjproject-devel sqlite2-devel libsqlite3x-devel + + renamed + radiusclient-ng-devel to radcli-devel + gmime22-devel to gmime-dev + + added + alsa-lib-devel bash binutils-devel bison doxygen flex hoard make pkgconfig + speexdsp-devel uriparser-devel uw-imap-devel wget xmlstarlet zlib-devel + codec2-devel fftw-devel libsndfile-devel unbound-devel + + ASTERISK-27599 + Reported by: Said Masoud + + Change-Id: I05bb0af98ae532b2d5f37478e38b8f0762b1c035 + +2018-01-28 05:20 +0000 [aaf14670b5] Alexander Traud + + * BuildSystem: Remove unused variables. + + Because of a copy-and-paste from the script build_tools/download_externals, + the script build_tools/list_valid_installed_externals got its local variables. + However in the latter, three variables were not used actually. + + Change-Id: I252de5a98c17ea54459174875357c22c2eebe8d5 + +2018-01-25 12:06 +0000 [84a6365164] Corey Farrell + + * loader: Use ast_cli_completion_add for 'module load' completion. + + This addresses all performance issues with 'module load' completion. In + addition to using ast_cli_completion_add we stop using libedit's + filename_completion_function, instead using ast_file_read_dir. This + ensures all results are produced from a single call to opendir. + + Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134 + +2018-01-27 09:44 +0000 [d9e42f27b9] Alexander Traud + + * core: Fix unused variable error in handle_show_sysinfo. + + The previous fix broke the case + HAVE_SYSINFO = no + HAVE_SYSCTL = yes + HAVE_SWAPCTL = no + which occurs on FreeBSD 11.1 for example. + + ASTERISK-26563 + + Change-Id: If77c39bc75f0b83a6c8a24ecb2fa69be8846160a + +2018-01-27 08:54 +0000 [3c26eec043] Alexander Traud + + * editline: Avoid shifting a negative signed value. + + clang 4.0 warned about this. + + ASTERISK-27630 + + Change-Id: Ie2725048c661c1792d8b1d498575144350b6e9ba + +2018-01-27 03:25 +0000 [c38da18ec6] Alexander Traud + + * headers: Consistent use of typeof and/or __typeof__. + + Because of a copy-and-paste error, the Asterisk project was using __typeof + instead of typeof. It works because typeof, __typeof, and __typeof__ are + supported by GCC, but here the escaped variant was not intended. Therefore, + for consistence, we change this to typeof. + + Change-Id: I2a962c3e596e882f691a19345445b14571a5f07c + +2018-01-24 18:25 +0000 [5d320d2d4b] Richard Mudgett + + * Update sounds release to fix siren7 and siren14 files. + + ASTERISK-16172 + + Change-Id: I2fb564258cd4db0f35952ad48b8687355c2dcad3 + +2018-01-15 11:08 +0000 [6da970bfb9] Alexander Traud + + * BuildSystem: Raise autoconf version requirement to 2.60a. + + AC_COMPUTE_INT requires at least autoconf 2.60a. + + This affects only those who contribute to Asterisk, only those who had to use + the script ./bootstrap.sh. Furthermore, this change just makes sure nobody is + using a too old autoconf. + + ASTERISK-16951 + + Change-Id: Ibca850e2fe0e77d935207bd959bacf7197d7f637 + +2018-01-26 06:48 +0000 [0afff31ed0] Alexander Traud + + * install_prereq: Download latest Jansson. + + ASTERISK-27603 + + Change-Id: I65c587534c0ae364f063d68da1bed40bb3d5e8aa + +2018-01-01 15:59 +0000 [39fcecad59] Corey Farrell + + * core: Tweak startup order. + + Move initialization of units which do not require configuration to occur + before preload modules. This leaves only units which load config between + module preload and regular load stages. + + Change-Id: I1d15384acad16a22c3498124421af474fa517478 + +2018-01-25 01:37 +0000 [23381d2c5e] Corey Farrell + + * Build System: Require __sync or __atomic functions. + + This change causes the configure script to throw an error if neither + __sync nor __atomic builtin functions are available. + + ASTERISK-27619 + + Change-Id: Ie01a281e0f5c41dfeeb5f250c1ccea8752f56ef9 + +2018-01-24 22:44 +0000 [a164b7ccfb] Corey Farrell + + * loader: Correct overly strict startup checks. + + The code which handled loading modules had too many situations which + would result in halting Asterisk startup. Treat most errors as declines + instead of failures. The exception is when the module load function + returns AST_MODULE_LOAD_FAILURE or an invalid code. + + Clear the missingdeps vector when appropriate to ensure the next loop + starts clean. + + ASTERISK-27620 + + Change-Id: I45547d9641fd45bd86d80250224417625631ad84 + +2018-01-24 18:49 +0000 [6fbd855228] Corey Farrell + + * Build System: Add support for __atomic built-in operators. + + Add a check to configure.ac for __atomic_fetch_add support. If found + use the __atomic built-in operators for ast_atomic_dec_and_test and + ast_atomic_fetchadd_int. + + ASTERISK~27619 + + Change-Id: I65b4feb02bae368904ed0fb03f585c05f50a690e + +2017-12-29 02:57 +0000 [527cf5a570] Corey Farrell + + * Remove redundant module checks and references. + + This removes references that are no longer needed due to automatic + references created by module dependencies. + + In addition this removes most calls to ast_module_check as they were + checking modules which are listed as dependencies. + + Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e + +2018-01-24 10:30 +0000 [b9e35bf6d3] Richard Mudgett + + * CHANGES: Add AMI action 'PJSIPShowContacts' note. + + ASTERISK-27581 + + Change-Id: If6af275764741a11030f0a4fd324fa29b376d74e + +2018-01-14 12:33 +0000 [5b8e71ab9f] Sungtae Kim + + * res_pjsip: Add AMI action 'PJSIPShowContacts' + + Add an AMI action which provides information on all + configured Contacts. + + ASTERISK-27581 + + Change-Id: I2eed42c74bbc725fad26b8b33b1a5b3161950c73 + +2018-01-18 20:19 +0000 [2f78dc2bfa] Richard Mudgett + + * pbx_variables.c: Misc fixes in variable substitution. + + * Copy more than one character at a time when there is nothing to + substitute. + + * Fix off by one error if a '}' or ']' is missing. + + * Eliminated the requirement that the "used" parameter had to point to a + variable. The current callers were always declaring a variable to meet + the requirement and discarding the value put into that variable. Now it + can be NULL. + + * In ast_str_substitute_variables_full() fixed using the bogus channel to + evaluate a function. We were not using the bogus channel we just created + to help evaluate a subexpression. + + Change-Id: Ia83d99f4f16abe47f329eb39b6ff2013ae7c9854 + +2018-01-18 09:01 +0000 [679fa5fb34] Corey Farrell + + * Add missing OPTIONAL_API and ARI dependences. + + I've audited all modules that include any header which includes + asterisk/optional_api.h. All modules which use OPTIONAL_API now declare + those dependencies in AST_MODULE_INFO using requires or optional_modules + as appropriate. + + In addition ARI dependency declarations have been reworked. Instead of + declaring additional required modules in res/ari/resource_*.c we now add + them to an optional array "requiresModules" in api-docs for each module. + This allows the AST_MODULE_INFO dependencies to include those missing + modules. + + Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606 + +2018-01-22 09:18 +0000 [140f937c7e] Alexander Traud + + * res_config_mysql: Avoid the header mysql_version.h. + + ASTERISK-27607 + + Change-Id: I23d00ded955c4afd5f2c3c9dc96dcb48b3f74eec + +2018-01-05 14:46 +0000 [fd557ad041] Alexander Traud + + * install_prereq: For PJProject, point users to configure script. + + The installation script and the new configure option --with-pjproject-bundled + aimed to accomplish the same. However, the installation script was out of + date. Users should go for the maintained configure option, or the Wiki. + + ASTERISK-24598 + + Change-Id: Icbf4b562f81f7c05bd24a3805bd46c0beb4ebd44 + +2018-01-20 12:58 +0000 [d427bb84a2] Alexander Traud + + * BuildSystem: Remove AC_CONFIG_AUX_DIR. + + ASTERISK-27602 + + Change-Id: I9f4d3d2bc1481748e39ad1e2b0a364d38e38978b + +2018-01-19 12:21 +0000 [693e509566] Alexander Traud + + * BuildSystem: Remove orphaned .PHONY targets. + + Change-Id: Ic44d75141b9bf99e7d72fcc82ee111b5cf6989d2 + +2018-01-19 12:14 +0000 [70137794e9] Alexander Traud + + * BuildSystem: Allow make clean all again. + + ASTERISK-27600 + Reported by: Hamid R. Hashmi + + Change-Id: I683d14d024650be04074b037b6300464519409f4 + +2018-01-19 06:16 +0000 [93471373f6] Alexander Traud + + * install_prereq: Update Debian/Ubuntu libraries. + + ASTERISK-27555 + + Change-Id: Ieb41b0cbf968af12882b39454b819ebb48b9ea46 + +2018-01-19 04:46 +0000 [4c511c1a4d] Alexander Traud + + * install_prereq: Support package manager DNF and yum option strict=1. + + This re-enables the script ./contrib/scripts/install_prereq on Fedora 22 and + newer, and on RHEL/CentOS when the option strict=1 was set for yum install. + + ASTERISK-27598 + Reported by: Hunter Stevens, Said Masoud + + Change-Id: I40f9517122aaa6906e8fc0962b4b8008dfddb368 + +2018-01-09 11:29 +0000 [77f2814d01] Benoît Dereck-Tricot + + * pbx: Reduce verbosity while loading extensions + + Each time the dial plan is reloaded, a lot of logs like these are generated: + "Added extension 'XXXXX' priority 1 to YYYYYYYYYYY" + This patch changes the log level for those logs. + + ASTERISK-27084 + + Change-Id: I5662902161c50890997ddc56835d4cafb456c529 + +2018-01-18 14:55 +0000 [5964061a21] Sean Bright + + * res_pjsip: Document tlsv1_1 and tlsv1_2 methods + + Change-Id: I67ed9039bf3f132fb20ee7a750e0aef0f704d7d3 + +2018-01-08 23:50 +0000 [33d5ab3e69] Igor Goncharovsky + + * chan_unistim: Fix hold function ability to lock/crash asterisk + + This patch fix chan_unistim hold functions to correctly support + hold function in different states possible in case of multiple lines + established on the phone + + ASTERISK-26596 #close + + Change-Id: Ib1e04e482e7c8939607a42d7fddacc07e26e14d4 + +2017-10-29 22:00 +0000 [25cb1ab05b] Corey Farrell + + * loader: Add support for built-in modules. + + * Add SRC_EMBEDDED variable to main/Makefile. Built-in module sources + must be listed in this variable to ensure they get the correct CFLAGS. + + Change-Id: I920852bc17513a9c2627061a4ad40511e3a20499 + +2017-12-09 00:03 +0000 [e6142a1282] Corey Farrell + + * loader: Rework load_resource_list. + + Use a single loop in a loop to scan the resource list attempting to + dlopen each module. The inner loop is repeated until it doesn't do any + work, then it is run one more time to allow printing of error messages. + + Change-Id: I60c15cd57ff9680b62e2a94c7519401fa4a38e45 + +2017-12-08 23:30 +0000 [a80cbb046e] Corey Farrell + + * loader: Remove global symbol only startup phase. + + Dependency loader is now in place so we no longer need a separate loader + phase for global symbols only. This simplifies the loader and allows us + to minimize calls to dlopen. + + Change-Id: I33e3174d67f3b4552d3d536326dcaf0ebabb097d + +2017-11-21 23:39 +0000 [3b73ed28c5] Corey Farrell + + * loader: Process module dependencies. + + * Add string vectors for requires, optional_apis and enhances. + * Add reffed_deps module vector for holding references to dependencies. + * Initialize string vectors after final dlopen of each module. + * Free string vectors and clear references from reffed_deps in + module_destroy. + * Create functions necessary to process module dependencies and enforce + load order. + + Module dependencies result in automatic references being managed by the + module loader. This enforces unload order. + + Change-Id: I9be08d1dd331aceadc1dcba00b804d71360b2fbb + +2017-12-27 17:44 +0000 [86b484dec7] Graham Mainwaring + + * app_followme: Add a prompt to be read when a call is connected + + This patch adds the ability to configure a prompt which will be read + to the "winner" who pressed 1 (or the configured value) and received + the call. + + ASTERISK-24372 #close + + Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272 + +2018-01-17 00:28 +0000 [4fd303b630] Corey Farrell + + * loader: Miscellaneous fixes. + + * Remove comment about lazy load. + * Improve message about module already being loaded and running. + * Handle allocation error in add_to_load_order. + * Dead code elimination from modules_shutdown. + + Change-Id: I22261599c46d0f416e568910ec9502f45143197f + +2018-01-17 08:36 +0000 [2a1b52cc67] Alexander Traud + + * BuildSystem: Use the detected name for MD5 everywhere. + + Affacted the (automatic) download script for external modules: + ./build_tools/download_externals + + ASTERISK-27596 + + Change-Id: If4c3176f7bf58df32fec6e02a659f1a78d57cf4b + +2018-01-17 07:11 +0000 [4cd3f5c162] Alexander Traud + + * BuildSystem: Invoke install not in GNU but POSIX style. + + ASTERISK-27594 + + Change-Id: Iaaa6a19d2fe031dffcba441d0502a7ea65c93cb3 + +2018-01-17 06:47 +0000 [7e7a20642c] Alexander Traud + + * BuildSystem: In OpenBSD, xmlstarlet is xml. + + ASTERISK-27593 + + Change-Id: I1c7087f7f7582e40b3312c690d912c9a86466805 + +2018-01-17 02:51 +0000 [8f31b70246] Alexander Traud + + * BuildSystem: Detect external library Lua in version 5.3. + + On some platforms, you decide to go for one specific version of Lua, for + example in OpenBSD. On other platforms, you are able to install several versions + side-by-side, for example in Ubuntu and Fedora. Asterisk already works with + Lua 5.3. Asterisk failed to detect Lua 5.3 on those platforms which allow + several versions. + + ASTERISK-27592 + + Change-Id: If7a4b395d844a464e9a1f4f626c5bff4ee67eed8 + +2017-12-22 19:50 +0000 [8494e78010] Richard Mudgett + + * res_pjsip: Split type=identify to IP address and SIP header matching priorities + + The type=identify endpoint identification method can match by IP address + and by SIP header. However, the SIP header matching has limited + usefulness because you cannot specify the SIP header matching priority + relative to the IP address matching. All the matching happens at the same + priority and the order of evaluating the identify sections is + indeterminate. e.g., If you had two type=identify sections where one + matches by IP address for endpoint alice and the other matches by SIP + header for endpoint bob then you couldn't predict which endpoint is + matched when a request comes in that matches both. + + * Extract the SIP header matching criteria into its own "header" endpoint + identification method so the user can specify the relative priority of the + SIP header and the IP address matching criteria in the global + endpoint_identifier_order option. The "ip" endpoint identification method + now only matches by IP address. + + ASTERISK-27491 + + Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095 + +2018-01-16 08:32 +0000 [7ed7d525fb] Richard Mudgett + + * taskprocessor.c: Increase the number of tps_singletons container buckets. + + Since v12 the number of taskprocessors in the system has increased a lot. + Small systems can easily have over a hundred and larger systems can have + thousands. + + Most uses of the tps_singletons container deal with creating and + destroying the taskprocessors. However, the pjsip distributor looks up + taskprocessors/serializers by name frequently. It needs to find the + serializer for incoming SIP responses to distribute them to the + appropriate serializer. + + Change-Id: Ice0603606614ba49f7c0c316c524735c064e7e43 + +2018-01-16 08:20 +0000 [f0a3c977d6] George Joseph + + * pjproject_bundled: Prevent crash on bad outgoing header + + We still need to figure out how a bad header is getting into the + outgoing message but this patch to pjproject prevents attempting + to print that header and causing a crash. + + For several users, this crash happens when sending 183 progress + messages. + + ASTERISK-26832 + Reported by: Ross Beer, Jan Rozhon + + Change-Id: Ie5c5a921c890c843587763e7f33f987dfe66bd16 + +2018-01-16 06:34 +0000 [a046305fae] Alexander Traud + + * BuildSystem: Avoid $EUID and use id -u instead. + + Makefile included a call to ${EUID} which requires the shell bash. To keep + compatibility with other shells like dash or ksh, use id -u instead. + + ASTERISK-27589 + + Change-Id: Ia6e74f5bc9aab4e6dc62b7439f647b7964e6f657 + +2018-01-15 18:03 +0000 [6fbe315f77] Richard Mudgett + + * cel_odbc.c: Fix menuslect module description display. + + Asterisk's makefile for menuselect has a very simple source file parsing + script that looks for AST_MODULE_INFO lines to extract the quoted string + as a module description. If it does not find a quoted string it uses the + whole line as the description. + + Change-Id: I80f13a63818e4e28d683639a94a4dfaea405c1d5 + +2017-11-19 16:30 +0000 [9cfdb81e91] Corey Farrell + + * loader: Add dependency fields to module structures. + + * Declare 'requires' and 'enhances' text fields on module info structure. + * Rename 'nonoptreq' to 'optional_modules'. + * Update doxygen comments. + + Still need to investigate dependencies among modules I cannot compile. + + Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf + +2017-11-19 20:10 +0000 [35ae99c712] Corey Farrell + + * vector: Additional string vector definitions. + + ast_vector_string_split: + This function will add items to an ast_vector_string by splitting values + of a string buffer. Items are appended to the vector in the order they + are found. + + ast_vector_const_string: + A vector of 'const char *'. + + Change-Id: I1bf02a1efeb2baeea11c59c557d39dd1197494d7 + +2018-01-15 10:57 +0000 [645297614e] Alexander Traud + + * BuildSystem: Resolve resolv.h not via Generic but Particular Header-Check. + + ASTERISK-27585 + + Change-Id: I27c67563788e6f67eeda5fb51a741823a50a95e2 + +2018-01-13 13:49 +0000 [cabe80631b] George Joseph + + * config_transport: Enable TCP_NODELAY on TLS transports + + We did this for TCP transports already but I'm not sure why we + didn't do it for TLS transports. + + ASTERISK_27474 #not_final_fix + + Change-Id: I5b1ef4b882f7b859e718236686b7898751dbb262 + +2018-01-12 18:37 +0000 [de7f2a6cb4] Corey Farrell + + * res_stasis_recording: Allow symbolic links in configured recordings dir. + + If any component of ast_config_AST_RECORDING_DIR is a symbolic link we + would incorrectly assume the ARI user was trying to escape the recording + path. Create additional check to check the recording directory's + realpath, only deny access if both do not match. + + This is needed by the testsuite when run by 'run-local'. + + Change-Id: I9145e841865edadcb5f75cead3471ad06bbb56c0 + +2018-01-12 12:00 +0000 [99535b0497] Corey Farrell + + * menuselect: Remove unused dev-mode option TRACE_FRAMES. + + ASTERISK-27575 #close + + Change-Id: Ica3a522892afed7a96816a5ecf140e1671f46ad4 + +2018-01-12 03:50 +0000 [eb9b85baec] Alexander Traud + + * res_config_pgsql: Avoid typecasting an int to unsigned char. + + clang 5.0 warned about this. + + ASTERISK-27576 + + Change-Id: If41f400a51973c06cdb9b75462e535b616bfe385 + +2018-01-12 03:17 +0000 [cff3add680] Alexander Traud + + * BuildSystem: Really do not pass unknown-warning options to the compiler. + + When an older GCC version is called with a too new warning option, GCC exited + with an error and Asterisk was not built. Therefore, the configure script tests + the installed compiler whether it supports that warning option. If not, Asterisk + does not pass it to the installed compiler. However, some compilers (like clang) + do not exit (error) but give just a warning in such a case. Because the compiler + did not exit, Asterisk passed the unknown-warning option. + + ASTERISK-27560 + + Change-Id: Ia9d148e689c173df4e91699113605dab2de36038 + +2018-01-12 04:27 +0000 [685bab254c] Alexander Traud + + * app_osplookup.c: Avoid two format truncations. + + GCC 7 warned about this. + + ASTERISK-27578 + + Change-Id: I4a00458dbe9b575ef04338b6a7852272745e1552 + +2018-01-12 04:03 +0000 [797747afa7] Alexander Traud + + * chan_ooh323: Avoid typecasting an int to unsigned short. + + clang 5.0 warned about this. + + ASTERISK-27577 + + Change-Id: I898fe4255023138a9e8b579fe4482fcf582f2b78 + +2018-01-05 15:13 +0000 [b9e2b72de6] Alexander Traud + + * install_prereq: Update Debian/Ubuntu libraries. + + ASTERISK-27555 + + Change-Id: I0818b6e42631be1b69237e2b41d3415275693e53 + +2018-01-11 12:05 +0000 [6d5f4768a4] Joshua Colp + + * chan_sip: Check that an iostream exists before accessing. + + Before getting the file descriptor for an iostream check + that it is present. + + ASTERISK-27534 + + Change-Id: Ie0aa1394007a37c30e337ea1176a6fb3a63bc99c + +2018-01-11 08:09 +0000 [30b5ec023f] Tzafrir Cohen + + * Ignore quilt .pc directory, used in deb packaging + + Debian packaging uses quilt to manage patches. Book-keeping for them is + done using quilt (either directly, or in a compatible format), and + tracked in the directory .pc . + + Change-Id: I22c90f3d7ab8918e6216e7b686de6fa0e1fdaa7b + Signed-off-by: Tzafrir Cohen + +2018-01-09 11:23 +0000 [f0eb00d1e7] Corey Farrell + + * stasis: Remove silly usage of RAII_VAR. + + Change-Id: Ib11193531e797bcb16bba560a408eab155f706d1 + +2018-01-09 11:09 +0000 [a383e1ddb1] Corey Farrell + + * stasis_cache_pattern: Remove silly usage of RAII_VAR. + + Change-Id: Ic98a51f555062cd863b6db3f8d76065943a9dea3 + +2018-01-09 16:23 +0000 [9e2fcb82ed] Sean Bright + + * cdr_syslog: Deprecate unmaintained module + + There has been an open issue against cdr_syslog (ASTERISK~14441) about + a race condition for 7.5 years that has never been addressed. Because + this module is effectively unmaintained and currently broken, there is + no sense in keeping it around. + + If logging CDRs to syslog is a desirable feature, it would probably be + better to write the logs directly to the syslog server via socket + instead of using the facilities provided by openlog/syslog/closelog. + Doing so would address the race condition referenced in the associated + issue. + + Change-Id: Ic77b94cd97f355a9cf5b1d3f3444964a6e0ba5dc + +2018-01-09 11:16 +0000 [0de004dd85] Corey Farrell + + * stasis_bridges: Remove silly usage of RAII_VAR. + + Change-Id: I0fa7ab05454f183dc4ff10e26d18776d2b0fcf1f + +2018-01-09 11:10 +0000 [01127e1664] Corey Farrell + + * stasis_cache: Remove silly usage of RAII_VAR. + + Change-Id: Ifa95e5801c949df296c7e4376347730fb0ed52ef + +2018-01-09 10:57 +0000 [175a9ef873] Corey Farrell + + * stasis_endpoints: Remove silly usage of RAII_VAR. + + Change-Id: Ic099dc552f36c353c89783a4bcfd09f010432733 + +2018-01-09 10:55 +0000 [4b655184b0] Corey Farrell + + * stasis_message_router: Remove silly usage of RAII_VAR. + + Change-Id: I50d6ae230920e0b878ed9cc8f79eef746e06701d + +2018-01-09 10:53 +0000 [3074c4165c] Corey Farrell + + * stasis_system: Remove silly usage of RAII_VAR. + + Change-Id: Iedbe5656cee68cd3a96a953558764aa02d4a0c3b + +2018-01-03 17:26 +0000 [8f3167c5f1] Richard Mudgett + + * res_pjsip.c: Update the endpoint identification documentation. + + * Endpoint identify_by documentation. + * IP/Header endpoint identifier documentation. + + Change-Id: Id92f00b495acca7be945daf749d2abd7f76a0b5a + +2018-01-03 15:20 +0000 [42a61d9db6] Richard Mudgett + + * res_pjsip_endpoint_identifier_ip.c: Remove unnecessary requirement. + + The requirement that "ip" must be in the endpoint identify_by list to + allow the type=identify method to identify the endpoint is not necessary. + The "ip" identifier method can match one and only one endpoint. To even + work, the "ip" identifier method configuration must explicitly specify the + identified endpoint. Therefore, why bother configuring the type=identify + identifier in the first place? The requirement only adds the potential + for configuration errors for no benefit. Even worse, those configuration + errors cannot be detected when the configuration loads. The requirement + was introduced with the ASTERISK_27206 patch. + + * Remove the code change that enforces the requiremnt. Listing the "ip" + method in the identify_by value is simply documentation. + + Change-Id: Ia057f92a33fb5d9f51dc5d5692e3d5ee1a6f2c11 + +2018-01-05 19:03 +0000 [a7bbb18e5c] Richard Mudgett + + * res_pjsip.c: Fix ident_to_str() and refactor ident_handler(). + + * Extracted sip_endpoint_identifier_type2str() and + sip_endpoint_identifier_str2type() to simplify the calling functions. + + * Fixed pjsip_configuration.c:ident_to_str() building the endpoint's + identify_by value string. + + Change-Id: Ide876768a8d5d828b12052e2a75008b0563fc509 + +2018-01-04 17:04 +0000 [be488eb14a] Richard Mudgett + + * res_pjsip_endpoint_identifier_ip.c: Allow multiple IdentifyDetail AMI events. + + The AMI PJSIPShowEndpoint action could only list one IdentifyDetail AMI + event per endpoint. However, there is no reason that multiple + type=identify sections cannot identify the same endpoint. + + * Reworked format_ami_endpoint_identify() to generate as many + IdentifyDetail AMI events as there are matching identifiers. + + Change-Id: Ie146792aef72d78e05416ab5b27bc552a30399db + +2018-01-05 05:51 +0000 [3a7d917256] Alexander Traud + + * translate: Avoid absolute value on unsigned substraction. + + ast_format_get_sample_rate(.) returns an unsigned type. The difference of a + substraction between two unsigned types does not get implicitly converted to a + signed type. Therefore, using abs(.) did not make sense. + + ASTERISK-27549 + + Change-Id: Ib904d9ee0d46b6fdd1476fbc464fbbf813304017 + +2018-01-09 08:22 +0000 [25022de875] Sean Bright + + * Revert "codec_opus: Make libcurl a dependency in menuselect" + + This reverts commit 028f4320de60a204e457ad606ab0a3318493b431. + + Change-Id: Ieb91f825cb55202a937f5361c01d356e7662b70c + +2018-01-08 10:54 +0000 [a21841bf40] Joshua Colp + + * res_pjsip_session: Always bundle streams if WebRTC is enabled. + + Some WebRTC clients can't handle renegotiation with the addition of + streams that include an offer to bundle. They instead expect the + newly added streams to already be bundled. This change does such a thing + if WebRTC support is enabled on an endpoint. + + ASTERISK-27566 + + Change-Id: I7fe9b7ac35a2798627d9c2c8369129f407af6461 + +2018-01-08 20:25 +0000 [d46cbe788a] Corey Farrell + + * bridge_softmix: Fix sfu_append_source_streams test. + + * validate_stream: zero result from ast_format_cap_identical indicates + they are not identical, rather than non-zero indicating an error. + * validate_original_streams: use num_streams instead of + ARRAY_LEN(params). + * Fix declaration of alice_dest_stream and bob_dest_stream. + + Change-Id: I6b1dd8bed10439d3c7406f033eb1896b6c419147 + +2018-01-08 18:47 +0000 [5380fb9978] Corey Farrell + + * app_confbridge: Fix NULL check in action_kick_last. + + The check for last_user == NULL needs to happen before we dereference + the variable, previously it was possible for us to check flags of a NULL + last_user. + + Change-Id: I274f737aa8af9d2d53e4a78cdd7ad57561003945 + +2018-01-06 02:17 +0000 [55a540272f] Corey Farrell + + * res_stasis: Reduce RAII_VAR usage. + + In addition to being a micro-optimization (RAII_VAR has overhead), this + change improves output of REF_DEBUG. Unfortunately when RAII_VAR calls + ao2_cleanup it does so from a generated _dtor_varname function. For + example this caused _dtor_app to release a reference instead of + __stasis_app_unregister. + + Change-Id: I4ce67120583a446babf9adeec678b71d37fcd9e5 + +2018-01-04 18:47 +0000 [faeb9e1b26] Sungtae Kim + + * res_pjsip: Add AMI action 'PJSIPShowAuths' + + Add an AMI action which provides information on all + configured Auths. + + ASTERISK-27547 + + Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817 + +2018-01-07 21:38 +0000 [8b3083cac5] Corey Farrell + + * res_stasis: Fix dial bridge unload. + + If the dial bridge has been created it must be released by calling + ast_bridge_destroy, simply releasing the ao2 reference is not enough. + + Also move stasis_app_control_shutdown earlier in unload to ensure the + bridge cannot be created or grabbed after the app_bridges container is + released. + + Change-Id: I372302de94ca63876069e2585a049c5060e5e767 + +2018-01-07 20:21 +0000 [6870ba5f26] Corey Farrell + + * res_stasis: Fix app_is_subscribed_bridge_id. + + Instead of searching for bridge_id provided in an argument this function + always searched for BRIDGE_ALL first. Rewrite this function to work + like the similar functions for channel and endpoint functions. + + Change-Id: Ib5caca69e11727c5c8a7284a1d00621f40f1e60a + +2018-01-05 07:44 +0000 [7e9781c25e] Alexander Traud + + * General: Silence modules on (un)load. + + Some (normally optional) modules created notices, warnings, and even errors + in normal situations like (un)load. This cluttered the command-line interface + (CLI) on start and while stopping gracefully. However, when an user went for + the script './contrib/scripts/install_prereq', those modules get compiled-in + because their prerequisites were met at compile time. Furthermore, because of + ASTERISK_27475, the former talkative module 'res_curl' is built as side-effect. + + ASTERISK-27553 + + Change-Id: I9f105f46d72553994e820679bfde3478a551b281 + +2018-01-06 15:40 +0000 [512286e3c8] Alexander Traud + + * BuildSystem: Really do not pass unknown-warning options to the compiler. + + When an older GCC version is called with a too new warning option, GCC exited + with an error and Asterisk was not built. Therefore, the configure script tests + the installed compiler whether it supports that warning option. If not, Asterisk + does not pass it to the installed compiler. However, some compilers (like clang) + do not exit (error) but give just a warning in such a case. Because the compiler + did not exit, Asterisk passed the unknown-warning option. + + ASTERISK-27560 + + Change-Id: Ia9b7747f649b27ff5e9f75c3db3fee4fe7a29621 + +2018-01-06 01:25 +0000 [f84fcc1fc1] Alexander Traud + + * General: Avoid implicit conversion to char when changes value to negative. + + clang 5.0 warned about this. + + ASTERISK-27557 + + Change-Id: I7cceaa88e147cbdf81a3a7beec5c1c20210fa41e + +2018-01-05 06:06 +0000 [b12c8cffad] Alexander Traud + + * bridge_softmix: Removed unused parameter from check_binaural_position_change(.). + + Found as a result of the function being passed an uninitalized variable by + clang. + + ASTERISK-27550 + + Change-Id: I8af3bd84656b685a956d498459f8db3613f68954 + +2018-01-06 06:45 +0000 [ad3252ccef] Alexander Traud + + * editline: Avoid comparison between pointer and zero character constant. + + gcc 7.2 warned about this. + + ASTERISK-27559 + + Change-Id: I48960dda9cf0a11b6a9426f775e632363f8caa74 + +2018-01-06 05:01 +0000 [ef68df9111] Alexander Traud + + * codec_gsm: Avoid shifting a negative signed value. + + clang 5.0 warned about this. + + ASTERISK-27558 + + Change-Id: Icc452ecb0d86bbeba78dae768cc472ec540699df + +2018-01-04 12:23 +0000 [b20b5758d9] Richard Mudgett + + * res_pjsip_endpoint_identifier_ip.c: Fix apply identify validation. + + The ip_identify_apply() did not validate the configuration for simple + static configuration errors or deal well with address resolution errors. + + * Added missing configuration validation checks. + * Fixed address resolution error handling. + * Demoted an error message to a warning since it does not fail applying + the identify object configuration. + + Change-Id: I8b519607263fe88e8ce964f526a45359fd362b6e + +2018-01-04 17:42 +0000 [705e6c04b3] Richard Mudgett + + * res_pjsip.c: Fix endpoint identifier registration name search. + + If an endpoint identifier name in the endpoint_identifier_order list is a + prefix to the identifier we are registering, we could install it in the + wrong position of the list. + + Assuming + endpoint_identifier_order=username,ip,anonymous + + then registering the "ip_only" identifier would put the identifier in the + wrong position of the priority list. + + * Fix incorrect strncmp() string prefix matching. + + Change-Id: Ib8819ec4b811da8a27419fd93528c54d34f01484 + +2018-01-05 03:33 +0000 [af064eaf13] Alexander Traud + + * BuildSystem: Find ptlib-config on Debian/Ubuntu. + + The current configure script requires that tool when libpt-dev is installed. + libpt-dev was installed by libopenh323-dev, bacause you wanted to go for H.323 + based channel drivers. + + ASTERISK-25329 + + Change-Id: I9c6ab78b7246c21536e1d252dcbffe682f63f83d + +2018-01-05 06:42 +0000 [f0c8f04c73] Alexander Traud + + * chan_ooh323: Limit outgoinglimit to positive values as intended. + + ASTERISK-27552 + + Change-Id: Ifbf9d51e7374ca2e8b27ec568f6770050fc1a854 + +2018-01-05 06:19 +0000 [09f339bda5] Alexander Traud + + * ooh323cDriver: Fix typo in header guard. + + ASTERISK-27551 + + Change-Id: I39ff66031e3373e895e2bc47b23a5e860ea4e012 + +2018-01-05 03:36 +0000 [bc1b4f4d43] Alexander Traud + + * BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf. + + ASTERISK-26046 + + Change-Id: I48f05698c235f709225b92bec5aa260fb57d69d1 + +2018-01-04 15:37 +0000 [cfb88f3ac1] Corey Farrell + + * pbx: Prevent execution of NULL pointer. + + pbx_extension_helper has a check for q->swo.exec == NULL but it doesn't + actually return so we would still run the function. Fix the return. + Move the 'int res' variable into the only scope which uses it. + + Also fix a copy-paste error in ast_pbx_init which could result in a + crash on allocation failure (we exit with a normal error instead). + + Change-Id: I0693af921fdc7f56b6a72a21fb816ed08b960a69 + +2018-01-04 10:50 +0000 [82cf585fb5] Corey Farrell + + * translators: Don't use ast_module_running_ref. + + Translators are run during module load before the module is actually + running, so it cannot use ast_module_running_ref. + + ASTERISK-20346 + + Change-Id: Iaa0e75da99c696e38000f1a41e340abbd7a88f56 + +2018-01-04 09:39 +0000 [da365affbd] Corey Farrell + + * rtp_engine: Add missing unlock. + + Change-Id: I380c31a255e060309f4916da11176e0d00813215 + +2018-01-04 09:30 +0000 [73bf5035b8] Corey Farrell + + * res_pjsip_history: Add missing unlock to CLI command. + + Change-Id: I872060a30543776a176a316309602d924a23eb29 + +2018-01-04 09:27 +0000 [aaed0b8b3a] Corey Farrell + + * aco: Fix NULL dereference in error path. + + Change-Id: Id505167cf0f9414a3c144fa2c1e181a2cf288694 + +2018-01-03 19:07 +0000 [e3c9314a2e] Corey Farrell + + * func_odbc: Add missing unlock's to acf_odbc_read. + + Change-Id: I828329ecbd252ae8f27a369a046d2b03102b07c6 + +2017-12-29 18:24 +0000 [55f1d69c43] Corey Farrell + + * loader: Create ast_module_running_ref. + + This function returns NULL if the module in question is not running. I + did not change ast_module_ref as most callers do not check the result + and they always call ast_module_unref. + + Make use of this function when running registered items from: + * app_stack API's + * bridge technologies + * CLI commands + * File formats + * Manager Actions + * RTP engines + * Sorcery Wizards + * Timing Interfaces + * Translators + * AGI Commands + * Fax Technologies + + ASTERISK-20346 #close + + Change-Id: Ia16fd28e188b2fc0b9d18b8a5d9cacc31df73fcc + +2018-01-03 10:41 +0000 [62f862e2cd] Kevin Harwell + + * res_pjsip_session: Check if sequence header is missing + + The pjsip_msg_find_hdr function can return NULL. This patch adds a check + when searching for the sequence header to make sure a NULL pointer is never + de-referenced. + + Change-Id: I19af23aeeded65be016be92360e8cb7ffe51fad2 + +2018-01-02 07:36 +0000 [9b5d1454b4] Tzafrir Cohen + + * cdr: submit: fix logic of test for batch mode + + ASTERISK-27539 #close + + Change-Id: I33cdf329d2bb4486dcae975c450f6aae94c515f7 + +2017-12-29 23:14 +0000 [ffbf5be116] Sungtae Kim + + * res_pjsip: Add AMI action 'PJSIPShowAors' + + Add an AMI action which provides information on all + configured AORs. + + ASTERISK-27537 + + Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb + +2018-01-02 00:26 +0000 [f298178583] Corey Farrell + + * aco: Add missing aco_option_type_string for OPT_TIMELEN_T. + + ASTERISK-27117 + + Change-Id: I8f6c34bb30830be9f7a40823723eb4dcaaa91c61 + +2017-12-31 10:26 +0000 [15f8b9b8bf] Sean Bright + + * ice: Increase foundation buffer size + + Per RFC 5245, the foundation specified with an ICE candidate can be up + to 32 characters but we are only allowing for 31. + + ASTERISK-27498 #close + Reported by: Michele Prà + + Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf + +2017-12-29 22:03 +0000 [b32d6d5e2d] Corey Farrell + + * astobj2: Create case-insensitive variants of container function macros. + + * AO2_STRING_FIELD_CASE_HASH_FN + * AO2_STRING_FIELD_CASE_CMP_FN + * AO2_STRING_FIELD_CASE_SORT_FN + + Change-Id: I11af8c6a0c43380a42732553f519c667abb842cf + +2017-12-29 22:59 +0000 [bc73337e07] Corey Farrell + + * core: Use macros to generate ao2_container callbacks where possible. + + This uses AO2_STRING_FIELD_HASH_FN and AO2_STRING_FIELD_CMP_FN where + possible in the Asterisk core. + + This removes CMP_STOP from the result of CMP_FN callbacks for the + following structure types: + * ast_bucket_metadata + * ast_bucket_scheme + * generic_monitor_instance_list (ccss.c) + * ast_bucket_file (media_cache.c) + * named_acl + + Change-Id: Ide4c1449a894bce70dea1fef664dade9b57578f1 + +2017-12-29 14:50 +0000 [0fe7df641a] Corey Farrell + + * datastore: Add automatic module references. + + Add a reference to the calling module when it is active to protect + access to datastore->info. Remove module references done by + func_periodic_hook as the datastore now handles it. + + ASTERISK-25128 #close + + Change-Id: I8357a3711e77591d0d1dd8ab4211a7eedd782c89 + +2017-12-28 13:27 +0000 [2dde5bef47] Richard Mudgett + + * stasis_channels.c: Misc cleanup. + + * Use current OBJ_SEARCH_xxx defines instead of the deprecated versions. + + * Fix hash_cb and cmp_cb container functions to correctly use the + OBJ_SEARCH_xxx values. + + * Remove incorrect usage of CMP_STOP. Most uses in the system have no + effect. This allows the collapse of channel_role_single_cmp_cb() and + channel_role_multi_cmp_cb() into channel_role_cmp_cb(). + + * Remove unnecessary usage of RAII_VAR(). + + Change-Id: I02c405518cab22aa2a082b61e2353bf7cd629a70 + +2017-12-13 15:43 +0000 [898b3b080a] Sean Bright + + * cdr_mysql: Make sure connection charset is always set + + When the MYSQL_OPT_RECONNECT option is enabled, the MySQL client API + will transparently reconnect when it needs to. Ideally this simplifies + our code, but when this reconnection occurs all connection state is + lost. Because we are not notified that this has happened, we don't know + to set our character set again (with "SET NAMES 'xyz'"). + + Rather than calling SET NAMES, we instead set the MYSQL_SET_CHARSET_NAME + option which will do it for us under the hood on each connect. This + option has been present in the MySQL C API for at least 15 years, so it + should be safe for most installations. + + I also snuck a few other changes into this patch: + + * Default the MySQL port to MYSQL_PORT (3306) instead of 0 if it's not + defined. + + * Fix some erroneous and/or silly checks on the contents of the + configuration ast_str values. + + ASTERISK-27366 #close + Reported by: Halil İbrahim YILDIZ + + Change-Id: I36bf8dc5d5f83584e803b3b1a151dea9396ab8f5 + +2017-12-27 20:48 +0000 [d69b7c6c6d] Richard Mudgett + + * manager.c: Update AMI Status event documentation + + The AMI Status event had linkedid listed twice and was missing the + effective connected line name and number headers. + + NOTE: The linkedid and other standard channel snapshot fields in the XML + documentation are part of the XML template defined in + doc/appdocsxml.xslt. + + Change-Id: I004c4c4f9e7b40ef55035c831702721bec82496c + +2017-12-27 22:36 +0000 [fa36f9c01b] Richard Mudgett + + * bridge_native_rtp.c: Fix reentrancy framehook crash. + + If two channels enter different native rtp bridges at the same time it is + possible that the framehook interface data pointer can be corrupted + because the struct variable was declared static. + + * Fixed the reentrancy corruption by changing the framehook interface + struct static variable to a stack local variable. + + * Moved the hook.data assignment outside of the channel lock. It did not + need the lock's protection. It probably was giving a false sense of + security. + + The testsuite + channels/pjsip/basic_calls/two_parties/nominal/alice_initiated/bob_hangs_up + test caught this with MALLOC_DEBUG and DO_CRASH enabled. + + Change-Id: If9e35b97d19209b0f984941c1d8eb5f7c55eea91 + +2017-12-27 20:22 +0000 [1d3dc9aea2] Richard Mudgett + + * func_channel.c: Update MASTER_CHANNEL documentation + + Be more explicit in what is meant by the master channel to eliminate + misunderstanding. + + ASTERISK-23133 + + Change-Id: I453bcaf4b99404a5a3e345dbf093ac6c1afcfc72 + +2017-12-27 19:27 +0000 [6338a03ce9] Corey Farrell + + * menuselect: Fix check for running configure. + + menuselect/Makefile checks that autoconfig.h and makeopts were newer + than the '.in' files. Unfortunately running ./configure does not touch + autoconfig.h unless the contents will change. + + Instead of looking at autoconfig.h we just need to ensure that makeopts + is newer than configure. + + Also make change to configure.ac so bootstrap.sh doesn't re-add the + extra trailing line-feed. + + Change-Id: Ief1f831d6717007f9cebb668c14e92782cd2b794 + +2017-12-21 23:56 +0000 [94eb12ca56] Corey Farrell + + * cdr: Missing NULL check and unlock. + + * handle_dial_message: Missing a check for NULL peer. + * cdr_generic_register: Missing unlock on allocation failure. + + cdr_generic_register is fixed by reordering so the new structure is + allocated and initialized before locking the list. + + Change-Id: I5799b99270d1a7a716a555c31ac85f4b00ce8686 + +2017-12-23 22:51 +0000 [23aa20bf20] Corey Farrell + + * loader: Add volatile to resource_being_loaded. + + Some compiler optimizers seem to assume that dlopen will not use + __attribute__((constructor)) functions to call back to the program. + This was causing resource_being_loaded to be optimized away completely. + + ASTERISK-27531 #close + Tested By: abelbeck + + Change-Id: If17a3b889e06811a0e7119f0539d052494d6ece9 + +2017-12-20 16:17 +0000 [553306548c] Kevin Harwell + + * AST-2017-014: res_pjsip - Missing contact header can cause crash + + Those SIP messages that create dialogs require a contact header to be present. + If the contact header was missing from the message it could cause Asterisk to + crash. + + This patch checks to make sure SIP messages that create a dialog contain the + contact header. If the message does not and it is required Asterisk now returns + a "400 Missing Contact header" response. Also added NULL checks when retrieving + the contact header that were missing as a "just in case". + + ASTERISK-27480 #close + + Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe + +2017-12-22 14:00 +0000 [c2529a352c] Corey Farrell + + * astobj.h: Remove from Asterisk core. + + This is the old ASTOBJ macro's which are no longer used except by the + deprecated netsock.c. Move it to the chan_iax2 include folder so it + does not get used elsewhere. + + Change-Id: I7e4ae96678b36b9f41d3cae14b167f110eb5d349 + +2017-12-22 08:23 +0000 [fd0ca1c3f9] Sean Bright + + * Remove as much trailing whitespace as possible. + + Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0 + +2017-12-21 09:51 +0000 [a1a179c09d] Sean Bright + + * Fix some invalid Unicode characters + + configs/samples/minivm.conf.sample contains invalid UTF-8, but that + appears to be intentional. + + Change-Id: I7b1e0d332f3380fd0425962a3c9c55f9b200c8cc + +2017-12-20 21:11 +0000 [f2f51ff4ea] Corey Farrell + + * app_voicemail: Fix file copy error handling. + + Fix error where input/output file descriptors would be closed multiple + times. + + Change-Id: Iba5140b60cb7de79e3d5d92be3c256947aa99da9 + +2017-12-20 14:54 +0000 [9415ec2877] Sean Bright + + * docs: Remove old API changes documentation + + Change-Id: I1bc7957121cc7ae27dca04acc3613f4e1858476a + +2017-12-20 11:14 +0000 [1b80ffa495] Corey Farrell + + * Fix Common Typo's. + + Fix instances of: + * Retreive + * Recieve + * other then + * different then + * Repeated words ("the the", "an an", "and and", etc). + * othterwise, teh + + ASTERISK-24198 #close + + Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31 + +2017-12-20 11:30 +0000 [3625e91586] Richard Mudgett + + * manager.h: Bump AMI version + + Change-Id: I62e6ddeb261ef012687e1fb6734c554e2499b6bf + +2017-12-20 10:23 +0000 [aaa3884d4a] Corey Farrell + + * bridge: Old channel video source not set to NULL after unref. + + The bridge holds onto the old channel video source after it's been + released. This can lead to use after free errors. + + ASTERISK-27229 #close + + Change-Id: Ib2dab61677dd8a21f7ad53cdc9b8ca93297838b3 + +2017-12-20 10:13 +0000 [c2850bfebc] Corey Farrell + + * core: Fix unused variable error in handle_show_sysinfo. + + Apparently in OSX it's possible for OSX to HAVE_SYSCTL but not + HAVE_SYSINFO or HAVE_SWAPCTL. In this case freeswap caused an unused + variable error. + + ASTERISK-26563 #close + + Change-Id: I8ec5b1897b786cc1abaf62264aa75039eea05510 + +2017-12-20 00:53 +0000 [fff7782cf5] Corey Farrell + + * app_festival: Fix fd leak on connection failure. + + Change-Id: If5efaddcf735ff7d17e55c36cc1388946cee9e0f + +2017-12-18 20:12 +0000 [d51837a1b9] Corey Farrell + + * CLI: Address multiple issues. + + * listen uses the variable `s` for the result from ast_poll() then + overwrites it with the result of accept(). Create a separate variable + poll_result to avoid confusion since ast_poll does not return a file + descriptor. + * Resolve fd leak that would occur if setsockopt failed in listen. + * Reserve an extra byte while processing completion results from remote + daemon. This fixes a bug where completion processing used strstr() on + a string that was not '\0' terminated. This was no risk to the Asterisk + daemon, the bug was only reachable the remote console process. + * Resolve leak in handle_showchan when the channel is not found. + * Multiple leaks and a deadlock in pbx_config CLI completion. + * Fix leaks in "manager show command". + + Change-Id: I8f633ceb1714867ae30ef4e421858f77c14485a9 + +2017-12-18 22:48 +0000 [b8f54f742f] Corey Farrell + + * dns_core: Protect against array index violation. + + Add a check to allocate_dns_record to prevent calling a pointer + retrieved from beyond dns_alloc_table. + + ASTERISK-27495 #close + + Change-Id: Ie2f6e4991cea46baa12e837bd64cc22b44d322bb + +2017-12-18 18:59 +0000 [3c037ef972] Corey Farrell + + * chan_sip: Fix memory leaks. + + In change_redirecting_information variables we use ast_strlen_zero to + see if a value should be saved. In the case where the value is not NULL + but is a zero length string we leaked. + + handle_response_subscribe leaked a reference to the ccss monitor + instance. + + Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f + +2017-12-16 07:51 +0000 [3b99a0332c] Ivan Poddubny + + * bridge: Stop music on hold on adding an arbitrary channel to a bridge + + When a channel that is on hold gets added to a bridge by + the Bridge AMI action or the dialplan application of the same name, + music continues to play, causing "robotic sound". + + This commit adds a call to ast_moh_stop to stop the music. + Also, it makes the AMI Park action use the right MOH class when the + channel gets parked. + + Reported by: Zane Conkle + + ASTERISK-25079 #close + + Change-Id: I4b129c5a20c15e63968842460ac5a1a85903cf9f + +2017-12-18 15:36 +0000 [b3e839debd] Corey Farrell + + * Remove constant conditionals (dead-code). + + Some variables are set and never changed, making them constant. This + means that code in the 'false' block of the conditional is unreachable. + + In chan_skinny and res_config_ldap I used preprocessor directive `#if 0` + as I'm unsure if the unreachable code could be enabled in the future. + + Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059 + +2017-12-19 02:50 +0000 [c02e256407] Oron Peled + + * chan_console: Use correct parameter for 'set active' + + chan_console supports multiple devices but the CLI only works on a + single device. 'console set active' selects this device. + + Sadly that CLI picks the wrong command-line parameter and will only + work for a device called 'active'. + + ASTERISK-27490 #close + + Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d + +2017-12-18 23:17 +0000 [bf33a09c37] Corey Farrell + + * core: Fix multiple trivial issues in the core. + + * Fix small leaks in from error conditions in sdp.c and translate.c. + * Check new file descriptor is less than 0, not less than or equal. + + Change-Id: Id7782775486175c739e0c4bf3ea5e17e3f452a99 + +2017-12-18 06:14 +0000 [81474dfb23] Aaron An + + * res_rtp_asterisk: Avoid close the rtp/rtcp fd twice. + + When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before + close the file descriptor. Close the FD twice will hangs the asterisk + under heavy load. + + ASTERISK-27299 #close + Reported-by: Aaron An + Tested-by: AaronAn + + Change-Id: I870a072d73fd207463ac116ef97100addbc0820a + +2017-12-18 19:47 +0000 [8dfc973d64] Corey Farrell + + * main/app: Fix leaks. + + * ast_linear_stream would leak a file descriptor if it failed to allocate + lin. + * ast_control_tone leaked zone and ts if ast_playtones_start failed. + + Additionally added whitespace to ast_linear_stream, pulled assignments + out of conditionals for improved readability. + + Change-Id: I6d1a10cf9161b1529d939b9b2d63ea36d395b657 + +2017-12-18 19:19 +0000 [a790ced2e8] Corey Farrell + + * func_callerid: Initialize app argument structures. + + This module uses AST_DEFINE_APP_ARGS_TYPE to define struct's instead of + directly using AST_DECLARE_APP_ARGS. Initialize the variables declared + in this way. + + Change-Id: If97fbdd8d63a204e2efd498a192effc14e90fb31 + +2017-08-11 17:02 +0000 [4c04e13783] Richard Mudgett + + * bridge_softmix.c: Change remove_destination_streams() return meaning. + + The return value of remove_destination_streams() now means we removed a + stream from the topology by making it a dead stream. Now we won't try to + request a topology change if we didn't remove any streams. + + Change-Id: Icd91571d856a1d04299a24c411e325c1d9d5c61d + +2017-08-11 16:57 +0000 [ea4179599f] Richard Mudgett + + * bridge_softmix.c: Don't match dead streams. + + * Made is_video_source() and is_video_dest() not match dead streams. + + * Optimized is_video_dest() to reduce duplicated code. + + Change-Id: I4e7ab762c7ee98395e78e6516399f57a2609b9a1 + +2017-12-18 18:40 +0000 [91d9eae79b] Corey Farrell + + * bridge_softmix: Fix memory leaks. + + Change-Id: Ifaf3e93b398595d21d07f535330fef77ff15a80c + +2015-11-11 17:20 +0000 [f6393b59af] Richard Mudgett + + * ast_json_pack(): Use safer json ref mechanism. + + Change-Id: I49204db2e57ae96eee43909c18ed007c09ac817e + +2017-12-18 18:04 +0000 [dc04d1ec93] Corey Farrell + + * app_voicemail: Fix memory management issues. + + * mwi_sub_event_cb: mwist leaked on separate_mailbox failure. + * add_email_attachment: A reference to sox_gain_tmpdir was used + after the storage was out of scope. + + Change-Id: I6282c542ff7b82fa091177a912d11234a8b00a30 + +2015-11-11 16:52 +0000 [7054fb8756] Richard Mudgett + + * rtp_engine.c: Eliminate rtcp_report_to_json() RAII_VAR usage. + + Change-Id: I58a22c2ca82e91d7537409b7b3af2d735827a54d + +2017-12-06 20:35 +0000 [5335ad117d] Rodrigo Ramírez Norambuena + + * app_queue: Add feature to set wrapuptime on the queue member + + This patch adds the ability to set the wrapuptime on the queue member + config. + + When the option is set the wrapuptime on the queue member is used instead + of the queue's wrapuptime. + + ASTERISK-27483 #close + + Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902 + +2017-12-18 14:00 +0000 [064c74e4af] Corey Farrell + + * netsock: Remove from Asterisk core. + + This moves netsock.c / netsock.h to the chan_iax2 module. netsock.h has + been marked deprecated since 13.0.0, chan_iax2 is the only remaining + user. + + Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3 + +2017-12-18 12:23 +0000 [731a23fba7] Corey Farrell + + * CLI: Fix 'core set debug channel' completion bug. + + The completion generator is missing a return so typing "core set debug + all off " causes the command to actually execute. + + Change-Id: Ibf6462088a74eee66967732b50445783ebefc20b + +2017-12-18 08:25 +0000 [1769d4a5c6] Joshua Colp + + * confbridge: Clarify mute sound documentation. + + The mute/unmute sounds are only played when the + action is initiated using the DTMF menu. + + ASTERISK-24756 + + Change-Id: I55b3dd5bc166096bf5e2f547ddd0ce355f36e3dc + +2017-12-18 06:36 +0000 [b40c00c97b] Joshua Colp + + * app_transfer: Remove LOCAL from documentation. + + The Local channel has never supported app_transfer + from what I can see so remove it from the documentation. + + ASTERISK-25649 + + Change-Id: Icbcfe297f6f866285a26b3e9fd5c6d00fa22e0e9 + +2017-12-15 19:01 +0000 [4a461bcde4] Richard Mudgett + + * chan_pjsip.c: Improve ast_request() diagnostic msgs. + + Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and + disable_multi_domain=no results in a misleading empty endpoint name + message. The message should say the endpoint was not found. + + * Added missing endpoint not found message. + + * Added more information to the empty endpoint name msgs if available. + + * Eliminated RAII_VAR in request(). + + Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4 + +2016-10-06 01:29 +0000 [6f8b34f9c1] Corey Farrell + + * chan_sip: Add security event for calls to invalid extension. + + Log a message to security events when an INVITE is received to an + invalid extension. + + ASTERISK-25869 #close + + Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f + +2017-12-15 10:26 +0000 [e6768c0f81] Corey Farrell + + * cdr: Minor optimizations. + + * bridge_candidate_process: remove SCOPED_AO2LOCK and return value. + * handle_standard_bridge_enter_message: replace recursive call with goto + statement. + + ASTERISK-24297 + + Change-Id: Id2eaa0822fb8dc799f63422bb3aa89de9d4ee2a2 + +2017-12-12 12:55 +0000 [bf2d35931d] Corey Farrell + + * aco: Minimize use of regex. + + Remove nearly all use of regex from ACO users. Still remaining: + * app_confbridge has a legitamate use of option name regex. + * ast_sorcery_object_fields_register is implemented with regex, all + callers use simple prefix based regex. I haven't decided the best + way to fix this in both 13/15 and master. + + Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b + +2017-12-12 12:36 +0000 [a455e18320] Corey Farrell + + * aco: Create ways to minimize use of regex. + + ACO uses regex in many situations where it is completely unneeded. In + some cases this doubles the total processing performed by + aco_process_config. + + * Create ACO_IGNORE category type for use in place of skip_category + regex source string. + * Create additional aco_category_op values to allow specifying category + filter using either a single plain string or a NULL terminated array + of plain strings. + * Create ACO_PREFIX to allow matching option names to case insensitive + prefixes. + + Change-Id: I66a920dcd8e2b0301f73f968016440a985e72821 + +2017-12-15 07:56 +0000 [03c25a869f] Corey Farrell + + * res_smdi: Fix shutdown ref. + + When adding shutdown refs for OPTIONAL_API components I accidentally + added it to the unload_module function in res_smdi. Move it to + load_module. + + Change-Id: I2b9da38fbc11ef78ea23dbb2df92b684be7f647c + +2017-12-11 17:07 +0000 [9d5797616c] Corey Farrell + + * loader: Use vector to build apha sorted module lists. + + Change-Id: I9c519f4dec3cda98b2f34d314255a31d49a6a467 + +2017-11-21 00:28 +0000 [7b54903313] Corey Farrell + + * loader: Replace priority heap with vector. + + This is needed for future changes which will require being able to + process the load priority out of order. + + Change-Id: Ia23421197f09789940510b03ebbbf3bf24d51bea + +2017-12-14 18:55 +0000 [9755eff46f] Sean Bright + + * res_hep: hepv3_is_loaded() should check if we are enabled + + res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of + unnecessary work even if 'enabled' is set to 'no' in hep.conf. + + Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b + +2017-11-20 23:10 +0000 [3505cc88e8] Corey Farrell + + * loader: Rework of load_dynamic_module. + + * Split off load_dlopen to perform actual dlopen, check results and log + warnings when needed. + * Always use RTLD_NOW. + * Use flags which minimize number of calls to dlopen required. First + attempt always uses RTLD_GLOBAL when global_symbols_only is enabled, + RTLD_LOCAL when it is not. + + This patch significantly reduces the number of dlopen's performed. With + 299 modules my system ran dlopen 857 times before this patch, 655 times + after this patch. + + Change-Id: Ib2c9903cfddcc01aed3e01c1e7fe4a3fb9af0f8b + +2017-11-21 20:34 +0000 [80bf0ee99a] Corey Farrell + + * loader: Minor fix to module registration. + + This protects the module loader itself against crashing if dlopen is + called on a module from outside loader.c. + + * Expand scope of lock inside ast_module_register to include reading of + resource_being_loaded. + * NULL check resource_being_loaded. + * Set resource_being_loaded NULL as soon as dlopen returns. This fixes + some error paths where it was not NULL'ed. + * Create module_destroy function to deduplicate code from + ast_module_unregister and modules_shutdown. + * Resolve leak that occured if a module did not successfully register. + * Simplify checking for successful registration. + + Change-Id: I40f07a315e55b92df4fc7faf525ed6d4f396e7d2 + +2017-12-14 15:27 +0000 [a8aa209901] Corey Farrell + + * res_clialiases: Fix completion pass-through. + + Never ignore contents of line when generating completion options. + + Change-Id: I74389efdfea154019d3b56a9f381610614c044c8 + +2017-12-11 18:20 +0000 [98f7e9251f] Richard Mudgett + + * res_rtp_asterisk.c: Disable packet flood detection for video streams. + + We should not do flood detection on video RTP streams. Video RTP streams + are very bursty by nature. They send out a burst of packets to update the + video frame then wait for the next video frame update. Really only audio + streams can be checked for flooding. The others are either bursty or + don't have a set rate. + + * Added code to selectively disable packet flood detection for video RTP + streams. + + ASTERISK-27440 + + Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70 + +2017-12-14 14:05 +0000 [283d2df680] George Joseph + + * res_pjsip_sdp_rtp: Add NULL check in add_crypto_to_stream + + add_crypto_to_stream wasn't checking for a NULL + session->inv_session->neg before calling pjmedia_sdp_neg_get_state. + This was causing a crash if the negotiation hadn't already been + completed and asterisk was compiled with --enable-dev-mode. + + Change-Id: I57c6229954a38145da9810fc18657bfcc4d9d0c9 + +2017-12-14 12:14 +0000 [c387beb456] Sean Bright + + * res_musiconhold: Start playlist after initial announcement + + Reset the samples counter to zero when we are done playing an + announcement so that we don't skip into the middle of the first file in + the playlist. + + Also add the selected annoucement to the output of 'moh show classes.' + + ASTERISK-24329 #close + Reported by: Thomas Frederiksen + + Change-Id: I2a5f986a31279c981592f49391409ebf38d6f6d0 + +2017-12-14 10:51 +0000 [7a8a187a56] Sean Bright + + * coverity: Fix warnings in res_smdi + + ASTERISK-19657 #close + Reported by: Matt Jordan III, Esq. + + Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a + +2017-12-14 10:22 +0000 [dac5e3a0df] Sean Bright + + * configs: Comment out and change IP of iax.conf [demo] + + This no longer appears to exist, so no sense in causing confusion. + + ASTERISK-27175 #close + Reported by: Tzafrir Cohen + + Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100 + +2017-12-13 14:26 +0000 [a51bfe5a79] George Joseph + + * README: Remove outdated references to tex docs + + Added links to the wiki to replace references to outdated + tex docs. + + ASTERISK-27430 + Reported by: Corey Farrell + + Change-Id: I5007e732b30bc7b63d124c530ae8857c89991209 + +2017-12-13 09:50 +0000 [5f6a3c4399] Corey Farrell + + * CLI: Remove special handling of 'core set verbose' from rasterisk. + + rasterisk does not need to handle setting verbose levels locally, it + should just tell the daemon what it wants and print what it is given. + Just max out the verbose level on the local client so all filtering + happens on the daemon. + + ASTERISK-20281 #close + + Change-Id: Ia305f75f1fc424a9169bfa30ef70d626ace2c8a8 + +2017-12-08 06:48 +0000 [daa3a3009a] sungtae kim + + * Add new AMI action for app_voicemail + + Currently, to figure out specified voicemail's status, there's only one + way to do it, which is use a VoicemailUserEntry AMI message. + But it consumed it too much resource(it check everything). + So, added new AMI action. + + ASTERISK-27470 + + Change-Id: Ie4eba1424a142e5fbd1d9fb1821a3fc1a1e238b7 + +2017-11-30 10:12 +0000 [62f2860c39] Joshua Colp + + * AST-2017-012: Place single RTCP report block at beginning of report. + + When the RTCP code was transitioned over to Stasis a code change + was made to keep track of how many reports are present. This count + controlled where report blocks were placed in the RTCP report. + + If a compound RTCP packet was received this logic would incorrectly + place a report block in the wrong location resulting in a write + to an invalid location. + + This change removes this counting logic and always places the report + block at the first position. If in the future multiple reports are + supported the logic can be extended but for now keeping a count + serves no purpose. + + ASTERISK-27382 + ASTERISK-27429 + + Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116 + +2017-12-13 06:54 +0000 [3370cd21df] Joshua Colp + + * res_pjsip_session: Reinvite using active stream topology if none requested. + + When a connected line update is sent to an endpoint we do not request + a specific stream topology to be used. Previously this resulted in the + configured stream topology being used which may actually differ from the + currently negotiated topology. PJSIP is helpful in this regard in that + it will fill in any missing streams with removed ones. This results in + our own state not matching the SDP, though, and we do not apply the + negotiated SDP. + + This change tweaks the code to use the actively negotiated stream + topology if it is present with a fallback to the configured one. This + results in the SDP and the state having matching information and the + world is happy. + + ASTERISK*27397 + + Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604 + +2017-12-06 08:24 +0000 [0b532367bd] Joshua Colp + + * pjsip: Ignore state changes from old transactions. + + When we fail over to a new target we create a new transaction + and it becomes the current INVITE transaction. This does not + prevent the previous transaction from raising state changes + and causing the session to be prematurely disconnected if a + transport error occurs immediately. + + This change backports a fix from PJSIP that eliminates the + incorrect state change and reduces when they would be raised + in the first place. + + ASTERISK-27408 + + Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34 + +2017-12-12 22:42 +0000 [cb249b2419] Yasuhiko Kamata + + * chan_sip: 3PCC patch for AMI "SIPnotify" + + A patch for sending in-dialog SIP NOTIFY message + with "SIPnotify" AMI action. + + ASTERISK-27461 + + Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4 + +2017-12-12 15:38 +0000 [c7f94e570e] Ivan Poddubny + + * app_queue: Fix extension state subscriptions removed on dialplan reload + + The approach with having a single global subscription to all extension + state changes has one issue: dynamically created hints don't have any + watchers and are therefore garbage collected on the first dialplan + reload. + + This change creates a state subscription for every queue member with a + hint as state_interface, thus increasing the count of watches for + hints, so they are not destroyed prematurely anymore. + + There are 2 side effects: + 1. The state change callback in app_queue is not executed when + there are no members referring to the extension. + 2. The callback is called multiple times for the same hint if it's + associated with more than one queue member. + + Reported by: Steven T. Wheeler + + ASTERISK-18411 #close + + Change-Id: I4956af2136ea2a7f110ac9272eae5f6e676d8f89 + +2017-12-12 15:28 +0000 [0c9cc7e975] Sean Bright + + * chan_sip: Don't send trailing \0 on keep alive packets + + This is a partial fix for ASTERISK~25817 but does not address the + comments regarding RFC 5626. + + Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420 + +2017-12-12 15:19 +0000 [5039b5741c] Dwayne Hubbard + + * chan_sip: Don't crash in Dial on invalid destination + + Stripping the DNID in a SIP dial string can result in attempting to call + the argument parsing macros on an empty string, causing a crash. + + ASTERISK-26131 #close + Reported by: Dwayne Hubbard + Patches: + dw-asterisk-master-dnid-crash.patch (license #6257) patch + uploaded by Dwayne Hubbard + + Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e + +2017-12-12 15:16 +0000 [6a67828b46] Corey Farrell + + * menuselect: Tweak check for recently run configure. + + Recently menuselect has randomly produced an error stating that + configure was just run and make had to be restarted. I believe this is + due to an incorrect menuselect/Makefile rule. The original rule + produced an error if makeopts or autoconfig.h were older than + makeopts.in or autoconfig.h.in. I believe this can create an issue if + makeopts is older than autoconfig.h.in or if autoconfig.h is older than + makeopts.in. The new rules compare files independently. + + Change-Id: Ibca155035fa1392c95e33cbf25f257902abba17b + +2017-12-07 17:51 +0000 [22810fc635] Richard Mudgett + + * chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) + + This patch does three things associated with the initial incoming INVITE + request URI. + + 1) Add access to the full initial incoming INVITE request URI. + + 2) We were not setting DNID on incoming PJSIP channels. The DNID is the + user portion of the initial incoming INVITE Request-URI. The value is + accessed by reading CALLERID(dnid). + + 3) Fix CHANNEL(pjsip,target_uri) documentation. + + * The initial incoming INVITE request URI is now available using + CHANNEL(pjsip,request_uri). + + * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the + initial incoming INVITE request URI user portion. + + * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is + the contact URI. + + * Refactored print_escaped_uri() out of channel_read_pjsip() to handle + pjsip_uri_print() error condition when the buffer is too small. + + ASTERISK-27478 + + Change-Id: I512e60d1f162395c946451becb37af3333337b33 + +2017-12-12 09:28 +0000 [ec1f4bf48d] Sean Bright + + * res_pjsip: Add TLSv1.1 and TLSv1.2 support + + Support for these protocols was added in the same commit as the 'proto' + field, so we can safely use the same ./configure check. + + For reference: https://trac.pjsip.org/repos/changeset/4968 + + Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac + +2017-12-12 08:06 +0000 [0b9d2135a9] Sean Bright + + * res_pjsip: Assign support levels to a few modules + + Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d + +2017-12-09 00:35 +0000 [c01ba7437e] Corey Farrell + + * CLI: Fix 'core show sysinfo' function ordering. + + Handle CLI initialization before any processing occurs. + + Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c + +2017-12-11 15:27 +0000 [b088cddc03] Kevin Harwell + + * pjsip_options: wrongly applied "UNKNOWN" status + + A couple of places were setting the status to "UNKNOWN" when qualifies were + being disabled. Instead this should be set to the "CREATED" status that + represents when a contact is given (uri available), but the qualify frequency + is set to zero so we don't know the status. + + This patch updates the relevant places with "CREATED". It also updates the + "CREATED" status description (value shown in CLI/AMI/ARI output) to a value + of "NonQualified"/"NonQual" as this description is hopefully less confusing. + + ASTERISK-27467 + + Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89 + +2017-12-08 12:04 +0000 [c2ec82bf36] Richard Mudgett + + * stasis_channels.c: Don't set channel snapshot caller_dnid twice. + + Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471 + +2017-12-11 09:45 +0000 [00578fae0a] Sean Bright + + * codec_opus: Make libcurl a dependency in menuselect + + ASTERISK-27475 #close + + Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800 + +2017-12-08 12:48 +0000 [521f741b04] Sean Bright + + * pjsip: Improve CLI completion performance + + Use the new ast_cli_completion_add() function to improve completion + performance for commands like 'pjsip show endpoint.' + + Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348 + +2017-12-07 14:19 +0000 [9a9edc6c9e] Sean Bright + + * astdb: Improve prefix searches in astdb + + Using the LIKE operator requires a full table scan of 'astdb', whereas a + comparison operation is able to use the primary key index. + + This patch adds a new function to the AstDB API for quick prefix matches + and updates res_sorcery_astdb to utilize it. This showed substantial + performance improvement in my test environment. + + Related to ASTERISK~26806, but does not completely resolve it. + + Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1 + +2017-12-08 18:19 +0000 [d2e87b8e14] Corey Farrell + + * loader: Refactor resource_name_match. + + Optimize resource_name_match. This change eliminates use of + ast_strdupa, instead verifying that both basename's are the same length, + then using strncasecmp. + + Change-Id: I477275c0e954c99d74be5abfc8bb6545b04e5a3d + +2017-12-08 14:58 +0000 [dbb376f166] Sean Bright + + * pjsip_configuration: Add correct file header + + Change-Id: I25348c386a222bb704aff07f54375108a6402906 + +2017-12-07 09:52 +0000 [2ffe52a116] Sean Bright + + * utils: Add convenience function for setting fd flags + + There are many places in the code base where we ignore the return value + of fcntl() when getting/setting file descriptior flags. This patch + introduces a convenience function that allows setting or clearing file + descriptor flags and will also log an error on failure for later + analysis. + + Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7 + +2017-12-07 19:33 +0000 [e2dbc26376] Corey Farrell + + * res_stasis and res_speech: Fix load order. + + res_stasis was missing AST_MODFLAG_LOAD_ORDER. Set res_stasis and + res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for + dependent modules. + + Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3 + +2017-12-07 18:22 +0000 [0e4d31eb9c] Kevin Harwell + + * pjsip_options: contacts sometimes not being updated on reload + + For both dynamic and static contacts it was possible that potential AOR + changes were not being applied to all contacts. This was because the qualify + and schedule code was only retrieving AOR's, and contacts with frequencies + greater than zero. + + For instance the following could happen: and AOR/contact has a frequency of 5, + it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are + stopped, a list of AOR's is retrieved with frequency > 0, but none are + selected since in this scenario all are 0. The contact for the one previously + set to 5 though does not get updated, so it's status remains "AVAILABLE". + + This patch makes it so all contacts (static and dynamic) are selected, and + appropriately updated if need be. + + ASTERISK-27467 #close + + Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb + +2017-12-07 18:18 +0000 [bd2218ce63] Kevin Harwell + + * pjsip_options: dynamic contact's fields not updated on reload + + Dynamic contacts were not being properly updated on reload. As a matter of + fact any changes to the AOR that a dynamic contact was associated with were + not being applied. + + On reload, this patch makes it so for each dynamic contact, the associated + AOR is now retrieved and the AOR's fields are applied to the contact. + + ASTERISK-27467 + + Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d + +2017-12-06 23:35 +0000 [c2c9995830] Corey Farrell + + * translate: Skip matrix_rebuild during shutdown. + + Change-Id: I1e5eef4029cba56e33d786c5a5ade8091e531a1e + +2017-12-06 14:49 +0000 [ab191e9782] Corey Farrell + + * sounds_index: Avoid repeatedly reindexing. + + The sounds index is rebuilt each time a format is registered or + unregistered. This causes the index to be repeatedly rebuilt during + startup and shutdown. + + This patch significantly reduces the work done by delaying sound index + initialization until after modules are loaded. This way a reindex only + occurs if a format module is loaded after startup. We also skip + reindexing when format modules are unloaded during shutdown. + + Change-Id: I585fd6ee04200612ab1490dc804f76805f89cf0a + +2017-12-05 18:04 +0000 [3078b7adc2] Richard Mudgett + + * CDR: Fix deadlock setting some CDR values. + + Setting channel variables with the AMI Originate action caused a deadlock + when you set CDR(amaflags) or CDR(accountcode). This path has the channel + locked when the CDR function is called. The CDR function then + synchronously passes the job to a stasis thread. The stasis handling + function then attempts to lock the channel. Deadlock results. + + * Avoid deadlock by making the CDR function handle setting amaflags and + accountcode directly on the channel rather than passing it off to the CDR + processing code under a stasis thread to do it. + + * Made the CHANNEL function and the CDR function process amaflags the same + way. + + * Fixed referencing the wrong message type in cdr_prop_write(). + + ASTERISK-27460 + + Change-Id: I5eacb47586bc0b8f8ff76a19bd92d1dc38b75e8f + +2017-12-06 12:42 +0000 [2af59ebb3a] Corey Farrell + + * media_index: Improve startup. + + This eliminates some wasteful operations in media_index startup. + + * Replace statically set string-fields with char[0]. + * Eliminate pointless RAII_VAR's. + * alloc_variant: Avoid pointless ao2_find on new info->variant. + * Stop trying find_variant before alloc_variant. + * process_media_file: replace ast_str with ast_asprintf. This avoids + reallocation of file_id_str. + + Overall sounds_index.c is about 27% of Asterisk startup time when using + sample configs. This patch reduces it to 20%. This is a half-fix. The + real problem is that the media_index is regenerated repeatedly - 68 + times in my test. + + Change-Id: Ia50b752f8efb356f852b05c4be495a6631af8652 + +2017-12-06 07:36 +0000 [e97e41552e] Richard Mudgett + + * bridge_basic.c: Update transfer diagnostic messages addendum. + + * Added start DTMF transfer verbose messages. + * Made associated transfer messages use a similar message format. + * Adjusted message verbose level as requested by initial reporter. + + ASTERISK-27449 + + Change-Id: I2045714586414b3c5ef1f3cc56c1c4af4b31f551 + +2017-11-29 06:21 +0000 [9d00583164] Niklas Larsson + + * bridge_basic.c: Update transfer diagnostic messages. + + * Add the channel name to diagnostic messages so you will know which + channel failed to transfer. + + * Promoted some debug messages to verbose 4 messages. + + ASTERISK-27449 #close + + Change-Id: Idac66b7628c99379cc9269158377fd87dc97a880 + +2017-12-01 13:54 +0000 [8536a09b86] Richard Mudgett + + * security-events: Fix SuccessfulAuth using_password declaration. + + The SuccessfulAuth using_password field was declared as a pointer to a + uint32_t when the field was later read as a uint32_t value. This resulted + in unnecessary casts and a non-portable field value reinterpret in + main/security_events.c:add_json_object(). i.e., It would work on a 32 bit + architecture but not on a 64 bit big endian architecture. + + Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935 + +2017-11-30 12:50 +0000 [ab63448fa6] Richard Mudgett + + * res_rtp_asterisk.c: Increase strictrtp learning timeout time. + + More complicated direct media reinvite negotiations can result in longer + delays before direct media flows. The strictrtp learning timeout time + was too short. One log showed that the first RTP packet came in just + after three seconds. + + * Increase the strictrtp learning timeout time from 1.5 to 5 seconds. + + ASTERISK-27453 + + Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c + +2017-12-04 08:33 +0000 [e0354bbe82] Alexander Traud + + * res_rtp_asterisk: Correct default in sample configuration file. + + With Asterisk 12 (commit 866d968), the default of "icesupport" changed to + - "yes" in the module "res_rtp_asterisk" and + - "no" in the module "chan_sip". + The latter was reflected in the sample configuration file for "sip.conf". The + former did not make it into "rtp.conf.sample". + + ASTERISK-20643 + + Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36 + +2017-12-04 05:27 +0000 [b2c4e8660a] Alexander Traud + + * chan_sip: Peers with distinct source ports don't match, regardless of transport. + + Previously, peers connected via TCP (or TLS) were matched by ignoring their + source port. One cannot say anything when protocol:IP:port match, yes (see + ). However, when the ports do not match, the + peers do not match as well. + + This change allows two peers connected to an Asterisk server via TCP (or TLS) + behind a NAT (= same source IP address) to be differentiated via their port as + well. + + ASTERISK-27457 + Reported by: Stephane Chazelas + + Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa + +2017-12-04 03:40 +0000 [0611fe581c] Sungtae Kim + + * Add new object for VoicemailUserEntry + + Currently, when the app_voicemail sending VoicemailUserEntry AMI event, there's + no OldMessageCount info for default. + To check the OldMessageCount info, it required IMAP_STORAGE define, but this is + not correct. + Added OldMessageCount item as a default. + + ASTERISK-27456 + + Change-Id: I5c71521c2d1daf8b7b161e31c34d28cca6aea4c7 + +2017-12-03 18:49 +0000 [e2715d2cd4] Joshua Colp + + * pjproject: Clean up disabling of WebRTC support. + + The definition in config_site.h and the argument to the + configure script are not necessary to disable WebRTC + support. The correct argument, --disable-libwebrtc, is + already passed. + + ASTERISK-26980 + + Change-Id: I27da2c894f87914956a72710222e17462d8a44bc + +2017-12-02 15:55 +0000 [39939cecfa] Corey Farrell + + * autoconf: Remove use of m4_ifblank. + + The m4_ifblank macro is not available on CentOS 6, reverse conditionals + to allow use of m4_ifval instead. ./bootstrap.sh was run but this patch + does not result in any difference to the generated configure script. + + Change-Id: I280785deb872ed8d3339d99cce63a2b54d5f1438 + +2017-11-30 14:38 +0000 [075faac2fd] George Joseph + + * AST-2017-013: chan_skinny: Call pthread_detach when sess threads end + + chan_skinny creates a new thread for each new session. In trying + to be a good cleanup citizen, the threads are joinable and the + unload_module function does a pthread_cancel() and a pthread_join() + on any sessions that are active at that time. This has an + unintended side effect though. Since you can call pthread_join on a + thread that's already terminated, pthreads keeps the thread's + storage around until you explicitly call pthread_join (or + pthread_detach()). Since only the module_unload function was + calling pthread_join, and even then only on the ones active at the + tme, the storage for every thread/session ever created sticks + around until asterisk exits. + + * A thread can detach itself so the session_destroy() function + now calls pthread_detach() just before it frees the session + memory allocation. The module_unload function still takes care + of the ones that are still active should the module be unloaded. + + ASTERISK-27452 + Reported by: Juan Sacco + + Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd + (cherry picked from commit 8f5dff543e457ee3450d21e741901609af0cd779) + +2017-12-01 10:01 +0000 [d9fdeae6a4] Sean Bright + + * config: Speed up config template lookup + + ast_category_get() has an (undocumented) implementation detail where it + tries to match the category name first by an explicit pointer comparison + and if that fails falls back to a normal match. + + When initially building an ast_config during ast_config_load, this + pointer comparison can never succeed, but we will end up iterating all + categories twice. As the number of categories using a template + increases, this dual looping becomes quite expensive. So we pass a flag + to category_get_sep() indicating if a pointer match is even possible + before trying to do so, saving us a full pass over the list of current + categories. + + In my tests, loading a file with 3 template categories and 12000 + additional categories that use those 3 templates (this file configures + 4000 PJSIP endpoints with AOR & Auth) takes 1.2 seconds. After this + change, that drops to 22ms. + + Change-Id: I59b95f288e11eb6bb34f31ce4cc772136b275e4a + +2017-12-01 08:29 +0000 [1ad0fbc80e] Sean Bright + + * config: Speed up ACO & sorcery initialization + + When starting Asterisk in the foreground, there is a perceptible delay + when loading modules that use the ACO and sorcery config frameworks. + For example, a lightly configured res_pjsip took 853ms to load on my + VM. + + I tracked down the slowness to the XPath queries used to associate the + relevant documentation with the config options. One improvement was + adding a call to xmlXPathOrderDocElems after loading an XML document. + From the libxml2 docs: + + Call this routine to speed up XPath computation on static documents. + + The second change was to remove recursive descent and wildcard + operators from the XPath queries. After these changes, res_pjsip takes + 85ms to load on my VM and there is no longer a perceptible delay when + starting Asterisk in the foreground. + + Change-Id: I45d457f1580e26bf5a2b0dab16e8e9ae46dcbd82 + +2017-12-01 06:07 +0000 [892df22ccd] Joshua Colp + + * res_http_post: Not all versions of gmime have GMIME_MAJOR_VERSION. + + This change makes the presence of the GMIME_MAJOR_VERSION + definition optional, as not all versions of gmime actually + define it. + + ASTERISK-27454 + + Change-Id: I01d99590045971ed6787899147170a5954077238 + +2017-11-30 21:24 +0000 [35a7036a0d] Corey Farrell + + * README-SERIOUSLY.bestpractices.txt: Convert to markdown + + Follow-up to conversion of README.md. + + Change-Id: I17ee7cf25bc027ece844efa2c1dfe613aff1e35b + +2017-11-17 10:38 +0000 [ce5cfc8ffb] Corey Farrell + + * autoconf: Use m4 conditionals where possible. + + Change-Id: I530c0a72f965437acef6a9a4fbfe5c487f078b65 + +2017-11-17 09:15 +0000 [87a57e8d46] Corey Farrell + + * autoconf: Fix call to AC_CONFIG_AUX_DIR. + + The `pwd` parameter to AC_CONFIG_AUX_DIR is unnecessary, the default + value is $srcdir. + + Additionally remove the AC_REVISION call. It only added a comment and + is pointless without SVN tag replacements. + + Change-Id: I99299a3217f095bddcb2edefb3b9af0ab147bc29 + +2017-11-20 16:58 +0000 [d12a2ab400] Corey Farrell + + * CLI: Remove compatibility code. + + Previous commits maintained compatibility with older remote console + clients as well as maintaining all API's. + + Remove the following compatibility code: + * ast_cli_generatornummatches. + * Remote command "_command nummatches". + * Sorting / duplicate removal by remote console. + + Change-Id: I59e6ce94fa57ae564888442049695f7e46746437 + +2017-11-26 11:47 +0000 [58115e9c21] Alexander Traud + + * translate: Transcode siren14, speex32, silk24, and silk12 via slin16. + + When a format has no pre-recorded sound files, Asterisk has to transcode between + formats. For this, Asterisk has a fixed translation table. If the pre-recorded + sound files are not available in the same sample rate, Asterisk has not only to + transcode but also to resample. + + Asterisk has pre-recorded files for SLN (8000 kHz) and SLN16 (16000 kHz). + However before this change, Asterisk did not take the sample rate into account, + because the translation paths to SLN and SLN16 got the same score/weight in the + table. Consequently, you might have got narrow-band audio with siren14, speex32, + silk24, and silk12 although those are (ultra) wide-band audio codecs. + + With this change, the distance in sample-rates is taken into account. Now on the + Command-Line interface (CLI) 'core show channels', you should see: + (slin@16000)->(slin@32000)->(speex@32000). + + ASTERISK-23735 + Reported by: Richard Kenner + + Change-Id: I9448295c1978be26f8633b6066395e7bbbe2e213 + +2017-11-26 09:44 +0000 [55c4d8e008] Richard Mudgett + + * res_ari: Fix inverted test giving wrong error message. + + The patch for ASTERISK_24560 inverted a test checking if the bridge name + is being updated to a different name. + + * Fix the test to return "Changing bridge name is not implemented" when + someone attempts to change the bridge name. + + ASTERISK-27445 + + Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9 + +2017-11-25 04:09 +0000 [74e7005a74] Alexander Traud + + * translate: Show sample rate for silk, speex, and slin in translation table. + + ASTERISK-24662 + + Change-Id: I3822956984292c99c48bca8e97807e498ccc0e88 + +2017-11-23 13:27 +0000 [02a9952709] Richard Mudgett + + * features.conf.sample: Clarify ActivatedBy documentation wording. + + Change-Id: Id2899331fe05d1909a862ea879742879d086bc64 + +2017-11-22 18:37 +0000 [4b1262c94b] Corey Farrell + + * Add defaultbranch to .gitreview. + + Although the default value of defaultbranch is master I'm adding it + anyways. This way when new major branches are being created the value + can be updated instead of having to remember the name of the key. + + Change-Id: I3db009217c5ae399fb84bee95076f4dbb7fa52d2 + +2017-11-22 18:43 +0000 [fcd9ba2b87] Alexander Anikin + + * add cmd connection creation on creation ooh323 call data structure + + ASTERISK-27353 #close + + Reported by: Marco Giordani + + Change-Id: I455096bd7da016b871afe09af86067c2c7c9f33f + +2017-11-22 10:42 +0000 [db21f7f2e1] Kevin Harwell + + * pjsip: 183 without To tag does not negotiate media + + If a 183 with sdp response is receive without a To tag the sdp is not + negotiated. According to RFC 3261 section 12.1.2 while a To tag is required, + the client needs to still be able to handle the missing tag case for + backwards compatibility. + + This patch, accepted by and applied to pjproject, makes it so if an incoming + 180/183 with SDP comes in without a To tag it gets appropriately handled. + + ASTERISK-27442 #close + + Change-Id: Ic9d6b01e05e8f4874eebbd7adfe05d932025d203 + +2017-11-21 06:39 +0000 [1a349d832d] Alexander Traud + + * res_rtp_asterisk: ICE server-reflexive candidates (srflx) with Dual-Stack. + + Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now, + srflx is gathered and sent via SDP, even when Asterisk is enabled for + Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used. + + ASTERISK-27437 + + Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1 + +2017-11-20 13:05 +0000 [8e1506154f] Corey Farrell + + * res_parking: Set load_pri more appropriately. + + res_parking had an inplicit load_pri of 0 meaning it was one of the very + first modules loaded after modules with global symbols. Set it to + AST_MODPRI_DEVSTATE_PROVIDER as it provides device state for parking + lots. + + Change-Id: I297b6fb3ff6993ec004e667b22a74f5925906259 + +2017-11-17 21:33 +0000 [90f9885f73] Corey Farrell + + * README: Convert to README.md. + + Convert the README file to markdown format, remove the old README. This + causes websites like github to display the README in a much nicer + format with live links. The raw file is still very readable from + plain text editors and terminals. + + Change-Id: I7d13131764a9a9026e5f8a6ddb245a01bbd788e7 + +2017-11-20 16:48 +0000 [b79d04f8f8] Corey Farrell + + * CLI: Finish conversion of completion handling to vectors. + + Change-Id: Ib81318f4ee52a5e73b003316e13fe9be1dd897a1 + +2017-11-07 15:34 +0000 [fbb8c0d3e4] Corey Farrell + + * CLI: Refactor cli_complete. + + * Stop using "_COMMAND NUMMATCHES" on remote consoles. Using this + command had doubled the amount of work needed from the Asterisk + daemon for each completion request. + * Fix code formatting. + * Remove static buffer used to send the command, use the same buffer + that will receive the results. + * Move sort from ast_cli_display_match_list. + + Change-Id: Ie2211b519a3d4bec45bf46e0095bdd01d384cb69 + +2017-11-07 14:13 +0000 [1cd24cd726] Corey Farrell + + * CLI: Rewrite ast_el_strtoarr to use vector's internally. + + This rewrites ast_el_strtoarr to use vector's internally, but still + return the original NULL terminated array of strings. + + Change-Id: Ibfe776cbe14f750effa9ca360930acaccc02e957 + +2017-11-07 14:47 +0000 [9c0a2110f0] Corey Farrell + + * CLI: Refactor ast_cli_display_match_list. + + * Stop estimating line count, just print until we run out of matches. + * Stop freeing entries, the caller does that anyways. + * Stop calculating / returning numoutput, it was ignored. + + Change-Id: I7f92afa8bea92241a95227587367424c8c32a5cb + +2017-11-08 23:42 +0000 [9587a61f4c] Corey Farrell + + * CLI: Create ast_cli_completion_add function. + + Some completion generators are very inefficent due to the way CLI + requests matches one at a time. ast_cli_completion_add can be called + multiple times during one invokation of a CLI generator to add all + results without having to reinitialize the search state for each match. + + Change-Id: I73d26d270bbbe1e3e6390799cfc1b639e39cceec + +2017-11-09 00:39 +0000 [a02cbc2ef3] Corey Farrell + + * CLI: Remove calls to ast_cli_generator. + + The ability to add to localized storage cannot be supported by + ast_cli_generator. The only calls to ast_cli_generator should be by + functions that need to proxy the CLI generator, for example 'cli check + permissions' or 'core show help'. + + * ast_cli_generatornummatches now retrieves the vector of matches and + reports the number of elements (not including 'best' match). + * test_substitution retrieves and iterates the vector. + + Change-Id: I8cd6b93905363cf7a33a2d2b0e2a8f8446d9f248 + +2017-11-20 09:13 +0000 [491e2eba0d] Alexander Traud + + * chan_sip: ICE contained square brackets around IPv6 addresses. + + ASTERISK-27434 + + Change-Id: Iaeed89b4fa05d94c5f0ec2d3b7cd6e93d2d5a8f7 + +2017-11-19 21:23 +0000 [10b4b5d200] Corey Farrell + + * loader: Fix comments in struct ast_module. + + Make the comments follow doxygen format, move comments to the line + before each field they describe. + + Change-Id: Ic445468398b5e88f13910f7c2f70bd15aad33a27 + +2017-11-16 17:25 +0000 [9ae805c900] Corey Farrell + + * cli: Remove silly usage of RAII_VAR. + + Change-Id: I81aacfee7cd26e4fc5eef07bca582700c2975bd7 + +2017-11-16 13:19 +0000 [89ccab95c2] Corey Farrell + + * ccss: Remove silly usage of RAII_VAR. + + Change-Id: I5ce40035e0a940e4e56f6322c1dcd47fbd509b98 + +2017-11-16 12:51 +0000 [5e99c334d1] Corey Farrell + + * app: Remove silly usage of RAII_VAR. + + Change-Id: Ideb594f7aae134974fb78d5477ba0853b97b8625 + +2017-11-16 12:19 +0000 [abdd9fa1a8] Corey Farrell + + * aoc: Remove silly usage of RAII_VAR. + + Change-Id: I07907f833b81aeb0128bc9442a2abb52679c7511 + +2017-11-16 12:55 +0000 [48e1b39b28] Corey Farrell + + * abstract_jb: Remove silly usage of RAII_VAR. + + Change-Id: I9d56175369363d1dc735504cf78a3a5577069f49 + +2017-11-20 13:08 +0000 [d6bbcec571] Corey Farrell + + * res_mwi_external_ami: Remove incorrect load priority. + + res_mwi_external_ami specified AST_MODFLAG_LOAD_ORDER but didn't set + load_pri, resulting in an actual load priority of 0. This module only + provides AMI actions so it has no reason to load early. + + Change-Id: I82987fcf10d3ea42716b2f9df915b16687fd5839 + +2017-11-20 12:54 +0000 [58fa3885cc] Corey Farrell + + * Loader: Remove unneeded load_pri declarations. + + Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri + AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT. + + Change-Id: I0123258eafce324249433a69df15a85cc16e509f + +2017-11-20 09:49 +0000 [7397961b02] Alexander Traud + + * BuildSystem: pjsip_evsub_set_uas_timeout was not used. + + ASTERISK-27435 + + Change-Id: Id318a7ae6d7d69b53f911d30bf3eece64852f15c + +2017-11-19 09:57 +0000 [b4f7f8250f] Corey Farrell + + * Build: Fix OSX build issues. + + OSX does not support 'readlink -f' or 'sed -r'. Replace readlink with + the GNU make macro 'realpath'. Replace sed with grep in one place, cut + in the other. + + ASTERISK-27332 + + Change-Id: I5d34ecca905384decb22ead45c913ae5e8aff748 + +2017-11-19 13:52 +0000 [999e0c17d7] Corey Farrell + + * Build: Fix issues building without SSL. + + * Fix conditional in libasteriskssl. + * Use variables produced by configure to link the SSL and uuid libraries + into libasteriskpj.so instead of hard-coding them. + + ASTERISK-27431 + + Change-Id: I3977931fd3ef8c4e4376349ccddb354eb839b58d + +2017-11-19 13:28 +0000 [53f42cc052] Corey Farrell + + * res_pjsip: Fix warning by deferring implicit type cast. + + Mac doesn't like the comparison of -1 to an enum, so store the result of + ast_sip_str_to_dtmf to an int so we can check for the negative return + value. ast_sip_str_to_dtmf returns an int so this is only delaying the + implicit type cast. + + Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29 + +2017-11-18 21:13 +0000 [75cb403775] Corey Farrell + + * tests: Fix warnings found on Mac. + + test_pbx used raise without explicitly including signal.h. On Mac for + some reason nothing else includes it. + + test_logger checked if an unsigned int was negative. Switch the + variable to 'int' so that error check can be effective. + + Change-Id: Ie1db5dd1818ac25cc2ae41b644f848b5865b1362 + +2017-11-18 20:25 +0000 [83a2c4d2ae] Corey Farrell + + * res_snmp: Declare RONLY if net-snmp headers do not. + + Some net-snmp builds do not provide the RONLY declare, only + NETSNMP_OLDAPI_RONLY. Map RONLY to NETSNMP_OLDAPI_RONLY to get around + this error. + + Change-Id: Ida5c7ad9406515825485c4d3b4a34fd6ad0da577 + +2017-11-18 20:02 +0000 [5a899fc503] Corey Farrell + + * res_fax: Remove checks for unsigned values being >= 0. + + It's impossible for gwtimeout or fdtimeout to be less than 0 because + they are unsigned int's. Remove checks and unreachable branches. + + Change-Id: Ib2286960621e6ee245e40013c84986143302bc78 + +2017-11-18 19:50 +0000 [b4862e463c] Corey Farrell + + * iostream: Fix ast_iostream_printf declaration. + + This adds the printf attribute and changes 'fmt' from 'const void *' to + 'const char *'. This resolves a warning from some compiler for + vsnprintf needing a literal string for format. + + Change-Id: I71c33a8262590042ee451e1146760c10bb22fb78 + +2017-11-18 19:29 +0000 [2fab3aacd6] Corey Farrell + + * app_minivm: Fix possible uninitialized return value. + + Declare 'res' initialized to -1 to deal with earlier error paths that + could cause 'res' to be returned uninitialized. + + Change-Id: I8ac2a5755bf4174d89ef893e924c940f702b104e + +2017-11-16 02:47 +0000 [0ca406c202] Pirmin Walthert + + * res_rtp_asterisk.c: Fix rtp source address learning for broken clients + + Some clients do not send rtp packets every ptime ms. This can lead to + situations in which the rtp source learning algorithm will never learn + the address of the client. This has been discovered on a Mac mini with + a pjsip based softphone after updating to Sierra: as soon as USB + headsets are involved, the softphone will send the second packet 30ms + after the first, the third 30ms after the second and the fourth 1ms + after the third. So in the old implmentation the rtp source learning + algorithm was repeatedly reset on the fourth packet. + + The patch changes the algorithm in a way that doesn't take the arrival + time between two consecutive packets into account but the time between + the first and the last packet of a learning sequence. + + The patch also fixes a second problem: when a user was using a wrong + value for the probation setting there was a LOG_WARNING output stating + that the value had been set to the default value instead. However + the code for setting the value back to defaults was missing. + + ASTERISK-27421 #close + + Change-Id: If778fe07678a6fd2041eaca7cd78267d0ef4fc6c + +2017-11-17 19:36 +0000 [9316a064fd] Corey Farrell + + * README: Send people to secure websites where available. + + We should be sending people to secure web URL's where available. + Update README's and docs. + + Change-Id: Id5b1e049b0b18b49a784f1254605aefa244ce19a + +2017-11-17 19:54 +0000 [5d0529c4d9] Corey Farrell + + * doxygen: Remove obsolete contents. + + Remove doxygen contents that have nothing to do with the current state + of Asterisk. + + Change-Id: Ic072cc8641f9533a202990ccf275ce87e3efd95c + +2017-11-17 09:57 +0000 [1b6e4c1175] Sean Bright + + * res_pjsip: Use reasonable buffer lengths for endpoint identification + + Domains themselves can be up to 255 characters long (per RFC 1035), so + our current buffer sizes are wholly inadequate for many use cases. + + Change-Id: If3f30a68307f1365a1fe06bc4b854c62842c9292 + +2017-11-11 10:09 +0000 [b9f4bb5988] Corey Farrell + + * menuselect: Remove ineffective weak attribute detection. + + menuselect detects compiler support for multiple styles of weak + functions. This is a remnant from 2013 when OPTIONAL_API required weak + functions. It is no longer correct for menuselect to switch + dependencies from optional to required based on lack of weak function + support. + + Note an issue remains - dependencies should switch from optional to + required based on OPTIONAL_API being enabled or disabled. I don't think + this is possible. menuselect needs to know at startup if OPTIONAL_API + is enabled or disabled, so the only way to fix this is to remove + OPTIONAL_API from menuselect and create a configure option. I've left + the code that switches in place but it's preprocessed out. + + Additionally removed: + - WEAKREF variable from Asterisk makeopts.in. + - Related disabled code from test_utils. + - Pointless AC_REVISION call from menuselect/configure.ac. + + Change-Id: Ifa702e5f98eb45f338b2f131a93354632a8fb389 + +2017-11-16 09:48 +0000 [c4f11911ea] Corey Farrell + + * acl: Fix allocation related issues. + + Add checks for allocation errors, cleanup and report failure when they + occur. + + * ast_duplicate_acl_list: Replace log warnings with errors, add missing + line-feed. + * ast_append_acl: Add missing line-feed to logger message. + * ast_append_ha: Avoid ast_strdupa in loop by moving debug message to + separate function. + * ast_ha_join: Use two separate calls to ast_str_append to avoid using + ast_strdupa in a loop. + + Change-Id: Ia19eaaeb0b139ff7ce7b971c7550e85c8b78ab76 + +2017-11-16 09:04 +0000 [781a520b73] Joshua Colp + + * bridge_basic: Ignore answer from transfer target when they've timed out. + + This is a fun one. + + Given the following attended transfer scenario: + + 1. Transfer target is called + 2. Transferer hangs up + 3. Transfer target call attempt reaches timeout + 4. Transfer target is told to hang up + 5. Transfer target answers before channel is hung up + 6. Transferer recall target is called + + A crash would occur. This is because the transfer target call + attempt, despite being told to hang up, would raise a recall + target answer before the recall target had been answered. As it + had not answered there would be no recall target channel and it + would implode. + + This change makes it so that if the transfer target has been + hung up we don't tell the attended transfer code that it has + answered. We also clear out the stimulus that the recall target + has been answered after telling the transfer target to hang up, + in case it was able to raise the information before we told it + to hangup. + + ASTERISK-27361 + + Change-Id: Ifb8b255a9c4d2c5c1b8ad77bf54f659ed286df99 + +2017-11-16 19:39 +0000 [a95f2994c6] Corey Farrell + + * aoc: Fix memory management issues. + + aoc_publish_blob failed to check for msg allocation error and never + released msg. + + Change-Id: Ib31a9ffb81056a0d496a49d7eec795005a44bcd5 + +2017-11-16 16:18 +0000 [7a735d45e2] Sean Bright + + * res_pjsip_transport_websocket: Give transport a meaningful description + + We were not \0 terminating this string, so any attempt to print it would + in the best case show an empty string and in the worst case potentially + crash. + + Change-Id: I63d96ef8f7516ac02a0f91e22dfa8acdc615042c + +2017-11-16 15:00 +0000 [6c53fb5d21] Sean Bright + + * res_pjsip: Use sorcery prefix operation for contact lookup + + This improves performance for registrations assuming that + res_config_astdb is not in use. + + Change-Id: I86f37aa9ef07a4fe63448cb881bbadd996834bb1 + +2017-10-19 14:44 +0000 [d995064fb7] Nir Simionovich + + * This patch adds a beanstalk CEL backend. + + Beanstalkd is a simple to use job queue. It provides a means to + create multiple job queues called "tubes". Each tube can store + multiple jobs, with varying priorities with the queue. Queue + processing is available via a simple TCP socket or via well defined + libraries, avaialble at + https://github.com/kr/beanstalkd/wiki/client-libraries + + This module is based upon the beanstalk-client library, available + for download at: https://github.com/deepfryed/beanstalk-client + + This module currently doesn't support user defined events. + + Change-Id: Ic3a087faeeac045d69a2a018e60e29831ddb95ab + +2017-11-09 19:58 +0000 [e793501084] Richard Mudgett + + * chan_pjsip.c: Improve answer failure log messages. + + * Balanced the session->inv_session refs on answer failure. + + Change-Id: I33542d639d37e692cb46550b972a5fcfc3b804b8 + +2017-11-14 18:00 +0000 [b7b800b689] Richard Mudgett + + * audiohook.c: Fix freeing a frame and still using it. + + Memory corruption happened to the media frame caches when an audio hook + freed a frame when it shouldn't. I think the freed frame was because a + jitter buffer interpolated a missing frame and the audio hook + unconditionally freed it. + + * Made audiohook.c:audio_audiohook_write_list() not free an interpolated + frame if it is the same frame as what was passed into the routine. + + * Made plc.c:normalise_history() use memmove() instead of memcpy() on a + memory block that could overlap. Found by valgrind investigating this + issue. + + ASTERISK-27238 + ASTERISK-27412 + + Change-Id: I548d86894281fc4529aefeb9f161f2131ecc6fde + +2017-11-15 12:10 +0000 [f512707362] George Joseph + + * app_record: Don't set RECORD_STATUS chan var until file is closed + + We've been calling pbx_builtin_setvar_helper to set the + RECORD_STATUS variable before actually closing the recorded file. + If a client is watching VarSet events and tries to do something with + the file when a RECORD_STATUS event is seen, they might attempt to + do so while the file it's still open. + + We now delay calling pbx_builtin_setvar_helper until after we close + the file. + + ASTERISK-27423 + + Change-Id: I7fe9de99953e46b4bafa2b38cf151fe8f6488254 + +2017-11-07 08:25 +0000 [cf1cb3345e] George Joseph + + * ast_coredumper: Add ability to use directory other than /tmp + + The OUTPUTDIR environment variable can now be set either in the + environment itself or in ast_debug_tools.conf. If set, it's used + for all work products instead of /tmp. + + Also added the --tarball-config option that includes the contents + of /etc/asterisk when either --tarball-coredumps or --tarball-results + are used. + + Change-Id: I66b2553319df61caea5b313d084f51978f730b4c + +2017-11-13 07:14 +0000 [29e0add14f] Joshua Colp + + * pjsip / hep: Provide correct local address for Websockets. + + Previously for PJSIP the local address of WebSocket connections + was set to the remote address. For logging purposes this is + not particularly useful. + + The WebSocket API has been extended to allow the local + address to be queried and this is used in PJSIP to set the + local address to the correct value. + + The PJSIP HEP support has also been tweaked so that reliable + transports always use the local address on the transport + and do not try to (wrongly) guess. As they are connection + based it is impossible for the source to be anything else. + + ASTERISK-26758 + ASTERISK-27363 + + Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca + +2017-11-13 17:47 +0000 [14253f9535] Corey Farrell + + * alertpipe: Correct documented return of ast_alertpipe_write. + + Change-Id: I4ea49c441890a81384144479dc93ab5a3989486d + +2017-11-09 19:47 +0000 [edd1016dd8] Corey Farrell + + * core: Use ast_alertpipe for Asterisk signal monitoring thread. + + Reduce the signal monitoring thread file descriptor use from two to one + on systems that support eventfd. + + Change-Id: Id4041a237d481ff699639e153ea6982fee14a462 + +2017-11-13 16:20 +0000 [cdaaa14a5f] Corey Farrell + + * core: Fix configuration of remote console socket path. + + The remote console socket path is the combination of asterisk.conf + settings astrundir from [directories] and astctl from [files]. + Unconditionally combine the two strings after processing all values + to ensure we end up with the correct socket path. + + ASTERISK-27415 + + Change-Id: Ib1e2805d55d6b0955c6430a1a2a93acbf9b091e8 + +2017-11-10 10:37 +0000 [f6ebd16bb8] George Joseph + + * bundled_pjproject: sip_parser: Fix return code in pjsip_find_msg + + The default return code for pjsip_find_msg was PJ_SUCCESS so if + a Content-Length header wasn't found at all, pjsip_find_msg was + returning PJ_SUCCESS instead of PJSIP_EMISSINGHDR. + + Also added the volatile keyword to a few variables that are used + both inside and outside the PJ_TRY/PJ_CATCH block. + + Partial fix for ASTERISK_27408 + + Change-Id: If82ba9de921e3d57df9c68cf96ee45ccc1491f7a + +2017-11-13 14:35 +0000 [2e7f6cd31b] Ben Ford + + * bundled_pjproject: Update to 2.7.1 + + Update from 2.7 to 2.7.1 for bundled pjproject. Changed version + and removed patch files included in the update. + + Change-Id: I55cea8e734b318c2df9daf86aa0802c559ec8357 + +2017-11-09 08:21 +0000 [ffccce76d9] Sean Bright + + * sorcery: Add ast_sorcery_retrieve_by_prefix() + + Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex + only so that they can anchor the potential match as a prefix and not + because they truly need regular expressions. + + Rather than using regular expressions for simple prefix lookups, add + a new operation - ast_sorcery_retrieve_by_prefix - that does them. + + Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79 + +2017-11-07 17:07 +0000 [14d60cee0c] Corey Farrell + + * CLI: Create ast_cli_completion_vector. + + This is a rewrite of ast_cli_completion_matches using a vector to build + the list. The original function calls the vector version, NULL + terminates the vector and extracts the elements array. + + One change in behavior the results are now sorted and deduplicated. This + will solve bugs where some duplicate checking was done before the list + was sorted. + + Change-Id: Iede20c5b4d965fa5ec71fda136ce9425eeb69519 + +2017-11-07 14:00 +0000 [4930404715] Corey Farrell + + * vectors: Add new macro and a string vector definition. + + * AST_VECTOR_STEAL_ELEMENTS - steal the array of elements for use + with non-vector code. + * struct ast_vector_string - a vector of 'char *'. + + Change-Id: I104d1b204be03fccf67e02a195596adcb5ab1e42 + +2017-11-11 13:01 +0000 [90bb0a3e10] Richard Mudgett + + * core: Add cache_media_frames debugging option. + + The media frame cache gets in the way of finding use after free errors of + media frames. Tools like valgrind and MALLOC_DEBUG don't know when a + frame is released because it gets put into the cache instead of being + freed. + + * Added the "cache_media_frames" option to asterisk.conf. Disabling the + option helps track down media frame mismanagement when using valgrind or + MALLOC_DEBUG. The cache gets in the way of determining if the frame is + used after free and who freed it. NOTE: This option has no effect when + Asterisk is compiled with the LOW_MEMORY compile time option enabled + because the cache code does not exist. + + To disable the media frame cache simply disable the cache_media_frames + option in asterisk.conf and restart Asterisk. + + Sample asterisk.conf setting: + [options] + cache_media_frames=no + + ASTERISK-27413 + + Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00 + +2017-11-11 09:42 +0000 [b865d29f1c] Richard Mudgett + + * frame.c: Make ast_frame_free()/ast_frfree() NULL tolerant + + Change-Id: Ic49d821ef88ada38a31bdd835b9531443c55d793 + +2017-11-10 22:04 +0000 [96987737b9] Corey Farrell + + * menuselect: Delete and ignore aclocal.m4. + + This file is temporary output from the bootstrap.sh command, it does not + need to be committed. + + Change-Id: Ie0fd113aff6eac44924c0bd0c900833c6c86a6d9 + +2017-10-30 22:09 +0000 [e9f8b317c3] Corey Farrell + + * Build: Make function constructor/destructor attributes mandatory. + + This change causes the configure script to fail if the C compiler does + not support both function attributes constructor and destructor. These + were already required as modules cannot function without these attributes + and Asterisk requires modules. + + This also has AST_GCC_ATTRIBUTE set a variable + ax_cv_have_func_attribute_$1. This is the same variable name used by + autoconf-archive's AX_GCC_FUNC_ATTRIBUTE, used for the same purpose. + + Change-Id: Id68e8a1447f2a6d707c54b56350e7bfdb33fb663 + +2017-11-10 07:06 +0000 [96f2ee865e] Joshua Colp + + * pjsip: Add patch to allow all transports to be destroyed. + + If a transport is created with the same transport type, source + IP address, and source port as one that already exists the old + transport is moved into a linked list called "tp_list". + + If this old transport is later shutdown it will not be destroyed + as the process checks whether the transport is valid or not. This + check does not look at the "tp_list" when making the determination + causing the transport to not be destroyed. + + This change updates the logic to query not just the main storage + method for transports but also the "tp_list". + + Upstream issue https://trac.pjsip.org/repos/ticket/2061 + + ASTERISK-27411 + + Change-Id: Ic5c2bb60226df0ef1c8851359ed8d4cd64469429 + +2017-11-09 20:34 +0000 [bb77666620] Corey Farrell + + * core: Remove disabled code. + + handle_quit has been disabled since 2003, remove it. + + Change-Id: Idc3aaa6c81676160547078f9b71e8aa43de2db18 + +2017-11-09 13:24 +0000 [23b0ef3e9b] Corey Farrell + + * Build System: Disable parallel make in the root Makefile. + + This ensures that the root Makefile runs only a single target at a time. + SUBMAKE will still honor requested parallelism, so 'make -j8' will build + one directory at a time but allow 8 jobs at once when building a sub + directory. + + This will fix some display glitches related to rebuild of XML + documentation. It will also prevent some edge case errors where + bundled pjproject needs to be rebuild before other parts of Asterisk. + + Change-Id: I4f2ec6fbbec1ada0ccb1109a28ea303524239b1e + +2017-03-29 20:46 +0000 [12010fc5c0] Richard Mudgett + + * chan_pjsip.c: Fix uninitialized cause value on failure. + + Change-Id: I3f9dd3c31bd582e54a30381500077de2319d8cc3 + +2017-11-08 01:40 +0000 [0bda39c668] Corey Farrell + + * DEBUG_FD_LEAKS: Add missing FD creators. + + This adds FD tracking for the following functions: + * eventfd + * timerfd_create + * socketpair + * accept + + ASTERISK-27404 + + Change-Id: Id6848fe904ade2d34eb39d2a20bd6b223e1111fc + +2017-11-07 11:49 +0000 [05f557820b] Corey Farrell + + * bridge_softmix: Note why ast_stream_topology_set_stream cannot fail. + + This appeared in my audit of ast_stream_topology_set_stream callers + not checking for errors but in this situation the call cannot fail. + Add comment so this can be ignored in the future. + + Change-Id: I91d25704859efbe50b8b82cfe1cd3c40ba177c9f + +2017-10-19 13:35 +0000 [dd1a914495] Kevin Harwell + + * AST-2017-011 - res_pjsip_session: session leak when a call is rejected + + A previous commit made it so when an invite session transitioned into a + disconnected state destruction of the Asterisk pjsip session object was + postponed until either a transport error occurred or the event timer + expired. However, if a call was rejected (for instance a 488) before the + session was fully established the event timer may not have been initiated, + or it was canceled without triggering either of the session finalizing states + mentioned above. + + Really the only time destruction of the session should be delayed is when a + BYE is being transacted. This is because it's possible in some cases for the + session to be disconnected, but the BYE is still transacting. + + This patch makes it so the session object always gets released (no more + memory leak) when the pjsip session is in a disconnected state. Except when + the method is a BYE. Then it waits until a transport error occurs or an event + timeout. + + ASTERISK-27345 #close + + Reported by: Corey Farrell + + Change-Id: I1e724737b758c20ac76d19d3611e3d2876ae10ed + +2017-10-03 16:19 +0000 [b358e441cd] Richard Mudgett + + * AST-2017-010: Fix cdr_object_update_party_b_userfield_cb() buf overrun + + cdr_object_update_party_b_userfield_cb() could overrun the fixed buffer if + the supplied string is too long. The long string could be supplied by + external means using the CDR(userfield) function. + + This may seem reminiscent to AST-2017-001 (ASTERISK_26897) and it is. The + earlier patch fixed the buffer overrun for Party A's userfield while this + patch fixes the same thing for Party B's userfield. + + ASTERISK-27337 + + Change-Id: I0fa767f65ecec7e676ca465306ff9e0edbf3b652 + +2017-10-19 13:53 +0000 [74432f51f9] George Joseph + + * AST-2017-009: pjproject: Add validation of numeric header values + + Parsing the numeric header fields like cseq, ttl, port, etc. all + had the potential to overflow, either causing unintended values to + be captured or, if the values were subsequently converted back to + strings, a buffer overrun. To address this, new "strto" functions + have been created that do range checking and those functions are + used wherever possible in the parser. + + * Created pjlib/include/limits.h and pjlib/include/compat/limits.h + to either include the system limits.h or define common numeric + limits if there is no system limits.h. + + * Created strto*_validate functions in sip_parser that take bounds + and on failure call the on_str_parse_error function which prints + an error message and calls PJ_THROW. + + * Updated sip_parser to validate the numeric fields. + + * Fixed an issue in sip_transport that prevented error messages + from being properly displayed. + + * Added "volatile" to some variables referenced in PJ_CATCH blocks + as the optimizer was sometimes optimizing them away. + + * Fixed length calculation in sip_transaction/create_tsx_key_2543 + to account for signed ints being 11 characters, not 9. + + ASTERISK-27319 + Reported by: Youngsung Kim at LINE Corporation + + Change-Id: I48de2e4ccf196990906304e8d7061f4ffdd772ff + +2017-11-06 17:58 +0000 [2c4db2a3d5] Corey Farrell + + * res_pjsip_pubsub: Fix multiple leaks on failure to append vectors. + + Change-Id: I68ece0073ea79667ca41eb10405f516f1d30d482 + +2017-11-06 18:12 +0000 [48e96aba6a] Corey Farrell + + * res_pjsip_history: Fix multiple leaks on vector append failure. + + Change-Id: I41e8d5183ace284095cc721f3b1fb32ade3f940f + +2017-11-06 18:01 +0000 [ecb81ae4de] Corey Farrell + + * res_pjsip_session: Fix multiple leaks. + + * Pre-initialize cloned media state vectors to final size to ensure + vector errors cannot happen later in the clone initialization. + * Release session_media on vector replace failure in + ast_sip_session_media_state_add. + * Release clone and media_state in ast_sip_session_refresh if we fail to + append to the stream topology, return an error. + + Change-Id: Ib5ffc9b198683fa7e9bf166d74d30c1334c23acb + +2017-11-07 12:03 +0000 [9b3db9a7fd] Corey Farrell + + * main/sdp_state: Check for errors from ast_stream_topology_set_stream. + + Change-Id: I84a83ae69daba5d185cc1d939b133a4c23565497 + +2017-11-06 16:37 +0000 [0cfc3cbf02] Richard Mudgett + + * res_pjsip_registrar.c: Fix AOR and pjproject group deadlock. + + One of the patches for ASTERISK_27147 introduced a deadlock regression. + When the connection oriented transport shut down, the code attempted to + remove the associated contact. However, that same transport had just + requested a registration that we hadn't responded to yet. Depending + upon timing we could deadlock. + + * Made send the REGISTER response after we completed processing the + request contacts and released the AOR lock to avoid the deadlock. + + ASTERISK-27391 + + Change-Id: I89a90f87cb7a02facbafb44c75d8845f93417364 + +2017-11-07 11:40 +0000 [eba1179795] Corey Farrell + + * res_pjsip_session: Check for errors from ast_stream_topology_set_stream. + + Free memory and return error if ast_stream_topology_set_stream fails. + + Change-Id: I9f4dbf44bed627243d2f1dd8aea2eab6c38a028d + +2017-11-07 11:34 +0000 [4ac6dd4e95] Corey Farrell + + * res_pjsip_t38: Better error checking for t38_create_media_state. + + Change-Id: I81b2587427c6982aa3e2a3f9ad69cce8d316eb10 + +2017-11-06 15:38 +0000 [fb18895108] Corey Farrell + + * stream: Return error from ast_stream_topology_set_stream. + + ast_stream_topology_set_stream had suppressed error codes from + AST_VECTOR_APPEND. The result of AST_VECTOR_APPEND needs to be returned + to the caller so they can take appropriate action on the stream. + + Change-Id: I6c0d12755743eadba1357f6153526cc055592856 + +2017-11-06 17:21 +0000 [801094da7b] Corey Farrell + + * res_stasis: Fix multiple leaks. + + * res/stasis/app.c JSON passed to app_send needs to be released. + * res/stasis_message.c: objects leak if vector append fails. + + Change-Id: I8dd5385b9f50a5cadf2b1d16efecffd6ddb4db4a + +2017-11-07 06:56 +0000 [02329b9a34] Richard Mudgett + + * res_pjproject.c: Fix ast_strdup() alloc failure. + + Change-Id: I74688038e7afe3a279359cce53aadb28ade51ead + +2017-11-05 22:06 +0000 [a36d8cc533] Aaron An + + * res_pjsip: Avoid crash when contact uri is empty string + + Asterisk will crash if contact uri is invalid, so contact_apply_handler + should check if the uri is NULL or empty. + + ASTERISK-27393 #close + Reported-by: Aaron An + Tested-by: AaronAn + + Change-Id: Ia0309bdc6b697c73c9c736e1caec910b77ca69f5 + +2017-11-06 17:55 +0000 [7ef38d399a] Corey Farrell + + * res_pjsip_outbound_registration: Fix leak on vector add failure. + + Change-Id: I774b88b3c9da41edd4dc8d78f095481f52f2bd46 + +2017-11-06 17:48 +0000 [8684219f79] Corey Farrell + + * res_pjsip_exten_state: Check for vector append failure. + + Release reference to publisher if we fail to add it to the vector. + + Change-Id: I64dff3f481b67b9884f37cadba7a5ccf23d084f3 + +2017-11-06 17:44 +0000 [f899368cd6] Corey Farrell + + * res_pjsip_config_wizard: Fix leaks and add check for malloc failure. + + wizard_apply_handler(): + - Free host if we fail to add it to the vector. + + wizard_mapped_observer(): + - Check for otw allocation failure. + - Free otw if we fail to add it to the vector. + + Change-Id: Ib5d3bcabbd9c24dd8a3c9cc692a794a5f60243ad + +2017-11-06 17:38 +0000 [4016884ef3] Corey Farrell + + * res_stasis_playback: Check for failure to append vector. + + Free resources and return error if we fail to append the vector in + stasis_app_control_play_uri. + + Change-Id: I22c4a90dd859b253f2850c6511de48b25609422b + +2017-11-06 17:33 +0000 [24b9751aaa] Corey Farrell + + * test_sorcery_memory_cache_thrash: Handle error from vector append. + + Cleanup resources when we fail to append the vector and report test + failure. + + Change-Id: I6eb41586fd11dee8c0dfe35e91cb465a4cab7298 + +2017-11-06 17:28 +0000 [29205e7adc] Corey Farrell + + * res_pjsip: Fix leak on error in ast_sip_auth_vector_init. + + Change-Id: Ib0fc7a18f3135ca8990c3984c9e15f6d26e556e8 + +2017-11-06 17:17 +0000 [70fcc043bb] Corey Farrell + + * res_pjproject: Handle error from adding to the buildopts vector. + + Change-Id: I076c7bd207c7989a23005395ce1735392657be65 + +2017-11-06 17:11 +0000 [5247ba4b88] Corey Farrell + + * res_ari_events: Fix use after free / double-free of JSON message. + + When stasis_app_message_handler needs to queue a message for a later + connection it needs to bump the message reference so it doesn't get + freed when the caller releases it's reference. + + Change-Id: I82696df8fe723b3365c15c3f7089501da8daa892 + +2017-11-06 15:33 +0000 [adb4fdcb7b] Corey Farrell + + * stasis: Release object if vector append fails. + + Change-Id: I3e5cc669169aab6175ddfaf7486edeaeb4fdcfb1 + +2017-11-06 15:20 +0000 [2f4f216026] Corey Farrell + + * RTP Engine: Deal with errors returned from AST_VECTOR_REPLACE. + + Check for errors from AST_VECTOR_REPLACE and clean memory if needed. + + Change-Id: I124d15cc1d645f85a72a1279f623c1993b304b0b + +2017-11-06 15:16 +0000 [5762f72425] Corey Farrell + + * PBX: Handle errors from AST_VECTOR_APPEND. + + This resolves potentials leaks on AST_VECTOR_APPEND error in: + * ast_context_add_include2 + * ast_context_add_switch2 + * ast_context_add_ignorepat2 + + Change-Id: Ib60e95c4f622fa3b832d87227c0523a695d736b6 + +2017-11-06 15:10 +0000 [714026b32e] Corey Farrell + + * Messaging: Report error on failure to register tech or handler. + + Message tech and handler registrations use a vector which could fail to + expand. If it does log and error and return error. + + Change-Id: I593a8de81a07fb0452e9b0efd5d4018b77bca6f4 + +2017-11-06 15:07 +0000 [e43c8af77c] Corey Farrell + + * format_cap: Fix leak on AST_VECTOR_APPEND error. + + format_cap_framed_init can fail on AST_VECTOR_APPEND. This should + report failure to the caller and clean the newly allocated frame. + + Change-Id: Ica0661235bf09497bf23d844ceb01f21b41a55b0 + +2017-11-06 14:23 +0000 [64bcb65a78] Corey Farrell + + * stasis: Remove silly use of RAII_VAR in stasis_forward_all. + + Change-Id: I46de4c968d40144d5b049966304ff66c1469fb65 + +2017-11-06 12:51 +0000 [b7e1034009] Corey Farrell + + * CLI: Remove unused internal command. + + The internal CLI command "_command complete" was last used by Asterisk + 0.2.0. Since then we've been using "_command nummatches" and "_command + matchesarray". + + Change-Id: I682fe1e21a24a3bb5bd04146e639f1c5866bcfce + +2017-11-03 18:08 +0000 [923424019b] Richard Mudgett + + * stasis_bridges.c: Fix off-nominal json memory leaks. + + Change-Id: Ib1181a36b317c86bff1ef2e44a17a0b1c73cfdc8 + +2017-11-03 17:43 +0000 [f81970d3fc] Richard Mudgett + + * stasis_channels.c: Remove a very silly RAII_VAR(). + + Change-Id: I28b458b3c1a442c4ef0be7b4986a95ea4149e14f + +2017-11-06 10:29 +0000 [36fedea8c1] Joshua Colp + + * res_pjsip_pubsub: Ensure remote URI contains URI only. + + This change makes it so that any user of the pubsub + API that requests the remote URI receives only the URI. + Previously the entire string was returned, which could + contain a display name. + + ASTERISK-27290 + + Change-Id: If1d0cd6630f0a264856d31d2a67933109187a017 + +2017-11-03 16:14 +0000 [9771f089f5] Richard Mudgett + + * stasis/app.c: Optimize stasis_app_get_debug_by_name() + + * Eliminate RAII_VAR() + * Short circuit application name lookup if global debug enabled. + + Change-Id: I5f78b7bd6ca7fd2c3b07cbbe036c6a93b4681123 + +2017-11-02 18:40 +0000 [ee08f10d06] Richard Mudgett + + * Fix ast_(v)asprintf() malloc failure usage conditions. + + When (v)asprintf() fails, the state of the allocated buffer is undefined. + The library had better not leave an allocated buffer as a result or no one + will know to free it. The most likely way it can return failure is for an + allocation failure. If the printf conversion fails then you actually have + a threading problem which is much worse because another thread modified + the parameter values. + + * Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL + on failure. That is much more useful than either an uninitialized pointer + or a pointer that has already been freed. Many uses won't have to check + for failure to ensure that the buffer won't be double freed or prevent an + attempt to free an uninitialized pointer. + + * stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by + ast_asprintf(). + + * ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to + the wrong thing which is now not needed even if assigning to the right + thing. + + Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23 + +2017-11-06 08:05 +0000 [ca4e6b568f] Sean Bright + + * res_pjsip: Ignore empty TLS configuration + + When using realtime, fields that are not explicitly set by an + administrator are still presented to sorcery as empty strings. Handle + this case explicitly. + + In this particular case, if any of these fields are required for TLS + support, their existence should be validated in the 'apply' handler once + we have a complete transport definition. + + ASTERISK-27032 #close + Reported by: seanchann.zhou + + Change-Id: Ie3b5fb421977ccdb33e415d4ec52c3fd192601b7 + +2017-09-29 09:50 +0000 [04d3785a79] Sean Bright + + * dtls: Add support for ephemeral DTLS certificates. + + This mimics the behavior of Chrome and Firefox and creates an ephemeral + X.509 certificate for each DTLS session. + + Currently, the only supported key type is ECDSA because of its faster + generation time, but other key types can be added in the future as + necessary. + + ASTERISK-27395 + + Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4 + +2017-11-06 03:21 +0000 [4013bfa52b] Corey Farrell + + * configure: Add autoconf check for libopusfile. + + This check is being added to make it easier for end-users of third party + open source Opus modules. This was removed by ASTERISK-26426 but only + the module needed to be removed. + + Change-Id: I62b9cd0c4fa8a77596ab0e042948a643a1152677 + +2017-11-06 03:18 +0000 [19332e6968] Alexander Traud + + * tcptls: Print notice when TLS is enabled but not configured. + + Asterisk can be compiled without a SSL/TLS library, without the Development + Headers of OpenSSL. However, if TLS (SIP) or Secure-WebSockets (WebRTC) was + enabled in a configuration file, Asterisk did not notice the user. Asterisk + failed silently, only the corresponding TCP ports were not open. + + ASTERISK-27394 + Reported-by: mossley74 + + Change-Id: Ib8b7539a5b2af8154c22e5f7a40fc68f95d95b93 + +2017-11-04 06:05 +0000 [2ebea5aa03] Alexander Traud + + * install_prereq: Checkout of libSRTP 2.x. + + Since Asterisk 13.17, libSRTP 2.x is supported. Therefore, its latest version + is installed again via the script install_prereq. + + ASTERISK-27356 + + Change-Id: I13125839a79052356469e41edacbebff0a937d39 + +2017-11-01 17:47 +0000 [79ddcdbc70] Richard Mudgett + + * Stasis/ARI: Fix off-nominal path json memory leaks. + + Change-Id: Id569c624c426e3b22a99936473c730592d8b83fb + +2017-11-02 11:38 +0000 [229790ea3d] Richard Mudgett + + * AOC: Fix AOC-S json memory leak. + + Change-Id: I3a1d40a41a8a7d00fa4a187de6a343a79155d3ef + +2017-11-01 18:04 +0000 [de4a4796d0] Richard Mudgett + + * res_stasis_device_state.c: Optimize stasis_app_device_states_to_json() + + * Eliminate RAII_VAR() + * Replace looped alloca with a char[] since that is how it is used anyway. + + Change-Id: Ia27e64a884afa0f50b9ffdb1cf23da6bfa51ffdf + +2017-11-01 18:58 +0000 [103b05bb4b] Richard Mudgett + + * res_stasis_mailbox.c: Fix leak of mailbox container. + + Change-Id: I7d33c1635713047e7d1597c9d882f7dc006d94b4 + +2017-11-03 10:35 +0000 [290bad22c9] Corey Farrell + + * Build System: Fix build failure caused by recent CLI improvements. + + We use the editline library to help with filename completion in our CLI + interface. Some systems failed to find the header when included from + loader.c. This is fixed by setting the proper CFLAGS for the build of + loader.o. + + ASTERISK-27378 + + Change-Id: Ib7fd496f1d7ed48141a2eadd5dd61cab2f2308be + +2017-11-01 11:12 +0000 [f8e0f9be22] Ben Ford + + * res_pjsip: Add to list of valid characters for from_user. + + Fixes a regression where some characters were unable to be used in + the from_user field of an endpoint. Additionally, the backtick was + removed from the list of valid characters, since it is not valid, + and it was replaced with a single quote, which is a valid character. + + ASTERISK-27387 + + Change-Id: Id80c10a644508365c87b3182e99ea49da11b0281 + +2017-11-02 05:34 +0000 [8701479386] Joshua Colp + + * core: Don't attempt to write to a stream that does not exist. + + When a frame is provided to ast_write ensure that a multistream + capable channel has a stream for it before attempting to give it + to the channel driver. In some cases (such as a deferred SDP + negotiation) the stream may not yet exist. + + ASTERISK-27364 + + Change-Id: Icf84ca982a67cdd6e9a71851eb7eb1bd0e865276 + +2017-11-02 01:57 +0000 [606ae3484a] Corey Farrell + + * Add missing menuselect dependencies. + + This adds menuselect dependencies for modules that use symbols of other + modules. + + ASTERISK-27390 + + Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385 + +2017-11-01 22:57 +0000 [b616b7e4a9] Corey Farrell + + * res/ari/resource_bridges.h: Update from 'make ari-stubs'. + + A comment was updated when I ran 'make ari-stubs'. + + Change-Id: Ib5154ae3ad72aff53374c28ead540fe349c42175 + +2017-11-01 19:46 +0000 [79f111e1f3] Corey Farrell + + * Prevent unload of modules which implement an Optional API. + + Once an Optional API module is loaded it should stay loaded. Unloading + an optional API module runs the risk of a crash if something else is + using it. This patch causes all optional API providers to tell the + module loader not to unload except at shutdown. + + ASTERISK-27389 + + Change-Id: Ia07786fe655681aec49cc8d3d96e06483b11f5e6 + +2017-10-30 17:30 +0000 [b9f457eac0] Corey Farrell + + * Modules: Additional improvements to CLI completion. + + Replace 'needsreload' argument with a 'type' argument to specify which + type of modules you want completion. This provides more accurate CLI + completion for load and unload commands. + + * 'module unload' now excludes modules that have active references or are + not running. + * 'module load' now excludes modules that are already running. + * 'core set debug [atleast] [module]' shows running modules only. + + ASTERISK-27378 + + Change-Id: Iea3e00054461484196c46f688f02635cc886bad1 + +2017-11-01 13:58 +0000 [1bfd1cf640] Sean Bright + + * pjsip_message_filter: Only do interface lookup for wildcard addresses. + + Change-Id: Ie083987e69dc43b6861671c218cacacc11b2072f + +2017-10-31 15:08 +0000 [1e70011710] Kevin Harwell + + * features: Bridge application's BRIDGERESULT not appropriately set + + The dialplan application "Bridge" was not setting the BRIDGERESULT to failure + when a failure did occur. Even worse if it did fail to join the bridge it would + still report success. + + This patch now sets the BRIDGERESULT variable to an appropriate value for a + given condition state. Also, removed the value INCOMPATIBLE as a valid result + type since it is no longer used. + + ASTERISK-27369 #close + + Change-Id: I22588e7125a765edf35cff28c98ca143e9927554 + +2017-10-31 13:18 +0000 [f2175c5a39] Corey Farrell + + * res_ari_channels: Fix reference leak in channel_state_invalid. + + channel_state_invalid leaked a reference to the channel snapshot any + time it was aquired. + + ASTERISK-27067 #close + + Change-Id: I8c653f00416b39978513c5605c4be0f03b1df29a + +2017-10-25 17:31 +0000 [4c535f5c30] Joshua Colp + + * core / pjsip: Add support for grouping streams together. + + In WebRTC streams (or media tracks in their world) can be grouped + together using the mslabel. This informs the browser that each + should be synchronized with each other. + + This change extends the stream API so this information can + be stored with streams. The PJSIP support has been extended + to use the mslabel to determine grouped streams and store + this association on the streams. Finally when creating the + SDP the group information is used to cause each media stream + to use the same mslabel. + + ASTERISK-27379 + + Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad + +2017-10-30 09:20 +0000 [022de525be] Tzafrir Cohen + + * ast_coredumper: allow setting asterisk binary explicitly + + Adds an extra option, --asterisk-bin= to ast_coredumper. If + provided, the binary given to gdb will be the parameter, rather than + asterisk from the PATH. + + ASTERISK-27380 #close + + Change-Id: I25f5b91eb75059b0fb2f142e468c26b283b0a9f3 + +2017-10-25 01:10 +0000 [3052b56423] Florian Floimair + + * alembic: Add bundle column in ps_endpoints table + + The ps_endpoints table was missing the bundle column + introduced with the bundle feature in + commit 065c3005ad92. + + ASTERISK-27374 #close + + Change-Id: Ic900f4f2c20f64b99ea898d50f5c0a7117472d46 + +2017-10-30 00:32 +0000 [e82b921c35] Corey Farrell + + * Modules: Fix issues with CLI completion. + + * Stop using ast_module_helper to check if a module is loaded, use + ast_module_check instead (app_confbridge and app_meetme). + * Stop ast_module_helper from listing reload classes when needsreload + was not requested. + + ASTERISK-27378 + + Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239 + +2017-10-28 19:18 +0000 [9bad4c74cc] Igor Goncharovskiy + + * app_agent_spool: Fix typo in dtmf features usage desctiption + + Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function + instead of correct 'dtmf_features' + + ASTERISK-27377 #close + + Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930 + +2017-10-27 13:41 +0000 [0991874430] Corey Farrell + + * res_pjsip_pubsub: Resolve potential crash in allocate_subscription. + + When allocate_subscription fails to initialize fields of the new sub it + calls destroy_subscription. + + Change-Id: I5b79c915ec216dc00c13c1e4172137864a4bec85 + +2017-10-26 12:18 +0000 [26607e4e3b] Richard Mudgett + + * app_voicemail.c: Fix compiler warning with IMAP build. + + ASTERISK-27181 + + Change-Id: Ic4468b49860bd7f67e922baf4c9e96828c184d17 + +2017-10-25 14:38 +0000 [2ca3dbb197] Richard Mudgett + + * codec.c: Defensively check the returned samples. + + Earlier versions of the codec_opus samples_count callback can return + negative error values on undecodable frames. This resulted in a divide by + zero exception. + + * Added a defensive check in ast_codec_samples_count() for a "negative" + samples count return value. Log the event and set the count to zero. + + ASTERISK-27194 + + Change-Id: Icf69350307ecbbc80a3d74de46af9bd80ea17819 + +2017-10-24 10:33 +0000 [9e1fbab382] Joshua Colp + + * res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint. + + When the identify_by option on an endpoint is set to ip it will + only be identified using the res_pjsip_endpoint_identifier_ip module. + This ensures that it is not mistakenly matched using the username of + the From header. To ensure behavior has not changed the default has + been changed to "username,ip" for the identify_by option. + + ASTERISK-27206 + + Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd + +2017-10-25 12:26 +0000 [4aec70690d] George Joseph + + * ast_coredumper: Add gzipping of binaries and display of signal info + + The --tarball-coredump option now creates a gzipped tarball of + coredumps processed, their results txt files and copies of + /etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and + /usr/lib(64)/asterisk as those files are needed to properly examine + the coredump. The file will be named + /tmp/asterisk..coredumps.tar.gz or + /tmp/asterisk-.coredumps.tar.gz if --tarball-uniqueid was + specified. + + Added dumps of *_siginfo to the top of the txt files so you can + tell what signal was invoked. + + Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9 + +2017-10-25 09:23 +0000 [3821be1c68] Ben Ford + + * http.c: Fix http header send content. + + Currently ast_http_send barricades a portion of the content that + needs to be sent in order to establish a connection for things + like the ARI client. The conditional and contents have been changed + to ensure that everything that needs to be sent, will be sent. + + ASTERISK-27372 + + Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d + +2017-03-30 09:51 +0000 [5553adb8ba] Corey Farrell + + * Build System: Fix --disable-xmldoc option. + + The configure option to disable XML documentation does not currently + work. This patch makes it effective, but also causes an ABI change by + removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs + from being automatically generated, but they can still be manually + generated with 'make doc/core-en_US.xml'. + + ASTERISK-26639 + + Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44 + +2017-10-23 00:55 +0000 [569e9a8391] Corey Farrell + + * Single API for ast_store_lock_info and ast_remove_lock_info. + + This makes the 'bt' parameter unconditional for ast_store_lock_info and + ast_remove_lock_info. The 'bt' parameter is unused when HAVE_BKTR is + undefined. + + Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453 + +2017-10-24 09:43 +0000 [6474de5f72] Corey Farrell + + * chan_sip: Fix SUBSCRIBE with missing "Expires" header. + + When chan_sip receives a SUBSCRIBE request with no "Expires" header it + processes the request as an unsubscribe. This is incorrect, per RFC3264 + when the "Expires" header is missing a default expiry should be used. + + ASTERISK-18140 + + Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5 + +2017-10-24 07:24 +0000 [7126520b3e] Alexander Traud + + * lpc10: Avoid compiler warning when DONT_OPTIMIZE/COMPILE_DOUBLE. + + ASTERISK-23556 + Reported by: Marcello Ceschia + + Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c + +2017-10-07 12:14 +0000 [841ac3ded6] Corey Farrell + + * hashtab: Use ast_free. + + A few places in hashtab use free instead of ast_free, remove declaration + of ASTMM_LIBC from hashtab.c as it's no longer needed. + + Change-Id: I2ff089bad71640c03c3ce97f1b00fc962ef79427 + +2017-10-23 01:02 +0000 [fb585cf185] Corey Farrell + + * Bundled pjproject: Enable pj_assert when dev-mode is enabled. + + ASTERISK-27359 + + Change-Id: Ib01fb6c01f9bb87129374a51cb9318c474147517 + +2017-10-23 13:44 +0000 [ee21076151] Corey Farrell + + * main/Makefile: Remove rule for non-existant testexpr2. + + Change-Id: Ibb3e47f27a395d74d8c5263db015b05434f5969b + +2017-10-23 12:42 +0000 [a9e9608982] Corey Farrell + + * test_config: Fix failure and segfault when config_hook is run twice. + + On second run the config_hook test was unexpectedly failing to load + test_config.conf because it was still unmodified since the last load. + This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial + loads, only using it when we are tested that a reload of unmodified + files do not initiate the hook. + + ASTERISK-25960 + + Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856 + +2017-10-23 12:23 +0000 [6f0431798e] George Joseph + + * res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6 + + create_outgoing_sdp_stream was setting "addr_type = STR_IP6" only + when an ipv6 media_address was specified on the endpoint. If + rtp_ipv6 was set and ast_sip_get_host_ip_string returned an ipv6 + address, we were leaving the addr_type set at the default of + STR_IP4. This caused the address type to be set incorrectly on the + "o" and "c" SDP attributes even though the address was set + correctly. Some clients don't like the mismatch. + + * Removed the test for endpoint/media_address and now check all + addresses for ipv6. + + ASTERISK-27198 + Reported by: Martin Cisárik + + Change-Id: I5214fc31b728117842243807e7927a319cf77592 + +2017-10-23 07:53 +0000 [488f98310f] Richard Mudgett + + * app_agent_pool.c: Fix online documentation typo. + + Change-Id: Ib0bc95fd0ec288c78c313823254d7a84ebfc4429 + +2017-10-22 17:32 +0000 [252353e0a9] Joshua Colp + + * res_xmpp: Ensure the connection filter is available. + + Users of the API that res_xmpp provides expect that a + filter be available on the client at all times. When + OAuth authentication support was added this requirement + was not maintained. + + This change merely moves the OAuth authentication to + after the filter is created, ensuring users of res_xmpp + can add things to the filter as needed. + + ASTERISK-27346 + + Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886 + +2017-10-21 03:44 +0000 [840e08716b] Alexander Traud + + * chan_sip: Crypto attribute not last but first on SDP media level. + + This matches the behavior of the other SIP channel driver, chan_pjsip. + + ASTERISK-27365 + + Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6 + +2017-10-17 10:53 +0000 [e41561fc2a] Richard Mudgett + + * res_pjproject.c: Upgrade bundled PJPROJECT to 2.7 + + Update patches included in bundled PJPROJECT for the new version. + + ASTERISK-27355 + + Change-Id: I9ac5dbbffaadca25ad24fac8b9ab615e5ace6083 + +2017-10-16 16:46 +0000 [4559cd0e28] Nir Simionovich + + * This patch adds a beanstalk CDR backend. + + Beanstalkd is a simple to use job queue. It provides a means to + create multiple job queues called "tubes". Each tube can store + multiple jobs, with varying priorities with the queue. Queue + processing is available via a simple TCP socket or via well defined + libraries, avaialble at + https://github.com/kr/beanstalkd/wiki/client-libraries + + This module is based upon the beanstalk-client library, available + for download at: https://github.com/deepfryed/beanstalk-client + + Change-Id: I5fe4089a34ab3b39230786d9bbfddafa56715f48 + +2017-10-18 13:41 +0000 [4760b2445c] Corey Farrell + + * res_pjsip_pubsub: Prevent unload except during shutdown. + + Prevent unload of the module as certain pjsip initialization functions + cannot be reversed. This required a reorder of the module_load so that + the non-reversable pjsip functions are not called until all potential + errors have been ruled out. + + ASTERISK-24483 + + Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7 + +2017-10-18 13:37 +0000 [449ee66a11] Corey Farrell + + * res_pjsip_refer: Prevent unload except during shutdown. + + Prevent unload of the module as certain pjsip initialization functions + cannot be reversed. + + ASTERISK-24483 + + Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d + +2017-10-18 12:04 +0000 [c9e19b31f5] Corey Farrell + + * chan_sip: Fix output of 'sip set debug off'. + + When sip.conf contains 'sipdebug=yes' it is impossible to disable it + using CLI 'sip set debug off'. This corrects the output of that CLI + command to instruct the user to turn sipdebug off in the configuration + file. + + ASTERISK-23462 #close + + Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318 + +2017-10-16 10:53 +0000 [955a891a84] Corey Farrell + + * app_macro deprecation. + + * Mark the module deprecated. + * Disable the module by default. + * Produce a warning the first time a macro is used. + * Note deprecation related options in app_dial and app_queue. + + ASTERISK-27350 + + Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc + +2017-10-18 03:30 +0000 [95b45d1c46] Alexander Traud + + * res_srtp: Add support for libsrtp2 with AES-GCM. + + Beside allowing AES-GCM again, this adds AES-192 again. + + ASTERISK-27356 + + Change-Id: Ia97a435faf26300335d9552fa676b5d17e5f7233 + +2017-10-14 14:41 +0000 [5d8c517960] Joshua Colp + + * bridge_softmix: Reduce topology cloning and improve renegotiation. + + As channels join and leave an SFU the bridge_softmix module + needs to renegotiate to add and remove their streams from + the other participants. Previously this was done by constructing + the ideal stream topology every time but in the case of leave + this was incomplete. + + This change makes it so bridge_softmix keeps an ideal stream + topology for each channel and uses it when making changes. This + ensures that when we request a renegotiation we are always + certain that we are aiming for the best stream topology + possible. In the case of a channel leaving this ensures that + we try to have an existing participant fill their place if + a participant has a fixed limit on the maximum number of video + streams they allow. + + ASTERISK-27354 + + Change-Id: I58070f421ddeadd2844a33b869b052630cf2e514 + +2017-10-06 15:55 +0000 [73164d0d7f] Richard Mudgett + + * cdr.c: Rename the Party A CDR container. + + * Rename the Party A CDR container from active_cdrs_by_channel to + active_cdrs_master. + + * Renamed the support functions associated with active_cdrs_master + appropriately. + + ASTERISK-27335 + + Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7 + +2017-10-02 17:42 +0000 [fe1120cf88] Richard Mudgett + + * cdr.c: Add container to key off of Party B channel names. + + The CDR performance gets worse the further it gets behind in processing + stasis messages. One of the reasons is because of a n*m loop used when + processing Party B information. + + * Added a new CDR container that is keyed to Party B so we don't need such + a large loop when processing Party B information. + + NOTE: To reduce the size of the patch I deferred to another patch the + renaming of the Party A active_cdrs_by_channel container to + active_cdrs_master and renaming the container's hash and cmp functions + appropriately. + + ASTERISK-27335 + + Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249 + +2017-10-11 06:04 +0000 [da24d425eb] Torrey Searle + + * contrib/script/sip_to_pjsip: implement 'all' for allow/disallow + + when 'all' is specified in an allow or disallow section, it should erase + all values from the inverse section in the default config. E.G. + allow=all should erase any deny values from default config & + vice-versa + + ASTERISK-27333 #close + + Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6 + +2017-10-14 04:11 +0000 [c4f40b778a] Guido Falsi + + * chan_dahdi: wrap include file which is not present on BSD systems in #ifdef + + The sys/sysmacros.h include file does not exist in BSD systems and + is not required to build this module there. + Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section + already exist I moved that include line inside it's #else branch. + + ASTERISK-27343 #close + + Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1 + +2017-10-13 09:43 +0000 [8f65d91dfd] Alexander Traud + + * res_pjsip_session: Rewrite o= with external_media_address. + + PJSIP allows a domain name as external_media_address. This allows chan_pjsip to + be used behind a NAT with changing IP addresses. The IP address of that domain + is resolved to the c= line already. This change sets also the o= line to that + domain. + + ASTERISK-27341 #close + + Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4 + +2017-10-12 12:03 +0000 [7d51a79beb] Joshua Colp + + * bridge_simple: Improve renegotiation success rate. + + When making channels compatible the bridge_simple module + will renegotiate one to better match the other. Some + endpoints incorrectly terminate the call if this process + fails. + + To better handle this scenario the audio streams present + on the new requested topology will include any existing + negotiated formats that happen to exist on the first + valid audio stream. This ensures formats are persent that + are known to be acceptable to the remote endpoint. + + ASTERISK-27259 + + Change-Id: I8fc0cc03e8bcfd0be8302f13b9f32d8268977f43 + +2017-10-13 08:51 +0000 [ee65d5ac7c] Corey Farrell + + * ast_bt_get_symbols: Prevent double-free. + + It's possible for bfdobj to be created but syms not created. If syms + was not allocated in the current loop iteration but was allocated in the + previous iteration it would crash. + + ASTERISK-27340 + + Change-Id: I5b110c609f6dfe91339f782a99a431bca5837363 + +2017-10-13 08:12 +0000 [44d9446eb5] Alexander Traud + + * tcptls: NULL-check the parameter of ast_ssl_teardown before accessing it. + + This avoids a crash on stopping a chan_sip which failed to start its TLS server. + + ASTERISK-27339 #close + + Change-Id: I327fc70db68eaaca5b50a15c7fd687fde79263d5 + +2017-09-29 14:26 +0000 [f369be21a8] Richard Mudgett + + * cdr.c: Eliminated many calls to ao2_global_obj_ref(). + + The CDR performance gets worse the further it gets behind in processing + stasis messages. One of the reasons is we were getting the global config + to determine if we needed to log a debugging message. + + * Many calls to ao2_global_obj_ref() were just so we could determine if + debug mode is enabled. Made a global flag to check instead. + + * Eliminated many RAII_VAR() usages associated with the remaining + ao2_global_obj_ref() calls. + + * Added missing NULL checks for the returned ao2_global_obj_ref() value. + + ASTERISK-27335 + + Change-Id: Iceaad93172862f610cad0188956634187bfcc7cd + +2017-10-06 13:45 +0000 [2eea087401] Richard Mudgett + + * cdr.c: Defer getting ao2_global_obj_ref() until needed. + + The CDR performance gets worse the further it gets behind in processing + stasis messages. One of the reasons is we were getting the global config + even if we didn't need it. + + * Most uses of the global config were only needed on off nominal code + paths so it makes sense to not get it until absolutely needed. + + ASTERISK-27335 + + Change-Id: I00c63b7ec233e5bfffd5d976f05568613d3c2365 + +2017-10-05 18:08 +0000 [7c7a917874] Richard Mudgett + + * cdr.c: Set stringfields only if they are different. + + The CDR performance gets worse the further it gets behind in processing + stasis messages. One of the reasons is we were repeatedly setting string + fields to potentially the same string in base_process_party_a(). Setting + a string field involves allocating room for the new string out of a memory + pool which may have to allocate even more memory. + + * Check to see if the string field is already set to the desired string. + + ASTERISK-27335 + + Change-Id: I3ccb7e23f1488417e08cafe477755033eed65a7c + +2017-10-05 18:03 +0000 [c80c8f2ab9] Richard Mudgett + + * cdr.c: Fix setting dnid, callingsubaddr, and calledsubaddr + + The string comparisons for setting these CDR variables was inverted. We + were repeatedly setting these CDR variables only if the channel snapshots + had the same value. + + ASTERISK-27335 + + Change-Id: I9482073524411e7ea6c03805b16de200cb1669ea + +2017-08-25 08:19 +0000 [21c0283b78] Thomas Sevestre + + * features, manager : Add CancelAtxfer AMI action + + Add action to cancel feature attended transfer with AMI interface + + ASTERISK-27215 #close + + Change-Id: Iab8a81362b5a1757e2608f70b014ef863200cb42 + +2017-10-06 04:55 +0000 [6576e4320a] Daniel Tryba + + * res_pjsip_session: Prevent user=phone being added to anonimized URIs. + + Move ast_sip_add_usereqphone to be called after anonymization of URIs, + to prevent the user_eq_phone adding "user=phone" to URIs containing a + username that is not a phonenumber (RFC3261 19.1.1). An extra call to + ast_sip_add_usereqphone on the saved version before anonymization is + added to add user=phone" to the PAI. + + ASTERISK-27047 #close + + Change-Id: Ie5644bc66341b86dc08b1f7442210de2e6acdec6 + +2017-10-06 05:14 +0000 [a56316423f] Daniel Tryba + + * res_pjsip: Prevent "user=phone" being added multiple times to header + + ast_sip_add_usereqphone adds "user=phone" to the header every time is is + called without checking whether the param already exists. Preventing + this by searching to string representation of header for "user=phone". + + ASTERISK-26988 #close + + Change-Id: Ib84383b07254de357dc6a98d91fc1d2c2c3719e6 + +2017-10-05 18:12 +0000 [e5b9eb0460] Richard Mudgett + + * cdr.c: Defer misc checks. + + Try to defer some checks until needed in case there is an early exit. + + Change-Id: Ibc6b34c38a4f60ad4f9b67984b7d070a07257064 + +2017-10-06 20:48 +0000 [e8bde6916a] Seán C McCord + + * ari/bridge: Add mute, dtmf suppression controls + + Add bridge_features structure to bridge creation. Specifically, this + implements mute and DTMF suppression, but others should be able to be + easily added to the same structure. + + ASTERISK-27322 #close + Reported by: Darren Sessions + Sponsored by: AVOXI + + Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61 + +2017-10-11 07:03 +0000 [ab4d36533c] George Joseph + + * chan_vpb: Fix a gcc 7 out-of-bounds complaint + + chan_vpb was trying to use sizeof(*p->play_dtmf), where + p->play_dtmf is defined as char[16], to get the length of the array + but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf) + returns the size of the first array element, which is 1. gcc7 + validly complains because the context in which it's used could + cause an out-of-bounds condition. + + Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba + +2017-10-06 02:39 +0000 [be7da57546] Nathan Bruning + + * app_queue.c: clear moh field in init_queue + + ASTERISK-27301 #close + + Change-Id: Ic31361f34e2de3b6470e68fc37205a7711082eba + +2017-10-09 21:00 +0000 [b8dadccbe1] Corey Farrell + + * sorcery: Use ao2_weakproxy to hold list of instances. + + * Store weak proxy objects in instances container. + * Remove special unreference function and replace with macro that calls + ao2_cleanup. + * Add REF_DEBUG information to ast_sorcery_open. + + Change-Id: I5a150a4e13cee319d46b5a4654f95a4623a978f8 + +2017-10-09 21:55 +0000 [7774623804] Corey Farrell + + * named_locks: Use ao2_weakproxy_find. + + Change-Id: I0ce8a1b7101b6caac6a19f83a89f00eaba1e9d9c + +2017-10-09 17:51 +0000 [b058f8673a] Corey Farrell + + * astobj2: Add ao2_weakproxy_find function. + + This function finds a weak proxy in an ao2_container and returns the + real object associated with it. + + Change-Id: I9da822049747275f5961b5c0a7f14e87157d65d8 + +2017-10-10 15:09 +0000 [fd3101e8ad] Corey Farrell + + * astobj2: Run weakproxy callbacks outside of lock. + + Copy the list of weakproxy callbacks to temporary memory so they can be + run without holding the weakproxy lock. + + Change-Id: Ib167622a8a0f873fd73938f7611b2a5914308047 + +2017-10-10 12:01 +0000 [3ad7d2f36c] Sean Bright + + * app_originate: Set ORIGINATE_STATUS correctly on failure + + We were ignoring the return value from ast_pbx_outgoing_exten() and + ast_pbx_outgoing_app() which could fail before setting the reason code. + This resulted in failures being reported as success. + + ASTERISK-25266 #close + Reported by: Allen Ford + + Change-Id: Idf16237b7e41b527d2c69c865829128686beeb3b + +2017-10-03 15:16 +0000 [b1d9fc87bc] Torrey Searle + + * contrib/thirdparty/sip_to_pjsip: add additional flag mappings + + add mappings for udptl redundancy, rtptimeout, and debug flags + + Change-Id: Ie73cf5c83c05dee01eb9624ede76c1a30225d73a + +2017-10-02 16:46 +0000 [b0408d05c0] Richard Mudgett + + * cdr.c: Eliminated simple RAII_VAR usages. + + Change-Id: I150505db307249a962987e7b941bdd369bb91f35 + +2017-10-10 09:49 +0000 [11cefdf621] Tzafrir Cohen + + * cdr_mysql: avoid releasing a config string + + Fixes a memory corruption issue after a reload of cdr_mysql. + + Issue was accidentally included in 747beb1ed159f89a3b58742e4257740b3d6d6bba . + + ASTERISK-27270 #close + + Change-Id: I90b6a9d18710c0f9009466370bd5f4bac5d5d12e + +2017-10-10 07:42 +0000 [b228f5c5e6] Tzafrir Cohen + + * declare optional openssl dependencies in moduleinfo + + Declare optional openssl dependencies in: + * res_rtp_asterisk.c + * tcptls.c + + ASTERISK-27328 #close + + Change-Id: I2636f1c05b8104b4fe6f36cce0ebd9a98b9c78ab + +2017-10-09 22:51 +0000 [fae09c6676] Corey Farrell + + * res_pjproject: Fix cleanup of buildopts vector. + + ASTERISK-27306 + + Change-Id: I3bed0edf3f55b1d4adcbabb25ec14f11dc766c72 + +2017-10-03 16:09 +0000 [fdf9aacca3] Richard Mudgett + + * cdr.c: Replace redundant check with an ast_assert() + + The only caller of cdr_object_fn_table.process_party_b() explicitly does + the check before calling. + + Change-Id: Ib0c53cdf5048227842846e0df9d2c19117c45618 + +2017-10-02 17:41 +0000 [2e4b5fadbd] Richard Mudgett + + * cdr.c: Replace inlined code with ao2_t_replace() + + Change-Id: I9f424f5282ca7d833592f958d95f1b2bafb549b0 + +2017-09-29 12:07 +0000 [62980eedc3] Richard Mudgett + + * cdr.c: Use current ao2 flag names + + Change-Id: Ib59d7d2f2a4a822754628f2c48a308d6791a6e6e + +2017-09-29 12:31 +0000 [e769846f11] Richard Mudgett + + * cdr.h: Fix doxygen comments. + + * Also some misc formatting in cdr.c. + + Change-Id: Ied89a28802a662c37c43326a1aafdce596e0df4a + +2017-09-20 18:36 +0000 [fb19799b62] Richard Mudgett + + * res_pjsip_registrar.c: Update remove_existing AOR contact handling. + + When "rewrite_contact" is enabled, the "max_contacts" count option can + block re-registrations because the source port from the endpoint can be + random. When the re-registration is blocked, the endpoint may give up + re-registering and require manual intervention. + + * The "remove_existing" option now allows a registration to succeed by + displacing any existing contacts that now exceed the "max_contacts" count. + Any removed contacts are the next to expire. The behaviour change is + beneficial when "rewrite_contact" is enabled and "max_contacts" is greater + than one. The removed contact is likely the old contact created by + "rewrite_contact" that the device is refreshing. + + ASTERISK-27192 + + Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b + +2017-10-09 08:15 +0000 [ad38a55a2d] Sean Bright + + * res_config_sqlite: Don't enable SQLite CDRs when running 'make samples' + + Change-Id: I65a5190b2732b2246d67472db70dd37db64ddad4 + +2017-10-08 14:05 +0000 [a0a1f95abf] David Hajek + + * res/res_ari.c Fix: Memory leaks in ARI when using Content-Type: application/json + + ASTERISK-27305 + Reported by: David Hajek + Tested by: David Hajek + + Change-Id: Ife3e289062e6cf7d0e7d342dbf79ed96feff441e + +2017-10-08 09:11 +0000 [feeb0974eb] Alexander Traud + + * tcptls: Do not re-bind to wildcard on client creation. + + Since ASTERISK-26922, this issue affected only those chan_sip which were + * enabled for dual-stack (bindaddr=::), and + * enabled for TCP (tcpenable=yes) and/or TLS (tlsenable=yes), and + * tried to register and/or invite a IPv4-only service, + * via TCP and/or TLS. + Now, ast_tcptls_client_create does not re-bind to [::] anymore. + + ASTERISK-27324 #close + + Change-Id: I4b242837bdeb1ec7130dc82505c6180a946fd9b5 + +2017-10-07 15:47 +0000 [eb224fea5e] Corey Farrell + + * res_pjsip_session: Fix format_cap leak. + + ASTERISK-27306 + + Change-Id: I2c8d3fc148f9f53715c958314e1146f9611741f3 + +2017-10-06 10:51 +0000 [f4798faacc] Matt Jordan + + * res_corosync: Fix linking issue with Corosync 2.x + + At some point in time in the history of Corosync (certainly within the + 2.x branch), the corosync_cfg_state_track function was removed. + Unfortunately, the cfg library is only linked if this function is + present. Without the cfg library being linked to res_corosync, loading + of res_corosync will fail. + + This patch makes it so that detecting corosync's core libraries, + determined by the COROSYNC external library checks, links both the cpg + and cfg libraries with res_corosync. + + Change-Id: I674e9e1c8fea11c3bf81154aaa7c1fd43f945465 + +2017-10-05 16:26 +0000 [a68a91f722] Corey Farrell + + * res_pjsip: Fix leak of persistent endpoint references. + + Do not manually call sip_endpoint_apply_handler from load_all_endpoints. + This is not necessary and causes memory leaks. + + Additionally reinitialize persistent->aors when we reuse a persistent + object with a new endpoint. + + ASTERISK-27306 + + Change-Id: I59bbfc8da8a14d5f4af8c5bb1e71f8592ae823eb + +2017-10-05 17:59 +0000 [3bd00c4a7e] Corey Farrell + + * vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. + + Use temporary variable to prevent multiple evaluations of elem argument. + This resolves a memory leak in res_pjproject startup. + + ASTERISK-27317 #close + + Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d + +2017-10-05 15:54 +0000 [b35ac9e566] Corey Farrell + + * res_pjsip: Fix leak of fake_auth references. + + pjsip_distributor leaks references to fake_auth when the default realm + has not changed. + + ASTERISK-27306 + + Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202 + +2017-10-05 20:23 +0000 [0f3e725503] Corey Farrell + + * main/strings: Fix uninitialized value. + + ast_strings_match uses sscanf and checks for non-zero return to verify a + token was parsed. This is incorrect as sscanf returns EOF (-1) for errors. + + ASTERISK-27318 #close + + Change-Id: Ifcece92605f58116eff24c5a0a3b0ee08b3c87b1 + +2017-10-05 19:55 +0000 [0b6be1b2d4] Corey Farrell + + * res_sdp_translator_pjmedia: Fix test unregistration. + + ASTERISK-27306 + + Change-Id: Ib3ed47167cb697ab7bd0a56cab589893f491651b + +2017-10-02 07:48 +0000 [59b6e8467a] Daniel Tryba + + * res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacy + + Currently privacy requests are only granted if the Privacy header + value is exactly "id" (defined in RFC 3325). It ignores any other + possible value (or a combination there of). This patch reverses the + logic from testing for "id" to grant privacy, to testing for "none" and + granting privacy for any other value. "none" must not be used in + combination with any other value (RFC 3323 section 4.2). + + ASTERISK-27284 #close + + Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56 + +2017-10-04 10:59 +0000 [65399a5eda] Corey Farrell + + * res_pjsip: Add REF_DEBUG info to module references. + + This provides better information to REF_DEBUG log for troubleshooting + when the system is unable to unload res_pjsip.so during shutdown due to + module references. + + ASTERISK-27306 + + Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612 + +2017-10-04 10:46 +0000 [7d04544986] Corey Farrell + + * res_pjsip: Fix issues that prevented shutdown of modules. + + res_pjsip and res_pjsip_session had circular references, preventing both + modules from shutting down. + * Move session supplement registration to res_pjsip. + * Use create internal functions for use by pjsip_message_filter.c. + + ASTERISK-27306 + + Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b + +2017-09-28 02:56 +0000 [2301447a20] Benoît Dereck-Tricot + + * res_calendar_icalendar: Filter out occurrences superceded by another VEVENT + + When we are loading the calendars, we call libical's + icalcomponent_foreach_recurrence method for each VEVENT component that + we have in our calendar. + + That method has no knowledge concerning the existence of the other + VEVENT components and will feed our callback with all ocurrences + matching the requested time span. + + The occurrences generated by icalcomponent_foreach_recurrence while + expanding a recurring VEVENT's RRULE and RDATE properties can be + superceded by an other VEVENT sharing the same UID. + + I use an external iterator (in libical terminology) to avoid messing + with the internal ones from the calling function, and search for + VEVENTS which could supersede the current occurrence. + + The event which can invalidate this occurence needs to have: + + - the same UID as our recurrent component (comp) + - a RECURRENCE-ID property, which represents the start time of this + occurrence + + If one component is found, just clean and return. + + ASTERISK-27296 #close + Reported by: Benoît Dereck-Tricot + + Change-Id: I8587ae3eaa765af7cb21eda3b6bf84e8a1c87af8 + +2017-09-28 17:37 +0000 [b2dbfe23ef] Richard Mudgett + + * app_queue.c: Fix announcements when announce-to-first-user not enabled. + + The previous patch for ASTERISK-27216 made it so you wouldn't get any + position or periodic announcements unless you had announce-to-first-user + enabled. The announce-to-first-user feature was added by ASTERISK_21782 + as a result of the patch which introduced the redundant announcements that + ASTERISK-27216 removes. + + * By noting that the makeannouncement variable is used to suppresses the + first user announcement, we set its initial value to the + announce-to-first-user enable setting. + + ASTERISK-27216 + + Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a + +2017-09-21 14:43 +0000 [80097676e7] Richard Mudgett + + * heap.c: No need to calloc heap pointer array. + + Change-Id: I5ae2f316229f336eb90d99c7af7ed07a33097e68 + +2017-09-27 13:45 +0000 [d1de7948fe] George Joseph + + * logger: Bring back ability to turn debug on by source file + + Somewhere along the way we lost the ability to debug individual + source files. For modules, this wasn't a big deal but all the + source files in ./main are in the one "core" module so debugging + individual core capabilities was almost impossible. + + * Added a test to DEBUG_ATLEAST that also checks __FILE__ instead + of just module name. Any source file will work even if it's in + a module subdirectory. + + Change-Id: Icc0af41837f3b1679dec7af21fa32cd1f7469f6e + +2017-09-28 05:33 +0000 [f21408c866] Joshua Colp + + * res_stasis: Add 'video_sfu' as a requested bridge type. + + This change adds 'video_sfu' as a requested bridge type when + creating a bridge. By specifying this a mixing type bridge is + created that exchanges video in an SFU fashion. + + Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606 + +2017-09-27 11:16 +0000 [a6dc0527a2] Richard Mudgett + + * res_pjsip_outbound_publish.c: Fix misplaced parenthesis. + + The pjsip_publishc_init() call was referenced with a misplaced + parentheses. As a result, outbound publication messages went out with an + expiration of 1 second. + + ASTERISK-27298 + + Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08 + +2017-09-26 11:01 +0000 [61ea872233] George Joseph + + * pjsip_message_filter: Fix regression causing bad contact address + + The "res_pjsip: Filter out non SIP(S) requests" commit moved the + filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER + in order to filter out incoming bad uri schemes as early as possible. + Since the change affected outgoing messages as well and the TRANSPORT + layer is the last to be run on outgoing messages, we were overwriting + the setting of external_signaling_address (which is set earlier by + res_pjsip_nat) with an internal address. + + * pjsip_message_filter now registers itself as a pjproject module + twice. Once in the TSX layer for the outgoing messages (as it was + originally), then a second time in the TRANSPORT layer for the + incoming messages to catch the invalid uri schemes. + + ASTERISK-27295 + Reported by: Sean Bright + + Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c + +2017-09-13 21:31 +0000 [9d65057cdf] Richard Mudgett + + * res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential. + + The bridge_p2p_rtp_write() has potential reentrancy problems. + + * Accessing the bridged RTP members must be done with the instance1 lock + held. The DTMF and asymmetric codec checks must be split to be done with + the correct RTP instance struct locked. i.e., They must be done when + working on the appropriate side of the point to point bridge. + + * Forcing the RTP mark bit was referencing the wrong side of the point to + point bridge. The set mark bit is used everywhere else to set the mark + bit when sending not receiving. + + The patches for ASTERISK_26745 and ASTERISK_27158 did not take into + account that not everything carried by RTP uses a codec. The telephony + DTMF events are not exchanged with a codec. As a result when + RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is + enabled, the DTMF digits would always get passed to the core even though + the local native RTP bridge is active, and the DTMF digits would go out + using the wrong SSRC id. + + * Add protection for non-format payload types like DTMF when updating the + lastrxformat and lasttxformat. Also protect against non-format payload + types when checking for asymmetric codecs. + + ASTERISK-27292 + + Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186 + +2017-09-26 10:55 +0000 [c9e972a26a] Sean Bright + + * res_rtp_asterisk: Trim trailing byte off of SDES packet + + This could have been fixed by subtracting 1 from the final value of + 'len' but the way the packet was being constructed was confusing so I + took the opportunity to (I think) make it more clear. + + We were sending 1 extra byte at the end of the SDES RTCP packet which + caused Chrome to complain (in its debug log): + + Too little data (1 byte) remaining in buffer to parse + RTCP header (4 bytes). + + We now send the correct number of bytes. + + Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b + +2017-09-25 13:00 +0000 [721947ebae] Sean Bright + + * webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file + + If using a legitimate certificate from a trusted certificate authority, + you don't need to provide CA file. + + Change-Id: I8623973b4209b44889243716d7880274caed8a6d + +2017-09-25 13:09 +0000 [0cbeaa5589] Sean Bright + + * pjproject: Patch to correct STUN FINGERPRINT usage + + Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6 + +2017-09-25 12:30 +0000 [b74cbadd05] Kevin Harwell + + * res_pjsip_session: outgoing call did not offer all configured codecs + + For some scenarios when an outgoing call was made only a subset of the + configured codecs were offered. If the codecs being offered happened to + not have a codec supported by the phone then the call would fail. + + For instance Alice and Bob both are configured in Asterisk for g722 and ulaw( + allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's + only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with + Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains + g722 and not both g722 and ulaw, so the call ends. + + This patch makes it so all the audio codecs configured on the endpoint always + get sent, and not just a subset. However priority is given to those codecs that + are compatible with the "other side". + + ASTERISK-27259 #close + + Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696 + +2017-09-25 10:59 +0000 [08e67f814b] Richard Mudgett + + * channel.c: Fix invalid reference in conditionaled out code. + + ASTERISK-27289 + + Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c + +2017-09-25 07:25 +0000 [4275ca16a1] George Joseph + + * build: A few gcc 7 error fixes + + Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec + +2017-09-15 02:59 +0000 [c3c73b3511] Stefan Engström + + * app_queue: Only do announcement logic between ringing cycles + + This patch reverts the change by patch 2263 from old reviewboard. + Note that reverting that 2263-patch still preserves the behaviour that + the commit log of the 2263-patch claimed to add. The reason for this is: + + The function wait_for_answer is only called from try_calling which + in turn is only called from the main for loop in queue_exec, and + earlier in that loop we already check the things that's removed by + this patch. There's no need to check those things twice each loop + iteration, and I think the proper place to check it is before each + ringing cycle. By checking it in wait_for_answer, you allow the issue + explained in the jira - that the head caller hears announcements while + the agents' sip phones are actively ringing. + + Reported-by: Stefan Engström + Tested-by: Stefan Engström + ASTERISK-27216 #close + + Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0 + +2017-09-23 12:32 +0000 [0fad11f21c] Sean Bright + + * app_stream_echo: Don't echo declined streams + + Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic + Edition after accepting the audio request but declining the video one. + + Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c + +2017-09-22 17:49 +0000 [601e0c563f] Joshua Colp + + * res_pjsip_session: Reduce (and improve) SDP renegotiation. + + When pruning a request to change the topology of a channel be + more intelligent about the resulting topology that is actually + used for SDP renegotiation. + + In a case where a stream has not already been negotiated we + don't need to renegotiate and offer a declined stream. This can + occur if something in Asterisk (such as ConfBridge) requests + to add video to a PJSIP channel that has no video codecs configured. + In this case since the stream did not already exist we can safely + remove the stream from the requested topology, resulting in no + renegotiation occurring. + + In a case where a renegotiation is requested with a codec that is + not supported we can reuse the formats of the existing stream if + it exists to ensure that the stream continues to flow, instead of + removing it. + + Change-Id: I636540798d55922377318fe619c510fb6ed125fb + +2017-09-22 15:29 +0000 [36690c26f8] Kevin Harwell + + * res_pjsip_session: Don't end session when receiving a 500 on a reinvite + + During a reinvite, if a remote endpoint error occurs and it returns a 500 the + session would end. This patch makes it so the session is not terminated, but + continues as it was. + + The reason for this is because some endpoints may send non session terminating + "server errors" like a failed codec negotiation. So in this case instead of + ending the call it can hopefully continue. In the case of a real server error + the session is already "doomed", will be known soon enough and appropriately + ended by Asterisk later. + + Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5 + +2017-09-22 10:02 +0000 [ebd0a4bebf] Sean Bright + + * res_pjsip: Use ast_sip_is_content_type() where appropriate + + Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d + +2017-09-21 09:47 +0000 [6c0e13da22] George Joseph + + * res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint + + When an INVITE came in with both audio and video streams but there + were no audio codecs defined for the endpoint, we weren't declining + the audio stream. Since it's usually the first/transport stream, + when the video stream was processed and tried to use the transport, + it was empty and caused a crash. We now decline the the stream if + there are no matching codecs so when the video stream is processed, + it's now the first/transport stream and processes normally. + + Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692 + +2017-09-19 14:28 +0000 [7c93982e9d] Richard Mudgett + + * res_rtp_asterisk.c: Fix bundled SSRC handling. + + Assertions in the v15+ AST-2017-008 patches found that we were not + handling the case if the incoming SDP did not specify the required SSRC + attributes for bundled to work. + + * Be strict on matching SSRC for bundled instances including the parent + instance. If the SSRC doesn't match then discard the packet. Bundled has + to tell us in the SDP signaling what SSRC to expect. Otherwise, we will + not know how to find the bundled instance structure. + + Change-Id: I152830bbff71c662408909042068fada39e617f9 + +2017-09-16 09:19 +0000 [f2985e3106] Joshua Colp + + * bridge: Change participant SFU streams when source streams change. + + Some endpoints do not like a stream being reused for a new + media stream. The frame/jitterbuffer can rely on underlying + attributes of the media stream in order to order the packets. + When a new stream takes its place without any notice the + buffer can get confused and the media ends up getting dropped. + + This change uses the SSRC change to determine that a new source + is reusing an existing stream and then bridge_softmix renegotiates + each participant such that they see a new media stream. This + causes the frame/jitterbuffer to start fresh and work as expected. + + ASTERISK-27277 + + Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07 + +2017-09-20 10:45 +0000 [971548405b] George Joseph + + * res_pjsip_session: Change some asserts to warning/debug messages + + There was an issue reported where an SDP received on a 183 Session + Progress message caused a crash because the pending streams had + already been processed when the OK was received. In that case the + pending topology was legitimately NULL. There was an assert for an + incorrect number of streams in the topology but not one for + topology being NULL. In any case, if you're not in dev-mode the + asserts don't do anything and since the scenario is legit, the + asserts weren't appropriate anyway. + + * Changed several asserts to warning or debug messages and return + codes as appropriate. + + ASTERISK-27264 + Reported by: Daniel Heckl + + Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848 + +2017-09-19 05:22 +0000 [cad68137a7] Rodrigo Ramírez Norambuena + + * res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1 + + In PostgreSQL 9.1 the backslash are string literals and not the escape + of characters. + + In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the + support for old version of Postgresql than 9.1 was dropped. The sentence + before make was "ESCAPE '\'" but in version before than 9.1 need it to be + as follow "ESCAPE '\\'". + + ASTERISK-27283 + + Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949 + +2017-09-15 09:43 +0000 [e666051d79] Ben Ford + + * res_pjsip_session: Check for removed stream state. + + When a sip session is refreshed, the stream topology is looped + through, checking each stream for compatible formats. This would + cause a crash if the stream state was AST_STREAM_STATE_REMOVED, + since the formats would never be set for this stream, causing + a NULL value to be returned from ast_stream_get_formats. This + commit adds a check for streams with removed states. + + Also removed a stray semicolon. + + Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42 + +2017-09-19 05:44 +0000 [b6aa728a58] George Joseph + + * chan_pjsip: Ignore AST_CONTROL_STREAM_TOPOLOGY_CHANGED for now + + chan_pjsip_indicate was missing a case for the recently added + AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an + error and causing the call to be hung up instead of just ignoring + it. + + ASTERISK-27260 + Reported by: Daniel Heckl + + Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80 + +2017-09-07 04:41 +0000 [6b7d5671d1] Jean Aunis + + * bridge : Fix one-way direct-media when early bridging with native_rtp + + When two channels were early bridged in a native_rtp bridge, the RTP description + on one side was not updated when the other side answered. + This patch forbids non-answered channels to enter a native_rtp bridge, and + triggers a bridge reconfiguration when an ANSWER frame is received. + + ASTERISK-27257 + + Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df + +2017-09-18 09:51 +0000 [1e4c1cec7f] Alexander Traud + + * res_srtp: lower log level of auth failures + + Previously, sRTP authentication failures were reported on log level WARNING. + When such failures happen, each RT(C)P packet is affected, spamming the log. + Now, those failures are reported at log level VERBOSE 2. Furthermore, the + amount is further reduced (previously all two seconds, now all three seconds). + Additionally, the new log entry informs whether media (RTP) or statistics (RTCP) + are affected. + + ASTERISK-16898 #close + + Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0 + +2017-09-19 10:38 +0000 [b748038230] George Joseph + + * res_pjsip_pubsub: Check for Content-Type header in rx_notify_request + + pubsub_on_rx_notify_request wasn't checking for a null + Content-Type header before checking that it was + application/simple-message-summary. + + ASTERISK-27279 + Reported by: Ross Beer + + Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52 + +2017-09-19 09:34 +0000 [a5f1d58fe1] David J. Pryke + + * chan_sip: Expose read-only access to the full SIP INVITE Request-URI + + Provide a way to get the contents of the the Request URI from the initial SIP + INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}") + + ASTERISK-27278 + Reported by: David J. Pryke + Tested by: David J. Pryke + + Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e + +2017-09-19 07:53 +0000 [6fd3db51e8] Joshua Colp + + * app_confbridge: Only create a channel that records audio. + + This change makes it so that the conference recorder channel + that is created only contains audio formats and an audio stream. + This is because the underlying application used by ConfBridge to + record, MixMonitor, only allows recording audio. + + Having additional streams (and in particular a video stream) can + result in clients needlessly renegotiating to add a video stream + that will never receive video. + + Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0 + +2017-09-19 06:34 +0000 [56f0d5fc0f] Rodrigo Ramírez Norambuena + + * res_config_pgsql: Add missing \n in debug log and update copyright year + + Change-Id: I4ba338ecbdecc6a814a902eddc4121c8ef3cda58 + +2017-09-13 14:14 +0000 [55567ee1d8] Sean Bright + + * res_calendar: Plug memory leak and micro-optimization + + ast_variables_destroy is NULL safe, so there is no need to check its + argument before passing it. + + ASTERISK-25524 #close + Reported by: Jesper + + Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b + +2017-09-13 03:46 +0000 [1199927fc0] alex + + * cdr_mysql.c: Apply cdrzone to start and answer + + Change-Id: I7de0a5adc89824a5f2b696fc22c80fc22dff36b0 + +2017-08-25 17:01 +0000 [087f667ab1] Richard Mudgett + + * AST-2017-008: Improve RTP and RTCP packet processing. + + Validate RTCP packets before processing them. + + * Validate that the received packet is of a minimum length and apply the + RFC3550 RTCP packet validation checks. + + * Fixed potentially reading garbage beyond the received RTCP record data. + + * Fixed rtp->themssrc only being set once when the remote could change + the SSRC. We would effectively stop handling the RTCP statistic records. + + * Fixed rtp->themssrc to not treat a zero value as special by adding + rtp->themssrc_valid to indicate if rtp->themssrc is available. + + ASTERISK-27274 + + Make strict RTP learning more flexible. + + Direct media can cause strict RTP to attempt to learn a remote address + again before it has had a chance to learn the remote address the first + time. Because of the rapid relearn requests, strict RTP could latch onto + the first remote address and fail to latch onto the direct media remote + address. As a result, you have one way audio until the call is placed on + and off hold. + + The new algorithm learns remote addresses for a set time (1.5 seconds) + before locking the remote address. In addition, we must see a configured + number of remote packets from the same address in a row before switching. + + * Fixed strict RTP learning from always accepting the first new address + packet as the new stream. + + * Fixed strict RTP to initialize the expected sequence number with the + last received sequence number instead of the last transmitted sequence + number. + + * Fixed the predicted next sequence number calculation in + rtp_learning_rtp_seq_update() to handle overflow. + + ASTERISK-27252 + + Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c + +2017-09-13 16:23 +0000 [d178f497d2] George Joseph + + * res_pjsip: Filter out non SIP(S) requests + + Incoming requests with non sip(s) URIs in the Request, To, From + or Contact URIs are now rejected with + PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in + pjsip_message_filter (formerly pjsip_message_ip_updater) and is + done at pjproject's "TRANSPORT" layer before a request can even + reach the distributor. + + URIs read by res_pjsip_outbound_publish from pjsip.conf are now + also checked for both length and sip(s) scheme. Those URIs read + by outbound registration and aor were already being checked for + scheme but their error messages needed to be updated to include + scheme failure as well as length failure. + + Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460 + +2017-09-14 07:54 +0000 [01f2220bec] Joshua Colp + + * tcptls: Change error message to debug. + + The Websocket implementation will steal the underlying stream of + TCP/TLS sessions. This results in an error message being output + about a stream not being present when in reality this is actually + fine. + + This change moves it to a debug message instead. + + Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094 + +2017-09-13 14:08 +0000 [d8112cd98b] Sean Bright + + * res_calendar: Various fixes + + * The way that we were looking at XML elements for CalDAV was extremely + fragile, so use SAX2 for increased robustness. + + * Don't complain about a 'channel' not be specified if autoreminder is + not set. Assume that if 'channel' is not set, we don't want to be + notified. + + * Fix some truncated CLI output in 'calendar show calendar' and make the + 'Autoreminder' description a bit more clear + + ASTERISK-24588 #close + Reported by: Stefan Gofferje + + ASTERISK-25523 #close + Reported by: Jesper + + Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c + +2017-09-13 09:38 +0000 [eec0396395] Sean Bright + + * chan_rtp: Use μ-law by default instead of signed linear + + Multicast/Unicast RTP do not use SDP so we need to use a format that + cleanly maps to one of the static RTP payload types. Without this + change, an Originate to a Multicast or Unicast channel without a format + specified would produce no audio on the receiving device. + + ASTERISK-21399 #close + Reported by: Tzafrir Cohen + + Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3 + +2017-09-11 05:46 +0000 [446d48fd49] George Joseph + + * res_pjsip: Add handling for incoming unsolicited MWI NOTIFY + + A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to + receive unsolicited MWI NOTIFY requests and make them available to + other modules via the stasis message bus. + + res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" + that parses a simple-message-summary body and, if + endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state + with the voice-message counts from the message. + + Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c + +2017-09-08 21:41 +0000 [4889574ff5] Richard Mudgett + + * res_rtp_asterisk.c: Add doxygen to RTCP payload types. + + Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998 + +2017-09-11 05:52 +0000 [f9bad3bd61] George Joseph + + * alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode + + The downgrade function was missing "_v2" at the end of the + alter column type. + + Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e + +2017-09-10 06:17 +0000 [680aba21ec] Walter Doekes + + * res/res_pjsip: Fix localnet checks in pjsip, part 2. + + In 45744fc53, I mistakenly broke SDP media address rewriting by + misinterpreting which address was checked in the localnet comparison. + + Instead of checking the remote peer address to decide whether we need + media address rewriting, we check our local media address: if it's + local, then we rewrite. This feels awkward, but works and even made + directmedia work properly if you set local_net. (For the record: for + local peers, the SDP media rewrite code is not called, so the + comparison does no harm there.) + + ASTERISK-27248 #close + + Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f + +2017-09-08 21:19 +0000 [c8d53a1638] Rodrigo Ramírez Norambuena + + * cdr_pgsql: Refactor magic number by definition for version + + Change-Id: I43f25976aa3069793ddbe0086833965a6fb0a518 + +2017-09-05 11:13 +0000 [e9a81157ac] Florian Floimair + + * alembic: Add support for MS-SQL + + MS-SQL has no native Enum-type support and therefore + needs to work with constraints. + Since these constraints need unique names the suggested approach + referenced in the following alembic documentation has been applied: + http://bit.ly/2x9r8pb + + ASTERISK-27255 #close + + Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95 + +2017-09-05 07:31 +0000 [525f84bb35] Jacek Konieczny + + * func_cdr: honour 'u' flag on dummy channel + + Fixes ${CDR(...,u)} when used in cdr_custom.conf + + ASTERISK-27165 #close + + Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83 + +2017-09-06 10:50 +0000 [2b3f903e6f] Sean Bright + + * app_waitforsilence: Cleanup & don't treat missing frames as 'noise' + + * WaitForSilence completes successfully if it receives no media in the + specified timeout, but when acting as WaitForNoise that logic needs + to be reversed. + + * Use standard argument parsing macros and add some error checking for + invalid values. + + * The documentation indicated that the first argument to both + WaitForSilence and WaitForNoise was required when it was not. Update + the documentation to reflect that. + + * Wrap up some behavior in structs to avoid boolean checks all over the + place. + + ASTERISK-24066 #close + Reported by: M vd S + + Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9 + +2017-09-06 16:05 +0000 [5553644284] Scott Griepentrog + + * chan_sip: when getting sip pvt return failure if not found + + In handle_request_invite, when processing a pickup, a call + is made to get_sip_pvt_from_replaces to locate the pvt for + the subscription. The pvt is assumed to be valid when zero + is returned indicating no error, and is dereferenced which + can cause a crash if it was not found. + + This change checks the not found case and returns -1 which + allows the calling code to fail appropriately. + + ASTERISK-27217 #close + Reported-by: Bryan Walters + + Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612 + +2017-09-06 13:38 +0000 [23571f31ac] Richard Mudgett + + * stasis/control.c: Fix set_interval_hook() ref leak. + + Change-Id: Ia0edb7dc0dbbb879c079ff7000f1b722d86ce7dc + +2017-09-01 05:17 +0000 [94091c7b96] George Joseph + + * stasis/control: Fix possible deadlock with swap channel + + If an error occurs during a bridge impart it's possible that + the "bridge_after" callback might try to run before + control_swap_channel_in_bridge has been signalled to continue. + Since control_swap_channel_in_bridge is holding the control lock + and the callback needs it, a deadlock will occur. + + * control_swap_channel_in_bridge now only holds the control + lock while it's actually modifying the control structure and + releases it while the bridge impart is running. + * bridge_after_cb is now tolerant of impart failures. + + Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3 + +2017-09-06 05:23 +0000 [67a2ca31f5] Vitezslav Novy + + * chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE + + If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE + to both parties to set up media path directly between the endpoints. + In this reINVITE msg SDP origin line (o=) contains IP address of endpoint + instead of IP of asterisk. This behavior violates RFC3264, sec 8: + "When issuing an offer that modifies the session, + the "o=" line of the new SDP MUST be identical to that in the + previous SDP, except that the version in the origin field MUST + increment by one from the previous SDP." + This patch assures IP address of Asterisk is always sent in + SDP origin line. + + ASTERISK-17540 + Reported by: saghul + + Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e + +2017-09-06 07:54 +0000 [0cbb17ce8f] George Joseph + + * alembic: Fix enum creation for dtls_fingerprint + + Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db + +2017-09-05 11:08 +0000 [a133c5cc53] Florian Floimair + + * alembic: fix erroneous commit for add_prune_on_boot + + Added include for postgresql ENUM type and + redefined values in the same way as in the + other migration scripts. + + ASTERISK-27254 #close + + Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa + +2017-09-06 03:02 +0000 [2d395793b7] Alexander Traud + + * res_srtp: Add support for libsrtp2.1. + + Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro + SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with + previous versions of libSRTP, this change allows libSRTP 2.1.x as well. + + ASTERISK-27253 #close + + Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315 + +2017-09-05 09:35 +0000 [bfc29de3ea] Ben Ford + + * chan_pjsip: Suppress frame warnings. + + When rtp_keepalive is on for a PJSIP endpoint dialing to another + Asterisk instance also using PJSIP, Asterisk will continue to print + warning messages about not being able to send frames of a certain + type. This suppresses that warning message. + + Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67 + +2017-09-05 10:05 +0000 [c3a6c8fd2d] Sean Bright + + * formats: Restore previous fread() behavior + + Some formats are able to handle short reads while others are not, so + restore the previous behavior for the format modules so that we don't + have spurious errors when playing back files. + + ASTERISK-27232 #close + Reported by: Jens T. + + Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300 + +2017-09-05 09:16 +0000 [f856d9b42b] Walter Doekes + + * res/res_pjsip: Standardize/fix localnet checks across pjsip. + + In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was + confusion about whether the transport_state->localnet ACL has ALLOW or + DENY semantics. + + For the record: the localnet has DENY semantics, meaning that "not in + the list" means ALLOW, and the local nets are in the list. + + Therefore, checks like this look wrong, but are right: + + /* See if where we are sending this request is local or not, and if + not that we can get a Contact URI to modify */ + if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { + ast_debug(5, "Request is being sent to local address, " + "skipping NAT manipulation\n"); + + (In the list == localnet == DENY == skip NAT manipulation.) + + And conversely, other checks that looked right, were wrong. + + This change adds two macro's to reduce the confusion and uses those + instead: + + ast_sip_transport_is_nonlocal(transport_state, addr) + ast_sip_transport_is_local(transport_state, addr) + + ASTERISK-27248 #close + + Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934 + +2017-09-05 08:39 +0000 [68bcfccd52] Joshua Colp + + * res_pjsip_session: Preserve stream name during renegotiation. + + Stream names within Asterisk can have meaning so when an externally + initiated renegotiation occurs we need to preserve the name of + the stream if it already exists. + + Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596 + +2017-09-05 07:50 +0000 [0ec95515f3] George Joseph + + * res_calendar*, res_smdi: Move to "extended" support + + Change-Id: I31eee8be30c6b0fc3dadb31111dd47742da8892d + +2017-09-05 05:23 +0000 [9b3f6d26bd] George Joseph + + * res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel + + t38_reinvite_response_cb can get called by res_pjsip_session's + session_inv_on_tsx_state_changed in situations where session->channel + is NULL. If it is, the ast_log warning segfaults because it tries + to get the channel name from a NULL channel. + + * Check session->channel and print "unknown channel" when it's NULL. + + ASTERISK-27236 + Reported by: Ross Beer + + Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7 + +2017-09-01 16:17 +0000 [60b44d1e38] Sean Bright + + * rtp_engine: Prevent possible double free with DTLS config + + ASTERISK-27225 #close + Reported by: Richard Kenner + + Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89 + +2017-09-01 13:15 +0000 [ef8eb9d11b] Sean Bright + + * chan_ooh323: Fix confusing indentation warning + + ASTERISK-27177 #close + Reported by: Tzafrir Cohen + + Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97 + +2017-09-01 09:51 +0000 [1bdbefbe76] Sean Bright + + * app_directory: Handle a NULL mailbox without crashing + + ASTERISK-27241 #close + Reported by: David Moore + + Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6 + +2017-07-24 10:48 +0000 [f78f5278ff] George Joseph + + * pjsip_message_ip_updater: Fix issue handling "tel" URIs + + sanitize_tdata was assuming all URIs were SIP URIs so when a non + SIP uri was in the From, To or Contact headers, the unconditional + cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused + a segfault when trying to access uri->other_param. + + * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) + checks before attempting to cast or use the returned uri. + + ASTERISK-27152 + Reported-by: Ross Beer + + Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f + +2017-07-01 19:24 +0000 [1bf3dfffd7] Corey Farrell + + * AST-2017-006: Fix app_minivm application MinivmNotify command injection + + An admin can configure app_minivm with an externnotify program to be run + when a voicemail is received. The app_minivm application MinivmNotify + uses ast_safe_system() for this purpose which is vulnerable to command + injection since the Caller-ID name and number values given to externnotify + can come from an external untrusted source. + + * Add ast_safe_execvp() function. This gives modules the ability to run + external commands with greater safety compared to ast_safe_system(). + Specifically when some parameters are filled by untrusted sources the new + function does not allow malicious input to break argument encoding. This + may be of particular concern where CALLERID(name) or CALLERID(num) may be + used as a parameter to a script run by ast_safe_system() which could + potentially allow arbitrary command execution. + + * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() + instead of ast_safe_system() to avoid command injection. + + * Document code injection potential from untrusted data sources for other + shell commands that are under user control. + + ASTERISK-27103 + + Change-Id: I7552472247a84cde24e1358aaf64af160107aef1 + +2017-05-22 10:36 +0000 [7f2a60fb38] Joshua Colp + + * res_rtp_asterisk: Only learn a new source in learn state. + + This change moves the logic which learns a new source address + for RTP so it only occurs in the learning state. The learning + state is entered on initial allocation of RTP or if we are + told that the remote address for the media has changed. While + in the learning state if we continue to receive media from + the original source we restart the learning process. It is + only once we receive a sufficient number of RTP packets from + the new source that we will switch to it. Once this is done + the closed state is entered where all packets that do not + originate from the expected source are dropped. + + The learning process has also been improved to take into + account the time between received packets so a flood of them + while in the learning state does not cause media to be switched. + + Finally RTCP now drops packets which are not for the learned + SSRC if strict RTP is enabled. + + ASTERISK-27013 + + Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c + +2017-08-30 07:28 +0000 [5ba82cedc6] Joshua Colp + + * res_rtp_asterisk: Allow remote SSRC to change on an RTP instance. + + When SDP renegotiation occurs it is possible for an RTP + instance to be reused for a new stream, resulting in the remote + SSRC changing if it is part of a bundle group. This change + allows this and updates its mapping in the current bundle + group. + + ASTERISK-27231 + + Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490 + +2017-08-25 21:06 +0000 [71be8d5bbe] Andre Nazario + + * chan_pjsip: Add tag info in CHANNEL function + + Create local_tag and remote_tag in CHANNEL info to get tag from From and + To headers of a SIP dialog. + + ASTERISK-27220 + + Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524 + +2017-08-29 14:22 +0000 [4650fc477a] Richard Mudgett + + * bridge_native_rtp.c: Fixup native_rtp_framehook() + + * Fix framehook to test frame type for control frame. + * Made framehook exit early if frame type is not a control frame. + * Eliminated RAII_VAR in framehook. + * Use switch instead of else-if ladder for control frame handling. + + Change-Id: Ia555fc3600bd85470e3c0141147dbe3ad07c1d18 + +2017-08-29 09:26 +0000 [06cc5ae9ff] Sean Bright + + * confbridge: Handle user hangup during name recording + + This prevents orphaned CBAnn channels from getting stuck in the bridge. + + ASTERISK-26994 #close + Reported by: James Terhune + + Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457 + +2017-08-24 11:45 +0000 [9a9589e8e1] Joshua Colp + + * core: Reduce video update queueing. + + A video update frame is used to indicate that a channel + with video negotiated should provide a full frame so the + decoder decoding the stream is able to do so. In situations + where a queue is used to store frames it makes no sense + for the queue to contain multiple video update frames. One + is sufficient to have a full frame be sent. + + ASTERISK-27222 + + Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7 + +2017-08-25 13:44 +0000 [da13cdb9e7] Sean Bright + + * voicemail: Fix various abuses of mkstemp + + mkstemp() returns a unique filename, but appending an extension to that + filename does not guarantee uniqueness. Instead, use mkdtemp() and we + can put whatever extension we want on the files that we create inside + the directory. + + In the case of app_minivm, we also now properly clean up any temporary + files that we create. + + ASTERISK-20858 #close + Reported by: Walter Doekes + + Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43 + +2017-08-25 12:20 +0000 [43670e471f] Sean Bright + + * app_record: Resolve some absolute vs. relative filename bugs + + If the Record() application is called with a relative filename that + includes directories, we were not properly creating the intermediate + directories and Record() would fail. + + Secondarily, updated the documentation for RECORDED_FILE to mention + that it does not include a filename extension. + + Finally, rewrote the '%d' functionality to be a bit more straight + forward and less noisy. + + ASTERISK-16777 #close + Reported by: klaus3000 + + Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2 + +2017-08-23 10:01 +0000 [2ee644aacf] Florian Floimair + + * alembic: Add dtls_fingerprint column in ps_endpoints table + + The ps_endpoints table was missing the dtls_fingerprint column + introduced with commit adba2a8d7fd. + + ASTERISK-27168 #close + + Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd + +2017-08-21 04:28 +0000 [33a648d4c6] Torrey Searle + + * res/res_pjsip_session: allow SDP answer to be regenerated + + If an SDP answer hasn't been sent yet, it's legal to change it. + This is required for PJSIP_DTMF_MODE to work correctly, and can + also have use in the future for updating codecs too. + + ASTERISK-27209 #close + + Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1 + +2017-08-24 09:42 +0000 [02f95d290f] Sean Bright + + * app_queue: Evaluate realtime queues when running dialplan functions + + ASTERISK-19103 #close + Reported by: Jim Van Meggelen + + Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b + +2017-08-23 09:19 +0000 [b1097be134] Eelco Brolman (License 6442) + + * app_voicemail: Honor escape digits in "greeting only" mode + + ASTERISK-21241 #close + Reported by: Eelco Brolman + Patches: + Patch uploaded by Eelco Brolman (License 6442) + + Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe + +2017-08-24 08:35 +0000 [7937d5b8b3] Sean Bright + + * res_smdi: Clean up memory leak + + Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8 + +2017-08-18 17:37 +0000 [f2c14f00b8] Richard Mudgett + + * res_pjsip_session.c: Fix crash when declining an active stream. + + If a previously active stream is declined we could crash because the + channel's thread is still using the stream while we are updating the + topology in the serializer thread. + + * Defer removing any declined stream's handler until we have blocked the + channel's thread with the channel lock. + + ASTERISK-27212 + + Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420 + +2017-08-16 17:50 +0000 [17976d1b4e] Richard Mudgett + + * bridge_channel.c: Fix FRACK when mapping frames to the bridge. + + * Add protection checks when mapping streams to the bridge. The channel + and bridge may be in the process of updating the stream mapping when a + media frame comes in so we may not be able to map the frame at the time. + + * We need to map the streams to the bridge's stream numbers right before + they are written into the bridge. That way we don't have to keep + locking/unlocking the bridge and we won't have any synchronization + problems before the frames actually go into the bridge. + + * Protect the deferred queue with the bridge_channel lock. + + ASTERISK-27212 + + Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a + +2017-08-11 16:31 +0000 [9c70c88369] Richard Mudgett + + * channel: Fix topology API locking. + + * ast_channel_request_stream_topology_change() must not be called with any + channel locks held. + + * ast_channel_stream_topology_changed() must be called with only the + passed channel lock held. + + ASTERISK-27212 + + Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691 + +2017-08-16 15:22 +0000 [6ad8249233] Richard Mudgett + + * bridge: Fix softmix bridge deadlock. + + * Fix deadlock in + bridge_softmix.c:softmix_bridge_stream_topology_changed() between + bridge_channel and channel locks. + + * The new bridge technology topology change callbacks must be called with + the bridge locked. The callback references the bridge channel list, the + bridge technology could change, and the bridge stream mapping is updated. + + ASTERISK-27212 + + Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be + +2017-08-14 12:20 +0000 [850a3fd017] Richard Mudgett + + * chan_pjsip.c: Fix topology refresh response code accuracy. + + There are other 1xx and 2xx codes than 100 and 200 respectively. + + Change-Id: I680db0997343256add1478714f5bf5b5569aee17 + +2017-08-11 17:06 +0000 [87c7a1c79c] Richard Mudgett + + * bridge_softmix.c: Restored softmix_bridge_leave() shortcut exit. + + Change-Id: I13026cd90954e0265eab94a0faf635a3e11f0e35 + +2017-08-17 17:07 +0000 [5bbf7b2aad] Richard Mudgett + + * app_confbridge: Document sfu video_mode value. + + Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204 + +2017-08-17 17:06 +0000 [f96536b1ea] Richard Mudgett + + * confbridge.h: Fix doxygen comments. + + Change-Id: I16133166a85fdb557c66ffcbfe8128d0b4725b0e + +2017-08-11 11:40 +0000 [946ef2d711] Richard Mudgett + + * bridge_softmix.c: Remove always true test. + + Change-Id: I26238df2ff0d0f6dfe95c3aa35da588f1ee71727 + +2017-08-17 16:46 +0000 [22af5e3784] Sungtae Kim + + * app_queue: Fix initial hold time queue statistic + + Fixed to use correct initial value and fixed to use the + correct queue info to check the first value. + + ASTERISK-27204 + + Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73 + +2017-08-20 08:15 +0000 [83b81d1f8d] Michael Kuron + + * res_xmpp: fix inverted return code check in OAuth + + fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon + success and -1 if the function is not available. + This commit inverts the return code check so that an error is printed if the + module is not loaded and not if it is loaded. + + ASTERISK-27207 #close + + Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb + +2017-08-17 12:00 +0000 [667986d875] Sean Bright + + * res_calendar_icalendar: Properly handle recurring events + + When looking for recurring events, use the correct end time based on the + configured 'timeframe.' + + ASTERISK-27174 #close + Reported by: Mark Thompson + + Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef + +2017-08-16 15:43 +0000 [0e777258be] George Joseph + + * Fix downloader not working with curl + + The codec/dpma downloader wasn't handling curl correctly. The logic + that transforms makeopts into a bash-sourceable file wasn't + handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was + looking for an 'or' command. + + That logic has been eliminated. Instead of trying to transform + and source makeopts, the downloader now calls a make scriptlet + to print the value of a specific variable. This way, make handles + the ors (or any other make construct that happens to creep into + that file). + + ASTERISK-27202 + Reported by: Sean McCord + + Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99 + +2017-08-15 13:12 +0000 [e4e2e53c8a] Kevin Harwell + + * manager: hook event is not being raised + + When the iostream code went in it introduced a conditional that made it so the + hook event was not being raised even if a hook is present. This patch adds a + check to see if a hook is present in astman_append. If so then call into the + send_string function, which in turn raises the even for specified hook. + + Also updated the ami hooks unit test, so the test could be automated. + + ASTERISK-27200 #close + + Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36 + +2017-08-15 15:15 +0000 [c049d1c3b2] Richard Mudgett + + * configure: Check cache for valid pjproject tarball before downloading. + + On a fresh Asterisk source directory, the bundled pjproject tarball is + unconditionally downloaded even if the tarball is already in a specified + cache directory. + + * Made check if the pjproject tarball is valid in the cache directory + before downloading the tarball on a fresh source directory. + + Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5 + +2017-08-15 11:14 +0000 [9e2b2a9837] Richard Mudgett + + * res_pjsip: Fix prune_on_boot to remove only contacts for the host. + + * Check that the contact's reg_server matches the host's name before + deleting any prune_on_boot contacts. We don't want to delete reliable + transport contacts made with other servers if the ps_contacts database + table is shared with other servers. + + Thanks to Ross Beer for pointing out that the original prune logic would + delete reliable transport contacts from other servers. + + ASTERISK-27147 + + Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0 + +2017-08-04 09:25 +0000 [15fbcc74d8] Andrey Egorov + + * res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif + + Add ability to use tokens instead of passwords according to Google OAuth 2.0 + protocol. + + ASTERISK-27169 + Reported by: Andrey Egorov + Tested by: Andrey Egorov + + Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db + +2017-08-10 14:18 +0000 [bd28a9bbd8] Richard Mudgett + + * STUN/netsock2: Fix some valgrind uninitialized memory findings. + + * netsock2.c: Test the addr->len member first as it may be the only member + initialized in the struct. + + * stun.c:ast_stun_handle_packet(): The combinded[] local array could get + used uninitialized by ast_stun_request(). The uninitialized string gets + copied to another location and could overflow the destination memory + buffer. + + These valgrind findings were found for ASTERISK_27150 but are not + necessarily a fix for the issue. + + Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57 + +2017-08-02 18:44 +0000 [1bec781cce] Richard Mudgett + + * res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown. + + The fix for the issue is broken up into three parts. + + This is part three which handles the client side of REGISTER requests. + The registered contact may no longer be valid on the server when the + transport used is reliable and the connection is broken. + + * Re-REGISTER our contact if the reliable transport is broken after + registration completes. We attempt to re-REGISTER immediately to minimize + the time we are unreachable. Time may have already passed between the + connection being broken and the loss being detected. + + * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's + are still correct if an allocation failure happens. + + ASTERISK-27147 + + Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83 + +2017-07-31 14:21 +0000 [82f4ade959] Richard Mudgett + + * res_pjsip: Remove ephemeral registered contacts on transport shutdown. + + The fix for the issue is broken up into three parts. + + This is part two which handles the server side of REGISTER requests when + rewrite_contact is enabled. Any registered reliable transport contact + becomes invalid when the transport connection becomes disconnected. + + * Monitor the rewrite_contact's reliable transport REGISTER contact for + shutdown. If it is shutdown then the contact must be removed because it + is no longer valid. Otherwise, when the client attempts to re-REGISTER it + may be blocked because the invalid contact is there. Also if we try to + send a call to the endpoint using the invalid contact then the endpoint is + not likely to see the request. The endpoint either won't be listening on + that port for new connections or a NAT/firewall will block it. + + * Prune any rewrite_contact's registered reliable transport contacts on + boot. The reliable transport no longer exists so the contact is invalid. + + * Websockets always rewrite the REGISTER contact address and the transport + needs to be monitored for shutdown. + + * Made the websocket transport set a unique name since that is what we use + as the ao2 container key. Otherwise, we would not know which transport we + find when one of them shuts down. The names are also used for PJPROJECT + debug logging. + + * Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state + event. Now the global keep_alive_interval option, initially idle shutdown + timer, and the server REGISTER contact monitor can work on wetsocket + transports. + + * Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction. + Now initially idle websockets will automatically shutdown. + + ASTERISK-27147 + + Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4 + +2017-07-28 18:26 +0000 [1dcb92bba8] Richard Mudgett + + * res_pjsip: PJSIP Transport state monitor refactor. + + The fix for the issue is broken up into three parts. + + This is part one which refactors the transport state monitor code to allow + more modules to be able to monitor transports. + + * Pull the management of PJPROJECT's transport state callback code from + res_pjsip_transport_management.c into res_pjsip. Now other modules can + dynamically add and remove themselves from transport monitoring without + worrying about breaking PJPROJECT's callback chain. + + * Add the ability for other modules to get a callback whenever a specific + transport is shutdown. + + ASTERISK-27147 + + Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912 + +2017-07-27 15:36 +0000 [ee5edfb050] Richard Mudgett + + * res_pjsip_transport_management.c: Rename some variables. + + * Use monitored instead of the misleading keepalive name. + + Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6 + +2017-08-09 15:24 +0000 [ecd1f87edf] Richard Mudgett + + * UPGRADE notes: Prepare for the eventual 16 branch. + + Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c + +2017-08-10 09:09 +0000 [4ed2733dde] Scott Griepentrog + + * res_pjsip_messaging: IPv6 receive address needs brackets + + When handling an incoming SIP MESSAGE, PJSIP + attaches the IP address that the message was + received from to the message in the variable + PJSIP_RECVADDR. When the IP address is IPv6 + the :PORT appended results in an unparseable + mess. By using an additional bit flag on the + pj_sockaddr_print call, the conventional use + of brackets around the address is achieved. + + ASTERISK-27193 #close + + Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9 + +2017-07-26 09:17 +0000 [d430f718f5] Torrey Searle + + * res_rtp_asterisk: enable rtcp & QOS stats on native bridge + + Asterisk wasn't generating or forwarding RTCP packets when native + bridge was activated. Also the stats weren't available via + CHANNEL(qos). Now the RTCP stats are always calculated. + + ASTERISK-27158 #close + + Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b + +2017-07-28 07:53 +0000 [a2dde59154] Torrey Searle + + * res_rtp_asterisk: Make P2P bridge Asymmetric codec aware + + Introduce a new property to rtp-engine to make it aware of + the desire for assymetric codecs or not. If asymmetric codecs + is not allowed, the bridge will compare read/write formats + and shut down the p2p bridge if needed + + ASTERISK-26745 #close + + Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f + +2017-08-08 13:33 +0000 [305bd0d99f] George Joseph + + * Make --with-pjproject-bundled the default for Asterisk 15 + + '--with-pjproject-bundled' is now the default when running + ./configure. It can be disabled with '--without-pjproject-bundled'. + + To make building without an internet connection easier, a new + ./configure option '--with-download-cache' was added that sets + the cache for externals (like pjproject, the codecs and the DPMA), + AND the sounds files. It can also be specified as an environment + variable named "AST_DOWNLOAD_CACHE". The existing + '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and + '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable + remain and if specified, will override '--with-downloads-cache'. + + ASTERISK-27189 + + Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce + +2017-08-05 06:36 +0000 [62092bc114] Joshua Colp + + * res_pjsip_session: Release media resources on session end quicker. + + A change was made long ago where the session was kept around + until the underlying INVITE session had been destroyed. This + had the side effect of also keeping the underlying media resources + around for this time as well. + + This change ensures that when we are told to terminate the + session we immediately release any media sessions associated + with it. + + ASTERISK-27110 + + Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82 + +2017-07-29 20:03 +0000 [4b58609c33] Kirill Katsnelson + + * chan_sip: Access incoming REFER headers in dialplan + + This adds a way to access information passed along with SIP headers in + a REFER message that initiates a transfer. Headers matching a dialplan + variable GET_TRANSFERRER_DATA in the transferrer channel are added to + a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH. + + The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for + headers that should be put into the hash. If not set, no headers are + included. If set to a string (perhaps 'X-' in a typical case), all headers + starting this string are added. Empty string matches all headers. + + If there are multiple of the same header, only the latest occurrence in + the REFER message is available in the hash. + + Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the + referrer channel, and should be set with the '_' or '__' prefix. + + I avoided a specific reference to SIP or REFER, as in my mind the mechanism + can be generalized to other channel techs. + + ASTERISK-27162 + + Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e + +2017-08-06 11:15 +0000 [88c65f7cb6] Joshua Colp + + * bridge: Fix stream topology/participant locking and video misrouting. + + This change fixes a few locking issues and some video misrouting. + + 1. When accessing the stream topology of a channel the channel lock + must be held to guarantee the topology remains valid. + + 2. When a channel was joined to a bridge the bridge specific + implementation for stream mapping was not invoked, causing video + to be misrouted for a brief period of time. + + ASTERISK-27182 + + Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03 + +2017-08-05 14:43 +0000 [16cfc3a954] Corey Farrell + + * channel: Fix leak on successful call to chan->tech->requester. + + joint_cap needs to be released unconditionally as chan->tech->requester + does not steal the reference even on success. + + ASTERISK-27180 #close + + Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6 + +2017-08-04 16:47 +0000 [104a8047a5] Kevin Harwell + + * res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect + + Currently, the handling of the msid attribute is not quite right. According to + the spec the msid's between the offer/answer are not dependent upon one another. + Meaning the same msid's given in an offer do not have to be returned in the + answer for a given stream. And they probably shouldn't be (copied/reused) since + this can potentially cause some browser side confusion. + + This patch generates new msids when both an offer and answer are sent from + Asterisk. However, Asterisk does reuse the original msid it sent out for a + reinvite. Also audio+video streams are paired together by sharing the same + stream id, but a different track id. + + ASTERISK-27179 #close + + Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643 + +2017-08-03 20:58 +0000 [7f8f3ca4dd] Corey Farrell + + * Correct some leaks in unit tests. + + * chan_sip: channel in test_sip_rtpqos_1. + * test_config: config hook, config info and global config holder. + * test_core_format: format in format_attribute_set_without_interface. + * test_stream: unneeded frame duplication. + * test_taskprocessor: task_data. + + Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31 + +2017-07-26 17:49 +0000 [842e1414d0] Richard Mudgett + + * res_pjsip_transport_websocket.c: Fix serializer ref leak. + + Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57 + +2017-08-02 18:41 +0000 [615b6a200a] Richard Mudgett + + * res_pjsip_outbound_registration.c: Misc fixes. + + * Remove unnecessary CMP_STOP. + + * In handle_client_registration() use DEBUG_ATLEAST() to only do work + needed for the debug log message when the debug log message is needed. + + * In sip_outbound_registration_state_destroy() check state->registration + for NULL. + + Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80 + +2017-07-31 20:20 +0000 [564927c5ed] Richard Mudgett + + * res_pjsip_nat.c: Remove unnecessary CMP_STOP. + + Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da + +2017-07-31 14:20 +0000 [5655cded78] Richard Mudgett + + * res_pjsip_registrar.c: Remove unnecessary CMP_STOP. + + Most uses of CMP_STOP are superfluous and are only respected when + OBJ_MULTIPLE is used to search the container. + + Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8 + +2017-08-03 13:13 +0000 [123c93a77c] Tzafrir Cohen + + * Support GMIME 3.0 + + Support building the Asterisk httpd with version 3.0 of gmime as + well as earlier versions of that library. + + ASTERISK-27173 + + Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f + +2017-08-02 09:43 +0000 [521b6fed12] Kevin Harwell + + * alembic/res_pjsip: Add "webrtc" configuration option + + When the "webrtc" option was added in res_pjsip it was not added to the alembic + scripts. This patch adds the option for alembic. + + Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of + an OPT_BOOL_T so if this field is ever written to a database it will write out + the correct value. + + ASTERISK-27119 #close + + Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b + +2017-07-30 01:17 +0000 [4c0798e91d] Kirill Katsnelson + + * chan_sip: Add dialplan function SIP_HEADERS + + Syntax: SIP_HEADERS([prefix]) + + If the argument is specified, only the headers matching the given prefix + are returned. + + The function returns a comma-separated list of SIP header names from an + incoming INVITE message. Multiple headers with the same name are included + in the list only once. The returned list can be iterated over using the + functions POP() and SIP_HEADER(). + + For example, '${SIP_HEADERS(Co)}' might return the string + 'Contact,Content-Length,Content-Type'. + + Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional + extended headers sent by a peer. + + ASTERISK-27163 + + Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267 + +2017-08-02 14:16 +0000 [4b03eb5c38] Corey Farrell + + * Fix compile error for old versions of GCC. + + Use -Wno-format-truncation only if supported by compiler. + + ASTERISK-27171 #close + + Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6 + +2017-08-02 16:08 +0000 [148cf2e0f7] Corey Farrell + + * app_privacy: remove unused header asterisk/image.h + + Change-Id: I56ed530633a642633b18383821069e806c92ae82 + +2017-07-26 08:48 +0000 [2be8d91c0f] snuffy (license 5024) + + * res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation + + This change fixes PIDF content generation when the underlying device + state is considered in use. Previously it was incorrectly marked + as closed meaning they were offline/unavailable. The code now + correctly marks them as open. + + Additionally: + + * Generate an XML element for our activity instead of a using a text + node. + + * Consider every extension state other than "unavailable" to be 'open' + status. + + * Update the XML namespaces and structure to reflect those + documented in RFC 4480 + + * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the + "in use" activity. This change results in eyeBeam using the + appropriate icon for the watched user. + + This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. + + ASTERISK-26659 #close + Reported by: Abraham Liebsch + patches: + ASTERISK-26659.diff submitted by snuffy (license 5024) + + Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810 + +2017-07-23 18:34 +0000 [2a4283f3e7] Joshua Colp + + * res_pjsip: Add support for dnsmgr to external_media_address. + + The "external_media_address" option on transports is now + resolved using dnsmgr. This allows it to be automatically + refreshed regularly if refreshes are enabled in dnsmgr. + If the system is using a dynamic IP address a dynamic DNS + hostname can be provided to keep the IP address up to + date. + + Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2 + +2017-07-27 20:58 +0000 [58d032112b] Corey Farrell + + * Fix compiler warnings on Fedora 26 / GCC 7. + + GCC 7 has added capability to produce warnings, this fixes most of those + warnings. The specific warnings are disabled in a few places: + + * app_voicemail.c: truncation of paths more than 4096 chars in many places. + * chan_mgcp.c: callid truncated to 80 chars. + * cdr.c: two userfields are combined to cdr copy, fix would break ABI. + * tcptls.c: ignore use of deprecated method SSLv3_client_method(). + + ASTERISK-27156 #close + + Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88 + +2017-07-26 09:27 +0000 [3f98488279] Sean Bright + + * app_queue: Add announce-position-only-up option + + Setting this option will cause the Queue application to only announce + the caller's position if it has improved since the last time that we + announced it. + + Change-Id: I173a124121422209485b043e2bf784f54242fce6 + +2017-07-27 06:35 +0000 [ac6d98b28d] Ian Gilmour (license 6889) + + * bundled_pjproject: Improve SSL/TLS error handling + + OpenSSL has 2 levels or error processing. It's possible for the + top layer to return SSL_ERROR_SYSCALL but the lower layer return + no error, in which case processing should continue. Only the top + layer was being examined though so connections were being torn + down when they didn't need to be. This patch adds the examination + of the lower level codes, and if they return no errors, allows + processing to continue. + + ASTERISK-27001 + Reported-by: Ian Gilmour + patches: + pjproject-2.6.patch submitted by Ian Gilmour (license 6889) + + Updated-by: George Joseph and Sauw Ming (Teluu) + + Merged to upstream pjproject on 7/27/2017 (commit 5631) + + Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2 + +2017-06-26 07:52 +0000 [65c560894d] Torrey Searle + + * chan_pjsip: add a new function PJSIP_DTMF_MODE + + This function is a replica of SIPDtmfMode, allowing the DTMF mode of a + PJSIP call to be modified on a per-call basis + + ASTERISK-27085 #close + + Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612 + +2017-07-25 15:17 +0000 [b3914df10b] Sean Bright + + * res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours + + Change-Id: Ia578ede1a55b21014581793992a429441903278b + +2017-07-20 08:08 +0000 [4f4936fd72] Sergej Kasumovic + + * res_stasis_device_state: Unsubscribe should remove old subscriptions + + Case scenario with Applications ARI: + + * Once you subscribe to deviceState with Applications REST API, it will be + added into subscription pool. + + * When you unsubscribe it will remove from the device_state_subscription + hash table but not from the subscription pool. + + * When you subscribe again, it will add it to pool again. + + * Now you will have two subscriptions and you will receive same event + twice. + + This fix should now remove deviceState subscription from pool and it + should fix unsubscribe on deviceState. + + ASTERISK-27130 #close + + Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4 + +2017-07-24 13:30 +0000 [a6eb9ee7d2] Joshua Colp + + * core: Add VP9 passthrough support. + + This change adds VP9 as a known codec and creates a cached + "vp9" media format for use. + + Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc + +2017-07-19 18:11 +0000 [922930753c] Richard Mudgett + + * app_voicemail.c: Allow mailbox entry on authentication retry prompt. + + The following testsuite voicemail tests were failing to re-enter the + mailbox after the first login attempt. + + tests/apps/voicemail/authenticate_invalid_mailbox + tests/apps/voicemail/authenticate_invalid_password + + The tests were noting the start of the vm-incorrect-mailbox prompt and + immediately sending the mailbox for the next login attempt. Since the + invalid message playback had to complete before the digits were + recognized, the test passed for the wrong reason and added approximately + 20 seconds to the test times. + + * Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox + digits like the initial vm-login prompt so the tests are able to enter the + intended mailbox. + + Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8 + +2017-07-21 15:57 +0000 [2697e45157] Matthew Fredrickson + + * format.h: Fix a few minor errors in comments. + + A few minor problems were found in comments in format.h. This patch fixes them. + + Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94 + +2017-07-14 13:47 +0000 [19b080b547] Rusty Newton + + * say.c: Fix file locations for second, seconds, minute, minutes files + + The seconds and minutes files have always existed in the base language + directory of the Core package. So say.c has always been calling the wrong + location (under digits/) for those two files and in the case of second and + minute they didn't exist in the Core packages at all. + + The 1.6 sounds release moves the second and minute files into Core from + Extra for the languages that already had them. A future release will include + the second and minute files for languages that didn't already have them. + + This patch just changes all the target locations for second, seconds, + minute, and minutes that were under the digits subdir to be under the root of + sounds instead. Which is where the sounds will be for some languages after 1.6 + sounds and for all languages after a future release. + + ASTERISK-25810 #close + + Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 + Reported-by: Nicolas Riendeau + +2017-07-21 14:20 +0000 [a2f6028a51] Rusty Newton + + * Sounds: Update Makefile for Extra sounds 1.5.1 release + + Incrementing version for the Extra sounds release. 1.5.1 Extra sounds + removes two prompts that were moved into the Core packages in the 1.6 Core + sounds release. + + ASTERISK-27142 #close + + Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7 + +2017-07-21 11:17 +0000 [063c9a935f] George Joseph + + * Update make_ari_stubs in master to make the version 16 + + Ready for next major version + + Change-Id: If9dc99b3b78768529e69a297d8f87e23582ca6d0 + +2017-07-21 11:24 +0000 [ba52a36ff2] George Joseph + + * Restore the incorrectly deleted spandspflow2pcap.log + + Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5 + +2017-07-20 09:57 +0000 [25c9464325] Sean Bright + + * corosync: Fix corosync library name in configure.ac + + Also add new corosync packages to install_prereq. + + Reported by Travis Ryan in #asterisk-dev + + Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db + +2017-07-17 11:01 +0000 [680c491a62] Joshua Colp + + * bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. + + This change does a few things to improve packet loss and renegotiation: + + 1. On outgoing RTP streams we will now properly reflect out of order + packets and packet loss in the sequence number. This allows the + remote jitterbuffer to better reorder things. + + 2. Video updates can now be discarded for a period of time + after one has been sent to prevent flooding of clients. + + 3. For declined and removed streams we will now release any + media session resources associated with them. This was not + previously done and caused an issue where old state was being + used for a new stream. + + 4. RTP bundling was not actually removing bundled RTP instances + from the parent. This has been resolved by removing based on + the RTP instance itself and not the SSRC. + + 5. The code did not properly handle explicitly unbundling an + RTP instance from its parent. This now works as expected. + + ASTERISK-27143 + + Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45 + +2017-05-19 23:28 +0000 [d2fbbdd692] Richard Mudgett + + * SDP: Create declined m= SDP lines using remote SDP if applicable. + + * Update SDP unit tests to test negotiating with declined streams. + Generation of declined m= lines created and responded tested. + + Change-Id: I5cb99f5010994ab0c7d9cf2d395eca23fab37b98 + +2017-05-02 18:51 +0000 [3a18a09030] Richard Mudgett + + * SDP: Rework SDP offer/answer model and update capabilities merges. + + The SDP offer/answer model requires an answer to an offer before a new SDP + can be processed. This allows our local SDP creation to be deferred until + we know that we need to create an offer or an answer SDP. Once the local + SDP is created it won't change until the SDP negotiation is restarted. + + An offer SDP in an initial SIP INVITE can receive more than one answer + SDP. In this case, we need to merge each answer SDP with our original + offer capabilities to get the currently negotiated capabilities. To + satisfy this requirement means that we cannot update our proposed + capabilities until the negotiations are restarted. + + Local topology updates from ast_sdp_state_update_local_topology() are + merged together until the next offer SDP is created. These accumulated + updates are then merged with the current negotiated capabilities to create + the new proposed capabilities that the offer SDP is built. + + Local topology updates are merged in several passes to attempt to be smart + about how streams from the system are matched with the previously + negotiated stream slots. To allow for T.38 support when merging, type + matching considers audio and image types to be equivalent. First streams + are matched by stream name and type. Then streams are matched by stream + type only. Any remaining unmatched existing streams are declined. Any + new active streams are either backfilled into pre-merge declined slots or + appended onto the end of the merged topology. Any excess new streams + above the maximum supported number of streams are simply discarded. + + Remote topology negotiation merges depend if the topology is an offer or + answer. An offer remote topology negotiation dictates the stream slot + ordering and new streams can be added. A remote offer can do anything to + the previously negotiated streams except reduce the number of stream + slots. An answer remote topology negotiation is limited to what our offer + requested. The answer can only decline streams, pick codecs from the + offered list, or indicate the remote's stream hold state. + + I had originally kept the RTP instance if the remote offer SDP changed a + stream type between audio and video since they both use RTP. However, I + later removed this support in favor of simply creating a new RTP instance + since the stream's purpose has to be changing anyway. Any RTP packets + from the old stream type might cause mischief for the bridged peer. + + * Added ast_sdp_state_restart_negotiations() to restart the SDP + offer/answer negotiations. We will thus know to create a new local SDP + when it is time to create an offer or answer. + + * Removed ast_sdp_state_reset(). Save the current topology before + starting T.38. To recover from T.38 simply update the local topology to + the saved topology and restart the SDP negotiations to get the offer SDP + renegotiating the previous configuration. + + * Allow initial topology for ast_sdp_state_alloc() to be NULL so an + initial remote offer SDP can dictate the streams we start with. We can + always update the local topology later if it turns out we need to offer + SDP first because the remote chose to defer sending us a SDP. + + * Made the ast_sdp_state_alloc() initial topology limit to max_streams, + limit to configured codecs, handle declined streams, and discard + unsupported types. + + * Convert struct ast_sdp to ao2 object. Needed to easily save off a + remote SDP to refer to later for various reasons such as generating + declined m= lines in the local SDP. + + * Improve converting remote SDP streams to a topology including stream + state. A stream state of AST_STREAM_STATE_REMOVED indicates the stream is + declined/dead. + + * Improve merging streams to take into account the stream state. + + * Added query for remote hold state. + + * Added maximum streams allowed SDP config option. + + * Added ability to create new streams as needed. New streams are created + with configured default audio, video, or image codecs depending on stream + type. + + * Added global locally_held state along with a per stream local hold + state. Historically, Asterisk only has a global locally held state + because when the we put the remote on hold we do it for all active + streams. + + * Added queries for a rejected offer and current SDP negotiation role. + The rejected query allows the using module to know how to respond to a + failed remote SDP set. Should the using module respond with a 488 Not + Acceptable Here or 500 Internal Error to the offer SDP? + + * Moved sdp_state_capabilities.connection_address to ast_sdp_state. There + seems no reason to keep it in the sdp_state_capabilities struct since it + was only used by the ast_sdp_state.proposed_capabilities instance. + + * Callbacks are now available to allow the using module some customization + of negotiated streams and to complete setting up streams for use. See the + typedef doxygen for each callback for what is allowable and when they are + called. + * Added topology answerer modify callback. + * Added topology pre and post apply callbacks. + * Added topology offerer modify callback. + * Added topology offerer configure callback. + + * Had to rework the unit tests because I changed how SDP topologies are + merged. Replaced several unit tests with new negotiation tests. + + Change-Id: If07fe6d79fbdce33968a9401d41d908385043a06 + +2017-06-18 19:24 +0000 [70d2ccb9da] Corey Farrell + + * Core: Add support for systemd socket activation. + + This change adds support for socket activation of certain SOCK_STREAM + listeners in Asterisk: + * AMI / AMI over TLS + * CLI + * HTTP / HTTPS + + Example systemd units are provided. This support extends to any socket + which is initialized using ast_tcptls_server_start, so any unknown + modules using this function will support socket activation. + + Asterisk continues to function as normal if socket activation is not + enabled or if systemd development headers are not available during + build. + + ASTERISK-27063 #close + + Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d + +2017-09-01 19:29 +0000 Asterisk Development Team + + * asterisk 15.0.0-rc1 Released. + +2017-07-24 10:48 +0000 [35c8fb1590] George Joseph + + * pjsip_message_ip_updater: Fix issue handling "tel" URIs + + sanitize_tdata was assuming all URIs were SIP URIs so when a non + SIP uri was in the From, To or Contact headers, the unconditional + cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused + a segfault when trying to access uri->other_param. + + * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) + checks before attempting to cast or use the returned uri. + + ASTERISK-27152 + Reported-by: Ross Beer + + Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f + +2017-07-01 19:24 +0000 [231ee5e6c6] Corey Farrell + + * AST-2017-006: Fix app_minivm application MinivmNotify command injection + + An admin can configure app_minivm with an externnotify program to be run + when a voicemail is received. The app_minivm application MinivmNotify + uses ast_safe_system() for this purpose which is vulnerable to command + injection since the Caller-ID name and number values given to externnotify + can come from an external untrusted source. + + * Add ast_safe_execvp() function. This gives modules the ability to run + external commands with greater safety compared to ast_safe_system(). + Specifically when some parameters are filled by untrusted sources the new + function does not allow malicious input to break argument encoding. This + may be of particular concern where CALLERID(name) or CALLERID(num) may be + used as a parameter to a script run by ast_safe_system() which could + potentially allow arbitrary command execution. + + * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() + instead of ast_safe_system() to avoid command injection. + + * Document code injection potential from untrusted data sources for other + shell commands that are under user control. + + ASTERISK-27103 + + Change-Id: I7552472247a84cde24e1358aaf64af160107aef1 + +2017-05-22 10:36 +0000 [ba2c8f1458] Joshua Colp + + * res_rtp_asterisk: Only learn a new source in learn state. + + This change moves the logic which learns a new source address + for RTP so it only occurs in the learning state. The learning + state is entered on initial allocation of RTP or if we are + told that the remote address for the media has changed. While + in the learning state if we continue to receive media from + the original source we restart the learning process. It is + only once we receive a sufficient number of RTP packets from + the new source that we will switch to it. Once this is done + the closed state is entered where all packets that do not + originate from the expected source are dropped. + + The learning process has also been improved to take into + account the time between received packets so a flood of them + while in the learning state does not cause media to be switched. + + Finally RTCP now drops packets which are not for the learned + SSRC if strict RTP is enabled. + + ASTERISK-27013 + + Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c + +2017-08-30 07:28 +0000 [663fe3e31f] Joshua Colp + + * res_rtp_asterisk: Allow remote SSRC to change on an RTP instance. + + When SDP renegotiation occurs it is possible for an RTP + instance to be reused for a new stream, resulting in the remote + SSRC changing if it is part of a bundle group. This change + allows this and updates its mapping in the current bundle + group. + + ASTERISK-27231 + + Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490 + +2017-08-24 11:45 +0000 [dab0389e24] Joshua Colp + + * core: Reduce video update queueing. + + A video update frame is used to indicate that a channel + with video negotiated should provide a full frame so the + decoder decoding the stream is able to do so. In situations + where a queue is used to store frames it makes no sense + for the queue to contain multiple video update frames. One + is sufficient to have a full frame be sent. + + ASTERISK-27222 + + Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7 + +2017-08-14 12:20 +0000 [0a0ef8a1b1] Richard Mudgett + + * chan_pjsip.c: Fix topology refresh response code accuracy. + + There are other 1xx and 2xx codes than 100 and 200 respectively. + + Change-Id: I680db0997343256add1478714f5bf5b5569aee17 + +2017-08-18 17:37 +0000 [00b10fa1e1] Richard Mudgett + + * res_pjsip_session.c: Fix crash when declining an active stream. + + If a previously active stream is declined we could crash because the + channel's thread is still using the stream while we are updating the + topology in the serializer thread. + + * Defer removing any declined stream's handler until we have blocked the + channel's thread with the channel lock. + + ASTERISK-27212 + + Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420 + +2017-08-16 17:50 +0000 [6acc945533] Richard Mudgett + + * bridge_channel.c: Fix FRACK when mapping frames to the bridge. + + * Add protection checks when mapping streams to the bridge. The channel + and bridge may be in the process of updating the stream mapping when a + media frame comes in so we may not be able to map the frame at the time. + + * We need to map the streams to the bridge's stream numbers right before + they are written into the bridge. That way we don't have to keep + locking/unlocking the bridge and we won't have any synchronization + problems before the frames actually go into the bridge. + + * Protect the deferred queue with the bridge_channel lock. + + ASTERISK-27212 + + Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a + +2017-08-11 16:31 +0000 [efbf0aa8df] Richard Mudgett + + * channel: Fix topology API locking. + + * ast_channel_request_stream_topology_change() must not be called with any + channel locks held. + + * ast_channel_stream_topology_changed() must be called with only the + passed channel lock held. + + ASTERISK-27212 + + Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691 + +2017-08-16 15:22 +0000 [6bad253669] Richard Mudgett + + * bridge: Fix softmix bridge deadlock. + + * Fix deadlock in + bridge_softmix.c:softmix_bridge_stream_topology_changed() between + bridge_channel and channel locks. + + * The new bridge technology topology change callbacks must be called with + the bridge locked. The callback references the bridge channel list, the + bridge technology could change, and the bridge stream mapping is updated. + + ASTERISK-27212 + + Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be + +2017-08-17 17:07 +0000 [40faa22ce8] Richard Mudgett + + * app_confbridge: Document sfu video_mode value. + + Change-Id: I26e17df2c93f3933b23f78070603adbcc84ba204 + +2017-08-16 15:43 +0000 [e52f9b041a] George Joseph + + * Fix downloader not working with curl + + The codec/dpma downloader wasn't handling curl correctly. The logic + that transforms makeopts into a bash-sourceable file wasn't + handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was + looking for an 'or' command. + + That logic has been eliminated. Instead of trying to transform + and source makeopts, the downloader now calls a make scriptlet + to print the value of a specific variable. This way, make handles + the ors (or any other make construct that happens to creep into + that file). + + ASTERISK-27202 + Reported by: Sean McCord + + Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99 + +2017-08-15 13:12 +0000 [d7b04f22de] Kevin Harwell + + * manager: hook event is not being raised + + When the iostream code went in it introduced a conditional that made it so the + hook event was not being raised even if a hook is present. This patch adds a + check to see if a hook is present in astman_append. If so then call into the + send_string function, which in turn raises the even for specified hook. + + Also updated the ami hooks unit test, so the test could be automated. + + ASTERISK-27200 #close + + Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36 + +2017-08-15 15:15 +0000 [44d316ef4a] Richard Mudgett + + * configure: Check cache for valid pjproject tarball before downloading. + + On a fresh Asterisk source directory, the bundled pjproject tarball is + unconditionally downloaded even if the tarball is already in a specified + cache directory. + + * Made check if the pjproject tarball is valid in the cache directory + before downloading the tarball on a fresh source directory. + + Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5 + +2017-08-09 15:24 +0000 [012391920c] Richard Mudgett + + * UPGRADE notes: Fixup for the 15 branch + + Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c + +2017-08-04 16:47 +0000 [4d3e66eabc] Kevin Harwell + + * res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect + + Currently, the handling of the msid attribute is not quite right. According to + the spec the msid's between the offer/answer are not dependent upon one another. + Meaning the same msid's given in an offer do not have to be returned in the + answer for a given stream. And they probably shouldn't be (copied/reused) since + this can potentially cause some browser side confusion. + + This patch generates new msids when both an offer and answer are sent from + Asterisk. However, Asterisk does reuse the original msid it sent out for a + reinvite. Also audio+video streams are paired together by sharing the same + stream id, but a different track id. + + ASTERISK-27179 #close + + Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643 + +2017-08-06 11:15 +0000 [71d0424ed5] Joshua Colp + + * bridge: Fix stream topology/participant locking and video misrouting. + + This change fixes a few locking issues and some video misrouting. + + 1. When accessing the stream topology of a channel the channel lock + must be held to guarantee the topology remains valid. + + 2. When a channel was joined to a bridge the bridge specific + implementation for stream mapping was not invoked, causing video + to be misrouted for a brief period of time. + + ASTERISK-27182 + + Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03 + (cherry picked from commit 0e352ec5100331c6a32008acc88d69d0fc58ccdd) + +2017-08-08 13:33 +0000 [84600e2682] George Joseph + + * Make --with-pjproject-bundled the default for Asterisk 15 + + '--with-pjproject-bundled' is now the default when running + ./configure. It can be disabled with '--without-pjproject-bundled'. + + To make building without an internet connection easier, a new + ./configure option '--with-download-cache' was added that sets + the cache for externals (like pjproject, the codecs and the DPMA), + AND the sounds files. It can also be specified as an environment + variable named "AST_DOWNLOAD_CACHE". The existing + '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and + '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable + remain and if specified, will override '--with-downloads-cache'. + + ASTERISK-27189 + + Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce + +2017-08-05 14:43 +0000 [afd7875e82] Corey Farrell + + * channel: Fix leak on successful call to chan->tech->requester. + + joint_cap needs to be released unconditionally as chan->tech->requester + does not steal the reference even on success. + + ASTERISK-27180 #close + + Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6 + (cherry picked from commit 3dbb1b9f48b0fa23cec2d8e3f94173004da320a4) + +2017-08-02 14:16 +0000 [53bba12340] Corey Farrell + + * Fix compile error for old versions of GCC. + + Use -Wno-format-truncation only if supported by compiler. + + ASTERISK-27171 #close + + Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6 + (cherry picked from commit cd79a15b2f9411c6e77f0f6594ff0c46f0ece080) + +2017-08-02 09:43 +0000 [c042ad8343] Kevin Harwell + + * alembic/res_pjsip: Add "webrtc" configuration option + + When the "webrtc" option was added in res_pjsip it was not added to the alembic + scripts. This patch adds the option for alembic. + + Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of + an OPT_BOOL_T so if this field is ever written to a database it will write out + the correct value. + + ASTERISK-27119 #close + + Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b + (cherry picked from commit b0c016cf6e0bcbe743f4f8286fb9b5ded830ccf7) + +2017-08-02 11:44 +0000 Asterisk Development Team + + * asterisk 15.0.0-beta1 Released. + +2017-07-27 20:58 +0000 [aba08692df] Corey Farrell + + * Fix compiler warnings on Fedora 26 / GCC 7. + + GCC 7 has added capability to produce warnings, this fixes most of those + warnings. The specific warnings are disabled in a few places: + + * app_voicemail.c: truncation of paths more than 4096 chars in many places. + * chan_mgcp.c: callid truncated to 80 chars. + * cdr.c: two userfields are combined to cdr copy, fix would break ABI. + * tcptls.c: ignore use of deprecated method SSLv3_client_method(). + + ASTERISK-27156 #close + + Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88 + +2017-07-27 06:35 +0000 [64edb4ed21] George Joseph + + * bundled_pjproject: Improve SSL/TLS error handling + + OpenSSL has 2 levels or error processing. It's possible for the + top layer to return SSL_ERROR_SYSCALL but the lower layer return + no error, in which case processing should continue. Only the top + layer was being examined though so connections were being torn + down when they didn't need to be. This patch adds the examination + of the lower level codes, and if they return no errors, allows + processing to continue. + + ASTERISK-27001 + Reported-by: Ian Gilmore + Patch-by: Ian Gilmore (pjproject-2.6.patch License 6889) + Updated-by: George Joseph and Sauw Ming (Teluu) + + Merged to upstream pjproject on 7/27/2017 (commit 5631) + + Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2 + +2017-07-25 15:17 +0000 [d056f6b2fe] Sean Bright + + * res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours + + Change-Id: Ia578ede1a55b21014581793992a429441903278b + +2017-07-26 08:48 +0000 [11cd3be506] Sean Bright + + * res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation + + This change fixes PIDF content generation when the underlying device + state is considered in use. Previously it was incorrectly marked + as closed meaning they were offline/unavailable. The code now + correctly marks them as open. + + Additionally: + + * Generate an XML element for our activity instead of a using a text + node. + + * Consider every extension state other than "unavailable" to be 'open' + status. + + * Update the XML namespaces and structure to reflect those + documented in RFC 4480 + + * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the + "in use" activity. This change results in eyeBeam using the + appropriate icon for the watched user. + + This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. + + ASTERISK-26659 #close + Reported by: Abraham Liebsch + patches: + ASTERISK-26659.diff submitted by snuffy (license 5024) + + Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810 + +2017-07-26 09:27 +0000 [76270c0f78] Sean Bright + + * app_queue: Add announce-position-only-up option + + Setting this option will cause the Queue application to only announce + the caller's position if it has improved since the last time that we + announced it. + + Change-Id: I173a124121422209485b043e2bf784f54242fce6 + +2017-06-26 07:52 +0000 [154e74eced] Torrey Searle + + * chan_pjsip: add a new function PJSIP_DTMF_MODE + + This function is a replica of SIPDtmfMode, allowing the DTMF mode of a + PJSIP call to be modified on a per-call basis + + ASTERISK-27085 #close + + Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612 + +2017-07-17 11:01 +0000 [451d86d62e] Joshua Colp + + * bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. + + This change does a few things to improve packet loss and renegotiation: + + 1. On outgoing RTP streams we will now properly reflect out of order + packets and packet loss in the sequence number. This allows the + remote jitterbuffer to better reorder things. + + 2. Video updates can now be discarded for a period of time + after one has been sent to prevent flooding of clients. + + 3. For declined and removed streams we will now release any + media session resources associated with them. This was not + previously done and caused an issue where old state was being + used for a new stream. + + 4. RTP bundling was not actually removing bundled RTP instances + from the parent. This has been resolved by removing based on + the RTP instance itself and not the SSRC. + + 5. The code did not properly handle explicitly unbundling an + RTP instance from its parent. This now works as expected. + + ASTERISK-27143 + + Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45 + +2017-07-20 08:08 +0000 [2128dc7c87] Sergej Kasumovic + + * res_stasis_device_state: Unsubscribe should remove old subscriptions + + Case scenario with Applications ARI: + + * Once you subscribe to deviceState with Applications REST API, it will be + added into subscription pool. + + * When you unsubscribe it will remove from the device_state_subscription + hash table but not from the subscription pool. + + * When you subscribe again, it will add it to pool again. + + * Now you will have two subscriptions and you will receive same event + twice. + + This fix should now remove deviceState subscription from pool and it + should fix unsubscribe on deviceState. + + ASTERISK-27130 #close + + Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4 + +2017-07-24 13:30 +0000 [927fc6bbd9] Joshua Colp + + * core: Add VP9 passthrough support. + + This change adds VP9 as a known codec and creates a cached + "vp9" media format for use. + + Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc + +2017-07-21 15:57 +0000 [9aa4942a49] Matthew Fredrickson + + * format.h: Fix a few minor errors in comments. + + A few minor problems were found in comments in format.h. This patch fixes them. + + Change-Id: I07f0bdb47b93359b361c4c3d8ecc87cd3199dd94 + +2017-07-23 18:34 +0000 [0219d25e4e] Joshua Colp + + * res_pjsip: Add support for dnsmgr to external_media_address. + + The "external_media_address" option on transports is now + resolved using dnsmgr. This allows it to be automatically + refreshed regularly if refreshes are enabled in dnsmgr. + If the system is using a dynamic IP address a dynamic DNS + hostname can be provided to keep the IP address up to + date. + + Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2 + +2017-07-19 18:11 +0000 [85c631294a] Richard Mudgett + + * app_voicemail.c: Allow mailbox entry on authentication retry prompt. + + The following testsuite voicemail tests were failing to re-enter the + mailbox after the first login attempt. + + tests/apps/voicemail/authenticate_invalid_mailbox + tests/apps/voicemail/authenticate_invalid_password + + The tests were noting the start of the vm-incorrect-mailbox prompt and + immediately sending the mailbox for the next login attempt. Since the + invalid message playback had to complete before the digits were + recognized, the test passed for the wrong reason and added approximately + 20 seconds to the test times. + + * Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox + digits like the initial vm-login prompt so the tests are able to enter the + intended mailbox. + + Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8 + +2017-07-21 14:20 +0000 [e0ad75ec2a] Rusty Newton + + * Sounds: Update Makefile for Extra sounds 1.5.1 release + + Incrementing version for the Extra sounds release. 1.5.1 Extra sounds + removes two prompts that were moved into the Core packages in the 1.6 Core + sounds release. + + ASTERISK-27142 #close + + Change-Id: I82f017812b0ea9599e19dd4635afd55611f13ee7 + +2017-07-14 13:47 +0000 [715d79b60d] Rusty Newton + + * say.c: Fix file locations for second, seconds, minute, minutes files + + The seconds and minutes files have always existed in the base language + directory of the Core package. So say.c has always been calling the wrong + location (under digits/) for those two files and in the case of second and + minute they didn't exist in the Core packages at all. + + The 1.6 sounds release moves the second and minute files into Core from + Extra for the languages that already had them. A future release will include + the second and minute files for languages that didn't already have them. + + This patch just changes all the target locations for second, seconds, + minute, and minutes that were under the digits subdir to be under the root of + sounds instead. Which is where the sounds will be for some languages after 1.6 + sounds and for all languages after a future release. + + ASTERISK-25810 #close + + Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 + Reported-by: Nicolas Riendeau + +2017-06-18 19:24 +0000 [eea9da2f42] Corey Farrell + + * Core: Add support for systemd socket activation. + + This change adds support for socket activation of certain SOCK_STREAM + listeners in Asterisk: + * AMI / AMI over TLS + * CLI + * HTTP / HTTPS + + Example systemd units are provided. This support extends to any socket + which is initialized using ast_tcptls_server_start, so any unknown + modules using this function will support socket activation. + + Asterisk continues to function as normal if socket activation is not + enabled or if systemd development headers are not available during + build. + + ASTERISK-27063 #close + + Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d + +2017-07-21 11:24 +0000 [94de9d3eea] George Joseph + + * Restore the incorrectly deleted spandspflow2pcap.log + + Change-Id: Iafe78cf0fb1e7064223d4dea279eeb776c8fa8e5 + +2017-07-21 07:56 +0000 [6239203628] George Joseph + + * Update make_ari_stubs to correct version + + Change-Id: I18575b46db48d62edc72f37dc23b4ab22b43a8b1 + +2017-07-20 09:57 +0000 [6650ae43e1] Sean Bright + + * corosync: Fix corosync library name in configure.ac + + Also add new corosync packages to install_prereq. + + Reported by Travis Ryan in #asterisk-dev + + Change-Id: Ib861c95ba630fed62dc54e56784ad8446ed9d2db + +2017-07-20 13:06 +0000 [b172474728] George Joseph + + * Update MAINLINE_BRANCH to 15 + + Change-Id: I425d542b600ceabeef2342e9adfeb68c484a043d + +2017-07-20 10:52 +0000 [3e8d628c0e] George Joseph + + * Update AMI and ARI versions for master/15 and update UPDATE.txt + + AMI goes from 3.2.0 to 4.0.0 + ARI goes from 2.0.0 to 3.0.0 + + Copied UPGRADE.txt -> UPGRADE-15.txt + Created new UPGRADE.txt + Removed a log file that was accidentally checked in a while ago + + Change-Id: I1c794f910038459b13e16f9c3a12c44e56f142f7 + +2017-07-18 15:04 +0000 [e7d9e42616] Benjamin Keith Ford + + * pjsip: Increase maximum packet size. + + The maximum packet size for PJSIP has been increased to handle the + multiple streams being added for WebRTC. + + Change-Id: I9ea1e8d02668c544acadcb1c6200e1cc1bd588b3 + +2017-07-17 07:19 +0000 [bcd3f65174] Joshua Colp + + * bridge_softmix: Don't reorder streams on participant leaving. + + When a participant leaves a bridge while operating in SFU mode + their respective stream on every other participant needs to be + removed. Leaving the stream out of the new topology results in + every stream after it being moved and reordered. This causes + problems with clients. Instead simply mark the stream as removed + which leaves it in place in the SDP and doesn't reorder or touch + any other streams. + + ASTERISK-27136 + + Change-Id: I4b3f840adcdf69b83842b0d8a737665ba0ef9cb1 + +2017-07-16 12:31 +0000 [f48695ce5b] Joshua Colp + + * bridge_softmix: Use removed stream spots when renegotiating. + + Streams are never truly removed in SDP, they still occupy + a location within the SDP. This location can be reused by + another stream if it so chooses. + + This change takes advantage of this such that if a new stream + is needing to be added for a new participant any removed streams + are instead replaced first. This reduces the size of the SDP + and the number of streams. + + ASTERISK-27134 + + Change-Id: I95cdcfd55cf47e02ea52abb5d94008db3fb68b1d + +2017-07-16 12:18 +0000 [942ee54b53] Joshua Colp + + * res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use. + + This change makes it so that if an RTCP packet is being sent + the RTP ICE component is used for sending if RTCP-MUX is in use. + + ASTERISK-27133 + + Change-Id: I6200f611ede709602ee9b89501720c29545ed68b + +2017-07-14 01:25 +0000 [26f149ab0a] Sergej Kasumovic + + * app_confbridge: Make sure name recordings are always removed from the filesystem + + This commit fixes two possible scenarios: + + * When recording name and if during recording you hangup, file is never + removed. This is due to the fact file location is nulled. + * When recording name and if you hangup during thank-you prompt, file + is never removed. + + ASTERISK-27123 #close + + Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625 + +2017-07-14 01:11 +0000 [d3f5b265c7] Sergej Kasumovic + + * chan_iax2: On reload make sure to check for existing MWI subscription + + On every reload of chan_iax2 module, MWI subscription was added, which + results in additional taskprocessors being accumulated over time. + + This commit fixes it by making sure we check for existing subscription + first. + + This was verified with 'core show taskprocessors' CLI command. + + ASTERISK-27122 #close + + Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9 + +2017-07-10 18:17 +0000 [7da6ddda30] Kevin Harwell + + * res_pjsip: Add "webrtc" configuration option + + This patch creates a new configuration option called "webrtc". When enabled it + defaults and enables the following options that are needed in order for webrtc + to work in Asterisk: + + rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled + media_encryption=dtls + dtls_verify=fingerprint + dtls_setup=actpass + + When "webrtc" is enabled, this patch also parses the "msid" media level + attribute from an SDP. It will also appropriately add it onto the outgoing + session when applicable. + + Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. + + ASTERISK-27119 #close + + Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd + +2017-07-13 15:43 +0000 [3fbb4a0a08] Rusty Newton + + * Sounds: Update for core sounds 1.6 release + + Added necessary lines to make the en_NZ language set selectable and to get + core sounds 1.6 pulled down. + + ASTERISK-26807 #close + ASTERISK-25816 #close + ASTERISK-26274 #close + + Change-Id: I84e4dd4696568cc1ba318d12ac4b075461d6eed4 + +2017-07-10 14:04 +0000 [78a50b0343] Corey Farrell + + * core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. + + This adds support for parsing timelen values from config files. This + includes support for all flags which apply to PARSE_INT32. Support for + this parser is added to ACO via the OPT_TIMELEN_T option type. + + Fixes an issue where extra characters provided to ast_app_parse_timelen + were ignored, they now cause an error. + + Testing is included. + + ASTERISK-27117 #close + + Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554 + +2017-06-30 13:55 +0000 [065c3005ad] Joshua Colp + + * res_rtp_asterisk / res_pjsip: Add support for BUNDLE. + + BUNDLE is a specification used in WebRTC to allow multiple + streams to use the same underlying transport. This reduces + the number of ICE and DTLS negotiations that has to occur + to 1 normally. + + This change implements this by adding support for it to + the RTP SDP module in PJSIP. BUNDLE can be turned on using + the "bundle" option and on an offer we will offer to + bundle streams together. On an answer we will accept any + bundle groups provided. Once accepted each stream is bundled + to another RTP instance for transport. + + For the res_rtp_asterisk changes the ability to bundle + an RTP instance to another based on the SSRC received + from the remote side has been added. For outgoing traffic + if an RTP instance is bundled to another we will use the + other RTP instance for any transport related things. For + incoming traffic received from the transport instance we + look up the correct instance based on the SSRC and use it + for any non-transport related data. + + ASTERISK-27118 + + Change-Id: I96c0920b9f9aca7382256484765a239017973c11 + +2017-07-11 09:55 +0000 [8b535a406b] Torrey Searle + + * res/res_stasis_snoop: generate silence when audiohook returns null + + Currently when rtp is paused, no packets are written to the + recorded audio file, causing the silence to be skipped and recording + not properly time aligned. The read handler as been adapted to + return a silence frame of the correct size. + + ASTERISK-27128 #close + + Change-Id: I2d7f60650457860b9c70907b14426756b058a844 + +2017-06-22 07:47 +0000 [d42a9cc9dc] Torrey Searle + + * res/res_pjsip_t38 ensure t38 requests get rejected quickly + + arm the t38 webhook always, so we can correctly reject a + T38 negotiation request when t38 is disabled on a channel + + Change-Id: Ib1ffe35aee145d4e0fe61dd012580be11aae079d + +2017-07-12 13:24 +0000 [6b138046e7] Corey Farrell + + * core: Add digit filtering to ast_waitfordigit_full + + This adds a parameter to ast_waitfordigit_full which can be used to only + stop waiting when certain expected digits are received. Any unexpected + DTMF digits are simply ignored. + + This also creates a new dialplan application WaitDigit. + + ASTERISK-27129 #close + + Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9 + +2017-07-11 04:48 +0000 [b54eb167b4] Holger Hans Peter Freyther + + * app_playback.c: Use the timezonename parameter + + In say_date_generic the timezonename parameter is passed but never + used. Fix it by passing it to the ast_localtime function. + + ASTERISK-27124 + + Change-Id: I63106b8db10426d417d7275f22554a616e92fae4 + +2017-07-12 15:07 +0000 [e83b9d141a] Sean Bright + + * basic-pbx: Remove res_pjsip_multihomed from sample config + + ASTERISK-27127 #close + Reported by: HZMI8gkCvPpom0tM + + Change-Id: I2b0c54570d58156e37166ac536728af3b6c01789 + +2017-07-11 14:33 +0000 [7f09fd2c2f] Joshua Colp + + * bridge/core_unreal: Fix SFU bugs with forwarding frames. + + This change fixes a few things uncovered during SFU testing. + + 1. Unreal channels incorrectly forwarded video frames when + no video stream was present on them. This caused a crash when + they were read as the core requires a stream to exist for the + underlying media type. The Unreal channel will now ensure a + stream exists for the media type before forwarding the frame + and if no stream exists then the frame is dropped. + + 2. Mapping of frames during bridging from the stream number of + the underlying channel to the stream number of the bridge was + done in the wrong location. This resulted in the frame getting + dropped. This mapping now occurs on reading of the frame from + the channel. + + 3. Bridging was using the wrong ast_read function resulting in + it living in a non-multistream world. + + 4. In bridge_softmix when adding new streams to existing channels + the wrong stream topology was copied resulting in no streams + being added. + + Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8 + +2017-07-11 07:26 +0000 [b7a875778a] George Joseph + + * res_musiconhold: Add kill_escalation_delay, kill_method to class + + By default, when res_musiconhold reloads or unloads, it sends a HUP + signal to custom applications (and all descendants), waits 100ms, + then sends a TERM signal, waits 100ms, then finally sends a KILL + signal. An application which is interacting with an external + device and/or spawns children of its own may not be able to exit + cleanly in the default times, expecially if sent a KILL signal, or + if it's children are getting signals directly from + res_musiconhoild. + + * To allow extra time, the 'kill_escalation_delay' + class option can be used to set the number of milliseconds + res_musiconhold waits before escalating kill signals, with the + default being the current 100ms. + + * To control to whom the signals are sent, the "kill_method" class + option can be set to "process_group" (the default, existing + behavior), which sends signals to the application and its + descendants directly, or "process" which sends signals only to the + application itself. + + Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b + +2017-07-05 12:44 +0000 [5d86da61a6] Benjamin Keith Ford + + * manager: Remove AMI "Queues" action. + + When performing the "Queues" action via AMI, it outputs the same + text that the Asterisk CLI outputs when running a "queue show" + command, which does not conform with the AMI spec. "QueueStatus" + already does what the "Queues" action should do, so instead of + correcting the output, the "Queues" action will be removed and + "QueueStatus" should be used instead. + + ASTERISK-27073 #close + Reported by: Brian + + Change-Id: Id11743859758255b69cc3a557750d7a56c6d16f8 + +2017-07-03 07:30 +0000 [d58ef31acd] Tzafrir Cohen + + * Avoid setting maxfiles for a remote asterisk + + Setting maxfiles (maximum number of open files) has no practical + effect on a remote asterisk (rasterisk, rasterisk -x). + + It has an ill effect of printing an extra message, which + may be annoying in case of -x. + + ASTERISK-27105 #close + + Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2 + +2017-07-05 15:31 +0000 [303f935a50] George Joseph + + * http.c: Reduce log spam + + Messages like "fwrite() failed: Connection reset by peer" are no + help whatsoever, especially since they can be caused simply by a + client disconnecting. + + * Make those WARNINGs DEBUGs. + * Check the return from ast_iostream_printf of headers. + + Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b + +2017-07-07 11:19 +0000 [8f72128e66] Benjamin Keith Ford + + * res_pjsip: Fix crash with from_user containing invalid characters. + + If the from_user field contains certain characters (like @, {, ^, etc.), + PJSIP will return a null value for the URI when attempting to parse it. + This causes a crash when trying to dial out through a trunk that contains + these invalid characters in its from_user field. + + This change checks the configuration and ensures that an endpoint will + not be created if the from_user contains an invalid character. It also + adds a null check to the PJSIP URI parsing as a backup. + + ASTERISK-27036 #close + Reported by: Maxim Vasilev + + Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0 + +2017-06-27 19:27 +0000 [03ae8b0105] Richard Mudgett + + * json.c: Add backtrace log to find 'Invalid UTF-8 string' errors + + Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929 + +2017-07-05 13:39 +0000 [9cd8a1df79] Richard Mudgett + + * res_rtp_asterisk.c: Fix TURN deadlock by using ICE session group lock. + + When a message is received on the TURN socket, the code processing the + message needs to call into the ICE/STUN session for further processing. + This code path locks the TURN group lock then the ICE/STUN group lock. In + another thread an ICE/STUN timer can fire off to send a keep alive message + over the TURN socket. In this code path, the ICE/STUN group lock is + obtained then the TURN group lock is obtained to send the packet. A + classic deadlock case if the group locks are not the same. + + * Made TURN get created using the ICE/STUN session's group lock. + + NOTE: I was originally concerned that the ICE/STUN session can get + recreated by ice_reset_session() for an event like RTCP multiplexing + causing a change during SDP negotiation. In this case the TURN group lock + would become different. However, TURN is also recreated as part of the + ICE/STUN recreation in ice_create() when all known ICE candidates are + added to the new ICE session. While the ICE/STUN and TURN sessions are + being recreated there is a period where the group locks could be + different. + + ASTERISK-27023 #close + Patches: + res_rtp_asterisk-turn-deadlock-fix.patch (license #6502) + patch uploaded by Michael Walton (modified) + + Change-Id: Ic870edb99ce4988a8c8eb6e678ca7f19da1432b9 + +2017-07-06 05:55 +0000 [7a4f577eb7] George Joseph + + * Fix alembic branches + + Change-Id: I04f607f084bda9b1b7f626e8e9735c37dc751187 + +2017-06-23 11:17 +0000 [1028f64be4] Richard Mudgett + + * bridge_native_rtp.c: Fix direct media video RTP instance ACL check. + + The video stream was using the audio stream RTP instance addresses to + check if the video RTP gets directed to an allowed direct media Access + Control List (ACL) address. There is no guarantee that the video RTP + instance uses the same addresses as the audio RTP instance. + + This looks like it has been a bug since v11 when direct media ACL was + first added to chan_sip and then faithfully reproduced through a couple + code refactorings into the new bridging architecture. + + Change-Id: I8ddd56320e0eea769f3ceed3fa5b6bdfb51d681a + +2017-07-05 10:29 +0000 [325eeced6a] Sean Bright + + * core: Remove 'Data Retrieval API' + + This API was not actively maintained, was not added to new modules + (such as res_pjsip), and there exist better alternatives to acquire the + same information, such as the ARI. + + Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83 + +2017-06-19 11:22 +0000 [d556c67f9f] Rodrigo Ramírez Norambuena + + * app_queue: Add change priority of call + + This patch include a feature to change the priority a caller in a + queue by CLI and AMI. + + Change-Id: I55d520d71cc1cefe9a9b81fefaefc14679e96133 + +2017-07-03 10:59 +0000 [910c05455d] Alexander Traud + + * chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support). + + When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was + added in any case, because of a local Boolean-negation error of the return value + of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was + still always added with tlsenable=yes, because the domains were not compared + just on the address but also on the port – and TLS is always on a different port + than UDP/TCP. + + ASTERISK-27106 + + Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c + +2017-07-03 10:38 +0000 [4398aa8fa4] Alexander Traud + + * chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support). + + Because of a copy-and-paste error when the struct ast_sockaddr changed, + tlsbindaddr was not added, when sip.conf contained autodomain=yes; see + "show sip domains" on the command-line interface (CLI) of Asterisk. + + ASTERISK-27106 + + Change-Id: I3d0957150017c223136968ef1266f275d0d6695e + +2017-06-29 13:58 +0000 [950b39a4f5] Sean Bright + + * app_voicemail: Cleanup ODBC connection handling + + The primary focus of this patch is adding a missing call to + ast_odbc_release_obj(), but is also a general cleanup of the ODBC + related code in app_voicemail. + + ASTERISK-27093 #close + + Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b + +2017-06-30 23:57 +0000 [50ddb56dad] Corey Farrell + + * channel: Clear channel flag in error branch. + + Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when + ast_read returns NULL. + + ASTERISK-27100 #close + + Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d + +2017-06-29 18:27 +0000 [b485f6c59c] Richard Mudgett + + * pjsip_distributor.c: Fix deadlock with TCP type transports. + + When a SIP message comes in on a transport, pjproject obtains the lock on + the transport and pulls the data out of the socket. Unlike UDP, the TCP + transport does not allow concurrent access. Without concurrency the + transport lock is not released when the transport's message complete + callback is called. The processing continues and eventually Asterisk + starts processing the SIP message. The first thing Asterisk tries to do + is determine the associated dialog of the message to determine the + associated serializer. To get the associated serializer safely requires + us to get the dialog lock. + + To send a request or response message for a dialog, pjproject obtains the + dialog lock and then obtains the transport lock. Deadlock can result + because of the opposite order the locks are obtained. + + * Fix the deadlock by obtaining the serializer associated with the dialog + another way that doesn't involve obtaining the dialog lock. In this case, + we use an ao2 container to hold the associated endpoint and serializer. + The new locks are held a brief time and won't overlap other existing lock + times. + + ASTERISK-27090 #close + + Change-Id: I9ed63f4da9649e9db6ed4be29c360968917a89bd + +2017-06-29 18:22 +0000 [65a5ac0168] Richard Mudgett + + * pjsip_distributor.c: Fix unidentified_requests hash functions. + + The OBJ_SEARCH_xxx defines should not be used as if they were individual + bits. They represent a multi-bit enumeration value field. + + Change-Id: I32abc9a475396dab02402a7014357dd94284e17b + +2017-06-29 15:06 +0000 [e7d41050e0] Kevin Harwell + + * app_stream_echo: misc bug fixes + + Fixed the following bugs: + + * calls to stream_echo_write had the last two parameters swapped + * ast_read should have been ast_read_stream + * added a null check on the frame's subclass format + + This also resets the update_sent flag upon receiving SRRCHANGE control frame. + This will then force a video update. + + ASTERISK-26997 + + Change-Id: I6ad7c8253559b800800433c52339e7f5aa583566 + +2017-06-29 14:56 +0000 [7df7b8a90c] Kevin Harwell + + * res_rtp_asterisk: trigger source change control frame when dtls is established + + There needed to be a way to notify handlers upstream that DTLS had been + established. This patch makes it so once DTLS has been estalished a source + change control frame is put into the read queue. Any handlers can then watch + for that frame and trigger off of it. + + ASTERISK-27096 #close + + Change-Id: I27ff344f5a8c691a1890dfe3254a4b1a49e7f4a0 + +2017-06-30 08:31 +0000 [f573e599c0] George Joseph + + * pjproject_bundled: Allow passing configure options to bundled + + There wasn't any good way to pass options like --host or --build + down to the pjproject configure which makes cross-compiling difficult. + + * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which + can be used to pass arbitrary options to pjproject configure. + * Automatically set the pjproject configure --host and --build + options to match those supplied for the asterisk configure. + + ASTERISK-27097 #close + Reported-by: Kinsey Moore + + Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e + +2017-06-29 14:50 +0000 [c0c99c7618] George Joseph + + * chan_pjsip: Fix ability to send UPDATE on COLP + + When connected_line_method is "invite", we're supposed to determine + if the client can support UPDATE and if it can, send UPDATE instead + of INVITE to avoid the SDP renegotiation. Not only was pjproject + not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing + that invite_tsx wasn't NULL which isn't always the case. + + * Updated chan_pjsip/update_connected_line_information to drop the + requirement that invite_tsx isn't NULL. + * Submitted patch to pjproject sip_inv.c that sets the + PJSIP_INV_SUPPORT_UPDATE flag correctly. + * Updated pjsip.conf.sample to clarify what happens when "invite" + is specified. + + ASTERISK-27095 + + Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560 + +2017-06-15 03:12 +0000 [fb7247c57c] Torrey Searle + + * res_pjsip: Add DTMF INFO Failback mode + + The existing auto dtmf mode reverts to inband if 4733 fails to be + negotiated. This patch adds a new mode auto_info which will + switch to INFO instead of inband if 4733 is not available. + + ASTERISK-27066 #close + + Change-Id: Id185b11e84afd9191a2f269e8443019047765e91 + +2017-06-29 03:47 +0000 [ab7d99e62d] Niklas Larsson + + * app_queue: Add priority to AMI QueueStatus + + Add priority to callers in AMI QueueStatus response + + ASTERISK-27092 #close + + Change-Id: I8d1f737a72c7c38f4cfe1a4ee3ecc0a4f85bd199 + +2017-05-30 09:12 +0000 [45df25a579] Mark Michelson + + * chan_pjsip: Add support for multiple streams of the same type. + + The stream topology (list of streams and order) is now stored with the + configured PJSIP endpoints and used during the negotiation process. + + Media negotiation state information has been changed to be stored + in a separate object. Two of these objects exist at any one time + on a session. The active media state information is what was previously + negotiated and the pending media state information is what the + media state will become if negotiation succeeds. Streams and other + state information is stored in this object using the index (or + position) of each individual stream for easy lookup. + + The ability for a media type handler to specify a callback for + writing has been added as well as the ability to add file + descriptors with a callback which is invoked when data is available + to be read on them. This allows media logic to live outside of + the chan_pjsip module. + + Direct media has been changed so that only the first audio and + video stream are directly connected. In the future once the RTP + engine glue API has been updated to know about streams each individual + stream can be directly connected as appropriate. + + Media negotiation itself will currently answer all the provided streams + on an offer within configured limits and on an offer will use the + topology created as a result of the disallow/allow codec lines. + + If a stream has been removed or declined we will now mark it as such + within the resulting SDP. + + Applications can now also request that the stream topology change. + If we are told to do so we will limit any provided formats to the ones + configured on the endpoint and send a re-invite with the new topology. + + Two new configuration options have also been added to PJSIP endpoints: + + max_audio_streams: determines the maximum number of audio streams to + offer/accept from an endpoint. Defaults to 1. + + max_video_streams: determines the maximum number of video streams to + offer/accept from an endpoint. Defaults to 1. + + ASTERISK-27076 + + Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7 + +2017-06-28 09:03 +0000 [642f8356ab] Joshua Colp + + * res_rtp_asterisk: Fix issues with ICE renegotiation. + + When re-inviting to add more streams it is possible for + the role of existing ICE sessions to be changed to the + incorrect value. This results in subsequent refreshes + within the sessions getting a role conflict and the ICE + session breaking down. This change only sets the role to + be the new value if an ICE renegotiation is actually + going to happen, otherwise the existing role is preserved. + + As well if we encounter a situation where a unidirectional + ICE negotiation happens and the other side does not send us + candidates we will not store any information for sending + traffic, even though we know where they are reachable. This + change fixes this by using the source of the ICE traffic + itself as the target if no candidates are known and we + receive some ICE traffic. + + ASTERISK-27088 + + Change-Id: I71228181e358917fcefc3100fad21b2fc02a59a9 + +2017-06-27 10:46 +0000 [a48d3e4d31] Torrey Searle + + * res/res_pjsip_t38: fix incorrect increment of media_count + + The T38 sdp callback incorrectly has a side effect of incrementing + the media_count. This can lead to core dumps. + + Change-Id: I7bb2f4987de4046ec52cfc34e5ea0662dae32af8 + +2017-06-08 22:50 +0000 [80e11bd79b] George Joseph + + * bridge_native_rtp: Keep rtp instance refs on bridge_channel + + There have been reports of deadlocks caused by an attempt to send a frame + to a channel's rtp instance after the channel has left the native bridge + and been destroyed. This patch effectively causes the bridge channel to + keep a reference to the glue and both the audio and video rtp instances + so what gets started will get stopped. + + ASTERISK-26978 #close + Reported-by: Ross Beer + + Change-Id: I9e1ac49fa4af68d64826ccccd152593cf8cdb21a + +2017-06-27 04:37 +0000 [7827755570] Ivan Poddubny + + * app_queue: Fix returning to dialplan when a queue is empty + + The fix for ASTERISK-25665 introduced a regression. + The return value of queue_exec used to be 0 in case of leavewhenempty + but it was changed to -1 (returned from wait_our_turn and passed + transparently by queue_exec), thus leading to hangup instead of returning + back to dialplan. + + This commit resets the value back to 0 in this case, restoring + original behavior. + + ASTERISK-27065 #close + Reported by: Marek Cervenka + + Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac + +2017-06-19 17:21 +0000 [0cef7b9d4e] Alexei Gradinari + + * app_voicemail: IMAP connection control + + A new global option "imap_poll_logout" was added to specify whether need to + disconnect from the IMAP server after polling of mailboxes. + + ASTERISK-27068 #close + + Closing IMAP connection after loading mailbox from voicemail.conf + + ASTERISK-24052 #close + + Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a + +2017-06-21 17:57 +0000 [975e271b01] Richard Mudgett + + * res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer + + Change-Id: I0bc97c6608de1d1a4228826b3b3be43f162f05f3 + +2017-06-16 18:08 +0000 [34db4c3993] Alexei Gradinari + + * res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact + + Do not need to unsubscribe/subscribe on creating the ednpoint's contact. + The modified function create_mwi_subscriptions_for_endpoint adds + the subscription only if it does not exist. + + The subscriptions aren't added for active contacts + which are retrieved on startup from realtime + if mwi_disable_initial_unsolicited=yes. + Because the mwi_contact_added is not called. + So the subscriptions also should be created on updating contact. + + ASTERISK-26230 #close + + Change-Id: I47e265af9296ca09aa42a316fdacac104148cee4 + +2017-06-20 16:05 +0000 [27dae55fb6] Kevin Harwell + + * core_local: local channel data not being properly unref'ed and unlocked + + In an earlier version of Asterisk a local channel [un]lock all functions were + added in order to keep a crash from occurring when a channel hung up too early + during an attended transfer. Unfortunately, when a transfer failure occurs and + depending on the timing, the local channels sometime do not get properly + unlocked and deref'ed after being locked and ref'ed. This happens because the + underlying local channel structure gets NULLed out before unlocking. + + This patch reworks those [un]lock functions and makes sure the values that get + locked and ref'ed later get unlocked and deref'ed. + + ASTERISK-27074 #close + + Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09 + +2017-06-20 16:01 +0000 [45a1f4e2ae] Kevin Harwell + + * bridge: stuck channel(s) after failed attended transfer + + If an attended transfer failed it was possible for some of the channels + involved to get "stuck" because Asterisk was not hanging up the transfer target. + + This patch ensures Asterisk hangs up the transfer target when an attended + transfer failure occurs. + + ASTERISK-27075 #close + + Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9 + +2017-06-19 11:28 +0000 [a7488f8a70] Rodrigo Ramírez Norambuena + + * cdr: fix mistake spelling of a word for Unanswered. + + Change-Id: I7a610bef369924523a445c7e849ee88cc45dc5df + +2017-06-12 16:17 +0000 [d7b6e06abb] Alexei Gradinari + + * res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact + + If the endpoint's last contact is deleted unsolicited MWI has to be + unsubscribed. + + ASTERISK-27051 #close + + Change-Id: I33e174e0b9dba0998927d16d6d100fda5c7254e0 + +2017-06-16 09:31 +0000 [854a6de819] George Joseph + + * res_stasis: Plug reference leak on stolen channels + + When a stasis channel is stolen by another app, the control + structure is unreffed but never unlinked from the app_controls + container. This causes the channel reference to leak. + + Added OBJ_UNLINK to the callback in channel_stolen_cb. + + Also added some additional channel lifecycle debug messages to + channel.c. + + ASTERISK-27059 #close + Repoorted-by: George Joseph + + Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14 + +2017-06-16 14:56 +0000 [e33bd96638] Matthew Fredrickson + + * formats/format_g729: Fix typo in comment + + There was a typo in a comment. This commit is to fix the typo. + + ASTERISK-27060 #close + + Change-Id: Ic2699f8dbeaacd58ccb6ec3203e853e1babe3235 + +2017-06-08 12:28 +0000 [0ad95bc8a0] Frederic LE FOLL + + * Core/PBX: Deadlock between dialplan execution and application unregistration. + + Not easy to reproduce, but we have noticed deadlocks when unloading a module + while dialplan is handling a request. + + The deadlock is between : + 1) Dialplan execution: pbx_extension_helper() first taking conlock, + then pbx_findapp() [when called] asking for lock on apps list. + 2) Application unregistration: ast_unregister_application() first taking lock + on apps list, then unreference_cached_app() [when called] asking for conlock. + + As a protection, I suggest to modify ast_unregister_application(), so that it + anticipates the need of conlock, before taking the lock on apps list. + The side effect is a longer unavailability of conlock when unregistering an + application. + + ASTERISK-27041 + + Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2 + +2017-06-12 09:23 +0000 [7a46309d3d] Alexei Gradinari + + * res_pjsip: New endpoint option "notify_early_inuse_ringing" + + This option was added to control whether to notify dialog-info state + 'early' or 'confirmed' on Ringing when already INUSE. + The value "yes" is useful for some SIP phones (Cisco SPA) + to be able to indicate and pick up ringing devices. + + ASTERISK-26919 #close + + Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711 + +2017-06-15 13:48 +0000 [53b7df82f4] Alexei Gradinari + + * app_voicemail: IMAP logout on reload/unload + + Closing IMAP connection on module reload or unload. + + ASTERISK-24052 #close + + Change-Id: I2a40182aa9ef249fa6865d33570430e9ada68525 + +2017-03-30 09:33 +0000 [9aeab4aced] Jan Friesse + + * res_corosync: Change thread stack size + + In Corosync 2.x libraries were changed to use LibQB IPC. + Sadly LibQB IPC doesn't support copy-free access to received buffer, so + Corosync libraries were rewritten to use stack as buffer. Mostly the + needed stack size is quite small, but for all *_dispatch functions, 1MiB + is needed. + + Asterisk function ast_pthread_create_background set stack size for new + thread to much smaller AST_BACKGROUND_STACKSIZE (~500KiB). + + This results in Asterisk crash when running with Corosync 2.x. + + Patch solves this issue by creating it's own version of + ast_pthread_create_background which sets stack size to much higher value + (actually it's AST_BACKGROUND_STACKSIZE + 3MiB). + + Another problem may appear when "corosync show members" netconsole + command is executed. It is also executed in thread and also has only + 500KiB stack size. Sadly it calls corosync_cfg_get_node_addrs which + again needs at least 1MiB stack. + + Solution is to use HAVE_COROSYNC_CFG_STATE_TRACK as a discriminator + between Corosync 1.x and 2.x. If 1.x is found, nothing changes. If 2.x + is found, NodeID is displayed instead of IP address. + + ASTERISK-25370 #close + Reported by: mdu113 + + Change-Id: Id95b0d21ab6e708e7d74ad8786c587211676fa08 + +2017-06-13 11:33 +0000 [1ac0096512] George Joseph + + * res_ari: Add "module loaded" check to ari stubs + + The recent change to make the use of LOAD_DECLINE more consistent + caused res_ari to unload itself before declining if the ari.conf + file wasn't found. The ari stubs though still tried to use the + configuration resulting in segfaults. + + This patch creates a new CHECK_ARI_MODULE_LOADED macro which tests + to see if res_ari is actually loaded and causes the stubs to also + decline if it isn't. The macro was then added to the mustache + template's "load_module" function. + + ASTERISK-27026 #close + Reported-by: Ronald Raikes + + Change-Id: I263d56efa628ee3c411bdcd16d49af6260c6c91d + +2017-06-15 12:33 +0000 [11ec2945c7] Richard Mudgett + + * chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read. + + The construction of the returned string assumed incorrectly that the + supplied buffer would always be initialized as an empty string. If it is + not an empty string we could overrun the supplied buffer by the length of + the non-empty buffer string plus one. It is also theoreticaly possible + for the supplied buffer to be overrun by a string terminator during a read + operation even if the supplied buffer is an empty string. + + * Fix the assumption that the supplied buffer would already be an empty + string. The buffer is not guaranteed to contain an empty string by all + possible callers. + + * Fix string terminator buffer overrun potential. + + Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9 + +2017-06-08 11:38 +0000 [e563a1920e] Richard Mudgett + + * SDP: Add get/set option calls for RTP sched context per type. + + Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4 + +2017-05-11 18:49 +0000 [716abaf33d] Richard Mudgett + + * SDP: Search for the ice-lite attribute in the right place. + + * Pulled finding the rtcp-mux attribute flag out of the ICE candidate for + loop. Also ordered the RTCP ICE candidate skip test to fail earlier. + + Change-Id: I8905d9c68563027a46cd3ae14dbcc27e9c814809 + +2017-05-11 18:46 +0000 [a95584d079] Richard Mudgett + + * SDP: Set the remote c= line in RTP instance. + + Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c + +2017-06-09 19:03 +0000 [06265b8c8a] Richard Mudgett + + * stream: Add ast_stream_topology_del_stream() and unit test. + + Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9 + +2017-05-11 14:09 +0000 [0fdb99c268] Richard Mudgett + + * SDP: Add t= line in sdp_create_from_state() + + Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141 + +2017-06-14 13:07 +0000 [4797a8bb81] Richard Mudgett + + * stream: Ignore declined streams for some topology calls. + + * Made ast_format_cap_from_stream_topology() not include any formats from + declined streams. + + * Made ast_stream_topology_get_first_stream_by_type() ignore declined + streams to return the first active stream of the type. + + * Updated unit tests to check these changes have the expected effect. + + Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df + +2017-06-15 07:32 +0000 [bd16c3c524] Joshua Colp + + * channel: Fix reference counting in ast_channel_suppress. + + The ast_channel_suppress function wrongly decremented the + reference count of the underlying structure used to keep + track of what should be suppressed on a channel if the + function was called multiple times on the same channel. + + This change cleans up the reference counting a bit so + this no longer occurs. + + ASTERISK-27016 + + Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136 + +2017-06-14 12:34 +0000 [b8b0b61a24] Richard Mudgett + + * app_voicemail.c: Fix compile error when IMAP enabled. + + Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1 + +2017-06-12 17:55 +0000 [023eede265] Alexei Gradinari + + * app_voicemail: IMAP logout on MWI unsubscribe + + Closing IMAP connection on MWI unsubscribe. + + ASTERISK-24052 #close + + Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd + +2017-06-14 11:12 +0000 [65ed2ea311] George Joseph + + * res_pjsip_pubsub: Fix reference to released endpoint + + destroy_subscription was attempting to get the id of the + subscription tree's endpoint after we'd already called ao2_cleanup + on it causing a segfault. + + Moved the cleanup until after the debug statement and since + endpoint could also be NULL at this point, check for that as well. + + ASTERISK-27057 #close + Reported-by: Ryan Smith + + Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678 + +2017-06-14 08:29 +0000 [ea3f8c6889] George Joseph + + * res_pjsip_session: Correct inverted test in session_outgoing_nat_hook + + There was a typo introduced in commit 776ffd77 which was preventing + the transport's external media address from being used. + + ASTERISK-27024 #close + Reported-by: Christopher van de Sande + patches: + patch.diff submitted by Florian Floimair (license 6892) + + Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27 + +2017-06-14 08:54 +0000 [88f18faf2a] George Joseph + + * res_rtp_asterisk: Fix ssrc change for rtcp srtp + + It looks like there was a copy/paste error in ast_rtp_change_source + where if there was a rtcp srtp instance, instead of updating its + ssrc we were updating the srtp instance ssrc twice. + + ASTERISK-27022 #close + Reported-by: Michael Walton + + Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095 + +2017-06-08 14:38 +0000 [d6386a8f0c] Joshua Colp + + * bridge: Add a deferred queue. + + This change adds a deferred queue to bridging. If a bridge + technology determines that a frame can not be written and + should be deferred it can indicate back to bridging to do so. + Bridging will then requeue any deferred frames upon a new + channel joining the bridge. + + This change has been leveraged for T.38 request negotiate + control frames. Without the deferred queue there is a race + condition between the bridge receiving the T.38 request + negotiate and the second channel joining and being in the + bridge. If the channel is not yet in the bridge then the T.38 + negotiation fails. + + A unit test has also been added that confirms that a T.38 + request negotiate control frame is deferred when no other + channel is in the bridge and that it is requeued when a new + channel joins the bridge. + + ASTERISK-26923 + + Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415 + +2017-06-13 14:17 +0000 [9e53c30610] Kevin Harwell + + * res_pjsip_refer/session: Calls dropped during transfer + + When doing an attended transfer it's possible for the transferer, after + receiving an accepted response from Asterisk, to send a BYE to Asterisk, + which can then be processed before Asterisk has time to start and/or + complete the transfer process. This of course causes the transfer to not + complete successfully, thus dropping the call. + + This patch makes it so any BYEs received from the transferer, after the REFER, + that initiate a session end are deferred until the transfer is complete. This + allows the channel that would have otherwise been hung up by Asterisk to + remain available throughout the transfer process. + + ASTERISK-27053 #close + + Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a + +2017-06-13 10:47 +0000 [b2fd7e5069] George Joseph + + * pjproject_bundled: Use the asterisk github mirror for download + + We now mirror the pjproject tarball and md5 at + https://github.com/asterisk/third-party/tree/master/pjproject + + To improve download reliability, we now get the tarball from + our mirror instead of from pjsip.org. + + ASTERISK-27052 #close + Reported-by: 'alex' + + Change-Id: I60236587a8935bfa71fcc391f4e2ecb31918c08a + +2017-06-12 09:57 +0000 [42f738e052] Alexei Gradinari + + * res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled + + If sending unsolicited mwi to all endpoints on startup is disabled + (mwi_disable_initial_unsolicited=yes) do not need to create subscriptions. + If there are many (thousands) realtime endpoints configured with unsolicited mwi + and Vociemail Storage configured as ODBC or IMAP there will be huge number of + DB/IMAP requests on startup. + + ASTERISK-26230 #close + + Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5 + +2017-06-11 12:06 +0000 [847087a4ff] Sean Bright + + * codecs.conf.sample: Fix max_bandwidth speling error + + Reported by Sylvain Boily via asterisk-dev mailing list. + + Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d + +2017-06-08 17:31 +0000 [8d1f54b92e] Jørgen H + + * res_pjsip_transport_websocket: Add NULL check in get_write_timeout + + Added check for NULL return value when calling + ast_sorcery_retrieve_by_id in function get_write_timeout + + ASTERISK-27046 + + Change-Id: I9357717278da631c3a1cb502c412693929b0cb41 + +2017-06-08 10:54 +0000 [d27168d36f] Guido Falsi + + * BuildSystem: Add patches to allow building with recent LibreSSL + + Add some #if defined checks which allow building against LibreSSL. + These patchess come from OpenBSD ports: + https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/ + + ASTERISK-27043 #close + Reported by: OpenBSD ports + + Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572 + +2017-06-06 14:54 +0000 [fcb1a0d7e8] David M. Lee + + * CFLAGS for BIND8 support + + Some systems (like macOS) require BIND_8_COMPAT to be defined so that + the nameser libraries are, well, BIND8 compatible. + + Change-Id: If79fc27a64f90de1835b5aa3aadfa9be22bd16b0 + +2017-06-08 10:36 +0000 [7b668297f3] Guido Falsi + + * BuildSystem: Fix build on FreeBSD due to missing crypt.h + + FreeBSD does not include a crypt.h include file. Definitions for + crypt() and crypt_r() are in unistd.h + + ASTERISK-27042 #close + + Change-Id: Ib307ee5e384870c6af50efa89fb73722dd0c3a7e + +2017-06-07 15:19 +0000 [5b80496b42] Joshua Colp + + * chan_pjsip: Update device state when in early media. + + The chan_pjsip module uses a calculation approach for + determining device state. This means that in situations + where we would expect device state to change we need to + tell the core to query. A scenario that was missed is + when early media was signaled. + + This change adds the notification for the core to + query device state when we are told that early media + is being provided. + + ASTERISK-27039 + + Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f + +2017-06-07 14:32 +0000 [e497a76d24] Sean Bright + + * eventfd: Disable during cross compilation + + Reported by Lonnie Abelbeck via private e-mail. + + Change-Id: Icc80f12b8d8d591e14a8e0ed9f1c02cbd193a89b + +2017-06-07 11:21 +0000 [19da99df2f] Alexei Gradinari + + * CHANGES: correct version for a new option 'refer_blind_progress' + + Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97 + +2017-06-06 07:04 +0000 [d3e951edf5] Joshua Colp + + * pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. + + PJSIP support in Asterisk differs from chan_sip in that it + allows media to be sent as-is without transcoding provided + the codecs were negotiated in the SDP. This is allowed + according to the RFC. Support for this differs quite a lot + though and some endpoints do not handle it well. + + This change extends the 'asymmetric_rtp_codec' option to + also cover this case. When set to no (the default) the code + behaves as chan_sip does - the best codec is selected and + we will only ever send that, unless we change what we are + sending if the remote side changes. When set to yes we + will send media as-is without transcoding if the codec + has been negotiated in the SDP. + + ASTERISK-26996 + + Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51 + +2017-06-06 10:04 +0000 [b3ca24d216] Sean Bright + + * res_rtp_multicast: Use consistent timestamps when possible + + When a frame destined for a MulticastRTP channel does not have timing + information (such as when an 'originate' is done), we generate the RTP + timestamps ourselves without regard to the number of samples we are + about to send. + + Instead, use the same method as res_rtp_asterisk and 'predict' a + timestamp given the number of samples. If the difference between the + timestamp that we generate and the one we predict is within a specific + threshold, use the predicted timestamp so that we end up with timestamps + that are consistent with the number of samples we are actually sending. + + Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f + +2017-05-31 10:41 +0000 [861984eac0] Joshua Colp + + * res_pjsip: Add support for returning only reachable contacts and use it. + + This introduces the ability for PJSIP code to specify filtering flags + when retrieving PJSIP contacts. The first flag for use causes the + query code to only retrieve contacts that are not unreachable. This + change has been leveraged by both the Dial() process and the + PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt + calls to contacts which are not unreachable. + + ASTERISK-26281 + + Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c + +2017-06-05 11:27 +0000 [d8802a6a0f] Kevin Harwell + + * channel: ast_write frame wrongly freed after call to audiohooks + + ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in + ast_write. It would free the frame given to ast_write if the frame returned + by ast_audiohook_write_list was different than the given one. The frame give + to ast_write should never be freed within that function. It is the caller's + resposibility to free the frame after writing (or when it its done with it). + By freeing it within ast_write this of course led to some memory corruption + problems. + + This patch makes it so the frame given to ast_write is no longer freed within + the function. The frame returned by ast_audiohook_write_list is now subsequently + used in ast_write and is freed later. It is freed either after translate if the + frame returned by translate is different, or near the end of ast_write prior to + function exit. + + ASTERISK-26973 #close + + Change-Id: Ic9085ba5f555eeed12f6e565a638c3649695988b + +2017-05-31 11:45 +0000 [001f4ddda4] Sean Bright + + * pbx_builtin: Properly handle hangup during Background + + Before this patch, when a user hung up during a Background, we would + stuff 0xff into a char and attempt a dialplan lookup of it. This caused + problems for some realtime engines which interpreted the value as the + beginning of an invalid UTF-8 sequence. + + ASTERISK-19291 #close + Reported by: Andrew Nowrot + + Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358 + +2017-05-31 04:25 +0000 [f6eeaaafd5] Joshua Colp + + * channel / app_meetme: Fix parentheses. + + ASTERISK-27025 + + Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed + +2017-05-30 16:07 +0000 [9dce4a947b] Sean Bright + + * stasis_recording: Correct ast_asprintf error checking + + ASTERISK-27021 #close + Reported by: Tim Morgan + + Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75 + +2017-05-28 15:43 +0000 [5c27fe2187] Sean Bright + + * format: Reintroduce smoother flags + + In review 4843 (ASTERISK-24858), we added a hack that forced a smoother + creation when sending signed linear so that the byte order was adjusted + during transmission. This was needed because smoother flags were lost + during the new format work that was done in Asterisk 13. + + Rather than rolling that same hack into res_rtp_multicast, re-introduce + smoother flags so that formats can dictate their own options. + + Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16 + +2017-05-24 10:09 +0000 [39d14834f8] Mark Michelson + + * Confbridge: Add "sfu" video mode to bridge profile options. + + A previous commit added plumbing to bridge_softmix to allow for an SFU + experience with Asterisk. This commit adds an option to app_confbridge + that allows for a confbridge to actually make use of the SFU video mode. + + SFU mode is implemented in a "set it and forget it" kind of way. That + is, when the bridge is created, if SFU mode is enabled, then the video + mode gets set to SFU and cannot be changed. Future improvements may + allow for a hybrid experience (e.g. forward multiple video streams, + specifically those of the most recent talkers), but for this addition, + no such capability is present. + + Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020 + +2017-05-05 11:56 +0000 [2da869408a] Mark Michelson + + * Add primitive SFU support to bridge_softmix. + + This sets up the "plumbing" in bridge_softmix to + be able to accommodate Asterisk asking as an SFU + (selective forwarding unit) for conferences. + + The way this works is that whenever a channel enters or leaves a + conference, all participants in the bridge get sent a stream topology + change request. The topologies consist of the channels' original + topology, along with video destination streams corresponding to each + participants' source video streams. So for instance, if Alice, Bob, and + Carol are in the conference, and each supplies one video stream, then + the topologies for each would look like so: + + Alice: + Audio, + Source video(Alice), + Destination Video(Bob), + Destination video (Carol) + + Bob: + Audio, + Source video(Bob) + Destination Video(Alice), + Destination video (Carol) + + Carol: + Audio, + Source video(Carol) + Destination Video(Alice), + Destination video (Bob) + + This way, video that arrives from a source video stream can then be + copied out to the destination video streams on the other participants' + channels. + + Once the bridge gets told that a topology on a channel has changed, the + bridge constructs a map in order to get the video frames routed to the + proper destination streams. This is done using the bridge channel's + stream_map. + + This change is bare-bones with regards to SFU support. Some key features + are missing at this point: + + * Stream limits. This commit makes no effort to limit the number of + streams on a specific channel. This means that if there were 50 video + callers in a conference, bridge_softmix will happily send out topology + change requests to every channel in the bridge, requesting 50+ + streams. + + * Configuration. The plumbing has been added to bridge_softmix, but + there has been nothing added as of yet to app_confbridge to enable SFU + video mode. + + * Testing. Some functions included here have unit tests. + However, the functionality as a whole has only been verified by + hand-tracing the code. + + * Selectivenss. For a "selective" forwarding unit, this does not + currently have any means of being selective. + + * Features. Presumably, someone might wish to only receive video from + specific sources. There are no external-facing functions at the moment + that allow for users to select who they receive video from. + + * Efficiency. The current scheme treats all video streams as being + unidirectional. We could be re-using a source video stream as a + desetnation, too. But to simplify things on this first round, I did it + this way. + + Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d + +2017-05-30 09:34 +0000 [045d7b8cb7] Sean Bright + + * format_mp3: Re-work menuselect/build issues + + Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3 + to not show up in menuselect), use .PHONY targets when the necessary + source files are not present. + + ASTERISK-23951 + Reported by: Tzafrir Cohen + + Change-Id: I0a7512c51acc9e86043671795020b0de725bd9e8 + +2017-05-30 09:43 +0000 [80206cdc65] George Joseph + + * test_json: Fix test names with reserved words + + Some of the test names were actually reserved words (true, false, + int, null, string, bool). When the jenkins test results analyzer + does its thing it tries to create a map using the test names as + keys and fails because they're reserved words. + + Added "type_" to those test names. + + Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b + +2017-05-26 11:41 +0000 [9c4f63263c] Joshua Colp + + * manager: Clear the flag on the other channel. + + During the channel flag audit an incorrect change was + done. The flag should be cleared on the second channel. + + ASTERISK-26469 + + Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8 + +2017-05-26 11:15 +0000 [1f136fe885] Sean Bright + + * res_srtp: Add support for libsrtp2 + + ASTERISK-25294 #close + Reported by: Tzafrir Cohen + + ASTERISK-26976 #close + Reported by: Alex + + Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40 + +2017-05-25 11:10 +0000 [59348aa182] Sean Bright + + * format_mp3: Don't try to build format_mp3 if we don't have sources + + ASTERISK-23951 #close + Reported by: Tzafrir Cohen + + Change-Id: Iebf181d44bb735787fde4b5be863c4d7e2478a30 + +2017-05-23 11:07 +0000 [44c5a144ce] Martin Tomec + + * Sqlite3: make busy_timeout configurable. + + Enables runtime configuration of busy_timeout for sqlite databases. + Default timeout remains 1000ms. + + ASTERISK-27014 #close + + Change-Id: I8921a3aac3c335843be4cb17d2dd0a5c157a36da + +2017-05-24 15:50 +0000 [08edd54c1b] George Joseph + + * unittests: Add a unit test that causes a SEGV and... + + ...that can only be run by explicitly calling it with + 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' + + This allows us to more easily test CI and debugging tools that + should do certain things when asterisk coredumps. + + To allow this a new member was added to the ast_test_info + structure named 'explicit_only'. If set by a test, the test + will be skipped during a 'test execute all' or + 'test execute category ...'. + + Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed + +2017-05-23 15:42 +0000 [d847fe6585] Sean Bright + + * res_agi: Allow configuration of audio format of EAGI pipe + + This change allows the format of the EAGI audio pipe to be changed by + setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of + the loaded formats. + + ASTERISK-26124 #close + + Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd + +2017-05-23 13:33 +0000 [e2e6baa8d8] Sean Bright + + * res_agi: Clarify 'RECORD FILE' documentation + + Documented the 'beep' option in both the parameters list and the command + description. + + ASTERISK-23839 #close + + Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea + +2017-05-23 13:06 +0000 [3dcb3c88aa] Sean Bright + + * res_agi: Prevent crash when SET VARIABLE called without arguments + + Explicitly check that the appropriate number of arguments were passed to + SET VARIABLE before attempting to reference them. Also initialize the + arguments array to zeroes before populating it. + + ASTERISK-22432 #close + + Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97 + +2017-05-23 12:35 +0000 [e490aa3176] Sean Bright + + * res_agi: Fix malformed AGI usage response + + If the generated XML documentation for a command does not end with a \n, + the postamble of the usage message does not appear on its own line. + + ASTERISK-25662 #close + + Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020 + +2017-05-23 10:06 +0000 [8ae0227cf3] Sean Bright + + * res_format_attr_h26x: Trim blanks in fmtp attributes + + Some devices separate format attributes with a semicolon followed by a + space, so trim blanks before trying to match them. + + ASTERISK-27008 #close + + Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc + +2017-05-15 15:03 +0000 [faab058014] Joshua Colp + + * app_queue: Fix members showing as being in call when not. + + A change was done which added an 'in_call' flag to queue + members that was set to true while talking to an agent. + Unfortunately in practice this does not accurately reflect + whether they are talking to an agent or not. If a Local + channel is involved and a transfer is performed then the + app_queue application would incorrectly think the agent + was still in a call with the caller. This was done to + fix a race condition between an agent becoming available + by device state and the checking of the last call information + for the wrapup time. There was a small window where the + last call information would be the previous value instead + of the new one. + + This change goes about fixing the original issue in a + different way by considering the call completed if device + state is received which would make the agent available + and if they are currently in a call. If this occurs the + last call information is updated before the agent becomes + available ensuring that old information is not present + when checking if the member should be called. This also + improves the transfer situation by actually updating + and enforcing the wrapup time. + + ASTERISK-26399 + ASTERISK-26400 + ASTERISK-26715 + ASTERISK-26975 + + Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea + +2017-05-23 05:45 +0000 [36e90952ec] Robert Mordec + + * app_confbridge: Race between removing and playing name recording while leaving + + When user leaves a conference, its channel calls async_play_sound_file() + in order to play the name announcement and then unlinks the sound file. + The async_play_sound_file() function adds a task to conference playback queue, + which then runs playback_common() function in a different thread. + + It leads to a race condition when, in some cases, channel thread may unlink + the sound file before playback_common() had a chance to open it. + + This patch creates a file deletion task, that is queued after playback. + + ASTERISK-27012 #close + + Change-Id: I412f7922d412004b80917d4e892546c15bd70dd3 + +2017-05-22 13:51 +0000 [440ff38c08] Kevin Harwell + + * res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm + + When using rtcp mux if an rtcp payload came in it would still use the srtp + unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp + data was being passed to the rtp unprotect method this would result in an + error. + + This patch ensures that the correct unprotect method is chosen by making + sure the passed in rtcp flag is appropriately set when rtcp mux is enabled + and an rtcp payload is received. + + ASTERISK-26979 #close + + Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241 + +2017-05-19 10:05 +0000 [0f487978a9] Sean Bright + + * chan_sip: Better ICE handling for RTCP-MUX + + If we are offered or are offering RTCP-MUX, don't consider RTCP ICE + candidates. This confuses certain browsers (current Firefox for + example) and causes intial audio setup delays. + + ASTERISK-26982 #close + + Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91 + +2017-05-12 10:38 +0000 [be4beff3e4] Steve Davies + + * app_queue: Add QUEUE_RAISE_PENALTY feature + + Additional variable to work alongside QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY, + including an extra parameter in queuerules.conf. This value causes lower + Agent penalty values to "raise up" so that they can join higher penalty agents + and be treated equally after a period of time. + + ASTERISK-26995 #close + + Change-Id: If1c6421a983667a5ac4c359f6dac25b212b4c459 + +2017-04-13 17:17 +0000 [7c0466092c] Mark Michelson + + * AST-2017-003: Handle zero-length body parts correctly. + + ASTERISK-26939 #close + + Change-Id: I7ea235ab39833a187db4e078f0788bd0af0a24fd + +2017-04-13 11:14 +0000 [949e9147bf] George Joseph + + * AST-2017-004: chan_skinny: Add EOF check in skinny_session + + The while(1) loop in skinny_session wasn't checking for EOF so + a packet that was longer than a header but still truncated + would spin the while loop infinitely. Not only does this + permanently tie up a thread and drive a core to 100% utilization, + the call of ast_log() in such a tight loop eats all available + process memory. + + Added poll with timeout to top of read loop + + ASTERISK-26940 #close + Reported-by: Sandro Gauci + + Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898 + +2017-04-13 17:16 +0000 [2bb98d8fac] Mark Michelson + + * AST-2017-002: Ensure transaction key buffer is large enough. + + ASTERISK-26938 #close + + Change-Id: I266490792fd8896a23be7cb92f316b7e69356413 + +2017-05-18 16:35 +0000 [4141748e85] Sean Bright + + * res_hep_rtcp: Add support level to module info + + Change-Id: I5661478f9cf12d431f730e42be79323b62831e92 + +2017-05-15 13:26 +0000 [a60d1f3974] Kevin Harwell + + * app_stream_echo: Added a multi-stream echo application + + If the channel does not have multi-stream support then this application acts + just like app_echo. If it does have multi-stream support then each stream is + echoed back to itself (one-to-one). + + If a "num" is specified, then a new topology is made that contains clones (from + the channel's topology) of all media types that are not equal to the given + "type". If the media type differs then the first stream matching the "type" is + cloned into the new topology and then up to "num" - 1 of the same stream are + also cloned into it. Any additional streams from the original topology matching + the "type" are subsequently ignored (i.e. not added to the new topology). + + For this same case when a frame is read from a stream that frame is still + echoed back like before, but now that frame is also echoed out to the + additional streams that matched on the specified "type". + + ASTERISK-26997 #close + + Change-Id: I254144486734178e196c7f590a26ffc13543ff2c + +2017-05-15 13:25 +0000 [51375686f7] Kevin Harwell + + * core/conversions: Added string to unsigned integer and long conversions + + Added functions that convert a string to an unsigned integer or unsigned long. + A couple of unit test were also created to test the routines. The reasons for + adding these conversion utilities (and hopefully eventually more) are as + follows: + + * Conversion routines are functionally contained with consistent and + better error checking + * The function names offer a better description of what is happening + * It encourages code reuse for easier bug fixing at a single source + * It's simpler to use + * It's unit testable + + For instance, currently in a lot of places when converting to an integer or + similar the "sscanf" function is used. When using "sscanf" it may not be + immediately clear what's happening as it lacks semantic naming. Limited error + checking is usually done as well. For example, most of the time a check is done + to make sure the value converted, but does not check for overflows or negative + valued conversions when converting unsigned numbers. + + Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the + built in error handling that "strtoul" has. For instance "strtoul" contains + overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly + more complex in its use. And maybe a bit controversial, but it may be ("big if") + potentially slower than "strtoul" in some cases. + + Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb + +2017-05-13 11:40 +0000 [5a7af00e80] Joshua Colp + + * asterisk: Audit locking of channel when manipulating flags. + + When manipulating flags on a channel the channel has to be + locked to guarantee that nothing else is also manipulating + the flags. This change introduces locking where necessary to + guarantee this. It also adds helper functions that manipulate + channel flags and lock to reduce repeated code. + + ASTERISK-26789 + + Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10 + +2017-05-12 21:04 +0000 [30fbed65f1] Richard Mudgett + + * res_pjsip_session.c: Process initial INVITE sooner. (key exists) + + Retransmissions of an initial INVITE could be queued in the serializer + before we have processed the first INVITE message. If the first INVITE + message doesn't get completely processed before the retransmissions are + seen then we could try to setup the same call from the retransmissions. A + symptom of this is seeing a (key exists) message associated with an + INVITE. An earlier change attempted to address this kind of problem by + calculating a distributor serializer to use for unassociated messages. + Part of that change also made incoming calls keep using that distributor + serializer. (ASTERISK-26088) However, some leftover code was still + deferring the INVITE processing to the session's serializer even though we + were already in that serializer. This not only is unnecessary but would + cause the same call resetup problem. + + * Removed the code to defer processing the initial INVITE to the session's + serializer because we are already running in that serializer. + + ASTERISK-26998 #close + + Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6 + +2017-05-14 00:37 +0000 [6e7b78414f] Rodrigo Ramírez Norambuena + + * Fix spelling queues.conf.sample file + + Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee + +2017-05-08 13:40 +0000 [93b7f84c1a] Vitezslav Novy + + * chan_sip: Change sip_get_codec() to return correct codec list + + Return cahnnel nativeformats to fix bridge technology selection process. + Same approach as in pjsip module. + + ASTERISK-26143 + Reported-by: Henning Holtschneider + + Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48 + +2017-05-08 15:56 +0000 [808f299808] Alexei Gradinari + + * res_pjsip: New endpoint option "refer_blind_progress" + + This option was added to turn off notifying the progress details + on Blind Transfer. If this option is not set then the chan_pjsip + will send NOTIFY "200 OK" immediately after "202 Accepted". + + Some SIP phones like Mitel/Aastra or Snom keep the line busy until + receive "200 OK". + + ASTERISK-26333 #close + + Change-Id: Id606fbff2e02e967c02138457badc399144720f2 + +2017-05-11 00:25 +0000 [045dbcc2d6] Ivan Poddubny + + * app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON + + There are 2 places in app_queue.c that log EXITEMPTY event: one in + wait_our_turn, and another one in queue_exec in the loop trying to + call an agent after wait_our_turn. + + In most cases it leads to logging EXITEMPTY twice. + + ABANDON is also logged on two places, and in the rare case when an agent + and caller hang up simultaneously it's also possible to get duplicates + in queue_log. + + This commit changes wait_our_turn to return -1 ("the caller should exit + the queue") instead of 0 ("the caller's turn has arrived") in case of + leaving when empty, so queue_exec skips the agent calling loop. + + Also, leave_queue is now executed only once in this case, because 2nd + time is just a noop when the queue entry has already been removed. + + Also, it sets qe->handled to -1 to indicate that the call was not + answered by an agent, but the necessary handling has already been done + in order to avoid logging an extra ABANDON entry. + + ASTERISK-25665 #close + Reported by: Ove Aursand + + Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e + +2017-04-27 19:37 +0000 [b8659be9b0] Richard Mudgett + + * SDP: Make process possible multiple fmtp attributes per rtpmap. + + Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349 + +2017-04-28 11:53 +0000 [c2906dfa05] Richard Mudgett + + * SDP: Remove sdp_state.remote_capabilities + + The sdp_state.remote_capabilities was only used inside merge_sdps() and + subsequent calls to merge_sdps() by re-INVITE's would leak them. + + Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce + +2017-05-05 14:30 +0000 [16785c0908] Richard Mudgett + + * SDP: Add interface_address to specify our address to use. + + When we optionally set the interface_address we are forcing the media to + go out a specific interface address. This allows us to optionally have + the media go out the interface that SIP signalling came in on or if we are + configured to have the media always go out a specific address. + + Change-Id: I160d9fac322a075bd2557b430632544178196189 + +2017-05-05 14:49 +0000 [367042bd3e] Richard Mudgett + + * SDP: Explicitly stop a RTP instance before destoying it. + + * Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream() + handle generating disabled/declined streams. + + * Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not + check the offerer side negotiated SDP because that isn't the purpose of + this patch and there is much to be done to handle declined/dummy streams. + + * Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and + /main/sdp/sdp_merge_crisscross unit tests. + + Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31 + +2017-04-28 19:48 +0000 [be5809fac8] Richard Mudgett + + * SDP: Rework merge_capabilities(). + + * Tried to give better variable names. + * Made our SDP answer use the offer's RTP payload types as the SDP RFC + says we SHOULD. + * Updating the local topology now takes the stream format caps. We are + likely preparing to send an offer. + + Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0 + +2017-04-28 12:30 +0000 [ae7689f093] Richard Mudgett + + * SDP: Update ast_get_topology_from_sdp() to keep RTP map. + + * Add failure exits to ast_get_topology_from_sdp(). + + Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049 + +2017-05-09 10:34 +0000 [cbbd119c21] Joshua Colp + + * tcptls: Improve error messages for TLS connections. + + This change uses the functions provided by OpenSSL to query + and better construct error messages for situations where + the connection encounters a problem. + + ASTERISK-26606 + + Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b + +2017-05-04 17:28 +0000 [10a4439ac9] Joshua Elson + + * Prevent Undefined Capath Crash + + It is possible to initialize a valid config without a capath + or cafile definition. This will cause a crash on a reload. + + This fix ensures capath is always allocated. + + ASTERISK-26983 #close + + Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12 + +2017-05-05 11:33 +0000 [1a1c86239d] George Joseph + + * cel_odbc: Fix timestamp processing for microseconds + + When a column is of type timestamp, the fraction part of the event + field's seconds was frequently parsed incorrectly especially if + there were leading zeros. For instance "2017-05-23 23:55:03.023" + would be parsed into an int as "23" then when the timestamp was + formatted again to be inserted into the database column it'd be + "2017-05-23 23:55:03.23" which is now 230 milliseconds instead of + 23 milliseconds. "03.000001" would be transformed to "03.1", etc. + + * If the event field is 'eventtime' and the db column is timestamp, + then existing processing has already correctly formatted the + timestamp so now we simply use it rather than parsing it and + re-printing it. This is the most common use case anyway. + + * If the event field is other than 'eventtime' and the db column + is timestamp, we now parse the seconds, including the fractional + part into a double rather than 2 ints. This preserves the + magnitude and precision of the fractional part. When we print + it, we now print it as a "%09.6lf" which correctly represents the + input. + + To be honest, why we parse the string timestamp into components, + test the components, then print the components back into a string + timestamp is beyond me. We should use parse it, test it, then if + it passes, use the original string representation in the database + call. Maybe someone thought that some implementations wouldn't + take a partial timestamp string like "2017-05-06" and decided to + always produce a full timestamp string even if an abbreviated one + was supplied. Anyway, I'm leaving it as it is. + + ASTERISK-25032 #close + Reported-by: Etienne Lessard + + Change-Id: Id407e6221f79a5c1120e1a70bc7e893bbcaf1938 + +2017-05-09 05:25 +0000 [3c36c29c81] Joshua Colp + + * res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. + + This change adds the required logic to allow the SIP + Call-ID to be placed into the HEP RTCP traffic if the + chan_sip module is used. In cases where the option is + enabled but the channel is not either SIP or PJSIP then + the code will fallback to the channel name as done + previously. + + Based on the change on Nir's branch at: + team/nirs/hep-chan-sip-support + + ASTERISK-26427 + + Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d + +2017-05-08 16:11 +0000 [201346fb7d] George Joseph + + * logger: Added logger_queue_limit to the configuration options. + + All log messages go to a queue serviced by a single thread + which does all the IO. This setting controls how big that + queue can get (and therefore how much memory is allocated) + before new messages are discarded. The default is 1000. + Should something go bezerk and log tons of messages in a tight + loop, this will prevent memory escalation. + + When the limit is reached, a WARNING is logged to that effect + and messages are discarded until the queue is empty again. At + that time another WARNING will be logged with the count of + discarded messages. There's no "low water mark" for this queue + because the logger thread empties the entire queue and processes it + in 1 batch before going back and waiting on the queue again. + Implementing a low water mark would mean additional locking as + the thread processes each message and it's not worth it. + + A "test" was added to test_logger.c but since the outcome is + non-deterministic, it's really just a cli command, not a unit + test. + + Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1 + +2017-05-02 18:05 +0000 [56c5c51076] Richard Mudgett + + * stream: ast_stream_clone() cannot copy the opaque user data. + + ast_stream_clone() cannot copy the opaque user data stored on a stream. + We don't know how to clone the data so it isn't copied into the clone. + + Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367 + +2017-05-04 17:32 +0000 [924628812b] Richard Mudgett + + * netsock2.c: Made get/set addr port avoid potential uninitialized memory. + + Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647 + +2017-05-05 08:48 +0000 [4146facfec] Joshua Colp + + * func_cdr: Allow empty value for CDR dialplan function. + + A regression was introduced in 12 where passing an empty value + to the CDR dialplan function was not longer allowed. This + change returns to the behavior of 11 where it is permitted. + + ASTERISK-26173 + + Change-Id: I3f148203b54ec088007e29e30005a5de122e51c5 + +2017-05-04 16:04 +0000 [0001834157] George Joseph + + * app_confbridge: Fix reference to cfg in menu_template_handler + + menu_template_handler wasn't properly accounting for the fact that + it might be called both during a load/reload (which isn't really + valid but not prevented) and by a dialplan function. In both cases + it was attempting to use the "pending" config which wasn't valid in + the latter case. aco_process_config is also partly to blame because + it wasn't properly cleaning "pending" up when a reload was done and + no changes were made. Both of these contributed to a crash if + CONFBRIDGE(menu,template) was called in a dialplan after a reload. + + * aco_process_config now sets info->internal->pending to NULL + after it unrefs it although this isn't strictly necessary in the + context of this fix. + * menu_template_handler now uses the "current" config and silently + ignores any attempt to be called as a result of someone uses the + "template" parameter in the conf file. + + Luckily there's no other place in the codebase where + aco_pending_config is used outside of aco_process_config. + + ASTERISK-25506 #close + Reported-by: Frederic LE FOLL + + Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7 + +2017-04-30 16:40 +0000 [c90d81ef51] Joshua Colp + + * bridge: Fix returning to dialplan when executing Bridge() from AMI. + + When using the Bridge AMI action on the same channel multiple times + it was possible for the channel to return to the wrong location in + the dialplan if the other party hung up. This happened because the + priority of the channel was not preserved across each action + invocation and it would fail to move on to the next priority in + other cases. + + This change makes it so that the priority of a channel is preserved + when taking control of it from another thread and it is incremented + as appropriate such that the priority reflects where the channel + should next be executed in the dialplan, not where it may or may not + currently be. + + The Bridge AMI action was also changed to ensure that it too + starts the channels at the next location in the dialplan. + + ASTERISK-24529 + + Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a + +2017-04-25 11:49 +0000 [7b0e3b92fd] Kevin Harwell + + * bridge_simple: Added support for streams + + This patch is the first cut at adding stream support to the bridging framework. + Changes were made to the framework that allows mapping of stream topologies to + a bridge's supported media types. + + The first channel to enter a bridge initially defines the media types for a + bridge (i.e. a one to one mapping is created between the bridge and the first + channel). Subsequently added channels merge their media types into the bridge's + adding to it when necessary. This allows channels with different sized + topologies to map correctly to each other according to media type. The bridge + drops any frame that does not have a matching index into a given write stream. + + For now though, bridge_simple will align its two channels according to size or + first to join. Once both channels join the bridge the one with the most streams + will indicate to the other channel to update its streams to be the same as that + of the other. If both channels have the same number of streams then the first + channel to join is chosen as the stream base. + + A topology change source was also added to a channel when a stream toplogy + change request is made. This allows subsystems to know whether or not they + initiated a change request. Thus avoiding potential recursive situations. + + ASTERISK-26966 #close + + Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163 + +2017-05-01 13:04 +0000 [008e25def9] Kevin Harwell + + * res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures + + When a call gets put on hold RTP is temporarily stopped and Asterisk was + setting the remote RTCP address to NULL. Then when RTCP data was received + from the remote endpoint, Asterisk would be missing this information when + publishing the rtcp_message stasis event. Consequently, message subscribers + (in this case res_hep_rtcp) trying to parse the "from" field output the + following error: + + "ast_sockaddr_split_hostport: Port missing in (null)" + + This patch makes it so the remote RTCP address is no longer set to NULL when + stopping RTP. There was only one place that appeared to check if the remote + RTCP address was NULL as a way to tell if RTCP was running. This patch added + an additional check on the RTCP schedid for that case to make sure RTCP was + truly not running. + + ASTERISK-26860 #close + + Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b + +2017-05-02 11:34 +0000 [675e058e77] Sean Bright + + * cleanup: Change severity of fread short-read warning + + Many sound files don't have a full frame's worth of data at EOF, so the + warning messages were a bit too noisy. So we demote them to debug + messages. + + Change-Id: I6b617467d687658adca39170a81797a11cc766f6 + +2017-04-26 16:22 +0000 [cd272da7a8] Richard Mudgett + + * SDP: Replace SDP telephone_event option with dtmf option + + The telephone_event option was used as a flag and a bit mapped value in + different places when it is a boolean. It is also inadequate to configure + the DTMF operation of the RTP instance created for the stream. + + Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b + +2017-04-29 16:11 +0000 [52e4f02b1a] Richard Mudgett + + * res_pjsip_t38.c: Fix deadlock in T.38 framehook. + + A deadlock can happen between a channel lock and a pjsip session media + container lock. One thread is processing a reINVITE's SDP and walking + through the session's media container when it waits for the channel lock + to put the determined format capabilities onto the channel. The other + thread is writing a frame to the channel and processing the T.38 frame + hook. The T.38 frame hook then waits for the pjsip session's media + container lock. The two threads are now deadlocked. + + * Made the T.38 frame hook release the channel lock before searching the + session's media container. This fix has been done to several other + frame hooks to fix deadlocks. + + ASTERISK-26974 #close + + Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186 + +2017-04-28 10:56 +0000 [8170793be6] George Joseph + + * res_pjsip_outbound_authenticator_digest: Add context to log messages + + There was no context info in this module's log messages so it was + impossible to toubleshoot. + + Added endpoint or host to all messages and added the realms in the + challenge for the "No auth credentials for any realm" message. + + Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b + +2017-04-27 16:46 +0000 [48566b8c66] Richard Mudgett + + * res_sdp_translator_pjmedia.c: Add TODO notes. + + Change-Id: If27ca61f79accc882c3376d2e876d2b44aa1347b + +2017-04-24 18:13 +0000 [ede90e4aa5] Richard Mudgett + + * SDP: Make SDP translation to/from internal representation more const. + + Change-Id: I473a174b869728604b37c60853896b0c458bc504 + +2017-04-20 19:25 +0000 [5c1851cbc0] Richard Mudgett + + * stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap. + + Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08 + +2017-04-24 16:55 +0000 [d71c6e3bfd] Richard Mudgett + + * SDP: Make ast_sdp_state_set_remote_sdp() return error. + + Change-Id: I7707c9d872c476d897ff459008652b35142a35e1 + +2017-04-14 11:52 +0000 [176123e76c] Richard Mudgett + + * SDP: Misc cleanups (Mostly memory leaks) + + Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2 + +2017-04-27 18:15 +0000 [bad091b317] Richard Mudgett + + * chan_vpb.cc: Fix compile error. + + Change-Id: I6d9edd34d8b2474222c86f44e379ead61e57a54f + +2017-04-26 16:14 +0000 [d6535c0080] Mark Michelson + + * SDP API: Add SSRC-level attributes + + RFC 5576 defines how SSRC-level attributes may be added to SDP media + descriptions. In general, this is useful for grouping related SSRCes, + indicating SSRC-level format attributes, and resolving collisions in RTP + SSRC values. These attributes are used widely by browsers during WebRTC + communications, including attributes defined by documents outside of RFC + 5576. + + This commit introduces the addition of SSRC-level attributes into SDPs + generated by Asterisk. Since Asterisk does not tend to use multiple + SSRCs on a media stream, the initial support is minimal. Asterisk + includes an SSRC-level CNAME attribute if configured to do so. This at + least gives browsers (and possibly others) the ability to resolve SSRC + collisions at offer-answer time. + + In order to facilitate this, the RTP engine API has been enhanced to be + able to retrieve the SSRC and CNAME on a given RTP instance. + + res_rtp_asterisk currently does not provide meaningful CNAME values in + its RTCP SDES items, and therefore it currently will always return an + empty string as the CNAME value. A task in the near future will result + in res_rtp_asterisk generating more meaningful CNAMEs. + + Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789 + +2017-04-27 08:02 +0000 [d6b2a58736] George Joseph + + * res_pjsip_session: Add cleanup to ast_sip_session_terminate + + If you use ast_request to create a PJSIP channel but then hang it + up without causing a transaction to be sent, the session will + never be destroyed. This is due ot the fact that it's pjproject + that triggers the session cleanup when the transaction ends. + app_chanisavail was doing this to get more granular channel state + and it's also possible for this to happen via ARI. + + * ast_sip_session_terminate was modified to explicitly call the + cleanup tasks and unreference session if the invite state is NULL + AND invite_tsx is NULL (meaning we never sent a transaction). + + * chan_pjsip/hangup was modified to bump session before it calls + ast_sip_session_terminate to insure that session stays valid + while it does its own cleanup. + + * Added test events to session_destructor for a future testsuite + test. + + ASTERISK-26908 #close + Reported-by: Richard Mudgett + + Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9 + +2017-04-24 10:59 +0000 [2b22c3c84b] Joshua Colp + + * channel: Add ability to request an outgoing channel with stream topology. + + This change extends the ast_request functionality by adding another + function and callback to create an outgoing channel with a requested + stream topology. Fallback is provided by either converting the + requested stream topology into a format capabilities structure if + the channel driver does not support streams or by converting the + requested format capabilities into a stream topology if the channel + driver does support streams. + + The Dial application has also been updated to request an outgoing + channel with the stream topology of the calling channel. + + ASTERISK-26959 + + Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6 + +2017-04-26 14:20 +0000 [c6b757fa05] Kevin Harwell + + * res_pjsip/res_pjsip_callerid: NULL check on caller id name string + + It's possible for a name in a party id structure to be marked as valid, but the + name string itself be NULL (for instance this is possible to do by using the + dialplan CALLERID function). There were a couple of places where the name was + validated, but the string itself was not checked before passing it to functions + like 'strlen'. This of course caused a crashed. + + This patch adds in a NULL check before attempting to pass it into a function + that is not NULL tolerant. + + ASTERISK-25823 #close + + Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a + +2017-04-25 11:43 +0000 [cf3429b934] Kevin Harwell + + * vector: defaults and indexes + + Added an pre-defined integer vector declaration. This makes integer vectors + easier to declare and pass around. Also, added the ability to default a vector + up to a given size with a default value. Lastly, added functionality that + returns the "nth" index of a matching value. + + Also, updated a unit test to test these changes. + + Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5 + +2017-04-26 05:38 +0000 [985a5fd7aa] Joshua Colp + + * frame: Better handle interpolated frames. + + Interpolated frames are frames which contain a number of + samples but have no actual data. Audiohooks did not + handle this case when translating an incoming frame into + signed linear. It assumed that a frame would always contain + media when it may not. If this occurs audiohooks will now + immediately return and not act on the frame. + + As well for users of ast_trans_frameout the function has + been changed to be a bit more sane and ensure that the data + pointer on a frame is set to NULL if no data is actually + on the frame. This allows the various spots in Asterisk that + check for an interpolated frame based on the presence of a + data pointer to work as expected. + + ASTERISK-26926 + + Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b + +2017-04-26 09:22 +0000 [99dea9ba84] Yasin CANER + + * res_pjsip_session : fixed wrong From Header number On Re-invite + + ASTERISK-26964 #close + + Change-Id: I55a9caa7dc90e6c4c219cb09b5c2ec08af84a302 + +2017-04-26 08:45 +0000 [858ed60446] George Joseph + + * pjproject_bundled: Add --disable-libwebrtc to configure + + Without the disable, pjproject tries to build it's internal + webrtc implementation which requires sse2. This fails on + platforms without sse2. + + ASTERISK-26930 #close + Reported-by: abelbeck + + Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410 + +2017-04-26 07:58 +0000 [585f9405b1] Thierry Magnien + + * channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections + + For outgoing TCP connections, Asterisk uses the first IP address of the + interface instead of the IP address we asked him to bind to. + + ASTERISK-26922 #close + Reported-by: Ksenia + + Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb + +2017-04-21 12:04 +0000 [f5b67871df] Sean Bright + + * cleanup: Fix fread() and fwrite() error handling + + Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in + the format modules. Neither of these functions will ever return a value + less than 0, which we were checking for in some cases. + + I've introduced a fair amount of duplication in the format modules, but + I plan to change how format modules work internally in a subsequent + patch set, so this is simply a stop-gap. + + Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872 + +2017-04-25 07:52 +0000 [199d4776c0] Joshua Colp + + * alembic: Add table for 'resource_list' PJSIP RLS type. + + This change adds an Alembic migration which adds a + ps_resource_list table that can contain resource_list + RLS configuration objects. + + ASTERISK-26929 + + Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05 + +2017-04-14 05:21 +0000 [19a79ae12c] Joshua Colp + + * sdp: Add support for T.38 + + This change adds a T.38 format which can be used in a stream + topology to specify that a UDPTL stream needs to be created. + The SDP API has been changed to understand T.38 and create + the UDPTL session, add the attributes, and parse the attributes. + + This change does not change the boundary of the T.38 state + machine. It is still up to the channel driver to implement and + act on it (such as queueing control frames or reacting to them). + + ASTERISK-26949 + + Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7 + +2017-03-21 15:44 +0000 [32b3e36c68] Mark Michelson + + * SDP: Ensure SDPs "merge" properly. + + The gist of this work ensures that when a remote SDP is received, it is + merged properly with the local capabilities. The remote SDP is converted + into a stream topology. That topology is then merged with the current + local topology on the SDP state. That new merged topology is then used + to create an SDP. Finally, adjustments are made to RTP instances based + on knowledge gained from the remote SDP. + + There are also a battery of tests in this commit that ensure that some + basic SDP merges work as expected. + + While this may not sound like a big change, it has the property that it + caused lots of ancillary changes. + + * The remote SDP is no longer stored on the SDP state. Biggest reason: + there's no need for it. The remote SDP is used at the time it is being + set and nowhere else. + + * Some new SDP APIs were added in order to find attributes and convert + generic SDP attributes into rtpmap structures. + + * Writing tests made me realize that retrieving a value from an SDP + options structure, the SDP options needs to be made const. + + * The SDP state machine was essentially gutted by a previous commit. + Initially, I attempted to reinstate it, but I found that as it had + been defined, it was not all that useful. What was more useful was + knowing the role we play in SDP negotiation, so the SDP state machine + has been transformed into an indicator of role. + + * Rather than storing separate local and joint stream state + capabilities, it makes more sense to keep track of current stream + state and update it as things change. + + Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d + +2017-04-24 13:16 +0000 [0611f2ca17] Sean Bright + + * res_hep: Add additional config initialization and validation + + * Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor + does not close fd 0 + + * Add logging output when the required option - capture_address - is not + specified. + + * Remove a no longer relevant #define and correct related documentation + + * Pass appropriate flags to aco_option_register so that capture_address + cannot be the empty string. + + ASTERISK-26953 #close + + Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076 + +2017-04-17 19:06 +0000 [59203c51cc] Sean Bright + + * core: Use eventfd for alert pipes on Linux when possible + + The primary win of switching to eventfd when possible is that it only + uses a single file descriptor while pipe() will use two. This means for + each bridge channel we're reducing the number of required file + descriptors by 1, and - if you're using timerfd - we also now have 1 + less file descriptor per Asterisk channel. + + The API is not ideal (passing int arrays), but this is the cleanest + approach I could come up with to maintain API/ABI. + + I've also removed what I believe to be an erroneous code block that + checked the non-blocking flag on the pipe ends for each read. If the + file descriptor is 'losing' its non-blocking mode, it is because of a + bug somewhere else in our code. + + In my testing I haven't seen any measurable difference in performance. + + Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d + +2017-04-21 12:33 +0000 [f1d20c84a1] Richard Mudgett + + * res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions. + + If ICE is enabled and a STUN server does not respond then we will block + until we give up on the STUN response. This will take nine seconds. In + the mean time the peer that sent the INVITE will send retransmissions. + + * Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out + earlier to prevent these retransmissions. + + ASTERISK-26890 + + Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8 + +2017-04-21 12:07 +0000 [835c209445] Richard Mudgett + + * res_pjsip_session.c: Restructure ast_sip_session_alloc() + + * Restructure ast_sip_session_alloc() to need less cleanup on off nominal + error paths. + + * Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid + unnecessary ref manipulation to return a session. This is faster than + calling a function. That function may do logging of the ref changes with + REF_DEBUG enabled. + + Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a + +2017-04-20 02:13 +0000 [b4b1943c5d] Jean Aunis + + * chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK + + Some equipments may send a re-INVITE containing an SDP in the final ACK + request. If this happens in the context of direct media, the remote end + should be updated with a re-INVITE. + This patch queues an "update RTP peer" frame to trigger the re-INVITE, + instead of the "source change" frame wich was used previously. + + ASTERISK-26951 + + Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6 + +2017-04-19 15:08 +0000 [c47b3e74d2] Sean Bright + + * pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified + + Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core + to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE + is passed to these functions, the calling thread will be blocked until + the newly created channel has been hung up. + + After this patch, we run the dial on the current thread rather than + spawning a new one. The only in-tree code that passes + AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced + thread usage if you are using .call files. + + Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913 + +2017-04-19 13:23 +0000 [afad2ffd9f] Richard Mudgett + + * res_rtp_asterisk.c: Fix crash in RTCP DTLS operation. + + Occasionally a crash happens when processing the RTCP DTLS timeout + handler. The RTCP DTLS timeout timer could be left running if we have not + completed the DTLS handshake before we place the call on hold or we + attempt direct media. + + * Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP. + + * Made some sanity tweaks to ast_rtp_prop_set() when switching from + standard RTCP mode to RTCP multiplexed mode. + + ASTERISK-26692 #close + + Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0 + +2017-03-22 16:05 +0000 [d165079cbc] Richard Mudgett + + * rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. + + The struct ast_rtp_instance has historically been indirectly protected + from reentrancy issues by the channel lock because early channel drivers + held the lock for really long times. Holding the channel lock for such a + long time has caused many deadlock problems in the past. Along comes + chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock + because sometimes there may not be an associated channel created yet or + the channel pointer isn't available. + + In the case of ASTERISK-26835 a pjsip serializer thread was processing a + message's SDP body while another thread was reading a RTP packet from the + socket. Both threads wound up changing the rtp->rtcp->local_addr_str + string and interfering with each other. The classic reentrancy problem + resulted in a crash. + + In the case of ASTERISK-26853 a pjsip serializer thread was processing a + message's SDP body while another thread was reading a RTP packet from the + socket. Both threads wound up processing ICE candidates in PJPROJECT and + interfering with each other. The classic reentrancy problem resulted in a + crash. + + * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP + instance struct. + + * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP + instance struct for the API call. + + * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy + problem with rtp->rtcp->local_addr_str in the scheduler thread running + ast_rtcp_write(). + + * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in + bridge_p2p_rtp_write() because there are two RTP instance structs + involved. + + * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler + callbacks. We cannot hold the instance lock when trying to stop a + scheduler callback. + + * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the + struct ast_rtp_instance ao2 object lock instead. The lock was used to + synchronize two threads to prevent a race condition between starting and + stopping a timeout timer. The race condition is no longer present between + dtls_perform_handshake() and __rtp_recvfrom() because the instance lock + prevents these functions from overlapping each other with regards to the + timeout timer. + + * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct + ast_rtp_instance ao2 object lock instead. The lock was used to + synchronize two threads using a condition signal to know when TURN + negotiations complete. + + * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN + ioqueue_worker_thread(). We cannot hold the instance lock when trying to + create or shut down the worker thread without a risk of deadlock. + + This patch exposed a race condition between a PJSIP serializer thread + setting up an ICE session in ice_create() and another thread reading RTP + packets. + + * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we + have re-locked the RTP instance to prevent the other thread from trying to + process ICE packets on an incomplete ICE session setup. + + A similar race condition is between a PJSIP serializer thread resetting up + an ICE session in ice_create() and the timer_worker_thread() processing + the completion of the previous ICE session. + + * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an + uninitialized/null remote_address after calling + update_address_with_ice_candidate(). + + * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() + destroying and setting the rtp->ice pointer to NULL while other threads + are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. + Now when we have to unlock the RTP instance object to call a PJPROJECT ICE + function we will hold a ref to the wrapper. Also added some rtp->ice NULL + checks after we relock the RTP instance and have to do something with the + ICE structure. + + ASTERISK-26835 #close + ASTERISK-26853 #close + + Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4 + +2017-04-19 08:39 +0000 [b8b3380944] Sean Bright + + * build: Update config.guess and config.sub + + Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172 + +2017-04-14 13:52 +0000 [6c0ab9afa7] Sean Bright + + * format_wav: Read 16khz wav samples properly + + When opening a PCM wave file for reading, we aren't tracking the + frequency of the opened file, so we treat 16khz files as 8khz and do + half reads. + + This patch also cleans up some of the data types and an unnecessarily + complex `if` expression. + + ASTERISK-26613 #close + Reported by: Vitaly K + + Change-Id: I05f8b263058dc573ea8ffe0c62e7964506e11815 + +2017-04-16 19:59 +0000 [b55d21ad91] George Joseph + + * make ari-stubs so doc periodic jobs can run + + The periodic doc job does a make ari-stubs and checks that + there are no changes before generating the docs. Since I changed + the mustache template (and the generated code directly) recently + and forgot to regenerate the stubs, the doc job thinks they're out + of date. + + Change-Id: I94b97035311eccf52b0101b8590223265a7881d4 + +2017-04-14 12:51 +0000 [4fb9f5d60e] Sean Bright + + * format_ogg_vorbis: Clear ogg/vorbis data structures on close + + On filestream close, we need to clear out the ogg & vorbis data + structures to prevent a memory leak. + + ASTERISK-26169 #close + Reported by: Ivan Myalkin + + Change-Id: Iee94c5a5d5bdafbf8b181c5c064d15d90ace8274 + +2017-04-14 17:32 +0000 [a3e623dd70] Richard Mudgett + + * Revert "bridging: Ensure successful T.38 negotation" + + This reverts commit 7819f95791fe0ca0e0cdc417e2687a5900444053. + + Change-Id: Ib91a7e6c9856f5f41329e42f40ba2394fee861a4 + +2017-04-14 16:50 +0000 [f6600f2c2e] Sean Bright + + * res_stun_monitor: Don't fail to load if DNS resolution fails + + res_stun_monitor will fail to load if DNS resolution of the STUN server + fails. Instead, we continue without the STUN server being resolved and + we will re-attempt the resolution on the STUN refresh interval. + + ASTERISK-21856 #close + Reported by: Jeremy Kister + + Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254 + +2017-04-14 14:36 +0000 [be71be7ed2] Sean Bright + + * format_pcm: Track actual header size of .au files + + Sun's Au file format has a minimum data offset 24 bytes, but this + offset is encoded in each .au file. Instead of assuming the minimum, + read the actual value and store it for later use. + + ASTERISK-20984 #close + Reported by: Roman S. + Patches: + asterisk-1.8.20.0-au-clicks-2.diff (license #6474) patch + uploaded by Roman S. + + Change-Id: I524022fb19ff2fd5af2cc2d669d27a780ab2057c + +2017-04-12 07:50 +0000 [2e6075c51f] George Joseph + + * modules: change module LOAD_FAILUREs to LOAD_DECLINES (master) + + Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258 + +2017-04-10 05:13 +0000 [72c5f3b0ba] Alexander Traud + + * res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP. + + This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in + SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over + UDP, if many codecs are allowed in Asterisk. This new feature is enabled + together with the optional feature compact_headers=yes via the file pjsip.conf. + + ASTERISK-26932 #close + + Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689 + +2017-04-12 07:47 +0000 [6db0939b96] George Joseph + + * modules: change module LOAD_FAILUREs to LOAD_DECLINES (14) + + Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e + +2017-04-11 11:07 +0000 [747beb1ed1] George Joseph + + * modules: change module LOAD_FAILUREs to LOAD_DECLINES + + In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed + to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting + if a module can't be loaded. If the user wishes to retain the + FAILURE behavior for a specific module, they can use the "require" + or "preload-require" keyword in modules.conf. + + A new API was added to logger: ast_is_logger_initialized(). This + allows asterisk.c/check_init() to print to the error log once the + logger subsystem is ready instead of just to stdout. If something + does fail before the logger is initialized, we now print to stderr + instead of stdout. + + Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25 + +2017-04-05 06:41 +0000 [7819f95791] Torrey Searle + + * bridging: Ensure successful T.38 negotation + + When a T.38 happens immediatly after call establishment, the control + frame can be lost because the other leg is not yet in the bridge. + + This patch detects this case an makes sure T.38 negotation happens + when the 2nd leg is being made compatible with the negotating + first leg + + ASTERISK-26923 #close + + Change-Id: If334125ee61ed63550d242fc9efe7987e37e1d94 + +2017-04-07 08:58 +0000 [7901225261] Torrey Searle + + * strings.h: Avoid overflows in the string hash functions + + On 2's compliment machines abs(INT_MIN) behavior is undefined and + results in a negative value still being returnd. This results in + negative hash codes that can result in crashes. + + ASTERISK-26528 #close + + Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b + +2017-04-07 16:14 +0000 [7312cbe803] Richard Mudgett + + * res_rtp_asterisk.c: Add stun_blacklist option + + Added the stun_blacklist option to rtp.conf. Some multihomed servers have + IP interfaces that cannot reach the STUN server specified by stunaddr. + Blacklist those interface subnets from trying to send a STUN packet to + find the external IP address. Attempting to send the STUN packet + needlessly delays processing incoming and outgoing SIP INVITEs because we + will wait for a response that can never come until we give up on the + response. Multiple subnets may be listed. + + ASTERISK-26890 #close + + Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342 + +2017-04-06 17:31 +0000 [7c37365f03] Richard Mudgett + + * stun.c: Fix ast_stun_request() erratic timeout. + + If ast_stun_request() receives packets other than a STUN response then we + could conceivably never exit if we continue to receive packets with less + than three seconds between them. + + * Fix poll timeout to keep track of the time when we sent the STUN + request. We will now send a STUN request every three seconds regardless + of how many other packets we receive while waiting for a response until we + have completed three STUN request transmission cycles. + + Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266 + +2017-04-06 18:30 +0000 [8d323c74fa] Richard Mudgett + + * sorcery.c: Speed up ast_sorcery_retrieve_by_id() + + Return early if ast_sorcery_retrieve_by_id() is not passed an id to find. + Also eliminated the RAII_VAR() usage in the function. + + Change-Id: I871dbe162a301b5ced8b4393cec27180c7c6b218 + +2017-04-10 11:30 +0000 [5b4e2ec267] Richard Mudgett + + * res_pjsip: Fix pointer use after unref. + + Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1 + +2017-04-06 18:18 +0000 [6f793ac149] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member. + + * create_rtp(): Eliminate use of deprecated transport struct member. That + member and several others in the transport structure were deprecated + because of an infinite loop created when using realtime configuration. + See 2451d4e4550336197ee2e482750cc53f30afa352 + + ASTERISK-26851 + + Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc + +2017-04-10 17:45 +0000 [d76bc0565c] Richard Mudgett + + * tcptls.c: Cleanup TCP/TLS listener thread on abnormal exit. + + Temporarily running out of file descriptors should not terminate the + listener thread. Otherwise, when there becomes more file descriptors + available, nothing is listening. + + * Added EMFILE exception to abnormal thread exit. + + * Added an abnormal TCP/TLS listener exit error message. + + * Closed the TCP/TLS listener socket on abnormal exit so Asterisk does not + appear dead if something tries to connect to the socket. + + ASTERISK-26903 #close + + Change-Id: I10f2f784065136277f271159f0925927194581b5 + +2017-04-08 03:05 +0000 [2b8dbc9e00] Walter Doekes + + * samples: Undo removal of include from canonicalize-app-names commit. + + This include was accidentally removed in changeset + Ia79aea64de89531362e993e34230c2044a70aa93. My bad. + + Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082 + +2017-04-07 08:35 +0000 [270b485f04] Joshua Colp + + * pjsip: Add Alembic for PUBLISH support. + + This change adds database tables for the PUBLISH support so it + can be configured using realtime. A minor fix to the + res_pjsip_publish_asterisk module was done so that it read the + sorcery configuration from the correct section. Finally the + sample configuration files have been updated. + + ASTERISK-26928 + + Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952 + +2017-04-07 08:06 +0000 [7a46cd7433] Alexander Traud + + * pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). + + When the Asterisk channel driver res_pjsip offers SIP-over-TLS, sometimes, not + reproducible, Asterisk crashed in pj_ssl_sock_get_info() because a NULL pointer + was read. This change avoids this crash. + + ASTERISK-26927 #close + + Change-Id: I24a6011b44d1426d159742ff4421cf806a52938b + +2017-04-05 09:10 +0000 [e6ae3651b8] Walter Doekes + + * samples: Canonicalize app names in extensions.conf.sample. + + This takes care of warnings by ossobv/asterisklint. + + Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93 + +2017-04-04 16:20 +0000 [01e9eaf3a6] George Joseph + + * pjproject_bundled: Add 3 upstream patches + + 0035-r5572-svn-backport-dialog-transaction-deadlock.patch + 0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch + 0037-r5576-svn-backport-session-timer-crash.patch + + Also removed the progress bar from wget download to stdout. + + ASTERISK-26905 #close + Reported-by: Ross Beer + + Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256 + +2017-04-04 11:44 +0000 [fac5115c43] Troy Bowman + + * app_queue: Log reason for PAUSEALL/UNPAUSEALL + + We needed the reason for our reporting when agents pause/unpause all of + their queues at once. This is a small, simple patch that adds a reason + for PAUSEALL and UNPAUSEALL. I have been using it in production for years. + + ASTERISK-26920 #close + + Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d + +2017-04-05 14:50 +0000 [40e9d5e8b7] George Joseph + + * sample_config: Add samples for pubsub to pjsip.conf.sample + + Added: + * outbound-publish + * resource_list + * inbound-publication + * asterisk-publication + + Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a + +2017-04-03 15:38 +0000 [f2ee8ac21e] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Don't alter global addr variable. + + * create_rtp(): Fix unexpected alteration of global address_rtp if a + transport is bound to an address. + + * create_rtp(): Fix use of uninitialized memory if the endpoint RTP media + address is invalid or the transport has an invalid address. + + ASTERISK-26851 + + Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7 + +2017-03-27 09:03 +0000 [380973cc47] Corey Farrell + + * CDR: Protect from data overflow in ast_cdr_setuserfield. + + ast_cdr_setuserfield wrote to a fixed length field using strcpy. This could + result in a buffer overrun when called from chan_sip or func_cdr. This patch + adds a maximum bytes written to the field by using ast_copy_string instead. + + ASTERISK-26897 #close + patches: + 0001-CDR-Protect-from-data-overflow-in-ast_cdr_setuserfie.patch submitted + by Corey Farrell (license #5909) + + Change-Id: Ib23ca77e9b9e2803a450e1206af45df2d2fdf65c + +2017-03-25 19:01 +0000 [6c3ae397cb] Daniel Journo + + * Unused realtime MOH classes not purged on 'moh reload' + + Purge Realtime MOH classes on 'moh reload' even when musiconhold.conf + hasn't changed. + + ASTERISK-25974 #close + + Change-Id: I42c78ea76528473a656f204595956c9eedcf3246 + +2017-03-31 12:09 +0000 [8e36064109] Corey Farrell + + * core: Improve/simplify handling of required headers. + + * Report failures if configure finds a required header is missing. + * Deduplicate includes between asterisk.h, astmm.h and compat.h. + * Unconditionally include headers in compat.h if required elsewhere. + + Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5 + +2017-04-03 13:56 +0000 [a889621b14] Richard Mudgett + + * res_pjsip: Fix transport ref leak. + + We were leaking a transport ref in multihomed_on_rx_message() which + resulted in the FRACK about excessive ref counts. + + ASTERISK-26916 #close + + Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f + +2017-04-03 02:30 +0000 [4fc22c7673] Alexander Traud + + * chan_sip: Session Timers required but refused wrongly. + + SIP user-agents indicate which protocol extensions are allowed in headers + like Supported and Required. Such protocol extensions are Session Timers + (RFC 4028) for example. Session Timers are supported since Mantis-10665. + Since ASTERISK-21721, not only the first but multiple Supported/Required + headers in a message are parsed. In that change, an existing variable was + re-used within a newly added do-loop. Currently, at the end of that loop, + that variable is an empty string always. Previously, that variable was used + within log output. However, the log output was not changed. + + ASTERISK-26915 #close + + Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990 + +2017-03-31 16:31 +0000 [48be02c5d8] Joshua Colp + + * res_pjsip_session: Allow BYE to be sent on disconnected session. + + It is perfectly acceptable for a BYE to be sent on a disconnected + session. This occurs when we respond to a challenge to the BYE + for authentication credentials. + + ASTERISK-26363 + + Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045 + +2017-03-31 13:14 +0000 [e8b1bb3041] Richard Mudgett + + * chan_vpb.cc: Fix compiler error. + + Added missing channel technology read/write stream callback + initialization. + + Change-Id: I829043a327d987e0d964485dd3d27964bebbd623 + +2017-03-30 18:28 +0000 [f9695dc057] Corey Farrell + + * Forward declare 'struct ast_json' in asterisk.h + + The ast_json structure is used in many Asterisk headers and is often the + only part of json.h used. This adds a forward declaration to asterisk.h + and removes the include of json.h from many headers. The declaration + has been left in endpoints.h and stasis.h to avoid problems with source + files that use ast_json functions without directly including json.h. + + ari.h continues to include json.h as it uses enum + ast_json_encoding_format. + + Change-Id: Id766aabce6bed56626d27e8d29f559b5e687b769 + +2017-03-30 08:11 +0000 [c537f99488] Sean Bright + + * cdr_pgsql: Fix buffer overflow calling libpq + + Implement the same buffer size checking done in cel_pgsql. + + ASTERISK-26896 #close + Reported by: twisted + + Change-Id: Iaacfa1f1de7cb1e9414d121850d2d8c2888f3f48 + +2017-03-28 13:01 +0000 [a7d94f504f] Walter Doekes + + * build: Fix deb build issues with fakeroot + + If DESTDIR is set, don't call ldconfig. Assume that DESTDIR is used to + create a binary archive. The ldconfig call should be delegated to the + archive postinst script. This fixes the case where fakeroot wraps 'make + install' causing $EUID to be 0 even though it doesn't have permission to + call ldconfig. + + The previous logic in configure.ac to detect and correct libdir + has been removed as it was not completely accurate. CentOS 64-bit + users should again specifiy --libdir=/usr/lib64 when configuring + to prevent install to /usr/lib. + + Updated Makefile:check-old-libdir to check for orphans in + lib64 when installing to lib as well as orphans in lib when installing + to lib64. + + Updated Makefile and main/Makefile uninstall targets to remove the + orphans using the new logic. + + ASTERISK-26705 + + Change-Id: I51739d4a03e60bff38be719b8d2ead0007afdd51 + +2017-03-27 15:32 +0000 [f3290d6b66] Joshua Colp + + * sdp: Add support for setting connection address and clean up state. + + This change cleans up state management for media streams by moving + RTP instances into their own session structure and adding additional + details that are not relevant to the core (such as connection address). + These can live either in the local capabilities or joint capabilities. + + The ability to set explicit connection address information for + the purposes of direct media and NAT has also been added at the + global and stream specific level. + + ASTERISK-26900 + + Change-Id: If7e5307239a9534420732de11c451a2705b6b681 + +2017-03-29 10:11 +0000 [5c1ea3ebbd] Sean Bright + + * astobj2: Prevent potential deadlocks with ao2_global_obj_release + + The ao2_global_obj_release() function holds an exclusive lock on the + global object while it is being dereferenced. Any destructors that + run during this time that call ao2_global_obj_ref() will deadlock + because a read lock is required. + + Instead, we make the global object inaccessible inside of the write + lock and only dereference it once we have released the lock. This + allows the affected destructors to fail gracefully. + + While this doesn't completely solve the referenced issue (the error + message about not being able to create an IQ continues to be shown) + it does solve the backtrace spew that accompanied it. + + ASTERISK-21009 #close + Reported by: Marcello Ceschia + + Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385 + +2017-03-30 10:18 +0000 [4e5cc70fb4] Corey Farrell + + * CEL: Remove header declarations of non-existant functions. + + ast_cel_alloc and ast_cel_destroy do not exist in code, remove them from + the headers. + + Change-Id: I99ce848e2e109e7d61771559f559b9e57973e45c + +2017-03-27 11:49 +0000 [f66edcb8b0] Josh Roberson + + * cel_pgsql.c: Fix buffer overflow calling libpq + + PQEscapeStringConn() expects the buffer passed in to be an + adequitely sized buffer to write out the escaped SQL value string + into. It is possible, for large values (such as large values to + Dial with a lot of devices) to have more than our 512+1 byte + allocation and thus cause libpq to create a buffer overrun. + + glibc will nicely ABRT asterisk for you, citing a stack smash. + + Let's only allocate it to be as large as needed: + If we have a value, then (strlen(value) * 2) + 1 (as recommended + by libpq), and if we have none, just one byte to hold our null + will do. + + ASTERISK-26896 #close + + Change-Id: If611c734292618ed68dde17816d09dd16667dea2 + +2017-03-29 08:04 +0000 [e76cc51d5e] Alexander Traud + + * srtp: Allow zero as tag value for a sRTP Crypto Suite. + + ASTERISK-25490 #close + + Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f + +2017-03-28 13:10 +0000 [2fe52174de] George Joseph + + * res_pjsip_config_wizard: Add 2 new parameters to help with proxy config + + Two new parameters have been added to the pjsip config wizard. + + * Setting 'sends_line_with_registrations' to true will cause the wizard + to skip the creation of an identify object to match incoming request + to the endpoint and instead add the line and endpoint parameters to + the outbound registration object. + + * Setting 'outbound_proxy' is a shortcut for adding individual + endpoint/outbound_proxy, aor/outbound_proxy and + registration/outbound_proxy parameters. + + Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 + (cherry picked from commit a827892ff77cd37912b528d9c45b446be091bbc0) + (cherry picked from commit 27344675be1941d30508c6e6bd684acdd0791e1a) + +2017-03-28 09:29 +0000 [7c0b12dc41] Sean Bright + + * alembic: Turn off execute bit on non-executable python scripts + + Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c + +2017-03-27 12:37 +0000 [3d8899bacf] Richard Mudgett + + * Add DTLS sanity check. + + Change-Id: Ib32612cf6c7ce9213a11b9cba82f630f8cd3564b + +2017-03-08 07:24 +0000 [5d938045d4] Joshua Colp + + * channel: Remove old epoll support and fixed max number of file descriptors. + + This change removes the old epoll support which has not been used or + maintained in quite some time. + + The fixed number of file descriptors on a channel has also been removed. + File descriptors are now contained in a growable vector. This can be + used like before by specifying a specific position to store a file + descriptor at or using a new API call, ast_channel_fd_add, which adds + a file descriptor to the channel and returns its position. + + Tests have been added which cover the growing behavior of the vector + and the new API call. + + ASTERISK-26885 + + Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928 + +2017-03-27 09:35 +0000 [fd204d5c65] Sean Bright + + * res_musiconhold: Document the 'format' option + + ASTERISK-26086 #close + Reported by: Jens Bürger + + Change-Id: I6aab666c0bf01fd0c64d7a5bcb22fa7f5d41335e + +2017-03-24 07:43 +0000 [cf6a6226ab] Sean Bright + + * core: Remove embedded module support + + This has not worked for some time and is no longer actively maintained. + + Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99 + +2017-03-27 08:58 +0000 [d22c678999] Sean Bright + + * res_musiconhold: Don't chdir() when scanning MoH files + + There doesn't appear to be any reason that we are chdir'ing in + moh_scan_files, and in the event of an Asterisk crash, the core files + may not get written because we have changed into a read-only directory. + + ASTERISK-23996 #close + Reported by: Walter Doekes + + Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354 + +2017-03-23 09:48 +0000 [d5a8799c4b] Sean Bright + + * res_xmpp: Use incremental backoff when a read error occurs + + If a read error occurs, we immediately attempt a reconnect without any + delay. Instead, let's sleep and backoff up to 60 seconds before we try + again. + + ASTERISK-24712 #close + Reported by: Matthias Urlichs + + Change-Id: I6fe10ef4734837727437beab715e336777f13f48 + +2017-03-24 11:29 +0000 [d08c69a9e2] Sean Bright + + * res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts + + chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL + (44) when a channel is hung up due to an RTP timeout. So do the same + when it happens with PJSIP for parity. + + Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8 + +2017-03-23 14:01 +0000 [d2f2cdf476] Kevin Harwell + + * AMI: Updated version + + Updated the AMI version for the following reason (see CHANGES for more details): + + The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now + contains a new optional parameter, 'MatchHeader'. + + Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9 + +2017-03-23 12:07 +0000 [12dde3b568] Kevin Harwell + + * pjproject_bundled: raise timeout value used when downloading + + After configuring Asterisk with '--with-pjproject-bundled' the configure/build + process attempts to download pjproject from its download site. Currently, a + timeout of 10 seconds is used that will stop the download process if pjproject + has not been fully downloaded in that time. For some systems this was not enough + time and the process was timing out too early. + + This patch raises the download timeout value to '60'. Also, this patch fixes + another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported + due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to + DOWNLOAD_TIMEOUT. + + ASTERISK-26814 #close + + Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842 + +2017-03-22 20:33 +0000 [98a88e9ffa] Sean Bright + + * res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus + + The documentation for JABBER_STATUS (and the deprecated JabberStatus + app) indicate that a return value of 7 indicates that the specified + buddy was not in the roster. It also indicates that you can specify a + "bare" JID (one without a resource). Unfortunately the actual behavior + does not match the documented behavior. + + Assuming that our roster includes the buddy online and available + "valid@example.org/Valid" and does *not* include the buddy + "invalid@example.org", the JABBER_STATUS() function returns the + following before this patch: + + +------------------------------+------------+--------------------------+ + | Buddy | Status | Result | + +------------------------------+------------+--------------------------+ + | valid@example.org | Online | 7 (Not in roster) | + | valid@example.org/Valid | Online | 1 (Online) | + | valid@example.org/Invalid | N/A | 7 (Not in roster) | + | invalid@example.org | N/A | Error logged, no return | + | invalid@example.org/Valid | N/A | Error logged, no return | + +------------------------------+------------+--------------------------+ + + And after this patch: + + +------------------------------+------------+--------------------------+ + | Buddy | Status | Result | + +------------------------------+------------+--------------------------+ + | valid@example.org | Online | 1 (Online) | + | valid@example.org/Valid | Online | 1 (Online) | + | valid@example.org/Invalid | N/A | 6 (Offline) | + | invalid@example.org | N/A | 7 (Not in roster) | + | invalid@example.org/Valid | N/A | 7 (Not in roster) | + +------------------------------+------------+--------------------------+ + + This brings the behavior in line with the documentation. + + ASTERISK-23510 #close + Reported by: Anthony Critelli + + Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf + +2017-03-23 09:45 +0000 [be94105d6d] Sean Bright + + * res_xmpp: Try to provide useful errors messages from OpenSSL + + If any errors occur during the TLS connection setup, we currently dump a + fairly generic error message. So instead we try to pull in something + useful from OpenSSL to report instead. + + ASTERISK-24712 + Reported by: Matthias Urlichs + + Change-Id: I288500991a9681f447d92913b11fedaf426087f4 + +2017-03-23 05:19 +0000 [ee81ee1f14] Sean Bright + + * res_xmpp: Fix ref counting issue + + The only remaining reference to the endpoint is in the endpoints + container, and because it is unlinked in ast_endpoint_shutdown, we don't + have to explicitly cleanup the endpoint ourselves. + + Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8 + +2017-03-23 09:30 +0000 [9493981419] Sean Bright + + * res_xmpp: Correctly check return value of SSL_connect + + SSL_connect returns non-zero for both success and some error conditions + so simply negating is inadequate. + + Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1 + +2017-03-22 17:32 +0000 [7657c279b5] Sean Bright + + * res_xmpp: Don't crash when trying to send a message without a connection + + If we never establish a connection to our Jabber server, iksemel never sets up + its internal transport pointer, so attempting to send a message dereferences a + NULL pointer and causes a crash. + + ASTERISK-21855 #close + Reported by: Jeremy Kister + + Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c + +2017-03-22 15:40 +0000 [0136ec12a3] Sean Bright + + * res_xmpp: Include client name in connection related error messages + + ASTERISK-25622 #close + Reported by: Sean Darcy + + Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9 + +2017-03-20 13:27 +0000 [9b103e7bea] Kevin Harwell + + * rtp_engine: allocate RTP dynamic payloads per session + + Dynamic payload types were statically defined in Asterisk. This unfortunately + limited the number of dynamic payloads that could be registered. With this patch + dynamic payload type numbers are now assigned dynamically and per RTP instance. + However, in order to limit any issues where some clients expect the old + statically defined value this patch makes it so the value Asterisk used to pre- + designate is used for the dynamic assignment if available. + + An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) + that turns the new dynamic behavior on or off. When off it reverts back to using + statically defined payload values. This option defaults to "yes" in Asterisk 15. + + ASTERISK-26515 #close + patches: + ASTERISK-26515.diff submitted by jcolp (license 5000 + + Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc + +2017-03-21 12:32 +0000 [bb2936f3e4] Sebastian Gutierrez + + * cdr: Allow setting of user field from 'h' extension + + The CDR code previously did not allow the user field to be set + from the 'h' extension in the dialplan. This change removes that + limitation and allows it to be set. + + ASTERISK-26818 + + Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6 + +2017-03-14 16:45 +0000 [6b7697ed48] Richard Begg + + * res_pjsip_session: Enable RFC3578 overlap dialing support. + + Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched + destinations) as currently provided by chan_sip is missing from res_pjsip. + This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] + which when set to yes enables 484 responses to partial destination + matches rather than the current 404. + + ASTERISK-26864 + + Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6 + +2017-03-21 06:59 +0000 [d4fcf196a2] Sean Bright + + * res_hep: Capture actual transport type in use + + Rather than hard-coding UDP, allow consumers of the HEP API to specify + which protocol is in use. Update the PJSIP provider to pass in the + current protocol type. + + ASTERISK-26850 #close + + Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978 + +2017-03-21 09:57 +0000 [1bf839d44b] Sean Bright + + * Revert "app_queue: Handle the caller being redirected out of a queue bridge" + + This reverts commit 163e9e53dc7d84dd42721e733b7706c8147bdd27. + + Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b + +2017-03-21 08:26 +0000 [6b4b87787c] Sean Bright + + * res_pjsip_messaging: Check URI type before dereferencing + + We aren't validating that the URI we just parsed is a SIP/SIPS one before + trying to access the user, host, and port members of a possibly uninitialized + structure. + + Also update the MessageSend documentation to indicate what 'from' formats are + accepted. + + ASTERISK-26484 #close + Reported by: Vinod Dharashive + + Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30 + +2017-03-13 15:21 +0000 [65ad554c98] Joshua Elson + + * pjsip: prevent memory corruption on creation of xml bodies + + ASTERISK-26776 #close + + Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2 + +2017-03-20 16:27 +0000 [fc794de756] Sean Bright + + * bridge_softmix: Ignore non-voice frames from translator + + Some codecs - codec_speex specifically - take voice frames and return + other types of frames, like CNG. If we subsequently treat those as + voice frames, we'll run into trouble when destroying the frame because + of the requirement that each voice frame have an associated format. + + ASTERISK-26880 #close + Reported by: Kirsty Tyerman + + Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c + +2017-03-14 23:49 +0000 [25016a74f8] Aaron An + + * audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. + + Fixed a bug in function "ast_audiohook_write_frame" that checked the + variable other_factory_samples and only flushed the factories, so they + would be in sync, when other_factory_samples > 0. When there is not any + rtp incoming the variable other_factory_samples will be 0, and although + the result of "our_factory_ms - other_factory_ms" may be very large, + this led to the record file not syncing. + + ASTERISK-26875 #close + Reported-by: Aaron An + Tested-by: Aaron An + + Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22 + +2017-03-18 12:30 +0000 [fc71c18a9b] Sean Bright + + * thread safety: Don't use getprotobyname() + + POSIX does not require getprotobyname() to be thread safe and some + implementations use static memory which causes issues when multiple + threads are used. + + Further, our usage of it today is just to ultimately get IPPROTO_TCP + for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. + + Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48 + +2017-03-19 13:26 +0000 [516e028b44] Sean Bright + + * res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret + + We are currently passing in the capacity of the read buffer instead of the + number of bytes that we actually read off the wire. + + Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36 + +2017-03-14 09:27 +0000 [79069f8ccb] Robert Mordec + + * app_queue: Member stuck as pending after forwarding previous call from queue + + Queue member will get stuck in pending_members if queue calls a device + that is different from the one observed for state changes. + + This patch removes members from pending_members as a result of channel stasis + events such as blind or attended transfers and hangup. + + ASTERISK-26862 #close + + Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727 + +2017-02-22 23:26 +0000 [8cb4f9cea1] Richard Mudgett + + * CHANNEL(callid): Give dialplan access to the callid. + + * Added CHANNEL(callid) to retrieve the call identifier log tag associated + with the channel. Dialplan now has access to the call log search key + associated with the channel so it can be saved in case there is a problem + with the call. + + ASTERISK-26878 + + Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f + +2017-03-16 08:42 +0000 [c13ea6080e] Sean Bright + + * app_queue: Fix locking behavior in stasis message handlers + + The queue_stasis_data structure contains various mutable fields that require + appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and + 'caller_uniqueid' fields need to be locked when read from or written to. + + Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088 + +2017-03-07 19:28 +0000 [15aa3c0a23] Sean Bright + + * chan_sip: Add rtcp-mux support + + ASTERISK-26846 #close + + Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639 + +2017-03-16 16:50 +0000 [57656e2b5b] Richard Mudgett + + * app_confbridge: Fix ConfbridgeTalking AMI event description. + + Thanks to Chris Howard for pointing this out on the wiki. + + Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705 + +2017-03-16 16:37 +0000 [82982a191c] Richard Mudgett + + * res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed. + + struct ast_rtcp does not define the dtls member if SRTP is not enabled. + + ASTERISK-26732 + + Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e + +2017-03-16 15:45 +0000 [49b1f1ca16] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Fix cut-n-paste error + + We were inadvertenly referencing the cos_video option to determine if we + should set the tos_audio and cos_audio value on the RTP instance. + + Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0 + +2017-03-16 10:39 +0000 [e6dc28b78f] Matt Jordan + + * res/res_pjsip_session: Only check localnet if it is defined + + If local_net is not defined on a transport, transport_state->localnet + will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in + this case, causing the external_media_address, if set, to be skipped. + + This patch causes us to only check if we are sending within a network if + local_net is defined. + + ASTERISK-26879 #close + + Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb + +2017-03-14 16:22 +0000 [44568fc712] Richard Begg + + * res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport + + Currently a wildcard address is used for the local RTP socket, which + will not always result in the same address as used by the SIP socket + (e.g. if explicit transport addresses are configured). + Use the transport's host address when binding new local RTP sockets if + available. + + ASTERISK-26851 + + Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a + +2017-03-07 08:33 +0000 [5013d8f5d3] George Joseph + + * res_pjsip: Symmetric transports + + A new transport parameter 'symmetric_transport' has been added. + + When a request from a dynamic contact comes in on a transport with + this option set to 'yes', the transport name will be saved and used + for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. + It's saved as a contact uri parameter named 'x-ast-txp' and will + display with the contact uri in CLI, AMI, and ARI output. On the + outgoing request, if a transport wasn't explicitly set on the + endpoint AND the request URI is not a hostname, the saved transport + will be used and the 'x-ast-txp' parameter stripped from the + outgoing packet. + + * config_transport was modified to accept and store the new parameter. + + * config_transport/transport_apply was updated to store the transport + name in the pjsip_transport->info field using the pjsip_transport->pool + on UDP transports. + + * A 'multihomed_on_rx_message' function was added to + pjsip_message_ip_updater that, for incoming requests, retrieves the + transport name from pjsip_transport->info and retrieves the transport. + If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter + containing the transport name is added to the incoming Contact header. + + * An 'ast_sip_get_transport_name' function was added to res_pjsip. + It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a + transport name if endpoint->transport is set or if there's an + 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or + ipv6 address. Otherwise it returns NULL. + + * An 'ast_sip_dlg_set_transport' function was added to res_pjsip + which takes an ast_sip_endpoint, a pjsip_dialog, and an optional + pjsip_tpselector. It calls ast_sip_get_transport_name() and if + a non-NULL is returned, sets the selector and sets the transport + on the dialog. If a selector was passed in, it's updated. + + * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas + were modified to call ast_sip_dlg_set_transport() instead of their + original logic. + + * res_pjsip/create_out_of_dialog_request was modified to call + ast_sip_get_transport_name() and pjsip_tx_data_set_transport() + instead of its original logic. + + * Existing transport logic was removed from endpt_send_request + since that can only be called after a create_out_of_dialog_request. + + * res_pjsip/ast_sip_create_rdata was converted to a wrapper around + a new 'ast_sip_create_rdata_with_contact' function which allows + a contact_uri to be specified in addition to the existing + parameters. (See below) + + * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated + since all it did was transport selection and that is now done in + ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. + + * 'contact_uri' was added to subscription_persistence. This was + necessary because although the parsed rdata contact header has the + x-ast-txp parameter added (if appropriate), + subscription_persistence_update stores the raw packet which + doesn't have it. subscription_persistence_recreate was then + updated to call ast_sip_create_rdata_with_contact with the + persisted contact_uri so the recreated subscription has the + correct transport info to send the NOTIFYs. + + * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since + all it did was transport selection and that is now done in + ast_sip_create_dialog_uac. + + * pjsip_message_ip_updater/multihomed_on_tx_message was updated + to remove all traces of the x-ast-txp parameter from the + outgoing headers. + + NOTE: This change does NOT modify the behavior of permanent + contacts specified on an aor. To do so would require that the + permanent contact's contact uri be updated with the x-ast-txp + parameter and the aor sorcery object updated. If we need to + persue this, we need to think about cloning permanent contacts into + the same store as the dynamic ones on an aor load so they can be + updated without disturbing the originally configured value. + + You CAN add the x-ast-txp parameter to a permanent contact's uri + but it would be much simpler to just set endpoint->transport. + + Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f + +2017-03-16 09:07 +0000 [68749a9fa7] Joshua Colp + + * res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped. + + This change removes an assumption that when DTLS is stopped + an RTCP session will be present on the RTP session. This is not + always the case. + + ASTERISK-26732 + + Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611 + +2017-03-15 13:24 +0000 [c87e7dd9ec] Richard Mudgett + + * autochan/mixmonitor/chanspy: Fix unsafe channel locking and references. + + Dereferencing struct ast_autochan.chan without first calling + ast_autochan_channel_lock() is unsafe because the pointer could change at + any time due to a masquerade. Unfortunately, ast_autochan_channel_lock() + itself uses struct ast_autochan.chan unsafely and can result in a deadlock + if the original channel happens to get destroyed after a masquerade in + addition to the pointer getting changed. + + The problem is more likely to happen with v11 and earlier because + masquerades are used to optimize out local channels on those versions. + However, it could still happen on newer versions if the channel is + executing a dialplan application when the channel is transferred or + redirected. In this situation a masquerade still must be used. + + * Added a lock to struct ast_autochan to safely be able to use + ast_autochan.chan while trying to get the channel lock in + ast_autochan_channel_lock(). The locking order is the channel lock then + the autochan lock. Locking in the other direction requires deadlock + avoidance. + + * Fix unsafe ast_autochan.chan usages in app_mixmonitor.c. + + * Fix unsafe ast_autochan.chan usages in app_chanspy.c. + + * app_chanspy.c: Removed unused autochan parameter from next_channel(). + + ASTERISK-26867 + + Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592 + +2017-03-07 14:13 +0000 [10fa49e327] Mark Michelson + + * Add rtcp-mux support + + This commit adds support for RFC 5761: Multiplexing RTP Data and Control + Packets on a Single Port. Specifically, it enables the feature when + using chan_pjsip. + + A new option, "rtcp_mux" has been added to endpoint configuration in + pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with + whatever it communicates with. Asterisk follows the rules set forth in + RFC 5761 with regards to falling back to standard RTCP behavior if the + far end does not indicate support for rtcp-mux. + + The lion's share of the changes in this commit are in + res_rtp_asterisk.c. This is because it was pretty much hard wired to + have an RTP and an RTCP transport. The strategy used here is that when + rtcp-mux is enabled, the current RTCP transport and its trappings (such + as DTLS SSL session) are freed, and the RTCP session instead just + mooches off the RTP session. This leads to a lot of specialized if + statements throughout. + + ASTERISK-26732 #close + Reported by Dan Jenkins + + Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5 + +2017-03-14 08:49 +0000 [dc4cdafd42] Torrey Searle + + * res/res_pjsip_refer: call xfer w/o extension + + When transfering to a URI without an extension, ensure that the + s extension of the dialplan is entered + + ASTERISK-26869 #close + + Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525 + +2017-03-09 11:05 +0000 [982d6173c5] Sean Bright + + * app_queue: Handle the caller being redirected out of a queue bridge + + A caller can leave the Queue() application after being bridged with a + member in a few ways: + + * Caller or member hangup + * Caller is transferred somewhere else (blind or atx) + * Caller is externally redirected elsewhere + + The first 2 scenarios are currently handled by subscribing to stasis + messages, but the 3rd is not explicitly covered. If a caller is + redirected away from the Queue() application, the member who was last + bridged with that caller will remain in an "In use" state until the + caller hangs up. + + This patch adds handling of the caller leaving the queue via + redirection. We monitor the caller-member bridge, and if the caller is + the one that leaves, we treat it the same as we would a caller hangup. + + ASTERISK-26400 #close + Reported by: Etienne Lessard + + Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334 + +2017-03-15 08:44 +0000 [0b8a57af6d] Joshua Colp + + * res_pjsip_endpoint_identifier_ip: Don't output error if no header_match. + + This change ensures that if no header_match option is set on an + identify an error message is not output stating the option is set + to an invalid value. + + ASTERISK-26863 + + Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a + +2017-03-14 07:50 +0000 [1475604eff] Matt Jordan + + * res_pjsip_endpoint_identifier_ip: Add an option to match requests by header + + This patch adds a new features to the endpoint identifier module, + 'match_header'. When set, inbound requests are matched by a provided SIP + header: value pair. This option works in conjunction with the existing + 'match' configuration option, such that if any 'match*' attribute + matches an inbound request, the request is associated with the specified + endpoint. + + Since this module now identifies by more than just IP address, + appropriate renaming of the module and/or variables can be done in a + non-release branch. + + ASTERISK-26863 #close + + Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 + (cherry picked from commit 30f52d79d7fc9ab0b628bef2b61ea515413795a2) + +2017-03-14 16:16 +0000 [f997090877] Richard Mudgett + + * pbx.c: Fix crash from malformed exten pattern. + + Forgetting to indicate an exten is a pattern can cause a crash if the + "pattern" has a character set range. e.g., "9999[3-5]" The crash is due + to a buffer overwrite because the '-' exten eye-candy wasn't removed as + expected and overran the allocated space. + + The buffer overwrite is fixed two ways in this patch. + + 1) Fix ext_strncpy() to distinguish between pattern and non-pattern + extens. Now '-' characters are removed when they are eye-candy and not + when they are part of a pattern character set. Since the function is + private to pbx.c, the return value now returns the number of bytes written + to the destination buffer instead of the strlen() of the final buffer so + the callers that care don't need to add one. + + 2) Fix callers to ext_strncpy() to supply the correct available buffer + size of the destination buffer. + + ASTERISK-26668 + + Change-Id: I555d97411140e47e0522684062d174fbe32aa84a + +2017-03-14 16:51 +0000 [0dc007e94d] Richard Begg + + * chan_iax2: Reload of iax peer results in loss of host address/port + + When using a non-dynamic peer address, build_peer() invalidates the + peer address structure by setting the address family to unspecified. + However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() + will not amend the peer address if the cache is still valid, resulting + in peer connectivity failures. + To fix this, we call ast_dnsmgr_refresh() instead. + + ASTERISK-26865 + + Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082 + +2017-03-14 15:12 +0000 [59130260e7] Matt Jordan + + * configure: Don't use the progress bar with curl when downloading to stdout + + In some scenarios, such as when there may not be a terminal (such as + inside a Docker container), curl will apparently direct the progress bar + to stdout. This can cause extra data to be appended to a file curl'd + down to stdout, resulting in md5 verification failures. + + This patch removes the progress bar, and tells curl to download the file + silently. + + ASTERISK-26872 #close + + Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c + +2017-03-02 17:11 +0000 [8470c2bdea] George Joseph + + * RFC sdp: Initial SDP creation + + * Added additional fields to ast_sdp_options. + * Re-organized ast_sdp. + * Updated field names to correspond to RFC4566 terminology. + * Created allocs/frees for SDP children. + * Created getters/setters for SDP children where appropriate. + * Added ast_sdp_create_from_state. + * Refactored res_sdp_translator_pjmedia for changes. + + Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48 + +2017-03-14 09:55 +0000 [05713c36ea] Matt Jordan + + * configs/samples/hep.conf.sample: Clarify how the HEP stack works + + This patch updates the documenation in hep.conf.sample to better specify + how the various HEP modules interact. + + ASTERISK-26717 #close + + Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124 + +2017-03-14 09:59 +0000 [0ded269bfa] Matt Jordan + + * funcs/func_devstate: Remove new line in Device field of during module load + + During module loading of func_devstate, Asterisk emits the current + device state of all Custom device states currently stored in the AstDB. + This was erroneously including a new line character ('\n') to the end of + the device state, causing two new lines to be emitted in + DeviceStateChange AMI events. + + Note that this only happened for those device state changes that + occurred during startup. Regular device state changes for Custom device + states are handled elsewhere, and did not have the newline. + + ASTERISK-26643 #close + Reported by: Roman Bedros + Tested by: Matt Jordan + patches: + ami_devstate.diff uploaded by Roman Bedros (License 6842) + + Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93 + +2017-03-14 09:37 +0000 [b03b72717f] Matt Jordan + + * main/stasis_cache: Demote the ERROR message when removing a nonexistent item + + This patch demotes the ERROR message that is displayed when a + nonexistent item is removed from the Stasis cache. The genesis of this + demotion is due to chan_sip's realtime peers and their interaction with + Asterisk's core ast_endpoint code, but ostensibly it could happen from + other channel drivers as well. + + Since Mark Michelson already did an excellent job of explaining on this + issue, it is quoted here for posterity: + + "Internally, when a realtime peer is retrieved, Asterisk creates an + ast_endpoint structure. When that peer is destroyed, the ast_endpoint is + destroyed as well. Part of the destruction of the ast_endpoint involves + clearing the Stasis cache of all information about that endpoint. The + problem here is that the act of creating the ast_endpoint is not enough + to actually put any information in the Stasis cache. Instead, something + has to happen, such as a state change, in order for the Stasis cache to + have any information about that endpoint. When a device registers, + chan_sip creates an ast_endpoint structure, processes the REGISTER, and + then destroys the ast_endpoint. When the ast_endpoint is destroyed, + there is nothing to destroy in the Stasis cache, so an error message is + emitted. When you use rtcachefriends, ast_endpoint structures persist + for the lifetime of the module and so you do not see this error + message." + + ASTERISK-25237 #close + + Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70 + +2017-03-08 12:39 +0000 [2d7e68c075] Matt Jordan + + * res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue + + Tabs > spaces. Always. + + Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1 + +2017-03-12 09:21 +0000 [12460b05c1] Joshua Colp + + * chan_pjsip: Don't assume a session will have a channel. + + When querying for PJSIP specific information using the dialplan + function CHANNEL() it is possible that the underlying session + will no longer have a channel associated with it. This is + most likely to occur when the RTCP HEP module attempts to get + the channel name. If this happens then a crash will occur. + + This change just adds a check that the channel exists on the + session before querying it. + + ASTERISK-26857 + + Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01 + +2017-03-10 20:29 +0000 [d1ef127084] George Joseph + + * pjproject_bundled: Reduce the need for rebuilds + + Bundled pjproject should now only rebuild if one of the menuselect + "Compiler Flags" options changes. + + Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43 + +2017-03-05 15:26 +0000 [36fed72614] Daniel Journo + + * pjsip/cli_commands: pjsip show channelstats shows wrong codec + + * cli_commands.c Fixed CLI output + + ASTERISK-26822 #close + + Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01 + +2017-03-08 14:29 +0000 [b14724adb3] Daniel Journo + + * res_musiconhold: moh general section is a class and issues warning + + * res_musiconhold.c: Ensure the general section is not treated as + a moh class. + + ASTERISK-26353 #close + + Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d + +2017-03-08 17:08 +0000 [35cfd2c0cc] Sean Bright + + * media_cache: Prefer ast_file_is_readable() over access() + + Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def + +2017-03-07 06:25 +0000 [bc2c66b594] Sean Bright + + * pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel + + Set a variable on the channel that indicates which attempt number we + are currently performing to allow for attempt-specific behavior. + + ASTERISK-26568 #close + Reported by: Roman Shubovich + + Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89 + +2017-03-07 07:37 +0000 [4e3b0cedba] Joshua Colp + + * res_pjsip_transport_websocket: Add support for IPv6. + + This change adds a PJSIP patch (which has been contributed upstream) + to allow the registration of IPv6 transport types. + + Using this the res_pjsip_transport_websocket module now registers + an IPv6 Websocket transport and uses it for the corresponding + traffic. + + ASTERISK-26685 + + Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647 + +2017-03-08 08:16 +0000 [60998371e3] Daniel Journo + + * app_voicemail: Cannot set fromstring on a per-mailbox basis + + * apps/app_voicemail.c fromstring field added to mailbox which will + override the global fromstring if set. + + ASTERISK-24562 #close + + Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe + +2017-03-07 13:38 +0000 [5d0371d743] Mark Michelson + + * res_http_websocket: Fix faulty read logic. + + When doing some WebRTC testing, I found that the websocket would + disconnect whenever I attempted to place a call into Asterisk. After + looking into it, I pinpointed the problem to be due to the iostreams + change being merged in. + + Under certain circumstances, a call to ast_iostream_read() can return a + negative value. However, in this circumstance, the websocket code was + treating this negative return as if it were a partial read from the + websocket. The expected length would get adjusted by this negative + value, resulting in the expected length being too large. + + This patch simply adds an if check to be sure that we are only updating + the expected length of a read when the return from a read is positive. + + ASTERISK-26842 #close + Reported by Mark Michelson + + Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab + +2017-03-07 08:12 +0000 [d51ca4b406] Jean Aunis + + * chan_sip: Call not cancelled after receiving a 422 response + + When receiving a 422 response, the invitestate variable must be reset to + INV_CALLING. + + ASTERISK-26841 + + Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099 + +2017-03-07 05:22 +0000 [3ed05badb9] Joshua Colp + + * core: Add stream topology changing primitives with tests. + + This change adds a few things to facilitate stream topology changing: + + 1. Control frame types have been added for use by the channel driver + to notify the application that the channel wants to change the stream + topology or that a stream topology change has been accepted. They are + also used by the indicate interface to the channel that the application + uses to indicate it wants to do the same. + + 2. Legacy behavior has been adopted in ast_read() such that if a + channel requests a stream topology change it is denied automatically + and the current stream topology is preserved if the application is + not capable of handling streams. + + Tests have also been written which confirm the multistream and + non-multistream behavior. + + ASTERISK-26839 + + Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9 + +2017-03-06 15:54 +0000 [272259a2c6] Daniel Journo + + * Saynumber is trying to get "and" from "digits/" subfolder + + * say.c Changed 'digits/and' to 'vm-and' for en_GB + + ASTERISK-26598 #close + + Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe + +2017-03-06 13:15 +0000 [5a74abc53b] Sean Bright + + * pbx_spool: Gracefully handle long lines in call files + + Per the linked issue, we aren't checking the buffer filled by fgets() + to determine if it contains a newline, so we will fail to correctly + parse the trailing portion of a long line. + + This patch increases the buffer size from 256 to 1024, and skips any + line that exceeds that length, logging a warning in the process. + + ASTERISK-17067 #close + Reported by: Dave Olszewski + + Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0 + +2017-03-02 21:27 +0000 [c9296b23d1] Richard Mudgett + + * core: Cleanup ast_get_hint() usage. + + * manager.c:manager_state_cb() Fix potential use of uninitialized hint[] + if a hint does not exist for the requested extension. Ran into this when + developing a testsuite test. The AMI event ExtensionStatus came out with + the hint header value containing garbage. The AMI event PresenceStatus + also had the same issue. + + * manager.c:action_extensionstate() no need to completely initialize the + hint[]. Only initialize the first element. + + * pbx.c:ast_add_hint() Remove unnecessary assignment. + + * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care + about the return value of ast_get_hint() there. + + Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b + +2017-02-16 04:22 +0000 [7922f26cb0] Jørgen H + + * res_pjsip WebRTC/websockets: Fix usage of WS vs WSS. + + According to the RFC[1] WSS should only be used in the Via header + for secure Websockets. + + * Use WSS in Via for secure transport. + + * Only register one transport with the WS name because it would be + ambiguous. Outgoing requests may try to find the transport by name and + pjproject only finds the first one registered. This may mess up unsecure + websockets but the impact should be minimal. Firefox and Chrome do not + support anything other than secure websockets anymore. + + * Added and updated some debug messages concerning websockets. + + * security_events.c: Relax case restriction when determining security + transport type. + + * The res_pjsip_nat module has been updated to not touch the transport + on Websocket originating messages. + + [1] https://tools.ietf.org/html/rfc7118 + + ASTERISK-26796 #close + + Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12 + +2017-02-24 15:30 +0000 [0560c32375] George Joseph + + * stream: Unit tests for stream read and tweaks framework + + * Removed the AST_CHAN_TP_MULTISTREAM tech property. We now rely + on read_stream being set to indicate a multi stream channel. + * Added ast_channel_is_multistream convenience function. + * Fixed issue where stream and default_stream weren't being set on + a frame retrieved from the queue. + * Now testing for NULL being returned from the driver's read or + read_stream callback. + * Fixed issue where the dropnondefault code was crashing on a + NULL f. + * Now enforcing that if either read_stream or write_stream are + set when ast_channel_tech_set is called that BOTH are set. + * Added the unit tests. + + ASTERISK-26816 + + Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2 + +2017-03-01 07:23 +0000 [1dacf317f3] Sean Bright + + * res_config_pgsql: Make 'require' return consistent with other backends + + res_config_pgsql should match the behavior of other realtime backend + drivers so that queue_log can disable adaptive logging. + + ASTERISK-25628 #close + Reported by: Dmitry Wagin + + Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372 + +2017-02-22 15:11 +0000 [9c55a71798] Mark Michelson + + * SDP: Add initial SDP state machine. + + This introduces and documents the various states in the state machine. + This also introduces API functions that induce state changes, and places + TODO comments telling what needs to be done in addition to what is + already there. Those TODOs will be replaced with real code in upcoming + changes. + + Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed + +2017-02-28 13:48 +0000 [60e9e4fcc0] Sean Bright + + * media_cache: Mark cache entry stale if cache file is removed + + In the event that a cache file is removed out from under us, we should + treat the cache entry as stale and force a refresh. + + ASTERISK-26774 #close + Reported by: Igor Gamayunov + + Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52 + +2017-02-28 09:41 +0000 [e5b44c26b4] Sean Bright + + * res_config_pgsql: Release table locks where appropriate + + The find_table() functions NULL or a locked table pointer. We are + not consistently calling release_table() in failure paths. + + Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544 + +2017-02-28 05:41 +0000 [6ebdcfe27d] Tzafrir Cohen + + * pjsip.conf.sample: user_agent: not a specific version + + Use the description of useragent from sip.conf here. + + ASTERISK-26825 #close + + Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755 + +2017-02-27 20:07 +0000 [fb68db87b1] George Joseph + + * res_pjsip_pubsub: Remove unneeded endpoint unref + + When a subscription was being recreated and the endpoint wasn't + found, we were trying to unref the endpoint. This was causing + FRACKs. Removed the unref. + + ASTERISK-26823 #close + + Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164 + +2017-02-16 04:16 +0000 [ee0a123f43] Jørgen H + + * res_pjsip: Fix crash when contact has no status + + This change fixes an assumption in res_pjsip that a contact will + always have a status. There is a race condition where this is + not true and would crash. The status will now be unknown when + this situation occurs. + + ASTERISK-26623 #close + + Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5 + +2017-02-21 18:06 +0000 [22242fef5d] George Joseph + + * res_pjsip_outbound_registration: Subscribe to network change events + + Outbound registration now subscribes to network change events + published by res_stun_monitor and refreshes all registrations + when an event happens. + + The 'pjsip send (un)register' CLI commands were updated to accept + '*all' as an argument to operate on all registrations. + + The 'PJSIP(Un)Register' AMI commands were also updated to + accept '*all'. + + ASTERISK-26808 #close + + Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25 + +2017-02-27 12:25 +0000 [4692a32ed7] George Joseph + + * build: Warn if asterisk is installed in both 32 and 64 bit sys dirs + + ... and clean them both up on uninstall. + + We've fixed the issue where 'make install' was installing to + /usr/lib on 64-bit systems that use /usr/lib64. Now we need + to clean up the remnants in /usr/lib. + + * 'make install' now prints a warning if DESTDIR/ASTLIBDIR + contains 'lib64' and libasterisk* shared libraries or modules + are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed + to 'lib'. + + * 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and + DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'. + + ASTERISK-26705 + + Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f + +2017-02-27 07:02 +0000 [ff2b4308d1] Joshua Colp + + * bridge_native_rtp: Handle case where channel joins already suspended. + + The bridge_native_rtp module did not properly handle the case where + a smart bridge operation occurs while a channel is suspended. In this + scenario the module would incorrectly set up local or remote RTP + bridging despite the media having to flow through Asterisk. The remote + endpoint would see two media streams and experience wonky audio. + + The module has been changed so that it ensures both channels are + not suspended when performing the native RTP bridging and this + requirement has been documented in the bridge technology. + + ASTERISK-26781 + + Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c + +2016-08-12 11:23 +0000 [5b1796f59d] frahaase + + * Binaural synthesis (confbridge): DTMF conference management. + + DTMF configuration options for the binaural softmix bridge: + toggle binaural rendering (per channel). + + ASTERISK-26292 + + Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8 + +2017-02-24 11:49 +0000 [2046743938] Joshua Colp + + * config: Improve documentation and behavior of outbound_proxy option. + + This change updates the documentation for the outbound_proxy option + to ensure it is consistently stated that a full SIP URI must be + provided for the option. + + The res_pjsip_outbound_registration module has also been changed so + that the provided outbound_proxy value is checked to ensure it is a + URI and if not an error is output stating so. + + ASTERISK-26782 + + Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593 + +2017-02-23 13:03 +0000 [c07c6714f2] Joshua Colp + + * channel: Add ast_read_stream function for reading frames from all streams. + + This change introduces an ast_read_stream function and callback in + the channel technology which allows reading frames from all streams + and not just the default streams. + + The stream number has also been added to frames. This is to allow the + case where frames are queued onto the channel instead of being read + directly from the driver. + + This change does impose a restriction on reading though: a chain of + frames can only contain frames from the same stream. + + ASTERISK-26816 + + Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce + +2017-02-09 18:05 +0000 [a537dae6d0] George Joseph + + * pjproject_bundled: Update for pjproject 2.6 + + * Removed all 2.5.5 functional patches. + * Updated usages of pj_release_pool to be "safe". + * Updated configure options to disable webrtc. + * Updated config_site.h to disable webrtc in pjmedia. + * Added Richard Mudgett's recent resolver patches. + + Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7 + +2017-02-23 15:49 +0000 [b0067bcf2c] George Joseph + + * build: Execute ldconfig to build cache. (take two) + + On some platforms a multiarch approach is used for libraries. + The build system does not take this into account and still + places libraries into the lib directory if no --libdir is + specified to configure. On initial startup this results in + libasteriskssl.so not being found, as it is not in the multiarch + lib directory. To make matters worse, options were being passed + to ldconfig on both Linux and FreeBSD that actually prevented + the rebuild of the cache. + + * Fedora has a /usr/share/config.site that automatically tells + autoconf to use /usr/lib64 but CentOS does not. This logic was + copied to configure.ac and modified so systems like Ubuntu, + which still use /usr/lib for 64-bit systems, aren't affected. + + Now that we have them in the correct directory... + + In order for the system loader to find libasteriskssl and + libasteriskpj, one of 3 things has to happen... + + - The linker cache must be rebuilt including the directory + where the libasterisk* libraries were installed. Only root + can rebuild the cache. This was busted. + - We have to link the asterisk binary with an rpath pointing + to the directrory where the libasterisk* libraries were + installed. This makes things very complicated and will happen + over the collective dead bodies of everyone who's had to + package a distribution with an rpath. + - Finally, you can start asterisk with LD_LIBRARY_PATH set to the + directrory where the libasterisk* libraries were installed. + + There are no other options. So... + + * The invokation of ldconfig has been moved from main/Makefile + to ASTTOPDIR/Makefile, the options have been removed, and + DESTDIR/ASTLIBDIR appended. If you aren't root, you will be + warned after the "Asterisk Installation Compete" banner that + you must re-run 'make install' as root, manually run + 'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with + LD_LIBRARY_PATH. + + ASTERISK-26705 + + Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982 + +2017-02-23 14:48 +0000 [0f4b349d37] Sean Bright + + * res_config_pgsql: Fix thread safety problems + + * A missing AST_LIST_UNLOCK() in find_table() + + * The ESCAPE_STRING() macro uses pgsqlConn under the hood and we were + not consistently locking before calling it. + + * There were a handful of other places where pgsqlConn was accessed + directly without appropriate locking. + + Change-Id: Iea63f0728f76985a01e95b9912c3c5c6065836ed + +2017-02-22 05:00 +0000 [6cc890b880] Joshua Colp + + * channel: Add support for writing to a specific stream. + + This change adds an ast_write_stream function which allows + writing a frame to a specific media stream. It also moves + ast_write() to using this underneath by writing media + frames provided to it to the default streams of the channel. + Existing functionality (such as audiohooks, framehooks, etc) + are limited to being applied to the default stream only. + + Unit tests have also been added which test the behavior of + both non-multistream and multistream channels to confirm that + the write() and write_stream() callbacks are invoked + appropriately. + + ASTERISK-26793 + + Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c + +2016-08-12 11:23 +0000 [094c26aa68] frahaase + + * Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix. + + Adds binaural synthesis to bridge_softmix (via convolution using libfftw3). + Binaural synthesis is conducted at 48kHz. + For a conference, only one spatial representation is rendered. + The default rendering is applied for mono-capable channels. + + ASTERISK-26292 + + Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf + +2017-02-22 08:53 +0000 [e57961db84] Sean Bright + + * res_config_ldap: Various code improvements + + The initial motivation for this patch was to properly handle memory + allocation failures - we weren't checking the return values from the + various LDAP library allocation functions. + + In the process, because update_ldap() and update2_ldap() were + substantially the same code, they've been consolidated. + + Change-Id: Iebcfe404177cc6860ee5087976fe97812221b822 + +2017-02-22 13:08 +0000 [66a35e2451] Michael L. Young + + * build_tools: Fix download_externals to allow the use of curl or wget + + Not sure if this is really a bug versus an improvement. I can see it being + viewed as a bug though by some. + + The current build_tools/download_externals file depends on wget in order to + download external modules. The current build system is able to discover + which tool to use for fetching remote files - either wget or curl. + + This patch takes advantage of this capability by modifying the two calls to + the wget binary to instead use what was discovered by the build system. + + ASTERISK-26812 #close + + Change-Id: If9411a2554f009274d377445613ae91192d948a1 + +2017-02-22 11:12 +0000 [ced73d5b79] Joshua Colp + + * Revert "build: Execute ldconfig to build cache." + + This reverts commit 28c8e4f58f0f38792c7c79a05bd07788ebf15332. + + Change-Id: Ie2e1aaf61fd49045994974a4581545ac8348fe4c + +2017-02-21 10:47 +0000 [15ed7af027] Sean Bright + + * pbx_realtime: Prevent premature extension matching + + The patterns provided by pbx_realtime were checked in the order in + which they were returned from the realtime backend. If there was + overlap between multiple patterns, the first one to correctly match was + chosen even though it may not have been the best match. + + We now sort the patterns descending by their length and compare in that + order. There may be cases where this still results in a sub-optimal + match, but this patch should improve the overall behavior. + + ASTERISK-18271 #close + Reported by: Charlie Smurthwaite + + Change-Id: I56d9ac15810eb1775966b669c3028e32cc7bd809 + +2017-02-22 08:32 +0000 [f58aefba5b] Joshua Colp + + * core: Show streams in "core show channel". + + The "core show channel" CLI command will now output the streams + present on the channel with their details. + + ASTERISK-26811 + + Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a + +2017-02-21 15:09 +0000 [fc70ca9499] Sean Bright + + * pbx_dundi: DUNDi weight parameter not processed correctly + + The DUNDi weight field is not always converted from network byte order + to host byte order. This can result in incorrect weight values and + incorrect selection of DUNDi destinations. + + ASTERISK-18731 #close + Reported by: Peter Racz + Patches: + dundi_weight.patch (license #6290) patch uploaded by Peter Racz + + Change-Id: Iba3e1a700ff539db57211a7bbc26f7b22ea9a1be + +2017-02-15 14:43 +0000 [a738772edd] Mark Michelson + + * Add initial SDP state code. + + This establishes the basic allocation/destruction of an SDP state + object, plus some of the simpler getter methods involved. Subsequent + tasks will deal with adding a state machine, creating SDPs from + capabilities and options, and merging SDPs into a joint SDP. + + Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709 + +2017-02-21 10:47 +0000 [ab04a018e4] Sean Bright + + * realtime: Fix ast_load_realtime_multientry handling + + ast_load_realtime_multientry() returns an ast_config structure whose + ast_categorys are keyed with the empty strings. Several modules were + giving semantic meaning to the category names causing problems at + runtime. + + * app_directory: Treated the category name as the mailbox name, and + would fail to direct calls to the appropriate extension after an + entry was chosen. + + * app_queue: Queues, queue members, and queue rules were all affected + and needed to be updated. + + * pbx_realtime: Pattern matching would never succeed because the + extension entered by the user was always compared to the empty + string. + + Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7 + +2017-02-21 08:56 +0000 [6e6c96d713] Sean Bright + + * realtime: Centralize some common realtime backend code + + All of the realtime backends create artificial ast_categorys to pass + back into the core as query results. These categories have no filename + or line number information associated with them and the backends differ + slightly on how they create them. So create a couple helper macros to + help make things more consistent. + + Also updated the call sites to remove redundant error messages about + memory allocation failure. + + Note that res_config_ldap sets the category filename to the 'table name' + but that is not read by anything in the core, so I've dropped it. + + Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897 + +2017-02-16 10:30 +0000 [28c8e4f58f] Joshua Colp + + * build: Execute ldconfig to build cache. + + On some platforms a multiarch approach is used for libraries. + The build system does not take this into account and still + places libraries into the lib directory if no --libdir is + specified to configure. On initial startup this results in + libasteriskssl.so not being found, as it is not in the multiarch + lib directory. + + This change does the minimally invasive thing and executes + ldconfig so that the libraries in the lib directory are found + and their location cached. By doing so Asterisk starts up fine. + + If DESTDIR is specified, however, the old logic is executed as + the install process may not have permission to alter the ldconfig + cache. + + ASTERISK-26705 + + Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4 + +2017-01-08 20:32 +0000 [6f15500ced] Richard Mudgett + + * res_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation. + + The inbound authentication object is supposed to be immutable when it is + stored in sorcery. However, the immutable property is violated if the + authentication object does not have a realm set. + + The immutable contract violation has a different effect depending upon + what sorcery back end is used. If it is the config file back end you + would get the same object back until res_pjsip is reloaded. If it is the + real-time or AstDB back end you would get a new object on each query. If + it is cached you would get the same object back until it is refreshed from + the database. + + Once an inbound authentication object has its realm set it may or may not + get updated again if the default_realm changes. + + If the same authentication object is used for inbound and outbound + authentication then the immutable violation can make it very hard to + determine why the outbound authentication now fails. The only diagnostic + message is a complaint about no realms matching when it had worked + earlier. It fails because of the difference in behaviour for an empty + realm setting between inbound and outbound authentication objects. + + * Fixed the sorcery object immutable violation by creating a new object + and setting the default_realm on it instead. The new object is a shallow + copy for speed. + + * The auth_store thread storage no longer holds an auth ref. It + interferes with the shallow copy and never needed a ref anyway. + + ASTERISK-26799 #close + + Change-Id: I2328a52f61b78ed5fbba38180b7f183ee7e08956 + +2017-02-04 20:17 +0000 [6400f5f309] Richard Mudgett + + * res_pjsip: Update artificial auth whenever default_realm changes. + + There was code attempting to update the artificial authentication object + whenever the default_realm changed. However, once the artificial + authentication object was created it would never get updated. The + artificial authentication object would require a system restart for a + change to the default_realm to take effect. + + ASTERISK-26799 + + Change-Id: Id59036e9529c2d3ed728af2ed904dc36e7094802 + +2017-01-01 08:02 +0000 [0b660c9989] Richard Mudgett + + * res_pjsip: Update authentication realm documentation. + + Using the same auth section for inbound and outbound authentication is not + recommended. There is a difference in meaning for an empty realm setting + between inbound and outbound authentication uses. + + An empty inbound auth realm represents the global section's default_realm + value when the authentication object is used to challenge an incoming + request. An empty outgoing auth realm is treated as a don't care wildcard + when the authentication object is used to respond to an incoming + authentication challenge. + + ASTERISK-26799 + + Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce + +2017-02-13 17:11 +0000 [7f83bcd63d] Richard Mudgett + + * pjproject: Fixes to resolve DNS SRV crashes. + + * Re #1945 (misc): Don't trigger SRV complete callback when there is a + parse error. + + * srv_resolver.c: Don't try to send query if already considered resolved. + + ** In resolve_hostnames() don't try to resolve a query that is already + considered resolved. + + ** In resolve_hostnames() fix DNS typo in comments. + + ** In build_server_entries() move a common expression assigning to cnt + earlier. + + * sip_transport.c: Fix tdata object name to actually contain the pointer. + + It helps if the logs referencing a tdata object buffer actually have a + name that includes the correct pointer as part of the name. Also since + the tdata has its own pool it helps if any logs referencing the pool have + the same name as the tdata object. This change brings tdata logging in + line with how tsx objects are named. + + ASTERISK-26669 #close + ASTERISK-26738 #close + + Change-Id: I56af2ded25476b3e870ca586ee69ed6954ef75af + +2017-02-20 13:38 +0000 [bf78c3c9c3] Richard Mudgett + + * pjproject: Increase SENDER_WIDTH column size for 64-bit system logs. + + ASTERISK-26669 + ASTERISK-26738 + + Change-Id: Ibae6fc8cae69a1f04df0c577c4c11200499d6fe0 + +2017-02-06 14:26 +0000 [54812f18b5] Richard Mudgett + + * pjsip_distributor.c: Update some debug messages to get transaction name. + + * Removed overloaded unmatched response ignore. We obviously sent the + request so we shouldn't ignore it because it isn't new work. + + ASTERISK-26669 + ASTERISK-26738 + + Change-Id: I55fb5cadc83a8e6699b347c6dc7fa32c5a617d37 + +2017-02-20 06:28 +0000 [b18f1bfb13] Sean Bright + + * app_voicemail: vm_authenticate accesses uninitialized memory + + vm_authenticate doesn't always set the passed ast_vm_user argument, so + we initialize to 0 before passing it in. + + ASTERISK-25893 #close + Reported by: Filip Jenicek + + Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a + +2017-02-20 11:19 +0000 [7739b0b3ae] Joshua Colp + + * Revert "build: Execute ldconfig to build cache." + + This reverts commit 8851c3e0885cb704a5a6159a51768ea5297e9b10. + + Change-Id: I124380be5e3bd57da978428a2a93604336ccd0db + +2017-02-20 08:04 +0000 [ffa7d69766] George Joseph + + * pjproject cli: Add object count after object lists + + When listing a container, we now print the number of objects + in the container at the end of the list. + + Change-Id: I791cbc3ee9da9a2af9adc655164b5d32953df812 + +2017-02-20 05:53 +0000 [e84353b8a8] Sean Bright + + * res_config_ldap: Don't try to delete non-existent attributes + + OpenLDAP will raise an error when we try to delete an LDAP attribute + that doesn't exist. We need to filter out LDAP_MOD_DELETE requests + based on which attributes the current LDAP entry actually has. There + is of course a small window of opportunity for this to still fail, + but it is much less likely now. + + Change-Id: I3fe1b04472733e43151563aaf9f8b49980273e6b + +2017-02-20 05:49 +0000 [9f392574f9] Sean Bright + + * res_config_ldap: Remove extraneous line numbers from log messages + + Extraneous line numbers were being output in many log messages. These + have been removed. + + Change-Id: Ice9efa3d252ee87f37fa8f5ea852fda482675431 + +2017-02-20 05:45 +0000 [ef0944395e] Sean Bright + + * res_config_ldap: Make memory allocation more consistent + + The code in update_ldap() and update2_ldap() was using both Asterisk's + memory allocation routines as well as OpenLDAP's. I've changed it so + that everything that is passed to OpenLDAP's functions are allocated + with their routines. + + Change-Id: Iafec9c1fd8ea49ccc496d6316769a6a426daa804 + +2017-02-20 05:30 +0000 [dd3efdf525] Sean Bright + + * res_config_ldap: Fix configuration inheritance from _general + + The "_general" configuration section allows administrators to provide + both general configuration options (host, port, url, etc.) as well as a + global realtime-to-LDAP-attribute mapping that is a fallback if one of + the later sections do not override it. This neglected to exclude the + general configuration options from the mapping. As an example, during + my testing, chan_sip requested 'port' from realtime, and because I did + not have it defined, it pulled in the 'port' configuration option from + "_general." We now filter those out explicitly. + + Change-Id: I1fc61560bf96b8ba623063cfb7e0a49c4690d778 + +2017-02-20 05:27 +0000 [d6d86f1c09] Sean Bright + + * res_config_ldap: Fix erroneous LDAP_MOD_REPLACE in LDAP modify + + We always treat the first change of our modification batch as a + replacement when it sometimes is actually a delete. So we have to pass + the correct arguments to the OpenLDAP library. + + ASTERISK-26580 #close + Reported by: Nicholas John Koch + Patches: + res_config_ldap.c-11.24.1.patch (license #6833) patch uploaded + by Nicholas John Koch + + Change-Id: I0741d25de07c9539f1edc6eff3696165dfb64fbe + +2017-02-15 11:55 +0000 [44abe214d2] Sean Bright + + * res_config_sqlite3: Fix crash when loading with invalid config + + When ast_config_load() fails with CONFIG_STATUS_FILEINVALID, it has + already destroyed the ast_config struct for us. Trying to do it again + results in a crash. + + Change-Id: If6a5c0ca718ad428e01a1fb25beb209a9ac18bc6 + +2017-02-17 17:06 +0000 [51e3b11989] Sean Bright + + * pjproject-bundled: Fix checksum verification when using cURL + + ASTERISK-26802 #close + Reported by: Michael L. Young + + Change-Id: Iad293080f55d4d69ab615717a15211d916eed613 + +2017-02-17 16:57 +0000 [0b427f9b59] Richard Mudgett + + * tcptls.c: Add some missing allocation failure checks. + + * Fix tcptls_session ref and fd leak in ast_tcptls_server_root(). + + Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb + +2017-02-17 14:58 +0000 [dbc3598014] Mark Michelson + + * Remove extra ast_iostream_close() calls. + + When AMI encounters an error at the beginning of a session, it would + explicitly call ast_iostream_close() on its tcptls session's iostream. + It then would jump to a label where it would shut down the tcptls + session instance. The tcptls session instance would again attempt to + close the iostream. + + Under normal circumstances, this might go by unnoticed. However, when + MALLOC_DEBUG is enabled, all fields on the iostream get set to + 0xdeaddead when the iostream is freed. Thus a second call to + ast_iostream_close() after the iostream has been freed would reslt in an + attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn + horribly. + + The fix here is to not directly close the iostream from the dangerous + scenarios. The specific scenarios are: + * Exceeding the configured authlimit + * Failing to build a mansession on a new connection + + Change-Id: I908f98d516afd5a263bd36b072221008a4731acd + +2017-02-14 09:54 +0000 [5a130b2e17] Mark Michelson + + * Add SDP translator and PJMEDIA implementation. + + This creates the following: + * Asterisk's internal representation of an SDP + * An API for translating SDPs from one format to another + * An implementation of a translator for PJMEDIA + + Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b + +2017-02-07 09:50 +0000 [8af6342555] Mark Michelson + + * Add initial SDP options. + + This is step one of adding an SDP API: defining some + configurable settings for SDPs. This is based on options + that are currently supported in Asterisk. + + Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7 + +2017-02-16 10:30 +0000 [8851c3e088] Joshua Colp + + * build: Execute ldconfig to build cache. + + On some platforms a multiarch approach is used for libraries. + The build system does not take this into account and still + places libraries into the lib directory if no --libdir is + specified to configure. On initial startup this results in + libasteriskssl.so not being found, as it is not in the multiarch + lib directory. + + This change does the minimally invasive thing and executes + ldconfig so that the libraries in the lib directory are found + and their location cached. By doing so Asterisk starts up fine. + + ASTERISK-26705 + + Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519 + +2017-02-16 08:38 +0000 [e93f2a5142] Sean Bright + + * realtime: Fix LIKE escaping in SQL backends + + The realtime framework allows for components to look up values using a + LIKE clause with similar syntax to SQL's. pbx_realtime uses this + functionality to search for pattern matching extensions that start with + an underscore (_). + + When passing an underscore to SQL's LIKE clause, it will be interpreted + as a wildcard matching a single character and therefore needs to be + escaped. It is (for better or for worse) the responsibility of the + component that is querying realtime to escape it with a backslash before + passing it in. Some RDBMs support escape characters by default, but the + SQL92 standard explicitly says that there are no escape characters + unless they are specified with an ESCAPE clause, e.g. + + SELECT * FROM table WHERE column LIKE '\_%' ESCAPE '\' + + This patch instructs 3 backends - res_config_mysql, res_config_pgsql, + and res_config_sqlite3 - to use the ESCAPE clause where appropriate. + + Looking through documentation and source tarballs, I was able to + determine that the ESCAPE clause is supported in: + + MySQL 5.0.15 (released 2005-10-22 - earliest version available from + archives) + PostgreSQL 7.1 (released 2001-04-13) + SQLite 3.1.0 (released 2005-01-21) + + The versions of the relevant libraries that we depend on to access MySQL + and PostgreSQL will not work on versions that old, and I've added an + explicit check in res_config_sqlite3 to only use the ESCAPE clause when + we have a sufficiently new version of SQLite3. + + res_config_odbc already handles the escape characters appropriately, so + no changes were required there. + + ASTERISK-15858 #close + Reported by: Humberto Figuera + + ASTERISK-26057 #close + Reported by: Stepan + + Change-Id: I93117fbb874189ae819f4a31222df7c82cd20efa + +2017-02-16 08:28 +0000 [f8f513d363] George Joseph + + * stream: Rename creates/destroys to allocs/frees + + To be consistent with sdp implementation. + + Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500 + +2017-02-16 05:46 +0000 [30aaeec5a1] Sean Bright + + * res_config_sqlite3: Properly create missing columns when necessary + + There were two specific issues resolved here: + + 1) The code that iterated over the required fields + (via ast_realtime_require) was broken for the RQ_INTEGER1 field + type. Iteration would stop when the first RQ_INTEGER1 (0) field + was encountered. + + 2) sqlite3_changes() was used to try and count the number of rows + returned by a SELECT statement. sqlite3_changes() only counts + affected rows, so this was always returning the value from the + most recent data modification statement. We now separate read-only + queries from data modification queries and count rows appropriately + in both cases. + + ASTERISK-23457 #close + Reported by: Scott Griepentrog + + Change-Id: I91ed20494efc3fcfbc2a96ac7646999a49814884 + +2017-02-15 14:44 +0000 [ac7a34c531] Joshua Elson + + * http: Ensure capath is defined on all http creations + + ASTERISK-26794 #close + + Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1 + +2017-02-15 23:09 +0000 [135bea931c] Igor Goncharovsky + + * chan_unistim: fix char type to have consistent behavior on ARM + + There is difference exists in behaviour of char type on x86 and ARM. + On x86 by default char variable type means signed char, but in ARM + unsigned char used. This make binary calculations and negative values + works wrong on ARM. + + This patch change type of char variables used for store negative + values and binary calculations to signed char. + + ASTERISK-26714 + + Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab + +2017-02-07 13:17 +0000 [4bdf5d329f] George Joseph + + * res_pjsip_pubsub: Correctly implement persisted subscriptions + + This patch fixes 2 original issues and more that those 2 exposed. + + * When we send a NOTIFY, and the client either doesn't respond or + responds with a non OK, pjproject only calls our + pubsub_on_evsub_state callback, no others. Since + pubsub_on_evsub_state (which does the sub_tree cleanup) does not + expect to be called back without the other callbacks being called + first, it just returns leaving the sub_tree orphaned. Now + pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE + which is what pjproject will set to tell us that it was the + transaction that timed out or failed and not the subscription + itself timing our or being terminated by the client. If is + TSX_STATE, pubsub_on_evsub_state now does the proper cleanup + regardless of the state of the subscription. + + * When a client renews a subscription, we don't update the + persisted subscription with the new expires timestamp. This causes + subscription_persistence_recreate to prune the subscription if/when + asterisk restarts. Now, pubsub_on_rx_refresh calls + subscription_persistence_update to apply the new expires timestamp. + This exposed other issues however... + + * When creating a dialog from rdata (which sub_persistence_recreate + does from the packet buffer) there must NOT be a tag on the To + header (which there will be when a client refreshes a + subscription). If there is one, pjsip_dlg_create_uas will fail. + To address this, subscription_persistence_update now accepts a flag + that indicates that the original packet buffer must not be updated. + New subscribes don't set the flag and renews do. This makes sure + that when the rdata is recreated on asterisk startup, it's done + from the original subscribe packet which won't have the tag on To. + + * When creating a dialog from rdata, we were setting the dialog's + remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. + When the client tried to resubscribe after a restart with the + correct cseq, we'd reject the request with an Invalid CSeq error. + + * The acts of creating a dialog and evsub by themselves when + recreating a subscription does NOT restart pjproject's subscription + timer. The result was that even if we did correctly recreate the + subscription, we never removed it if the client happened to go away + or send a non-OK response to a NOTIFY. However, there is no + pjproject function exposed to just set the timer on an evsub that + wasn't created by an incoming subscribe request. To address this, + we create our own timer using ast_sip_schedule_task. This timer is + used only for re-establishing subscriptions after a restart. + + An earlier approach was to add support for setting pjproject's + timer (via a pjproject patch) and while that patch is still included + here, we don't use that call at the moment. + + While addressing these issues, additional debugging was added and + some existing messages made more useful. A few formatting changes + were also made to 'pjsip show scheduled tasks' to make displaying + the subscription timers a little more friendly. + + ASTERISK-26696 + ASTERISK-26756 + + Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e + +2017-02-15 11:03 +0000 [11886dea82] Sean Bright + + * res_rtp_asterisk: Use PJ_ICE_MAX_CAND instead of hard-coding 16 + + pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND, + which is a compile-time constant. Instead of hard-coding 16 when we + enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can + potentially collect more interfaces if the compile time options are + changed. + + Tangentially related to ASTERISK~24464 + + Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd + +2016-12-22 09:42 +0000 [b58de2fab7] Dennis Guse + + * Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer + + Adds the import tool for converting a HRIR database to hrirs.h + + ASTERISK-26292 + + Change-Id: I51eb31b54c23ffd9b544bdc6a09d20c112c8a547 + +2017-02-14 12:33 +0000 [a9c15a0e4c] Joshua Colp + + * stream: Add unit tests for channel stream usage. + + This change adds unit tests cover the following: + + 1. That retrieving the first media stream of a specific media + type from a stream topology retrieves the expected media + stream. + + 2. That setting the native formats of a channel which does + not support streams results in the creation of streams on + its behalf according to the formats of the channel. + + 3. That setting a stream topology on a channel which supports + streams sets the topology to the provided one. + + ASTERISK-26790 + + Change-Id: Ic53176dd3e4532e8c3e97d9e22f8a4b66a2bb755 + +2017-02-13 16:50 +0000 [275f469a4d] Sean Bright + + * app_voicemail: Allow 'Comedian Mail' branding to be overriden + + Original patch by John Covert, slight modifications by me. + + ASTERISK-17428 #close + Reported by: John Covert + Patches: + app_voicemail.c.patch (license #5512) patch uploaded by + John Covert + + Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6 + +2017-02-13 11:50 +0000 [bf2f091bbb] George Joseph + + * stream: Add stream topology to channel + + Adds topology set and get to channel. + + ASTERISK-26790 + + Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4 + +2017-01-25 16:25 +0000 [2b245b12d9] Ryan Rittgarn + + * app_voicemail: VoiceMailPlayMsg did not play database stored messages + + When attempting to use VoiceMailPlayMsg with a realtime data backend + the message is located, but never retrieved. This patch adds the + required RETRIEVE and DISPOSE calls that will fetch the message from + the database (and IMAP storage as well for that matter). + + Also, removed extraneous make_file call. + + ASTERISK-26723 #close + + Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c + +2017-02-14 08:12 +0000 [662c9e69fa] Sean Bright + + * app_record: Add option to prevent silence from being truncated + + When using Record() with the silence detection feature, the stream is + written out to the given file. However, if only 'silence' is detected, + this file is then truncated to the first second of the recording. + + This patch adds the 'u' option to Record() to override that behavior. + + ASTERISK-18286 #close + Reported by: var + Patches: + app_record-1.8.7.1.diff (license #6184) patch uploaded by var + + Change-Id: Ia1cd163483235efe2db05e52f39054288553b957 + +2017-02-07 11:13 +0000 [9f394d074a] Sebastian Gutierrez + + * app_queue: reset abandoned in sl for sl2 calculations + + ASTERISK-26775 #close + + Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284 + +2017-02-13 11:00 +0000 [6c4657e28e] Joshua Colp + + * stream: Add stream topology unit tests and fix uncovered bugs. + + This change adds unit tests for the various API calls relating + to stream topologies. This includes creation, destruction, + inspection, and manipulation. + + Through this a few bugs were uncovered in the implementation: + + 1. Creating a topology using a format capabilities would fail as + the code considered a return value of 0 from the append stream + function to indicate an error which is incorrect. + + 2. Not all functions which placed a stream into a topology + set the position on the stream itself. + + 3. Appending a stream would cause a frack if the position + provided was the last one. This occurred because the existing + stream was queried but the index was outside of what the + vector was currently at for size. + + ASTERISK-26786 + + Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0 + +2017-02-11 09:57 +0000 [3f94373778] Sean Bright + + * cli: Fix various CLI documentation and completion issues + + * app_minivm: Use built-in completion facilities to complete optional + arguments. + + * app_voicemail: Use built-in completion facilities to complete + optional arguments. + + * app_confbridge: Add missing colons after 'Usage' text. + + * chan_alsa: Use built-in completion facilities to complete optional + arguments. + + * chan_sip: Use built-in completion facilities to complete optional + arguments. Add completions for 'load' for 'sip show user', 'sip show + peer', and 'sip qualify peer.' + + * chan_skinny: Correct and extend completions for 'skinny reset' and + 'skinny show line.' + + * func_odbc: Correct completions for 'odbc read' and 'odbc write' + + * main/astmm: Use built-in completion facilities to complete arguments + for 'memory' commands. + + * main/bridge: Correct completions for 'bridge kick.' + + * main/ccss: Use built-in completion facilities to complete arguments + for 'cc cancel' command. + + * main/cli: Add 'all' completion for 'channel request hangup.' Correct + completions for 'core set debug channel.' Correct completions for 'core + show calls.' + + * main/pbx_app: Remove redundant completions for 'core show + applications.' + + * main/pbx_hangup_handler: Remove unused completions for 'core show + hanguphandlers all.' + + * res_sorcery_memory_cache: Add completion for 'reload' argument of + 'sorcery memory cache stale' and properly implement. + + Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca + +2017-02-10 15:45 +0000 [8b72ec312b] George Joseph + + * stream: Add media stream topology definition and API + + This change adds the media stream topology definition and API for + accessing and using it. + + Some refactoring of the stream was also done. + + ASTERISK-26786 + + Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568 + +2017-01-13 11:21 +0000 [75f8167e66] Norbert Varga + + * chan_pjsip: Multidomain endpoint finding on call + + When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com), + the user part is stripped down as it would be a trunk with a specified user, + and only the host part is called as a PJSIP endpoint and can't be found. + This is not correct in the case of a multidomain SIP account, so the stripping + after the @ sign is done only if the whole endpoint (in multidomain case + 1000@test.com) can't be found. + + ASTERISK-26248 + + Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6 + +2017-02-13 05:05 +0000 [89871576b9] Joshua Colp + + * channel: Protect flags in ast_waitfor_nandfds operation. + + The ast_waitfor_nandfds operation will manipulate the flags + of channels passed in. This was previously done without + the channel lock being held. This could result in incorrect + values existing for the flags if another thread manipulated + the flags at the same time. + + This change locks the channel during flag manipulation. + + ASTERISK-26788 + + Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed + +2017-02-11 11:25 +0000 [07abb39d6a] Richard Mudgett + + * res_pjsip.c: Fix inconsistency between warning and action. + + The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE + but we have no authenticator registered to create the challenge. + + Change-Id: I62368180d774b497411b80fbaabd0c80841f8512 + +2017-02-11 11:26 +0000 [ce810a892b] Richard Mudgett + + * pjsip_distributor.c: Fix off-nominal tdata ref leak. + + Change-Id: I571f371d0956a8039b197b4dbd8af6b18843598d + +2017-02-09 10:01 +0000 [0910773077] Sean Bright + + * manager: Restore Originate failure behavior from Asterisk 11 + + In Asterisk 11, if the 'Originate' AMI command failed to connect the provided + Channel while in extension mode, a 'failed' extension would be looked up and + run. This was, I believe, unintentionally removed in 51b6c49. This patch + restores that behavior. + + This also adds an enum for the various 'synchronous' modes in an attempt to + make them meaningful. + + ASTERISK-26115 #close + Reported by: Nasir Iqbal + + Change-Id: I8afbd06725e99610e02adb529137d4800c05345d + +2017-02-08 14:27 +0000 [16fdb11bc3] Richard Mudgett + + * core: Cleanup some channel snapshot staging anomalies. + + We shouldn't unlock the channel after starting a snapshot staging because + another thread may interfere and do its own snapshot staging. + + * app_dial.c:dial_exec_full() made hold the channel lock while setting up + the outgoing channel staging. Made hold the channel lock after the called + party answers while updating the caller channel staging. + + * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. + Also we need to use ast_hangup() instead of ast_channel_unref() at that + location. + + * channel.c:__ast_channel_alloc_ap() added a comment about not needing to + complete the channel snapshot staging on off-nominal exit paths. + + * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel + locks while staging the channels for the stats channel variables. + + Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a + +2017-02-07 06:56 +0000 [bab4885f1e] Joshua Colp + + * stream: Add media stream definition and API with unit tests. + + This change adds the media stream definition and API for + accessing and using it. Unit tests have also been written + which exercise aspects of the API. + + ASTERISK-26773 + + Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87 + +2017-02-10 09:35 +0000 [648d181d2f] George Joseph + + * configs/samples: Fix placement of 'identify' entry in sorcery.conf + + The entry for 'identify' was incorrectly placed in the + res_pjsip section when it should be in + res_pjsip_endpoint_identifier_ip. + + ASTERISK-26785 #close + + Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a + +2017-02-08 11:50 +0000 [46147a8f30] Mark Michelson + + * Revert "Update qualifies when AOR configuration changes." + + This reverts commit 6492e91392b8fd394193e411c6eb64b45486093f. + + The change in question was intended to prevent the need to reload in + order to update qualifies on contacts when an AOR changes. However, this + ended up causing a deadlock instead. + + Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e + +2017-02-07 12:01 +0000 [5422ec140c] Joshua Colp + + * srv: Fix crash when ast_srv_lookup is used and 0 records are returned. + + When performing an SRV lookup using the ast_srv_lookup function it + did not properly handle the situation where 0 records are returned. + If this happened it would wrongly assume that at least one record + was present. + + This change fixes the code so it will exit early if an error occurs + or if 0 records are returned. + + ASTERISK-26772 + patches: + srv_lookup.patch submitted by nappsoft (license 6822) + + Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351 + +2017-02-06 11:40 +0000 [b79cc62057] Joshua Colp + + * res_stasis_device_state: Protect the adding/removing of subscriptions. + + The adding and removing of device state subscriptions did not protect + fully against simultaneous manipulation. In particular the subscribe + case allowed a small window where two subscriptions could be added for + the same device state instead of just one. + + This change makes the code hold the subscriptions lock for the entirety + of each operation to ensure that two are not occurring at the same time. + + ASTERISK-26770 + + Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b + +2017-02-01 17:56 +0000 [b47cf1a7d6] Richard Mudgett + + * res_pjsip: Fix some off nominal tdata leaks. + + Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d + +2017-02-03 15:26 +0000 [7b280e7ccf] Sebastien Duthil + + * res_ari: fix memory leak for channelvars + + In ari.conf, when setting the option channelvars, every Stasis channel + snapshot would create a list of variable/value that would not be freed + when the snapshot is freed, resulting in a often-recurring memory + leak. + + ASTERISK-26767 #close + + Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d + +2017-02-03 02:25 +0000 [c6c7f17206] Tzafrir Cohen + + * libasteriskssl: do nothing with OpenSSL >= 1.1 + + OpenSSL 1.1 requires no explicit initialization. The hacks in the + library are not needed. They also happen to fail running Asterisk. + + Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100 + +2017-01-20 23:59 +0000 [bc041ca14a] Tzafrir Cohen + + * tcptls: use TLS_client_method with OpenSSL 1.1 + + OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous + version-specific methods (such as TLSv1_client_method(). Other than + being simpler to use and more correct (gain support for TLS newer that + TLS1, in our case), the older ones produce a deprecation warning that + fails the build in dev-mode. + + Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07 + +2017-01-20 23:57 +0000 [2c8d0764de] Tzafrir Cohen + + * openssl 1.1 support: use OPENSSL_VERSION_NUMBER + + Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect + the openssl 1.1 API. + + Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458 + +2017-01-31 18:28 +0000 [50029f585e] Richard Mudgett + + * channel.c: Fix unbalanced read queue deadlocking local channels. + + Using the timerfd timing module can cause channel freezing, lingering, or + deadlock issues. The problem is because this is the only timing module + that uses an associated alert-pipe. When the alert-pipe becomes + unbalanced with respect to the number of frames in the read queue bad + things can happen. If the alert-pipe has fewer alerts queued than the + read queue then nothing might wake up the thread to handle received frames + from the channel driver. For local channels this is the only way to wake + up the thread to handle received frames. Being unbalanced in the other + direction is less of an issue as it will cause unnecessary reads into the + channel driver. + + ASTERISK-26716 is an example of this deadlock which was indirectly fixed + by the change that found the need for this patch. + + * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue + did not add the same number of alerts to the alert-pipe. Correspondingly, + when there is an exceptionally long queue event, any removed frames did + not also remove the corresponding number of alerts from the alert-pipe. + + ASTERISK-26632 #close + + Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6 + +2017-01-31 16:38 +0000 [97c308471d] Richard Mudgett + + * res_agi: Prevent an AGI from eating frames it should not. (Re-do) + + A dialplan intercept routine is equivalent to an interrupt routine. As + such, the routine must be done quickly and you do not have access to the + media stream. These restrictions are necessary because the media stream + is the responsibility of some other code and interfering with or delaying + that processing is bad. A possible future dialplan processing + architecture change may allow the interception routine to run in a + different thread from the main thread handling the media and remove the + execution time restriction. + + * Made res_agi.c:run_agi() running an AGI in an interception routine run + in DeadAGI mode. No touchy channel frames. + + ASTERISK-25951 + + ASTERISK-26343 + + ASTERISK-26716 + + Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43 + +2017-01-31 16:32 +0000 [72e3fc5845] Richard Mudgett + + * Frame deferral: Revert API refactoring. + + There are several issues with deferring frames that are caused by the + refactoring. + + 1) The code deferring frames mishandles adding a deferred frame to the + deferred queue. As a result the deferred queue can only be one frame + long. + + 2) Deferrable frames can come directly from the channel driver as well as + the read queue. These frames need to be added to the deferred queue. + + 3) Whoever is deferring frames is really only doing the __ast_read() to + collect deferred frames and doesn't care about the returned frames except + to detect a hangup event. When frame deferral is completed we must make + the normal frame processing see the hangup as a frame anyway. As such, + there is no need to have varying hangup frame deferral methods. We also + need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. + That fake hangup is to cause the PBX thread to break out of loops to go + execute a new dialplan location. + + 4) To properly deal with deferrable frames from the channel driver as + pointed out by (2) above, means that it is possible to process a dialplan + interception routine while frames are deferred because of the + AST_CONTROL_READ_ACTION control frame. Deferring frames is not + implemented as a re-entrant operation so you could have the unsupported + case of two sections of code thinking they have control of the media + stream. + + A worse problem is because of the bad implementation of the AMI PlayDTMF + action. It can cause two threads to be deferring frames on the same + channel at the same time. (ASTERISK_25940) + + * Rather than fix all these problems simply revert the API refactoring as + there is going to be only autoservice and safe_sleep deferring frames + anyway. + + ASTERISK-26343 + + ASTERISK-26716 #close + + Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496 + +2017-02-02 11:26 +0000 [4c51ad158d] Sean Bright + + * res_odbc: Remove deprecated settings from sample configuration file + + ASTERISK-26704 #close + Reported by: Anthony Messina + + Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f + +2017-02-01 17:14 +0000 [7d9b50a7b2] Richard Mudgett + + * res_resolver_unbound.c: Fix frequent ref leak caught by excessive ref trap. + + ASTERISK-26765 + + Change-Id: I27eb97df7f8d7e624b0b9a61c0fcee4718c86d8d + +2017-02-01 15:56 +0000 [2849b726b6] Sean Bright + + * audiohooks: Muting a hook can mute underlying frames + + If an audiohook is placed on a channel that does not require transcoding, + muting that hook will cause the underlying frames to be muted as well. + + The original patch is from David Woolley but I have modified slightly. + + ASTERISK-21094 #close + Reported by: David Woolley + Patches: + ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded + by David Woolley + + Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed + +2017-02-01 13:54 +0000 [bbed75c3ba] Mark Michelson + + * Update qualifies when AOR configuration changes. + + Prior to this change, qualifies would only update in the following + cases: + * A reload of res_pjsip.so was issued. + * A dynamic contact was re-registered after its AOR's qualify_frequency + had been changed + This does not work well if you are using realtime for your AORs. You can + update your database to have a new qualify_frequency, but the permanent + contacts on that AOR will not have their qualifies updated. And the + dynamic contacts on that AOR will not have their qualifies updated until + the next registration, which could be a long time. + + This change seeks to fix this problem by making it so that whenever AOR + configuration is applied, the contacts pertaining to that AOR have their + qualifies updated. + + Additions from this patch: + * AOR sorcery objects now have an apply handler that calls into a newly + added function in the OPTIONS code. This causes all contacts + associated with that AOR to re-schedule qualifies. + * When it is time to qualify a contact, the OPTIONS code checks to see + if the AOR can still be retrieved. If not, then qualification is + canceled on the contact. + + Alterations from this patch: + * The registrar code no longer updates contact's qualify_frequence and + qualify_timeout. There is no point to this since those values already + get updated when the AOR changes. + * Reloading res_pjsip.so no longer calls the OPTIONS initialization + function. Reloading res_pjsip.so results in re-loading AORs, which + results in re-scheduling qualifies. + + Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121 + +2017-01-31 11:17 +0000 [aeea634bc0] Joshua Colp + + * res_pjsip: Handle invocation of callback on outgoing request when error occurs. + + There are some error cases in PJSIP when sending a request that will + result in the callback for the request being invoked. The code did not + handle this case and assumed on every error case that the callback was not + invoked. + + The code has been changed to check whether the callback has been invoked + and if so to absorb the error and treat it as a success. + + ASTERISK-26679 + ASTERISK-26699 + + Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91 + +2017-01-30 09:02 +0000 [7a16524a83] Sean Bright + + * res_rtp_asterisk: Swap byte-order when sending signed linear + + Before Asterisk 13, signed linear was converted into network byte order by a + smoother before being sent over the network. We restore this behavior by + forcing the creation of a smoother when slinear is in use and setting the + appropriate flags so that the byte order conversion is always done. + + ASTERISK-24858 #close + Reported-by: Frankie Chin + + Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9 + +2017-01-31 12:46 +0000 [e252aff9ad] George Joseph + + * debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts + + Forgot to install it with the original patch + + Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c + +2017-01-25 06:50 +0000 [ef4deb8ecd] George Joseph + + * debug_utilities: Add ast_logescalator + + The escalator works by creating a set of startup commands in cli.conf + that set up logger channels and issue the debug commands for the + subsystems specified. If asterisk is running when it is executed, + the same commands will be issued to the running instance. The original + cli.conf is saved before any changes are made and can be restored by + executing '$prog --reset'. + + The log output will be stored in... + $astlogdir/message.$uniqueid + $astlogdir/debug.$uniqueid + $astlogdir/dtmf.$uniqueid + $astlogdir/fax.$uniqueid + $astlogdir/security.$uniqueid + $astlogdir/pjsip_history.$uniqueid + $astlogdir/sip_history.$uniqueid + + Some minor tweaks were made to chan_sip, and res_pjsip_history + so their history output could be send to a log channel as packets + are captured. + + A minor tweak was also made to manager so events are output to verbose + when "manager set debug on" is issued. + + Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543 + +2017-01-23 09:35 +0000 [178b90af02] Torrey Searle + + * libastssl/pj: libastssl/pj should have an so_version + + Issue introduced in b59956a87. In the non-darwin case libastssl/pj + should be versioned. This causes the symbol file for this lib + to not be generated. + + Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c + (cherry picked from commit 54b027916a71f2b83b2050cef5ef704ea5de39b2) + +2017-01-24 19:51 +0000 [138cd8d019] Kirill Katsnelson + + * make_build_h: handle backslashes in external strings + + LikewiseOpen creates user names with a backslash in them. A gentle + massage with sed(1) allows such strings to be inserted into build.h + properly quoted. I am also adding the same for host name and other + strings used in the script that are more or less user-controlled. + + ASTERISK-26754 + + Change-Id: Iac5ef2b67a68ee58f35ddbf86bb818ba6eabecae + +2017-01-24 22:31 +0000 [8270d2436d] Kirill Katsnelson + + * app_queue: Fix queues randomly disappearing on reload + + With 500+ queues and a reload every minute, a random queue disappears + upon reload. The cause is mususe of the 'dead' flag. Namely, all queues + were marked dead up front, and then "resurrected" by dropping this flag + for those found in the configuration. But a queue marked dead can be + removed also when control leaves the app entry point on a PBX thread. + + With this change, the queue is marked only not found, and at the end of + reload only the queues that are still not found are actually marked as + dead, so the dead flag is never reset, and set only on positively dead + queues. + + ASTERISK-26755 + + Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf + +2017-01-26 07:57 +0000 [7fa3de7ae9] Joshua Colp + + * res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving. + + This change adds a missing unreference of the hostname when resolving and + also cleans up the iterator. + + ASTERISK-26735 + + Change-Id: Ic012ebaf3d89e714eec340b7b0c5e63c66af857a + +2017-01-25 15:26 +0000 [d32bd63860] Mark Michelson + + * Add reload options to CLI/AMI stale object commands. + + Marking an object as stale in a memory cache is supposed to prime the + cache so that the next time the item is retrieved, the stale item is + deleted from the cache and a background task is run to re-populate the + cache with a fresh version of the object. + + The problem is, there are some object types out there for which there is + no natural reason that they would be retrieved from the backend with any + regularity. Outbound PJSIP registrations are a good example of this. At + startup, they are read, and an object-specific state is created that + refers to the initially-retrieved object for all time. + + Adding the "reload" option to the CLI/AMI commands gives the cache the + opportunity to manually re-retrieve the object from the backend, both + storing the new object in the cache and applying the new object's + configuration to the module that uses that object. + + Change-Id: Ieb1fe7270ceed491f057ec5cbf0e097bde96c5c8 + +2017-01-10 17:39 +0000 [20aed30d9a] Richard Mudgett + + * T.140: Fix format ref and memory leaks. + + * channel.c:ast_sendtext(): Fix T.140 SendText memory leak. + + * format_compatibility.c: T.140 RED and T.140 were swapped. + + * res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak. + + * res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic + scheduled red_write(). + + * res_rtp_asterisk.c: Some other minor misc tweaks. + + Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb + +2017-01-24 15:39 +0000 [ee2b0f2eef] Joshua Colp + + * res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0. + + When configuring a match using a netmask the error variable was + not defaulting to 0. For some people this would cause the code + to think an error occurred when adding the match when in reality + it added perfectly fine. + + ASTERISK-26693 + + Change-Id: I850c250813742bddde65c84e739093c9e01dfe56 + +2017-01-10 17:37 +0000 [930a24a730] Richard Mudgett + + * astobj2.c: Add excessive ref count trap. + + Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a + +2017-01-10 13:11 +0000 [de28c1b9f1] Richard Mudgett + + * main/app.c: Memory corruption from early format destruction. + + * make_silence() created a malloced silence slin frame without adding a + slin format ref. When the frame is destroyed it will unref the slin + format that never had a ref added. Memory corruption is expected to + follow. + + * Simplified and fixed counting the number of samples in a frame list for + make_silence(). + + * Eliminated an unnecessary RAII_VAR associated with the make_silence() + frame. + + Change-Id: I47de3f9b92635b7f8b4d72309444d6c0aee6f747 + +2017-01-11 14:59 +0000 [2039eb8edf] Richard Mudgett + + * frame.c: Fix off-nominal format ref leaks. + + * ast_frisolate() could leak frame format refs on allocation + failures. + + * Similified code in ast_frisolate() and code used by + ast_frisolate(). + + Change-Id: I79566d4d36b3d7801bf0c8294fcd3e9a86a2ed6d + +2017-01-13 19:08 +0000 [e922979d49] Richard Mudgett + + * stasis_bridge.c: Fix off-nominal stasis control ref leak. + + Change-Id: Ib17218343a6596832060180e19386da9df150ac8 + +2017-01-10 12:30 +0000 [56854f22d2] Richard Mudgett + + * res_musiconhold.c: Fix format ref leak when parsing MOH config class. + + Change-Id: Ica8e8e2ce7604c2c61ec55bef07dc675361d2ea5 + +2017-01-10 14:03 +0000 [d87f81ddb1] Richard Mudgett + + * chan_oss.c: Fix format ref leak in oss_read(). + + Change-Id: I0a5d56c7dcf327d60f86a4c25a23571733709fd0 + +2017-01-10 17:48 +0000 [36bdd7c1a0] Richard Mudgett + + * Add notes about embedded ast_frame structs holding a format ref. + + mod_format.h: Note ast_filestream.fr holds a format ref. + + translate.h: Note ast_trans_pvt.f holds a format ref. + + Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749 + +2017-01-20 21:13 +0000 [6f3e8c8e01] Richard Mudgett + + * PJPROJECT logging: Fix detection of max supported log level. + + The mechanism used for detecting the maximum log level compiled into the + linked pjproject did not work. The API call simply stores the requested + level into an integer and does no range checking. Asterisk was assuming + that there was range checking and limited the new value to the allowable + range. To get the actual maximum log level compiled into the linked + pjproject we need to get and save off the initial set log level from + pjproject. This is the maximum log level supported. + + * Get and save off the initial log level setting before altering it to the + desired level on startup. This has to be done by a macro rather than + calling a core function to avoid incorrectly linking pjproject. + + * Split the initial log level warning messages to warn if the linked + pjproject cannot support the requested startup level and if it is too low + to get the pjproject buildopts for "pjproject show buildopts". + + * Adjust the CLI "pjproject set log level" to check the saved max log + level and to generate normal output messages instead of a warning message. + + ASTERISK-26743 #close + + Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4 + +2017-01-05 13:21 +0000 [0ea3c371c5] Richard Mudgett + + * res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands. + + ASTERISK-23828 #close + + Change-Id: Ifb8a3b61f447aedc58a8e6b36a810f7566018567 + +2017-01-23 16:18 +0000 [4bfeda6ee4] Mark Michelson + + * Free endpoint ACLs when destroying PJSIP endpoints. + + If endpoint ACLs were specified, they were not being freed + when endpoints were destroyed. On systems with realtime endpoints, this + could add up quickly since each DB lookup would allocate the ACL without + freeing it. + + ASTERISK-26731 #close + Reported by Ustinov Artem + + Change-Id: Ie1f8bf5b7a0de628c975beba01e69c56893331ad + +2017-01-19 09:05 +0000 [6691606723] George Joseph + + * ari: Implement 'debug all' and request/response logging + + The 'ari set debug' command has been enhanced to accept 'all' as an + application name. This allows dumping of all apps even if an app + hasn't registered yet. To accomplish this, a new global_debug global + variable was added to res/stasis/app.c and new APIs were added to + set and query the value. + + 'ari set debug' now displays requests and responses as well as events. + This required refactoring the existing debug code. + + * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} + to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. + * In order to print the body of incoming requests even if a request + failed, the consumption of the body was moved from the ari stubs + to ast_ari_callback in res_ari.c and the moustache templates were + then regenerated. The body is now passed to ast_ari_invoke and then + on to the handlers. This results in code savings since that template + was inserted multiple times into all the stubs. + + An additional change was made to the ao2_str_container implementation + to add partial key searching and a sort function. The existing cli + code assumed it was already there when it wasn't so the tab completion + was never working. + + Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf + (cherry picked from commit 1d890874f39a5a81b20da44358143ed9b54ab0fe) + +2017-01-20 23:41 +0000 [f3f9175df0] Tzafrir Cohen + + * test_voicemail_api: order of params to VERIFY macros + + Fix order of parameters in calls to VM_API_INT_VERIFY and + VM_API_STRING_VERIFY + + ASTERISK-26739 #close + + Change-Id: I30dc6b36893aadad6012be3f16f93aa5720870d6 + Note: status: builds. Not tested any further. + +2017-01-23 09:10 +0000 [96e7291cbd] George Joseph + + * pjproject_bundled: Fix setting max log level + + An earlier attempt to prevent pjsua from spitting out an extra 6795 + lines of debug output every time the testsuite called it was also + turning off the ability for asterisk to output debug info when it + needed to. This patch reverts the earlier fix and instead adds + a pjproject patch that sets the startup log level to 1 for pjsua + pjsystest and the pjsua python binding. This is an asterisk-only + patch that does not affect pjproject functionality and will not be + submitted upstream. + + Change-Id: I347a8b58b2626f2906ccfc1d339e907627a0c9e8 + +2017-01-23 10:08 +0000 [23690c1b35] Joshua Colp + + * res_pjsip_endpoint_identifier_ip: Read settings before resolving. + + An option has been added, srv_lookups, which controls whether + SRV lookups are performed on the provided match hosts or not. + It was possible for this option to be applied after resolution + had already happened. + + This change makes it so hosts are stored away, settings are read + and applied, and then resolution is done. This ensures that no + matter the ordering the srv_lookups option is in effect. + + ASTERISK-26735 + + Change-Id: I750378cb277be0140f8c5539450270afbfc43388 + +2016-11-29 09:31 +0000 [1061539b75] Lorenzo Miniero + + * media: Add experimental support for RTCP feedback. + + This change adds experimental support for providing RTCP + feedback information to codec modules so they can dynamically + change themselves based on conditions. + + ASTERISK-26584 + + Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857 + +2017-01-22 17:25 +0000 [cfe72c39cf] Richard Mudgett + + * LISTFILTER: Remove outdated ERROR message. + + Feeding LISTFILTER an empty variable results in an invalid ERROR message. + Earlier changes made the message useless because we can no longer tell if + the variable is empty or does not exist. It is valid to try to remove a + value from an empty list just as it is valid to try to remove a value that + is not in a non-empty list. + + * Removed the outdated ERROR message. + + * Added more test cases to the LISTFILTER unit test. + + Change-Id: Ided9040e6359c44a335ef54e02ef5950a1863134 + +2017-01-21 14:43 +0000 [dbb9c8141d] Tzafrir Cohen + + * tests: use datadir for sound files + + Some (voicemail-related) tests API symlinks beep.gsm and other files + from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR. + + ASTERISK-26740 #close + + Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89 + +2017-01-05 15:11 +0000 [ef9164b9ca] Richard Mudgett + + * res_pjsip_pubsub.c: Fix AMI event list counts. + + Fix the AMI PJSIPShowSubscriptionsInbound, PJSIPShowSubscriptionsOutbound, + and PJSIPShowResourceLists actions event counts. The reported counts may + not necessarily be accurate depending on what happens. + + The subscriptions count would be wrong if Asterisk ever has outbound + subscriptions. + + The resource list count could be wrong if a list were added or removed + during the AMI action being processed. + + Change-Id: I4344301827523fa174960a42c413fd19abe4aed5 + +2017-01-05 13:02 +0000 [ab858295a2] Richard Mudgett + + * res_pjsip_pubsub.c: Fix incorrect message string wrapping. + + Change-Id: Id771e6fe56d89ce365ddcbb423f820af97211120 + +2017-01-05 13:01 +0000 [6d648185bc] Richard Mudgett + + * res_pjsip_pubsub.c: Eliminate trivial SCOPED_LOCK usage. + + Change-Id: Ie0b69a830385452042fa19e7d267c6790ec6b6be + +2017-01-05 12:58 +0000 [90f3b1270c] Richard Mudgett + + * res_pjsip: alloca can never fail. + + Change-Id: Ia2a6158e5fdf311bc2a1c0c43417978de504b1f1 + +2017-01-13 11:03 +0000 [d16b3a9917] George Joseph + + * debug_utilities: Create ast_loggrabber + + ast_loggrabber gathers log files from customizable search patterns, + optionally converts POSIX timestamps to a readable format and + tarballs the results. + + Also a few tweaks were made to ast_coredumper. + + Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 + (cherry picked from commit c70915287837704090d75f181525765de7a17221) + +2017-01-01 03:47 +0000 [48730ae65e] Richard Mudgett + + * res_pjsip_outbound_authenticator_digest.c: Fix spacing in warning messages. + + Change-Id: I573f0343c0c63a785cd4da60d57cc9f8b9ce7f49 + +2016-12-22 04:07 +0000 [40b9766a31] Martin Tomec + + * app_queue: add RINGCANCELED log event on caller hang up + + QueueLog did not log ringnoanswer when the caller abandoned call + before first timeout. It was impossible to get agent membername + and ringing duration for this short calls. After some discusions + it seems that the best way is to add new event RINGCANCELED, + which is generated after caller hangup during ringing. + + ASTERISK-26665 + + Change-Id: Ic70f7b0f32fc95c9378e5bcf63865519014805d3 + +2017-01-12 15:58 +0000 [283c16c6b6] Kevin Harwell + + * abstract/fixed/adpative jitter buffer: disallow frame re-inserts + + It was possible for a frame to be re-inserted into a jitter buffer after it + had been removed from it. A case when this happened was if a frame was read + out of the jitterbuffer, passed to the translation core, and then multiple + frames were returned from said translation core. Upon multiple frames being + returned the first is passed on, but sebsequently "chained" frames are put + back into the read queue. Thus it was possible for a frame to go back into + the jitter buffer where this would cause problems. + + This patch adds a flag to frames that are inserted into the channel's read + queue after translation. The abstract jitter buffer code then checks for this + flag and ignores any frames marked as such. + + Change-Id: I276c44edc9dcff61e606242f71274265c7779587 + +2016-11-06 06:30 +0000 [8cc1cd5df7] Sebastian Gutierrez + + * app_queue: Add QueueUpdate application. + + Add an application that allows tracking outbound calls + using app_queue. + + ASTERISK-19862 + + Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e + +2017-01-13 21:23 +0000 [f4e77a5678] Richard Mudgett + + * taskprocessor.c: Change when high water warning logged. + + The task processor queue reached X scheduled tasks message was originally + intended to get logged only once per task processor to prevent spamming + the log. This is no longer necessary since high and low water thresholds + can better control when the message is logged. + + It is beneficial to generate the warning each time a task processor + reaches the high water level because PJSIP stops processing new requests + while any high water alert is active. Without this change you would have + to enable at least debug level 3 logging to know about a repeated alert + trigger. + + * Made generate the warning message whenever a task is pushed into the + task processor that triggers the high water alert. + + * Appended 'again' to the warning for a repeated high water alert trigger. + + Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999 + +2017-01-10 05:54 +0000 [e0e502d9d2] Aaron An + + * res_rtp_asterisk: Fix bug in function CHANNEL(rtcp, all_rtt) + + Function CHANNEL(rtcp,all_rtt) CHANNEL(rtcp,all_loss) CHANNEL(rtcp,all_jitter) + always return 0.0 due to wrong define of macro "AST_RTP_SATA_SET" and + "AST_RTP_STAT_STRCPY". + It should compare "combined" with "stat" not "current_stat". + + ASTERISK-26710 #close + Reported-by: Aaron An + Tested-by: AaronAn + + Change-Id: Id4140fafbf92e2db689dac5b17d9caa009028a15 + +2017-01-10 18:10 +0000 [0d53c91fba] George Joseph + + * debug_utilities: Create the ast_coredumper utility + + This utility allows easy manipulation of asterisk coredumps. + + * Configurable search paths and patterns for existing coredumps + * Can generate a consistent coredump from the running instance + * Can dump the lock_infos table from a coredump + * Dumps backtraces to separate files... + - thread apply 1 bt full -> .thread1.txt + - thread apply all bt -> .brief.txt + - thread apply all bt full -> .full.txt + - lock_infos table -> .locks.txt + * Can tarball corefiles and optionally delete them after processing + * Can tarball results files and optionally delete them after processing + * Converts ':' in coredump and results file names '-' to facilitate + uploading. Jira for instance, won't accept file names with colons + in them. + + Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. + + [1] For *BSDs, the "devel/gdb" package might have to be installed to + get a recent gdb. The utility will check all instances of gdb + it finds in $PATH and if one isn't found that can run python, it + prints a friendly error. + + Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd + (cherry picked from commit cb47b4556053cd50d9102eef913671ad0306062d) + +2017-01-08 10:29 +0000 [e54c8aec34] George Joseph + + * pjproject_bundled: Fix compilation with MALLOC_DEBUG + + When MALLOC_DEBUG was specified, make was failing. Immediately + remaking would work. The issues was in the ordering of the make + dependencies. + + Change-Id: If6030b54fc693f3179f32bfd20c6b5d5f1b3f7cd + +2017-01-05 06:11 +0000 [a7d856cd96] Joshua Colp + + * res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. + + This change implements SRV support for the IP based endpoint + identifier module. All possible addresses through SRV are looked + up and added as matches. If no SRV records are available a + fallback to normal host resolution is done. If an IP address + is provided then no SRV lookup occurs. + + This is configured using the "srv_lookups" option on the + identify section and defaults to "yes". + + ASTERISK-26693 + + Change-Id: I6b641e275bf96629320efa8b479737062aed82ac + +2016-11-06 06:37 +0000 [740ca862e4] Sebastian Gutierrez + + * app_queue: add new Service Level calculation + + Adds a new formula for SL2 and documentation + + ASTERISK-26559 + + Change-Id: I0970c620460507cd9d45b0d43600779c8915e770 + +2016-12-19 15:03 +0000 [d96e350256] Jonathan R. Rose + + * core/pbx: dialplan show - display filename/line# + + Adds the ability for extensions to be registered to include filename and + line number so that dialplan show output can show the filename and line + number of a config file responsible for generating a given extension. + + This only affects config modules that are written to use the new extension + registering functions. In this patch, that only includes pbx_config, so + extensions registered in extensions.conf and any included extension will + be shown in this manner. Extensions registered in this manner will show + the filename and line number *instead* of the registrar. + + ASTERISK-26658 #close + Reported by: Jonathan R. Rose + + Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30 + +2016-12-22 09:13 +0000 [aea2285865] Alexander Traud + + * res_pjsip_session: Access SIPDOMAIN via Dialplan. + + This feature was available in the SIP channel driver chan_sip. For example, + Asterisk is the outbound proxy and has to handle all SIP-URIs, even domains not + local to Asterisk. In that case, SIPDOMAIN is used in the Dialplan, to detect + and dial remote SIP-URIs. This change here sets the SIP destination domain of + an inbound call (SIPDOMAIN) in the SIP channel driver res_pjsip as well. + + ASTERISK-26670 #close + + Change-Id: I27c880dc404a3c1c6792e1ba3545475339577243 + +2017-01-04 05:50 +0000 [e220c11bec] Alexander Traud + + * chan_sip: Remember SDP negotiation on SIP_CODEC_INBOUND. + + After a SIP_CODEC_INBOUND in the dialplan, do not continue with cached formats + but remember the joint format. Cached formats contain default parameters, + often create an empty fmtp line. However, a joint format might have passed + format_get_joint(.) in a res_format_attr_* module (like Opus Codec) and + contain the resulting format parameters from a SDP negotiation. + + ASTERISK-26691 #close + + Change-Id: I35712d98a793d4c3efdd156cec57deab9014b1dc + +2017-01-03 15:14 +0000 [ceb9dae566] George Joseph + + * pjproject_bundled: Compile pjsua with max log level = 2 + + A while back, we changed config_site.h to set PJ_LOG_MAX_LEVEL = 6. + This allowed us to control the log level better from inside Asterisk. + An unfortunate side effect of this was that the pjsua binary and + python bindings were also compiled with log level set to 6 so whenever + a testsuite test that uses pjsua runs, it spits out 6795 lines of + debug in an instant even before the test starts. I believe this + overruns the Jenkins capture buffer and prevents the test from + properly terminating. In turn, this results in the testsuite just + hanging until the job is killed. It's more frequent on the higher + end agents because they can spit out the messages faster. + + Unfortunately, the messages are all spit out before we have control + of the python pj.Lib instance where we can set logging levels so the + only alternative was to actually compile pjsua and _pjsua.so with an + overridden PJ_LOG_MAX_LEVEL. Although defining a lower max level was + done in the Makefile, the define in config_site.h had to be wrapped + with "#ifndef" so the change would take effect. + + Change-Id: I2af9e7d48dde1927279c586c9c725d868fe6f3ff + +2016-12-22 16:00 +0000 [ae57652983] Joshua Colp + + * chan_pjsip: Use session for retrieving CHANNEL() information. + + The CHANNEL() dialplan function implementation for PJSIP allows + querying of PJSIP specific information. This used the channel + passed in to get the PJSIP session and associated information. + It is possible for this channel to be masqueraded and end + up as a different channel type by the time the information + request is actually acted upon. + + This change retrieves the PJSIP session safely and accesses + data from it (including channel). This provides a guarantee + that the session and channel will not be altered when the + request is being acted upon. + + ASTERISK-26673 + + Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6 + +2016-12-31 19:56 +0000 [386e3a01b3] Joshua Elson + + * res_pjsip: Fix known compact header issues + + ASTERISK-26684 #close + + Change-Id: Ifd7e401c45015119dd5e8421dbfe3afa6381744a + +2016-12-30 06:59 +0000 [aad29b9bca] Martin Tomec + + * res_calendar: delete old calendars after reload + + When "fetch_again_at_reload" is set in config, we create now + new object and thread for each reloaded calendar (with new + configuration). Old calendar should be then unlinked, so the + old thread can exit and free memory. + + ASTERISK-26683 + + Change-Id: Ic17fba9371c5a8b26a6bc54ea4957c13a32a343e + +2016-12-30 09:10 +0000 [5a5953f98c] George Joseph + + * res_pjsip_refer: Handle compact Refer-To header. + + refer_incoming_refer_request needed to look for the "r" header as well + as the "Refer-To" header. + + ASTERISK-26655 #close + patches: + refer_compact_fix.diff submitted by JoshE (license 6075) + + Change-Id: I610410a99b02427ea5db887aeb454d5f12c2259f + +2016-12-23 12:11 +0000 [ac04e63ac2] Richard Mudgett + + * bridge_native_rtp.c: Minor code cleanups. + + In native_rtp_bridge_compatible_check() + + * Made one variable declaration per line. + + * Extracted if test assignment to make the test easier to see. + + * Made long if tests easier to see the combinatorial logic. + + * Added bridge id to a couple debug messages. + + Change-Id: I65bc5732aa7c9a2537f062f106fbea711cf2daad + +2016-12-23 12:10 +0000 [da6f40c9ff] Richard Mudgett + + * bridge_native_rtp.c: Fix native rtp bridge data race. + + native_rtp_bridge_compatible() didn't lock the bridge channels before + checking the channels for native bridging ability. As a result, one of + the channel's native format capabilities structure got replaced out from + under the native bridge check. Use of a stale pointer to freed memory + causes bad things to happen. + + MALLOC_DEBUG, DO_CRASH, and the + tests/channels/pjsip/transfers/blind_transfer/caller_direct_media + testsuite test caught this. + + * Add missing channel locking in native_rtp_bridge_compatible(). + + Change-Id: If25fdb3ac8e85563c4857fb8216b3d9dc3d0fa53 + +2016-12-21 16:28 +0000 [b576b58d74] Richard Mudgett + + * res_rtp_asterisk.c: Fix uninitialized memory crash. + + ast_rtp_remote_address_set() could pass an uninitialized 'us' parameter to + ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' + parameter may not get initialized. Thus when the code tries to save the + 'us' parameter to the local address we could try to copy a ridiculous + sized memory buffer and segfault. + + * Made pass an initialized 'us' parameter to ast_ouraddrfor(). + + * Optimized out the 'us' struct variable. + + ASTERISK-26672 #close + + Change-Id: I4acea5dcdf0813da2c7d3e11c2d6067d160d17dc + +2016-12-21 16:25 +0000 [67cc8499a2] Richard Mudgett + + * acl.c: Improve ast_ouraddrfor() diagnostic messages. + + * Made not generate strings unless they will actually be used. + + ASTERISK-26672 + + Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3 + +2016-12-21 17:54 +0000 [67b47191e9] Richard Mudgett + + * chan_rtp.c: Fix uninitialized memory crash. + + unicast_rtp_request() could pass an uninitialized 'us' parameter to + ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' + parameter may not get initialized. Thus when the code tries to save the + 'us' parameter to the local address we could try to copy a ridiculous + sized memory buffer and segfault. + + * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort + the UnicastRTP channel request if it fails. + + ASTERISK-26672 + + Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0 + +2016-12-21 17:55 +0000 [2fc65173e5] Richard Mudgett + + * res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip(). + + We access uninitialized memory when the 'ourip' parameter does not + have an initial guess to our IP address. + + ASTERISK-26672 + + Change-Id: I35507ea1ad7455d2be188f6ccdd4add7bd150e15 + +2016-12-07 15:23 +0000 [8b7d252987] Richard Mudgett + + * res_rtp_asterisk.c: Fix off nominal memory leak. + + Change-Id: I95b1088d11244a2edae6607c12fbf33b38658a75 + +2016-12-14 02:21 +0000 [bab253ac9f] Tzafrir Cohen + + * Fixes to various issues reported by pyflakes + + Pyflake is a python (2) source checker. This patch fixes various + (mostly trivial) errors and warnings it reports. + + Change-Id: Ia35c5ac61751b927814cf693994c632c412386ea + +2016-12-09 12:23 +0000 [f461f65dea] Martin Tomec + + * app_queue: Ensure member is removed from pending when hanging up. + + In some cases member is added to pending_members, and the channel + is hung up before any extension state change. So the member would + stay in pending_members forever. So when we call do_hang, we + should also remove member from pending. + + ASTERISK-26621 #close + + Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54 + +2016-12-18 15:23 +0000 [d29eb3b99d] George Joseph + + * pjproject_bundled: Make build single threaded + + There were just too many issues in various environments with + multi threaded building of pjproject. It doesn't really speed + things up anyway since asterisk is already being compiled in + parallel. + + Change-Id: Ie5648fb91bb89b4224b6bf43a0daa1af793c4ce1 + +2016-12-08 20:00 +0000 [8fbb384ea2] Corey Farrell + + * chan_sip: Reorder unload_module to deal with stuck TCP threads. + + In some situations TCP threads may become frozen. This creates the + possibility that Asterisk could segfault if they become unfrozen after + chan_sip has been dlclose'd. This reorders the unload_module process to + allow abort if threads do not exit within 5 seconds. + + High level order as follows: + 1) Unregister from the core to stop new requests. + 2) Signal threads to stop + 3) Clear config based tables (but do not free the table itself). + 4) Verify that threads have shutdown, cancel unload if not. + 5) Clean all remaining resources. + + ASTERISK-26586 + + Change-Id: Ie23692041d838fbd35ece61868f4c640960ff882 + +2016-12-16 01:32 +0000 [147b8e636e] David M. Lee + + * configure: fix with-pjproject-bundled + + The AC_ARG_WITH macro's shell variable is withval; not enableval. Purely + coincidentally, the option would work when --enable-dev-mode is given. + + Also fixed a portability problem with bootstrap.sh, since -printf is not + a portable option for find. + + Change-Id: I0f0e5b1a934b5af5737713834361e9c95b96b376 + +2016-12-15 13:25 +0000 [d27dee3cca] Richard Mudgett + + * autosupport: Add 'pjproject show buildopts' + + Change-Id: I8aa55a7c3fb175235ddc7f85e9457d5102d06fa7 + +2016-12-14 14:21 +0000 [9404efa6f4] Richard Mudgett + + * chan_dahdi.c: Fix bounds check regression. + + Caused by ASTERISK-25494 + + Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb + +2016-12-13 14:34 +0000 [45a5e2abc6] Richard Mudgett + + * res_pjsip: Add/update ERROR msg if invalid URI. + + ASTERISK-24499 + + Change-Id: Ie305153e47e922233b2ff24715e0e326e5fa3a6c + +2016-12-12 18:38 +0000 [44e72c9d44] Richard Mudgett + + * MESSAGE: Flush Message/ast_msg_queue channel alert pipe. + + ASTERISK-25083 + + Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2 + +2016-12-13 14:06 +0000 [19328de2ab] George Joseph + + * res_sorcery_memory_cache: Change an error to a debug message + + When a sorcery user calls ast_sorcery_delete on an object that + may have already expired from the cache, res_sorcery_memory_cache + spits out an ERROR. Since this can happen frequently and validly when + an inbound registration expires after the cache entry expired, the + errors are unnecessary and misleading. Changed to a debug/1. + + Change-Id: Idf3a67038c16e3da814cf612ff4d6d18ad29ecd7 + +2016-12-09 08:14 +0000 [31268e0a28] George Joseph + + * pjproject_bundled: Retry download if previously saved tarball is bad + + If a tarball is corrupted during download, the makefile will attempt to + download it again. If the tarball somehow gets corrupted after it's + downloaded however, the makefile was just failing. We now + retry the download. + + ASTERISK-26653 #close + + Change-Id: I1b24d454852d80186f60c5a65dc4624ea8a1c359 + +2016-12-08 12:43 +0000 [4c6ba1dbba] Badalyan Vyacheslav + + * Fix typo in chan_sip + + The conditional expressions of the 'if' operators + situated alongside each other are identical. + + Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb + +2016-12-08 12:30 +0000 [934aa2c768] Badalyan Vyacheslav + + * res_pjsip: Fix 'A = B != C' kind. + + Consider reviewing the expression of the 'A = B != C' kind. + The expression is calculated as following: 'A = (B != C)' + + Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d + +2016-12-08 12:54 +0000 [51118e7d70] Badalyan Vyacheslav + + * chan_sip: Delete unneeded check + + P is always true. We check it before + + Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb + +2016-12-08 12:58 +0000 [fe5be81821] Badalyan Vyacheslav + + * Small code cleanup in chan_sip + + The conditional expressions of the 'if' operators situated + alongside each other are identical. + + Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a + +2016-12-08 12:34 +0000 [149d8db96c] Badalyan Vyacheslav + + * Fix IO conversion bug + + Expression 'rlen < 0' is always false. + Unsigned type value is never < 0. + + Change-Id: Id9f393ff25b009a6c4a6e40b95f561a9369e4585 + +2016-11-30 09:31 +0000 [c796f00c35] Walter Doekes + + * chan_sip: Do not allow non-SP/HTAB between header key and colon. + + RFC says SIP headers look like: + + HCOLON = *( SP / HTAB ) ":" SWS + SWS = [LWS] ; sep whitespace + LWS = [*WSP CRLF] 1*WSP ; linear whitespace + WSP = SP / HTAB ; from rfc2234 + + chan_sip implemented this: + + HCOLON = *( LOWCTL / SP ) ":" SWS + LOWCTL = %x00-1F ; CTL without DEL + + This discrepancy meant that SIP proxies in front of Asterisk with + chan_sip could pass on unknown headers with \x00-\x1F in them, which + would be treated by Asterisk as a different (known) header. For + example, the "To\x01:" header would gladly be forwarded by some proxies + as irrelevant, but chan_sip would treat it as the relevant "To:" header. + + Those relying on a SIP proxy to scrub certain headers could mistakenly + get unexpected and unvalidated data fed to Asterisk. + + This change fixes so chan_sip only considers SP/HTAB as valid tokens + before the colon, making it agree on the headers with other speakers of + SIP. + + ASTERISK-26433 #close + AST-2016-009 + + Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b + +2016-11-14 18:18 +0000 [5c89604a32] Joshua Colp + + * res_format_attr_opus: Fix crash when fmtp contains spaces. + + When an opus offer or answer was received that contained an + fmtp line with spaces between the attributes the module would + fail to properly parse it and crash due to recursion. + + This change makes the module handle the space properly and + also removes the recursion requirement. + + ASTERISK-26579 + + Change-Id: I01f53e5d9fa9f1925a7365f8d25071b5b3ac2dc3 + +2016-12-06 14:54 +0000 [79b09b5f18] George Joseph + + * res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command + + The PJSIPShowRegistrationsInbound AMI command was just dumping out + all AORs which was pretty useless and resource heavy since it had + to get all endpoints, then all aors for each endpoint, then all + contacts for each aor. + + PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail + events which meets the intended purpose of the other command and has + significantly less overhead. Also, some additional fields that were + added to Contact since the original creation of the ContactStatusDetail + event have been added to the end of the event. + + For compatibility purposes, PJSIPShowRegistrationsInbound is left + intact. + + ASTERISK-26644 #close + + Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a + +2016-12-07 14:22 +0000 [3b6e6cd01c] snuffy + + * tests_dns: Make DNS tests older nameser.h compatible + + Fix the tests for DNS to use older style nameser.h as + in ASTERISK-26608. + + Tested on: OpenBSD 6.0, Debian 8 + + ASTERISK-26647 #close + + Change-Id: I285913c44202537c04b3ed09c015efa6e5f9052d + +2016-12-06 16:45 +0000 [76d52dc228] Richard Mudgett + + * Bundled pjproject: Fix finding SIP transactions. + + Occasionally SIP message transactions are not found when they should be. + In the particular case an incoming INVITE transaction is CANCELed but the + INVITE transaction cannot be found so a 481 response is returned for the + CANCEL. The problematic calls have a '_' character in the Via branch + parameter. + + The problem is in the pjproject PJ_HASH_USE_OWN_TOLOWER feature's code. + The problem with the "own tolower" code is that it does not calculate the + same hash value as when the pj_tolower() function is used. The "own + tolower" code will erroneously modify the ASCII characters '@', '[', '\\', + ']', '^', and '_'. Calls to pj_hash_calc_tolower() can use the + PJ_HASH_USE_OWN_TOLOWER substitute algorithm when enabled. Calls to + pj_hash_get_lower(), pj_hash_set_lower(), and pj_hash_set_np_lower() call + find_entry() which never uses the PJ_HASH_USE_OWN_TOLOWER algorithm. As a + result you may not be able to find a hash tabled entry because the + calculated hash values would differ. + + * Simply disable PJ_HASH_USE_OWN_TOLOWER. + + ASTERISK-26490 #close + + Change-Id: If89bfdb5f301b8b685881a9a2a6e0c3c5af32253 + +2016-12-01 16:49 +0000 [503006123a] Mark Michelson + + * http: Send headers and body in one write. + + This is a semi-regression caused by the iostreams change. Prior to + iostreams, HTTP headers were written to a FILE handle using fprintf. + Then the body was written using a call to fwrite(). Because of internal + buffering, the result was that the HTTP headers and body would be sent + out in a single write to the socket. + + With the change to iostreams, the HTTP headers are written using + ast_iostream_printf(), which under the hood calls write(). The HTTP body + calls ast_iostream_write(), which also calls write() under the hood. + This results in two separate writes to the socket. + + Most HTTP client libraries out there will handle this change just fine. + However, a few of our testsuite tests started failing because of the + change. As a result, in order to reduce frustration for users, this + change alters the HTTP code to write the headers and body in a single + write operation. + + ASTERISK-26629 #close + Reported by Joshua Colp + + Change-Id: Idc2d2fb3d9b3db14b8631a1e302244fa18b0e518 + +2016-12-06 10:56 +0000 [bf6423a336] Mark Michelson + + * Iostreams: Correct off-by-one error. + + ast_iostream_printf() attempts first to use a fixed-size buffer to + perform its printf-like operation. If the fixed-size buffer is too + small, then a heap allocation is used instead. The heap allocation in + this case was exactly the length of the string to print. The issue here + is that the ensuing call to vsnprintf() will print a NULL byte in the + final space of the string. This meant that the final character was being + chopped off the string and replaced with a NULL byte. For HTTP in + particular, this caused problems because HTTP publishes the expected + Contact-Length. This meant HTTP was publishing a length one character + larger than what was actually present in the message. + + This patch corrects the issue by adding one to the allocation length. + + ASTERISK-26629 + Reported by Joshua Colp + + Change-Id: Ib3c5f41e96833d0415cf000656ac368168add639 + +2016-12-06 12:06 +0000 [fe9f070885] George Joseph + + * pjproject_bundled: Fix missing inclusion of symbols + + Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to + the CFLAGS. Not sure how they went missing. + + Also fixed an uninstall problem where we weren't removing the + symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was + there, I fixed it for libasteriskssl as well. + + Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556 + +2016-11-30 18:25 +0000 [4b3d3fc741] Richard Mudgett + + * res_pjsip_outbound_registration.c: Filter redundant statsd reporting. + + Increasing the testsuite shutdown timeout before forcibly killing + Asterisk allowed more events to be sent out. Some tests failed as + a result. The tests/channels/pjsip/statsd/registrations failed + because we now get the statsd events that a comment in the test + configuration stated couldn't be intercepted. Unfortunately, we + get a variable number of events because of internal status state + transition races generating redundant statsd events. + + We were reporting redundant statsd PJSIP.registrations.state changes + for internal state changes that equated to the same thing publicly. + + * Made update_client_state_status() filter out redundant statsd + updates. + + ASTERISK-26527 + + Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646 + +2016-06-28 16:26 +0000 [26c8552fff] Tzafrir Cohen + + * OpenSSL 1.1.0 support + + OpenSSL 1.1.0 includes some major changes in the interface. See + https://wiki.openssl.org/index.php/1.1_API_Changes . + + Status: Right now there are still a few deprecation notes with OpenSSL + 1.1.0. But it's a start. + + Changes: + * CRYPTO_LOCK is no longer available. Replace it with its value for now. + I don't completely understand what it is used for there. + * Remove several functions from libasteriskssl that seem to no longer be + needed. + * Structures have become opaque and are accesses with accessors. + * ERR_remove_thread_state() no longer needed. + * SSLv2 code now could no longer be used in 1.1. + + ASTERISK-26109 #close + + Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b + +2016-11-22 11:20 +0000 [75230f4c01] Guido Falsi + + * res_rtp: Fix regression when IPv6 is not available. + + The latest Release candidate fails to create RTP streams when IPv6 + is not available. Due to the changes made in September the ast_sockaddr + structure passed around to create these streams is always of AF_INET6 + type, causing failure when used for IPv4. This patch adds a utility + function to check for availability of IPv6 and applies such check + at startup to determine how to create the ast_sockaddr structures. + + ASTERISK-26617 #close + + Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e + +2016-11-23 18:27 +0000 [1dfa11b65c] Richard Mudgett + + * PJPROJECT logging: Made easier to get available logging levels. + + Use of the new logging is as simple as issuing the new CLI command or + setting the new pjproject.conf option. + + Other options that can affect the logging are how you have the pjproject + log levels mapped to Asterisk log types in pjproject.conf and if you have + configured Asterisk to log the DEBUG type messages. Altering the + pjproject.conf level mapping shouldn't be necessary for most installations + as the default mapping is sensible. Configuring Asterisk to log the DEBUG + message type is standard practice for collecting debug information. + + * Added CLI "pjproject set log level" command to dynamically adjust the + maximum pjproject log message level. + + * Added CLI "pjproject show log level" command to see the currently set + maximum pjproject log message level. + + * Added pjproject.conf startup section "log_level" option to set the + initial maximum pjproject log message level so all messages could be + captured from initialization. + + * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into + bundled pjproject. Pjproject will use the currently set run time log + level to determine if a log message is generated just like Asterisk + verbose and debug logging levels. + + * In log_forwarder(), made always log enabled and mapped pjproject log + messages. DEBUG mapped log messages are no longer gated by the current + Asterisk debug logging level. + + * Removed RAII_VAR() from res_pjproject.c:get_log_level(). + + ASTERISK-26630 #close + + Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389 + +2016-11-30 10:48 +0000 [621d886ca7] Mark Michelson + + * Frame deferral: Re-queue deferred frames one-at-a-time. + + The recent change that made frame deferral into an API had a behavior + change to it. When frame deferral was completed, we would take all of + the deferred frames and queue them all onto the channel in one call to + ast_queue_frame_head(). Before frame deferral was API-ized, places that + performed manual frame deferral would actually take each deferred frame + and queue them onto the channel. + + This change in behavior caused the confbridge_recording test to start + failing consistently. Without going too crazily deep into the details, + a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect + was attempting to break it out of the sleep, but because there were more + frames in the channel read queue than expected, the channel ended up + being unable to break from its sleep loop. + + By restoring the behavior of individual frame queuing after deferral, + the test starts passing again. + + Note, this points to a potential underlying issue pointing to an + "unbalance" that can occur when queuing multiple frames at once, + and so a follow-up issue is being created to investigate that + possibility. + + Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d + +2016-11-15 15:01 +0000 [e5e887be53] Alexei Gradinari + + * chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no + + The sending codec is switched to the receiving codec and then + is switched back to the best native codec on EVERY receiving RTP packets. + This is because after call of ast_channel_set_rawwriteformat there is call + of ast_set_write_format which calls set_format which sets rawwriteformat + to the best native format. + + This patch adds a new function ast_set_write_format_path which set + specific write path on channel and uses this function to switch + the sending codec. + + ASTERISK-26603 #close + + Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d + +2016-11-21 15:43 +0000 [ddc951060a] David Kerr + + * app_originate: Add option to execute gosub prior to dial + + Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 + that requested ability to add callerid into app_originate. + Comments in that issue suggested that it was better solved by + adding an option to gosub prior to originating the call. The + attached patch implements this much like app_dial with two + options one to gosub on the originating channel and one to gosub + on the newly created channel and behaves just like app_dial. + I have tested this patch by adding callerid info to the new + channel and also SIPAddHeader (to e.g. add header to force auto + answer) and confirmed it works. Have also tested both 'exten' + and 'app' versions of app_originate. + + Opened by: dkerr + Patch by: dkerr + + Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57 + +2016-11-28 19:43 +0000 [0e214c4932] Eduardo S. Libardi + + * res_calendar_caldav: Add support reading gmail calendar + + The response from gmail calendar includes the string name + "caldav:calendar-data". res_calendar_caldav implements + the example included in RFC 4791: string "C:calendar-data". + When reading the calendar, res_calendar_caldav compare the + string and if does not match just discards the event. + This commit compares the response to both strings, + successfully loading gmail calendar events. + Writing to gmail calendar is working prior to this fix. + + ASTERISK-26624 + Reported by: Eduardo S. Libardi + + Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a + +2016-11-28 15:12 +0000 [a3f48be0da] Matt Jordan + + * res/res_pjsip: Fix documentation whitespace issues + + Tabs > Spaces. + + Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0 + +2016-11-22 10:27 +0000 [0e15760795] Matt Jordan + + * res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter + + Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise + 'ws' when WebSockets are to be used as the transport. This applies to + both secure and insecure WebSockets. + + There were two bugs in Asterisk with respect to this: + + (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for + insecure websockets and 'wss' for secure websockets. While this + would seem to make sense - since 'WS' and 'WSS' are used for the Via + Transport parameter - this is not the case for the SIP URI. This + patch corrects that by registering the secure websockets with + pjproject using the shorthand 'WS', and by returning 'ws' when asked + for the transport parameter. Note that in pjproject, it is perfectly + valid to have multiple transports use the same shorthand. + + (2) In chan_sip, we return an upper-case version of the transport 'WS' + instead of 'ws'. Since we should be strict in what we send and + liberal in what we accept (within reason), this patch lower-cases + the transport before appending it to the parameter. + + ASTERISK-24330 #close + Reported by: cervajs, Inaki Baz Castillo + + Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42 + +2016-11-28 11:03 +0000 [8a68289766] George Joseph + + * build_tools: Fix download_externals to handle certified branches + + download_externals wasn't handling the "certified/13.x" version + correctly. + + Change-Id: I124d195bb117ca36fd7bf1150c630f3b474a9d9a + +2016-11-28 07:36 +0000 [e3dae763ee] Joshua Colp + + * iostream: Move include of asterisk.h + + The asterisk.h header file needs to be included first or else + some things go awry, such as: + + implicit declaration of function 'vasprintf' + + Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c + +2016-11-26 10:57 +0000 [0b588778c0] Michael Kuron + + * chan_sip: Fix segfault during module unload + + If a TCP/TLS connection was pending (not accepted and not timed out) during + unload of chan_sip, Asterisk would segfault when trying to send a signal to + a thread whose thread ID hadn't been recorded yet. This commit fixes that by + recording the thread ID before calling the blocking connect() syscall. + This was a regression introduced by 776a14386a55b5425c7e9617eff8af8b45427144. + + The above wasn't enough to fix the segfault, which was now delayed to the + point where connect() timed out. Therefore, it was necessary to also remove + the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be + used to interruput the connect() syscall. + This was a regression introduced by 5d313f51b982a18f7321adcf7c7a4e822d8b2714. + + ASTERISK-26586 #close + + Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b + +2016-11-23 14:52 +0000 [ead773f801] Dennis Guse + + * pbx_lua: On configuration errors report module load failure instead of decline. + + Switched from AST_MODULE_LOAD_DECLINE to AST_MODULE_LOAD_FAILURE. + Therefore, if pbx_lua fails to load and pbx_lua is marked as required, + Asterisk exits as expected. + If extensions.lua cannot be opened, AST_MODULE_LOAD_DECLINE is reported. + + Change-Id: I8e5a0037e69b41743db60c568541ebb2f52a7a8f + +2016-11-11 08:16 +0000 [d9b24cce0a] gestoip2 + + * res_rtp_asterisk: RTT miscalculation in RTCP + + When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. + RTT calculation is correct, but the data representation isn't. RTT is + represented by a 32-bit fixed-point number with the integer part in the + first 16 bits and the fractional part in the last 16 bits. In order to + get the RTT value, the fractional part is miscalculated, there is an + unnecessary 16 bit shift that causes overflow. Besides this there is + another mistake, when transforming the integer value to the fixed point + fractional part via bitwise operation, that loses precision. + + * RTT fractional part is no longer shifted, avoiding overflow. + + * RTT fractional part is transformed to its fixed-point value more + precisely. + + * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. + + * Fixed NTP timestamp report logging. The usec was inexplicably + multiplied by 4096. + + ASTERISK-26566 #close + Reported by Hector Royo Concepcion + + Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f + +2016-11-15 13:44 +0000 [635b0a0a55] Michael Kuron + + * tcptls: Use new certificate upon sip reload + + Previously, a TLS server socket would only be restarted upon sip reload if the + bind address had changed. This commit adds checking for changes to TLS + parameters like certificate, ciphers, etc. so they get picked up without + requiring a reload of the entire chan_sip module. This does not affect open + connections in any way, but new connections will use the new TLS parameters. + The changes also apply to HTTP and Manager. + + ASTERISK-26604 #close + + Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6 + +2016-11-21 09:49 +0000 [abae3dc36e] George Joseph + + * pjproject_bundled: Use $(LIB_RT) for link of libasteriskpj + + libasteriskpj was hard coded to use -lrt but librt is linux specific + so we now use the LIB_RT variable which gets set by configure. + + Change-Id: I41148884517e3031f7675a413d524c86e8614694 + +2016-11-19 16:19 +0000 [b546497fe0] snuffy + + * Add support for older name resolving version libraries like openBSD + + Fix support of OS's like openBSD that use an older nameser.h, + this change reverts the defines to the older style which on other + systems is found in nameser_compat.h + + Tested on openBSD 6.0, Debian 8 + + ASTERISK-26608 #close + + Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a + +2016-11-18 09:46 +0000 [7a8d6bc81b] Mark Michelson + + * Bump ARI version to 2.0.0 + + In order to not have version number overlap between different versions + of Asterisk, each new major version of Asterisk will mean we also bump + the ARI major version number. + + This particular change does NOT introduce any known breaking changes to + ARI. + + For discussion relating to this topice, see: + http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html + + Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665 + +2016-11-16 12:05 +0000 [d3f070c7a2] George Joseph + + * pjproject_bundled: Improve reliability of pjproject download + + The download process now has a timeout which will cause wget to retry + if it stops retrieving data for 5 seconds and fetch and curl to timeout + if the whole retrieval take smore than 30 seconds. + + If the tarball retrieval works, the MD5SUM file is retrieved from + the downloads site and the md5 checksum is verified. + + If either the tarball retrieval or MD5SUM retrieval fails, or the + checksums don't match, the entire process is retried once. If it + fails again, any incomplete tarball is deleted. + + .DELETE_ON_ERROR: was also added to the Makefile. Not only does + this delete the tarball on failure, it till also delete corrupted + library files from the pjproject source directory should they + fail to build correctly. + + Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and + Ubuntu 14. + + Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1 + +2016-11-11 07:13 +0000 [e822a50f86] Mikheili Dautashvili + + * main/app.c: Transmit Silence on ControlPlayback pause + + ASTERISK-26562 #close + + Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8 + +2016-11-17 10:52 +0000 [d670ea6297] Mark Michelson + + * manager: update minor version + + Based on bridge video AMI event changes, bump the minor version of AMI. + + Change-Id: Idf84507354170400813cda780906c94c9f1b60b4 + +2016-11-17 08:25 +0000 [349e08cb48] Timo Teräs + + * codec_dahdi: Fix poll.h include. + + POSIX defines poll.h. sys/poll.h should not be used as it is c-library + internal header which may or may not exist. Notably in musl including + sys/poll.h generates warning of being incorrect. + + Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252 + +2016-11-16 20:24 +0000 [935f5d003b] George Joseph + + * build: Various OpenBSD issues + + OpenBSD's 'find' doesn't take the -delete argument so you have to pipe + through 'xargs rm -rf'. + + 'echo -e' doesn't like \t starting a line. It just prints 't' which + causes the libasteriskpj.exports file to be garbage. They were just + cosmetic so they were removed. + + librt doesn't exist so the link of libasteriskpj.so fails. It's not + actually needed for linux anyway so -lrt was removed from the link. + + res_rtp_asterisk was failing to load because of an undefined + DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if + so DTLSv1_method is used instead. + + ASTERISK-26608 + + Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c + +2016-11-16 15:42 +0000 [dc8f99ee27] Mark Michelson + + * res_format_attr_opus: Fix fmtp generation. + + res_format_attr_opus assumed that the string being passed into it was + empty. It tried to determine if the only thing it had written was + + a=fmtp: + + And if it had, it would reset the string. Its calculation was off when + working with chan_sip, though. chan_sip passes the entire built SDP + rather than an empty string. This resulted in always putting an empty + fmtp line in the SDP. + + ASTERISK-26520 #close + Reported by scgm11 + + Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5 + +2016-11-15 16:23 +0000 [ed9ced0531] Richard Mudgett + + * codec_opus: Fix warning when Opus negotiated but codec_opus not loaded. + + When Opus is negotiated but not loaded, the log is spammed with messages + because the system does not know how to calculate the number of samples in + a frame. + + * Suppress the warning by supplying a function that assumes 20ms of + samples in the frame. For pass through support it doesn't really seem to + matter what number of samples is returned anyway. + + ASTERISK-26605 #close + + Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f + +2016-11-14 14:36 +0000 [0cd0e70c16] Richard Mudgett + + * res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak. + + Responding to authentication challenges leaks PJSIP memory pools. + + The leak was introduced with a pjproject 2.5.5 API change. + https://trac.pjsip.org/repos/ticket/1929 changed the API usage of + pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to + clean up cached authentication allocations that get allocated with + pjsip_auth_clt_reinit_req(). + + ASTERISK-26516 #close + + Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8 + +2016-11-15 12:01 +0000 [3017f09f22] George Joseph + + * file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type + + One of the code paths in __ast_file_read_dirs will only get executed if + the OS doesn't support dirent->d_type OR if the filesystem the + particular file is on doesn't support it. So, while standard Linux + systems support the field, some filesystems like XFS do not. In this + case, we need to call stat() to determine whether the directory entry + is a file or directory so we append the filename to the supplied + directory path and call stat. We forgot to truncate path back to just + the directory afterwards though so we were passing a complete file name + to the callback in the dir_name parameter instead of just the directory + name. + + The logic has been re-written to only create a full_path if we need to + call stat() or if we need to descend into another directory. + + Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba + +2016-06-02 14:10 +0000 [070a51bf7c] Timo Teräs + + * Implement internal abstraction for iostreams + + fopencookie/funclose is a non-standard API and should not be used + in portable software. Additionally, the way FILE's fd is used in + non-blocking mode is undefined behaviour and cannot be relied on. + + This introduces internal abstraction for io streams, that allows + implementing the desired virtualization of read/write operations + with necessary timeout handling. + + ASTERISK-24515 #close + ASTERISK-24517 #close + + Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85 + +2016-11-15 08:07 +0000 [d3b61a98f4] Joshua Colp + + * manager: Bump AMI version number. + + During the development of Asterisk 14 the behavior of + the Command AMI action was altered such that the result + was returned on lines with a prefix of "Output: ". While + this was documented in the UPGRADE.txt file it is also + reasonable that this should bump the AMI version number. + + ASTERISK-26556 + + Change-Id: Idf1bf01608e53f7bfdf43ddb4d0683e53f74ee42 + +2016-11-14 15:57 +0000 [edd7ae85e8] Matt Jordan + + * pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS + + The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how + many pairs of local/remote candidates will be made. If for some reason + we reach this upper bound, ICE will generally fail and no media will + flow between the browser and Asterisk. + + This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of + pairs of candidates we'd theoretically allow, which is + PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied + PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame + Docker), this is far too low to allow WebRTC calls to succeed. + + Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed + even when the system Asterisk was running on had quite a few virtual + interfaces. + + Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55 + +2016-11-14 15:32 +0000 [cc86329228] Matt Jordan + + * apps/app_echo: Only relay a single video source change frame + + In 9785e8d0, app_echo was updated to relay video source updates to the + channel for the purposes of displaying video in WebRTC tests. + Unfortunately, this can cause a Kafkaesque nightmare if two or more + Local channels are in a bridge together where their ends are in + app_echo. When this situation occurs, a video update sent into app_echo + will cause the video update to be relayed to the other Local channels, + causing another round of video updates, etc. In not much time at all, + the channel length queues will be overwhelmed, channel alert pipes will + fail, and all hell will break loose as Asterisk merrily continues to + throw more video update requests onto the channels. + + This patch updates app_echo to *only* relay a single video update. Once + a video update has been made, all further video updates are dropped. + This meets the intended purpose of the original patch: if we get a video + update and we're in app_echo, go ahead and ask the sender to update + themselves. However, once we've got that video stream sync'd up, don't + keep spamming the world. + + Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74 + +2016-11-08 10:11 +0000 [a72ef38113] Matt Jordan + + * res/ari/resource_bridges: Add the ability to manipulate the video source + + In multi-party bridges, Asterisk currently supports two video modes: + * Follow the talker, in which the speaker with the most energy is shown + to all participants but the speaker, and the speaker sees the + previous video source + * Explicitly set video sources, in which all participants see a locked + video source + + Prior to this patch, ARI had no ability to manipulate the video source. + This isn't important for two-party bridges, in which Asterisk merely + relays the video between the participants. However, in a multi-party + bridge, it can be advantageous to allow an external application to + manipulate the video source. + + This patch provides two new routes to accomplish this: + (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} + Sets a video source to an explicit channel + (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource + Removes any explicit video source, and sets the video mode to talk + detection + + ASTERISK-26595 #close + + Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621 + +2016-11-14 14:03 +0000 [7263a17ca0] George Joseph + + * channel: Fix issues in hangup scenarios caused by frame deferral + + ASTERISK-26343 + + Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 + (cherry picked from commit 0be46aaf6b8b9eb5b0160ec591cdc2c6e1802a6d) + +2016-11-14 13:55 +0000 [0dc4567133] George Joseph + + * Revert "Revert "channel: Use frame deferral API for safe sleep."" + + This reverts commit e5365dada5052b87275c048f6e29ac7d5e2b2415. + + Change-Id: Icc40cf0c7687454760762912dd29e4ae79e8e9ee + +2016-11-14 13:55 +0000 [6d61f7bfd1] George Joseph + + * Revert "Revert "autoservice: Use frame deferral API"" + + This reverts commit edca6911f392f47c1a5a25d1d3a357c72b04a78a. + + Change-Id: I76030b87333a2c390cd05392b74b75678d78ddfa + +2016-11-14 13:55 +0000 [f62c9c42fa] George Joseph + + * Revert "Revert "AGI: Only defer frames when in an interception routine."" + + This reverts commit 6bce938c2fcb60b7a77a0e997a6518860c0bfa39. + + Change-Id: Iadbf462bf2a52e8b2fa9ebc75b37b1f688ba51d9 + +2016-11-14 13:54 +0000 [2966fa5ad7] George Joseph + + * Revert "Revert "Add API for channel frame deferral."" + + This reverts commit fa749866c17f91860d3e9f89742eab3e6f03ecbc. + + Change-Id: Idcd1b88fa0766b1326dcc87d8905dbc314c71bd7 + +2016-11-11 10:45 +0000 [c6d755de11] Sebastien Duthil + + * res_ari: Add support for channel variables in ARI events. + + This works the same as for AMI manager variables. Set + "channelvars=foo,bar" in your ari.conf general section, and then the + channel variables "foo" and "bar" (along with their values), will + appear in every Stasis websocket channel event. + + ASTERISK-26492 #close + patches: + ari_vars.diff submitted by Mark Michelson + + Change-Id: I5609ba239259577c0948645df776d7f3bc864229 + +2016-11-14 12:16 +0000 [72da2ef9ff] George Joseph + + * cli: Fix ast_el_read_char to work with libedit >= 3.1 + + Libedit 3.1 is not build with unicode on as a default and so the + prototype for the el_gets callback changed from expecting a char buffer + to accepting a wchar buffer. If ast_el_read_char isn't changed, + the cli reads garbage from teh terminal. + + Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and + updated ast_el_read_char to use the HAVE_ define to detemrine whether + to use char or wchar. + + ASTERISK-26592 #close + + Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a + +2016-11-12 12:15 +0000 [97a75e3829] Tzafrir Cohen + + * Add support for building RADIUS with radcli + + Radcli is yet another RADIUS client library, generally compatible with + freeradius and radiusclient-ng. + + This commit adds autoconf option for detecting it as well and changes + cdr_radius and cel_radius to use its header file in that case. + + ASTERISK-26540 #close + + Change-Id: I271f0715406334874865ffbce0b354b3a2ca148f + +2016-11-10 10:57 +0000 [1bd49040c4] Joshua Colp + + * res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp. + + When optimistic SRTP was on it was possible for us to still + set up a call without an audio stream if an offer was received + with required SRTP. + + This change makes it so this scenario will now fail with a 488 + response. + + ASTERISK-26575 + + Change-Id: I7d14187037681f48879bd20319ac79d0877318f3 + +2016-11-11 02:41 +0000 [dfb951817f] Igor Goncharovskiy + + * Fix closing rtp ports after call finished in chan_unistim. + + Fix ASTERISK-26565 by adding ast_rtp_instance_stop before + rtp instance destroy for chan_unistim. Also several fixes + for displayed text translation. + + Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc + +2016-11-11 00:29 +0000 [939dcf66b0] Timo Teräs + + * addons/chan_mobile: do not use strerror_r + + The two reasons why it might be used are that some systems do not + implement strerror in thread safe manner, and that strerror_r returns + the error code in the string in case there's no error message. + + However, all of asterisk elsewhere uses strerror() and assumes it + to be thread safe. And in chan_mobile the errno is also explicitly + printed so neither of the above reasons are valid. + + The reasoning to remove usage is that there are actually two versions + of strerror_r: XSI and GNU. They are incompatible in their return + value, and there's no easy way to figure out which one is being + used. glibc gives you the GNU version if _GNU_SOURCE is defined, + but the same feature test macro is needed for other symbols. On + all other systems you assumedly get XSI symbol, and compilation warnings + as well as non-working error printing. + + Thus the easiest solution is to just remove strerror_r and use + strerror as rest of the code. Alternative is to introduce ast_strerror + in separate translation unit so it can request the XSI symbol in + glibc case, and replace all usage of strerror. + + Change-Id: I84d35225b5642d85d48bc35fdf399afbae28a91d + +2016-09-23 17:54 +0000 [338f35edcc] Richard Mudgett + + * res_pjsip.c: Rework endpt_send_request() req_wrapper code. + + * Don't hold the req_wrapper lock too long in endpt_send_request(). We + could block the PJSIP monitor thread if the timeout timer expires. + sip_get_tpselector_from_endpoint() does a sorcery access that could take + awhile accessing a database. pjsip_endpt_send_request() might take awhile + if selecting a transport. + + * Shorten the time that the req_wrapper lock is held in the callback + functions. + + * Simplify endpt_send_request() req_wrapper->timeout code. + + * Removed some redundant req_wrapper->timeout_timer->id assignments. + + Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9 + +2016-09-21 15:10 +0000 [bb196323f9] Richard Mudgett + + * res_pjsip: Fix tdata leaks in off nominal paths. + + Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b + +2016-10-24 12:41 +0000 [9df59d9ff4] Richard Mudgett + + * res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. + + Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94 + +2016-11-10 13:38 +0000 [73524bde9c] C.J. Collier + + * chan_sip: Fix typo and re-wrap surrounding docs + + Correct typo of end-pints to end-points + Re-wrap session timer parameter docs to max 80 chars wide; this + eases reading on terminals with lower resolution, commonly the case + for those with visual impairments. + + ASTERISK-26573 + + Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b + Signed-off-by: C.J. Collier + +2016-11-09 15:14 +0000 [bdb6d928c5] Joshua Colp + + * res_pjsip: Perform resolution when explicit IPv6 transport is used. + + This change fixes the SIP resolver such that if an IPv6 transport + is explicitly used it will resolve NAPTR, SRV, and AAAA records. + + You can explicitly use one by specifying it on an endpoint. + + ASTERISK-26571 + + Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36 + +2016-11-10 08:33 +0000 [93a0de1f0e] Joshua Colp + + * app_queue: Add mention of 'ABANDON' variable to CHANGES. + + ASTERISK-26558 + + Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e + +2016-11-10 07:34 +0000 [fa749866c1] George Joseph + + * Revert "Add API for channel frame deferral." + + This reverts commit f073f648b87d45e4729969fd2d83695c300757d1. + Multiple testsuite failures were detected after the fact. + + Change-Id: I968c380418bf65c7166f6ecff30fe8e247ea6682 + +2016-11-10 07:33 +0000 [6bce938c2f] George Joseph + + * Revert "AGI: Only defer frames when in an interception routine." + + This reverts commit 28926d1c81540bbeb16802814d3f2e63c2347bd2. + Multiple testsuite failures were detected after the fact. + + Change-Id: I8d4f5ccbb421a351d616254844ae7e5a31053edb + +2016-11-10 07:32 +0000 [edca6911f3] George Joseph + + * Revert "autoservice: Use frame deferral API" + + This reverts commit afef1b8e4a311d33b3e485b9bab3c6e7fd13fbc9. + Multiple testsuite failures were detected after the fact. + + Change-Id: Ib4cb0c0a6475681ce817f71b4050be25640ab67f + +2016-11-10 07:31 +0000 [e5365dada5] George Joseph + + * Revert "channel: Use frame deferral API for safe sleep." + + This reverts commit 392202304d248147378f1e16f1f012285dc1221f. + + Multiple testsuite issues were discovered after the fact. + + Change-Id: I848c4196dca2994b1a368087004326ea354cff95 + +2016-11-09 18:18 +0000 [edea41126b] George Joseph + + * build: Fix default values for some SANITIZER options + + 2 of the sanitizers didn't have default values so in systems that + don't support sanitizers menuselect would spit out warnings. They + were harmless but confusing. They've now been set to "0". + + Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58 + +2016-11-06 06:04 +0000 [4e8ab6cda9] Sebastian Gutierrez + + * app_queue: new variable set when abandoned + + sets the variable ABANDONED to TRUE if the call was not answered. + + ASTERISK-26558 + + Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3 + +2016-11-08 10:48 +0000 [e5860ce07d] Mark Michelson + + * res_pjsip_session: Do not call session supplements when it's too late. + + res_pjsip_sesssion was hooking into transaction and invite state + changes. One of the reasons for doing so was due to the + PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the + message sending process, and so we should call session supplements to + alter the outgoing message. + + In reality, this event was meant to indicate that the message either + a) had already been sent, or + b) required a DNS lookup and would be sent when the DNS query + completed. + + In case (a), this meant we were altering an already-sent + request/response for no reason. In case (b), this potentially meant we + could be trying to alter a request/response at the same time that the + DNS resolution completed. In this case, it meant we might be stomping on + memory being used by the thread actually sending the message. This + caused potential crashes and memory corruption. + + This patch removes the calls to session supplements from the case where + the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to + alter the message at this point is too late, and it can cause nothing + but harm to try to do it. Because there were no longer any calls to the + handle_outgoing() function, it has been removed. + + Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92 + +2016-11-03 16:46 +0000 [392202304d] Mark Michelson + + * channel: Use frame deferral API for safe sleep. + + This is another case where manual frame deferral can be replaced with + centralized routines instead. + + Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e + (cherry picked from commit d149c4b9e07eeb880d8428ad52c6fdb315cc15f5) + +2016-11-03 16:46 +0000 [afef1b8e4a] Mark Michelson + + * autoservice: Use frame deferral API + + Rather than use manual frame deferral, just let the channel API do it + for us. + + ASTERISK-26343 + + Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49 + (cherry picked from commit 8ba3e2fc27f9966b8c7ce75c1eca6208613a9315) + +2016-11-03 16:42 +0000 [28926d1c81] Mark Michelson + + * AGI: Only defer frames when in an interception routine. + + AGI recently was modified to defer important frames. This was because + when AGI was used in a connected line interception routine, the + resulting connected line frame would end up getting discarded by the + AGI. + + However, this caused bad behavior in other cases. Specifically, during a + transfer, if someone attempted to manually set the Caller ID on a + channel in an AGI, the deferred connected line frame would end up + overwriting what had been manually set in the AGI. + + Since the initial issue was specific to interception routines, this + change removes the manual frame deferral from AGI and instead uses the + new frame deferral API in interception routines. + + ASTERISK-26343 #close + Reported by Morton Tryfoss + + Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208 + +2016-11-03 16:36 +0000 [f073f648b8] Mark Michelson + + * Add API for channel frame deferral. + + There are several places in Asterisk that have duplicated logic + for deferring important frames until later. + + This commit adds a couple of API calls to facilitate this automatically. + + ast_channel_start_defer_frames(): Future reads of deferrable frames on + this channel will be deferred until later. + + ast_channel_stop_defer_frames(): Any frames that have been deferred get + requeued onto the channel. + + ASTERISK-26343 + + Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641 + +2016-11-02 10:52 +0000 [d30415bfa1] Joshua Colp + + * res_stasis: Don't unsubscribe from a NULL bridge. + + A NULL bridge has special meaning in res_stasis for + unsubscribing. It means that a subscription to ALL + bridges should be removed. This should not be done + as part of the normal subscription management in + the res_stasis channel loop. + + ASTERISK-26468 + + Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0 + +2016-11-03 07:42 +0000 [0a698cd932] Alexander Anikin + + * chan_ooh323: Fixes to work right with Cisco devices + + Changed output packets queue processing algo to one read-one write + instead of all read-all send + + Remove h.245 tunneling parameter from ReleaseComplete packet + + ASTERISK-24400 #close + Reported by: Dmitry Melekhov + Tested by: Dmitry Melekhov + + Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6 + +2016-11-03 13:10 +0000 [a1cdc3891a] Alexander Anikin + + * chan_ooh323: reset rrq count on gk registration + + reset registration attempts count on success registration on gatekeeper + + Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336 + +2016-11-06 05:40 +0000 [b2b5f9d897] frahaase + + * ast_format: Adds an identifier for interleaved audio formats to the ast_format + + Adds an identifier (with a getter and setter) to detect channels with + interleaved audio. + This is needed by the binaural bridge_softmix patch (ASTERISK-26292) and + was already discussed here: + http://lists.digium.com/pipermail/asterisk-dev/2016-October/075900.html + The identifier can be set during fmtp parsing (to be seen in the + res_format_attr_opus.c change). + + ASTERISK-26292 + + Change-Id: I359801cc5f98c35671c48dabc81a7f4ee1183d63 + +2016-11-06 03:46 +0000 [fbbbd0add9] Michael Kuron + + * automon: restore mixing of the both channels after recording stops + + This is a regression over Asterisk 11, introduced by + 2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via + the automon DTMF code would automatically be mixed together using sox because + app_monitor would be called with the m option. This commit restores this + behavior. + + Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759 + +2016-11-04 15:42 +0000 [367d4903cc] Matt Jordan + + * res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems + + Not surprisingly, using Respoke (and possibly other systems) it is + possible to blow past the 16k limit for a WebSocket packet size. This + patch bumps it up to 32k, which, at least for Respoke, is sufficient. + For now. + + Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that + matter), this patch adds a LOW_MEMORY directive that sets the buffer to + 8k for systems who have asked for their reduced memory availability to + be considered. + + Change-Id: Id235902537091b58608196844dc4b045e383cd2e + +2016-11-04 15:40 +0000 [7a449b6819] Matt Jordan + + * res_stasis: Set a video source mode on Stasis created bridges + + When a bridge is created via ARI (through res_stasis), no video source + mode is set by default. As a result, any endpoint sending video media + won't ever see any video reflected back to it. + + This patch defaults a bridge to a 'follow the talker' video mode. + Further work can be done to add routes that allow for the video mode to + be controlled through the /bridges resource. + + Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866 + +2016-11-04 15:37 +0000 [bbe943729a] Matt Jordan + + * main/bridge_channel: Fix channel reference leak on video source + + When a channel is made the video source, the bridge holds a reference to + it. Whenever the video source changes, that reference is released. + However, a ref leak does occur if the channel leaves the bridge (such as + being hung up) while it is the video source, as the bridge never + releases the ref in such a case. + + This patch adds a line to the bridge_channel_internal_join routine such + that, when a channel finishes its time in the bridge, it notifies the + bridge via ast_bridge_remove_video_src that if it is a video source its + reference should be released. + + ASTERISK-26555 #close + + Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a + +2016-11-04 15:36 +0000 [a70d6dba8c] Matt Jordan + + * main/bridge: Add some verbose logging for video source changes + + It's actually quite useful to see the source of a video stream change. + This doesn't happen terribly often, even with talk detection - but when + it does, it's nice to know which channel is now providing your video + stream. + + As a verbose 5 level message, it shouldn't be terribly spammy or costly + to have, and is 'lower level' then most other verbose messages that the + bridge system emits. + + ASTERISK-26555 + + Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c + +2016-11-04 15:33 +0000 [fb17b630a5] Matt Jordan + + * bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source + + WebRTC clients really, really want to know the SSRC of the media they're + getting. Changing the SSRC is generally not a good thing. + + bridge_softmix, starting in Asterisk 12, started changing the SSRC of + parties as they joined or left the bridge. With most phones, this isn't + a problem: phones just play back the stream they're getting. With WebRTC + clients, however, the SSRC is tied to a media stream that may be + negotiated. When a new SSRC just shows up, the media can be dropped. + + As it turns out, the SSRC change shouldn't even be necessary. From the + perspective of the client, it's still talking to Asterisk with the same + media stream: why indicate that the far party has suddenly changed to a + different source of media? + + This patch opts to just remove the SSRC changes. With this patch, video + clients that join/leave a softmix bridge actually get the video stream + instead of freaking out. + + ASTERISK-26555 + + Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf + +2016-10-28 15:11 +0000 [70d5f90e3d] Kevin Harwell + + * stasis_recording/stored: remove calls to deprecated readdir_r function. + + The readdir_r function has been deprecated and should no longer be used. This + patch removes the readdir_r dependency (replaced it with readdir) and also moves + the directory search code to a more centralized spot (file.c) + + Also removed a strict dependency on the dirent structure's d_type field as it + is not portable. The code now checks to see if the value is available. If so, + it tries to use it, but defaults back to using the stats function if necessary. + + Lastly, for most implementations of readdir it *should* be thread-safe to make + concurrent calls to it as long as different directory streams are specified. + glibc falls into this category. However, since it is possible that there exist + some implementations that are not safe, locking has been added for those other + than glibc. + + ASTERISK-26412 + ASTERISK-26509 #close + + Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba + +2016-11-04 10:57 +0000 [bf01ff53f8] Kevin Harwell + + * Revert "chan_sip: Fix lastrtprx always updated" + + This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc. + + Unfortunately, the aforementioned commit caused a regression (incoming calls + would eventually disconnect). Thus it is being removed. + + ASTERISK-26523 #close + ASTERISK-25270 + + Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d + +2016-11-03 13:45 +0000 [1504194215] Alexander Anikin + + * chan_ooh323: Fix infinite loop on read second part of H.225 packet + + Fix logic on read second part of H.225 packet. There was infinite loop on + wrong connections due to read before poll. + + Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff + +2016-11-03 11:55 +0000 [78dc6ceaf6] George Joseph + + * pjproject_bundled: Fix issue with libasteriskpj needing libresample + + libresample is only needed by pjproject if we're building pjsua, which + we only do if TEST_FRAMEWORK is selected. It's required by pjsua to + process audio which is needed by some testsuite tests. Unfortunately, + pjproject relies on a newer version of libresample than the version + that ships by most distros so we need to compile the version that's + bundled with pjproject. Since we only need it for pjsua, we DON'T want + it's symbols exposed when we actually build asterisk. + + There was a problem however... TEST_FRAMEWORK is only known AFTER we've + already run ./configure on both asterisk and pjproject but pjproject's + ./configure needs to test it to know whether to set up to build + libresample or not. The previous way of figuring this out was to + always tell ./configure "yes" but not actually build the library. This + caused an issue where building libasteriskpj was being told to include + libresample but it wasn't actually there. + + The solution is to still do a default pjproject configure during an + asterisk ./configure but if makeopts or menuselect.makeopts changes + subsequently, we now reconfigure pjproject, taking into account the + current state of TEST_FRAMEWORK. Previously, if makeopts or + menuselect.makeopts changed, only a recompile of pjproject was done. + + Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685 + +2016-11-01 19:48 +0000 [0904c1f4cc] Sebastian Gutierrez + + * chan_sip: add missing account code + + Added missing account to AMI event of sip show peers + + ASTERISK-26176 #close + + Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482 + +2016-11-02 09:15 +0000 [4de5454ef1] Joshua Colp + + * app_dial: Fix incorrect device state when channel is picked up. + + Given the scenario where multiple channels are dialed using Dial() + but the caller is picked up using PickupChan() all outgoing channels + except the channel specified to PickupChan() would be marked + as ringing until the call had been hung up. + + When using the PickupChan application the channel executing the + application is swapped into place of another channel. As part + of this process the channel is answered. The Dial application + has explicit logic which checks if the channel is answered, + cancels all other outgoing channels, and bridges. This logic is + different than the normal logic that is executed when an outgoing + channel is answered. This different logic failed to publish dial + events stating that the other outgoing channels had been canceled. + As a result references to the outgoing channels were held onto by + the dial masquerade process until the call had been ended and + the channels had gone away. This would result in the channels + appearing in the "core show channels" list despite not being present + anymore and would also result in incorrect device state. + + This change makes it so that this logic also publishes + dial events stating that the other outgoing channels have been + canceled. + + ASTERISK-26549 + + Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f + +2016-09-13 04:08 +0000 [9ac53877f6] Alexander Traud + + * rtp_engine: Allow more than 32 dynamic payload types. + + Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK + (Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the + dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges. + Consequently, when the dynamic range is exhausted, this change utilizes payload + types in the range between 35 and 63 giving room for another 29 payload types. + + ASTERISK-26311 #close + + Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964 + +2016-11-02 05:05 +0000 [6a99f007d6] Tzafrir Cohen + + * autoconf: more variants for OSARCH linux-gnu + + There are quite a few odd GNU/Linux platforms. Just call all of them + linux-gnu. + + Specifically this fixes building the Debian platforms mips64el and x32. + And maybe also others. + + ASTERISK-26546 #close + + Change-Id: I06ec4bd7f0ee1c84b6b24d81538223b07c4174b1 + +2016-11-01 13:13 +0000 [f29b8d62bb] Richard Mudgett + + * bundled pjproject: Fix DNS write to freed memory. + + PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS + patch. + + The patch below fixes a write to freed memory under cartain DNS lookup + conditions. + + 0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch + + ASTERISK-26516 + Reported by: Richard Mudgett + + Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5 + +2016-11-01 06:56 +0000 [6233e146c6] Joshua Colp + + * res_pjsip_sdp_rtp: Limit number of formats to defined maximum. + + The res_pjsip_sdp_rtp module did not restrict the number of + formats added to a media stream in the SDP to the defined + limit. If allow=all was used with additional loaded codecs this + could result in the next media stream being overwritten some. + + This change restricts the module to limit it to the defined + maximum and also increases the maximum in our bundled pjproject. + + ASTERISK-26541 #close + + Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8 + +2016-10-31 17:35 +0000 [8060cd1ec1] Kevin Harwell + + * codecs.conf.sample: Add sample and option descriptions for codec_opus + + codecs.conf.sample was missing codec opus's configuration options, descriptions, + and examples. This patch adds the configuration options and examples to + codecs.conf.sample that can be used with codec_opus. + + ASTERISK-26538 #close + + Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b + +2016-10-20 07:27 +0000 [c30d677333] Matt Jordan + + * res/stasis: Add CLI commands for displaying/debugging ARI apps + + This patch adds three new CLI commands: + - ari show apps: list the registered ARI applications + - ari show app: show detailed information about an ARI application + - ari set debug: dump events being sent to an ARI application + + Note that while these CLI commands live in the res_stasis module, we use + the 'ari' family for these commands. This was done as most users of + Asterisk aren't aware of the semantic differences between ARI and + res_stasis, and some 'ari' CLI commands already exist. + + ASTERISK-26488 #close + + Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5 + +2016-11-01 08:32 +0000 [2526dff94d] Grachev Sergey + + * chan_sip: Incorrect display option Outbound reg. retry 403 + + If in sip.conf (general section) set option register_retry_403=no, + the command "sip show settings" return value: + Outbound reg. retry 403:0 + If in sip.conf (general section) set option register_retry_403=yes, + the command "sip show settings" return value: + Outbound reg. retry 403:-1 + + * In static char "sip show settings" for "Outbound.reg. retry 403" + option use AST_CLI_YESNO + + ASTERISK-26476 #close + + Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9 + +2016-11-01 04:18 +0000 [ed08811e64] Tzafrir Cohen + + * netsock.c: fix includes for HURD + + ASTERISK-25070 + + Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814 + +2016-11-01 04:00 +0000 [69fed26deb] Tzafrir Cohen + + * define PATH_MAX for HURD + + PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD + define it to a constant. It is indeed not safe to assume there won't be + longer paths and Asterisk generally does err safely on such cases. + + So even for HURD we'll just pretend PATH_MAX is 4096. + + ASTERISK-25070 #close + + Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3 + +2016-10-31 16:12 +0000 [f27f837a9f] George Joseph + + * pjproject_bundled: Fix compile of pjsua so it handles audio + + In order for pjsua and its python binding to actually negotiate + audio for the testsuite tests, it needs g711 and resample. The + pj* libraries themselves do not. Unfortunately, pjproject relies + on a brand new libresample that most distros don't ship so we need + to use the libresample already bundled with pjproject. Only the pjsua + executable and the _pjsua.so python library are linked with it so it + shouldn't interfere with asterisk itself. + + Also it was pointed out that apply_patches couldn't handle multiple + patches that depended on each other during the dry-run, so the + dry-run was removed. + + Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098 + +2016-10-31 13:46 +0000 [1648ca06c3] Etienne Lessard + + * manager: Add documentation for NewConnectedLine event. + + The NewConnectedLine event has been added by commit fe7671f, but the + documentation was missing. + + ASTERISK-26537 #close + + Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6 + +2016-10-30 13:33 +0000 [273debd261] Corey Farrell + + * vector: Prevent NULL argument to memcpy. + + Headers declare that memcpy does not accept NULL argument for the first + two parameters. Add a conditional block to prevent memcpy and ast_free + from running on vectors with NULL element array. + + ASTERISK-26526 #close + + Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71 + +2016-10-29 10:19 +0000 [ad60927a40] Corey Farrell + + * astobj2: Declare private variable data_size for AO2_DEBUG only. + + Every ao2 object contains storage for a private variable data_size, + though the value is never read if AO2_DEBUG is disabled. This change + makes the variable conditional, reducing memory usage. + + ASTERISK-26524 #close + + Change-Id: If859929e507676ebc58b0f84247a4231e11da07f + +2016-10-28 14:55 +0000 [6feee22e09] Richard Mudgett + + * bundled pjproject: Crashes while resolving DNS names. + + PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS + patch. + + The patches below fix the DNS lookup race condition crash caused by + attempting to send the same message twice for the single DNS lookup. + + 0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch + 0006-r5473-svn-backport-Fix-pending-query.patch + + The patch below removes a cached DNS response from the hash table when + another thread is referencing the old entry. The table still contained + the entry when it was destroyed which can result in inexplicable crashes. + + 0006-r5475-svn-backport-Remove-DNS-cache-entry.patch + + ASTERISK-26344 #close + Reported by: Ian Gilmour + + ASTERISK-26387 #close + Reported by: Harley Peters + + Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4 + +2016-10-28 16:59 +0000 [12bdde6a6c] George Joseph + + * pjproject_bundled: Fix issue where "/version.mak" wasn't found + + main/Makefile includes third-party/pjproject/build.mak but + doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak" + evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile + before the include. + + Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604 + +2016-10-28 13:30 +0000 [9d8b9b6ca5] Matt Krokosz + + * res_pjsip_outbound_publish: Fix crash when publishing device state. + + While publishing device state between multiple instances of Asterisk, + a crash will sporadically occur under high CPS which looks to be a + race condition operating on the publisher queue. + + ASTERISK-26506 + + Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6 + +2016-10-27 21:49 +0000 [d6ad867897] Corey Farrell + + * Fix shutdown crash caused by modules being left open. + + It is only safe to run ast_register_cleanup callbacks when all modules + have been unloaded. Previously these callbacks were run during graceful + shutdown, making it possible to crash during shutdown. + + ASTERISK-26513 #close + + Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21 + +2016-10-28 09:50 +0000 [badd38f031] Rusty Newton + + * SAC documentation: don't specify transports for endpoints and registrations + + Removing explicit transport definition for endpoints and registrations. It + isn't necessary and isn't generally advised. + + ASTERISK-26514 #close + + Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb + +2016-10-18 09:06 +0000 [0646b48ece] Tzafrir Cohen + + * chan_dahdi: remove by_name support + + Support for referring to DAHDI channels by logical names was added in + (FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support + of refering to channels by name. + + While technically usable, it has never been properly supported in + dahdi-tools, as using it would require many changes at the Asterisk + level. Instead logical mapping was added at the kernel level. + + Thus it seems that refering to DAHDI channels by name is not really used + by anyone, and therefore should probably be removed. + + Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485 + +2016-10-26 18:48 +0000 [4f45d62653] George Joseph + + * pjproject_bundled: Remove usage of tar's --strip-components option + + Older versions of tar don't support the --strip-components option so + instead of doing 'tar --strip-components=1 -C source', we now just + untar to the tarball's root directory (pjproject-) and + rename that directory to 'source'. + + Also fixed an issue where the pjproject source directory is a hard + coded absolute pathname. + + ASTERISK-26510 #close + ASTERISK-22480 #close + + Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0 + +2016-10-26 21:40 +0000 [a6e5bae3ef] Corey Farrell + + * Remove ASTERISK_REGISTER_FILE. + + ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes + all traces of it. + + Previously exported symbols removed: + * __ast_register_file + * __ast_unregister_file + * ast_complete_source_filename + + This also removes the mtx_prof static variable that was declared when + MTX_PROFILE was enabled. This variable was only used in lock.c so it + is now initialized in that file only. + + ASTERISK-26480 #close + + Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 + +2016-10-27 08:07 +0000 [6993f3c9c3] Joshua Colp + + * res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls. + + The res_pjsip_caller_id module wrongly assumed that a + saved From header would always exist on sessions. This + is true until an inbound call is received and a session + timer causes an UPDATE to be sent. In this case there will + be no saved From header and a crash will occur. This change + makes it fall back to the From header of the outgoing request + if no saved From header is present. + + ASTERISK-26307 #close + + Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa + +2016-10-26 07:51 +0000 [95062fe220] Joshua Colp + + * app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. + + When executing the MailboxExists dialplan application and + MAILBOX_EXISTS dialplan function the passed in temporary voice + mailbox was not cleared, causing it to try to free garbage. + + ASTERISK-26503 #close + + Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3 + +2016-10-23 07:38 +0000 [aed6c219a3] Joshua Colp + + * pjsip: Fix a few media bugs with reinvites and asymmetric payloads. + + When channel format changes occurred as a result of an RTP + re-negotiation the bridge was not informed this had happened. + As a result the bridge technology was not re-evaluated and the + channel may have been in a bridge technology that was incompatible + with its formats. The bridge is now unbridged and the technology + re-evaluated when this occurs. + + The chan_pjsip module also allowed asymmetric codecs for sending + and receiving. This did not work with all devices and caused one + way audio problems. The default has been changed to NOT do this + but to match the sending codec to the receiving codec. For users + who want asymmetric codecs an option has been added, asymmetric_rtp_codec, + which will return chan_pjsip to the previous behavior. + + The codecs returned by the chan_pjsip module when queried by + the bridge_native_rtp module were also not reflective of the + actual negotiated codecs. The nativeformats are now returned as + they reflect the actual negotiated codecs. + + ASTERISK-26423 #close + + Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc + +2016-10-26 06:32 +0000 [7925f60cd9] Joshua Colp + + * res_pjsip_sdp_rtp: Fix address family of explicit media_address. + + When an explicit media_address is provided the address family + in the SDP needs to be set to reflect it. + + ASTERISK-26309 + + Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79 + +2016-10-25 11:20 +0000 [802bbf8752] George Joseph + + * test_astobj2_thrash: Fix multithreaded issues + + The test uses 4 threads to grow, count, lookup and shrink 15K objects + in a container. If there's only 1 execution engine available, the test + will complete in <50ms. If each threads gets its own execution engine, + the test may timeout after 60 seconds because the count thread does a + locked ao2_callback on the whole container in a tight loop with only + a sched_yield to give up time. The lock contention makes the test + execution times wildly variable and mostly timeout. 2 execution + engines are OK, 3 results in about 33% failure rate and >=4 causes + a 80% failure rate. + + To fix, the sched_yield was changed to a usleep(500). + + Also, the number of buckets specified for the container was an even + number so that was changed to the next prime number greater than + (MAX_HASH_ENTRIES / 100). That's 151 currently. + + Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77 + +2016-10-18 09:04 +0000 [2b9ad3a5f7] Alexei Gradinari + + * chan_pjsip: segfault on already disconnected session + + On heavy loaded system the TCP/TLS incoming calls could be + disconnected by pjproject while these calls are being + processed by asterisk. + + This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref + to inform pjproject that an INVITE session is in use. + + ASTERISK-26482 #close + + Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33 + +2016-10-10 11:49 +0000 [01d1d3763f] Badalyan Vyacheslav + + * cdr_radius,cel_radius: Fix old memleak in unload + + - Call "rc_openlog" optional. If you do not call, + you will simply NULL instead of a name. + + - On the one PID can be only one syslog channel. + And it can already be run in logger.c + + - Calling rc_openlog we assigns a new name for + the channel syslog. This unexpected behavior for logger.c. + + Most lesser evil, is to agree on a NULL name syslog + if the channel was not launched in logger.c. + + It also solves the problem of memory leaks. + + ASTERISK-26455 #close + + Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd + +2016-10-24 10:55 +0000 [16c23b57c7] George Joseph + + * pjproject_bundled: Fixed various build issues + + * CFLAGS is now properly set when using older gcc. + * All third-party pjproject targets have been removed. This fixes + an issue with older libsrtp in some distros. + * Manually removing the source directory now causes a rebuild. + * EXTERNALS_CACHE_DIR is now properly checked. + * Whitespace fixes. + + Change-Id: I98fec6847efc5602a9f41cb95096fd660a49fa60 + +2016-10-24 14:13 +0000 [1d277e7cb6] Pascal Cadotte Michaud + + * typo: s/paranthesis/parenthesis/ in a comment + + Change-Id: I7c1f4eb051177ee22cbe97e063d4a3effe29be30 + +2016-09-19 06:13 +0000 [403c4f5833] Joshua Colp + + * pjsip: Support dual stack automatically. + + This change adds support for dual stack automatically. No + configuration is required and the IP address and version + in the SIP messages and SDP will be automatically changed + based on the transport over which the message is being + sent. RTP usage has also been changed to listen on both + IPv4 and IPv6 simultaneously to allow media to flow, and + to allow ICE support on both simultaneously. This also + allows failover between IPv6 and IPv4 to work as expected. + + ASTERISK-26309 #close + + Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d + +2016-10-19 12:05 +0000 [3bd76dd679] Mark Michelson + + * ARI: Add duplicate channel ID checking for channel creation. + + This is similar to what is done for origination, but for the 14 and up + channel creation method. When attempting to create a channel, if a + channel ID is specified and a channel already exists with that ID, then + a 409 is returned. + + Change-Id: I77f9253278c6947939c418073b6b31065489187c + +2016-10-17 14:18 +0000 [e459b8dadf] Mark Michelson + + * ARI: Detect duplicate channel IDs + + ARI and AMI allow for an explicit channel ID to be specified + when originating channels. Unfortunately, there is nothing in + place to prevent someone from using the same ID for multiple + channels. Further complicating things, adding ID validation to channel + allocation makes it impossible for ARI to discern why channel allocation + failed, resulting in a vague error code being returned. + + The fix for this is to institute a new method for channel errors to be + discerned. The method mirrors errno, in that when an error occurs, the + caller can consult the channel errno value to determine what the error + was. This initial iteration of the feature only introduces "unknown" and + "channel ID exists" errors. However, it's possible to add more errors as + needed. + + ARI uses this feature to determine why channel allocation failed and can + return a 409 error during origination to show that a channel with the + given ID already exists. + + ASTERISK-26421 + + Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06 + +2016-10-19 17:53 +0000 [e03364c40a] snuffy + + * Fix issue with CLI not returning to prompt after running "features show" + + ASTERISK-26444 #close + + Change-Id: I91d645b7e6e5dba35f8c410df2be77a8c0e3acb8 + +2016-10-04 18:24 +0000 [3e96d491d0] Michael Walton + + * res_rtp_asterisk: Add ice_blacklist option + + Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the + form ice_blacklist = , e.g. ice_blacklist = + 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay + discovery. This is useful for optimizing the ICE process where a system + has multiple host address ranges and/or physical interfaces and certain + of them are not expected to be used for RTP. Multiple ice_blacklist + configuration lines may be used. If left unconfigured, all discovered + host addresses are used, as per previous behavior. + + Documention in rtp.conf.sample. + + ASTERISK-26418 #close + + Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9 + +2016-10-18 16:30 +0000 [f14ef51ead] Mark Michelson + + * CDR: Alter destruction pattern for CDR chains. + + CDRs form chains. When the root of the chain is destroyed, it then + unreferences the next CDR in the chain. That CDR is destroyed, and it + then unreferences the next CDR in the chain. This repeats until the end + of the chain is reached. While this typically does not cause any sort of + problems, it is possible in strange scenarios for the CDR chain to grow + way longer than expected. In such a scenario, the destruction pattern + can result in a stack overflow. + + This patch fixes the problem by switching from a recursive pattern to an + iterative pattern for destruction. When the root CDR is destroyed, it is + responsible for iterating over the rest of the CDRs and unreferencing + each one. Other CDRs in the chain, since they are not the root, will + simply destroy themselves and be done. This causes the stack depth not + to increase. + + ASTERISK-26421 #close + Reported by Andrew Nagy + + Change-Id: I3ca90c2b8051f3b7ead2e0e43f60d2c18fb204b8 + +2016-10-18 11:51 +0000 [f31772ec20] Joshua Colp + + * ari: Update model validator based on addition of asterisk_id. + + ASTERISK-26470 + + Change-Id: I9c386f7a1c7d969161b28f189eb6298bbc5b7541 + +2016-09-11 10:13 +0000 [18a6f250e2] Tzafrir Cohen + + * menuselect: invalid test for GTK2 + + configuire.ac was only checking for the existence of pkg-config + and not the gtk2 package itself. Now it calls AST_PKG_CONFIG_CHECK + for gtk+-2.0. + + ASTERISK-26356 #close + + Change-Id: I93e9d0166341f0e7f84b52955bb6f81da42f2ef6 + +2016-10-18 03:01 +0000 [a43ee21211] Alexander Traud + + * cli: Auto-complete File not Module for core set debug. + + Since Asterisk 1.8, the command "core set debug" on the command-line interface + asks not for a file (.c) but a module name. This change shows modules (.so) on + the auto-completion via a tabulator or the question mark. Now, when you + partially type a module name, TAB or ?, you get the correct candidiates. + + ASTERISK-26480 + + Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0 + +2016-08-12 11:22 +0000 [dce31f90ba] frahaase + + * Binaural synthesis (confbridge): On/off setting for binaural synthesis. + + Adds setting to confbridge.conf (binaural_active) that determines if binaural + synthesis can be available in bridge_softmix. + + ASTERISK-26292 + + Change-Id: I59dfcb8e55fe1df4ef32045882fea5bb58fc71db + +2016-10-17 11:39 +0000 [2a808b2fa6] George Joseph + + * pjproject_bundled: Add patch to address SSL crash + + Addresses crashes when an attempt is made to operate on an SSL socket + after the socket has been closed. + + ASTERISK-26477 #close + + Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002 + +2016-10-13 14:09 +0000 [973e57d5ce] Leandro Dardini + + * app_queue: Added initialization for "context" parameter + + When using Asterisk Realtime Architecture, empty fields are skipped and the + default values are used. If the "context" parameter in queue was set and then + cleared from the database, the old value remains in memory and it continues + to be used. This change initialize the "context" parameter with an empty value, + allowing clearing the parameter. + + ASTERISK-26462 #close + + Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905 + +2016-10-15 20:05 +0000 [dd5129d84a] Matt Jordan + + * res/ari: Add the Asterisk EID field to outgoing events + + This patch adds the Asterisk EID field to all outgoing ARI events. + Because this field should be added to all events as they are + transmitted, it is appended to the JSON message just prior to it being + handed off to the application message handler. This makes it somewhat + resilient to both new events being added to ARI, as well as other + potential event transport mechanisms. + + ASTERISK-26470 #close + + Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d + +2016-10-13 02:06 +0000 [2b03017022] Moises Silva + + * chan_rtp: Set a sane default rtp engine for unicast. + + ASTERISK-26439 + + Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011 + +2016-10-16 17:25 +0000 [6651c66e68] George Joseph + + * utils.c: Fix ast_set_default_eid for multiple platforms + + ast_set_default_eid was searching for ethX, emX, enoX, ensX and even + pciD#U interface names. While this was a good attempt, it wasn't + inclusive enough to capture interfaces like enp6s0 or ens6d1, etc. + + Rather than relying on interface names, we now simply find the first + interface returned by the OS that has a hardware address and that + address isn't all 0x00 or all 0xff. The code IS different for BSD, + Solaris and Linux based on what method is available for enumerating + interfaces. + + Tested on: + FreeBSD9 + CentOS6 + Ubuntu14 + Fedora24 + + I was unable to test on Solaris at this time but the code for Solaris + is used elsewhere at Digium. + + Change-Id: Iaa6db87ca78a9a375e47d70e043ae08c1448cb72 + +2016-10-15 04:58 +0000 [e9315791b3] Michael Kuron + + * chan_sip: Only send video on outgoing channel if incoming channel supports it + + Previously, the settings videosupport=always and videosupport=yes behaved + identically and unconditionally caused a video offer to be sent in the SDP on + an outgoing call. This was a regression introduced with commit + 5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 in Asterisk 1.6.1. + + This commit restores correct behavior: videosupport=always causes a video offer + to be sent unconditionally, while videosupport=yes will only offer video on an + outbound channel if the incoming channel it is bridged to also supports video. + That way, the device receiving the outgoing call can display the correct user + interface elements for audio or video and will not unnecessarily show a blank + video window on an audio-only call. + + ASTERISK-17470 #close + + Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae + +2016-10-14 00:18 +0000 [aa39a87697] Corey Farrell + + * Fix issues with bundled pjproject cached download. + + Previously when testing I had a preexisting makeopts in ASTTOPDIR. The + ordering of configure.ac causes --with-externals-cache to be processed + after third-party configure. In cases where the Asterisk clone is + cleaned it would cause pjproject to be downloaded to /tmp. This + moves processing of the externals cache and sounds cache to happen + before third-party configure. + + This also addresses a possible issue with the third-party Makefile. If + TMPDIR is set by the environment it would override the path given to + --with-externals-cache. + + ASTERISK-26416 + + Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d + +2016-10-12 16:24 +0000 [9c49b96374] Richard Mudgett + + * Audit ast_json_pack() calls for needed UTF-8 checks. + + Added needed UTF-8 checks before constructing json objects in various + files for strings obtained outside the system. In this case string values + from a channel driver's peer and not from the user setting channel + variables. + + * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json + object construction. + + ASTERISK-26466 + Reported by: Richard Mudgett + + Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096 + +2016-10-12 16:20 +0000 [774d5f7ef7] Richard Mudgett + + * json: Check party id name, number, subaddresses for UTF-8. + + * Updated unit test as ast_json_name_number() is now NULL tolerant. + + ASTERISK-26466 #close + Reported by: Richard Mudgett + + Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6 + +2016-10-11 18:14 +0000 [1c4c6c082d] Richard Mudgett + + * json: Add UTF-8 check call. + + Since the json library does not make the check function public we + recreate/copy the function in our interface module. + + ASTERISK-26466 + Reported by: Richard Mudgett + + Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99 + +2016-10-12 17:42 +0000 [6fe5202c2c] Richard Mudgett + + * aoc.c: Whitespace cleanup + + * In s_to_json() removed unnecessary ast_json_ref() to ast_json_null() + when creating the type json object. The ref is a noop. + + Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a + +2016-10-12 16:22 +0000 [c3bf1632cd] Richard Mudgett + + * app_minivm.c: Fix malformed ast_json_pack() call. + + Change-Id: I082b239022fac462666e52a14a44304748908dc0 + +2016-10-12 17:27 +0000 [9c54964dc5] Richard Mudgett + + * app_queue.c: Fix clearing of pause reason string. + + The pause reason is not always cleared when it should be cleared. + + * Made set_queue_member_pause() always clear pause reason if not pausing + with a reason string. + + Change-Id: I993dad19626ec017478a230e980989438b778c53 + +2016-10-12 16:30 +0000 [3b3d06884c] George Joseph + + * res_config_mysql: Fix several issues related to recent table changes + + Unlike any of the other database drivers, res_config_mysql checks that + the table definition matches the requirements for every insert and + update statement. Since all requirements are forced to 'char', any + column that isn't a char, like ps_contacts' expiration_time, + qualify_timeout, etc., will throw a warning. It's kinda harmless but + very misleading. Since no other driver does those checks on insert + or update, they've been removed from res_config_mysql. Also, all + the logic that actually attempted to ALTER the table to fix the issue + has been removed. With the move to alembic, the auto-alter + functionality is not only unnecessary, it's also dangerous. + + The other issue is that res_config_mysql calls the mysql_insert_id + function inside store_mysql. Presumably the intention was to return + the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE + IS NON_PORTABLE AND MAY CHANGE. That value is then returned to + config realtime as the number of rows inserted. Guess what? The value + changed. It now only returns the number of rows inserted if there's an + auto increment column on the table, which ps_contacts doesn't have. + Otherwise it returns 0. So now, the insert worked but we tell config + realtime and sorcery that no rows were inserted. That call to + mysql_insert_id was removed and we now always return 1 if the insert + succeeded. We're only inserting 1 row at a time anyway. If the insert + fails, we still return -1. + + ASTERISK-26362 #close + Reported-by: Carlos Chavez + + Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4 + +2016-08-12 11:22 +0000 [dd6fc1bb7d] frahaase + + * Binaural synthesis (confbridge): Adds libfftw3 as dependency. + + Adds libfftw3 to the build chain that is is going to be used for binaural + synthesis by bridge_softmix. + + ASTERISK-26292 + + Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b + +2016-09-29 13:08 +0000 [20c3dba39e] Torrey Searle + + * res_fax: Fix a tight race condition causing fax to crash in audio fallback + + When T.38 gets rejected and G711 failback occurs there is a period of + time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set, + leading to a crash. + + Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982 + +2016-10-06 09:58 +0000 [86e8716952] George Joseph + + * app_dial: Add the "Q" option to set the cause on unanswered channels + + The "Q" option will set the cause on the unanswered channels when + another channel answers. It overrides the default of + ANSWERED_ELSEWHERE. + + NOTE: chan_sip does not support setting the cause on a CANCEL to + anything other than ANSWERED_ELSEWHERE. + + ASTERISK-26446 #close + + Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47 + +2016-10-11 06:55 +0000 [4f7f8a7e95] Alexander Traud + + * chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia. + + In the SIP channel driver chan_sip, auto_comedia was expected to be used in + tandem with auto_force_rport. Or stated differently: Only when auto_force_rport + was chosen (the default), auto_comedia worked. This change allows auto_comedia + to be set independently of the state of (auto_)force_rport. For example, + nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments + when IPv6 clients are behind a Firewall. + + ASTERISK-26457 #close + + Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2 + +2016-10-10 16:59 +0000 [17031f12fe] Badalyan Vyacheslav + + * vector: After remove element recheck index + + Small fix. It is necessary to double-check + the index that we just removed because there + is a new element. + + ASTERISK-26453 #close + + Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7 + +2016-09-29 12:52 +0000 [cc269766b8] Torrey Searle + + * res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge + + If a bridge switched to P2P when a DTMF was in progress it + was possible for the DTMF to continue being sent indefinitely. + + Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29 + +2016-10-09 21:28 +0000 [fafdde322c] Corey Farrell + + * logger: Prevent output of verbose messages initiated from rasterisk. + + Remote asterisk consoles should only display verbose log messages + created by the daemon. The first patch for ASTERISK-26410 caused + a couple verbose messages to be printed when the rasterisk process + ended. + + ASTERISK-26410 + + Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a + +2016-10-04 20:46 +0000 [7af7490e42] Michael Walton + + * audiohooks: Remove redundant codec translations when using audiohooks + + The main frame read and write handlers in main/channel.c don't use the + optimum placement in the processing flow for calling audiohooks + callbacks, as far as codec translation is concerned. This change places + the audiohooks callback code: + * After the channel read translation if the frame is not linear before + the translation, thereby increasing the chance that the frame is linear + as required by audiohooks + * Before the channel write translation if the frame is linear at this + point + This prevents the audiohooks code from instantiating additional + translation paths to/from linear where a linear frame format is already + available, saving valuable CPU cycles + + ASTERISK-26419 + + Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f + +2016-10-10 10:59 +0000 [3ab7fae96b] Badalyan Vyacheslav + + * res_pjsip_config_wizard: Memory leak in module_unload + + Fixed a memory leak. It removes only the first element. + Added a useful feature in vector.h to remove all items + under the CMP through a callback function / macro. + + ASTERISK-26453 #close + + Change-Id: I84508353463456d2495678f125738e20052da950 + +2016-09-29 12:45 +0000 [9f62feca60] Ludovic Gasc (GMLudo) + + * res_calendar: Add support for fetching calendars when reloading + + We use a lot res_calendar, we are very happy with that, especially + because you use libical, the almost alone opensource library that + supports really ical format with all types of recurrency. + + Nevertheless, some features are missed for our business use cases. + + This first patch adds a new option in calendar.conf: + fetch_again_at_reload. Be my guest for a better name. + + If it's true, when you'll launch "module reload res_calendar.so", + Asterisk will download again the calendar. + + The business use case is that we have a WebUI with a scheduler planner, + we know when the calendars are modified. + + For now, we need to define 1 minute of timeout to have a chance that + our user doesn't wait too long between the modification and the real + test. But it generates a lot of useless HTTP traffic. + + + ASTERISK-26422 #close + + Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077 + +2016-10-09 21:53 +0000 [ca2f3e5b99] Badalyan Vyacheslav + + * cel_odbc: Fix memory leak on module unload + + Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715 + +2016-10-03 11:30 +0000 [5fb848eebd] George Joseph + + * bundled_pjproject: Add tests for programs used by the Makefile, et al. + + Added tests for bzip2, tar, patch, sed and nm to configure.ac. + + Set DOWNLOAD_TO_STDOUT to a working command line regardless of + whether the download program is wget, curl or fetch. + + Added a 'configure.m4' file to the third-party directory which takes + care of calling any third-party project setup. Had to move some + pjproject_bundled stuff up in configure.ac so it was called before + the third-party configure macro. + + The pjproject tarball is now downloaded to the externals_cache_dir if + it was specified on the ./configure command line + + Removed regeneration of the pjproject aconfigure file. It was only + needed for an old patch that no longer applies. + + Converted the tests for symbols to explicit tests since we know that + they're now available in the bundled version. Saves a little time + during configure. + + ASTERISK-26416 #close + Reported-by: Corey Farrell + + Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b + (cherry picked from commit e6b0053d7561032b7adbf6f3afaecf30f5046605) + (cherry picked from commit a0d02f38322c2c4d7743504003fd376d32a133db) + +2016-10-09 18:54 +0000 [73f75c246b] Joshua Colp + + * Revert "Packet-Loss Concealment (PLC) for supporting codecs." + + This change introduced some fax test failures + that have not yet been addressed. So this is + not forgotten I'm submitting a change which + reverts it. + + This reverts: + d56fc3b36b7bb59b5506129b9895b6c3341350c9. + + ASTERISK-25629 + + Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5 + +2016-10-05 14:53 +0000 [c5e8f50169] George Joseph + + * pjproject_bundled: Add MALLOC_DEBUG capability + + pjproject_bundled will now use the asterisk memory debugging APIs + if MALLOC_DEBUG is turned on in menuselect. + + Because this required stubs for the executable programs and the python + bindings, some Makefile reorganization was needed to properly handle + the dependencies. As a result, the makefile now individually makes + each of the pjproject libraries separately instead of making them all + in 1 shot. The only visible change is that there are separate status + lines printed for each library instead oif 1 for all libs. Also, the + making of the pjproject dependency files was eliminated. They're not + needed for building unless you're actively modifying pjproject source + files and it makes the build process faster. Finally, any issues with + parallel builds should be resolved again making the build faster. + + Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0 + +2016-10-04 16:59 +0000 [442b597929] George Joseph + + * alembic: Allow cdr, config and voicemail to exist in the same schema + + cdr, config and voicemail are all separate alembic trees. Because + alembic's default is to use a table named 'alembic_version' to store + the current tree revision, the 3 trees can't exist in the same schema + without stepping on each other. + + Now each tree uses 'alembic_version_' as the version table. + Each tree's env.py script now first checks for 'alembic_version'. If + it finds it AND its revision is in the tree's history, the script + renames it to 'alembic_version_'. Regardless, the script + then continues with the migration using 'alembic_version_' + and creates that table if it's not found. The result is that if an + existing 'alembic_version' table was found but it didn't belong to this + tree, it's left alone and 'alembic_version_' is used or + created. + + WARNING: If multiple trees are using the same schema, they MUST NOT + CRU or D any objects with names that might exist in the other trees. + An example would be 'yesno_values' type. If two trees perform + operations on it, one tree could pull it out from under the other. + Thankfully we currently don't share any names among cdr, config and + voicemail. + + NOTE: Since the env.py scripts in each tree were identical, a common + env.py has been placed in the ast-db-manage directory and a symlink + to it has been placed in each tree directory. + + ASTERISK-24311 #close + Reported-by: Dafi Ni + + Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898 + +2016-10-05 04:25 +0000 [c4268ec734] Alexander Traud + + * chan_sip: Honor support of Symmetric Response (rport) for SIP requests. + + In the SIP channel driver chan_sip, the default is "auto_force_rport". When no + NAT was detected, for example in case of IPv6, Asterisk uses the IP address + from the headers within the SIP-REGISTER for subsequent SIP signaling. When + the remote party specifies support for Symmetric Response (RFC 3581) via the + parameter "rport", Asterisk should not extract the port from the SIP headers + but reuse the port of the transport. This did not happen because of a typo. + + ASTERISK-26438 #close + + Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6 + +2016-08-12 11:22 +0000 [c455823657] frahaase + + * Binaural synthesis (confbridge): interleaved two-channel audio. + + Asterisk only supports mono audio at the moment. + This patch adds interleaved two-channel audio to Asterisk's channels. + + ASTERISK-26292 + + Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a + +2016-09-16 18:54 +0000 [2a03575c30] Corey Farrell + + * astobj2: Add backtrace to log_bad_ao2. + + * Compile __ast_assert_failed unconditionally. + * Use __ast_assert_failed to log messages from log_bad_ao2 + * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run. + + Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751 + +2016-09-30 16:29 +0000 [79532bca75] Rodrigo Ramírez Norambuena + + * Add text of cdr directory into README.md for ast-db-manage + + Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636 + +2016-09-09 12:38 +0000 [806d08b675] Etienne Lessard + + * app_queue: Update dynamic members ringinuse on reload. + + Previously, when reloading the members of a queue, the members added statically + (i.e. defined in queues.conf) would see their "ringinuse" value updated but not + the members added dynamically. + + This change makes dynamic members ringuse value to be updated on reload. + + Note that it's impossible to add a dynamic member with a specific ringinuse + value. For both static and dynamic members, the ringinuse value can always be + changed later on with command like "queue set ringinuse" or with the AMI action + "QueueMemberRingInUse". So it's possible this commit could break a user workflow + if he was changing the ringinuse value of dynamic members via such commands and + was also relying on the fact that a queue reload would not update the dynamic + members ringinuse value. + + ASTERISK-26330 + + Change-Id: I3745cc9a06ba7e02c399636f1ee9e58c04081f3f + +2016-09-29 14:02 +0000 [d31ffb421c] Kevin Harwell + + * Remove "format_ogg_opus: New format" + + This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c. + + ASTERISK-26426 #close + + Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5 + +2016-09-19 04:46 +0000 [8c5c95ad89] Corey Farrell + + * core: Remove ABI effects of LOW_MEMORY. + + This allows asterisk to compiled with LOW_MEMORY to load modules built + without LOW_MEMORY. + + ASTERISK-26398 #close + + Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d + +2016-09-27 16:10 +0000 [a77ebb2017] George Joseph + + * download_externals: Fix issue with re-install + + Needed to ignore an xmlstarlet return code for optional element. + + Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873 + Reported-by: Dan Jenkins + +2016-09-27 15:35 +0000 [2d2a8944be] Corey Farrell + + * logger: Output early verbose messages to console. + + Verbose messages should be printed to the console if the sublevel is + less than option_verbose. This fix ensures the welcome message with + copyright and license are printed at daemon and interactive rasterisk + startup. + + ASTERISK-26410 #close + + Change-Id: Ia44235e30ec328aba92ea2c8a837b094e65c9a03 + +2016-09-22 09:49 +0000 [c7ef1e0af3] George Joseph + + * codec_opus: Add download ability to menuselect + + Updated codecs/codecs.xml to add codec_opus to the external + download list. + + ASTERISK-26409 + + Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4 + (cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c) + (cherry picked from commit e9684f3acd0e8def0df582c1505dd39dd3fd1610) + +2016-07-23 14:50 +0000 [5cc3c6679f] George Joseph + + * codec_opus: Replace res_format_attr_opus with the one from codec_opus + + Preparation + + ASTERISK-26409 + + Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 + (cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9) + +2016-07-23 15:56 +0000 [40aa28131b] George Joseph + + * format_ogg_opus: New format + + Add Ogg/Opus playback support. + + This uses libopusfile in order to be able to read .opus files and play + them back. + + Writing/recording support is not present at this time. + + ASTERISK-26409 + + Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955 + (cherry picked from commit daee8bbd5209b4158bc1785eede845a26e6cbeaa) + +2016-09-24 19:05 +0000 [43901e9418] George Joseph + + * build_tools: Add ability to download variants to download_externals + + Some external packages have multiple variants that apply to different + builds of asterisk. The DPMA for instance has a "bundled" variant that + needs to be downloaded if asterisk was configured with + --with-pjproject-bundled. + + There are 2 ways to specify variants: + + If you need the user to make the decision about which variant to + download, simply create multiple menuselect "member" entries like so... + + + external + xmlstarlet + bash + no + + + + external + xmlstarlet + bash + no + + + Note that the second entry has "-" appended to the name. + You can then use the existing menuselect facilities to restrict which + members to enable or disable. Youy probably don't want the user to + enable multiple at the same time. + + If you want to hide the details of the variants, the better way to + do it is to create 1 member with "variant" elements. + + + external + xmlstarlet + bash + no + + + + + + + + + + The condition must be a bash expression suitable for use with an "if" + statement. Any environment variable can be used plus those available + in makeopts. + + In this case, if asterisk was configured with --with-pjproject-bundled + the bundled variant will be automatically downloaded. Otherwise the + normal version will be downloaded. + + Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e + +2016-09-23 09:54 +0000 [5dd99465d3] Alexander Traud + + * chan_sip: Resolve externhost not to IPv6; instead go for IPv4. + + For the channel driver chan_sip, you specify externhost=example.com in sip.conf + when your Asterisk is behind a NAT and your IP address is assigned dynamically. + Or stated differently: You do not have a static IP address to use "externaddr" + directly. This NAT support is quite handy but just about IPv4. Previously, + Asterisk resolved "externhost" to any IP version. When the first DNS answer + resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and + connection (c=). This happened in outgoing SIP-REGISTER and while answering + SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an + IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". + + ASTERISK-18232 #close + Reported by: Jacek Kowalski + Tested by: Alexander Traud + patches: + changes.patch submitted by Alessandro Crespi + + Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac + +2016-09-20 09:42 +0000 [d425971009] George Joseph + + * chan_sip: Address runaway when realtime peers subscribe to mailboxes + + Users upgrading from asterisk 13.5 to a later version and who use + realtime with peers that have mailboxes were experiencing runaway + situations that manifested as a continuous stream of taskprocessor + congestion errors, memory leaks and an unresponsive chan_sip. + + A related issue was that setting rtcachefriends=no NEVER worked in + asterisk 13 (since the move to stasis). In 13.5 and earlier, when a + peer tried to register, all of the stasis threads would block and + chan_sip would again become unresponsive. After 13.5, the runaway + would happen. + + There were a number of causes... + * mwi_event_cb was (indirectly) calling build_peer even though calls to + mwi_event_cb are often caused by build_peer. + * In an effort to prevent chan_sip from being unloaded while messages + were still in flight, destroy_mailboxes was calling + stasis_unsubscribe_and_join but in some cases waited forever for the + final message. + * add_peer_mailboxes wasn't properly marking the existing mailboxes + on a peer as "keep" so build_peer would always delete them all. + * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions + then just creating them again. + + All of this was causing a flood of subscribes and unsubscribes on + multiple threads all for the same peer and mailbox. + + Fixes... + * add_peer_mailboxes now marks mailboxes correctly and build_peer only + deletes the ones that really are no longer needed by the peer. + * add_peer_mwi_subs now only adds subscriptions marked as "new" instead + of unsubscribing and resubscribing everything. It also adds the peer + object's address to the mailbox instead of its name to the subscription + userdata so mwi_event_cb doesn't have to call build_peer. + + With these changes, with rtcachefriends=yes (the most common setting), + there are no leaks, locks, loops or crashes at shutdown. + + rtcachefriends=no still causes leaks but at least it doesn't lock, loop + or crash. Since making rtcachefriends=no work wasnt in scope for this + issue, further work will have to be deferred to a separate patch. + + Side fixes... + * The ast_lock_track structure had a member named "thread" which gdb + doesn't like since it conflicts with it's "thread" command. That + member was renamed to "thread_id". + + ASTERISK-25468 #close + + Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0 + +2016-09-22 01:40 +0000 [18a8ca06eb] Aaron An + + * channels/chan_pjsip: fix HANGUPCAUSE function bug. + + HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. + This patch change the call order of ast_queue_control_data + and ast_queue_control in chan_pjsip_incoming_response. + + ASTERISK-26396 #close + Reported by: AaronAn + Tested by: AaronAn + + Change-Id: Ide2d31723d8d425961e985de7de625694580be61 + +2016-09-21 14:24 +0000 [a805d779e8] Joshua Colp + + * core: Ensure presencestate subtype and message are NULL. + + When retrieving presence state information there is no + guarantee that the subtype and message passed in are + set to NULL. This change ensures they are. + + ASTERISK-26397 #close + + Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86 + +2016-09-21 10:48 +0000 [077caf566e] Joshua Colp + + * res_odbc: Make pooling option deprecation notice more useful. + + This changes the notice for the deprecation of the old + pooling options to point to the new option for doing + pooling. This gives a clearer direction as to what to + look into. + + ASTERISK-26389 #close + + Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10 + +2016-09-21 08:46 +0000 [78b6190a11] Joshua Colp + + * odbc: Remove options that are no longer applicable. + + The pooling, shared_connection, limit, and idlecheck options + are no longer used in res_odbc. + + ASTERISK-26389 + + Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6 + +2016-08-16 15:21 +0000 [923edf2596] Corey Farrell + + * logger: Simplify ast_callid handling code. + + Routines responsible for managing ast_callid's are overly complicated. + This is left-over code from when ast_callid was an AO2 object. Now that + it is an integer the code can be reduced. + + ast_callid handler code no longer prints it's own error message upon failure + to allocate threadstorage as ast_calloc would have already printed a + message. Debug messages that were printed when TEST_FRAMEWORK was + enabled have been also been removed. + + Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e + +2016-09-20 15:17 +0000 [5cb905a227] Corey Farrell + + * core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. + + Move the function outside the conditional block that excludes + LOW_MEMORY. + + ASTERISK-26273 #close + + Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4 + +2016-09-20 09:22 +0000 [00f1d05d34] Corey Farrell + + * logger: Always enable verbose for console channel. + + Previous versions of Asterisk did not require verbose to be specified in + logger.conf for the console channel, if it was requested by command line + or asterisk.conf it just worked. This change causes Asterisk to always + enable verbose in the console channel level mask. Verbose is displayed + on consoles if requested by command line, option_verbose or 'core set + verbose'. + + This also delays initialization of the logger until after threadstorage + is initialized. Initializing too early can cause messages to be printed + multiple times to the console (stdout). + + ASTERISK-26391 #close + + Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04 + +2016-09-20 10:16 +0000 [74f562a8e2] Corey Farrell + + * logger: Fix default console settings. + + When logger.conf is missing or invalid we should be printing notices, + warnings and errors to the console. The logmask was incorrectly + calculated. + + Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3 + +2016-09-19 14:21 +0000 [0bc9912739] Walter Doekes + + * asterisk.c: Non-root users also get the astcanary after core restart. + + Without this change, a 'core restart' would kill the astcanary forever + if you're not running as root. Both with and without this patch, the + scheduling priority was still SCHED_RR after restart. + + Additionally, the astcanary is now spawned if you start with high + priority and Asterisk doesn't get a chance to lower it. For example + through: `chrt -r 10 sudo -u asterisk asterisk -c` + + Also reap killed astcanary processes on core restart. + + ASTERISK-26352 #close + + Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55 + +2016-09-19 09:40 +0000 [bffaf46690] Walter Doekes + + * asterisk.c: When astcanary dies on linux, reset priority on all threads. + + Previously only the canary checking thread itself had its priority set + to SCHED_OTHER. Now all threads are traversed and adjusted. + + ASTERISK-19867 #close + Reported by: Xavier Hienne + + Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39 + +2016-09-12 18:00 +0000 [2820b13393] Richard Mudgett + + * res_config_odbc.c: Fix buffer size limitation creating invalid SQL. + + Creating ODBC SQL queries resulted in queries too large to fit into the + supplied buffer. The resulting truncated buffer contained an invalid SQL + query. + + * Made SQL query generation code use a thread storage buffer that can + increase in size as needed. + + * Fixed bad multi-line warning messages. + + ASTERISK-26263 #close + Reported by: Jeppe Ryskov Larsen + + Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae + +2016-09-14 06:53 +0000 [0376af9519] Joshua Colp + + * rtp: Only accept the first payload for a format in SDP. + + When receiving an SDP offer with multiple payloads for + the same format we would generate an answer with the first + payload, but during the payload crossover operation + (to set the payloads for receiving) we would remove all + payloads but the last. This would result in incoming + traffic being matched against the wrong format and outgoing + traffic being sent using the wrong payload. + + This change makes it so that once a format has a payload + number put into the mapping all subsequent ones are ignored. + This ensures there is only ever one payload in the mapping + and that it is the payload placed into the answer SDP. + + ASTERISK-26365 #close + + Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60 + +2016-09-14 08:42 +0000 [9d894ee0a1] Joshua Colp + + * res_pjsip_multihomed: Change Contact port to listening port. + + The res_pjsip_multihomed module determines what interface and transport + a request is going out on and updates the SIP message accordingly with + the address information. This currently incorrectly updates the Contact + header for connectionful protocols to the ephemeral connection port, + instead of the bound address for the listening socket which can actually + accept the connection back. If the remote side attempts to connect back on + the epehemeral port it will fail. + + This change makes it so the port is updated to the bound port on + connectionful protocols and is maintained on UDP (as there can be + multiple of those). + + ASTERISK-26374 #close + + Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab + +2016-09-07 14:48 +0000 [47c527df0a] George Joseph + + * pjproject_bundled: Prevent SERVFAIL from marking name server bad + + A name server that returns "Server Failure" is indicating only that + the server couldn't process that particular request. We should NOT + assume that the name server is incapable of serving other requests. + + Here's the scenario we've been encountering... + + * 2 local name servers configured in resolv.conf. + * An OPTIONS request causes a request for A and AAAA records to go out + to both nameservers. + * The A responses both come back successfully resolved. + * Because of an issue at some upstream nameserver, the AAAA responses + for that particular query come back as "SERVFAIL" from both local + name servers. + * Both local servers are marked as bad and no further queries can be + sent until the 60 second ttl expires. Only previously cached results + can be used. + * In this case, 60 seconds is just enough time for another OPTIONS + request to go out to the same host so the cycle repeats. + + We could set the bad ttl really low but that also affects REFUSED and + NOTAUTH which probably DO signal a real server issue. Besides, even + a really low bad ttl would be an issue on a pbx. + + Although we use our own resolver in 14 and master and don't have this + issue there, Teluu has merged this patch upstream so it's appropriate + to cherry-pick to 14 and master to keep pjproject consistent. + + + Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0 + +2016-09-12 07:37 +0000 [d3ddf4b0fd] Tzafrir Cohen + + * cdr_mysql: fix UTC support + + * Make 'cdrzone=UTC' work properly. + * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone + + ASTERISK-26359 #close + + Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778 + +2016-06-27 14:26 +0000 [07b95f7c65] Tzafrir Cohen + + * sd_notify (systemd status notifications) support + + sd_notify() is used to notify systemd of changes to the status of the + process. This allows the systemd daemon to know when the process + finished loading (and thus only start another program after Asterisk has + finished loading). + + To use this, use a systemd unit with 'Type=notify' for Asterisk. + + This commit also adds the function ast_sd_notify(), a wrapper around + sd_notify that does nothing if not built with systemd support. + + Also adds support for libsystemd detection in the configure script. + + Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 + +2016-09-09 06:35 +0000 [bc81765bb4] Timo Teräs + + * Fix showing of swap details when sysinfo() is available + + If sysinfo() is available, but not sysctl() or swapctl() the + printing code for swap buffer sizes is incorrectly omitted. + The above condition happens with musl c-library. + + Fix #if rule to consider defined(HAVE_SYSINFO). And also + remove the redundant || defined(HAVE_SYSCTL) which was + incorrectly there to start with. Now swap information is + displayed only if an actual libc function to get it is + available. + + This also fixes warnings previously seen with musl libc: + + [CC] asterisk.c -> asterisk.o + asterisk.c: In function 'handle_show_sysinfo': + asterisk.c:773:6: warning: variable 'totalswap' set but not used + [-Wunused-but-set-variable] + int totalswap = 0; + ^~~~~~~~~ + asterisk.c:770:11: warning: variable 'freeswap' set but not used + [-Wunused-but-set-variable] + uint64_t freeswap = 0; + ^~~~~~~~ + + Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca + +2016-09-14 07:59 +0000 [89764f7ae9] Joshua Colp + + * rtp: Preserve timestamps on video frames. + + Currently when receiving video over RTP we store only + a calculated samples on the frame. When starting the video + it can take some time for this calculation to actually yield + a value as it requires constant changing timestamps. As well + if a video frame passes over multiple RTP packets this calculation + will fail as the timestamp is the same as the previous RTP + packet and the number of samples calculated will be 0. + + This change preserves the timestamp on the frame and allows + it to pass through the core. When sending the video this timestamp + is used instead of a new one being calculated. + + ASTERISK-26367 #close + + Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd + +2016-09-14 09:51 +0000 [5f54ac3a80] Joshua Colp + + * res_pjsip_transport_management: Convert time in log message to seconds. + + ASTERISK-26375 #close + + Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc + +2016-09-13 05:34 +0000 [6ba68b486e] Steve Davies + + * chan_sip: Fix session timeout on retransmit of non-UDP packets + + Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for + SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP + connections, allowing the TCP layer to handle the retransmits. Unfortunately, + this caused sessions to be terminated with a retransmit timeout becasue it + stopped at the point of the first retrans call. + + This patch waits for the 64*T1 timer to expire instead. + + ASTERISK-19968 + + Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204 + +2016-09-13 06:08 +0000 [e3487b9360] Joshua Colp + + * res_pjsip: Don't assume a request will have any addresses. + + When performing DNS resolution the failover code present in + res_pjsip currently assumes that a request will always have + at least one viable address. In practice this is not true. + A domain may be used that has no records. + + The code now checks that at least one address exists on the + request which prevents looping. + + ASTERISK-26364 #close + + Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c + +2016-09-12 12:25 +0000 [7d7b23f04f] Richard Mudgett + + * app_queue: Fix CLI "queue show" and AMI Queues action output truncation. + + The output of CLI "queue show" and AMI Queues action is truncated and + "failed to extend from 240 to 327" messages are generated if the queue + member and interface names are lengthy. + + * Increase the string buffer size from 240 to 512 in order to accommodate + for more information fields added to the output since v1.8. + + ASTERISK-26360 #close + Reported by: Richard Mudgett + + Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d + +2016-09-12 03:28 +0000 [740292e6ae] Walter Doekes + + * chan_sip: Allow target refresh (Contact update) on re-INVITE. + + Previously, the Contact was stored only on initial INVITE and on any + 18X and 200. That meant that after re-INVITEs from *us* the Contact + could get updated, but after re-INVITEs from the *peer*, it did not. + + This changeset fixes this inconsistency, properly allowing target + refreshes through re-INVITES (RFC3261, 12.2). + + If your strictrtp setting allows it, this change allows you to switch + the source IP of a connected/calling device mid-call with a simple + re-INVITE from the new IP. + + ASTERISK-26358 #close + + Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435 + +2016-08-31 15:22 +0000 [82ec58aa91] Richard Mudgett + + * sip_to_pjsip.py: Map legacy_useroption_parsing. + + Map the sip.conf general section legacy_useroption_parsing to the + new pjsip.conf global ignore_uri_user_options. + + ASTERISK-26316 + Reported by: Kevin Harwell + + Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc + +2016-08-29 18:08 +0000 [ba362822f3] Richard Mudgett + + * res_pjsip: Add ignore_uri_user_options option. + + This implements the chan_sip legacy_useroption_parsing option but with a + better name. + + * Made the caller-id number and redirecting number strings obtained from + incoming SIP URI user fields always truncated at the first semicolon. + People don't care about anything after the semicolon showing up on their + displays even though the RFC allows the semicolon. + + ASTERISK-26316 #close + Reported by: Kevin Harwell + + Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62 + +2016-09-09 06:26 +0000 [56caf5402c] Walter Doekes + + * contrib: Let safe_asterisk script continue without /dev/tty9. + + If you use the safe_asterisk script, it uses hardcoded defaults before + running configurable values from /etc/asterisk/startup.d. The hardcoded + default has TTY=9. Some containerized environments don't have such a + TTY, and safe_asterisk would stop. + + The custom configuration from /etc/asterisk/startup.d/* isn't read until + after it stopped, so changing TTY in a custom config did not help. + + This changeset changes safe_asterisk to continue if the TTY setting was + untouched and /dev/tty9 and /dev/vc/9 aren't found. + + Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc + +2016-09-09 05:39 +0000 [901e612739] Joshua Colp + + * res_pjsip: Only invoke unidentified endpoint logic when unidentified. + + The code was incorrectly invoking the unidentified logic when + an endpoint had actually been identified, causing log messages + to be output. + + ASTERISK-26349 #close + + Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f + +2016-08-29 22:26 +0000 [2a50c29101] Aaron An + + * res/res_pjsip: Add preferred_codec_only config to pjsip endpoint. + + This patch add config to pjsip by endpoint. + ;preferred_codec_only=yes + ; Respond to a SIP invite with the single most preferred codec + ; rather than advertising all joint codec capabilities. This + ; limits the other side's codec choice to exactly what we prefer. + + ASTERISK-26317 #close + Reported by: AaronAn + Tested by: AaronAn + + Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762 + +2016-08-16 15:34 +0000 [28b2aeba0b] Mark Michelson + + * res_pjsip: Do not crash on ACKs from unknown endpoints. + + The endpoint identification PJSIP module is intended to identify which + endpoint an incoming request is from. If an endpoint is not identified, + then an artificial endpoint is used in its place when proceeding. + + The problem is that the ACK request type is an exception to the rule. + The artificial endpoint is not used when processing an ACK. This results + in the possibility of having a NULL endpoint being used further on. + + The reason ACK is an exception is an attempt not to spam security logs + with unidentified requests. Presumably, you've already logged the + unidentified request on the preceeding INVITE. + + Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion + didn't cause an issue. A new change in 13.10 added endpoint ACL checking + shortly after endpoint identification. Because we are accessing a NULL + endpoint, this ACL check resulted in a crash. + + The fix here is to be sure to retrieve the artificial endpoint for all + request types. ACKs still do not generate unidentified request security + events. + + ASTERISK-26264 #close + Reported by nappsoft + + AST-2016-006 + + Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703 + +2016-08-23 06:35 +0000 [82a3d659dc] Joshua Colp + + * chan_sip: Don't allocate new RTP instances on top of old ones. + + In some scenarios dialog_initialize_rtp can be called multiple times on + the same dialog. This can cause RTP instances to be leaked along with + multiple file descriptors for each instance. + + This change makes it so the existing RTP instances are destroyed and + not overwritten, stopping the memory leak. + + ASTERISK-26272 #close + patches: + ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) + + Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 + +2016-09-06 11:46 +0000 [f369dbb705] Richard Mudgett + + * res_pjsip_messaging.c: Misc cleanups and fixes. + + * Eliminated RAII_VAR in get_outbound_endpoint(). + + * Simplify update_to() coding. However, this function can only be a NoOp + because the To string can only be a URI and not a name-address formatted + string. + + * Simplify update_from() coding. Also fixed a code path modifying the + from string when the caller could still want to use the original string. + + * Fixed msg_data_create() incompletely removing the "pjsip:" to then add + back the "sip:" string if needed. The code didn't handle the "pjsip:sip:" + case because it left the colon after pjsip in the string. + + Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db + +2016-09-07 16:00 +0000 [2e5da0c715] Joshua Colp + + * res_pjsip: Allow global headers to be overridden. + + Currently when you add global headers from the dialplan both + the header in the dialplan and the globally configured header + are added to the resulting SIP INVITE. This change makes it + so the headers in the dialplan take precedence and are the + only ones added. + + Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad + +2016-08-10 15:14 +0000 [ac02bbd9a0] Mark Michelson + + * ConfBridge: Make some announcements asynchronous. + + Confbridge announcements tend to block a channel while they are being + played. In some circumstances, this is warranted since you want that + particular channel not to hear the announcement (Example: "John Doe has + entered the conference"). For others it makes less sense. + + This change first introduces methods for playing sounds asynchronously + into the conference. This is very similar to how synchronous sounds are + played, except the channel initiating the playback does not wait for the + sound to complete before moving on. + + Asynchronous announcements are used for two circumstances: + * Sounds played for a user after they have left the bridge + * Sounds that play first to a single user and then the rest of the + conference (if the channel and conference use the same language) + + ASTERISK-26289 #close + Reported by Mark Michelson + + Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a + +2016-07-19 09:41 +0000 [7a12355dbd] Alexander Traud + + * chan_sip: Allow Preferred sRTP. + + Following the Encrypt-all-the-things paradigm: + + The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone + determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts + the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile + (sRTP is preferred aka optional; not mandatory). If the VoIP server does not + support sRTP and TLS, the phone shows an open padlock icon. + + This paradigm is supported by several VoIP/SIP clients on default. Some + implementations even cannot be changed to RTP/sAVP. Therefore here, this + change allows Preferred sRTP for ingress. For egress, please, create a dial + plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. + + ASTERISK-20234 #close + Reported by: tootai + Tested by: tootai, Alexander Traud + patches: + srtp_patches.diff submitted by Matt Jordan + + Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd + +2016-09-07 05:59 +0000 [baa7dba180] Joshua Colp + + * res_resolver_unbound: Fix config documentation. + + The code was referencing the config section as 'globals' + instead of 'general'. This change swaps it over to 'general'. + + Change-Id: I9dfe7788f41c4a6754c77e103880dc1a747de7fe + +2016-09-06 15:25 +0000 [e769c19a31] Matt Jordan + + * res/res_stasis_playback: Cancel the entire playlist when a stop occurs + + Prior to this patch, a stop issued by a delete of a Playback resource + (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop + the current media URI playing. Subsequent URIs specified by a playback + operation would then proceed on, even though we had just indicated to + the User that the Playback was finished *and* after they had just + 'deleted' the resource. Whoops. + + This patch corrects it by bailing out of the sequence of URIs to play if + one of them is terminated with an AST_CONTROL_STREAM_STOP indication. + + ASTERISK-26341 #close + + Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42 + +2016-08-01 20:55 +0000 [6caf6bcdad] George Joseph + + * build: Add download capability for external packages + + The DPMA and g729a, silk, siren7 and siren14 codecs hosted at + http://downloads.digium.com/pub/telephony/ are now listed in the + "External" sections of the "Resource Modules" and "Codec Translators" + pages in menuselect. Any that are selected will automatically be + downloaded and installed when "make install" is run. Their LICENSE and + README (if avaialble) files will be installed to + ASTVARLIBDIR/documentation/thirdparty/. + + Example use with codecs: + + The codecs/codecs.xml file is a menuselect style xml file that lists + the codecs to be included. Their support levels are 'external', which + triggers the download and install, and defaultenabled is no. Also + because codec_g729a is actually in a directory named codec_g729 on the + download server, the newly added 'member_data' element is used to + override the default of the directory name being the package name. You + can use the 'directory_name' attribute to keep default base URL + (http://downloads.digium.com/pub/telephony/) but use the new directory, + or you use the 'remote_url' attribute to specify a full URL to the + download directory. In this case, you must still follow the same + subdirectory naming conventions as that used for the packages located + at 'http://downloads.digium.com/pub/telephony'. + + A new configure option '--with-externals-cache' was added and like + '--with-sounds-cache' it allows the installer to cache tarballs so + they're not downloaded every time. + + To assist with the download and install process, each external package + now has a manifest.xml file that, among other things, contains a package + version and checksums for each file in the tarball. The manifest is + saved to both the cache directory and ASTMODDIR and together with the + manifest.xml on the downloads site, tells the install scripts whether + a download and/or update is needed. + + bash and xmlstarlet are required for downloader operation. If they're + not installed, the external items in menuselect will be unavailable. + + Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a + +2016-08-18 14:45 +0000 [7bb7f7b9d5] Alexei Gradinari + + * res_pjsip_session: segfault on already disconnected session + + On heavy loaded system the TCP/TLS incoming calls could be + disconnected by pjproject while these calls are being + processed by asterisk which could use the session's memory pools. + If the session in the disconnected state then the session memory + pools were already freed, so we get segfault. + + This patch adds a lifetime control on an INVITE session to pjproject. + The lifetime of the session is manipulated by calling + pjsip_inv_add_ref/pjsip_inv_dec_ref. + This patch uses these functions to inform pjproject that the + session is in use. + + This patch adds check if the session state is not disconnected + and also checks if the memory pool is not NULL. + + This patch also places tasks 'session_end' and 'session_end_completion' + into session's serializer to avoid race condition. + + ASTERISK-26291 #close + + Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7 + +2016-09-06 02:41 +0000 [d80b28560c] Walter Doekes + + * chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. + + Certain SNOM phones send so-called "optional crypto" in their SDP body. + Regular SRTP setup looks like this: + + m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 + a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... + + SNOM-style "optional crypto" looks like this: + + m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 + a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... + + A crypto line is supplied, but the m-line does not have SAVP. + + When res_srtp.so is *not* loaded, then chan_sip.so treats the optional + crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the + incoming call with the following message: + + WARNING: process_sdp: Failed to receive SDP offer/answer with + required SRTP crypto attributes for audio + + For platforms that want to start providing SRTP this presents a + compatibility problem. + + This changeset lets chan_sip handle the SDP as if no crypto-line was + supplied: i.e. accept the call as regular RTP, just like it did before + res_srtp was loaded. + + Now you'll get this informative warning instead: + + WARNING: Ignoring crypto attribute in SDP because RTP transport is + insecure + + ASTERISK-23989 #close + Reported by: Olle Johansson + + Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2 + +2016-09-03 16:04 +0000 [730cb3b0b7] Matt Jordan + + * apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option + + In any scenario in which the callee is not connected to the caller, the + current code in app_dial will crash due to raising a Dial End Stasis + Message after the callee channel has been hung up. This patch corrects + the error by simply moving the explicit hangup of the callee (peer) + channel until after the dial end message. + + ASTERISK-25691 #close + + Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d + +2016-09-03 16:02 +0000 [6e1a3b924e] Matt Jordan + + * apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 + + If the callee selects option '5' using the Dial application's privacy + (P) option, the DIALSTATUS is erroneously set to ANSWER. This option + reflects the callee sending the caller to VoiceMail one time; the call + is definitely *not* ANSWERed in such a scenario. With this patch, the + DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that + is set when the 'send to VoiceMail every time' option is set. + + ASTERISK-25691 + + Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358 + +2016-08-30 16:40 +0000 [68c7694abb] Richard Mudgett + + * res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). + + Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09 + +2016-08-29 18:06 +0000 [35ce4d25c7] Richard Mudgett + + * pjsip_configuration.c: Ignore repeated identify by methods. + + Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838 + +2016-08-30 17:26 +0000 [c1e438fdf7] Richard Mudgett + + * config_global.c: Comments and a default expression adjustment. + + Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3 + +2016-08-31 15:14 +0000 [edcf09e47c] Richard Mudgett + + * sip_to_pjsip.py: Map canreinvite as directmedia alias. + + Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2 + +2016-08-31 15:37 +0000 [47336a0bdd] Richard Mudgett + + * sip_to_pjsip.py: Fix typo converting outboundproxy registration. + + Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15 + +2016-08-31 15:13 +0000 [dba02575fc] Richard Mudgett + + * sip_to_pjsip.py: Fix comment typo and tabs. + + Change-Id: If35174614545727817d329c60ba4456c028941b5 + +2016-08-31 15:56 +0000 [4aaa27e532] Richard Mudgett + + * Sample configs: Eliminate false multiline comment block starts. + + Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6 + +2016-09-02 11:36 +0000 [c3b965a2c0] Richard Mudgett + + * format_cap.c: Fix CLI "core show channeltype Surrogate" crash. + + * Make ast_format_cap_get_names() NULL tolerant. + + ASTERISK-26331 #close + Reported by: CGI.NET + + Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3 + +2016-08-26 17:22 +0000 [e875e1c12a] Corey Farrell + + * sorcery: Create function ast_sorcery_lockable_alloc. + + Create an alternative to ast_sorcery_generic_alloc which uses astobj2 + shared locking. Use this new method for the 'struct ast_sip_aor' allocator. + + Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f + +2016-08-18 13:28 +0000 [131baf70d6] Corey Farrell + + * named_locks: Use ao2_weakproxy to deal with cleanup from container. + + This allows standard ao2 functions to be used to release references to + an ast_named_lock. This change can cause less frequent locking of the + global named_locks container. The container is no longer locked when a + named_lock reference is being release except when this causes the + named_lock to be destroyed. + + Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6 + +2016-08-26 13:18 +0000 [0c5b6e9ff5] Corey Farrell + + * astobj2: Support using a separate object for locking. + + Create ao2_alloc_with_lockobj function to support shared locking. + + Change-Id: Iba687eb9843922be7e481e23a32c0700ecf88a80 + +2016-08-31 12:23 +0000 [48fd4c815c] Michael Kuron + + * app_mp3: Use correct buffer size and the same sample rate as the channel + + Previously, the buffer used for MP3 streamed from HTTP servers had a size of + 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 + minute. Only when the buffer is full does audio start to play. + For MP3 files streamed from a server, that is usually not a big deal as long as + the connection to the server is fast enough to supply that much data within a + second or two. For MP3 live streams however, it takes 1 minute to download 1 + minute of audio, so without this change, app_mp3 wasn't really usable for MP3 + live streams. + This commit changes the buffer size so that it covers 6 seconds of an MP3 file + streamed from a server and 0.5 seconds of an MP3 live stream. The latter is + identified by the use of a .m3u file extension. + + app_mp3 so far only supported 8 kHz audio. + Now it always runs at the sample rate of the channel. + + ASTERISK-26085 #close + + Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0 + +2016-08-31 05:33 +0000 [91993ebaa5] Jean Aunis + + * resource_channels.c: add hangup reason "answered_elsewhere". + + In ARI, the channels API allows to hangup a channel with a hangup reason. + This commit adds a new reason "answered_elsewhere". + When using a SIP channel, this will eventually allow Asterisk to add a proper + "Reason" header to a CANCEL message. + + ASTERISK-26321 + + Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d + +2016-08-26 10:39 +0000 [faf9bdebb7] Alexei Gradinari + + * res_pjsip: qualify/unqualify added/deleted realtime endpoints + + If the PJSIP endpoint's AOR with the permanent contact + was deleted from the realtime storage the res_pjsip module + continues trying to qualify this contact. + The error 'Unable to find an endpoint to qualify contact' + appeares every 'qualify_frequency' seconds. + This patch deletes this contact in this case. + + The PJSIP endpoint's AOR with the permanent contact + is never qualified if it is added to realtime storage + after asterisk started. + This patch adds qualifying for the AOR's permanent contacts + on the first handling of this AOR. + + ASTERISK-26319 #close + + Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe + +2016-08-22 17:08 +0000 [c98a047ee6] Mark Michelson + + * res_pjsip: Default endpoints to the "offline" status. + + A recent change attempted to optimize startup by not updating contact + status. Instead, code responsible for qualifying contacts updates the + status as it becomes known. The code even accounts for contacts/AORs + that are not set to be qualified. + + The problem, though, is when there are no contacts associated with an + endpoint. A common case is when an endpoint is set to register its + contacts but has not done so yet. In this case, prior to registration, + the endpoint's device state will appear to be "not in use" and hints + associated with that device will appear to be "idle". In actuality, the + device state and hint should both appear as "unavailable". The reason + for the failure is that the optimization change made all persistent + endpoint states set to "unknown". + + The fix here is to change the hard-coded "unknown" to be "offline" + instead. The default state will be offline until the qualifying code + determines that the contact is actually online. This way, if there are + no contacts at all, then the state stays as offline, and device state + and hints appear correctly. + + ASTERISK-26269 #close + Reported by nappsoft + + Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a + +2016-08-29 07:07 +0000 [5e0758575c] Etienne Lessard + + * pbx.c: Prevent infinite recursion in manager_show_dialplan_helper. + + Previously, if context A was including context B and context B was including + context A, i.e. if there was a circular dependency between contexts, then + calling manager_show_dialplan_helper could lead to an infinite recursion, + resulting in a crash. + + This commit applies the same solution as the one implemented in the + show_dialplan_helper function. The manager_show_dialplan_helper and + show_dialplan_helper functions contain lots of code in common, but the former + was missing the "infinite recursion avoidance" code. + + ASTERISK-26226 #close + + Change-Id: I1aea85133c21787226f4f8442253a93000aa0897 + +2016-08-25 07:06 +0000 [c21e6764f1] Joshua Colp + + * app_queue: Ensure member is removed from pending when hanging up. + + When dialing channels it is possible that they may not ever + leave the not in use state (Local channels in particular) by + the time we cancel them. If this occurs but we know they were + dialed we explicitly remove them from the pending members + container so that subsequent call attempts occur. + + ASTERISK-26299 #close + + Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65 + +2016-08-26 14:34 +0000 [a7487e9261] George Joseph + + * pjproject_bundled: Disable srtp use by pjmedia + + The reason for the disable is that while Asterisk works fine with older + libsrtp versions, newer versions of pjproject won't compile with them. + Debian 6 for instance, has libsrtp 1.4.4 which is older than what + pjproject is expecting. + + We don't use most of pjmedia but we DO use it for SDP negotiation. + Luckily disabling srtp in pjmedia doesn't interfere with it's ability + to negitiate a secure channel. The proper crypto attributes are + negotiated in both directions. + + ASTERISK-26279 #close + + Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2 + +2016-08-26 08:41 +0000 [858fa5eb2c] Alexander Traud + + * channel: No hung-up on failing security requirements. + + In your Diaplan, if you specify + same => n,Set(CHANNEL(secure_bridge_media)=1) + same => n,Set(CHANNEL(secure_bridge_signaling)=1) + only the SIP channel driver chan_sip supports this. All other channels drivers + like res_pjsip fail. In case of failure, the original sRTP source code released + the whole channel, even if not hung-up, yet. This change does not release the + channel but instead hangs-up the channel. + + ASTERISK-26306 + + Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db + +2016-08-20 09:04 +0000 [f35501b8c9] Alexander Traud + + * sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. + + When using the migration script sip_to_pjsip.py, and your sip.conf is + configured with bindaddr=::, two transports are written to pjsip.conf, one for + 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 + and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface + like in chan_sip. + + Furthermore, the script internal functions "build_host" and "split_hostport" + did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change + makes sure, even such addresses are parsed correctly. + + ASTERISK-26309 + + Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48 + +2016-08-04 20:11 +0000 [ea929d766d] Richard Mudgett + + * res_pjsip: Cache global config options. + + We may check a global config option hundreds of times a second or more. + Asking sorcery for the global configuration from the config files backend + involves several allocations and container traversals. Using realtime + without a memory cache is a lot worse because you have to lookup in the + realtime database each time to reconstitute the sorcery object. With a + memory cache for realtime, there is about the same amount of overhead as + for config files. Either way, it is still fairly expensive to access the + sorcery object that much. + + * Cache the global config options so we can access them faster. You must + now always perform a res_pjsip reload to change the global options. + + Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7 + +2016-08-23 11:02 +0000 [5eb6cb969f] Richard Mudgett + + * res_fax: Fix deadlock in ast_channel_get_t38_state(). + + ast_channel_get_t38_state() calls ast_channel_queryoption() with + AST_OPTION_T38_STATE. If the passed in channel is a local channel then a + deadlock can happen if a channel lock is held when called. + + * Made ast_channel_get_t38_state() callers not hold a channel lock before + calling. + + * Update ast_channel_get_t38_state() doxygen to note that no channel locks + can be held when calling the function. + + ASTERISK-26203 #close + Reported by: Etienne Lessard + + ASTERISK-24822 #close + Reported by: David Brillert + + ASTERISK-22732 #close + Reported by: Richard Mudgett + + Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214 + +2016-08-23 10:39 +0000 [277a2d667a] Richard Mudgett + + * res_fax: Fix deadlock setting FAXMODE channel variable. + + ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. + Unfortunately, it also introduced a deadlock potential because + set_channel_variables() which sets FAXMODE can be called during a + masquerade. The ast_channel_get_t38_state() which gets the value used to + set FAXMODE cannot be called with the channel locked. As a result, local + channels can deadlock because of how they must acquire the locks necessary + to operate. + + The intent of FAXMODE is for dialplan to know how a fax was transferred + after the fax completes. However, the previous patch sets FAXMODE to the + channel's current T.38 state AFTER the fax has completed and where T.38 + may have already disconnected. + + * Set FAXMODE based upon T.38 negotiations exchanged either with the fax + applications or the fax framehooks. + + ASTERISK-26203 + Reported by: Etienne Lessard + + ASTERISK-24822 + Reported by: David Brillert + + ASTERISK-22732 + Reported by: Richard Mudgett + + Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1 + +2016-08-22 12:31 +0000 [edca14c8a5] Richard Mudgett + + * res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). + + fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be + called with any channel locks already held. A deadlock can happen if the + function is operating on a local channel. + + * Made fax_gateway_indicate_t38() unlock the channel before calling + ast_indicate_data() since fax_gateway_indicate_t38() is always called with + the channel locked. + + * Made fax_gateway_indicate_t38() return void since nothing cared about + its return value. + + ASTERISK-26203 + Reported by: Etienne Lessard + + ASTERISK-24822 + Reported by: David Brillert + + ASTERISK-22732 + Reported by: Richard Mudgett + + Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407 + +2016-08-23 11:16 +0000 [141cd42880] Richard Mudgett + + * res_fax.c: Add chan locked precondition comments. + + Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7 + +2016-08-23 10:42 +0000 [b86771d1bf] Richard Mudgett + + * ast_framehook_detach() must be called with the channel locked. + + The framehook container could become corrupted if the channel lock is not + held before calling. + + Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584 + +2016-08-22 15:01 +0000 [5744f434f0] Richard Mudgett + + * ast_framehook_attach() must be called with the channel locked. + + The framehook container could become corrupted if the channel lock is not + held before calling. + + Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438 + +2016-08-17 02:51 +0000 [93b7533d74] chris de rock + + * app_macro: Consider '~~s~~' as a macro start extension. + + As described in issue ASTERISK-26282 the AEL parser creates macros with + extension '~~s~~'. app_macro searches only for extension 's' so the + created extension cannot be found. with this patch app_macro searches for + both extensions and performs the right extension. + + ASTERISK-26282 #close + + Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb + +2016-08-24 04:44 +0000 [d2e03c252d] Eugene + + * chan_iax2: Set plaintext auth to deprecated as per ASTERISK-22820 + + Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth + is not supported in IAX2 protocol. Please refer to section 8.6.13 of + RFC 5456. + + But plaintext auth is still supported by Asterisk implementation of IAX2. + This support should be dropped. + + Patch, based on asterisk-dev discussion, adds deprecation warning on + startup if 'auth' is set to 'plaintext', changes default values of + 'auth' from 'md5, plaintext' to 'md5'. + + Patch is safe in terms of backwards compatibility, will work even if + remote peers have auth=plaintext and we have defaults. + + auth=plaintext setting will remain deprecated in Asterisk 14 and 15, + and IAX2 plaintext support will be removed in Asterisk 16. + + ASTERISK-22820 #close + + Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf + +2016-08-24 14:42 +0000 [e40aa40aca] George Joseph + + * res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options + + ast_multicast_rtp_create_options now checks for NULL or empty options + + Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362 + +2016-07-19 13:14 +0000 [2e79f52d71] Alexander Traud + + * codecs: Add Codec 2 mode 2400. + + ASTERISK-26217 #close + + Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6 + +2016-08-10 15:14 +0000 [ded22c712a] Mark Michelson + + * ConfBridge: Rework announcer channel methodology + + NOTE: This patch was submitted earlier and reverted because of a failing + test. The test has been patched so that it adjusts for the changes here, + so this is being resubmitted for review. + + One feature that confbridge has is the ability to play sounds to all + participants in the conference. Prior to this commit, the algorithm for + this was as follows: + + * Grab the playback lock + * Push the conference announcer channel into the bridge + * Play back the sound + * Pull the conference announcer channel from the bridge + * Release the playback lock + + The issue here is that the act of adding the playback channel to the + bridge and removing it for each announcement is expensive. Amongst the + expenses: + + * The announcer channel is imparted into the bridge, meaning a new + thread is spun up for each playback. + * When the announcer is added or removed from the bridge, it results + in the BRIDGEPEER channel variable being set on all channels in the + bridge. This requires keeping the bridge locked and locking each + individual channel in order to set it. + * There's also just the general overhead of adding the channel and + removing it from the bridge. The bridge potentially has to reconfigure + every single time + + With this commit, the paradigm for playing back announcements has + shifted. + + * The announcer channel is now added to the bridge when the conference + is allocated, and it is hung up when the conference is destroyed. + * A taskprocessor is used to queue playbacks onto the announcer channel. + This keeps the behavior from before where playbacks do not overlap. + * The announcer channel is no longer placed into the bridge as + departable. Since we are not constantly removing the channel from + the bridge, it is safe to add the channel using an independent thread + and simply hang the channel up when it is time for the conference to + be destroyed. + + The use of the taskprocessor for playbacks opens up the interesting + possibility of having asynchronous announcements played. In this commit, + however, the behavior is still exactly the same as it previously was. + + ASTERISK-26289 + Reported by Mark Michelson + + Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0 + +2016-08-23 05:54 +0000 [065d810d3f] Joshua Colp + + * Revert "ConfBridge: Rework announcer channel methodology" + + This reverts commit 5aa877305223faab5a1119276a934893ab9dc138. + + Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491 + +2016-08-19 10:21 +0000 [41ee14bfae] Alexei Gradinari + + * compilation failed with -Werror=maybe-uninitialized + + The compilation failed for devmode + --enable DONT_OPTIMIZE + --enable BETTER_BACKTRACES + --enable DO_CRASH + --enable TEST_FRAMEWORK + + res_pjsip/pjsip_configuration.c: In function dtls_handler: + res_pjsip/pjsip_configuration.c:974:20: error: + back may be used uninitialized in this function [-Werror=maybe-uninitialized] + int size = strlen(front); + ^ + cc1: all warnings being treated as errors + + Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580 + +2016-08-20 14:51 +0000 [eb0c9c476f] David M. Lee + + * res_odbc_transaction: add dep on generic_odbc + + When res_odbc_transaction depended on res_odbc, it got the generic_odbc + headers and libs implicitly. Now that it no longer depends on res_odbc, + its dependency on generic_odbc must be explicit. + + Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911 + +2016-08-20 11:18 +0000 [12752c64cc] Alexander Traud + + * pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations. + + PJProject supports a lot of platforms even Windows, some with different defaults + when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS, + "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows. + + Because of this, if configured with just an IPv6 address/transport, PJProject + listens to both IPv4 and IPv6. However, this is not supported by the PJProject + team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP, + incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only + server which accepts incoming connections on IPv4. + + If you try to configure two transports, one with IPv4 and one with IPv6 on the + same interface, as expected by the PJProject team, the IPv4 transport is not + able to bind because the IPv6 transport listens to both already. + + One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide. + Then, you are able to configure two transports, one for each IP version on the + same interface. That way, you get a server which works with IPv4 clients and + IPv6 clients at the same time over the same interface. + + Here, this change sets this parameter directly within PJProject to match the + expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack + servers out of the box like in chan_sip. This change was accepted by the + PJProject team as and is expected + to arrive in the next version, PJProject 2.6.0. Until then, this change is + incorporated in the bundled PJProject of Asterisk. + + ASTERISK-26309 + + Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae + +2016-08-19 18:19 +0000 [55ccdf93c3] Corey Farrell + + * Fix checks for allocation debugging. + + MALLOC_DEBUG should not be used to check if debugging is actually + enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only + indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it + is active. + + Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53 + +2016-08-19 14:09 +0000 [8061d9f66f] Corey Farrell + + * Fix naming mismatch of allocator functions. + + Allocator functions that take file/line/func parameters are prefixed + with single-underscore when MALLOC_DEBUG is not defined, + double-underscore when it is defined. This change updates all + allocators that accept file/line/func to have the same prototype in + either ABI mode. The parameter order of __ast_vasprintf and + __ast_asprintf in utils.h have been changed to match that of astmm.h. + + End-use allocator macro's have been removed from astmm.h and moved to an + unconditional part of utils.h. + + Change-Id: I823bb6ce2b5675b3a4735948f10a3b420e9a023a + +2016-08-17 08:10 +0000 [c1b6a79686] Torrey Searle + + * res_ari: Add http prefix to generated docs + + updated the uri handler to include the url prefix of the http server + this enables res_ari to add it to the uris when generating docs + + Change-Id: I279335a2625261a8492206c37219698f42591c2e + (cherry picked from commit 6f448f32fe9b7379e2630fab7b06205f901f2ded) + +2016-08-19 03:59 +0000 [02a82f758e] Alexander Traud + + * sip_to_pjsip: Add cert_file. + + When using the migration script sip_to_pjsip.py, cert_file was not migrated to + pjsip.conf. A previous change regarding this contained a copy/paste error. + + ASTERISK-22374 + + Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b + +2016-08-18 09:21 +0000 [1a9555f036] Alexander Traud + + * sip.conf: tlsclientmethod is using sslv23 as default. + + When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL + SSLv23_method. This was documented incorrectly in the file sip.conf.sample. + + SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method + enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that + function should have been called 'secure_method' or 'automatic_method' back in + the 90s. + + Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if + you face a server which has problems like not falling back to TLSv1.0 + automatically. + + ASTERISK-24425 + + Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3 + +2016-08-16 15:57 +0000 [53a2f7dc88] Kevin Harwell + + * res_format_attr_g729: Add annexb=no format parameter to SDPs + + Historically, Asterisk has always specified annexb=no for the g729 format. + However, when using res_pjsip no format attribute was specified. This patch + makes it so the SDP now contains a format attribute line with annexb=no. + + Note, that this means only g729a is negotiated. Even for pass through support. + According to rfc7261 the type of annex used (a or b) is dependent upon the + answerer. However, Asterisk being a back to back user agent makes this tricky + to support at this time, thus we only allow annex 'a' for now. + + ASTERISK-26228 #close + patches: + res_format_attr_g729.c submitted by Jason Parker (license 4993) + + Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0 + +2016-08-18 17:02 +0000 [7ea133f2ab] Kevin Harwell + + * rest-api: Swagger scripts were not replacing format variable in file brief + + Given resource paths did not have 'json' substituted in for the '{format}'. For + some auto generated documentation/comment strings it resulted in something like + the following: + + "... REST handler for /api-docs/sounds.{format}" + + This patch makes sure the resource api's path is properly substituted. + + ASTERISK-25472 #close + + Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23 + +2016-08-18 15:15 +0000 [c7ffd6111d] George Joseph + + * res_odbc: Correct the dependency relationship with res_odbc_transaction + + The MODULEINFO dependencies between these 2 modules was reversed. + res_odbc should depend on res_odbc_transaction, not the other way + around. + + ASTERISK-25984 #close + + Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f + +2016-08-18 12:04 +0000 [966527249e] Kevin Harwell + + * sip_to_pjsip: Set correct tls transport method + + A recent update had a copy/paste error where the unused variable 'val' was + being passed to the set_value function instead of the 'method' value itself. + + This patch passes in the right variable. + + ASTERISK-22374 + + Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06 + +2016-08-10 15:14 +0000 [5aa8773052] Mark Michelson + + * ConfBridge: Rework announcer channel methodology + + One feature that confbridge has is the ability to play sounds to all + participants in the conference. Prior to this commit, the algorithm for + this was as follows: + + * Grab the playback lock + * Push the conference announcer channel into the bridge + * Play back the sound + * Pull the conference announcer channel from the bridge + * Release the playback lock + + The issue here is that the act of adding the playback channel to the + bridge and removing it for each announcement is expensive. Amongst the + expenses: + + * The announcer channel is imparted into the bridge, meaning a new + thread is spun up for each playback. + * When the announcer is added or removed from the bridge, it results + in the BRIDGEPEER channel variable being set on all channels in the + bridge. This requires keeping the bridge locked and locking each + individual channel in order to set it. + * There's also just the general overhead of adding the channel and + removing it from the bridge. The bridge potentially has to reconfigure + every single time + + With this commit, the paradigm for playing back announcements has + shifted. + + * The announcer channel is now added to the bridge when the conference + is allocated, and it is hung up when the conference is destroyed. + * A taskprocessor is used to queue playbacks onto the announcer channel. + This keeps the behavior from before where playbacks do not overlap. + * The announcer channel is no longer placed into the bridge as + departable. Since we are not constantly removing the channel from + the bridge, it is safe to add the channel using an independent thread + and simply hang the channel up when it is time for the conference to + be destroyed. + + The use of the taskprocessor for playbacks opens up the interesting + possibility of having asynchronous announcements played. In this commit, + however, the behavior is still exactly the same as it previously was. + + ASTERISK-26289 + Reported by Mark Michelson + + Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5 + +2016-08-18 08:19 +0000 [e55d1e47aa] Alexander Traud + + * sip_to_pjsip: Map the TLS method correctly. + + When using the migration script sip_to_pjsip.py and tlsclientmethod is not set + in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to + overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is + offering/using not just TLSv1.0 but TLSv1.2 as well. + + ASTERISK-22374 + + Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f + +2016-08-18 08:17 +0000 [da14c439a3] Alexander Traud + + * sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent. + + When using the migration script sip_to_pjsip.py, no section of type=system or + type=general were created. Therefore the keys compactheaders, timerb, timert1, + and useragent were not migrated to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1 + +2016-08-18 08:16 +0000 [675721a7ab] Alexander Traud + + * sip_to_pjsip: Map (session-)timers correctly. + + When using the migration script sip_to_pjsip.py, session-timers=accept and + session-timers=refuse were mapped to wrong values. + + ASTERISK-22374 + + Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092 + +2016-08-18 08:15 +0000 [acc5237e91] Alexander Traud + + * sip_to_pjsip: Write username even without authname. + + When using the migration script sip_to_pjsip.py, now the (mandatory) username is + written to pjsip.conf, even if there was no (optional) authname in the register + string in sip.conf. + + ASTERISK-22374 + + Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f + +2016-08-18 08:14 +0000 [3eb02235f5] Alexander Traud + + * sip_to_pjsip: Parse register even with transport. + + When using the migration script sip_to_pjsip.py and the register string + started with a transport in sip.conf - like tls://... - register was not parsed + correctly and therefore not migrated correctly to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2 + +2016-08-18 08:13 +0000 [9907e2b1c1] Alexander Traud + + * sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit. + + When using the migration script sip_to_pjsip.py, those keys got missing. These + keys might appear several times and the function "merge_value" tried to collect + those. However, because these keys have different names in sip.conf and + pjsip.conf, "merge_value" was not able to find the new key name in sip.conf. + This change lets "merge_value" search with the old key name in sip.conf and + write with the new key name in pjsip.conf. + + ASTERISK-22374 + + Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2 + +2016-08-18 08:11 +0000 [c0e0075718] Alexander Traud + + * sip_to_pjsip: Map externhost/ip to Transports. + + When using the migration script sip_to_pjsip.py, the externhost or externip of + sip.conf were erroneously written to Endpoints instead to Transports. + + ASTERISK-22374 + + Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4 + +2016-08-18 08:04 +0000 [a937c2ccb1] Alexander Traud + + * sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry. + + When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and + minexpiry were not migrated to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b + +2016-08-18 08:03 +0000 [163cc2d68f] Alexander Traud + + * sip_to_pjsip: Write media_encryption. + + When using the migration script sip_to_pjsip.py, encryption=yes got missing and + media_encryption=sdes was not written to pjsip.conf, because of a typo. + + ASTERISK-22374 + + Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05 + +2016-08-18 08:02 +0000 [d8b5970749] Alexander Traud + + * sip_to_pjsip: Write cos and tos. + + When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got + missed, because of a typo. Therefore, cos and tos were not written to + pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused + by a copy-and-paste error. + + ASTERISK-22374 + + Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2 + +2016-08-18 07:55 +0000 [38491401b5] Alexander Traud + + * sip_to_pjsip: Add cert_file and ca_list_path. + + When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were + not migrated to pjsip.conf. + + ASTERISK-22374 + + Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825 + +2016-08-16 15:36 +0000 [534063fd67] George Joseph + + * res_pjsip: Add contact_user to endpoint + + contact_user, when specified on an endpoint, will override the user + portion of the Contact header on outgoing requests. + + Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4 + +2016-08-17 14:13 +0000 [0b4fa65532] Richard Mudgett + + * res_pjsip_session.c: Fix unbound srv failover tests. + + Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover + functionality if a TCP connection gets disconnected. Under these + conditions, session_inv_on_state_changed() gets a + PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new + transport. Unfortunately, session_inv_on_tsx_state_changed() also gets + the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates + the session. + + * Made session_inv_on_tsx_state_changed() complete terminating the session + on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still + PJSIP_INV_STATE_DISCONNECTED. + + ASTERISK-26305 #close + Reported by: Richard Mudgett + + Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d + +2016-08-11 12:10 +0000 [046069011b] Tzafrir Cohen + + * followme: initialize all config items on reload + + Some configuration directives were not initialized on reload, and hence + were not reset to default if they were removed from followme.conf. + + ASTERISK-26288 #close + + Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150 + +2016-08-17 06:12 +0000 [57f4e4428a] Alexander Traud + + * BuildSystem: Detect ca_list_path capabilities in external PJProject. + + Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the + bundled PJProject but not in an external PJProject. Therefore, ca_list_path + could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2 + is detected again to enable ca_list_path again. + + ASTERISK-26303 #close + + Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0 + +2016-08-16 12:24 +0000 [a5c0cf4922] George Joseph + + * ari: Add documentation that path parameters are case-sensitive + + Added to api.wiki.mustache so that the generated object pages + have the notation in the table header as well as under each method + that has path parameters. + + ASTERISK-25492 #close + + Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf + +2016-08-15 15:29 +0000 [824a4e84d1] Corey Farrell + + * Refactor usage pattern of xmldoc info tag. + + This updates func_channel.c and main/message.c to use a generic xpointer + include instead of including info from each channel driver. Now the + name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in + documentation for func_channel. Setting the name attribute of info to + MessageToInfo or MessageFromInfo causes it to be included in the + MessageSend application and AMI action. + + Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea + +2016-06-15 17:10 +0000 [957df73301] Evgeniy Tsybra + + * chan_sip: Fix lastrtprx always updated + + Packets are read regulary, when there is no data in buffer fr->frametype + is AST_FRAME_NULL. There was no check of frametype and lastrtprx always + updated and, therefore, rtptimeout did not work at all. + + ASTERISK-25270 #close + + Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d + +2016-08-10 14:41 +0000 [e85adbd947] Alexei Gradinari + + * core: Entity ID is not set or invalid + + The Exchanging Device and Mailbox States could not working + if the Entity ID (EID) is not set manually and can't be obtained + from ethernet interface. + + This patch replaces debug message to warning + and addes missing description about option 'entityid' to + asterisk.conf.sample. + + With this patch the asterisk also: + (1) decline loading the modules which won't work without EID: + res_corosync and res_pjsip_publish_asterisk. + (2) warn if EID is empty on loading next modules: + pbx_dundi, res_xmpp + + Starting with v197 systemd/udev will automatically assign "predictable" + names for all local Ethernet interfaces. + This patch also addes some new ethernet prefixes "eno" and "ens". + + ASTERISK-26164 #close + + Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6 + +2016-08-04 20:00 +0000 [13450c80ce] Richard Mudgett + + * res_sorcery_config.c: Cleanup ao2 container usage idioms. + + Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a + +2016-08-04 15:57 +0000 [d526aa5cbe] Richard Mudgett + + * sorcery.c: Minor optimizations. + + * Remove some unused parameters from internal functions: + sorcery_wizard_create() + sorcery_wizard_update() + sorcery_wizard_delete() + + * Created the struct sorcery_observer_invocation ao2 object without a lock + since it is not needed in sorcery_observer_invocation_alloc(). + + * Cleanup generic ao2 container sorcery object id hash, sort, and cmp + functions. + + Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e + +2016-08-01 11:04 +0000 [45e143576f] Richard Mudgett + + * sorcery.c: Tweak some container declaration formatting. + + * Tweak sorcery_object_type_alloc() formatting. + * Tweak ast_sorcery_init() formatting. + + Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944 + +2016-08-11 23:30 +0000 [eca3d2698a] Corey Farrell + + * pbx.c: Additional fixes to ast_context_remove_extension_callerid2. + + Do not check registrar of the first extension head. We should only check + the registrar when we match the priority. + + Additionally fix a couple calls to strcmp which used the input callerid + instead of the clean version ex.cidmatch. + + ASTERISK-26233 + + Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1 + +2016-08-13 22:02 +0000 [9202ca34a8] Matt Jordan + + * app_dial: Improve documentation + + * Add some helpful and other embedded paragraph tags + + * Document some of the lesser known channel variables set by Dial + + * Add examples for some common Dial uses, along with some more + challenging but useful options + + Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1 + +2016-08-13 20:16 +0000 [e9fe08ea37] Matt Jordan + + * manager: Add tags to relate interrelated events/actions together + + Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4 + +2016-08-13 20:15 +0000 [a93cd39ac1] Matt Jordan + + * manager: Add tags to relate Bridge related events,actions, and apps + + Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85 + +2016-08-13 20:14 +0000 [d8a7594ffd] Matt Jordan + + * manager: Add tags to relate AoC events and actions + + Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e + +2016-08-13 20:13 +0000 [243f0cf99a] Matt Jordan + + * manager: Add tags to relate UserEvent actions/apps/events + + Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4 + +2016-08-12 15:53 +0000 [3269cf4c17] Matt Jordan + + * res_agi: Improve documentation + + * Groups of AGI commands that have similar functionality now reference + each other, and all reference the AGI application for ease of wiki + reference. + + * The documentation for the AGI application has been improved, in + particular noting the various AGI types and how they are invoked. + + * A warning message has been added to DeadAGI, noting that it is + deprecated. + + Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d + +2016-08-12 13:53 +0000 [a19f4affe8] Matt Jordan + + * manager: Add links between related events + + This patch adds some see-also references between related AMI events. It + focuses primarily on those events that are guaranteed to come in pairs, + such as DTMFBegin/DTMFEnd, as well as those that occur during the life + cycle of an Asterisk channel, such as Newchannel/Hangup. + + Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3 + +2016-08-12 11:15 +0000 [ddab42e296] Matt Jordan + + * func_channel: Reorganize documentation + + * Following the example of the PJSIP channel driver, the channel + technology specific documentation has been moved to the respective + channel drivers that provide that functionality. This has the benefit + of locating the documentation of items with those modules that provide + it. + + * Examples of using the CHANNEL function for both standard items as well + as for PJSIP have been added. + + * The 'max_forwards' standard item has been documented. + + Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b + +2016-08-15 07:17 +0000 [922b74169f] Joshua Colp + + * manager: Clarify that dialplan manipulation actions are under system class. + + ASTERISK-26246 #close + + Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657 + +2016-08-11 22:12 +0000 [9debe1ca26] Corey Farrell + + * Run mandatory cleanup when startup fails. + + Errors during startup result in an exit. These error branches should be + calling ast_run_atexit(0) to ensure mandatory cleanup is run. + + ASTERISK-26267 #close + + Change-Id: If226f2326ae2df7add20040696132214cf2bb680 + +2016-08-11 11:24 +0000 [d7534e016b] George Joseph + + * res_pjsip_caller_id: Copy header name to short header name + + When compact_headers was set, we were sending a zero-length header name + for PAI and RPID because we always forced the short header name length + to 0. We did this because we cloned the header from "From" and wanted + to clear "f" from the sname. By cloning however, we bypass pjproject's + automatic logic that sets sname to name if there's no compact form of + the header, which there isn't for PAI and RPID. So now we force sname + to be the same as name right after we set name. + + res_pjsip_diversion needed the same treatment for the Diversion header. + + ASTERISK-26241 #close + + Change-Id: I633ec139630cd83809aae00336cee4a10077e467 + +2016-08-11 11:13 +0000 [225fd1003f] Matt Jordan + + * app_queue: Prevent crash when a call is forwarded to an invalid location + + When a call forward attempt is made from a Queue member, the current + code will hang up the forwarding channel in an off-nominal condition + prior to raising the Stasis events informing the rest of Asterisk that + the call was forwarded. This will result in a slew of dreaded FRACKs, + most likely leading to a crash. + + This patch modifies the code such that we don't hang up the forwarding + channel even in an off-nominal condition until we've safely raised the + Stasis messages. + + ASTERISK-25797 #close + + Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38 + +2016-08-11 12:18 +0000 [aeb859dba9] George Joseph + + * res_pjsip: Fail global load if debug or default_from_user are empty + + If debug was specified in the global configuration but left blank, + the logger would treat it as a wildcard and log all hosts. If + default_from_user was empty, a crash would result. + + The global apply handler now checks for empty strings. + + ASTERISK-26239 #close + ASTERISK-26238 #close + + Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336 + +2016-08-01 15:07 +0000 [2275494e80] Richard Mudgett + + * res_pjsip res_pjsip_mwi: Misc fixes and cleanups. + + * Eliminated RAII_VAR() usage in + ast_sip_persistent_endpoint_update_state(). + + * Added a missing allocation failure check to + persistent_endpoint_find_or_create(). + + * Made persistent_endpoint_find_or_create() create the new object without + a lock as it isn't needed. + + * Cleaned up some ao2 container allocation idioms. + + * Reordered res_pjsip_mwi.c load_module() and unload_module() + + Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8 + +2016-08-04 18:03 +0000 [d4ffbccef6] Richard Mudgett + + * location.c: Misc fixes and cleanups. + + * Eliminated most RAII_VAR() usage. + + * Added several missing allocation failure checks. + + * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without + a lock as it is not needed. + + Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc + +2016-08-11 12:01 +0000 [36b2a40533] George Joseph + + * autohints: Update CHANGES and extensions.conf.sample + + Make it clear that we're talking about device state hints and add + an entry to the sample config. + + Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433 + +2016-08-02 13:53 +0000 [4a5da6c9b4] Richard Mudgett + + * taskprocessor.c: Tweak high water checks. + + * The high water check in ast_taskprocessor_alert_set_levels() would + trigger immediately if the new high water level is zero and the queue was + empty. + + * The high water check in taskprocessor_push() was off by one. + + Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d + +2016-08-03 16:24 +0000 [5ba6357be2] Richard Mudgett + + * res_pjsip: Make aor named lock a mutex. + + The named aor lock was always being locked for writes so a rwlock adds no + benefit and may be slower because rwlocks are biased toward read locking. + + Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28 + +2016-07-29 17:41 +0000 [b6e03a5ff3] Richard Mudgett + + * pjsip_distributor.c: Add missing allocation failure check. + + Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28 + +2016-08-11 10:50 +0000 [ac0454f9fa] David M. Lee + + * Fixed compile flags for non-module libs + + The non-module libs libasteriskssl.dylib and libasteriskpj.dylib have + long been missing the AST_NOT_MODULE compile flag. This was mostly + okay, until a recent fix to improve compiler warnings when the + AST_MODULE_SELF_SYM is missing broke the build on OS X/macOS/whatever + they are calling it these days. + + Change-Id: I2cb51c890824f001280a5114f2e775f97c163516 + +2016-08-11 10:50 +0000 [b3c2f1164b] Kevin Harwell + + * alembic: add auth_username to endpoint's identify_by enum + + A new identify_by option was added recently, auth_username. However, this + setting was not added as an allowable choice in the database enumeration + value. + + This patch updates the current enumeration, adding in the new setting. + + ASTERISK-26268 #close + + Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8 + +2016-08-08 14:50 +0000 [41aba83ff6] Richard Mudgett + + * res_srtp: Move SDP SRTP code from the core to res_srtp. + + A patch made to the master branch (Now the 14 branch) inadvertently made + libsrtp a required dependency in order to compile Asterisk. Rather than + create dummy defines to substitute for the defines supplied by libsrtp + when libsrtp is not available, most of the code in sdp_srtp.c is moved + into res_srtp.c. This gets more code out of Asterisk's core that isn't + used when SRTP is not available. This also makes another inadvertent + required dependency on libsrtp by Asterisk's core unlikely. + + ASTERISK-26253 #close + Reported by: Ben Merrills + + Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7 + +2016-08-06 10:57 +0000 [820879415f] Alexei Gradinari + + * pjsip: Fix deadlock with suspend taskprocessor on masquerade + + If both channels which should be masqueraded + are in the same serializer: + 1st channel will be locked waiting condition 'complete' + 2nd channel will be locked waiting condition 'suspended' + + On heavy load system a chance that both channels will be in + the same serializer 'pjsip/distibutor' is very high. + + To reproduce compile res_pjsip/pjsip_distributor.c with + DISTRIBUTOR_POOL_SIZE=1 + + Steps to reproduce: + 1. Party A calls Party B (bridged call 'AB') + 2. Party B places Party A on hold + 3. Party B calls Voicemail app (non-bridged call 'BV') + 4. Party B attended transfers Party A to voicemail using REFER. + 5. When asterisk masquerades calls 'AB' and 'BV', + a deadlock is happened. + + This patch adds a suspension indicator to the taskprocessor. + When a session suspends/unsuspends the serializer + it sets the indicator to the appropriate state. + The session checks the suspension indicator before + suspend the serializer. + + ASTERISK-26145 #close + + Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b + +2016-08-09 12:07 +0000 [d4170df40a] Kevin Harwell + + * alembic/sqlalchemy: auto increment only allowed on a single column + + The extensions table defined two columns (id and priority) as primary key + autoincrement columns. However only one is allowed when defining the primary + key. + + This patch removes the autoincrement attribute from the priority column since + it does not need to be as such and really should not have been on there in the + first place. + + This patch also removes 'context', 'exten', and 'priority' from the primary key + index and creates a new combined unique contraint index on them. + + ASTERISK-26183 #close + + Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b + +2016-08-10 11:47 +0000 [8d42ff784d] George Joseph + + * res_resolver_unbound: Allow compilation with libunbound version < 1.5 + + libunbound at version 1.4.20 (which CentOS still uses) declared all + of their string function parameters as as 'char *'. 1.4.21 changed + them all to 'const char *'. Thankfully 1.4.21 also introduced the + UNBOUND_VERSION_MAJOR define so configure now checks for that and + sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks + that and casts away the 'const' if it's not set. + + Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and + Fedora24 (1.5.4). There are a few failing tests to be addressed though. + + ASTERISK-26283 #close + + Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148 + +2016-08-07 09:58 +0000 [c315460abb] Matt Jordan + + * channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH + + This patch adds a new PJSIP specific dialplan function, + PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media + session will be refreshed via either an UPDATE or re-INVITE request. + When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, + the formats in use on a PJSIP channel can be re-negotiated and changed + dynamically after call setup. + + ASTERISK-26277 #close + + Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b + (cherry picked from commit eec60dd77394f0519895fc6abce3a6f90f6470f1) + +2016-08-09 16:19 +0000 [8fe9f1f7f1] Mark Michelson + + * res_rtp_asterisk: Cache local RTCP address. + + When an RTCP packet is sent or received, res_rtp_asterisk generates a + Stasis event that contains the RTCP report as well as the local and + remote addresses that the report pertains to. + + The addresses are determined using ast_find_ourip(). For the local + address, this will typically result in a lookup of the hostname of the + server, and then a DNS lookup of that hostname. If you do not have the + host in /etc/hosts, then this results in a full DNS lookup, which can + potentially block for some time. + + This is especially problematic when performing RTCP reads, since those + are done on the same thread responsible for reading and writing media. + + This patch addresses the issue by performing a lookup of the local + address when RTCP is allocated. We then use this cached local address + for the Stasis events when necessary. + + ASTERISK-26280 #close + Reported by Mark Michelson + + Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556 + +2016-08-08 19:14 +0000 [827457dca0] Corey Farrell + + * Produce friendly error when AST_MODULE_SELF_SYM is not defined. + + Modules must define AST_MODULE_SELF_SYM to be used as the name of a + generated function. This produces a friendly error when it's not + defined. + + ASTERISK-26278 #close + + Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b + +2016-08-08 12:53 +0000 [403b63571c] Alexei Gradinari + + * res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack + + The PJSIP taskprocessors could be overflowed on startup + if there are many (thousands) realtime endpoints + configured with unsolicited mwi. + The PJSIP stack could be totally unresponsive for a few minutes + after boot completed. + + This patch creates a separate PJSIP serializers pool for mwi + and makes unsolicited mwi use serializers from this pool. + This patch also adds 2 new global options to tune taskprocessor + alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. + + This patch also adds new global option 'mwi_disable_initial_unsolicited' + to disable sending unsolicited mwi to all endpoints on startup. + If disabled then unsolicited mwi will start processing + on next endpoint's contact update. + + ASTERISK-26230 #close + + Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a + +2016-08-06 01:37 +0000 [0749f6e6f3] Rodrigo Ramírez Norambuena + + * res_odbc: Show only when there a fail attempt of connection in CLI + + When is executed CLI command "odbc show all" every time is show + information about variable last_negative_connect. If not there a fail + attempt of connection will show date like "1969-12-31 21:00:00". + + This patch fix there situation for to show only this information when + exists a fail attempt before. + + Change-Id: I7c058b0be6f7642e922de75ee6b82c7276c9f113 + +2016-08-05 22:06 +0000 [b156a291af] Rodrigo Ramírez Norambuena + + * cdr_adaptive_odbc: Fix DNSs mixed config quote quoted_identifiers + + When haved more than once DNSs config and one of their dont set + quoted_identifiers and before this is with configurated with + quoted_identifiers resulting a truncate statement for a reference null + for quote character identifier. + + This patch initializes quoted flag before build SQL Query + + Example config for this bugfix case in cdr_adaptive_odbc.conf file + + [first] + connection=asterisk-server1 + table=cdr + quoted_identifiers=" + + [second] + connection=asterisk-server2 + table=cdr + + [third] + connection=asterisk-server3 + table=cdr + quoted_identifiers=` + + Change-Id: Ibd95667b468e10d4a19a2b9d88b9934ec7207e1d + +2016-08-05 15:34 +0000 [9042ad40f2] Alexei Gradinari + + * app_voicemail: Add taskprocessor alert level options. + + On heavy loaded system with IMAP or DB storage, + 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. + It could happen when the IMAP or DB server dies or is unreachable. + It could happen on startup when there are many (thousands) + realtime endpoints configured with unsolicited mwi. + If the taskprocessor queue reaches the high water level + then the alert is triggered and pjsip stops processing new requests + until the queue reaches the low water level to clear the alert. + + This patch adds 2 new 'general' configuration options + to tune taskprocessor alert levels: + 'tps_queue_high' - Taskprocessor high water alert trigger level. + 'tps_queue_low' - Taskprocessor low water clear alert level + + ASTERISK-26229 #close + + Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8 + +2016-08-04 10:16 +0000 [54869e4823] Joshua Colp + + * res_pjsip_outbound_publish: Use a serializer shutdown group for unload. + + This change replaces the custom unload process for the outbound + publish module with the common serializer shutdown group. + + ASTERISK-25217 #close + + Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6 + +2016-08-04 10:27 +0000 [e711e57106] Kevin Harwell + + * resource_channels: Sync with ARI stubs + + This file was out of sync with the current ARI definitions. + + Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a + +2016-08-03 15:41 +0000 [29b0f733a0] Corey Farrell + + * Add missing checks during startup. + + This ensures startup is canceled due to allocation failures from the + following initializations. + * channel.c: ast_channels_init + * config_options.c: aco_init + + ASTERISK-26265 #close + + Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611 + +2016-08-03 09:47 +0000 [90b30b21ac] Joshua Colp + + * astconfigparser: Really handle case where line is simply a comment. + + The regular expression would match causing the code that handled + the line if it was merely a comment to never get executed. + + Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819 + +2016-08-01 11:08 +0000 [73bce50ef8] Joshua Colp + + * sorcery: Use more compatible regex for local expressions. + + This changes the use of an empty regex for both res_sorcery_config + and res_sorcery_memory to "." instead. This is a more compatible + regular expression which also works on FreeBSD. + + ASTERISK-26206 #close + + Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388 + +2016-08-02 03:08 +0000 [3ff964c6b6] Alexander Traud + + * res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. + + ASTERISK-26256 #close + + Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058 + +2016-08-01 16:13 +0000 [f6276441b1] George Joseph + + * menuselect: Add an opaque "member_data" string to the acceptable xml + + Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe + +2016-07-29 13:13 +0000 [1cd79d6ee5] Mark Michelson + + * Remove SILK payload mappings from Asterisk core. + + SILK is a bit of a hog when it comes to using up our limited number of + dynamic payload types in the RTP engine. By freeing up four slots, it + allows for other codecs to potentially take the place. + + Now, codec_silk.so will dynamically use the payload slots in the RTP + engine when it loads. + + A better fix would be make RTP dynamic payload types actually + dynamic. However, at this stage of Asterisk 14 development, this is a + risky move that would be imprudent. + + Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612 + +2016-07-29 04:48 +0000 [a7ae48441f] Joshua Colp + + * astconfigparser: Handle case where line is simply a comment. + + Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5 + +2016-07-28 14:10 +0000 [89a0a1eb45] Corey Farrell + + * pbx.c: Fix handling of '-' in extension name and callerid + + This adds a two strings to ast_exten. name to go with exten and + cidmatch_display to go with cidmatch. The new fields contain input used + to add the extension in the first place. The existing fields now + contain stripped input that excludes insignificant spaces and dashes. + These stripped fields should always be used for comparisons. The + unstripped fields should normally be used for display, but displaying + stripped values will not cause runtime errors. + + Note the actual string is only stored twice if it contains dashes. If + no dashes are found then both 'char *' fields point to the same memory. + So this change has a minimum effect on memory usage. + + The existing functions ast_get_extension_name and + ast_get_extension_cidmatch return unstripped values as they did before + this change. Other similar bugs likely still exist where unstripped + extensions are saved outside pbx.c then passed back in. + + ASTERISK-26233 #close + + Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f + +2016-07-27 17:17 +0000 [68ebf86e2f] Richard Mudgett + + * pbx.c: Allow dangerous functions when adding a hint to dialplan. + + We can allow dangerous functions when adding a hint since altering + dialplan is itself a privileged activity. Otherwise, we could never + execute dangerous functions. + + ASTERISK-25996 #close + Reported by: Andrew Nagy + + Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba + +2016-07-21 10:36 +0000 [b5bc2fdda8] Alexei Gradinari + + * pjproject: fixed a few bugs + + This patch fixes the issue in pjsip_tx_data_dec_ref() + when tx_data_destroy can be called more than once, + and checks if invalid value (e.g. NULL) is passed to. + + This patch updates array limit checks and docs + in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability(). + + Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40 + +2016-07-17 18:28 +0000 [b4f1c6380e] George Joseph + + * pjproject_bundled: Update for pjproject 2.5.5 + + Add more --disable-* switches to Makefile.rules including + --disable-opus which was causing bundled pjproject to fail with + "undefined reference" errors in libasteriskpj. + + Changed PJ_ENABLE_EXTRA_CHECK to 1. + + Removed 2 obsolete patches and added a new one. + The new one was merged by Teluu on 6/27/2016. + + ASTERISK-26148 #close + + Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 + +2016-07-27 10:33 +0000 [feb1a43412] David M. Lee + + * Portably sscanf tv_usec + + In a timeval, tv_usec is defined as a suseconds_t, which could be + different underlying types on different platforms. Instead of trying to + scanf directly into the timeval, scanf into a long int, then copy that + into the timeval. + + Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95 + +2016-07-27 12:36 +0000 [1d364ac54f] Kevin Harwell + + * rtp_engine: Failed assertion and wrong name given for codec + + Fixed an assert check that would trigger when the passed in value was negative. + The negative value was being cast to an unsigned value. This resulted in the + check failing. + + Also fixed another problem when loading formats in the engine. When setting the + mime type the format's name was being passed in instead of the codec's name. + + Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c + +2016-07-27 09:56 +0000 [8802e55c26] David M. Lee + + * Replace strdupa with more portable ast_strdupa + + The strdupa function is a GNU extension, and not widely portable. We + have an ast_strdupa function used within Asterisk which is preferred. + I pulled the definition up from menuselect.c into the menuselect.h + header file so it can be shared across menuselect. + + Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e + +2016-07-21 22:44 +0000 [737471f131] Richard Mudgett + + * dsp.c: Add fax and DTMF detection unit tests. + + * Add fax amplitude and frequency sweep tests. + * Add DTMF amplitude and twist unit tests. + + Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7 + +2016-07-21 11:56 +0000 [a8cd5d255a] Richard Mudgett + + * dsp.c: Added descriptive comments to Goertzel calculations. + + * Added doxygen to describe some struct members and what is going on in + the code. + + Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d + +2016-07-13 13:48 +0000 [6dfb34cf13] Richard Mudgett + + * dsp.c: Fix incorrect format reference typo. + + Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896 + +2016-07-25 21:18 +0000 [327136088e] Richard Mudgett + + * dsp.c: Correct DTMF twist dsp.conf documentation. + + Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae + +2016-07-22 04:43 +0000 [1e7168aee0] Joshua Colp + + * astconfigparser.py: Update with realtime fixes. + + When configuring SIP URIs in the pjsip.conf file it is + necessary to escape the semicolon so the parser does not + treat it as a comment. This change allows this to work in + the astconfigparser implementation. + + A secondary bug where some data was lost if a configuration + option included a "=" in its value was also fixed. + + A bug where sections would be considered equal despite + being different has also been fixed. + + Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8 + +2016-07-21 22:28 +0000 [49461f37b7] Richard Mudgett + + * dsp.c: Fix erroneous fax tone detection. + + The Goertzel calculations get less accurate the lower the signal level + being worked with becomes because there is less resolution remaining. + If it is too low we can erroneously detect a tone where none really + exists. The searched for fax frequencies not only need to be so much + stronger than the background noise they must also be a minimum strength. + + * Add needed minimum threshold test to tone_detect(). + + * Set TONE_THRESHOLD to allow low volume frequency spread detection. + + ASTERISK-26237 #close + Reported by: Richard Mudgett + + Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc + +2016-07-24 18:27 +0000 [b4c5dcad01] George Joseph + + * menuselect: Various menuselect enhancements + + * Add 'external' as a support level. + * Add ability for module directories to add entries to the menu + by adding members to the /.xml file. + * Expand the description field to 3 lines in the ncurses implementation. + * Allow the description field to wrap in the newt implementation. + * Add description field to the gtk implementation. + + Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 + +2016-07-24 16:51 +0000 [9db420c69d] Joshua Colp + + * ari: Update version. + + New functionality has been added so the version has been + bumped to one over the 13 version. + + Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69 + +2016-07-23 08:51 +0000 [8852a4c3db] George Joseph + + * asterisk.c: Add auto generation and persistence of UUID + + Upcoming features will require the generation and persistence + of a UUID. + + Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d + +2016-07-22 14:44 +0000 [76781a0964] Mark Michelson + + * Fix sqlalchemy error regarding identifier length. + + sqlalchemy was complaining: + + sqlalchemy.exc.IdentifierError: Identifier + 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 + characters + + This fixes the problem by changing the index name to be + "ps_contacts_qualifyfreq_exp" instead. + + ASTERISK-26227 #close + Reported by Mark Michelson + + Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9 + +2016-07-19 06:16 +0000 [9be69c1636] Alexander Traud + + * chan_sip: Enable Session-Timers for SIP over TCP (and TLS). + + Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that + scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables + Session-Timers for SIP over TCP (and for SIP over TLS). + + However with longer international calls via TCP, the SIP channel might break, + because all hops on the Internet route must stay online (have not a single power + outage, for example). Therefore with Session-Timers enabled (which are enabled + at default), you might see dropped calls. Consequently even with this change, + you might be better-off going for session-timers=refuse in your sip.conf. + + ASTERISK-19968 #close + + Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 + +2016-07-19 13:39 +0000 [8fb807009f] Alexander Traud + + * codecs: Add iLBC 20. + + Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk + defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests + this. + + ASTERISK-26218 #close + ASTERISK-26221 #close + Reported by: Aaron Meriwether + + Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa + +2016-07-15 16:16 +0000 [4286a369a1] Richard Mudgett + + * res_pjsip: Whitespace and comment cleanup. + + Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38 + +2016-07-21 22:34 +0000 [68de3a9e51] Corey Farrell + + * pbx.c: Remove duplicate code. + + Merge code found in both branches of a conditional in + ast_add_extension2_lockopt. + + The updated code initializes peer_table and peer_label_table of the + extension before linking it to the context. + + Change-Id: Ic759e27cdc9906c6877df41d28ee9c5be8f41c20 + +2016-07-21 16:35 +0000 [15bf6a87dc] George Joseph + + * Create Asterisk-14: Update CHANGES and UPGRADE files + + Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc + +2016-07-21 09:05 +0000 [1b4922466b] George Joseph + + * chan_sip: Prevent deadlock when issuing "sip show channels" + + sip_show_channels locks the dialogs container first then locks each + sip_pvt so it can spit out the details. The rest of sip dialog + processing locks the sip_pvt first then locks the dialogs container + if it needs to. Both lock in the order they need but deadlocks can + result. To fix, sip_show_channels and sip_show_channelstats have + been converted to use an iterator rather than ao2_callback. This way + the container is locked only while getting the next entry and is + unlocked when the callback is called. + + ASTERISK-23013 #close + + Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 + +2016-07-15 19:28 +0000 [a36a174c4b] Corey Farrell + + * pbx: Create pbx_sw.c for management of 'struct ast_sw'. + + This changes context switches from a linked list to a vector, makes + 'struct ast_sw' opaque to pbx.c. + + Although ast_walk_context_switches is maintained the procedure is no + longer efficient except for the first call (inc==NULL). This + functionality is replaced by two new functions implemented by vector + macros. + * ast_context_switches_count (AST_VECTOR_SIZE) + * ast_context_switches_get (AST_VECTOR_GET) + + As with ast_walk_context_switches callers of these functions are + expected to have locked contexts. Only a few places in Asterisk walked + the switches, they have been converted to use the new functions. + + Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998 + +2016-07-21 10:28 +0000 [81ea024d93] Alexei Gradinari + + * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. + + This patch removed call of pjsip_tx_data_dec_ref in send_notify + if send_request failed. + The pjsip_dlg_send_request deletes the message on error by itself. + + It seems this patch fixes next issues: + ASTERISK-26199 + ASTERISK-26166 + ASTERISK-26174 + + Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a + +2016-07-13 05:24 +0000 [1d2173c7ae] Alexander Traud + + * res_srtp: Enable AES-256 and AES-GCM. + + ASTERISK-26190 #close + + Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b + +2016-07-18 22:46 +0000 [8f6e9ffcc6] Corey Farrell + + * Add conditional support for noreturn functions. + + This adds support for tagging functions with the noreturn attribute. + If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE + and DO_CRASH are enabled then failed assertions never return. This can + resolve a large number of false positives with static analyzers. + + ASTERISK-26220 #close + + Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753 + +2016-07-19 13:18 +0000 [3d62f317dd] Richard Mudgett + + * chan_dahdi.c: Fix deadlock potential in fax redirection. + + The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to + deadlock if an incoming fax happens during the Playback or similar + application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + ASTERISK-26216 #close + Reported by: Richard Mudgett + + Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa + +2016-07-13 18:49 +0000 [db4979fa79] Richard Mudgett + + * chan_sip.c: Fix deadlock potential in fax redirection. + + The sip_read() has the potential to deadlock if an incoming fax happens + during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e + +2016-07-13 18:48 +0000 [3db468ea9e] Richard Mudgett + + * chan_pjsip.c: Fix deadlock potential in fax redirection. + + The chan_pjsip_cng_tone_detected() has the potential to deadlock if an + incoming fax happens during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5 + +2016-07-12 17:33 +0000 [9abbea162c] Richard Mudgett + + * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. + + The fax_detect_framehook() has the potential to deadlock if an incoming + fax happens during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + * Made only detach the framehook if we detected a fax and not on other + possible frames. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d + +2016-07-12 17:24 +0000 [804fbd9c2b] Richard Mudgett + + * res_fax: Fix FAXOPT(faxdetect) timeout option. + + The fax detection timeout option did not work because basically the wrong + variable was checked in fax_detect_framehook(). As a result, the timer + would timeout immediately and disable fax detection. + + * Fixed ignoring negative timeout values. We'd complain and then go right + on using the negative value. + + * Fixed destroy_faxdetect() in the off-nominal case of an incomplete + object creation. + + * Added more range checking to FAXOPT(gateway) timeout parameter. + + ASTERISK-26214 #close + Reported by: Richard Mudgett + + Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 + +2016-07-18 16:16 +0000 [0d1744e132] Richard Mudgett + + * chan_dahdi: Add faxdetect_timeout option. + + The new option allows the channel driver's faxdetect option to timeout on + a call after the specified number of seconds into a call. The new feature + is disabled if the timeout is set to zero. The option is disabled by + default. + + * Don't clear dsp_features after passing them to the dsp code in + my_pri_ss7_open_media(). We should still remember them especially for the + new faxdetect_timeout option. + + ASTERISK-26214 + Reported by: Richard Mudgett + + Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 + +2016-07-15 20:44 +0000 [e739888d99] Richard Mudgett + + * res_pjsip: Add fax_detect_timeout endpoint option. + + The new endpoint option allows the PJSIP channel driver's fax_detect + endpoint option to timeout on a call after the specified number of + seconds into a call. The new feature is disabled if the timeout is set + to zero. The option is disabled by default. + + ASTERISK-26214 + Reported by: Richard Mudgett + + Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d + +2016-07-17 07:43 +0000 [d56fc3b36b] Alexander Traud + + * translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. + + ASTERISK-25629 #close + + Change-Id: I66c0086e6c17764b8141ec60a3e2aaefe088eb78 + +2016-09-19 14:18 +0000 Asterisk Development Team + + * asterisk 14.0.0-rc1 Released. + +2016-09-19 09:17 +0000 [a23b33576f] Joshua Colp + + * Release summaries: Add summaries for 14.0.0-rc1 + +2016-09-19 09:17 +0000 [e11354b864] Joshua Colp + + * Release summaries: Remove previous versions + +2016-09-19 09:17 +0000 [24fac2271a] Joshua Colp + + * .version: Update for 14.0.0-rc1 + +2016-09-19 09:17 +0000 [52c101d441] Joshua Colp + + * .lastclean: Update for 14.0.0-rc1 + +2016-09-19 09:17 +0000 [edae56dc65] Joshua Colp + + * realtime: Add database scripts for 14.0.0-rc1 + +2016-09-14 09:51 +0000 [205e2ea351] Joshua Colp + + * res_pjsip_transport_management: Convert time in log message to seconds. + + ASTERISK-26375 #close + + Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc + +2016-09-13 06:08 +0000 [bc085bba24] Joshua Colp + + * res_pjsip: Don't assume a request will have any addresses. + + When performing DNS resolution the failover code present in + res_pjsip currently assumes that a request will always have + at least one viable address. In practice this is not true. + A domain may be used that has no records. + + The code now checks that at least one address exists on the + request which prevents looping. + + ASTERISK-26364 #close + + Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c + +2016-09-09 05:39 +0000 [9a800b24ac] Joshua Colp + + * res_pjsip: Only invoke unidentified endpoint logic when unidentified. + + The code was incorrectly invoking the unidentified logic when + an endpoint had actually been identified, causing log messages + to be output. + + ASTERISK-26349 #close + + Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f + +2016-08-16 15:34 +0000 [137aa2f13c] Mark Michelson + + * res_pjsip: Do not crash on ACKs from unknown endpoints. + + The endpoint identification PJSIP module is intended to identify which + endpoint an incoming request is from. If an endpoint is not identified, + then an artificial endpoint is used in its place when proceeding. + + The problem is that the ACK request type is an exception to the rule. + The artificial endpoint is not used when processing an ACK. This results + in the possibility of having a NULL endpoint being used further on. + + The reason ACK is an exception is an attempt not to spam security logs + with unidentified requests. Presumably, you've already logged the + unidentified request on the preceeding INVITE. + + Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion + didn't cause an issue. A new change in 13.10 added endpoint ACL checking + shortly after endpoint identification. Because we are accessing a NULL + endpoint, this ACL check resulted in a crash. + + The fix here is to be sure to retrieve the artificial endpoint for all + request types. ACKs still do not generate unidentified request security + events. + + ASTERISK-26264 #close + Reported by nappsoft + + AST-2016-006 + + Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703 + +2016-08-23 06:35 +0000 [f877e62cc9] Corey Farrell (license 5909) + + * chan_sip: Don't allocate new RTP instances on top of old ones. + + In some scenarios dialog_initialize_rtp can be called multiple times on + the same dialog. This can cause RTP instances to be leaked along with + multiple file descriptors for each instance. + + This change makes it so the existing RTP instances are destroyed and + not overwritten, stopping the memory leak. + + ASTERISK-26272 #close + patches: + ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) + + Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 + +2016-09-06 15:25 +0000 [b17ee86148] Matt Jordan + + * res/res_stasis_playback: Cancel the entire playlist when a stop occurs + + Prior to this patch, a stop issued by a delete of a Playback resource + (indicated by the control frame AST_CONTROL_STREAM_STOP) would only stop + the current media URI playing. Subsequent URIs specified by a playback + operation would then proceed on, even though we had just indicated to + the User that the Playback was finished *and* after they had just + 'deleted' the resource. Whoops. + + This patch corrects it by bailing out of the sequence of URIs to play if + one of them is terminated with an AST_CONTROL_STREAM_STOP indication. + + ASTERISK-26341 #close + + Change-Id: I2da9ec43545ba46cdfffe287c7e4907eae7fca42 + +2016-08-29 12:30 +0000 Asterisk Development Team + + * asterisk 14.0.0-beta2 Released. + +2016-08-29 07:29 +0000 [9cdf44668d] Joshua Colp + + * Release summaries: Add summaries for 14.0.0-beta2 + +2016-08-29 07:29 +0000 [73d39f2029] Joshua Colp + + * Release summaries: Remove previous versions + +2016-08-29 07:29 +0000 [e8a97775ee] Joshua Colp + + * .version: Update for 14.0.0-beta2 + +2016-08-29 07:29 +0000 [345409825a] Joshua Colp + + * .lastclean: Update for 14.0.0-beta2 + +2016-08-29 07:29 +0000 [105c1168f7] Joshua Colp + + * realtime: Add database scripts for 14.0.0-beta2 + +2016-08-29 06:31 +0000 [8927b52634] Joshua Colp + + * alembic: Fix downgrade path. + + The 3772f8f828da version was referencing a previous version + that did not exist in the 14.0 branch. It has been fixed to + reference the correct previous version. + + Change-Id: I004d0fcfdfe1d1bb6f01c6dac2b69f6b1f40ae51 + +2016-08-11 12:18 +0000 [9a95c6dea3] gtjoseph + + * res_pjsip: Fail global load if debug or default_from_user are empty + + If debug was specified in the global configuration but left blank, + the logger would treat it as a wildcard and log all hosts. If + default_from_user was empty, a crash would result. + + The global apply handler now checks for empty strings. + + ASTERISK-26239 #close + ASTERISK-26238 #close + + Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336 + +2016-08-11 11:24 +0000 [aaee8160bc] gtjoseph + + * res_pjsip_caller_id: Copy header name to short header name + + When compact_headers was set, we were sending a zero-length header name + for PAI and RPID because we always forced the short header name length + to 0. We did this because we cloned the header from "From" and wanted + to clear "f" from the sname. By cloning however, we bypass pjproject's + automatic logic that sets sname to name if there's no compact form of + the header, which there isn't for PAI and RPID. So now we force sname + to be the same as name right after we set name. + + res_pjsip_diversion needed the same treatment for the Diversion header. + + ASTERISK-26241 #close + + Change-Id: I633ec139630cd83809aae00336cee4a10077e467 + +2016-08-11 12:01 +0000 [7af0eac02a] gtjoseph + + * autohints: Update CHANGES and extensions.conf.sample + + Make it clear that we're talking about device state hints and add + an entry to the sample config. + + Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433 + +2016-08-11 10:50 +0000 [ef0bf47bb3] Kevin Harwell + + * alembic: add auth_username to endpoint's identify_by enum + + A new identify_by option was added recently, auth_username. However, this + setting was not added as an allowable choice in the database enumeration + value. + + This patch updates the current enumeration, adding in the new setting. + + ASTERISK-26268 #close + + Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8 + +2016-08-08 14:50 +0000 [a1d6b14c40] Richard Mudgett + + * res_srtp: Move SDP SRTP code from the core to res_srtp. + + A patch made to the master branch (Now the 14 branch) inadvertently made + libsrtp a required dependency in order to compile Asterisk. Rather than + create dummy defines to substitute for the defines supplied by libsrtp + when libsrtp is not available, most of the code in sdp_srtp.c is moved + into res_srtp.c. This gets more code out of Asterisk's core that isn't + used when SRTP is not available. This also makes another inadvertent + required dependency on libsrtp by Asterisk's core unlikely. + + ASTERISK-26253 #close + Reported by: Ben Merrills + + Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7 + +2016-08-09 12:07 +0000 [a783e1e60d] Kevin Harwell + + * alembic/sqlalchemy: auto increment only allowed on a single column + + The extensions table defined two columns (id and priority) as primary key + autoincrement columns. However only one is allowed when defining the primary + key. + + This patch removes the autoincrement attribute from the priority column since + it does not need to be as such and really should not have been on there in the + first place. + + This patch also removes 'context', 'exten', and 'priority' from the primary key + index and creates a new combined unique contraint index on them. + + ASTERISK-26183 #close + + Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b + +2016-08-10 11:47 +0000 [9c56f798f6] gtjoseph + + * res_resolver_unbound: Allow compilation with libunbound version < 1.5 + + libunbound at version 1.4.20 (which CentOS still uses) declared all + of their string function parameters as as 'char *'. 1.4.21 changed + them all to 'const char *'. Thankfully 1.4.21 also introduced the + UNBOUND_VERSION_MAJOR define so configure now checks for that and + sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks + that and casts away the 'const' if it's not set. + + Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and + Fedora24 (1.5.4). There are a few failing tests to be addressed though. + + ASTERISK-26283 #close + + Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148 + +2016-08-01 16:13 +0000 [1ad00c1c30] gtjoseph + + * menuselect: Add an opaque "member_data" string to the acceptable xml + + Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe + +2016-07-17 18:28 +0000 [815b6f72f8] gtjoseph + + * pjproject_bundled: Update for pjproject 2.5.5 + + Add more --disable-* switches to Makefile.rules including + --disable-opus which was causing bundled pjproject to fail with + "undefined reference" errors in libasteriskpj. + + Changed PJ_ENABLE_EXTRA_CHECK to 1. + + Removed 2 obsolete patches and added a new one. + The new one was merged by Teluu on 6/27/2016. + + ASTERISK-26148 #close + + Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 + (cherry picked from commit 4cf02b5584ce33bb0a64408c27bf20c19bc4ce13) + +2016-07-29 13:13 +0000 [c95b611a73] Mark Michelson + + * Remove SILK payload mappings from Asterisk core. + + SILK is a bit of a hog when it comes to using up our limited number of + dynamic payload types in the RTP engine. By freeing up four slots, it + allows for other codecs to potentially take the place. + + Now, codec_silk.so will dynamically use the payload slots in the RTP + engine when it loads. + + A better fix would be make RTP dynamic payload types actually + dynamic. However, at this stage of Asterisk 14 development, this is a + risky move that would be imprudent. + + Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612 + +2016-07-27 12:36 +0000 [bc94ccbcdd] Kevin Harwell + + * rtp_engine: Failed assertion and wrong name given for codec + + Fixed an assert check that would trigger when the passed in value was negative. + The negative value was being cast to an unsigned value. This resulted in the + check failing. + + Also fixed another problem when loading formats in the engine. When setting the + mime type the format's name was being passed in instead of the codec's name. + + Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c + +2016-07-26 23:19 +0000 Asterisk Development Team + + * asterisk 14.0.0-beta1 Released. + +2016-07-26 17:22 +0000 [a7233fbf3e] Mark Michelson + + * Release summaries: Add summaries for 14.0.0-beta1 + +2016-07-26 16:24 +0000 [c327430ea0] Mark Michelson + + * Release summaries: Remove previous versions + +2016-07-26 16:24 +0000 [763a18bc9d] Mark Michelson + + * .version: Update for 14.0.0-beta1 + +2016-07-26 16:24 +0000 [ce6898bd3c] Mark Michelson + + * .lastclean: Update for 14.0.0-beta1 + +2016-07-26 16:24 +0000 [ebc477aa5d] Mark Michelson + + * realtime: Add database scripts for 14.0.0-beta1 + +2016-07-26 16:00 +0000 [1838b283aa] Mark Michelson + + * ChangeLog: Updated for 14.0.0 + +2016-07-26 15:02 +0000 [f196cf975d] Mark Michelson + + * Release summaries: Add summaries for 14.0.0 + +2016-07-26 14:01 +0000 [699a7390eb] Mark Michelson + + * .version: Update for 14.0.0 + +2016-07-26 14:01 +0000 [4b17a11d7d] Mark Michelson + + * .lastclean: Update for 14.0.0 + +2016-07-26 14:01 +0000 [bb9dcae98c] Mark Michelson + + * realtime: Add database scripts for 14.0.0 + +2016-07-24 18:27 +0000 [90f445729d] gtjoseph + + * menuselect: Various menuselect enhancements + + * Add 'external' as a support level. + * Add ability for module directories to add entries to the menu + by adding members to the /.xml file. + * Expand the description field to 3 lines in the ncurses implementation. + * Allow the description field to wrap in the newt implementation. + * Add description field to the gtk implementation. + + Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 + +2016-07-24 16:51 +0000 [f75401b1e3] Joshua Colp + + * ari: Update version. + + New functionality has been added so the version has been + bumped to one over the 13 version. + + Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69 + +2016-07-23 08:51 +0000 [58759bd77c] gtjoseph + + * asterisk.c: Add auto generation and persistence of UUID + + Upcoming features will require the generation and persistence + of a UUID. + + Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d + +2016-07-22 14:44 +0000 [46b4e673ae] Mark Michelson + + * Fix sqlalchemy error regarding identifier length. + + sqlalchemy was complaining: + + sqlalchemy.exc.IdentifierError: Identifier + 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 + characters + + This fixes the problem by changing the index name to be + "ps_contacts_qualifyfreq_exp" instead. + + ASTERISK-26227 #close + Reported by Mark Michelson + + Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9 + +2016-07-22 07:01 +0000 [633c34c411] gtjoseph + + * build_tools: Update make_version for 14 + + Also remove svn stuff + + Change-Id: I95d762f7cbbe5eb01117bde8779515d51a0bb06a + +2016-07-19 13:39 +0000 [c82f24f36a] Alexander Traud + + * codecs: Add iLBC 20. + + Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk + defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests + this. + + ASTERISK-26218 #close + ASTERISK-26221 #close + Reported by: Aaron Meriwether + + Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa + +2016-07-15 16:16 +0000 [6e2e3915c8] Richard Mudgett + + * res_pjsip: Whitespace and comment cleanup. + + Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38 + +2016-07-19 13:18 +0000 [5efb5b38e8] Richard Mudgett + + * chan_dahdi.c: Fix deadlock potential in fax redirection. + + The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to + deadlock if an incoming fax happens during the Playback or similar + application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + ASTERISK-26216 #close + Reported by: Richard Mudgett + + Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa + +2016-07-13 18:49 +0000 [a1d36c89e0] Richard Mudgett + + * chan_sip.c: Fix deadlock potential in fax redirection. + + The sip_read() has the potential to deadlock if an incoming fax happens + during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e + +2016-07-13 18:48 +0000 [4dfadcb025] Richard Mudgett + + * chan_pjsip.c: Fix deadlock potential in fax redirection. + + The chan_pjsip_cng_tone_detected() has the potential to deadlock if an + incoming fax happens during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5 + +2016-07-12 17:33 +0000 [964ae54ecf] Richard Mudgett + + * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. + + The fax_detect_framehook() has the potential to deadlock if an incoming + fax happens during the Playback or similar application. + + * Fixed the potential deadlock by not calling ast_async_goto() with the + channel lock held. + + * Made always eat the fax detection frame whether there is a fax extension + or not. + + * Made only detach the framehook if we detected a fax and not on other + possible frames. + + ASTERISK-26216 + Reported by: Richard Mudgett + + Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d + +2016-07-12 17:24 +0000 [c3462adeb8] Richard Mudgett + + * res_fax: Fix FAXOPT(faxdetect) timeout option. + + The fax detection timeout option did not work because basically the wrong + variable was checked in fax_detect_framehook(). As a result, the timer + would timeout immediately and disable fax detection. + + * Fixed ignoring negative timeout values. We'd complain and then go right + on using the negative value. + + * Fixed destroy_faxdetect() in the off-nominal case of an incomplete + object creation. + + * Added more range checking to FAXOPT(gateway) timeout parameter. + + ASTERISK-26214 #close + Reported by: Richard Mudgett + + Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 + +2016-07-18 16:16 +0000 [c03e27c1c8] Richard Mudgett + + * chan_dahdi: Add faxdetect_timeout option. + + The new option allows the channel driver's faxdetect option to timeout on + a call after the specified number of seconds into a call. The new feature + is disabled if the timeout is set to zero. The option is disabled by + default. + + * Don't clear dsp_features after passing them to the dsp code in + my_pri_ss7_open_media(). We should still remember them especially for the + new faxdetect_timeout option. + + ASTERISK-26214 + Reported by: Richard Mudgett + + Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 + +2016-07-15 20:44 +0000 [d11731ac2f] Richard Mudgett + + * res_pjsip: Add fax_detect_timeout endpoint option. + + The new endpoint option allows the PJSIP channel driver's fax_detect + endpoint option to timeout on a call after the specified number of + seconds into a call. The new feature is disabled if the timeout is set + to zero. The option is disabled by default. + + ASTERISK-26214 + Reported by: Richard Mudgett + + Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d + +2016-07-21 10:28 +0000 [56b4112659] Alexei Gradinari + + * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. + + This patch removed call of pjsip_tx_data_dec_ref in send_notify + if send_request failed. + The pjsip_dlg_send_request deletes the message on error by itself. + + It seems this patch fixes next issues: + ASTERISK-26199 + ASTERISK-26166 + ASTERISK-26174 + + Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a + +2016-07-21 09:05 +0000 [52cbdf2393] gtjoseph + + * chan_sip: Prevent deadlock when issuing "sip show channels" + + sip_show_channels locks the dialogs container first then locks each + sip_pvt so it can spit out the details. The rest of sip dialog + processing locks the sip_pvt first then locks the dialogs container + if it needs to. Both lock in the order they need but deadlocks can + result. To fix, sip_show_channels and sip_show_channelstats have + been converted to use an iterator rather than ao2_callback. This way + the container is locked only while getting the next entry and is + unlocked when the callback is called. + + ASTERISK-23013 #close + + Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 + +2016-07-13 05:24 +0000 [2103ad1fec] Alexander Traud + + * res_srtp: Enable AES-256 and AES-GCM. + + ASTERISK-26190 #close + + Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b + +2016-07-18 22:46 +0000 [05cfe1a76e] Corey Farrell + + * Add conditional support for noreturn functions. + + This adds support for tagging functions with the noreturn attribute. + If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE + and DO_CRASH are enabled then failed assertions never return. This can + resolve a large number of false positives with static analyzers. + + ASTERISK-26220 #close + + Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753 + +2016-07-15 19:28 +0000 [0c88fb460f] Corey Farrell + + * pbx: Create pbx_sw.c for management of 'struct ast_sw'. + + This changes context switches from a linked list to a vector, makes + 'struct ast_sw' opaque to pbx.c. + + Although ast_walk_context_switches is maintained the procedure is no + longer efficient except for the first call (inc==NULL). This + functionality is replaced by two new functions implemented by vector + macros. + * ast_context_switches_count (AST_VECTOR_SIZE) + * ast_context_switches_get (AST_VECTOR_GET) + + As with ast_walk_context_switches callers of these functions are + expected to have locked contexts. Only a few places in Asterisk walked + the switches, they have been converted to use the new functions. + + Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998 + +2016-07-19 04:48 +0000 [6fca2b3bf0] Alexander Traud + + * Makefile: Retain XML Declaration and DTD in docs. + + Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo, + the XML Declaration and DTD were overwritten by this. + + ASTERISK-26212 #close + + Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd + +2016-07-18 18:40 +0000 [cf1188a1be] Corey Farrell + + * Unit tests: Use AST_TEST_DEFINE in conditional code only. + + If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead + code. This places all existing unit tests into a conditional block if + they weren't already. + + ASTERISK-26211 #close + + Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686 + +2016-07-18 09:22 +0000 [e9daa34261] Alexei Gradinari + + * res_pjsip_mwi: remove unneeded check on endpoint's contacts. + + The function create_mwi_subscriptions_for_endpoint checks + if there is active contacts by retrieving aors and contacts. + + This function is used to create all unsolicited mwi subscriptions + on startup and is used when contact added. + + In both cases it's not necessary to check if there are contacts. + The contacts are needed when asterisk sends mwi. + + ASTERISK-26200 #close + + Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa + +2016-07-18 05:13 +0000 [cb5e3445be] Alexander Traud + + * res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets. + + With this change, the initial RTP sequence number is randomly chosen not between + 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over + counter (ROC) synchronization is not lost for sRTP, when the very first RTP + packets get lost; see http://srtp.sourceforge.net/faq.html#Q6 + + ASTERISK-26207 #close + + Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464 + +2016-07-18 04:14 +0000 [6428580e7f] Alexander Traud + + * Makefile: Suppress echoing of target 'config' again. + + ASTERISK-26038 #close + + Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f + +2016-07-15 02:59 +0000 [e2e8713b84] Corey Farrell + + * pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'. + + This changes context ignore patterns from a linked list to a vector, + makes 'struct ast_ignorepat' opaque to pbx.c. + + Although ast_walk_context_ignorepats is maintained the procedure is no + longer efficient except for the first call (inc==NULL). This + functionality is replaced by two new functions implemented by vector + macros. + * ast_context_ignorepats_count (AST_VECTOR_SIZE) + * ast_context_ignorepats_get (AST_VECTOR_GET) + + As with ast_walk_context_ignorepats callers of these functions are + expected to have locked contexts. Only a few places in Asterisk walked + the ignorepats, they have been converted to use the new functions. + + Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a + +2016-07-14 13:51 +0000 [be36bd7ca5] Corey Farrell + + * pbx: Create pbx_include.c for management of 'struct ast_include'. + + This changes context includes from a linked list to a vector, makes + 'struct ast_include' opaque to pbx.c. + + Although ast_walk_context_includes is maintained the procedure is no + longer efficient except for the first call (inc==NULL). This + functionality is replaced by two new functions implemented by vector + macros. + * ast_context_includes_count (AST_VECTOR_SIZE) + * ast_context_includes_get (AST_VECTOR_GET) + + As with ast_walk_context_includes callers of these functions are + expected to have locked contexts. Only a few places in Asterisk walked + the includes, they have been converted to use the new functions. + + const have been applied where possible to parameters for ast_include + functions. + + Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60 + +2016-07-14 03:25 +0000 [d3348c51b5] Corey Farrell + + * features.c: Remove unneeded adsi.h include. + + adsi.h is no longer used by features.c since parking was moved to a + module. + + Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59 + +2016-06-30 15:58 +0000 [273052f404] Mark Michelson + + * Update support for SILK format. + + This commit adds scaffolding in order to support the SILK audio format + on calls. Roughly, this is what is added: + + * Cached silk formats. One for each possible sample rate. + * ast_codec structures for each possible sample rate. + * RTP payload mappings for "SILK". + + In addition, this change overhauls the res_format_attr_silk file in the + following ways: + + * The "samplerate" attribute is scrapped. That's native to the format. + * There are far more checks to ensure that attributes have been + allocated before attempting to reference them. + * We do not SDP fmtp lines for attributes set to 0. + + These changes make way to be able to install a codec_silk module and + have it actually work. It also should allow for passthrough silk calls + in Asterisk. + + Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e + +2016-07-14 07:45 +0000 [31967dacdf] Richard Miller (license 5685) + + * app_queue: Only remove queue member from pending when state changes. + + It is possible for a not in use state change to occur multiple + times causing a queue member to be removed from the pending call + container prematurely. + + The first not in use state change will remove the queue member + from the container. At this moment the member may be called and + placed in the pending container. After this another not in use + state change can be received which will remove it from the + container. Despite being called at this point the code will + incorrectly see that there are no pending calls to it. + + This change only removes it from the pending container if the + state has actually changed. + + ASTERISK-26133 #close + patches: + app_queue.diff submitted by Richard Miller (license 5685) + + Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0 + +2016-07-14 02:40 +0000 [f3608b50d7] Corey Farrell + + * pbx: Fix leak of timezone for time based includes. + + Create include_free to run ast_destroy_timing and ast_free, use that in + all places that freed an ast_include structure. This fixes a couple of + paths that previously did not run ast_destroy_timing. + + ASTERISK-26196 #close + + Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838 + +2016-07-13 17:45 +0000 [63ac4c9487] Kevin Harwell + + * translate: explicit format destination not properly set + + If the destination format's name differed from the codec name then the + translator's explict_dst field would be improperly set. In some circumstances + it would end up setting it to a newly created format that has the same name + as the codec when it actually needed to be the given destination codec. + + This could cause the translation path to use the wrong format. For instance, + if an endpoint had specified 'myulaw' as a format the translator could end up + using a 'ulaw' format (with whatever/default settings) instead. If the format + attribute settings differed between the two then there may unexpected results + during processing. + + This patch removes the name check when building the translation path. This + should make it always set the translator's explicit_dst to the given destination + format as long as the sample rate and types match. + + Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5 + +2016-07-08 11:46 +0000 [2f26512fd8] Richard Mudgett + + * stasis_endpoint.c: Fix contactstatus_to_json(). + + The roundtrip_usec json member is optional. If it isn't present then + don't put it into the converted json structure where ast_json_pack() + will choke on it. + + Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0 + +2016-07-11 10:22 +0000 [bc1ff41be7] Richard Mudgett + + * pjsip_options.c: Fix container operation. + + aor_observer_deleted() needs to operate on all contacts found for the + deleted AOR instead of only the first one found. This is really only a + problem if there is more than one contact for the AOR. + + Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1 + +2016-07-11 10:21 +0000 [eabcfeeaa3] Richard Mudgett + + * pjsip_configuration.c: Misc cleanups. + + * Fix some whitespace in various routines. + + * Rename i to iter in persistent_endpoint_update_state(). + + * Fix off-nominal copy/paste message wording in + persistent_endpoint_contact_deleted_observer() + + Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8 + +2016-07-13 13:45 +0000 [f73ddde7d4] Corey Farrell + + * chan_sip: Fix reference leak in mwi_event_cb + + Cleanup the peer reference when stasis_subscription_final_message is + true. Also free peer_name even if peer exists, after reload a new + peer_name will be allocated. + + ASTERISK-26193 #close + + Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69 + +2016-07-13 11:30 +0000 [fd54d69feb] Corey Farrell + + * threadpool: Fix leak in ast_threadpool_serializer_group error path. + + ast_threadpool_serializer_group leaks a reference to ser when listener + is allocated but tps is not. Although listener takes the reference to + ser cleanup functions are not run without tps. + + ASTERISK-26191 #close + + Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585 + +2016-06-22 07:13 +0000 [85212f2799] Eugene Voityuk ,Alexander Traud + + * res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. + + Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) + support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added + for DTLS. The source code from main/tcptls.c should have been re-used to ease + security audits. Therefore, this change rolls back the change from July 2015 and + re-uses the code from July 2014. This has the additional benefits to work under + CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. + + ASTERISK-25659 #close + Reported by: StefanEng86, urbaniak, pay123 + Tested by: sarumjanuch, traud + patches: + res_rtp_asterisk.patch submitted by sarumjanuch + dtls_centos_step_1.patch submitted by traud + dtls_centos_step_2.patch submitted by traud + + Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c + +2016-06-24 19:55 +0000 [0d487b53b1] Matt Jordan + + * res/res_pjsip_session: Check for presence of an active negotiator + + It is possible in a hypothetical situation for a session refresh to be + invoked on a PJSIP when the negotiatior on the INVITE session has not + yet been established. While this shouldn't occur with existing uses of + ast_sip_session_refresh, the crashes that occur due to improperly + calling PJSIP functions that expect a non-NULL negotiatior are + avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this + means that simply checking for the presence of the negotiator before + passing it to other PJSIP functions that use it is allowable. As such, + this patch adds checks for the presence of the negotiator before calling + PJSIP functions that assume it is non-NULL. + + Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d + +2015-10-19 18:55 +0000 [c49833653b] Matt Jordan + + * res/res_pjsip_pubsub: Add additional debug statements + + When something very sad and wrong occurs, it's challenging sometimes to + figure out why. This patch adds some additional debug statements on + off-nominal paths to try and make debugging easier. + + Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640 + +2015-10-19 18:55 +0000 [f12311ee69] Matt Jordan + + * res/res_corosync: Raise a Stasis message on node join/leave events + + When res_corosync detects that a node leaves or joins, it currently is + informed of this via Corosync callbacks. However, there are a few + limitations with the information presented: + (1) While we have information that Corosync is aware of - such as the + Corosync nodeid - that information is really only useful inside of + Corosync or res_corosync. There's no way to translate a Corosync + nodeid to some other internally useful unique identifier for the + Asterisk instance that just joined or left the cluster. + (2) While res_corosync is notified of the instance joining or leaving + the cluster, it has no mechanism to inform the Asterisk core or + other modules of this event. This limits the usefulness of res_corosync + as a heartbeat mechanism for other modules. + + This patch addresses both issues. + + First, it adds the notion of a cluster discovery message both within the + Stasis message bus, as well as the binary event messages that + res_corosync uses to transmit data back and forth within the cluster. + When Asterisk joins the cluster, it sends a discovery message to the other + nodes in the cluster, which correlates the Corosync nodeid along with + the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids + to Asterisk EIDs, such that it can map changes in cluster state with the + Asterisk instance that has that nodeid. Likewise, when an Asterisk + instance receives a discovery message from a node in the cluster, it now + sends its own discovery message back to the originating node with the + local Asterisk EID. This lets Asterisk instances within the cluster + build a complete picture of the other Asterisk instances within the + cluster. + + Second, it publishes the discovery messages onto the Stasis message bus. + Said messages are published whenever a node joins or leaves the cluster. + Interested modules can subscribe for the ast_cluster_discovery_type() + message under the ast_system_topic() and be notified when changes in + cluster state occur. + + Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465 + +2016-07-13 08:57 +0000 [a3f4141f6f] Alexander Traud + + * BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf. + + Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. + + ASTERISK-26046 #close + + Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7 + +2016-07-11 20:07 +0000 [886f2cab23] gtjoseph + + * rest_api/channels: Fix multiple issues with create and dial + + * We weren't properly subscribing to the channel and it's originator + on create. + * We weren't doing a publish_dial after calling ast_call on dial. + * We weren't calling depart_bridge when a channel left the dial bridge. + + The first 2 issues were causing events to not be generated and the third + was actually causing channels to not get properly destroyed when hung up. + + Together these 3 issues were causing the new + rest_apichannels/create_dial_bridge tests to fail. + + As a result of the fixes, the cdr state machine had to be slightly + tweaked to allow bridge leave events without asserting and the tests + themselves had to be updated to account for the channels now cleaning + themselves up. + + Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8 + +2016-07-11 10:25 +0000 [b85446d039] Richard Mudgett + + * res_pjsip: Fix statsd regression. + + The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f + patch introduced several regressions when the newly created "Updated" + state goes out for each endpoint registration refresh. + + 1) It restarted any OPTIONS RTT ping cycle. + + 2) It would interfere with a currently active ping and throw off that + ping's resulting RTT calculation. + + 3) It cleared the RTT time each time the endpoint was refreshed. + + 4) The cleared RTT time was sent out as a statsd update each time. + + 5) It created two AMI events for each update. + + * Revert the original patch and reimplement it. Now the current contact + status state is re-sent instead of the state being momentarily toggled + every time the endpoint refreshes its registration. The statsd events are + not created for the re-sent refresh because they are sent after every + OPTIONS ping. + + ASTERISK-26160 #close + Reported by: Matt Jordan + + Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1 + +2016-07-10 19:08 +0000 [4ad333bb0e] Joshua Colp + + * func_odbc: Fix connection deadlock. + + The func_odbc module was modified to ensure that the + previous behavior of using a single database connection + was maintained. This was done by getting a single database + connection and holding on to it. With the new multiple + connection support in res_odbc this will actually starve + every other thread from getting access to the database as + it also maintains the previous behavior of having only + a single database connection. + + This change disables the func_odbc specific behavior if + the res_odbc module is running with only a single database + connection active. The connection is only kept for the + duration of the request. + + ASTERISK-26177 #close + + Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f + +2016-07-12 03:50 +0000 [110b01a0bc] Alexander Traud + + * BuildSystem: Allow own CFLAGS on ./configure. + + Before this change, make failed with the error + Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH + when CFLAGS were supplied to the configure script. This was introduced with + which disabled BUILD_NATIVE when + CFLAGS were supplied. Those who need different -march= values, please, go for + ./configure + make menuselect.makeopts or make menuselect + ./menuselect/menuselect --disable BUILD_NATIVE + + ASTERISK-25289 #close + + Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc + +2016-07-11 13:42 +0000 [44f16af7cc] Richard Mudgett + + * ast_expr2: Fix off-nominal memory leak. + + Thanks to ibercom for pointing out a memory leak that was missed + in the earlier patch for the issue. + + ASTERISK-26119 + Reported by: Alexei Gradinari + + Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71 + +2016-07-11 10:17 +0000 [8476a9332f] Alexander Traud + + * install_prereq: Checkout of libSRTP 1.5.x. + + Since 5th November 2014, the master branch of libSRTP changed the prefix of + several member names and is not compatible with the source code in Asterisk + anymore. Therefore instead, this change checks out the latest version of the + libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as + backend. This makes AES-GCM and AES-IN possible. + + ASTERISK-22131 #close + + Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6 + +2016-07-09 13:32 +0000 [ad30d60c69] Corey Farrell + + * chan_sip: Fix reference leaks in error paths. + + * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. + * build_peer leaks peer on failure to allocate the endpoint. + + This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed + with an unref in the appropriate place. + + ASTERISK-26184 #close + + Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12 + +2016-07-07 12:44 +0000 [7408c51a48] Corey Farrell + + * REF_DEBUG: Prevent logging of container node objects. + + Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being + recorded to the refs log for the node being replaced. This prevents + logging of those unrefs since they would produce errors in + refcounter.py. + + ASTERISK-26181 #close + + Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4 + +2016-07-04 16:38 +0000 [c832f100d9] Alexei Gradinari + + * res_sorcery_realtime: fix bug when successful UPDATE is treated as failed + + If the SQL UPDATE statement changes nothing then SQLRowCount returns 0. + This value should be treated as success. + But the function sorcery_realtime_update treats it as failed. + + This bug was found using stress tests on PJSIP. + If there are 2 consecutive SIP REGISTER requests with the same contact data + during 1 second then res_pjsip_registrar adds contact location on 1st request + and tries to update contact location on 2nd. + The update fails and res_pjsip_registrar even removes correct contact location. + + The test "object_update_uncreated" was removed from test_sorcery_realtime.c + because it's now a valid situation. + + This patch also adds missing debug of extra SQL parameter. + + ASTERISK-26172 #close + + Change-Id: I05a7f3051455336c9dda29efc229decf86071303 + +2016-07-07 10:38 +0000 [302be4809a] Joshua Colp + + * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. + + Some T.38 implementations may send another re-invite after the initial + one which adds additional negotiation details (such as the max bitrate). + Currently this will fail when passthrough is being done in chan_sip as we + do nothing if T.38 is already active. + + Other handlers of T.38 inside of Asterisk (such as res_fax) handle this + scenario so this change adds support for it to chan_sip and res_pjsip_t38. + If a request to negotiate is received while T.38 is already enabled a + new re-INVITE is sent and negotiation is done again. + + ASTERISK-26179 #close + + Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c + +2016-07-07 10:55 +0000 [fb96492ec4] Scott Griepentrog + + * PJSIP: provide valid tcp nodelay option for reuse + + When using TCP transport with chan_pjsip, the TCP_NODELAY + option value was allocated on the stack, then passed as a + pointer to the tcp transport configuration structure, and + later re-used on subsequently created sockets when it was + no longer valid. This patch changes the allocation to be + a static. + + ASTERISK-26180 #close + Reported by: Scott Griepentrog + + Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0 + +2016-07-06 09:29 +0000 [1c949eea6c] Alexei Gradinari + + * res_pjsip: Added "subscribe_context" to endpoint + + If specified, incoming SUBSCRIBE requests will be searched for the matching + extension in the indicated context. If no "subscribe_context" is specified, + then the "context" setting is used. + + ASTERISK-25471 #close + + Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514 + +2016-07-04 05:58 +0000 [32cb981d04] Alexander Traud + + * BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf. + + Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This + avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is + using AS_HELP_STRING everywhere else already. + + ASTERISK-26046 + + Change-Id: I8299faf504ceaeee3e39930c59293809e116c631 + +2016-06-22 17:26 +0000 [9f2c007254] Richard Mudgett + + * res_pjsip_session.c: Don't send extra BYE if SDP invalid. + + When an answer SDP is invalid we were disconnecting the outgoing call and + sending two BYE requests. The first BYE was sent by PJPROJECT because of + the invalid SDP answer. The second BYE was sent by Asterisk because it + thought the canceled call was the result of the RFC5407 section 3.1.2 race + condition. + + * Made not send the BYE on a canceled session if the SDP negotiation is + incomplete because PJPROJECT has already sent a BYE for the failed + negotiation. + + ASTERISK-25772 #close + Reported by: Dmitriy Serov + + Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836 + +2016-06-27 17:19 +0000 [08d3b9a89e] Richard Mudgett + + * res_pjsip_session.c: End call on initial invalid SDP negotiation. + + When an incoming call defers SDP negotiation and then sends us an invalid + SDP in the ACK, we need to send a BYE to disconnect the call. In this + case SDP negotiation has failed and we don't have valid media streams + negotiated. + + ASTERISK-25772 + + Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8 + +2016-06-23 15:13 +0000 [e6e12c752c] Richard Mudgett + + * res_pjsip.c: Register PJMEDIA error code decoder. + + Registering the PJMEDIA error codes allows errors found when parsing an + incoming SDP to be easier to figure out. + + "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" + is much easier to understand than "Unknown error 220030". + + ASTERISK-25772 + + Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0 + +2016-06-27 16:56 +0000 [5d2fc6bab7] Richard Mudgett + + * res_pjsip_session.c: Remove unused parameter from handle_incoming(). + + Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa + +2016-06-22 18:02 +0000 [656ed73ac6] Richard Mudgett + + * res_pjsip: Add missing NULL checks when using pjsip_inv_end_session(). + + pjsip_inv_end_session() is documented as being able to return the + passed in tdata parameter set to NULL on success. + + Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047 + +2016-06-30 15:17 +0000 [4f7b859726] Richard Mudgett + + * features: Fix channel datastore access. + + Found as a result of the testsuite tests/callparking test crashing. + + Several calls to ast_get_chan_featuremap_config() and + ast_get_chan_features_xfer_config() did not lock the channel before + calling so the channel's datastore list was accessed without the lock's + protection. Apparently another thread deleted a datastore on the + channel's list while the crashing thread was walking the list. Crash at + 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result. + + * Add missing channel locks to calls that were not already protected + as the doxygen for those calls indicates. + + Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1 + +2016-06-30 08:25 +0000 [5ad7e1c09a] gtjoseph + + * configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject + + There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK + from getting set when using an external pjproject. + + ASTERISK-26099 #close + Reported-by: Ross Beer + + Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae + +2016-06-29 15:31 +0000 [dab2a6b689] Matt Jordan + + * hep.conf.sample: Default 'enabled' to 'no' + + Following the principle of least surprise, we should not be sending + massive numbers of PJSIP and RTCP HEP packets out into the ether to some + only-slightly-random IP address. Having 'enabled' set to 'no' in the + sample configuration file should prevent this from happening for those + who run 'make samples'. + + ASTERISK-26159 #close + + Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1 + +2016-06-29 15:09 +0000 [9129ac8e73] Matt Jordan + + * pjproject/patches/config_site: Increase the max number of ICE candidates + + When negotiating ICE candidates with WebRTC capable endpoints, many + networks will result in a browser offering ICE candidates that exceeds + the default number of max candidates, 16. This patch bumps the max + candidates to 32, with the max checks at twice the number of candidates. + In practice, this has shown to be sufficient for browser/WebRTC + negotiation. + + Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5 + +2016-06-28 09:00 +0000 [4045e6d8ba] gtjoseph + + * codecs: Fix ABI incompatibility created by adding format_name to ast_codec + + Adding format_name even to the end of ast_codec caused issued with + binary codec modules because the pointer would be garbage in asterisk + when they registered. So, the ast_codec structure was reverted and an + internal_ast_codec structure was created just for use in codec.c. A new + internal-only API was also added (__ast_codec_register_with_format) so + that codec_builtin could register codecs with the format_name in a + separate parameter rather than in the ast_codec structure. + + ASTERISK-26144 #close + Reported-by: Alexei Gradinari + + Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba + +2016-06-28 08:22 +0000 [651290a809] gtjoseph + + * BuildSystem: Fix a few issues hightlighted by gcc 6.x + + gcc 6.1.1 caught a few more issues. + Made sure the unit tests still pass for the func_env and stdtime + issues. + + ASTERISK-26157 #close + + Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e + +2016-06-28 10:33 +0000 [83f2c2573b] Matt Jordan + + * configs/basic-pbx/modules.conf: Remove 'bad' modules + + This patch removes the following modules: + - pbx_functions: It never existed. + - res_pjsip_log_forwarder: It no longer exists. + - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs + aren't going to be installing HOMER + - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't + loaded, and we aren't configured to make use of the + module + + Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5 + +2016-06-22 11:19 +0000 [75818b4084] Joshua Colp + + * siren: Add format attribute modules for Siren7 and Siren14. + + This change removes hardcoded SDP parsing and generation for + Siren7 and Siren14 from chan_sip and moves it to format attribute + modules so it can also be used by chan_pjsip. + + With this the fmtp lines for both are added with the bitrate + information. + + ASTERISK-26021 + + Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037 + +2016-06-23 04:33 +0000 [6e87bf746a] Alexander Traud + + * BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf. + + Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C + but requires ANSI C anyway. + + ASTERISK-26046 + + Change-Id: I914c014385e1862102d90fe7650621def78db02e + +2016-06-22 15:04 +0000 [8c7017f76e] Corey Farrell + + * res_fax: Fix reference leak in fax_v21_session_new. + + fax_v21_session_new created a session details object but only released + the allocation reference during error conditions. fax_session_new adds + it's own reference to details if needed so the caller is always + responsible for cleaning it's own reference. + + ASTERISK-26141 #close + + Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88 + +2016-06-22 14:25 +0000 [6fa3ed0679] Alexei Gradinari + + * res_pjsip: improve realtime performance #2 + + The patch removes updating all Endpoints' status on startup. + Instead, only non-qualified aors with static contact + and non-qualified non-expired contacts are retrieved from the realtime to + update the endpoint status to ONLINE. + The endpoint name was added to the contact object to simply find the endpoint + that created this contact. + + The status of endpoints with qualified aors will be updated by 'qualify' + functions. + + ASTERISK-26061 #close + + Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df + +2016-06-22 13:41 +0000 [d293ead077] gtjoseph + + * res_rtp_asterisk: Fix a self-comparison identified by gcc 6 + + gcc 6 caught a previously unidentified self-comparison in + ice_candidate_cmp. Fixed it and re-ordered the predicates for better + short-circuiting. + + ASTERISK-26140 #close + + Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7 + +2016-06-22 10:37 +0000 [c7309a5254] gtjoseph + + * chan_unistim: Fix memcpy in get_to_address + + A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD) + was using a pointer to a pointer as the destination of a memcpy and a + '&' instead of '*' in the sizeof. + + ASTERISK-26138 #close + + Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708 + +2016-06-20 13:21 +0000 [b6bd97eea2] Mark Michelson + + * Fix Alembic upgrades. + + A non-existent constraint was being referenced in the upgrade script. + This patch corrects the problem by removing the reference. + + In addition, the head of the alembic branch referred to a non-existent + revision. This has been fixed by referring to the proper revision. + + This patch fixes another realtime problem as well. Our Alembic scripts + store booleans as yes or no values. However, Sorcery tries to insert + "true" or "false" instead. This patch introduces a new boolean type that + translates to "yes" or "no" instead. + + ASTERISK-26128 #close + + Change-Id: I51574736a881189de695a824883a18d66a52dcef + +2016-06-22 10:51 +0000 [3b4f5d1345] gtjoseph + + * test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO + + Since the file was missing the depends on pjproject, it wasn't + picking up the pjproject related include path. If there was no + system installed pjproject and pjproject-bundled was used, a compile + would fail because pjsip.h wasn't found. + + ASTERISK-26139 #close + + Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757 + +2016-06-22 10:55 +0000 [5f23aacda4] Alexander Traud + + * BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf. + + Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not + support the platform SVR2 from the year 1987 anymore. + + ASTERISK-26046 + + Change-Id: I28161b037feb2d29ab46ed20e785928460226c22 + +2016-06-21 06:52 +0000 [804005d251] Torrey Searle + + * res_rtp_asterisk: fix memory leak in dtls + + ensure that cert bios get freed after creating the fingerprint + + ASTERISK-26129 #close + + Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451 + +2016-06-21 17:42 +0000 [f572b26495] Richard Mudgett + + * res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro. + + Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c + +2016-06-12 11:19 +0000 [b57cd01404] gtjoseph + + * res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription + + Occasionally under load we'll attempt to send a final NOTIFY on a + subscription that's already been terminated and a SEGV will occur + down in pjproject's evsub_destroy function. This is a result of a + race condition between all the paths that can generate a notify + and/or destroy the underlying pjproject evsub object: + + * The client can send a SUBSCRIBE with Expires: 0. + * The client can send a SUBSCRIBE/refresh. + * The subscription timer can expire. + * An extension state can change. + * An MWI event can be generated. + * The pjproject transaction timer (timer_b) can expire. + + Normally when our pubsub_on_evsub_state is called with a terminate, + we push a task to the serializer and return at which point the dialog + is unlocked. This is usually not a problem because the task runs + immediately and locks the dialog again. When the system is heavily + loaded though, there may be a delay between the unlock and relock + during which another event may occur such as the subscription timer + or timer_b expiring, an extension state change, etc. These may also + cause a terminate to be processed and if so, we could cause pjproject + to try to destroy the evsub structure twice. There's no way for us to + tell that the evsub was already destroyed and the evsub's group lock + can't tolerate this and SEGVs. + + The remedy is twofold. + + * A patch has been submitted to Teluu and added to the bundled + pjproject which adds add/decrement operations on evsub's group lock. + + * In res_pjsip_pubsub: + * configure.ac and pjproject-bundled's configure.m4 were updated + to check for the new evsub group lock APIs. + * We now add a reference to the evsub group lock when we create + the subscription and remove the reference when we clean up the + subscription. This prevents evsub from being destroyed before + we're done with it. + * A state has been added to the subscription tree structure so + termination progress can be tracked through the asyncronous tasks. + * The pubsub_on_evsub_state callback has been split so it's not doing + double duty. It now only handles the final cleanup of the + subscription tree. pubsub_on_rx_refresh now handles both client + refreshes and client terminates. It was always being called for + both anyway. + * The serialized_on_server_timeout task was removed since + serialized_pubsub_on_rx_refresh was almost identical. + * Missing state checks and ao2_cleanups were added. + * Some debug levels were adjusted to make seeing only off-nominal + things at level 1 and nominal or progress things at level 2+. + + ASTERISK-26099 #close + Reported-by: Ross Beer. + + Change-Id: I779d11802cf672a51392e62a74a1216596075ba1 + +2016-06-21 07:05 +0000 [6eb0354f2d] Alexander Traud + + * res_rtp_asterisk: Use latest DTLS version available by underlying platform. + + Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the + underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for + WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based + cipher-suites. + + ASTERISK-26130 #close + + Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0 + +2016-06-21 10:53 +0000 [596d0b0bc3] Scott Griepentrog + + * PJSIP: provide transport type with received messages + + The receipt of a SIP MESSAGE may occur over any transport including TCP + and TLS. When the message is received, the original URI is added to the + message in the field PJSIP_RECVADDR, but this is insufficient to ensure + a reply message can reach the originating endpoint. This patch adds the + PJSIP_TRANSPORT field populated with the transport type. + + ASTERISK-26132 #close + + Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e + +2016-06-21 08:01 +0000 [9e222efbf2] Alexander Traud + + * BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf. + + Some configure scripts used both AC_HELP_STRING and its replacement + AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were + changed to AS_HELP_STRING. + + ASTERISK-26046 + + Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f + +2016-06-20 10:29 +0000 [e94aae00a7] Joshua Colp + + * res_pjsip_session: Handle race condition at shutdown with timer. + + When shutting down res_pjsip_session will get unloaded before res_pjsip. + The act of unloading unregisters all the PJSIP services and sets + their module IDs to -1. In some cases it is possible for a timer to + occur after this happens which calls into res_pjsip_session. The + res_pjsip_session module can then try to get the session from the + INVITE session using the module ID. Since the module ID is now -1 + this fails. + + This change stores a copy of the module ID and uses it for the timer + callback scenario. If the module ID is -1 the callback immediately + returns but if the module ID is valid then it continues as normal. + + This works as the original ID of the module is guaranteed to still + be valid when used with the INVITE session. + + ASTERISK-26127 #close + + Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573 + +2016-06-20 12:13 +0000 [0a30008224] Richard Mudgett + + * app_voicemail.c: Fix IMAP compile error. + + Fix compile error introduced by the patch for + ASTERISK-26045 + + Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3 + +2016-06-17 13:51 +0000 [820ed3d4b3] Alexei Gradinari + + * fix: memory leaks, resource leaks, out of bounds and bugs + + ASTERISK-26119 #close + + Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c + +2016-06-13 17:40 +0000 [11caa10cf5] Mark Michelson + + * ARI: Ensure announcer channels are destroyed. + + Announcer channels were not being destroyed because the + stasis_app_control structure that referenced them was not being + destroyed. The control structure was not being destroyed because it was + not being unlinked from its container. It was not being unlinked from + its container because the after bridge callback for the announcer + channel was not being run. The after bridge callback was not being run + because the after bridge datastore was not being removed from the + channel on destruction. The channel was not being destroyed because the + hangup that used to destroy the channel was now only reducing the + reference count to one. The reference count of the channel was only + being reduced to one because the stasis_app_control structure was + holding the final reference... + + The control structure used to not keep a reference to the channel, so + that loop described above did not happen. + + The solution is to manually remove the control structure from its + container when the playback on a bridge is complete. + + ASTERISK-26083 #close + Reported by Joshua Colp + + Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4 + +2016-06-20 08:05 +0000 [f72ffc1ff9] Alexander Traud + + * http: leverage 'bindaddr' for TLS in http.conf + + The internal HTTP/WebSocket server supports both TCP and TLS, which can be + activated separately via the file http.conf. The source code intends to re-use + the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified + explicitly. This did not work because of a typo. This change resolves this typo. + + ASTERISK-26126 #close + + Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f + +2016-05-18 17:37 +0000 [3c80f84cd0] Richard Mudgett + + * res_pjsip_transport_management.c: Misc cleanups to survive shutdown. + + * In unload_module(), reordered destroying things to minimize the window + that the global transports container could be used by other threads on + shutdown. When shutting down you need to stop things in the opposite + order of creation. + + * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to + eliminate the crash potential by other threads using the container on + shutdown. + + * Made struct monitored_transport.sip_received not use + ast_atomic_fetchadd_int() since it is used as a boolean value that is only + set TRUE. It was previously incremented for every received SIP message + and could theoretically overflow. + + * In monitored_transport_state_callback(), allocated the monitored + transport object without a lock since the lock was unused. + + * In keepalive_global_loaded(), removed releasing the transports container + if the keepalive_thread could not be started. I set it up to be tried + again if the user reloads the configuration. + + Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff + +2016-01-05 19:08 +0000 [7c59f2126f] Richard Mudgett + + * res_pjsip.c: Add check that timer actually got scheduled. + + Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1 + +2016-06-13 13:33 +0000 [51cc5c31c4] Richard Mudgett + + * res_rtp_multicast.c: Fix warning message typo. + + Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3 + +2016-02-11 18:15 +0000 [3d0632a9c2] Richard Mudgett + + * res_pjsip_session.c: Reorganize ast_sip_session_terminate(). + + Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b + +2016-06-08 06:15 +0000 [ac683f13c9] Alexander Traud + + * core: Not the configured but granted number of possible file descriptors. + + With CLI "core show settings", simply the parameter maxfiles of the file + asterisk.conf was shown. If that parameter was not set, nothing was displayed + although the environment might have set a default number itself. Or if maxfiles + were not granted (completely), still maxfiles was shown. Now, the maximum number + of possible file descriptors in the environment is shown. + + ASTERISK-26097 + + Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b + +2016-06-10 10:39 +0000 [4eb8cf2684] Joshua Colp + + * translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. + + This reverts commit 5bfef2a8b4674382f959b21a3b8e14cf1d942bab as it + caused fax test failures. + + ASTERISK-25629 + + Change-Id: I79de974dc4f63a1cafe0d2509169fd9a6b3cbaf4 + +2016-06-08 06:05 +0000 [0bf1a53db3] Alexander Traud + + * astfd: With RLIMIT_NOFILE only the current value is sensible. + + With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", both the maximum max + and current max of possible file descriptors were shown. Both show the same + value always. Not to confuse users, just the current maximum is shown now. + + ASTERISK-26097 + + Change-Id: I49cf7952d73aec9e3f6a88942842c39be18380fa + +2016-06-07 18:45 +0000 [d338343dac] Joshua Colp + + * cel: Ensure only one dial status per channel exists. + + CEL wrongly assumed that a channel would only have a single dial + event on it. This is incorrect. Particularly in a queue each + call attempt to a member will result in a dial event, adding + a new dial status in CEL without removing the old one. This + would cause the container to grow with only one dial status + being removed when the channel went away. The other dial status + entries would remain leaking memory. + + This change fixes the memory leak by ensuring that only one dial + status will only ever exist for each channel. + + The behavior during the scenario where multiple events are received + has also been improved. For failure cases the first failure will + be the dial status. If an answer dial status is received, though, + it will take priority and the dial status for the channel will be + answer. + + Memory usage has also been decreased by storing the minimal + amount of information and the code has been cleaned up slightly. + + ASTERISK-25262 #close + + Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe + +2016-06-01 13:48 +0000 [1fd3a7849e] Mark Michelson + + * ARI: Ensure proper channel state on operations. + + ARI was recently outfitted with operations to create and dial channels. + This leads to the ability to try funny stuff. You could create a channel + and then immediately try to play back media on it. You could create a + channel, dial it, and while it is ringing attempt to make it continue in + the dialplan. + + This commit attempts to fix this by adding a channel state check to + operations that should not be able to operate on outbound channels that + have not yet answered. If a channel is in an invalid state, we will send + a 412 response. + + ASTERISK-26047 #close + Reported by Mark Michelson + + Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8 + +2016-06-08 11:27 +0000 [10019dc70c] Mark Michelson + + * test_http_media_cache: Fix failing test. + + The retrieve_cache_control_directives test has been failing occasionally + in Jenkins. The apparent failure occurs when attempting to validate the + expiration of the retrieved file. + + After reproducing, the problem was pretty clear. At the beginning of the + test, the current time is retrieved. The seconds value of this timestamp + is X. When the file is retrieved, res_http_media_cache calculates the + expiration and in doing so retrieves the current time. In most cases, + since the test executes quickly, it will also retrieve a timestamp with + X seconds. However, if the test starts very near to when the timestamp + seconds are set to increment, res_http_media_cache may retrieve a + timestamp with X+1 seconds instead. + + The test attempted to account for this by allowing a tolerance of 1 + second when validating the expiration. However, the problem was that the + comparisons being used in the validation used > and < operations. This + meant that values that fell within the tolerance (because they equaled + the upper bound of the tolerance) would fail. + + The solution is to use >= and <= operators in the expiration validation. + + However, I estimated that while the one second tolerance should be + fine on most machines, it would still be possible on a very slow machine + to end up falling outside the one second tolerance. So I have also + relaxed the tolerance of expiration validation to be three seconds + instead. + + The final change here is to add a debug message when validating + expiration so that we can see what values are being compared. + + ASTERISK-25959 #close + Reported by Joshua Colp + + Change-Id: Ic1a0e10722c1c5d276d5a4d6a67136d6ec26c247 + +2016-06-03 01:20 +0000 [56bdf048d2] Timo Teräs + + * Add support for OGG/Speex file format + + ASTERISK-18995 #close + + Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a + +2016-06-09 10:33 +0000 [f0855358a6] gtjoseph + + * cdr.c: Remove assert in base_process_dial_end + + Scenario: Caller blonde transfer + Bob calls Charlie who answers. + Bob puts Charlie on hold and calls Alice. + Before Alice answers, Bob transfers Charlie to Alice. + + Charlie's channel triggers an assert because he gets an "ANSWERED" + event even though he never dialed anything. With recent changes to dial + events, this is now a valid scenario so the assert needed to be removed. + + ASTERISK-26103 #close + + Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2 + +2016-06-09 10:37 +0000 [cdb7edbe7b] Mark Michelson + + * chan_pjsip: Lock channel when checking for RTP changes. + + bridge_native_rtp can call into an RTP-capable channel driver in order + for the driver to update information about who the channel is + communicating with. For SIP channel drivers, this means deactivating + RTCP and sending a reinvite so that the endpoints can communicate + directly. + + bridge_native_rtp does the right thing and has the channel locked when + calling into the channel driver. chan_pjsip can't alter session + properties in this thread, though. chan_pjsip queues a task on the + session serializer in order to update properties there. + + The problem is that this queued task was not locking the channel. This + meant that the queued task could attempt to deactivate RTCP at the same + time that the channel thread was attempting to process an incoming RTCP + packet. This could lead to a crash. + + This patch fixes the issue by locking the channel in the queued task + when altering RTP properties. + + ASTERISK-26092 #close + Reported by Niklas Larsson + + Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159 + +2016-06-03 22:44 +0000 [04ec9c745e] Richard Mudgett + + * res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. + + This patch fixes a race condition processing received REGISTER requests + and their retransmissions caused by REGISTER requests being processed by + two threads. The "sip_transaction Unable to register REGISTER transaction + (key exists)" message is a notable symptom of this issue. + + This issue was more likely to happen before the pjsip/distributor + serializers were created. Instead of steps one and two below placing the + REGISTER messages into the same pjsip/distributor they were placed in + random pjsip/default serializers. + + 1) REGISTER requests come in and get placed on the pjsip/distributor + serializer. + + 2) Before the first request is processed a retransmission comes in and is + placed on the same pjsip/distributor serializer. + + 3) The first request goes up the pjsip stack and is then shunted off to + the pjsip/aor/ serializer. + + 4) Before the first request is completed processing in the pjsip/aor/ + serializer, the second request goes up the pjsip stack and is also shunted + off to the pjsip/aor/ serializer. + + 5) The first request completes processing and sends out its response. + + 6) The second request completes processing and tries to send out its + response but pjlib complains that the REGISTER transaction key already + exists. + + 7) Sadness ensues. + + * The race is eliminated by removing the pjsip/aor/ serializer and + continuing the processing in the pjsip/distributor serializer. Now any + retransmissions queued in the pjsip/distributor serializer will be + processed after the first message is completely processed. + + ASTERISK-26088 #close + Reported by: Richard Mudgett + + Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a + +2016-06-03 11:35 +0000 [dcfef53ee2] Richard Mudgett + + * stasis: Add setting subscription congestion levels. + + Stasis subscriptions and message routers create taskprocessors to process + the event messages. API calls are needed to be able to set the congestion + levels of these taskprocessors for selected subscriptions and message + routers. + + * Updated CDR, CEL, and manager's stasis subscription congestion levels + based upon stress testing. Increased the congestion levels to reduce the + potential for bursty call setup/teardown activity from triggering the + taskprocessor overload alert. CDRs in particular need an extra high + congestion level because they can take awhile to process the stasis + messages. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: Id0a716394b4eee746dd158acc63d703902450244 + +2016-06-02 18:19 +0000 [4879cd875c] Richard Mudgett + + * sorcery: Add setting object type congestion levels. + + Sorcery creates taskprocessors for object types to process object observer + callbacks. An API call is needed to be able to set the congestion levels + of these taskprocessors for selected object types. + + * Updated PJSIP's contact and contact_status sorcery object type observer + default congestion levels based upon stress testing. Increased the + congestion levels to reduce the potential for bursty register/unregister + and subscribe/unsubscribe activity from triggering the taskprocessor + overload alert. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6 + +2016-06-02 16:08 +0000 [2cd67d5b07] Richard Mudgett + + * taskprocessors: Implement high/low water mark alerts. + + When taskprocessors get backed up, there is a good chance that we are + being overloaded and need to defer adding new work to the system. + + * Implemented a high/low water alert mechanism for modules to check if the + system is being overloaded and take appropriate action. When a + taskprocessor is created it has default congestion levels set. A + taskprocessor can later have those congestion levels altered for specific + needs if stress testing shows that the taskprocessor is a symptom of + overloading or needs to handle bursty activity without triggering an + overload alert. + + * Add CLI "core show taskprocessor" low/high water columns. + + * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was + never a good thing to use when creating a taskprocessor because of the + nature of how its references needed to be cleaned up on a partial + creation. + + * Made res_pjsip's distributor check if the taskprocessor overload alert + is active before placing a message representing brand new work onto a + distributor serializer. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I182f1be603529cd665958661c4c05ff9901825fa + +2016-05-27 17:31 +0000 [c966a035e0] Richard Mudgett + + * res_pjsip_session: Use distributor serializer for incoming calls. + + We must continue using the serializer that the original INVITE came in on + for the dialog. There may be retransmissions already enqueued in the + original serializer that can result in reentrancy and message sequencing + problems. + + Outgoing call legs create the pjsip/outsess/ serializers for + their dialogs. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc + +2016-05-27 16:28 +0000 [5b7b16a87f] Richard Mudgett + + * res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. + + * Resolves potential reentrancy problems if system restarted in the middle + of subscription message transactions. + + * Fixes memory leak recreating persistent subscriptions when the + subscription resource tree could not be created. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be + +2016-05-27 12:50 +0000 [c2ae49249c] Richard Mudgett + + * res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. + + We must continue using the serializer that the original SUBSCRIBE came in + on for the dialog. There may be retransmissions already enqueued in the + original serializer that can result in reentrancy and message sequencing + problems. The "sip_transaction Unable to register SUBSCRIBE transaction + (key exists)" message is a notable symptom of this issue. + + Outgoing subscriptions still create the pjsip/pubsub/ + serializers for their dialogs. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0 + +2016-05-26 17:35 +0000 [2ff26e9746] Richard Mudgett + + * pjsip_distributor.c: Consistently pick a serializer for messages. + + Incoming messages that are not part of a dialog or a recognized response + to one of our requests need to be sent to a consistent serializer. Under + load we may be queueing retransmissions before we can process the original + message. We don't need to throw these messages onto random serializers + and cause reentrancy and message sequencing problems. + + * Created a pool of pjsip/distributor serializers that get picked by + hashing the call-id and remote tag strings of the received messages. + + * Made ast_sip_destroy_distributor() destroy items in the reverse order of + creation. + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I2ce769389fc060d9f379977f559026fbcb632407 + +2016-06-02 12:51 +0000 [df2791da8f] Richard Mudgett + + * pjsip_distributor.c: Ignore messages until fully booted. + + We should not be processing any incoming messages until we are fully + booted. We may not have dialplan or other needed configuration loaded + yet. + + ASTERISK-26089 #close + Reported by: Scott Griepentrog + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264 + +2016-06-09 09:20 +0000 [d21a77b325] gtjoseph + + * build: Fix ast_sockaddr initialization to be more portable + + A change to glibc 2.22 changed the order of the sockadddr_storage + members which caused the places where we do an initialization of + ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those + initializers (which we shouldn't have been using anyway) have been + replaced with memsets. + + Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4 + +2016-06-03 00:59 +0000 [72d190eb69] Timo Teräs + + * Detect and use proper libraries for musl toolchains + + Change-Id: I8d9b212f70813404b82918a3f99439e500d4bfcb + +2016-06-03 00:57 +0000 [39b69ab537] Timo Teräs + + * Fixes to include signal.h + + POSIX defines signal.h. sys/signal.h should not be used as it is + c-library internal header which may or may not exist. Notably with + musl it generates warning of being incorrect. + + Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc + +2016-06-08 12:26 +0000 [7f5ca67e5f] Matt Jordan + + * res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded + + A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not + loaded and does not have a configuration file. Previously when this + occurred, checks were put in to see if the configuration was loaded + successfully. While this is a good idea - and has been added to the + offending function in res_hep - the reality is res_hep_pjsip and + res_hep_rtcp have no business running if res_hep isn't also running. + + As such, this patch also adds a function to res_hep that returns whether + or not it successfully loaded. Oddly enough, ast_module_check returns + "everything is peachy" even if a module declined its load - so it cannot + be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this + function to see if they should continue to load; if it fails, they + decline their load as well. + + ASTERISK-26096 #close + + Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea + +2016-06-08 02:11 +0000 [784c18128b] Alexander Traud + + * chan_sip: No rtpmap for static RTP payload IDs in SDP. + + This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in + SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over + UDP, if many codecs are allowed in Asterisk. This new feature is enabled + together with the optional feature compactheaders=yes via the file sip.conf. + + ASTERISK-25578 #close + + Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044 + +2016-06-02 12:04 +0000 [31a5c28339] Joshua Colp + + * res_odbc: Implement a connection pool. + + Testing has shown that our usage of UnixODBC is problematic + due to bugs within UnixODBC itself as well as the heavy weight + cost of connecting and disconnecting database connections, even + when pooling is enabled. + + For users of UnixODBC 2.3.1 and earlier crashes would occur due + to insufficient protection of the disconnect operation. This was + fixed in UnixODBC 2.3.2 and above. + + For users of UnixODBC 2.3.3 and higher a slow-down would occur + under heavy database use due to repeated connection establishment. + A regression is present where on each connection the database + configuration is cached again, with the cache growing out of + control. + + The connection pool implementation present in this change helps + to mitigate these issues by reducing how much we connect and + disconnect database connections. We also solve the issue of + crashes under UnixODBC 2.3.1 by defaulting the maximum number of + connections to 1, returning us to the previous working behavior. + For users who may have a fixed version the maximum concurrent + connection limit can be increased helping with performance. + + The connection pool works by keeping a list of active connections. + If the connection limit has not been reached a new connection is + established. If the connection limit has been reached then the + request waits until a connection becomes available before + continuing. + + ASTERISK-26074 #close + ASTERISK-26054 #close + + Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff + +2016-05-31 09:10 +0000 [80ff7912a1] Vasil Kolev + + * chan_sip: bigger buffers for headers, better failure mode + + Currently chan_sip can give weird messages if the contacts don't + fit in the From: or To: headers. This fix changes the from,to and + invite variables to use ast_str, allocates and deallocates them and + resizes them if needed. + + ASTERISK-26069 #close + + Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3 + +2016-06-06 11:13 +0000 [60caebc738] Örn Arnarson + + * apps/app_voicemail.c and main/say.c: Add support for Icelandic language + + Icelandic has some weird grammar rules when dealing with dates and + numbers. There are different genders used depending on which number + you're dealing with, and only a handful of numbers do change depending + on the gender. There is also an implied gender in several cases. + + This patch was originally written for asterisk 1.6, and has been in use + for several years without crashes. I cleaned it up a bit and rewrote + what was necessary for Asterisk 13. + + The functions were copied from other similar languages and modified + where appropriate. If i recall correctly, the German and Danish + functions were used as a base. + + ASTERISK-26087 + Reported by: Örn Arnarson + Tested by: Örn Arnarson + + Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665 + +2016-06-07 05:45 +0000 [52120204c9] Alexander Traud + + * res_srtp: Instead of libSRTP use OpenSSL as random source. + + Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. + Therefore, the symbol RAND_bytes is used instead of crypto_get_random. + + ASTERISK-24436 #close + + Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96 + +2016-06-07 02:16 +0000 [da943ec5c0] Alexander Traud + + * BuildSystem: Avoid 'ar cru' and use 'ar cr' instead. + + In several internal library projects, the files are archived with the help of + 'ar cr'. Only the projects editline and the Objective Open H.323 stack + implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms + changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier + ignored since `D' is the default (see `U')". For consistency and to avoid this + message all projects use 'ar cr' now. + + ASTERISK-26091 #close + + Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40 + +2016-06-01 16:57 +0000 [dca052e531] Richard Mudgett + + * chan_rtp.c: Simplify options to UnicastRTP channel creation. + + Change the awkward and not as flexible UnicastRTP options format + From: + Dial(UnicastRTP/127.0.0.1[/[][/[]]]) + To: + Dial(UnicastRTP/127.0.0.1[/[]]) + + Where can be standard Asterisk flag options: + c() - Specify which codec/format to use such as 'ulaw'. + e() - Specify which RTP engine to use such as 'asterisk'. + + More option flags can be easily added later such as the codec's RTP + payload type to use when the codec does not have a static payload type + defined. + + Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9 + +2016-05-02 05:57 +0000 [5bfef2a8b4] Jaco Kroon + + * translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. + + ASTERISK-25629 #close + + Change-Id: Ibfcf0670e094e9718d82fd9920f1fb2dae122006 + +2016-05-25 10:34 +0000 [3e8d523d88] Alexei Gradinari + + * core/dial: New channel variable FORWARDERNAME + + Added a new channel variable FORWARDERNAME which indicates which + channel was responsible for a forwarding requests received on dial attempt. + + Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. + + ASTERISK-26059 #close + + Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2 + +2016-05-27 14:49 +0000 [a2f820e8dc] gtjoseph + + * ari/resource_channels: Add 'formats' to channel create/originate + + If you create a local channel and don't specify an originator channel + to take capabilities from, we automatically add all audio formats to + the new channel's capabilities. When we try to make the channel + compatible with another, the "best format" functions pick the best + format available, which in this case will be slin192. While this is + great for preserving quality, it's the worst for performance and + overkill for the vast majority of applications. + + In the absense of any other information, adding all formats is the + correct thing to do and it's not always possible to supply an + originator so a new parameter 'formats' has been added to the channel + create/originate functions. It's just a comma separated list of formats + to make availalble for the channel. Example: "ulaw,slin,slin16". + 'formats' and 'originator' are mutually exclusive. + + To facilitate determination of format names, the format name has been + added to "core show codecs". + + ASTERISK-26070 #close + + Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b + +2016-06-03 01:33 +0000 [538c6415c6] Timo Teräs + + * chan_sip: Support auth username for callbackextension feature + + ASTERISK-20527 #close + + Change-Id: I659cf7f00836a09d09d146ad226a40477d731239 + +2016-06-03 00:39 +0000 [797695c5cc] Timo Teräs + + * Make use of GLOB_BRACE and GLOB_NOMAGIC optional + + These flags are non-portable GNU extensions. Make their use + optional. This fixes complication error on e.g. musl c-library + based systems. + + Change-Id: I0aa06efc62aa8995f091445c8b762a75a91042f3 + +2016-06-02 14:57 +0000 [3c1fec8099] Timo Teräs + + * Fix res_search usage + + Resolver state is not part of res_search API. This fixes + compilation error: + + dns.c:261:8: error: too many arguments to function 'res_search' + ret = res_search(&dns_state, + + Change-Id: Ia600a58557040df83f744da3dde23225293845a5 + +2016-06-02 14:53 +0000 [9c1d95e873] Timo Teräs + + * Fix #include poll.h and sys/cdefs.h + + POSIX defines poll.h, sys/poll.h should not be used at is c-library + internal header which may or may not exist. Notable in musl it + generates warning of being incorrect. And add explict include of + sys/cdefs.h where needed. + + Change-Id: I142930df53fe7585a06b854b6faddc5301e024be + +2016-05-25 08:45 +0000 [8a5c2e736c] Niklas Larsson + + * core/manager: Add uptime field to FullyBooted + + Add Uptime and LastReload to event FullyBooted. + + ASTERISK-26058 #close + Reported by: Niklas Larsson + + Change-Id: I909b330801c0990d78df9b272ab0adc95aecb15e + +2016-06-02 04:59 +0000 [4505a59dc9] Joshua Colp + + * alembic: Fix migration. + + The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting + to use UniqueConstraint and failing. It was not imported and after + importing it also continued to fail. + + I've changed the script to use the explicit name of the constraint + instead. + + Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9 + +2016-06-01 13:57 +0000 [40d19f2e55] Richard Mudgett + + * logging,cdr,cel: Fix stringfield memory leak. + + The stringfields refactor to allow adding stringfields to the end of a + structure (f6f4cf459f43f072604927209b39646f84aaa2e2) exposed some + incomplete cleanup code by some stringfield users. + + The most noticeable leaker is the logging system where there is a leak for + every log message generated. + + ASTERISK-26078 #close + Reported by: Etienne Lessard + Patches: + jira_asterisk_26078_v13.patch (license #5621) patch uploaded + by Richard Mudgett + + Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782 + +2016-05-31 13:02 +0000 [aec7916595] Richard Mudgett + + * pjsip_distributor.c: Use correct rdata info access method (Part 2). + + The pjproject doxygen for rdata->msg_info.info says to call + pjsip_rx_data_get_info() instead of accessing the struct member directly. + You need to call the function mostly because the function will generate + the struct member value if it is not already setup. + + Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799 + +2016-05-09 15:00 +0000 [205a31f86c] Mark Michelson + + * Expand the scope of Dial Events + + Dial events up to this point have come in two flavors + * A Dial event with no status to indicate that dialing has begun + * A Dial event with a status to indicate that dialing has ended + + With this change, Dial events have been expanded to also give + intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS". + This is especially useful for ARI dialing, as it gives the application + writer the opportunity to place a channel into an early bridge when + early media is detected. + + AMI handles these in-progress dial events by sending a new event called + "DialState" that simply indicates that dial state has changed but has + not ended. ARI never distinguished between DialBegin and DialEnd, so no + change was made to the event itself. + + Another change here relates to dial forwards. A forward-related event + was previously only sent when a channel was successfully able to forward + a call to a new channel. With this set of changes, if forwarding is + blocked, we send a Dial event with a forwarding destination but no + forwarding channel, since we were prevented from creating one. This is + again useful for ARI since application writers can now handle call + forward attempts from within their own application. + + ASTERISK-25925 #close + Reported by Mark Michelson + + Change-Id: I42cbec7730d84640a434d143a0d172a740995543 + +2016-05-30 19:27 +0000 [8a6a14590d] gtjoseph + + * res_pjsip_mwi_body_generator: Re-order the body items + + Re-ordered the body items so Message-Account is second. + + Messages-Waiting: no + Message-Account: sip:1571@:5060 + Voice-Message: 0/0 (0/0) + + ASTERISK-26065 #close + Reported-by: Ross Beer + + Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3 + +2016-05-30 10:58 +0000 [7fa5766752] gtjoseph + + * pjproject_bundled: Move to pjproject 2.5 + + Although all the patches we had against 2.4.5 were applied by Teluu, + a new bug was introduced preventing re-use of tcp and tls transports + This patch removes all the previous patches against 2.4.5, updates + the version to 2.5, and adds a new patch to correct the transport + re-use problem. + + Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068 + +2016-05-27 12:25 +0000 [b56f611856] Rusty Newton + + * res_pjsip: Add clarifying documentation to PJSIP_HEADER help text + + Added notes about when you can read or write headers. Specifically + about being able to read on the inbound channel and write on an + outbound channel. + + ASTERISK-26063 #close + Reported by: Private Name + Tested by: Rusty Newton + + Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5 + +2016-05-26 15:14 +0000 [bb0f4a6310] Mark Michelson + + * multicast RTP: Add dialing options + + This adds a new parameter to the end of a multicast RTP dialing string. + This parameter defines the following options: + + * i: Set the interface from which multicast RTP is sent + * l: Set whether multicast packets are looped back to the sender + * t: Set the TTL for multicast packets + * c: Set the codec to use for RTP + + ASTERISK-26068 #close + Reported by Mark Michelson + + Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219 + +2016-05-09 14:48 +0000 [88d997913f] Mark Michelson + + * ARI: Re-implement the ARI dial command, allowing for early bridging. + + ARI dial had been implemented using the Dial API. This made great sense + when dialing was 100% separate from bridging. However, if a channel were + to be added to a bridge during the dial attempt, there would be a + conflict between the dialing thread and the bridging thread. Each would + be attempting to read frames from the dialed channel and act on them. + + The initial attempt to make the two play nice was to have the Dial API + suspend the channel in the bridge and stay in charge of the channel + until the dial was complete. The problem with this was that it was + riddled with potential race conditions. It also was not well-suited for + the case where the channel changed which bridge it was in during the + dial. + + This new approach removes the use of the Dial API altogether. Instead, + the channel we are dialing is placed into an invisible ARI dialing + bridge. The bridge channel thread handles incoming frames from the + channel. If the channel is added to a real bridge, it is departed from + the invisible bridge and then added to the real bridge. Similarly, if + the channel is removed from the real bridge, it is automatically added + back to the invisible bridge if the dial attempt is still active. + + This approach keeps the threading simple by always having the channel + being handled by bridge channel threads. + + ASTERISK-25925 + + Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb + +2016-05-19 14:56 +0000 [31f17abe44] Alexei Gradinari + + * res_pjsip: add "via_addr", "via_port", "call_id" to contact + + As res_pjsip_nat rewrites contact's address, only the last Via header + can contain the source address of registered endpoint. + Also Call-Id header may contain the source address of registered + endpoint. + + Added "via_addr", "via_port", "call_id" to contact. + Added new fields ViaAddress, CallID to AMI event ContactStatus. + + ASTERISK-26011 + + Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576 + +2016-05-24 16:56 +0000 [574c9e77eb] Alexei Gradinari + + * res_pjsip: chatty verbose messages + + There are a lot of verbose messages about Endpoint and Contact status + changes if there are many dynamic endpoints. + The patch sets verbose level 2 for Endpoint status changes + and verbose level 3 for Contact status changes. + + ASTERISK-26055 #close + + Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7 + +2016-05-20 13:56 +0000 [b3142e99e4] Alexei Gradinari + + * app_voicemail: fix bugs, imap mm_status log change to debug + + Fixed some bugs: + - create dirpath when save downloading message from IMAP storage. + - create IMAP folder if not exists when saving to IMAP storage + - check if file successfully opened before write to it + - some IMAP checks + - remove non-standard flag 'Unseen' + etc + + Change to debug IMAP mm_status log instead of verbose. + + Remove unused X-Asterisk-VM-Caller-channel message header + for security reason. The clients should not know name of peer/endpoint. + + ASTERISK-26045 #close + + Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b + +2016-05-25 18:30 +0000 [7d44d12816] Richard Mudgett + + * pjsip_distributor.c: Use correct rdata info access method. + + The pjproject doxygen for rdata->msg_info.info says to call + pjsip_rx_data_get_info() instead of accessing the struct member directly. + You need to call the function mostly because the function will generate + the struct member value if it is not already setup. + + Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2 + +2016-05-03 11:11 +0000 [1d60bfcdf1] Tzafrir Cohen + + * followme: allow disabling callee prompt + + Add the option 'enable_callee_prompt' to followme.conf. Enabled by + default. If disabled, a callee is not prompted to accept or reject + the forwarded call. + + ASTERISK-26064 #close + + Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5 + Signed-off-by: Tzafrir Cohen + +2016-02-12 09:59 +0000 [80ff2c2540] Corey Farrell + + * threadpool: Fix potential data race. + + worker_start checked for ZOMBIE status without holding a lock. All + other read/write of worker status are performed with a lock, so this + check should do the same. + + ASTERISK-25777 #close + + Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781 + +2016-05-24 05:28 +0000 [070eab6ed2] Joshua Colp + + * res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying. + + Recent changes to res_pjsip_outbound_publish have introduced a + race condition at shutdown where an outbound publish may be shutdown + twice. In this case the first succeeds as a result of the unpublish. + In the second invocation since it's been unpublished a task is + queued to just destroy the client. This task holds no ref to the + publish and as a result the publish may be destroyed before the + task is run, causing a crash. + + This explicit destruction task now holds a reference to the publish + to ensure it remains valid. + + ASTERISK-26053 #close + + Change-Id: I10789b98add3e50292ee3b33a55a1d9061cec94b + +2016-05-09 14:27 +0000 [f6c33771f6] Mark Michelson + + * Bridging: introduce "invisible" bridges. + + Invisible bridges function the same as normal bridges, but they have the + following restrictions: + + * They never show up in CLI, AMI, or ARI queries. + * They do not have Stasis messages published about them. + + Invisible bridges' main use is for when use of the bridging system is + desired, but the bridge should not be known to users of the Asterisk + system. + + ASTERISK-25925 + + Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670 + +2016-05-22 11:03 +0000 [85d0272e76] Joshua Colp + + * res_pjsip: Only check transaction on transaction state events. + + The send request callback function currently assumes that it + will only ever be called on transaction state changes. This is + not always true. If our own timer callback occurs we will call + the callback with a timer event instead of a transaction state + change event. In this case the transaction on the event is + invalid and accessing it will result in a crash. + + ASTERISK-26049 #close + + Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc + +2016-05-21 05:42 +0000 [31897d2d99] Jesper (License 5518) + + * func_curl: Don't trim response text on non-ASCII characters + + The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of + a signed comparison. + + ASTERISK-25669 #close + Reported by: Jesper + patches: + strings.curl.trim.patch submitted by Jesper (License 5518) + + Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a + +2016-05-20 19:03 +0000 [2a77af9ed0] Richard Mudgett + + * chan_rtp.c: Cleanup ast_request() parameter parsing. + + * Fixed NULL crash potential if parameters are missing. + + * Reordered some operations so further diagnostic messages can be + more helpful. + + Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70 + +2016-05-20 16:59 +0000 [ade5275a3e] Richard Mudgett + + * parking.h: Update ast_parking_park_call() doxygen to reality. + + ASTERISK-26029 + + Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260 + +2016-05-12 15:18 +0000 [c378b00a83] Alexei Gradinari + + * func_odbc: single database connection should be optional + + func_odbc was changed in Asterisk 13.9.0 + to make func_odbc use a single database connection per DSN + because of reported bug ASTERISK-25938 + with MySQL/MariaDB LAST_INSERT_ID(). + + This is drawback in performance when func_odbc is used + very often in dialplan. + + Single database connection should be optional. + + ASTERISK-26010 + + Change-Id: I7091783a7150252de8eeb455115bd00514dfe843 + +2016-05-20 09:39 +0000 [1c02b19b79] Mark Michelson + + * res_pjsip: Match dialogs on responses better. + + When receiving an incoming response to a dialog-starting INVITE, we were + not matching the response to the INVITE dialog. Since we had not + recorded the to-tag to the dialog structure, the PJSIP-provided method + to find the dialog did not match. + + Most of the time, this was not a problem, because there is a fall-back + that makes the response get routed to the same serializer that the + request was sent on. However, in cases where an asynchronous DNS lookup + occurs in the PJSIP core, the thread that sends the INVITE is not + actually a threadpool serializer thread. This means we are unable to + record a serializer to handle the incoming response. + + Now, imagine what happens when an INVITE is sent on a non-serialized + thread, and an error response (such as a 486) arrives. The 486 ends up + getting put on some random threadpool thread. Eventually, a hangup task + gets queued on the INVITE dialog serializer. Since the 486 is being + handled on a different thread, the hangup task can execute at the same + time that the 486 is being handled. The hangup task assumes that it is + the sole owner of the INVITE session and channel, so it ends up + potentially freeing the channel and NULLing the session's channel + pointer. The thread handling the 486 can crash as a result. + + This change has the incoming response match the INVITE transaction, and + then get the dialog from that transaction. It's the same method we had + been using for matching incoming CANCEL requests. By doing this, we get + the INVITE dialog and can ensure that the 486 response ends up being + handled by the same thread as the hangup, ensuring that the hangup runs + after the 486 has been completely handled. + + ASTERISK-25941 #close + Reported by Javier Riveros + + Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0 + +2016-05-18 06:19 +0000 [e773e3a9bb] Matt Jordan + + * ARI: Add the ability to download the media associated with a stored recording + + This patch adds a new feature to ARI that allows a client to download + the media associated with a stored recording. The new route is + /recordings/stored/{name}/file, and transmits the underlying binary file + using Asterisk's HTTP server's underlying file transfer facilities. + + Because this REST route returns non-JSON, a few small enhancements had + to be made to the Python Swagger generation code, as well as the + mustache templates that generate the ARI bindings. + + ASTERISK-26042 #close + + Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181 + +2016-05-19 11:41 +0000 [40cb032009] Joshua Colp + + * res_sorcery_astdb: Filter fields to only the registered ones. + + This change introduces the same filtering that is done in res_sorcery_realtime + to the res_sorcery_astdb module. This allows persisted sorcery objects + that may contain unknown fields to still be read in from the AstDB + and used. This is particularly useful when switching between different + versions of Asterisk that may have introduced additional fields. + + ASTERISK-26014 #close + + Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2 + +2016-05-09 21:40 +0000 [9766a12b4c] snuffy + + * res_pjsip_empty_info: Respond to empty SIP INFO packets + + Some SBCs require responses to empty SIP INFO packets + after establishing call via INVITE, if not responded to + they may drop your call after unspecified timeout of X minutes. + + They are identified by having no Content-Type, check for this + and respond with 200 - OK message. + + ASTERISK-24986 #close + Reported-by: Ilya Trikoz, Federico Santulli + + Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0 + +2016-05-18 10:58 +0000 [111c4b0324] Tzafrir Cohen + + * Makefile: remove OSARCH check for init install + + There are more specific checks for the platform. + + Specifically this allows installing OS/X init scripts. + + ASTERISK-26038 #close + + Change-Id: If08933621145b10362a0cfe73c079301d9c13f50 + Signed-off-by: Tzafrir Cohen + +2016-05-10 11:28 +0000 [d4b77dad1b] Joshua Colp + + * res_pjsip_exten_state: Use the extension for publishing to. + + This change uses the newly added multi-user support for + outbound publish to publish to the specific user that an + extension state change is for. + + This also extends the res_pjsip_outbound_publish support + to include the user specific From and To URI information in + the outbound publishing of extension state. Since the URI + is used when constructing the body it is important to ensure + that the correct local and remote URIs are used. + + Finally the max string growths for the dialog-info+xml + body generator has been increased as through testing it has + proven to be too conservative. + + ASTERISK-25965 + + Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1 + +2016-05-03 16:07 +0000 [3905997bae] Kevin Harwell + + * res_pjsip_outbound_publish: Add multi-user support per configuration + + Added a new multi_user option that when specified allows a particular + configuration to be used for multiple users. It does this by replacing + the user portion of the server uri with a dynamically created one. + + Two new API calls have been added in order to make use of the new + functionality: + + ast_sip_publish_user_send - Sends an outgoing publish message based on the + given user. If state for the user already exists it uses that, otherwise + it dynamically creates new outbound publishing state for the user at that + time. + + ast_sip_publish_user_remove - Removes all outbound publish state objects + associated with the user. This essentially stops outbound publishing for + the user. + + ASTERISK-25965 #close + + Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc + +2016-05-18 07:54 +0000 [6e5e84458f] gtjoseph + + * udptl: Don't eat sequence numbers until OK is received + + Scenario: + Local fax -> Asterisk w/ firewall -> Provider -> Remote fax + + * Local fax starts rtp call to remote fax + * Remote fax starts t38 call back to local fax. + * Local fax sends t38 no-signal to Asterisk before sending an OK. + * udptl processes the frame and increments the expected sequence number. + * chan_sip drops the frame because the call isn't up so nothing goes out + the external interface to open the port for incoming packets. + * Local fax sends OK and Asterisk sends OK to the remote fax. + * Remote fax sends t38 packets which are dropped by the firewall. + * Local fax re-sends t38 no-signal with the same sequence number. + * udptl drops the frame because it thinks it's a dup. + * Still no outgoing packets to open the firewall. + * t38 negotiation fails. + + The patch drops frames t38 received before udptl sequence processing + when the call hasn't been answered yet. The second no-signal frame + is then seen as new and is relayed out the external interface which + opens the port and allows negotiation to continue. + + ASTERISK-26034 #close + + Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 + +2016-05-15 12:22 +0000 [52148d93f4] Matt Jordan + + * CHANGES: Update formatting of items + + * Provide consistent indenting of lines in bulleted paragraphs + * Respect the 80 character column width + * Group all like items together, e.g., all dialplan applications under + "Applications", etc. + * Use a single blank line to break up functionality changes within a + larger section + * Use two blanks lines to delineate larger sections + + Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647 + +2016-04-18 18:17 +0000 [03d88b5656] Matt Jordan + + * ARI: Add the ability to play multiple media URIs in a single operation + + Many ARI applications will want to play multiple media files in a row to + a resource. The most common use case is when building long-ish IVR prompts + made up of multiple, smaller sound files. Today, that requires building a + small state machine, listening for each PlaybackFinished event, and triggering + the next sound file to play. While not especially challenging, it is tedious + work. Since requiring developers to write tedious code to do normal activities + stinks, this patch adds the ability to play back a list of media files to a + resource. + + Each of the 'play' operations on supported resources (channels and bridges) + now accepts a comma delineated list of media URIs to play. A single Playback + resource is created as a handle to the entire list. The operation of playing + a list is identical to playing a single media URI, save that a new event, + PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final + media URI. When the entire list is finished being played, a PlaybackFinished + event is raised. + + In order to help inform applications where they are in the list playback, the + Playback resource now includes a new, optional attribute, 'next_media_uri', + that contains the next URI in the list to be played. + + It's important to note the following: + - If an offset is provided to the 'play' operations, it only applies to the + first media URI, as it would be weird to skip n seconds forward in every + media resource. + - Operations that control the position of the media only affect the current + media being played. For example, once a media resource in the list + completes, a 'reverse' operation on a subsequent media resource will not + start a previously completed media resource at the appropiate offset. + - This patch does not add any new operations to control the list. Hopefully, + user feedback and/or future patches would add that if people want it. + + ASTERISK-26022 #close + + Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f + +2016-05-17 11:14 +0000 [5bd1bf2816] gtjoseph + + * chan_sip: Prevent extra Session-Expires headers from being added + + When chan_sip does a re-INVITE to refresh a session and authentication + is required, the INVITE with the Authorization header containes a + second Session-Expires header without the ";refersher=" parameter. + This is causing some proxies to return a 400. Also, when Asterisk is + the uas and the refresher, it is including the Session-Expires and + Min-SE headers in OPTIONS messages which is not allowed per RFC4028. + + This patch (based on the reporter's) Checks to see if a Session-Expires + header is already in the message before adding another one. It also + checks that the method is INVITE or UPDATE. + + ASTERISK-26030 #close + + Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9 + +2016-05-16 15:29 +0000 [ae81b55361] gtjoseph + + * res_pjsip_outbound_registration: Clean up state when registration is deleted + + Nothing was cleaning up the registration state object when ast_sorcery_delete + was called on a registration. So, the registration was deleted from sorcery + but the state object went right on refreshing the registration (or failing + to refresh the registration) with the peer. + + * Added a 'deleted' observer on registration that removes the state object. + + ASTERISK-25964 #close + Reported-by Matt Jordan + + Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23 + +2016-05-15 19:05 +0000 [8b5cee4a4f] gtjoseph + + * res_pjsip: Set TCP_NODELAY on TCP transports + + Although it's perfectly legal to place multiple SIP messages in the same packet, + it can cause problems because the Linux default is to enable Path MTU Discovery + which sets the Don't Fragment bit on the packets. If adding a second message to + the packet causes the MTU to be exceeded, and the destination isn't equipped to + send a FRAGMENTATION NEEDED response to a large packet, the packet will just be + dropped. + + We can't specifically tell the stack to send only 1 message per packet, but we + can turn on TCP_NODELAY when we create the transport. This will at least tell + the stack to send packets as soon as possible. + + ASTERISK-26005 #close + Reported-by: Ross Beer + + Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd + +2016-05-14 07:24 +0000 [3522376512] Matt Jordan + + * logger: Support JSON logging with Verbose messages + + When 2d7a4a3357 was merged, it missed the fact that Verbose log messages + are formatted and handled by 'verbosers'. Verbosers are registered + functions that handle verbose messages only; they exist as a separate + class of callbacks. This was done to handle the 'magic' that must be + inserted into Verbose messages sent to remote consoles, so that the + consoles can format the messages correctly, i.e., the leading + tabs/characters. + + In reality, verbosers are a weird appendage: they're a separate class of + formatters/message handlers outside of what handles all other log + messages in Asterisk. After some code inspection, it became clear that + simply passing a Verbose message along with its 'sublevel' importance + through the normal logging mechanisms removes the need for verbosers + altogether. + + This patch removes the verbosers, and makes the default log formatter + aware that, if the log channel is a console log, it should simply insert + the 'verbose magic' into the log messages itself. This allows the + console handlers to interpret and format the verbose message + themselves. + + This simplifies the code quite a lot, and should improve the performance + of printing verbose messages by a reasonable factor: + (1) It removes a number of memory allocations that were done on each + verobse message + (2) It removes the need to strip the verbose magic out of the verbose + log messages before passing them to non-console log channels + (3) It now performs fewer iterations over lists when handling verbose + messages + + Since verbose messages are now handled like other log messages (for the + most part), the JSON formatting of the messages works as well. + + ASTERISK-25425 + + Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96 + +2016-05-14 21:48 +0000 [a1803cb5f4] Matt Jordan + + * configs/samples/pjsip.conf.sample: Fix typo + + A ':' is not a valid token for starting a comment. + + Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad + +2016-05-12 07:08 +0000 [d29c17834c] Matt Jordan + + * res/res_hep_pjsip: Fix reported local IP address when bound to 'any' + + When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its + local address the 'any' address, as opposed to the IP address we + actually received the packet on. This can cause some confusion in Homer, + as it will dutifully report what we send it. + + This patch uses the PJSIP inspection routines to determine which IP + address we probably received the packet on based on the remote party's + IP address. In the event that this fails, it falls back to the IP + address natively reported by the transport. + + Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3 + +2016-05-14 12:29 +0000 [14938184a3] Sean Bright + + * res_ari: Correct Location headers returned by some ARI resources + + The Location headers returned by: + + * /bridges/{bridgeId}/play + * /bridges/{bridgeId}/record + * /channels/{channelId}/play + * /channels/{channelId}/record + + Did not have the '/ari' prefix, and in the case of the 'play' resources, were + using 'playback' instead of 'playbacks.' + + Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c + +2016-05-11 20:17 +0000 [e06a23681c] Matt Jordan + + * res_hep: Provide an option to pick the UUID type + + At one point in time, it seemed like a good idea to use the Asterisk + channel name as the HEP correlation UUID. In particular, it felt like + this would be a useful identifier to tie PJSIP messages and RTCP + messages together, along with whatever other data we may eventually send + to Homer. This also had the benefit of keeping the correlation UUID + channel technology agnostic. + + In practice, it isn't as useful as hoped, for two reasons: + 1) The first INVITE request received doesn't have a channel. As a + result, there is always an 'odd message out', leading it to be + potentially uncorrelated in Homer. + 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. + This causes RTCP information to be uncorrelated to the SIP message + traffic seen by those capture nodes. + + In order to support both (in case someone is trying to use res_hep_rtcp + with a non-PJSIP channel), this patch adds a new option, uuid_type, with + two valid values - 'call-id' and 'channel'. The uuid_type option is used + by a module to determine the preferred UUID type. When available, that + source of a correlation UUID is used; when not, the more readily available + source is used. + + For res_hep_pjsip: + - uuid_type = call-id: the module uses the SIP Call-ID header value + - uuid_type = channel: the module uses the channel name if available, + falling back to SIP Call-ID if not + For res_hep_rtcp: + - uuid_type = call-id: the module uses the SIP Call-ID header if the + channel type is PJSIP and we have a channel, + falling back to the Stasis event provided + channel name if not + - uuid_type = channel: the module uses the channel name + + ASTERISK-25352 #close + + Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c + +2016-05-13 11:46 +0000 [69a85a519f] Alexei Gradinari + + * res_pjsip: Endpoint IP Access Controls + + With the old SIP module we can use IP access controls per peer. + PJSIP module missing this feature. + + This patch added next configuration Endpoint options: + "acl" - list of IP ACL section names in acl.conf + "deny" - List of IP addresses to deny access from + "permit" - List of IP addresses to permit access from + "contact_acl" - List of Contact ACL section names in acl.conf + "contact_deny" - List of Contact header addresses to deny + "contact_permit" - List of Contact header addresses to permit + + This patch also better logging failed request: + add custom message instead of "No matching endpoint found" + add SIP method to logging + + ASTERISK-25900 + + Change-Id: I456dea3909d929d413864fb347d28578415ebf02 + +2016-05-12 14:36 +0000 [fd3f70598d] Mark Michelson + + * Use doubles instead of floats for conversions when comparing strings. + + In 13.9.0, there was an issue where PJSIP contacts added to an AOR would + be deleted at seemingly random times. + + One reason this was happening was because of an operation to retrieve + the contacts whose expiration time was less than or equal to the current + time. When retrieving existing contacts, the contact's expiration time + and the current time were converted from a string to a float, and those + two floats were compared. + + On some systems, including mine, this conversion was horribly off. For + instance, I could regularly see the string "1463079214" get converted + into 1463079168.000000. When switching from using a float to using a + double, the conversion was as expected. + + Why was the conversion to float off? My best guess is that the + conversion to float was attempting to store the entire value in the 23 + bit significand of the IEEE-754 floating point number. In particular, if + you take only the 23 most significant bits of 1463079214, you get the + messed up 1463079168 that we were seeing in the conversion. It likely + was possible to get a more precise value by composing the number using + an exponent, but the conversion did not work that way. With a double, + you have a 52 bit significand, allowing the entire value to fit there, + and thereby allowing an accurate conversion. + + ASTERISK-26007 #close + Reported by Greg Siemon + + Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070 + +2016-05-12 09:13 +0000 [4f8cfa0220] gtjoseph + + * pjsip_distributor: Add missing newline to NOTICE + + There was a newline missing from the end of the "no matching endpoint" notice. + + Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181 + +2016-05-10 10:19 +0000 [d14d1ba826] Sebastian Damm + + * res_pjsip_outbound_registration: generate correct Contact URI for TLS + + There are two types of SIP URIs indicating a secure transport: + * sips:user@example.org + * sip:user@example.org;transport=tls + + When using a sips URI, Asterisk checks incoming INVITEs and answers from + the other side for sips URIs, and rejects the packet if there are only + sip URIs. So Asterisk should only generate a sips Contact URI if the + other side supports it. + + This patch makes Asterisk generate either a sip or sips Contact URI + depending on the format of the server URI. + + If you want a sip URI, use: + server_uri=sip:example.org\;transport=tls + + If you want a sips URI, use: + server_uri=sips:example.org + + ASTERISK-25990 #close + Reported-by: Sebastian Damm + + Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2 + +2016-05-05 16:41 +0000 [9f996624b0] Alexei Gradinari + + * logger: Add PID to syslog messages. + + During refactoring of this support the addition of + the PID to messages was removed. This change adds it + back in. + + ASTERISK-25538 #close + + Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36 + +2016-05-11 14:07 +0000 [5236ffed97] Matt Jordan + + * configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER + + When running on a system that does not support or use AST_UNDEFINED_SANITIZER + or AST_LEAK_SANITIZER, the configure script would incorrectly set those + constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would + cause menuselect to error out, complaining that a blank value is not a + valid option. This patch corrects the issue by setting the value to 0 if + the options that those constants enable/disable is not found. + + Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba + +2016-05-10 08:17 +0000 [b5c471b339] Tzafrir Cohen + + * followme: delete the right recorded name file + + FollowMe with the option a records the name of the caller and plays it + to the callee. However it has failed to clean up that recorded file + as it tried to delete the file name without the '.sln' extension. + + ASTERISK-26008 #close + + Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec + Signed-off-by: Tzafrir Cohen + +2016-05-10 03:10 +0000 [ec85ea3c21] Tzafrir Cohen + + * basic-cfg: asterisk.conf: don't set languages + + * No need to set language in a miniml configuration. 'en' will do just + fine. + * It would be useful to have an example of setting it to a different + language. + * Setting the documentation language explicitly is likewise not + required. Setting it to a different value is not common. At least + until there is a set of translated documentation. + + Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7 + Signed-off-by: Tzafrir Cohen + +2016-05-10 03:08 +0000 [1b0a9bb2c4] Tzafrir Cohen + + * basic-cfg: asterisk.conf: debug level 5 spams + + Don't suggest users to use debug level 5, which spews (usually + non-useful) debug information. Reduce the suggestion to (an + arbitrarily-selected) level 2. + + Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60 + Signed-off-by: Tzafrir Cohen + +2016-05-10 03:06 +0000 [d0ba3e8196] Tzafrir Cohen + + * basic-cfg: asterisk.conf: defaults of options + + Note the default of remmed-out options. To clarify that those values are + not the defaults. + + Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738 + Signed-off-by: Tzafrir Cohen + +2016-05-10 02:56 +0000 [f943a1fd84] Tzafrir Cohen + + * basic-cfg: asterisk.conf: remove [directories] + + A minimal configuration does not need to explicitly spell out the + directories. The built-in defaults will do just fine. In many cases + they are wrong. + + Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c + Signed-off-by: Tzafrir Cohen + +2016-05-05 11:37 +0000 [1e876d6915] Kevin Harwell + + * res_pjsip_authenticator_digest: Don't use source port in nonce verification + + From the issue reporter: + "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of + the timestamp, the source address, the source port, a server UUID that is + calculated at startup, and the authentication realm. + + Rather than caching nonces that we create, we instead attempt to re-calculate + the nonce when receiving an incoming request with authentication. We then + compare the re-calculated nonce to the incoming nonce, and if they don't match, + then authentication has failed early. + + The problem is that it is possible, especially when using TCP, to receive two + requests from the same endpoint but have differing source ports for those + requests. Asterisk itself commonly will use different source ports for + outbound TCP requests." + + This patch removes the source port dependency when building the nonce. + + ASTERISK-25978 #close + + Change-Id: I871b5f4adce102df1c4988066283095ec509dffe + +2016-05-07 14:39 +0000 [dfefbf8731] gtjoseph + + * config_transport: Tell pjproject to allow all SSL/TLS protocols + + The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. + SSL is not allowed. So, even if you specify "sslv3" for a transport method, + it's silently ignored and one of the TLS protocols is used. This was a new + behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that + we never caught. + + Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). + This tells pjproject to set the socket protocol to match the method. + + ASTERISK-26004 #close + + Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078 + +2016-05-05 09:14 +0000 [d03e170ae7] Joshua Colp + + * res_pjsip_pubsub: Use common datastores container API. + + This migrates res_pjsip_pubsub over to using the newly + introduce common datastores management API instead of using + its own implementations for both subscriptions and + publications. + + As well the extension state data now provides a generic + datastores container instead of a subscription. This allows + the dialog-info+xml body generator to work for both + subscriptions and publications. + + ASTERISK-25999 #close + + Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1 + +2016-05-05 09:12 +0000 [94cd351ec4] Joshua Colp + + * datastore: Add common container based datastores API. + + This change introduces a common container based datastores + management API. This has been done in a few places across + the tree but this consolidates all of the logic into one + place in a generic fashion. + + ASTERISK-25999 + + Change-Id: I72eb15941dcdbc2a37bb00a33ce00f8755bd336a + +2016-05-04 02:40 +0000 [8923c9ac96] Jaco Kroon + + * app_confbridge: Add a regcontext option for confbridge bridge profiles. + + This patch allows for having app_confbridge register the name of the + conference as an extension into a specific context, similar to + regcontext for chan_sip. This variant is not quite as involved as the + one in chan_sip and doesn't allow for multiple contexts or custom + extensions, you can only specify the context and the conference name + will always be used as the extension to register. + + ASTERISK-25989 #close + + Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f + +2016-05-08 20:19 +0000 [facce6f632] gtjoseph + + * pjproject_bundled: Check for python-dev and TEST_FRAMEWORK + + The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set. + The python bindings are now built only if TEST_FRAMEWORK is set and a + python development package is installed. + + libresample was also disabled. + + ASTERISK-25993 #close + Reported-by: Joshua Colp + + Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03 + +2016-05-06 11:54 +0000 [322c3b4262] Alexei Gradinari + + * res_pjsip: module load priority + + The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_* + and res_pjsip_registrar modules should load ASAP + to avoid "No matching endpoint found" for legitimate endpoint. + + ASTERISK-25994 + + Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b + +2016-05-05 15:16 +0000 [516f49f316] Alexei Gradinari + + * stasis_endpoints: Add new Status and Headers to ContactStatus + + ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. + ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. + These additions should be also in stasis_endpoints + to include in command "manager show event ContactStatus" + + Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a + +2016-05-03 15:43 +0000 [64e058f75a] Kevin Harwell + + * res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches + + When reloading, or fetching realtime data, if the "apply" failed for any + numerous reasons the current state object would not be maintained. This + potentially resulted in publishes being stopped for some states/clients when + they should not have been. + + This patch makes it so the current state object is kept upon any type of reload/ + fetch failures. + + Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30 + +2016-05-03 15:35 +0000 [adc82a2260] Kevin Harwell + + * res_pjsip_outbound_publishing: After unloading the library won't load again + + The same thing was happening in res_pjsip_publish_asterisk. When the library + was unloaded it did not unregister the object type from sorcery. Subsequent + loads resulted in a failed load due to the sorcery type already existing. + + Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9 + +2016-05-03 15:39 +0000 [3b0ce5169d] Kevin Harwell + + * res_pjsip_outbound_publish: Won't unload if condition wait times out + + When res_pjsip_outbound_publish unloads it has to wait for all current + publishing objects to get done. However if the wait condition times out + then it does not fail the unload. This sometimes results in an infinite + loop check while unloading. This patch now fails the unload operation if + the condition times out. + + Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec + +2016-05-03 14:59 +0000 [41fccbfeb1] Kevin Harwell + + * res_pjsip_outbound_publish: Ref leak in off nominal callback paths + + There were a few spots where the client object's reference was being leaked in + sip_outbound_publish_callback. This patch cleans up those leaks. + + Change-Id: I485d0bc9335090f373026f77c548042e258461df + +2016-05-03 15:31 +0000 [dfbb03cc8e] Kevin Harwell + + * res_pjsip_outbound_publish: Potential crash due to off nominal path + + It was possible for the explicit publish destroy function to be called without + the pjsip client ever being initialized. This fix checks to make sure there is + a client to destroy before attempting. + + Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c + +2016-05-05 05:07 +0000 [17b6ba49ef] Joshua Colp + + * file: Ensure nativeformats remains valid for lifetime of use. + + It is possible for the nativeformats of a channel to change + throughout its lifetime. As a result a user of it needs to either + ensure the channel is locked when accessing the formats or keep + a reference to the nativeformats themselves. + + This change fixes the file playback support so it keeps a + reference to the nativeformats when accessing things. + + ASTERISK-25998 #close + + Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 + +2016-04-15 09:32 +0000 [cc4c5f5693] Alexei Gradinari + + * res_pjsip: improve realtime performance + + This patch modified pjsip_options to retrieve only + permament contacts for aor if the qualify_frequency is > 0 + and persisted contacts if the qualify_frequency is > 0. + + This patch also fixed a bug in res_sorcery_astdb. + res_sorcery_astdb doesn't save object data retrived from astdb. + + ASTERISK-25826 + + Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05 + +2016-05-02 16:52 +0000 [92f85fe766] Alexei Gradinari + + * res_fax/t38_gateway: Peer V.21 session is created on wrong channel + + The channel and peer V.21 sessions are created on the same channel now. + The peer V.21 session should be created only on peer channel + when one of channel can handle T.38. + + Also this patch enable debug for T.38 gateway session + if global fax debug enabled. + + ASTERISK-25982 + + Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e + +2016-05-04 16:11 +0000 [4df48581f1] Alexei Gradinari + + * pjsip: Added "reg_server" to contacts (fixed alembic) + + ASTERISK-25931 + + Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549 + +2016-05-04 03:17 +0000 [02f4ca1079] Chris Trobridge + + * config_options.c: Expand #ifdef to contain whole if statement. + + ASTERISK-25956 #close + + Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38 + +2016-05-02 16:08 +0000 [380ac201ac] Alexei Gradinari + + * res_fax: add FAXMODE variable + + The app_fax set FAXMODE variable, but res_fax missing this feature. + This patch add FAXMODE variable which is set to either "audio" or "T38". + + ASTERISK-25980 + + Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b + +2016-05-02 05:56 +0000 [0c9faaee47] Jean Aunis + + * app_chanspy: fix audiohook options in non read-only mode + + When option 'o' was not set, ChanSpy created its audiohook with the flag + AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one + direction only. + + ASTERISK-25866 #close + + Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0 + +2016-04-07 16:33 +0000 [a4cfcda036] Alexei Gradinari + + * res_pjsip/AMI: add contact.updated event + + With the old SIP module AMI sends PeerStatus event on every + successfully REGISTER requests, ie, on start registration, + update registration and stop registration. + + With PJSIP AMI sends ContactStatus only when status is changed. + Regarding registration: + on start registration - Created + on stop registration - Removed + but on update registration nothing + + This patch added contact.updated event. + + ASTERISK-25904 + + Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f + +2016-04-30 17:52 +0000 [e61716b774] gtjoseph + + * pjproject_bundled: Various fixes discovered during testing of OSes + + For all OSes: + * Disabled third-party codecs in pjproject and added + '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the + configure options since we don't use the pjsip codec capability. + + FreeBSD: + * Added FreeBSD support to install_prereq. + * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". + * Added __progname and environ to asterisk.exports.in. + * Reverted the use of ldconfig to create shared library symlinks to ln. + * Only enable epoll in pjproject if `uname -s` is Linux. + * Added a patch to pjproject to take the name of the 'make' command from + an environment variable if supplied. This is needed for the python bindings. + (merged by Teluu into pjproject trunk 5/3/2016) + FreeBSD support isn't complete. Still some general issues regarding + make/gmake having nothing to do with pjproject. With some handholding it DOES + build successfully. + + CentOS: + Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. + CentOS 6/7 32/64 build and run the pjsip testsuite successfully. + + Ubuntu: + No changes required. + Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. + + Debian: + No changes required. + Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. + + There will utimately be a follow-up patch to create an install_prereq for + the testsuite as I've discovered a few missing requirements. + + ASTERISK-25968 #close + + Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c + +2016-03-17 14:29 +0000 [080c6216b6] Andrew Nagy + + * app_voicemail: always copy dynamic struct to avoid race condition + + Voicemail email addresses can be corrupt or voicemail + emails can end up being sent to the wrong email address if asterisk is + reading voicemail.conf during a reload and processing an email at the + same time. This patch always copies the struct that would otherwise only + be copied once. + + ASTERISK-24463 #close + Reported by: John Campbell + Tested by: Etienne Lessard + Tested by: Andrew Nagy + Change-Id: I3a0643813116da84e2617291903d0d489b7425fb + +2016-04-15 14:26 +0000 [2b1edee772] Alexei Gradinari + + * pjsip: Added "reg_server" to contacts. + + If the Asterisk system name is set in asterisk.conf, it will be stored + into the "reg_server" field in the ps_contacts table to facilitate + multi-server setups. + + ASTERISK-25931 + + Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8 + +2016-05-01 02:21 +0000 [bf13b59062] Diederik de Groot + + * configs/basic-pbx/asterisk.conf: contains incorrect path separator + + Note: When packagers use these files (as an example) the paths are never + really used when they are split using '='. + + Note: Thirdparty applications will also have trouble parsing the file when + expecting '=>'. + + Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004 + +2016-04-27 17:19 +0000 [2c46063d54] Richard Mudgett + + * res_pjsip_exten_state: Create PUBLISH messages. + + Create PUBLISH messages to update a third party when an extension state + changes because of either a device or presence state change. + + A configuration example: + + [exten-state-publisher] + type=outbound-publish + server_uri=sip:instance1@172.16.10.2 + event=presence + ; Optional regex for context filtering, if specified only extension state + ; for contexts matching the regex will cause a PUBLISH to be sent. + @context=^users + ; Optional regex for extension filtering, if specified only extension + ; state for extensions matching the regex will cause a PUBLISH to be sent. + @exten=^[0-9]* + ; Required body type for the PUBLISH message. + ; + ; Supported values are: + ; application/pidf+xml + ; application/xpidf+xml + ; application/cpim-pidf+xml + ; application/dialog-info+xml (Planned support but not yet) + @body=application/pidf+xml + + The '@' extended variables are used because the implementation can't + extend the outbound publish type as it is provided by the outbound publish + module. That means you either have to use extended variables, or + implement some sort of custom extended variable thing in the outbound + publish module. Another option would be to refactor that stuff to have an + option which specifies the use of an alternate implementation's + configuration and then have that passed to the implementation. JColp + opted for the extended variables method originally. + + ASTERISK-25972 #close + + Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca + +2016-04-26 16:10 +0000 [0b5292525c] Richard Mudgett + + * res_pjsip_exten_state: Check if body generator is available. + + When starting the extension state publishers, check if the requested + message body generator is available. If not available give error message + and skip starting that publisher. + + * res_pjsip_pubsub.c: Create new API if type/subtype generator + registered. + + * res_pjsip_exten_state.c: Use new body generator API for validation. + + ASTERISK-25922 + + Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c + +2016-04-28 11:35 +0000 [369182d084] Richard Mudgett + + * res_pjsip: Start body generator users after suppliers. + + Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb + +2016-04-28 16:06 +0000 [3af83ea2fb] Richard Mudgett + + * res_pjsip_pubsub.c: Add useful information to some messages. + + Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a + +2016-04-26 15:58 +0000 [8e1b663b87] Richard Mudgett + + * res_pjsip_pubsub.c: Fix body generator registration race. + + Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67 + +2016-04-28 16:54 +0000 [30415944a8] gtjoseph + + * pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE + + When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled, + the input uri string will become corrupted if it contains escape sequences. + It's not possible to automatically strdup or strdupa the input string because + the output uri pj_str_t's will have pointers to chunks of the input string. + Getting around this would require more memory management code and wouldn't + be worth the savings of doing the unescape in place. + + ASTERISK-25970 #close + Reported-by: Dmitriy Serov + + Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88 + +2016-04-26 15:13 +0000 [906ea2c43f] Richard Mudgett + + * res_pjsip_pubsub.h: Fix doxygen association. + + Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632 + +2016-04-25 16:00 +0000 [76ea4cfaae] Richard Mudgett + + * res_pjsip_outbound_publish.c: Remove redundant flag check. + + Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353 + +2016-03-07 18:34 +0000 [4ebf9a938d] gtjoseph + + * res_pjsip: Add ability to identify by Authorization username + + A feature of chan_sip that service providers relied upon was the ability to + identify by the Authorization username. This is most often used when customers + have a PBX that needs to register rather than identify by IP address. From my + own experiance, this is pretty common with small businesses who otherwise + don't need a static IP. + + In this scenario, a register from the customer's PBX may succeed because From + will usually contain the PBXs account id but an INVITE will contain the caller + id. With nothing recognizable in From, the service provider's Asterisk can + never match to an endpoint and the INVITE just stays unauthorized. + + The fixes: + + A new value "auth_username" has been added to endpoint/identify_by that + will use the username and digest fields in the Authorization header + instead of username and domain in the the From header to match an endpoint, + or the To header to match an aor. This code as added to + res_pjsip_endpoint_identifier_user rather than creating a new module. + + Although identify_by was always a comma-separated list, there was only + 1 choice so order wasn't preserved. So to keep the order, a vector was added + to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar + to find the aor. The res_pjsip_endpoint_identifier_* modules are called in + globals/endpoint_identifier_order. + + Along the way, the logic in res_pjsip_registrar was corrected to match + most-specific to least-specific as res_pjsip_endpoint_identifier_user does. + + The order is: + + username@domain + username@domain_alias + username + + Auth by username does present 1 problem however, the first INVITE won't have + an Authorization header so the distributor, not finding a match on anything, + sends a securty_alert. It still sends a 401 with a challenge so the next + INVITE will have the Authorization header and presumably succeed. As a result + though, that first security alert is actually a false alarm. + + To address this, a new feature has been added to pjsip_distributor that keeps + track of unidentified requests and only sends the security alert if a + configurable number of unidentified requests come from the same IP in a + configurable amout of time. Those configuration options have been added to + the global config object. This feature is only used when auth_username + is enabled. + + Finally, default_realm was added to the globals object to replace the hard + coded "asterisk" used when an endpoint is not yet identified. + + The testsuite tests all pass but new tests are forthcoming for this new + feature. + + ASTERISK-25835 #close + Reported-by: Ross Beer + + Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d + +2016-04-27 13:23 +0000 [2b150f0b80] Mark Michelson + + * func_odbc: Check connection status before executing queries. + + A recent change to func_odbc made it so that a single connection was + maintained per DSN. The problem was that the code was optimistic about + the health of the connection after initially opening it and did nothing + to re-connect in case the connection had died. + + This change adds a check before executing a query to ensure that the + connection to the database is still up and running. + + ASTERISK-25963 #close + Reported by Ross Beer + + Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d + +2016-04-15 11:59 +0000 [860b135c88] Alexei Gradinari + + * res_pjsip: disable multi domain to improve realtime performace + + This patch added new global pjsip option 'disable_multi_domain'. + Disabling Multi Domain can improve Realtime performance by reducing + number of database requests. + + ASTERISK-25930 #close + + Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7 + +2016-04-01 07:50 +0000 [7281770710] Jean Aunis + + * app_chanspy: reduce audio loss on the spying channel. + + ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC + and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when + queues grow too large or when read and write queues go out of sync. + Now these flags are set conditionally: + - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set + - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not + be set on the audiohook + + ASTERISK-25866 + + Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd + +2016-04-14 07:03 +0000 [81ea80b74c] Joshua Colp + + * res_pjsip_exten_state: Add config support for exten state publishers. + + This change adds the ability to configure outbound publishing of + extension state. Right now stuff is merely set up to store the + configuration and to register a global extension state callback. The + act of constructing the body and sending is not yet complete. + + Configurable elements right now are a regex for filtering the context, + a regex for filtering the extension, and the body type to publish. + + ASTERISK-25922 #close + + Change-Id: Ia7e630136dfc355073c1cadff8ad394a08523d78 + +2016-04-26 11:13 +0000 [c480159045] Joshua Colp + + * chan_sip: Give more time for TCP/TLS threads to stop. + + The unload process currently tells each TCP/TLS to terminate but + does not wait for them to do so. This introduces a race condition + where the container holding the threads may be destroyed before + the threads are able to remove themselves from it. When they + finally do the container is invalid and can't be used causing a + crash. + + A previous change existed which waited a bit to wait for any + stranglers to finish. This change extends this and waits longer. + + ASTERISK-25961 #close + + Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6 + +2016-04-26 05:48 +0000 [8ae69cffef] Joshua Colp + + * app_queue: Fix crash when unloading module. + + When unloading the app_queue module the members in each queue are + destroyed and as part of this they are removed from the pending + members container. Unfortunately a crash would occur as the container + was destroyed before the members were removed. + + This change tweaks ordering so the container destruction occurs + after the members are destroyed. + + ASTERISK-16115 + + Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b + +2016-04-24 22:51 +0000 [284bb814ac] gtjoseph + + * config: Fix ast_config_text_file_save2 writability check for missing files + + A patch I did back in 2014 modified ast_config_text_file_save2 to check the + writability of the main file and include files before truncating and re-writing + them. An unintended side-effect of this was that if a file doesn't exist, + the check fails and the write is aborted. + + This patch causes ast_config_text_file_save2 to check the writability of the + parent directory of missing files instead of checking the file itself. This + allows missing files to be created again. A unit test was also added to + test_config to test saving of config files. + + The regression was discovered when app_voicemail's passwordlocation=spooldir + feature stopped working. + + ASTERISK-25917 #close + Reported-by: Jonathan Rose + + Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80 + +2016-04-25 08:11 +0000 [f99ec857c8] Javier Acosta + + * Fix case sensitive actions in AMI QueueSummary and QueueStatus + + ASTERISK-25954 #close + Reported by: Javier Acosta + + Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256 + (cherry picked from commit c0688a6398f27296ff849848a2e416e036d794e3) + +2016-04-21 14:23 +0000 [30ab21d5fa] Kevin Harwell + + * app_queue: queue members can receive multiple calls + + It was possible for a queue member that is a member of at least 2 or more + queues to receive mulitiple calls at the same time. This happened because + of a race between when a member was being rung and when the device state + notified the other queue(s) member object of the state change. + + This patch makes it so when a queue member is being rung it gets added to + a global pool of queue members. If that same member is tried again, e.g. + from another queue, and it is found to already exist in the pending member + container then it will not ring that member. + + ASTERISK-16115 #close + + Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48 + +2016-04-22 17:53 +0000 [99fcf2a791] gtjoseph + + * res_agi: Prevent run_agi from eating frames it shouldn't + + The run_agi function is eating control frames when it shouldn't be. This is + causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond + transfer. + + Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie + answers. + + Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE + and is left thinking he's connected to Bob. + + In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls + an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on + Charlie's channel. + + The fix was to accumulate deferrable frames in the "forever" loop instead of + dropping them, and re-queue them just before running the actual agi command + or exiting. + + ASTERISK-25951 #close + + Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645 + +2016-04-22 15:25 +0000 [757ec6172b] Richard Mudgett + + * test_message.c: Wait longer in case dialplan also processes the test message. + + Bumped the wait from 1 second to 5 seconds. The test message was hitting my + default call handler and failing the test because it took longer. + + Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba + +2016-04-21 23:53 +0000 [41ecf22587] Kirill Katsnelson + + * chan_sip: Make autocreated peers send PeerStatus events + + Since Stasis has been introduced, an attempt to send AMI messages by an + autocreated peer caused a crash, and all events from autocreated peers were + semi-inadvertently disabled altogether in 0b83761. This change restores the + disabled functionality. + + ASTERISK-25950 + + Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974 + +2016-04-13 17:09 +0000 [b3cc74fda9] Richard Mudgett + + * manager_channels.c: Fix allocation failure crash. + + An earlier allocation failure failed to create a channel snapshot for the + AMI HangupRequest/SoftHangupRequest event which resulted in a crash in + channel_hangup_request_cb(). Where the stasis message gets generated + cannot tell if the NULL snapshot returned was because of an allocation + failure or the channel was a dummy channel. + + * Made channel_hangup_request_cb() check if the channel blob has a + snapshot and exit if it doesn't. + + * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). + + Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24 + +2016-04-13 13:50 +0000 [a63656b419] Richard Mudgett + + * Bridge system: Fix memory leaks and double frees on impart failure. + + You cannot reference the passed in features struct after calling + ast_bridge_impart(). Even if the call fails. + + Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 + +2016-04-13 13:20 +0000 [71dfa35540] Richard Mudgett + + * bridge_softmix.c: Fix crash if channel fails to join mixing tech. + + softmix_bridge_join() failed because of an allocation failure. To address + this, the softmix bridge technology now checks if the channel failed to + join softmix successfully. In addition, the bridge now begins the process + of kicking the channel out of the bridge so we don't have channels + partially in the bridge for very long. + + * Fix the test_channel_feature_hooks.c unit tests. The test channel must + have a valid codec to join the simple_bridge technology. This patch makes + joining a bridge more strict by not allowing partially joined channels to + remain in the bridge. + + Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b + +2016-04-12 15:29 +0000 [06632a0d11] Richard Mudgett + + * Manager: Short circuit AMI message processing. + + Improve AMI message processing performance if there are no consumers + listening for the messages. We now skip creating the AMI event message + text strings. + + Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3 + +2016-04-13 17:54 +0000 [6ddd856b86] Richard Mudgett + + * manager.c: Eliminate most RAII_VAR usage. + + * Made ast_manager_event_blob_create() not allocate the ao2 event object + with a lock as it is not needed. + + Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c + +2016-04-22 13:49 +0000 [924738e950] Mark Michelson + + * func_odbc: Use one connection per DSN. + + res_odbc was changed in Asterisk 13.8.0 to remove connection management, + opting instead to let unixodbc maintain open connections and return + those to Asterisk as requested. + + This was a boon for realtime, since it meant that multiple threads could + potentially run parallel queries since they could each be using their + own database connections. + + However, on the user-facing side, func_odbc, there were some inherent + behaviors being relied on that no longer hold true after the change. + One such reported behavior was that MySQL's LAST_INSERTED_ID() works + per-connection. This means that if Asterisk uses separate connections + for every database operation, whereas before it used one connection for + everything, we have broken expectations and functionality. + + The fix provided in this patch is to make func_odbc use a single + database connection per DSN. This way, user-facing database usage will + have the same behavior as it did pre-13.8.0. However, realtime, which is + the real workhorse of database interaction, will continue to let + unixodbc manage connections. + + ASTERISK-25938 #close + Reported by Edwin Vandamme + + Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc + +2016-04-22 13:02 +0000 [6ede210c98] Leif Madsen + + * Remove reference to non-existent sip.conf option + + Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f + + ASTERISK-25927 #close + + Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8 + +2016-04-21 08:26 +0000 [c991e5472e] Diederik de Groot + + * lock.c: Check *lt before dereferencing it + + *lt is NULL if t->tracking == 0 + + ASTERISK-25948 #close + + Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba + +2016-04-15 14:36 +0000 [6b1a632290] Richard Mudgett + + * res_stasis: Handle re-enter stasis bridge with swap channel. + + We lose the fact that there is a swap channel if there is one. We + currently wind up rejoining the stasis bridge as a normal join after the + swap channel has already been kicked from the bridge. + + This patch preserves the swap channel so the AMI/ARI events can note that + the channel joining the bridge is swapping with another channel. Another + benefit to swaqpping in one operation is if there are any channels that + get lonely (MOH, bridge playback, and bridge record channels). The lonely + channels won't leave before the joining channel has a chance to come back + in under stasis if the swap channel is the only reason the lonely channels + are staying in the bridge. + + ASTERISK-25947 #close + Reported by: Richard Mudgett + + ASTERISK-24649 + Reported by: John Bigelow + + ASTERISK-24782 + Reported by: John Bigelow + + Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee + +2016-04-19 16:58 +0000 [1c5248c383] Richard Mudgett + + * bridge: Hold off more than one imparting channel at a time. + + An earlier patch blocked the ast_bridge_impart() call until the channel + either entered the target bridge or it failed. Unfortuantely, if the + target bridge is stasis and the imprted channel is not a stasis channel, + stasis bounces the channel out of the bridge to come back into the bridge + as a proper stasis channel. When the channel is bounced out, that + released the block on ast_bridge_impart() to continue. If the impart was + a result of a transfer, then it became a race to see if the swap channel + would get hung up before the imparted channel could come back into the + stasis bridge. If the imparted channel won then everything is fine. If + the swap channel gets hung up first then the transfer will fail because + the swap channel is leaving the bridge. + + * Allow a chain of ast_bridge_impart()'s to happen before any are + unblocked to prevent the race condition described above. When the channel + finally joins the bridge or completely fails to join the bridge then the + ast_bridge_impart() instances are unblocked. + + ASTERISK-25947 + Reported by: Richard Mudgett + + ASTERISK-24649 + Reported by: John Bigelow + + ASTERISK-24782 + Reported by: John Bigelow + + Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 + +2016-04-19 17:52 +0000 [70e860ec49] gtjoseph + + * res_pjsip_callerid: Clear out display name if id->name is not valid + + When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning + the From header, then it overwrites the display name and uri from the channel's + connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was + leaving the display name from the From header in the new RPID or PAI header. + On an attended transfer where the originator had a caller id number set but not + a display name, the re-INVITE to the final transferee had the number of the + originator but the display name of the transferer. + + Added a check to clear out the display name in the new header if + connected.id.name was invalid. + + ASTERISK-25942 #close + + Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b + +2016-04-19 13:02 +0000 [d95512a7dd] Joshua Colp + + * app_talkdetect: Make the module core supported. + + This module is used as part of testsuite tests to confirm + stuff works. I'm accordingly marking it as core as it is + required by those tests. + + Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88 + +2016-04-18 12:12 +0000 [0235a66532] Mark Michelson + + * PJSIP: Remove PJSIP parsing functions from uri length validation. + + The PJSIP parsing functions provide a nice concise way to check the + length of a hostname in a SIP URI. The problem is that in order to use + those parsing functions, it's required to use them from a thread that + has registered with PJLib. + + On startup, when parsing AOR configuration, the permanent URI handler + may not be run from a PJLib-registered thread. Specifically, this could + happen when Asterisk was started in daemon mode rather than + console-mode. If PJProject were compiled with assertions enabled, then + this would cause Asterisk to crash on startup. + + The solution presented here is to do our own parsing of the contact URI + in order to ensure that the hostname in the URI is not too long. The + parsing does not attempt to perform a full SIP URI parse/validation, + since the hostname in the URI is what is important. + + ASTERISK-25928 #close + Reported by Joshua Colp + + Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60 + +2016-04-18 17:00 +0000 [b8b60135ec] Mark Michelson + + * res_pjsip_registrar: Fix bad memory-ness with user_agent. + + Recent changes to the PJSIP registrar resulted in tests failing due to + missing AOR_CONTACT_ADDED test events. The reason for this was that the + user_agent string had junk values in it, resulting in being unable to + generate the event. + + I'm going to be honest here, I have no idea why this was happening. Here + are the steps needed for the user_agent variable to get messed up: + * REGISTER is received + * First contact in the REGISTER results in a contact being removed + * Second contact in the REGISTER results in a contact being added + * The contact, AOR, expiration, and user agent all have to be passed as + format parameters to the creation of a string. Any subset of those + parameters would not be enough to cause the problem. + + Looking into what was happening, the thing that struck me as odd was + that the user_agent variable was meant to be set to the value of the + User-Agent SIP header in the incoming REGISTER. However, when removing a + contact, the user_agent variable would be set (via ast_strdupa inside a + loop) to the stored contact's user_agent. This means that the + user_agent's value would be incorrect when attempting to process further + contacts in the incoming REGISTER. + + The fix here is to use a different variable for the stored user agent + when removing a contact. Correcting the behavior to be correct also + means the memory usage is less weird, and the issue no longer occurs. + + ASTERISK-25929 #close + Reported by Joshua Colp + + Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08 + +2016-04-18 13:41 +0000 [6cfa02394f] Joshua Colp + + * res_pjsip_transport_management: Allow unload to occur. + + At shutdown it is possible for modules to be unloaded that wouldn't + normally be unloaded. This allows the environment to be cleaned up. + + The res_pjsip_transport_management module did not have the unload + logic in it to clean itself up causing the res_pjsip module to not + get unloaded. As a result the res_pjsip monitor thread kept going + processing traffic and timers when it shouldn't. + + Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a + +2016-04-15 11:41 +0000 [6365f0018f] Richard Mudgett + + * bridge_channel.c: Ignore role setup failure in channel push. + + We have to setup the channel roles after the bridge class push is called + because the bridge class push callback may have set roles on the incoming + channel. Since we have already partially pushed the channel into the + bridge and reversing what we have already done could be problematic, the + only thing we can do is press on to complete pushing the channel into the + bridge. + + * Ignore any channel role setup errors after pushing the channel into a + bridge. The channel may behave incorrectly in the bridge but we can no + longer abort the push at this time. + + Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00 + +2016-04-17 15:37 +0000 [f06ce7f90a] Jaco Kroon + + * chan_sip: Don't verify table if rtupdate=no + + If rtupdate=no do not verify sipregs/peers table has updatable fields. + + ASTERISK-25934 #close + + Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d + +2016-04-18 04:53 +0000 [dbb47e0a47] ibercom + + * app_queue: Frequent segfaults in function can_ring_entry() + + ASTERISK-25888 #close + + Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117 + +2016-04-15 16:51 +0000 [af114edb8b] Richard Mudgett + + * stasis_bridge.c: Update stasis bridge push diagnostic messages. + + Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a + +2016-04-12 14:55 +0000 [5e64d7e7a3] Mark Michelson + + * Dial: Combine frame handling functions. + + There is a good amount of repetition in the two frame handling routines + in the Dial API. This commit combines the two functions into one. + + This is in preparation for an upcoming commit that adds the ability to + handle frames for a channel in a bridge. + + ASTERISK-25925 + Reported by Mark Michelson + + Change-Id: Iaae2f174e3058e774cb44e10659fcdfb85345c58 + +2016-04-11 16:20 +0000 [a6e2ba187a] Alexei Gradinari + + * Codecs: strip codec name while parsing allow/disallow options + + Failed registration using PJSIP/Realtime if one of the codec name + in allow/disallow option is wrong or contains space. + + This patch strip codec name. + + ASTERISK-25914 + + Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d + +2016-04-14 13:49 +0000 [be4333ddad] Mark Michelson + + * transport management: Register thread with PJProject. + + The scheduler thread that kills idle TCP connections was not registering + with PJProject properly and causing assertions if PJProject was built in + debug mode. + + This change registers the thread with PJProject the first time that the + scheduler callback executes. + + AST-2016-005 + + Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 + +2016-03-17 12:28 +0000 [e83499df56] gtjoseph + + * res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) + + There are several places that do scheduled tasks or periodic housecleaning, + each with its own implementation: + + * res_pjsip_keepalive has a thread that sends keepalives. + * pjsip_distributor has a thread that cleans up expired unidentified requests. + * res_pjsip_registrar_expire has a thread that cleans up expired contacts. + * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. + * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. + + There are also places where we should be doing scheduled work but aren't. + A good example are the places we have sorcery observers to start registration + or qualify. These don't work when changes are made to a backend database + without a pjsip reload. We need to check periodically. + + As a first step to solving these issues, a new ast_sip_sched facility has + been created. + + ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. + When a task is ready to run, ast_sip_task_pusk is called for it. This ensures + that the task is executed in a PJLIB registered thread and doesn't hold up the + ast_sched thread so it can immediately continue processing the queue. The + serializer used by ast_sip_sched is one of your choosing or a random one from + the res_pjsip pool if you don't choose one. + + Another feature is the ability to automatically clean up the task_data when the + task expires (if ever). If it's an ao2 object, it will be dereferenced, if + it's a malloc'd object it will be freed. This is selectable when the task is + scheduled. Even if you choose to not auto dereference an ao2 task data object, + the scheduler itself maintains a reference to it while the task is under it's + control. This prevents the data from disappearing out from under the task. + + There are two scheduling models. + + AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at + the specific interval. That is, every "interval" milliseconds, regardless of + how long the task takes. If the task takes longer than the interval, it will + be scheduled at the next available multiple of interval. For exmaple: If the + task has an interval of 60 secs and the task takes 70 secs (it better not), + the next invocation will happen at 120 seconds. + + AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should + start "interval" milliseconds after the current invocation has finished. + + Also, the same ast_sched facility for fixed or variable intervals exists. The + task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or + AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. + + One res_pjsip.h housekeeping change was made. The pjsip header files were + added to the top. There have been a few cases lately where I've needed + res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because + I didn't add the pjsip header files to my source even though I never referenced + any pjsip calls. + + Finally, a few new convenience APIs were added to astobj2 to make things a + little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and + ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros + were also copied from res_phoneprov because I got tired of having to duplicate + the same hash, sort and compare functions over and over again. The + AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for + aor_container_alloc into your source. + + This facility can be used immediately for the situations where we already have + a thread that wakes up periodically or do some scheduled work. For the + registration and qualify issues, additional sorcery and schema changes would + need to be made so that we can easily detect changed objects on a periodic + basis without having to pull the entire database back to check. I'm thinking + of a last-updated timestamp on the rows but more on this later. + + Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c + +2016-03-08 12:12 +0000 [216f22fd0f] Mark Michelson + + * res_pjsip_transport_management: Kill idle TCP connections. + + "Idle" here means that someone connects to us and does not send a SIP + request. PJProject will not automatically time out such connections, so + it's up to Asterisk to do it instead. + + When we receive an incoming TCP connection, we will start a timer + (equivalent to transaction timer D) waiting to receive an incoming + request. If we do not receive a request in that timeframe, then we will + shut down the TCP connection. + + ASTERISK-25796 #close + Reported by George Joseph + + AST-2016-005 + + Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 + +2016-03-08 10:52 +0000 [d9fba46016] Mark Michelson + + * Rename res_pjsip_keepalive res_pjsip_transport_management + + ASTERISK-25796 + Reported by George Joseph + + AST-2016-005 + + Change-Id: Id322a05f927392293570599730050bc677d99433 + +2016-04-14 07:23 +0000 [7b8b6e2e4f] Mark Michelson + + * AST-2016-004: Fix crash on REGISTER with long URI. + + Due to some ignored return values, Asterisk could crash if processing an + incoming REGISTER whose contact URI was above a certain length. + + ASTERISK-25707 #close + Reported by George Joseph + + Patches: + 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch + + AST-2016-004 + + Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d + +2016-04-12 13:10 +0000 [ff3af764de] Richard Mudgett + + * bridge_softmix.c: Fix crash if could not allocate the dsp. + + Fix off nominal crash where we could not setup the channel to process + frames for the softmix bridge technology because of allocation failure. + + Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372 + +2016-04-13 13:38 +0000 [caa416d5f3] gtjoseph + + * stringfields: Update extended string fields for master only. + + In 13, the new ast_string_field_header structure had to be dynamically + allocated and assigned to a pointer in ast_string_field_mgr to preserve ABI + compatability. In master, it can be converted to being a structure-in-place in + ast_string_field_mgr to eliminate the extra alloc and free calls. + + Change-Id: Ia97c5345eec68717a15dc16fe2e6746ff2a926f4 + +2016-04-12 15:41 +0000 [bd3671b397] gtjoseph + + * pjproject: Add patch for removing strip of '[]' from header params + + From the patch submitted to Teluu on 4/12/2016 + <<<<<<<<< + The wholesale stripping of '[]' from header parameters causes issues if + something (like a port) occurs after the final ']'. + + '[2001:a::b]' will correctly parse to '2001:a::b' + '[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left + with ':8080' and parsing stops with a syntax error. + + I can't even find a case where stripping the '[]' is a good thing anyway. Even + if you continued to parse and resulted in a string that looks like this... + '2001:a::b:8080', it's not valid. + + This came up in Asterisk because Kamailio sends us a Contact with an alias + URI parameter that has an IPv6 address in it like this: + Contact: + which should be legal but causes a syntax error because of the characters + after the final ']'. Even if it didn't, the '[]' should still not be stripped. + + I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6 + enabled. No issues were caused by removing the code that strips the '[]'. + >>>>>>>>>>> + + ASTERISK-25123 #close + Reported-by: Anthony Messina + + Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a + +2016-04-12 09:10 +0000 [5a0534dc62] Joshua Colp + + * app_voicemail: Fix test_voicemail_notify_endl test. + + The test_voicemail_notify_endl test checks the end-of-line + characters of an email message to confirm that they are consistent. + The test wrongfully assumed that reading from the email message + into a buffer will always result in more than 1 character being + read. This is incorrect. If only 1 character was read the test + would go outside of the buffer and access other memory causing + a crash. + + The test now checks to ensure that 2 or more characters are read + in ensuring the test stays within the buffer. + + ASTERISK-25874 #close + + Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710 + +2016-04-07 12:02 +0000 [c00c298a0e] Alexei Gradinari + + * app_voicemail/IMAP: function 'save_to_folder' creates wrong folder + + If try to move message to Cust1 (number 5) + the function 'save_to_folder' tries to create Greeting folder instead of Cust1. + + This patch fixed it by setting GREETINGS_FOLDER = -1 + + ASTERISK-24927 #close + + Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51 + +2016-04-07 16:18 +0000 [49813bc9e5] Alexei Gradinari + + * res_pjsip: Add headers to AMI Event ContactStatusDetail + + * Added Useragent and RegExpire headers to AMI Event + ContactStatusDetail with associated documentation. + + ASTERISK-25903 #close + + Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239 + +2016-04-05 16:56 +0000 [4e00e31ef1] Alexei Gradinari + + * res_pjsip_outbound_publish: Add transport for outbound PUBLISH + + The first available transport of the appropriate type is used now. + This patch adds new config option 'transport' for outbound-publish. + If transport is set then outbound PUBLISH requests will use this transport. + + ASTERISK-25901 #close + + Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151 + +2016-04-11 14:26 +0000 [2cc56573de] Jaco Kroon + + * core_unreal: Fix hangupcauses not getting set on Local channels + + ASTERISK-25912 #close + + Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa + +2016-04-01 13:30 +0000 [a621dd5e96] gtjoseph + + * res_pjsip contact: Lock expiration/addition of contacts + + Contact expiration can occur in several places: res_pjsip_registrar, + res_pjsip_registrar_expire, and automatically when anyone calls + ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar + may also be attempting to renew or add a contact. Since none of this was locked + it was possible for one thread to be renewing a contact and another thread to + expire it immediately because it was working off of stale data. This was the + casue of intermittent registration/inbound/nominal/multiple_contacts test + failures. + + Now, the new named lock functionality is used to lock the aor during contact + expire and add operations and res_pjsip_registrar_expire now checks the + expiration with the lock held before deleting the contact. + + ASTERISK-25885 #close + Reported-by: Josh Colp + + Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059 + +2016-04-10 14:16 +0000 [8637f29d24] gtjoseph + + * pjproject: Add patch to fix Via IPv6 parsing + + There's a bug in pjproject's sip_parser where the ":" wasn't correctly + interpreted. This is causing IPv6 addresses in the "received" parameter of the + Via header to cause a syntax check failure. + + This patch was submitted to Teluu on 4/10/2016. + + ASTERISK-25910 #close + Reported-by: Anthony Messina + + Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922 + +2016-03-31 20:04 +0000 [216abb0ae7] gtjoseph + + * lock: Add named lock capability + + Locking some objects like sorcery objects can be tricky because the underlying + ao2 object may not be the same for all callers. For instance, two threads that + call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 + different ao2 objects if the underlying wizard had to rehydrate the aor from a + database. Locking one ao2 object doesn't have any effect on the other even if + those objects had locks in the first place. + + Named locks allow access control by keyspace and key strings. Now an "aor" + named "1000" can be locked and any other thread attempting to lock "aor" "1000" + will wait regardless of whether the underlying ao2 object is the same or not. + Mutex and rwlocks are supported. + + This capability will initially be used to lock an aor when multiple threads may + be attempting to prune expired contacts from it. + + Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 + +2016-04-07 11:37 +0000 [f9dab80816] Alexei Gradinari + + * app_voicemail/IMAP: IMAP access FATAL error: Out of memory + + Sometimes uw-imap function 'mail_fetchbody' returns huge len + which then pass to uw-imap function 'rfc822_base64'. + uw-imap tries to allocate huge memory and abort() on fail. + + This patch check the len. + If the len more than max size (128 Mbytes) log error. + This patch also set variables len, newlen to avoid uninizialezed len. + This patch also check pointer returned by rfc822_base64. + + ASTERISK-25899 #close + + Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca + +2016-04-07 16:39 +0000 [b3be945415] Alexei Gradinari + + * res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event + + BLF pickup isn't working on Cisco SPA and Snom phones + if the direction="recipient" attribute is missing in 'dialog' tag. + + This patch adds direction="recipient" if extension state is + Ringing. + + ASTERISK-24601 #close + + Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c + +2016-04-06 17:57 +0000 [6138a75e8e] Richard Mudgett + + * pbx.h: Make ast_state_cb_type take more const. + + This eliminates some casts that I made a note saying v10 and above + would no longer need them. + + Better late than never :) + + Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572 + +2016-04-07 10:59 +0000 [72c19f7dc5] Richard Mudgett + + * pbx.c: Minor code rearangements. + + * Pull out a loop invariant. + + * Convert an else-if ladder to a switch statement. + + Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95 + +2016-04-07 12:26 +0000 [28cefc3e88] Richard Mudgett + + * pbx: Update doxygen for extension state watchers. + + Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e + +2016-04-07 11:49 +0000 [751d7a5a49] gtjoseph + + * alembic: Remove batch operations (and sqlite support) + + Because SQLite doesn't support full ALTER capabilities, alembic scripts + require batch operations. However, that capability wasn't available until + 0.7.0 which some distributions haven't reached yet. Therefore, the batch + operations introduced in commit 86d6e44cc (review 2319) have been reverted + and SQLite is unsupported again, for now anyway. + + Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql. + + ASTERISK-25890 #close + Reported-by: Harley Peters + + Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80 + +2016-04-07 11:05 +0000 [2eaeea690d] Joshua Colp + + * res_pjsip_registrar_expire: Fix race condition at shutdown. + + When shutting down, the PJSIP sorcery is destroyed. The registrar + expiration module queries the PJSIP sorcery to determine what + to expire. As there was no synchronization between termination + of the expiration thread and the unloading of the module it was + possible for the thread to try to access the PJSIP sorcery after + it had been destroyed. + + This change ensures that the thread is shut down before allowing + the module to be considered unloaded. + + Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b + +2016-04-06 16:28 +0000 [3e5672d843] Joshua Colp + + * res_pjsip: Fix configuration setting of "regcontext". + + Due to a merge problem two options were swapped causing the + regcontext setting to not get set. + + Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1 + +2016-04-06 08:01 +0000 [8ed5f61152] Jacek Konieczny + + * frame.c: Copy the whole subclass in ast_frdup(). + + The problem is ast_frdup() does not copy whole frame.subclass for voice, + video and image frames, only the format is copied. For video frames, the + subclass structure contains the .frame_ending flag used to put the RTP + marker where it needs to be. + + ASTERISK-25894 #close + + Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33 + +2016-03-30 17:18 +0000 [abbb2edd4c] Mark Michelson + + * ARI: Add method to Dial a created channel. + + This adds a new ARI method that allows for you to dial a channel that + you previously created in ARI. + + By combining this with the create method for channels, it allows for a + workflow where a channel can be created, manipulated, and then dialed. + The channel is under control of the ARI application during all stages of + the Dial and can even be manipulated based on channel state changes + observed within an ARI application. + + The overarching goal for this is to eventually be able to add a dialed + channel to a Stasis bridge earlier than the "Up" state. However, at the + moment more work is needed in the Dial and Bridge APIs in order to + facilitate that. + + ASTERISK-25889 #close + + Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205 + +2016-03-30 17:01 +0000 [dd48d60c5b] Mark Michelson + + * ARI: Add method to create a new channel. + + This adds a new ARI method to the channels resource that allows for the + creation of a new channel. The channel is created and then placed into + the specified Stasis application. + + This is different from the existing originate method that creates a + channel, dials it, and then places the answered channel into the + dialplan or a Stasis application. This method does not attempt to call + the channel at all. Dialing is left as a later step after channel + creation. This allows for pre-dialing channel manipulation if desired. + + ASTERISK-25889 + + Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8 + +2016-03-28 11:31 +0000 [1dc5e28624] Joshua Colp + + * pbx: Add support for autohints. + + This change introduces the concept of autohints. These are hints + which are created as a result of device state changes occurring within + the core. When this happens a hint will be created (if it does not + exist already) using the device name as the extension. + + For example if a device state change is received for "PJSIP/bob" + and autohints are enabled on a context then a hint will exist in + that context for "bob" with a device of "PJSIP/bob". + + For virtual or custom device states the name after the type will + be used. For example if the device state of "Custom:bob" changes + then a hint will exist in that context for "bob" with a device of + "Custom:bob". + + This functionality can be enabled in extensions.conf by placing + "autohints=yes" in a context. + + ASTERISK-25881 #close + + Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e + +2016-04-05 14:23 +0000 [a098251e7e] Mark Michelson + + * res_pjsip: Handle deferred SDP hold/unhold properly. + + Some SIP devices indicate hold/unhold using deferred SDP reinvites. In + other words, they provide no SDP in the reinvite. + + A typical transaction that starts hold might look something like this: + + * Device sends reinvite with no SDP + * Asterisk sends 200 OK with SDP indicating sendrecv on streams. + * Device sends ACK with SDP indicating sendonly on streams. + + At this point, PJMedia's SDP negotiator saves Asterisk's local state as + being recvonly. + + Now, when the device attempts to unhold, it again uses a deferred SDP + reinvite, so we end up doing the following: + + * Device sends reinvite with no SDP + * Asterisk sends 200 OK with SDP indicating recvonly on streams + * Device sends ACK with SDP indicating sendonly on streams + + The problem here is that Asterisk offered recvonly, and by RFC 3264's + rules, if an offer is recvonly, the answer has to be sendonly. The + result is that the device is not taken off hold. + + What is supposed to happen is that Asterisk should indicate sendrecv in + the 200 OK that it sends. This way, the device has the freedom to + indicate sendrecv if it wants the stream taken off hold, or it can + continue to respond with sendonly if the purpose of the reinvite was + something else (like a session timer refresher). + + The fix here is to alter the SDP negotiator's state when we receive a + reinvite with no SDP. If the negotiator's state is currently in the + recvonly or inactive state, then we alter our local state to be + sendrecv. This way, we allow the device to indicate the stream state as + desired. + + ASTERISK-25854 #close + Reported by Robert McGilvray + + Change-Id: I7615737276165eef3a593038413d936247dcc6ed + +2016-03-30 16:47 +0000 [ef4d3f1328] Mark Michelson + + * Dial: Add function to append already-created channel. + + The Dial API takes responsiblity for creating an outbound channel when + calling ast_dial_append(). This commit adds a new function, + ast_dial_append_channel(), which allows us to create the channel outside + the Dial API and then to append the channel to the ast_dial structure. + + This is useful for situations where the channel's creation and dialing + are distinct operations. Upcoming ARI early bridge work will illustrate + its usage. + + ASTERISK-25889 + + Change-Id: Id8179f64f8f99132f80dead8d5db2030fd2c0509 + +2016-03-27 23:33 +0000 [984d6fd95c] gtjoseph + + * config: Allow filters when appending to a category + + In sorcery based config files where there are multiple categories with the same + name, you can't use the (+) operator to reliably append to a category because + config.c stops looking when it finds the first one with the same name. + + Example: + + [1000] + type = endpoint + + [1000] + type = aor + + [1000](+) + authenticate_qualify = yes + + This config will fail because config.c appends authenticate_qualify to the + first category it finds, the endpoint, and that's not valid for endpoint. + + Solution: + + The capability to find a category that contains a certain variable already + exists so the only real change was to parse anything after the '+' that's not a + comma, as a filter string. + + [1000] + type = endpoint + + [1000] + type = aor + + [1000](+type=aor) + authenticate_qualify = yes + + This now works as expected. + + Although the following example doesn't make any sense for pjsip, you can even + specify multiple filters: + + [1000](+type=aor&qualify_frequency=10) + + ASTERISK-25868 #close + Reported-by: Nick Repin + + Change-Id: I10773da4c79db36fbf1993961992af63d3441580 + +2016-04-05 10:21 +0000 [784fb43f43] Joshua Colp + + * res_http_websocket: Make core supported. + + Websockets are a core part of ARI support and as such this + module should also be core supported. + + Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c + +2016-03-25 23:22 +0000 [4d40b161c3] gtjoseph + + * stringfields: Refactor to allow fields to be added to the end of structures + + String fields are great, except that you can't add new ones without breaking + ABI compatibility because it shifts down everything else in the structure. + The only alternative is to add your own char * field to the end of the + structure and manage the memory yourself which isn't ideal, especially since + you then can't use the OPT_STRINGFIELD_T type. + + Background: + + The reason string fields had to be declared inside the + AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared + fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS + declared the pool, then the fields, then the manager, you could use the offsets + of the pool and manager and iterate over the sequential addresses in between to + access the fields. The actual pool, field allocation and field set operations + don't actually care where the field is. It's just iteration over the fields + that was the problem. + + Solution: Extended String Fields + + An extended string field is one that is declared outside the + AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent + structure. Other than using AST_STRING_FIELD_EXTENDED instead of + AST_STRING_FIELD, it looks the same as other string fields. It's storage comes + from the pool and it participates in string field compare and copy operations + peformed on the parent structure. It's also a valid target for the + OPT_STRINGFIELD_T aco option type. + + Implementation: + + To keep track of the extended fields and make sure that ABI isn't broken, the + existing embedded_pool pointer in the manager structure was repurposed to be a + pointer to a separate header structure that contains the embedded_pool pointer + plus a vector of fields. The length of the manager structure didn't change and + the embedded_pool pointer isn't used in the macros, only the stringfields C + code. A side benefit of this is that changing the header structure in the + future won't break ABI. + + ast_string_fields_init initializes the normal string fields and appends them to + the vector, and subsequent calls to ast_string_field_init_extended initialize + and append the extended fields. Cleanup, ast_string_fields_cmp, and + ast_string_fields_copy can now work on the vector instead of sequentially + traversing the addresses between the pool and manager. + + The total size of a structure using string fields didn't change, whether using + extended fields or not, nor have the offsets of any structure members, either + inside the original block or outside. Adding an extended field to the end of a + structure is the same as adding a char *. + + Details: + + The stringfield C code was pulled out from utils.c and into stringfields.c. + It just made sense. + + Additional work was done in ast_string_field_init and + ast_calloc_with_stringfields to handle the allocation of the new header + structure and the vector, and the associated cleanup. In the process some + additional NULL pointer checking was added. + + A lot of work was done in stringfields.h since the logic for compare and copy + is there. Documentation was added as well as somne additional NULL checking. + + The ability to call ast_calloc_with_stringfields with a number of structures + greater than 1 never really worked. Well, the calloc worked but there was no + way to access the additional structures or clean them up. It was agreed that + there was no use case for requesting more than 1 structure so an ast_assert + was added to prevent it and the iteration code removed. + + Testing: + + The stringfield unit tests were updated to test both normal and extended + fields. Tests for ast_string_field_ptr_set_by_fields and + ast_calloc_with_stringfields were also added. + + As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except + res_pjsip itself, saved off. The patch was then added and a full compile and + install was performed. Then the older res_pjsip_* moduled were copied over the + installed versions so res_pjsip was new and the rest were old. No issues. + + contact->aor, which is a char * at the end of contact, was then changed to an + extended string field and a recompile and reinstall was performed, again + leaving stock versions of the the res_pjsip_* modules. Again, no issues with + the res_pjsip_* modules using the old stringfield implementation and with + contact->aor as a char *, and res_pjsip itself using the new stringfield + implementation and contact->aor being an extended string field. + + Finally, several existing string fields were converted to extended string + fields to test OPT_STRINGFIELD_T. Again, no issues. + + Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61 + +2016-04-04 18:02 +0000 [c07e1190ec] gtjoseph + + * res_pjsip_mwi: Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7 + + I forgot the new voicemail_extension wasn't a stringfield and didn't check + for NULL where I should have. + + Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb + +2016-04-03 11:47 +0000 [060b7b83bc] gtjoseph + + * install_prereq: Fix check_installed_debs remove subversion + + check_installed_debs wasn't handling virtual packages like libsrtp-dev and + libresample-dev and on multiarch systems it was accidentally filtering out all + packages if any :i386 packages were found instead of just filtering out the + :i386 packages themselves. + + Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda + +2016-04-01 13:09 +0000 [433d2c4bbf] gtjoseph + + * utils.c: Fix typo in handle_show_locks + + ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e). + + Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf + +2016-03-30 18:34 +0000 [304f81780d] gtjoseph + + * pjproject_bundled: Fix use of LDCONFIG for shared library link creation + + LDCONFIG apparently isn't set to something sane on all systems so the creation + of the shared library links fails. Instead of just testing for non-blank, + main/Makefile now checks that LDCONFIG is actually executable and reverts to + LN if it isn't. + + This applies to both libasteriskpj and libasteriskssl. + + Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG. + + ASTERISK-25873 #close + Reported-by: Hans van Eijsden + + Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729 + +2016-03-29 13:47 +0000 [0ea742d33a] Richard Mudgett + + * res_stasis: Add control ref to playback and recording structs. + + The stasis_app_playback and stasis_app_recording structs need to have a + struct stasis_app_control ref. Other threads can get a reference to the + playback and recording structs from their respective global container. + These other threads can then use the control pointer they contain after + the control struct has gone. + + * Add control ref to stasis_app_playback and stasis_app_recording structs. + + With the refs added, the control command queue can now have a circular + control reference which will cause the control struct to never get + released if the control's command queue is not flushed when the channel + leaves the Stasis application. Also the command queue needs better + protection from adding commands if the control->is_done flag is set. + + * Flush the control command queue on exit. + + ASTERISK-25882 #close + + Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d + +2016-03-28 18:10 +0000 [53f63ad770] Richard Mudgett + + * res_stasis: Fix crash on a hanging up channel. + + * Give the struct stasis_app_control ao2 object a ref to the channel held + in the object. Now the channel will still be around if a thread needs to + post a stasis message instead of crash because the topic was destroyed. + + * Moved stopping any lingering silence generator out of the struct + stasis_app_control destructor and made it a part of exiting the Stasis + application. Who knows which thread the destructor will be called under + so it cannot affect the channel's silence generator. Not only was the + channel unprotected when the silence generator was stopped, stasis may no + longer even control the channel. + + ASTERISK-25882 + + Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4 + +2016-03-30 13:31 +0000 [2fab4d7da8] Richard Mudgett + + * res_stasis.c: Protect channel datastore list from stasis end. + + Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95 + +2016-03-29 18:06 +0000 [ece2edaa04] Richard Mudgett + + * res_ari: Cannot get control also means channel is unavailable. + + The only caller of ari_bridges_play_found() has this note: + + If ari_bridges_play_found fails because the channel is unavailable for + playback, The channel will be removed from the playback list soon. We can + keep trying to get channels from the list until we either get one that + will work or else there isn't a channel for this bridge anymore, in which + case we'll revert to ari_bridges_play_new. + + Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6 + +2016-03-29 14:29 +0000 [2f36cba4b5] Richard Mudgett + + * res_stasis_recording.c: Cleanup stasis_app_recording_find_by_name(). + + Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7 + +2016-03-28 14:23 +0000 [34457dd9db] Richard Mudgett + + * core_unreal.c: Add clarification comment about channel ref. + + Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3 + +2016-03-30 12:38 +0000 [2b3261cd36] gtjoseph + + * res_pjsip_mwi: Allow subscribe to vm access extension as an alias + + Background: + + If your extension is 1000 and the voicemail access extension is 1571 and you + dial 1571, usually a dialplan rule calls voicemailmain with your extension and + you are placed directly in your mailbox. Therefore most admins program the + voicemail (or other speed dial) button on their phones to the access extension. + Some phones (Snom at least) use whatever is programmed there to also subscribe + for MWI and so can't dial one number and subscribe to another. This works fine + in chan_sip because chan_sip completely ignores the user portion of the + SUBSCRIBE message request URI. If it can match the peer, is subscribes to the + peer's mailbox. The user could be set to anything or nothing and you'd still + get subscribed to your mailbox. + + Issue: + + chan_pjsip actually uses the user portion of the URI to find an aor and its + mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can + create an aor for 1571 but you certainly can't add your entire voicemail + system's mailboxes to it and everyone would get notified of every MWI. + + Solution: + + When an MWI subscribe comes in and an aor can't be found that matches the + resource directly, check the resource against the endpoint's aors. If an aor + is found that has a voicemail_extension that matches the resource, use it. + + ASTERISK-25865 + Reported-by: Ross Beer + + Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e + +2016-03-24 22:55 +0000 [e2524fcee3] gtjoseph + + * res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited + + res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds + the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes + on endpoints for unsolicited mwi and on aors for subscriptions required + that the admin know in advance which the client wanted. If you specified + mailboxes on the endpoint, subscriptions were rejected even if you also + specified mailboxes on the aor. + + Voicemail extension: + * Added a global default_voicemail_extension which defaults to "". + * Added voicemail_extension to both endpoint and aor. + * Added ast_sip_subscription_get_dialog for support. + * Added ast_sip_subscription_get_sip_uri for support. + + When an unsolicited NOTIFY is constructed, the From header is parsed, the + voicemail extension from the endpoint is substituted for the user, and the + result placed in the Message-Account field in the body. + + When a subscribed NOTIFY is constructed, the subscription dialog local uri + is parsed, the voicemail_extension from the aor (looked up from the + subscription resource name) is substituted for the user, and the result + placed in the Message-Account field in the body. + + If no voicemail extension was defined, the Message-Account field is not added + to the NOTIFY body. + + mwi_subscribe_replaces_unsolicited: + * Added mwi_subscribe_replaces_unsolicited to endpoint. + + The previous behavior was to reject a subscribe if a previous internal + subscription for unsolicited MWI was found for the mailbox. That remains the + default. However, if there are mailboxes also set on the aor and the client + subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal + subscription is removed and replaced with the external subscription. This + allows an admin to configure mailboxes on both the endpoint and aor and allows + the client to select which to use. + + ASTERISK-25865 #close + Reported-by: Ross Beer + + Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea + +2016-03-30 09:46 +0000 [724b9ab28f] gtjoseph + + * res_rtp_asterisk: Fix placement of txcount increment + + Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount + for rtcp packets as well as rtp packets and that was causing sender reports + to be generated instead of receiver reports in cases where no rtp was actually + being sent. + + Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp, + to rtp_sento which only handles rtp packets. + + Discovered by the hep/rtcp-receiver test. + + Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5 + +2016-03-26 22:33 +0000 [c4064727d2] gtjoseph + + * chan_pjsip: Add 'pjsip show channelstats' + + Added the ability to show channel statistics to chan_pjsip (cli_functions.c) + + Moved the existing 'pjsip show channel(s)' functionality from + pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's + private header so it made sense to move the existing channel commands as well. + + Now using stasis_cache_dump to get the channel snapshots rather than retrieving + all endpoints, then getting each one's channel snapshots. Much more efficient. + + Change-Id: I03b114522126d27434030b285bf6d531ddd79869 + +2016-03-25 10:59 +0000 [970803efcb] Jacek Konieczny + + * res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS + + Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 + explicitly states: + + There MUST be a separate DTLS-SRTP session for each distinct pair of + source and destination ports used by a media session + + This means RTP keying material cannot be used for DTLS RTCP, which was + the reason why RTCP encryption would fail. + + ASTERISK-25642 + + Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a + +2016-03-25 10:42 +0000 [9785e8d090] Jacek Konieczny + + * app_echo: forward and generate VIDUPDATE frames + + When using app_echo via WebRTC with VP8 video the video would appear + only after a few minutes, because there would be nothing to request + a full reference frame. + + This fixes the problem in both ways: + - echos any VIDUPDATE frames received on the channel + - sends one such frame when first video frame is to be forwarded + + This makes the echo work with Firefox and Chrome WebRTC implementation. + + ASTERISK-25867 #close + + Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e + +2016-03-27 12:53 +0000 [44ffb5105a] gtjoseph + + * res_rtp_asterisk: Fix packet stats on bridged connection + + rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated + for bridged streams because the calulations were being done after the + bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated. + + Moved the calculations so they occur for all valid received packets and + all transmitted packets. Also added rxoctetcount and txoctetcount to + ast_rtp_instance_stat. + + Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb + +2016-03-10 19:52 +0000 [c971a64366] gtjoseph + + * res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS + + No one seemed to notice but every time an OPTIONS goes out, it goes + out with a From of "asterisk" (or whatever the default from_user is set to), + even if you specify an endpoint. + + The issue had several causes... + qualify_contact is only called with an endpoint if called from the CLI. + If the endpoint is NULL, qualify_contact only looks up the endpoint if + authenticate_qualify=yes. Even then, it never passes it on to + ast_sip_create_request where the From header is set. Therefore From + is always "asterisk" (or whatever the default from_user is set to). + Even if ast_sip_create_request were to get an endpoint, it only sets + the From if endpoint->from_user is set. + + The fix is 4 parts... + + First, create_out_of_dialog_request was modified to use the endpoint id + if endpoint was specified and from_user is not set. + + Second, qualify_contact was modified to always look up an endpoint if + one wasn't specified regardless of authenticate_qualify. It then passes + the endpoint on to create_out_of_dialog_request. + + Third (and most importantly), find_an_endpoint was modified to find + an endpoint by using an "aors LIKE %contact->aor%" predicate with + ast_sorcery_retrieve_by_fields. As such, this patch will only work + if the sorcery realtime optimizations patch goes in. Otherwise we'd + be pulling the entire endpoints database every time we send an OPTIONS. + Since we already know the contact's aor, the on_endpoint callback was also + modified to just check if the contact->aor is an exact match to one of + the endpoint's. + + Finally, since we now have an endpoint for every OPTIONS request, + res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was + updated to get the transport from the endpoint and set it on tdata. + Now the correct transport is used. + + Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af + +2016-03-08 15:55 +0000 [c948ce9651] gtjoseph + + * sorcery/res_pjsip: Refactor for realtime performance + + There were a number of places in the res_pjsip stack that were getting + all endpoints or all aors, and then filtering them locally. + + A good example is pjsip_options which, on startup, retrieves all + endpoints, then the aors for those endpoints, then tests the aors to see + if the qualify_frequency is > 0. One issue was that it never did + anything with the endpoints other than retrieve the aors so we probably + could have skipped a step and just retrieved all aors. But nevermind. + + This worked reasonably well with local config files but with a realtime + backend and thousands of objects, this was a nightmare. The issue + really boiled down to the fact that while realtime supports predicates + that are passed to the database engine, the non-realtime sorcery + backends didn't. + + They do now. + + The realtime engines have a scheme for doing simple comparisons. They + take in an ast_variable (or list) for matching, and the name of each + variable can contain an operator. For instance, a name of + "qualify_frequency >" and a value of "0" would create a SQL predicate + that looks like "where qualify_frequency > '0'". If there's no operator + after the name, the engines add an '=' so a simple name of + "qualify_frequency" and a value of "10" would return exact matches. + + The non-realtime backends decide whether to include an object in a + result set by calling ast_sorcery_changeset_create on every object in + the internal container. However, ast_sorcery_changeset_create only does + exact string matches though so a name of "qualify_frequency >" and a + value of "0" returns nothing because the literal "qualify_frequency >" + doesn't match any name in the objset set. + + So, the real task was to create a generic string matcher that can take a + left value, operator and a right value and perform the match. To that + end, strings.c has a new ast_strings_match(left, operator, right) + function. Left and right are the strings to operate on and the operator + can be a string containing any of the following: = (or NULL or ""), !=, + >, >=, <, <=, like or regex. If the operator is like or regex, the + right string should be a %-pattern or a regex expression. If both left + and right can be converted to float, then a numeric comparison is + performed, otherwise a string comparison is performed. + + To use this new function on ast_variables, 2 new functions were added to + config.c. One that compares 2 ast_variables, and one that compares 2 + ast_variable lists. The former is useful when you want to compare 2 + ast_variables that happen to be in a list but don't want to traverse the + list. The latter will traverse the right list and return true if all + the variables in it match the left list. + + Now, the backends' fields_cmp functions call ast_variable_lists_match + instead of ast_sorcery_changeset_create and they can now process the + same syntax as the realtime engines. The realtime backend just passes + the variable list unaltered to the engine. The only gotcha is that + there's no common realtime engine support for regex so that's been noted + in the api docs for ast_sorcery_retrieve_by_fields. + + Only one more change to sorcery was done... A new config flag + "allow_unqualified_fetch" was added to reg_sorcery_realtime. + "no": ignore fetches if no predicate fields were supplied. + "error": same as no but emit an error. (good for testing) + "yes": allow (the default); + "warn": allow but emit a warning. (good for testing) + + Now on to res_pjsip... + + pjsip_options was modified to retrieve aors with qualify_frequency > 0 + rather than all endpoints then all aors. Not only was this a big + improvement in realtime retrieval but even for config files there's an + improvement because we're not going through endpoints anymore. + + res_pjsip_mwi was modified to retieve only endpoints with something in + the mailboxes field instead of all endpoints then testing mailboxes. + + res_pjsip_registrar_expire was completely refactored. It was retrieving + all contacts then setting up scheduler entries to check for expiration. + Now, it's a single thread (like keepalive) that periodically retrieves + only contacts whose expiration time is < now and deletes them. A new + contact_expiration_check_interval was added to global with a default of + 30 seconds. + + Ross Beer reports that with this patch, his Asterisk startup time dropped + from around an hour to under 30 seconds. + + There are still objects that can't be filtered at the database like + identifies, transports, and registrations. These are not going to be + anywhere near as numerous as endpoints, aors, auths, contacts however. + + Back to allow_unqualified_fetch. If this is set to yes and you have a + very large number of objects in the database, the pjsip CLI commands + will attempt to retrive ALL of them if not qualified with a LIKE. + Worse, if you type "pjsip show endpoint " guess what's going to + happen? :) Having a cache helps but all the objects will have to be + retrieved at least once to fill the cache. Setting + allow_unqualified_fetch=no prevents the mass retrieve and should be used + on endpoints, auths, aors, and contacts. It should NOT be used for + identifies, registrations and transports since these MUST be + retrieved in bulk. + + Example sorcery.conf: + + [res_pjsip] + endpoint=config,pjsip.conf,criteria=type=endpoint + endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error + + ASTERISK-25826 #close + Reported-by: Ross Beer + Tested-by: Ross Beer + + Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67 + +2016-03-25 23:19 +0000 [8e8cf80cea] Philip Correia + + * res_parking: Fix blind transfer dynamic lots creation. + + Blind transfers to a recognized parking extension need to use the parker's + channel variable values to create the dynamic parking lot. This is + because there is always only one parker while the parkee may actually be a + multi-party bridge. A multi-party bridge can never supply the needed + channel variables to create the dynamic parking lot. In the multi-party + bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and + channel variables are inherited by the local channel used to park the + bridge. + + * In park_common_setup(), make use the parker instead of the parkee to + supply the dynamic parking lot channel variable values. In all but one + case, the parkee is the same as the parker. However, in the recognized + parking extension blind transfer scenario for a two party bridge they are + different channels. For consistency, we need to use the parker channel. + + * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the + local channel when blind transferring a multi-party bridge to a recognized + parking extension. + + * When a local channel starts a call, the Local;2 side needs to inherit + the CHANNEL(parkinglot) value from Local;1. + + The DTMF one-touch parking case wasn't even trying to create dynamic + parking lots before it aborted the attempt. + + * In parking_park_call(), add missing code to create a dynamic parking + lot. + + A DTMF bridge hook is documented as returning -1 to remove the hook. + Though the hook caller is really coded to accept non-zero. See the + ast_bridge_hook_callback typedef. + + * In feature_park_call(), don't remove the DTMF one-touch parking hook + because of an error. + + ASTERISK-24605 #close + Reported by: Philip Correia + Patches: + call_park.patch (license #6672) patch uploaded by Philip Correia + + Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9 + +2016-03-23 14:24 +0000 [3cf714031c] Richard Mudgett + + * res_parking: Cleanup find_channel_parking_lot_name() usage. + + Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02 + +2016-03-18 14:01 +0000 [13e75ee04f] Richard Mudgett + + * res_parking: Misc fixes. + + res/parking/parking_applications.c: + + * Add malloc fail checks in setup_park_common_datastore(). + + * Fix playing parking failed announcement to only happen on non-blind + transfers in park_app_exec(). It could never go out before because a test + was provedly always false. + + res/parking/parking_bridge.c: + + * Fix NULL tolerance in generate_parked_user() because + bridge_parking_push() can theoretically pass a NULL parker channel if the + parker channel went away for some reason. + + * Clarify some weird code dealing with blind_transfer in + bridge_parking_push(). + + res/parking/parking_bridge_features.c: + + * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel + which will be bulk copied to the Local;2 channel on the subsequent + ast_call(). The additional advantage is if the parker channel has the + BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed + to be overridden. + + res/parking/parking_manager.c: + + * Fix AMI Park action input range checking of the Timeout header in + manager_park(). + + * Reduced locking scope to where needed in manager_park(). + + res/res_parking.c: + + * Fix some off nominal missing unlocks by eliminating the returns. + + Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca + +2014-12-15 05:23 +0000 [e2853ae337] Philip Correia + + * res_parking: Update parking documentation for dynamic parking lots. + + * Remove duplicate res_parking.conf courtesytone config option + documentation. + + ASTERISK-24596 #close + Reported by: Philip Correia + + ASTERISK-24605 + Reported by: Philip Correia + Patches: + call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia + + Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5 + +2016-03-25 06:02 +0000 [72a897c534] Joshua Colp + + * media_cache: Demote warning to debug as it may occur often. + + The file playback system will now query the media cache and then + the old file functionality. Under normal conditions this will result + in the cache failing to retrieve a file causing a warning message + to get output each time a file is played back. + + This change demotes this warning to a debug message. + + Change-Id: Ib72246ba300b5cce32774bfb3c26634bfb708624 +2016-03-10 16:58 +0000 [89e94e886c] Mark Michelson + + * Restrict CLI/AMI commands on shutdown. + + During stress testing, we have frequently seen crashes occur because a + CLI or AMI command attempts to access information that is in the process + of being destroyed. + + When addressing how to fix this issue, we initially considered fixing + individual crashes we observed. However, the changes required to fix + those problems would introduce considerable overhead to the nominal + case. This is not reasonable in order to prevent a crash from occurring + while Asterisk is already shutting down. + + Instead, this change makes it so AMI and CLI commands cannot be executed + if Asterisk is being shut down. For AMI, this is absolute. For CLI, + though, certain commands can be registered so that they may be run + during Asterisk shutdown. + + ASTERISK-25825 #close + + Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990 + +2016-03-24 14:08 +0000 [3f720155b7] Alexander Traud + + * chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. + + Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those + codecs, which the caller did not request/support. That fix was not complete + because on the second Session Timer all codecs were sent again. Some VoIP/SIP + clients interpreted that complete codec-list as a change in the SIP session. + Because of that, Asterisk did not send the RTP audio via NAT anymore which + created a non-audio scenario after the second Session Timer fired. + + ASTERISK-24543 #close + + Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66 + +2016-03-19 07:34 +0000 [894071ea2c] Gianluca Merlo + + * config: fix flags in uint option handler + + The configuration unsigned integer option handler sets flags for the + parser as if the option should be a signed integer (PARSE_INT32), + leading to errors on "out of range" values. Fix flags (PARSE_UINT32). + + A fix to res_pjsip is also present which stops invalid flags from + being passed when registering sorcery object fields for qualify + status. + + ASTERISK-25612 #close + + Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e + +2016-03-24 07:51 +0000 [13cdf3e8a1] Walter Doekes + + * musiconhold: Only warn if music class is not found in memory and database. + + The log message when a MusicOnHold music class was not found was changed + from debug level to WARNING level in Asterisk 11.19 and 13.5. For those + using realtime musiconhold, this message is wrong because it warns + before checking the database. + + This changeset delays the warning until after the database has been + checked. + + Reported-by: Conrad de Wet + ASTERISK-25444 #close + + Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf + +2016-03-24 05:48 +0000 [87c9ab97ea] Walter Doekes + + * core/logging: Fix broken syslog levels on older glibc. + + The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However + this macro is broken in older glibc (< 2.17); it would left-shift the + facility a second time, causing the resultant priority to become + invalid. + + The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: + + The priority argument is formed by ORing the facility and the level + values [...]. + + ASTERISK-25510 #close + Reported by: Michael Newton + + Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03 +2016-03-24 06:18 +0000 [a72f3b5bb4] Joshua Colp + + * tests/test_http_media_cache: Fix file descriptor leak in test. + + Change-Id: Ie8a9ae3d13bdeaacafc8d28271adc6707f633a5f + +2016-02-28 19:05 +0000 [13efea24f7] Matt Jordan + + * main/app: Only look to end of file if ':end' is specified, and not just ':' + + There is a little known feature in app_controlplayback that will cause the + specified offset to be used relative to the end of a file if a ':end' is + detected within the filename. + + This feature is pretty bad, but okay. + + However, a bug exists in this code where a ':' detected in the filename + will cause the end pointer to be non-NULL, even if the full ':end' isn't + specified. This causes us to treat an unspecified offset (0) as being + "start playing from the end of the file", resulting in no file playback + occurring. + + This patch fixes this bug by resetting the end pointer if ':end' is not + found in the filename. + + Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35 + +2015-12-26 15:29 +0000 [ca14b99e6e] Matt Jordan + + * main/file: Add the ability to play media in the media cache + + This patch allows applications/APIs that access media through the core file + APIs to play media in the media cache. Prior to determining if a 'filename' + exists, the filename is passed to the media cache's retrieve API call. If + that call succeeds, the local file specified passed back by the API is + opened for streaming. When used in this fashion, the 'filename' is actually + a URI that the media cache process and understand. + + ASTERISK-25654 #close + + Change-Id: I73b6e2e90c3e91b8500581c45cdf9c0dc785f5f0 + +2015-12-30 10:52 +0000 [01962a3932] Matt Jordan + + * tests/test_http_media_cache: Add unit tests for res_http_media_cache + + This patch adds unit tests for res_http_media cache, that covers nominal + creation and retrieval - and through them as well, staleness and deletion + checks. In addition, this patch adds tests that covers the interaction of + various HTTP headers, including Expires, Etag, and Cache-Control. + + ASTERISK-25654 + + Change-Id: I2db101e307c863857fe416d6f5bf4cace9ac7cf5 + +2015-01-29 08:38 +0000 [22e2340813] Matt Jordan + + * res/res_http_media_cache: Add an HTTP(S) backend for the core media cache + + This patch adds a bucket backend for the core media cache that interfaces to a + remote HTTP server. When a media item is requested in the cache, the cache will + query its bucket backends to see if they can provide the media item. If that + media item has a scheme of HTTP or HTTPS, this backend will be invoked. + + The backend provides callbacks for the following: + * create - this will always retrieve the URI specified by the provided + bucket_file, and store it in the file specified by the object. + * retrieve - this will pull the URI specified and store it in a temporary + file. It is then up to the media cache to move/rename this file + if desired. + * delete - destroys the file associated with the bucket_file. + * stale - if the bucket_file has expired, based on received HTTP headers from + the remote server, or if the ETag on the server no longer matches + the ETag stored on the bucket_file, the resource is determined to be + stale. + + Note that the backend respects the ETag, Expires, and Cache-Control headers + provided by the HTTP server it is querying. + + ASTERISK-25654 + + Change-Id: Ie201c2b34cafc0c90a7ee18d7c8359afaccc5250 + +2015-12-26 15:31 +0000 [791b4c9f81] Matt Jordan + + * main/media_cache: Provide an extension on the local file associated with a URI + + This patch does the following: + + First, it addresses file extension handling in the media cache. The media core + in Asterisk is a bit interesting in that it wants: + * A file to have an extension on it. That extension is used to associate the + file with a defined format module. + * The filename passed to the core to not have an extension on it. This allows + the core to match the available file formats with the format a channel + is capable of handling. + + Unfortunately, this makes the current implementation a bit lacking in the media + cache. By default, we do not store the extension of a retrieved URI on the + local file that is created. As a result, the media core does not know what + format the file is, and the file is ignored. Modifying the file outside of the + media core is bad, as we would not be able to update the internal + ast_bucket_file's path. + + At the same time, we do not want to pass the extension out in the file_path + parameter in ast_media_cache_retrieve. This parameter is intended to be fed + into the media core; if we passed the extension, all callers would have to + strip it off. + + Thus, this patch does the following: + * If there is an extension specified in the URL, we append it to the local + file name (if a preferred file name isn't specified), and we store that + in the local file path. + * The extension, however, is stripped off of the file_path parameter passed + back out of ast_media_cache_retrieve. + + Second, this patch causes stale items to be completely removed from the system. + Prior to this patch, sound files could be orphaned due to the bucket + referencing the file being deleted, but the file itself not being removed. This + is now addressed by explicitly calling ast_bucket_file_delete on the + bucket_file when it is deemed to be stale. Note that this only happen when we + know we will attempt to retrieve the resource again. + + Finally, this patch changes the AO2 container holding media items to just use + a regular mutex. The usage for this container already assumed it was a plain + mutex, and - given that retrieval of an item can cause it to be replaced in + the container - a mutex makes more sense than a read/write lock. + + Change-Id: I51667fff86ae8d2e4a663555dfa85b11e935fe0f + +2014-10-25 20:21 +0000 [6bbcfb34bd] Matt Jordan + + * funcs/func_curl: Add the ability for CURL to download and store files + + This patch adds a write option to the CURL dialplan function, allowing it to + CURL files and store them locally. The value 'written' to the CURL URL + specifies the location on disk to store the file. As an example: + + same => n,Set(CURL(http://1.1.1.1/foo.wav)=/tmp/foo.wav) + + Would retrieve the file foo.wav from the remote server and store it in the + /tmp directory. + + Due to the potentially dangerous nature of this function call, APIs are + forbidden from using the write functionality unless live_dangerously is set + to True in asterisk.conf. + + ASTERISK-25652 #close + + Change-Id: I44f4ad823d7d20f04ceaad3698c5c7f653c41b0d + +2016-03-23 08:59 +0000 [392341ba37] gtjoseph + + * pjproject-bundled: Cleanups for reported issues + + PortAudio should no longer be required + PJSIP_MAX_PKT_LEN is now 6000 + Older autoconf issue fixed. (CentOS 6) + + Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd + +2015-11-20 08:02 +0000 [ac66999971] Francesco Castellano + + * chan_sip.c: Space after port causes unnecessary resolution attempt + + check_via() already skips leading blanks where the sent-by address (with the + optional port) should be placed. + + Since RFC 3261 allows for blanks between the port ant the Via parameters: + > https://tools.ietf.org/html/rfc3261#section-20.42 + (actually it allows a lot of blanks more ;-)). I just switched from + ast_skip_blanks() to ast_strip() on the local copy of the string. + + ASTERISK-21301 #close + + Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06 +2016-03-19 17:49 +0000 [1d3191b118] gtjoseph + + * progdocs: Exclude ./third-party from documentation generation + + We don't need pjproject's documentation embedded in Asterisk's. + + Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17 + +2016-03-18 20:32 +0000 [8f94f947f5] Gianluca Merlo + + * func_aes: fix misuse of strlen on binary data + + The encryption code for AES_ENCRYPT evaluates the length of the data to + be encoded in base64 using strlen. The data is binary, thus the length + of it can be underestimated at the first NULL character. + Reuse the write pointer offset to evaluate it, instead. + + ASTERISK-25857 #close + + Change-Id: If686b5d570473eb926693c73461177b35b13b186 +2016-03-18 14:31 +0000 [a3c9a74a02] Kevin Harwell + + * chan_pjsip: ref leak when checking direct_media_glare + + Fix the reference leak introduced in the following commit: + + c534bd58075e2e1a1e4f3b23c435186c71b155fd + + ASTERISK-25849 + + Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85 +2016-03-16 12:37 +0000 [c534bd5807] Kevin Harwell + + * chan_pjsip: transfers with direct media reinvite has wrong address/port + + During a transfer involving direct media a race occurs between when the + transferer channel is swapped out, initiating rtp changes/updates, and the + subsequent reinvites. + + When Alice, after speaking with Charlie (Bob is on hold), connects Bob and + Charlie invites are sent to each in order to establish the call between them. + Bob is taken off hold and Charlie is told to have his media flow through + Asterisk. However, if before those invites go out the bridge updates Bob's + and/or Charlie's rtp information with direct media data (i.e. address, port) + then the invite(s) will contain the remote data in the SDP instead of the + Asterisk data. + + The race occurs in the native bridge glue code when updating the peer. The + direct_media_address can get set twice before sending out the first invite + during call connection. This can happen because the checking/setting of the + direct_media_address happened in one thread while the sending of the invite(s) + happened in another thread. + + This fix removes the race condition by moving the checking/setting of the + direct_media_address to be in the same thread as the sending of the invites(s). + This serializes the checking/setting and sending so they can no longer happen + out of order. + + ASTERISK-25849 #close + + Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873 + +2016-03-03 04:43 +0000 [bdccb81157] Sergio Medina Toledo + + * res_pjsip_refer.c: Fix seg fault in process of Refer-to header. + + The "Refer-to" header of an incoming REFER request is parsed by + pjsip_parse_uri(). That function requires the URI parameter to be NULL + terminated. Unfortunately, the previous code added the NULL terminator by + overwriting memory that may not be safe. The overwritten memory results + could be benign, memory corruption, or a segmentation fault. Now the URI + is NULL terminated safely by copying the URI to a new chunk of memory with + the correct size to be NULL terminated. + + ASTERISK-25814 #close + + Change-Id: I32565496684a5a49c3278fce06474b8c94b37342 + +2016-02-25 10:29 +0000 [0da36fca6b] Leif Madsen + + * Add initial support to build Docker images + + This work-in-progress is the first step to being able to reliably + build Asterisk containers from the Asterisk source. I'm submitting + this based on feedback gained at AstriDevCon 2015. + + Information about how to use this is provided in contrib/docker/README.md + and will result in a local Asterisk container being built right from + your source. I believe this can eventually be automated via + hub.docker.com. + + Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1 + +2016-03-11 12:22 +0000 [810f92c9dc] Richard Mudgett + + * chan_sip.c: Fix mwi resub deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 #close + + Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6 + +2016-03-10 17:01 +0000 [72c444ba37] Richard Mudgett + + * chan_sip.c: Fix registration timeout and expire deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508 + +2016-03-09 16:26 +0000 [7ea1e181dc] Richard Mudgett + + * chan_sip.c: Fix waitid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + * Made always run check_pendings() under the scheduler thread so scheduler + ids can be checked safely. + + ASTERISK-25023 + + Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52 + +2016-03-10 12:17 +0000 [fbf8e04aed] Richard Mudgett + + * chan_sip.c: Fix t38id deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f + +2016-03-08 15:08 +0000 [02458cc6fd] Richard Mudgett + + * chan_sip.c: Fix session timers deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900 + +2016-03-09 16:34 +0000 [c7fdff2e37] Richard Mudgett + + * chan_sip.c: Fix reinviteid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1 + +2016-03-07 13:21 +0000 [69810b306d] Richard Mudgett + + * chan_sip.c: Fix autokillid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + * Fix clearing autokillid in __sip_autodestruct() even though we could + reschedule. + + ASTERISK-25023 + + Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f + +2016-03-09 16:32 +0000 [f484ddbdfe] Richard Mudgett + + * chan_sip.c: Fix packet retransid deadlock potential. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + * Fix retrans_pkt() to call check_pendings() with both the owner channel + and the private objects locked as required. + + * Refactor dialog retransmission packet list to safely remove packet + nodes. The list nodes are now ao2 objects. The list has a ref and the + scheduled entry has a ref. + + ASTERISK-25023 + + Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641 + +2016-03-07 18:28 +0000 [67c79c326d] Richard Mudgett + + * chan_sip.c: Fix provisional_keepalive_sched_id deadlock. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Stopping a scheduled event can result in a deadlock if the scheduled event + is running when you try to stop the event. If you hold a lock needed by + the scheduled event while trying to stop the scheduled event then a + deadlock can happen. The general strategy for resolving the deadlock + potential is to push the actual starting and stopping of the scheduled + events off onto the scheduler/do_monitor() thread by scheduling an + immediate one shot scheduled event. Some restructuring may be needed + because the code may assume that the start/stop of the scheduled events is + immediate. + + ASTERISK-25023 + + Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48 + +2016-03-09 11:22 +0000 [76be7093cd] Richard Mudgett + + * chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + * Make dialog_unlink_all() unschedule all items at once in the sched + thread. + + ASTERISK-25023 + + Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4 + +2016-03-10 21:54 +0000 [52f0932e4c] Richard Mudgett + + * chan_sip.c: Clear scheduled immediate events on unload. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + The reordering of chan_sip's shutdown is to handle any immediate events + that get put onto the scheduler so resources aren't leaked. The typical + immediate events at this time are going to be concerned with stopping + other scheduled events. + + ASTERISK-25023 + + Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20 + +2016-03-15 14:51 +0000 [0987a11cce] Richard Mudgett + + * sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + Delaying destruction of the chan_sip sip_pvt structures caused the + /channels/chan_sip/test_sip_rtpqos unit test to crash. That test + registers a special test ast_rtp_engine with the rtp engine module. When + the unit test completes it cleans up by unregistering the test + ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt + happens after the unit test returns, the destructor tries to call the rtp + engine destroy callback of the test ast_rtp_engine auto variable which no + longer exists on the stack. + + * Change the test ast_rtp_engine auto variable to a static variable. Now + the variable can still exist after the unit test exits so the delayed + sip_pvt destruction can complete successfully. + + ASTERISK-25023 + + Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13 + +2016-03-07 15:50 +0000 [9a7cfa2b61] Richard Mudgett + + * sched.c: Ensure oldest expiring entry runs first. + + This patch is part of a series to resolve deadlocks in chan_sip.c. + + * Updated sched unit test to check new behavior. + + ASTERISK-25023 + + Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3 + +2016-03-15 13:31 +0000 [7964e260d3] Andrew Nagy + + * app_stasis: Don't hang up if app is not registered + + This prevents pbx_core from hanging up the channel if the app isn't + registered. + + ASTERISK-25846 #close + + Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce +2016-03-07 18:56 +0000 [cb97198ca6] Richard Mudgett + + * chan_sip.c: Simplify sip_pvt destructor call levels. + + Remove destructor calling destroy_it calling really_destroy_it + for no benefit. Just make the destructor the really_destroy_it + function. + + Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a + +2016-03-04 18:25 +0000 [8be01398d9] Richard Mudgett + + * chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full(). + + Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12 + +2016-03-14 08:59 +0000 [4df7b3ae80] Joshua Colp + + * build: Add configure check for proto field of PJSIP TLS transport setting. + + Older versions of PJSIP do not have the proto field on the TLS transport + setting structure. This change adds a configure check so even if it is + not present we will still be able to build. + + Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9 + +2016-03-12 16:02 +0000 [0af6b5de62] gtjoseph + + * build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE + + I can't ever recall actually needing the intermediate files or the checking + that a double compile produces. What I CAN remember is every DONT_OPTIMIZE + build needing 3 invocations of gcc instead of 1 just to do the checks and + produce those intermediate files. + + Having said that, Richard pointed out that the reason for the double compile + was that there were cases in the past where a submitted patch failed to compile + because the submitter never tried it with the optimizations turned on. + + To get the best of both worlds, COMPILE_DOUBLE has been split into its own + option. If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected + BUT you can then turn it off if all you need are the debugging symbols. This + way you have to make an informed decision about disabling COMPILE_DOUBLE. + + To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature + was added to menuselect. The element can now contain an "autoselect" + attribute which will turn the used member on but not create a hard dependency. + The cflags.xml implementation for COMPILE_DOUBLE looks like this... + + + COMPILE_DOUBLE + core + + + + * app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. + + Channel masquerading had a conflict with autochannel locking. + + When locking autochannel->channel, the channel is fetched from the + autochannel and then locked. During the fetch, the autochannel -- which + has no locks itself -- can be modified by someone who owns the channel + lock. That means that the value of autochan->channel cannot be trusted + until you hold the lock. + + In practice, this caused problems with Local channels getting + masqueraded away while the ChanSpy attempted to get info from that + channel. The old channel which was about to get removed got locked, but + the new (replaced) channel got unlocked (no-op). Because the replaced + channel was now locked (and would never get unlocked), it couldn't get + removed from the channel list in a timely manner, and would now cause + deadlocks when iterating over the channel list. + + This change checks the autochannel after locking the channel for changes + to the autochannel. If the channel had been changed, the lock is + reobtained on the new channel. + + In theory it seems possible that after this fix, the lock attempt on the + old (wrong) channel can be on an already destroyed lock, maybe causing + a crash. But that hasn't been observed in the wild and is harder induce + than the current deadlock. + + Thanks go to Filip Frank for suggesting a fix similar to this and + especially to IRC user hexanol for pointing out why this deadlock was + possible and testing this fix. And to Richard for catching my rookie + while loop mistake ;) + + ASTERISK-25321 #close + + Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def + +2016-03-07 21:34 +0000 [fb28049de2] gtjoseph + + * pjproject_bundled: Remove --with-external-pa from configure options. + + Not sure why it was there in the first place as we already specify + --disable-sound. + + Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9 + +2016-03-06 14:38 +0000 [d2eb65f71e] gtjoseph + + * res_pjsip: Strip spaces from items parsed from comma-separated lists + + Configurations like "aors = a, b, c" were either ignoring everything after "a" + or trying to look up " b". Same for mailboxes, ciphers, contacts and a few + others. + + To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To + facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were + updated to handle null pointers. + + In some cases, an ast_strlen_zero() test was added to skip consecutive commas. + + There was also an attempt to ast_free an ast_strdupa'd string in + ast_sip_for_each_aor which was causing a SEGV. I removed it. + + Although this issue was reported for realtime, the issue was in the res_pjsip + modules so all config mechanisms were affected. + + ASTERISK-25829 #close + Reported-by: Mateusz Kowalski + + Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2 + +2016-03-07 02:02 +0000 [f690c105f3] Rodrigo Ramírez Norambuena + + * res_odbc_transaction: fix some format tab + + Change-Id: I265e4ac47c629c9a63dd86b59df82a7ab3c64384 + +2016-02-17 22:58 +0000 [0ec9fe5421] Rodrigo Ramírez Norambuena + + * main/cli.c: Refactor function to print seconds formatted + + Refactor and created function ast_cli_print_timestr_fromseconds to print + seconds formatted: year(s) week(s) day(s) hour(s) second(s) + + This function now is used in addons/cdr_mysql.c,cdr_pgsql.c, main/cli.c, + res_config_ldap.c, res_config_pgsql.c. + + Change-Id: Ibeb8634102cd11d3f8623398b279cb731bcde36c + +2016-03-04 20:37 +0000 [471ff375fd] gtjoseph + + * install_prereq: Add packages for bundled pjproject + + RedHat/CentOS needs python-devel + Debian/Ubuntu needs automake, libsrtp-dev and python-dev + + Ubuntu also needed libncurses5-dev for cmenuselect so while not + needed for pjproject, I adedd it anyway. + + Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089 + +2016-02-24 17:25 +0000 [2b9849625c] gtjoseph + + * res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited + + Per RFC3325, the 'From' header is now anonymized on outgoing calls when + caller id presentation is prohibited. + + TID = trust_id_outbound + PRO = Set(CALLERID(pres)=prohib) + USR = endpoint/from_user + DOM = endpoint/from_domain + PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) + + Conditions |Result + --------------------|---------------------------------------------------- + TID PRO USR DOM |PAI FROM + --------------------|---------------------------------------------------- + Y Y abc def.ghi |PRI "Anonymous" + Y Y abc |PRI "Anonymous" + Y Y def.ghi |PRI "Anonymous" + Y Y |PRI "Anonymous" + + Y N abc def.ghi |YES + Y N abc |YES > + Y N def.ghi |YES "Caller Name" @def.ghi> + Y N |YES "Caller Name" @> + + N Y abc def.ghi |NO "Anonymous" + N Y abc |NO "Anonymous" + N Y def.ghi |NO "Anonymous" + N Y |NO "Anonymous" + + N N abc def.ghi |YES + N N abc |YES > + N N def.ghi |YES "Caller Name" @def.ghi> + N N |YES "Caller Name" @> + + ASTERISK-25791 #close + Reported-by: Anthony Messina + + Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9 + +2016-03-03 17:34 +0000 [37472f7398] gtjoseph + + * third_party/Makefile.rules: Replace unsupported != operator with $(shell ...) + + Apparently the != operator is fairly new so I've replaced it with + the old $(shell ...) syntax. + + Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479 + Reported-by: Richard Mudgett + +2016-01-23 15:50 +0000 [195100e770] gtjoseph + + * loader: Retry dlopen when loading fails + + Although we use the RTLD_LAZY flag when calling dlopen + the first time on a module, this only defers resolution + for function calls. Pointer references to functions are + determined at link time so dlopen expects them to be there. + Since we don't cross-module link, pointers to functions + in other modules won't be available and dlopen will fail. + + Doing a "hardened" build also causes problems because it + typically sets "-z now" on the ld command line which + overrides RTLD_LAZY at run time. + + If the failing module isn't a GLOBAL_SYMBOLS module, then + dlopen will be called again after all the GLOBAL_SYMBOLS + modules have been loaded and they'll eventually resolve. + + If the calling module IS a GLOBAL_SYMBOLS module itself + and a third module depends on it, then there's an issue + because the second time through the dlopen loop, + GLOBAL_SYMBOLS modules aren't given any special treatment + and since the order in which dlopen is called isn't + deterministic, the dependent may again be tried before the + module it needs is loaded. + + Simple solution: Save modules that fail load_resource + because of a dlopen error in a list and retry them + immediately after the first pass. Keep retrying until + the failed list is empty or we reach a #defined max + retries. Error messages are suppressed until the final + pass which also gets rid of those confusing error messages + about module failures that are later corrected. + + Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb + +2016-03-01 16:18 +0000 [15c5743ac1] Kevin Harwell + + * bridge.c: Crash during attended transfer when missing a local channel half + + It's possible for the transferer channel to get hung up early during the + attended transfer process. For instance, a phone may send a "bye" immediately + upon receiving a sip notify that contains a sip frag 100 (I'm looking at you + Jitsi). When this occurs a race begins between the transferer being hung up + and completion of the transfer code. + + If the channel hangs up too early during a transfer involving stasis bridging + for instance, then when the created local channel goes to look up its swap + channel (and associated datastore) it can't find it (since it is no longer in + the bridge) thus it fails to enter the stasis application. Consequently, the + created local channel(s) hang up as well. If the timing is just right then the + bridging code attempts to add the message link with missing local channel(s). + Hence the crash. + + Unfortunately, there is no great way to solve the problem of the unexpected + "bye". While we can't guarantee we won't receive an early hangup, and in this + case still fail to enter the stasis application, we can make it so asterisk + does not crash. + + This patch does just that by locking the local channel structure, checking + that the local channel's peer has not been lost, and then continuing. This + keeps the local channel's peer from being ripped out from underneath it by + the local/unreal hangup code while attempting to set the stasis message link. + + ASTERISK-25771 + + Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880 + +2016-03-01 18:08 +0000 [0d2ccbca62] Kevin Harwell + + * res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 + + During the transfer process, some phones (okay it was the Jitsi softphone, + but maybe others are out there) send a "bye" immediately after receiving a + SIP Notify. When a "bye" is received early for some types of transfers the + transferer channel may no longer be available during late stage transfer + processing. + + For instance, during an attended transfer involving stasis bridging at one + point the created local channel looks for an associated swap channel in + order to retrieve the stasis application name. If the transferer has hung + up then the local channel will fail to find it. The local channel then has + no way to know which stasis app to enter, so it fails and hangs up as well. + Thus the transfer does not complete as expected. + + This patch delays the sending of the initial notify in order to give the + transfer process enough time to gather the necessary data for a successful + transfer. + + ASTERISK-25771 + + Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16 + +2016-03-03 08:26 +0000 [6af7fc4c37] Joshua Colp + + * res_pjsip_dtmf_info: NULL terminate the message body. + + PJSIP does not ensure that when printing the message body the + buffer will be NULL terminated. This is problematic when searching + for the signal and duration values of the DTMF. + + This change ensures the buffer is always NULL terminated. + + Change-Id: I52653a1a60c93092d06af31a27408d569cc98968 + +2016-03-01 20:03 +0000 [b8b7c2e428] gtjoseph + + * alembic: Fix downgrade and tweak for sqlite + + Downgrade had a few issues. First there was an errant 'update' statement in + add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we + weren't cleaning up the ENUMs so subsequent upgrades on postgres failed + because the types already existed. + + For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly. + Fortunately alembic batch_operations takes care of this for us if we + use it so the alter and drops were converted to use batch operations. + + Here's an example downgrade: + + with op.batch_alter_table('ps_endpoints') as batch_op: + batch_op.drop_column('tos_audio') + batch_op.drop_column('tos_video') + batch_op.add_column(sa.Column('tos_audio', yesno_values)) + batch_op.add_column(sa.Column('tos_video', yesno_values)) + batch_op.drop_column('cos_audio') + batch_op.drop_column('cos_video') + batch_op.add_column(sa.Column('cos_audio', yesno_values)) + batch_op.add_column(sa.Column('cos_video', yesno_values)) + + with op.batch_alter_table('ps_transports') as batch_op: + batch_op.drop_column('tos') + batch_op.add_column(sa.Column('tos', yesno_values)) + # Can't cast integers to YESNO_VALUES, so dropping and adding is required + batch_op.drop_column('cos') + batch_op.add_column(sa.Column('cos', yesno_values)) + + Upgrades from base to head and downgrades from head to base were tested + repeatedly for postgresql, mysql/mariadb, and sqlite3. + + Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8 + +2016-03-02 15:55 +0000 [7b71bca8a4] gtjoseph + + * config_transport: Fix objects returned by ast_sip_get_transport_states + + ast_sip_get_transport_states was returning a container of internal_state + objects instead of ast_sip_transport_state objects. This was causing + transport lookups to fail, most noticably in res_pjsip_nat, which + couldn't find the correct external addresses. This was causing contacts + to go out with internal ip addresses. + + ASTERISK-25830 #close + Reported-by: Sean Bright + + Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e + +2016-03-02 11:17 +0000 [0a3f0e85ac] Scott Griepentrog + + * CHAOS: cleanup possible null vars on msg alloc failure + + In message.c, if msg_alloc fails to init the string field, + vars may be null, so use a null tolerant cleanup. + + In res_pjsip_messaging.c, if msg_data_create fails, mdata + will be null, so use a null tolerant cleanup. + + ASTERISK-25323 + + Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56 + +2016-03-02 09:34 +0000 [60aa871be3] Scott Griepentrog + + * CHAOS: prevent crash on failed strdup + + This patch avoids crashing on a null pointer + if the strdup() allocation fails. + + ASTERISK-25323 + + Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5 + +2016-02-29 18:11 +0000 [0bdbf0d882] Richard Mudgett + + * func_callerid.c: Update REDIRECTING reason documentation. + + Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386 + +2016-02-26 18:57 +0000 [25de01f301] Richard Mudgett + + * SIP diversion: Fix REDIRECTING(reason) value inconsistencies. + + Previous chan_sip behavior: + + Before this patch chan_sip would always strip any quotes from an incoming + reason and pass that value up as the REDIRECTING(reason). For an outgoing + reason value, chan_sip would check the value against known values and + quote any it didn't recognize. Incoming 480 response message reason text + was just assigned to the REDIRECTING(reason). + + Previous chan_pjsip behavior: + + Before this patch chan_pjsip would always pass the incoming reason value + up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip + would send the reason value as passed down. + + With this patch: + + Both channel drivers match incoming reason values with values documented + by REDIRECTING(reason) and values documented by RFC5806 regardless of + whether they are quoted or not. RFC5806 values are mapped to the + equivalent REDIRECTING(reason) documented value and is set in + REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a + quoted string version ('"unconditional"') is converted to + REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal + with 'cfu' instead of any of the aliases. + + The incoming 480 response reason text supported by chan_sip checks for + known reason values and if not matched then puts quotes around the reason + string and assigns that to REDIRECTING(reason). + + Both channel drivers send outgoing known REDIRECTING(reason) values as the + unquoted RFC5806 equivalent. User custom values are either sent as is or + with added quotes if SIP doesn't allow a character within the value as + part of a RFC3261 Section 25.1 token. Note that there are still + limitations on what characters can be put in a custom user value. e.g., + embedding quotes in the middle of the reason string is silly and just + going to cause you grief. + + * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. + e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the + 'cfu' value. + + * Added missing malloc() NULL return check in res_pjsip_diversion.c + set_redirecting_reason(). + + * Fixed potential read from a stale pointer in res_pjsip_diversion.c + add_diversion_header(). The reason string needed to be copied into the + tdata memory pool to ensure that the string would always be available. + Otherwise, if the reason string returned by reason_code_to_str() was a + user's reason string then the string could be freed later by another + thread. + + Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87 + +2016-02-26 18:54 +0000 [8c8ef4efb0] Richard Mudgett + + * res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. + + Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd + +2016-02-29 20:41 +0000 [75ec137e91] Richard Mudgett + + * res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. + + * Fix double unref of other_party channel in off nominal path. + + * This is unlikely to be a real problem. However, for safety, + in handle_incoming_request() keep the datastore ref with the + other_party channel ref until we are finished with the other_party + channel. + + Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821 + +2016-01-18 21:54 +0000 [3173e91bab] gtjoseph + + * build-system: Allow building with static pjproject + + Background here: + http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html + + From CHANGES: + * To help insure that Asterisk is compiled and run with the same known + version of pjproject, a new option (--with-pjproject-bundled) has been + added to ./configure. When specified, the version of pjproject specified + in third-party/versions.mak will be downloaded and configured. When you + make Asterisk, the build process will also automatically build pjproject + and Asterisk will be statically linked to it. Once a particular version + of pjproject is configured and built, it won't be configured or built + again unless you run a 'make distclean'. + + To facilitate testing, when 'make install' is run, the pjsua and pjsystest + utilities and the pjproject python bindings will be installed in + ASTDATADIR/third-party/pjproject. + + The default behavior remains building with the shared pjproject + installation, if any. + + Building: + + All you have to do is include the --with-pjproject-bundled option on + the ./configure command line (and remove any existing --with-pjproject + option if specified). Everything else is automatic. + + Behind the scenes: + + The top-level Makefile was modified to include 'third-party' in the + list of MOD_SUBDIRS. + + The third-party directory was created to contain any third party + packages that may be needed in the future. Its Makefile automatically + iterates over any subdirectories passing on targets. + + The third-party/pjproject directory was created to house the pjproject + source distribution. Its Makefile contains targets to download, patch + configure, generate dependencies, compile libs, apps and python bindings, + sanitized build.mak and generate a symbols list. + + When bootstrap.sh is run, it automatically includes the configure.m4 + file in third-party/pjproject. This file has a macro to download and + conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR + and PJPROJECT_BUNDLED. It also tests for the capabilities like + PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to + trying to compile. Of course, bootstrap.sh is only run once and the + configure file is incldued in the patch. + + When configure is run with the new options, the macro in configure.m4 + triggers the download, patch, conifgure and tests. No compilation is + performed at this time. The downloaded tarball is cached in /tmp so + it doesn't get downloaded again on a distclean. + + When make is run in the top-level Asterisk source directory, it will + automatically descend all the subdirectories in third_party just as it + does for addons, apps, etc. The top-level Makefile makes sure that + the 'third-party' is built before 'main' so that dependencies from the + other directories are built first. + + When main does build, a new shared library (libasteriskpj) is created that + links statically to the pjproject .a files and exports all their symbols. + The asterisk binary links to that, just as it does with libasteriskssl. + + When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject + python bindings are installed in ASTDATADIR/third-party/pjproject. This + will facilitate testing, including running the testsuite which will be + updated to check that directory for the pjsua module ahead of the system + python library. + + Modules should continue to depend on pjproject if they use pjproject APIs + directly. They should not care about the implementation. No changes to any + res_pjsip modules were made. + + Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103 + +2016-02-22 16:59 +0000 [2dae4a1ccf] Richard Mudgett + + * chan_sip.c: Fix T.38 issues caused by leaving a bridge. + + chan_sip could not handle AST_T38_TERMINATED frames being sent to it when + the channel left the bridge. The action resulted in overlapping outgoing + reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not + happy. + + * Force T.38 to be remembered as locally bridged. Now when the channel + leaves the native RTP bridge after T.38, the channel remembers that it has + already reINVITEed the media back to Asterisk. It just needs to terminate + T.38 when the AST_T38_TERMINATED arrives. + + * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant + AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if + they happen before the T.38 state changes to disabled. Now the T.38 state + is set to disabled before the reINVITE is sent. + + ASTERISK-25582 #close + + Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce + +2016-02-18 18:27 +0000 [bf29a4e2e6] Richard Mudgett + + * res_pjsip_t38.c: Back out part of an earlier fix attempt. + + This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d + commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame + that it puts into the bridge may or may not be processed by the time the + bridged peer is kicked out of the bridge. If it is processed then all is + well. However, if it is not processed then that channel is stuck in fax + mode until it hangs up or maybe if it joins another bridge for T.38 + faxing. + + ASTERISK-25582 + + Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7 + +2016-02-22 13:54 +0000 [c7d45b84f9] Richard Mudgett + + * bridge core: Add owed T.38 terminate when channel leaves a bridge. + + The channel is now going to get T.38 terminated when it leaves the + bridging system and the bridged peers are going to get T.38 terminated as + well. + + ASTERISK-25582 + + Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7 + +2016-02-19 16:01 +0000 [0e296563d7] Richard Mudgett + + * channel api: Create is_t38_active accessor functions. + + ASTERISK-25582 + + Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b + +2016-02-19 19:06 +0000 [86f7336c91] Richard Mudgett + + * bridge_channel: Don't settle owed events on an optimization. + + Local channel optimization could cause DTMF digits to be duplicated. + Pending DTMF end events would be posted to a bridge when the local channel + optimizes out and is replaced by the channel further down the chain. When + the real digit ends, the channel would get another DTMF end posted to the + bridge. + + A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B + + 1) LocalA has the /n flag to prevent optimization. + 2) B is sending DTMF to A through the local channel chain. + 3) When LocalB optimizes out it can move B to the position of LocalB;1 + 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would + settle an owed DTMF end to the bridge toward LocalA;2. + 5) When B finally ends its DTMF it sends the DTMF end down the chain. + 6) Without this patch, A would hear the DTMF digit end when LocalB + optimizes out and when B ends the original digit. + + ASTERISK-25582 + + Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251 + +2016-02-22 12:15 +0000 [128c96456c] Richard Mudgett + + * channel.c: Route all control frames to a channel through the same code. + + Frame hooks can conceivably return a control frame in exchange for an + audio frame inside ast_write(). Those returned control frames were not + handled quite the same as if they were sent to ast_indicate(). Now it + doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a + channel or ast_indicate(). + + ASTERISK-25582 + + Change-Id: I5775f41421aca2b510128198e9b827bf9169629b + +2016-02-25 15:13 +0000 [4422905218] gtjoseph + + * sorcery: Refactor create, update and delete to better deal with caches + + The ast_sorcery_create, update and delete function have been refactored + to better deal with caches and errors. + + The action is now called on all non-caching wizards first. If ANY succeed, + the action is called on all caching wizards and the observers are notified. + This way we don't put something in the cache (or update or delete) before + knowing the action was performed in at least 1 backend and we only call the + observers once even if there were multiple writable backends. + + ast_sorcery_create was never adding to caches in the first place which + was preventing contacts from getting added to a memory_cache when they + were created. In turn this was causing memory_cache to emit errors if + the contact was deleted before being retrieved (which would have + populated the cache). + + ASTERISK-25811 #close + Reported-by: Ross Beer + + Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46 +2016-02-25 15:39 +0000 [acf329a3c7] gtjoseph + + * res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s. + + There are a few cases where we're emitting notices or warnings + for things that really need neither, like a client retrying to subscribe + to mwi when they're not conifgured for it. They get a 404 so there's no + need for non-debug messages. + + Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f +2016-02-25 14:17 +0000 [7e3e1ddf7e] gtjoseph + + * res_sorcery_memory_cache: Fix SEGV in some CLI commands + + A few of the CLI commands weren't checking for enough arguments + and were SEGVing. + + Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413 + +2016-02-22 19:31 +0000 [803a2fc2d5] Richard Mudgett + + * rtp_engine.h: Remove extraneous semicolons. + + Change-Id: Ib462633d396fa941379dfef648dcd2245e350084 + +2016-02-23 14:57 +0000 [886ee09471] Richard Mudgett + + * chan_sip.c: Suppress T.38 SDP c= line if addr is the same. + + Use the correct comparison function since we only care if the address + without the port is the same. + + Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0 + +2016-02-16 08:14 +0000 [b7970cabfa] Christof Lauber + + * res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables + + Introduced realloaction of ast_str buf in sqlite3_escape functions in case + the returned buffer from threadstorage was actually too small. + + Change-Id: I3c5eb43aaade93ee457943daddc651781954c445 + +2016-02-11 11:01 +0000 [ba8adb4ce3] gtjoseph + + * res_pjsip/config_transport: Allow reloading transports. + + The 'reload' mechanism actually involves closing the underlying + socket and calling the appropriate udp, tcp or tls start functions + again. Only outbound_registration, pubsub and session needed work + to reset the transport before sending requests to insure that the + pjsip transport didn't get pulled out from under them. + + In my testing, no calls were dropped when a transport was changed + for any of the 3 transport types even if ip addresses or ports were + changed. To be on the safe side however, a new transport option was + added (allow_reload) which defaults to 'no'. Unless it's explicitly + set to 'yes' for a transport, changes to that transport will be ignored + on a reload of res_pjsip. This should preserve the current behavior. + + Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf + +2016-02-19 04:30 +0000 [c00082329e] Walter Doekes + + * chan_sip: Optionally supply fromuser/fromdomain in SIP dial string. + + Previously you could add [!dnid] to the SIP dial string to alter the To: + header. This change allows you to alter the From header as well. + + SIP dial string extra options now look like this: + + [![touser[@todomain]][![fromuser][@fromdomain]]] + + INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To: + header, that is no longer possible. + + ASTERISK-25803 #close + + Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7 + +2016-02-07 17:34 +0000 [f8767a8804] gtjoseph + + * res_pjproject: Add ability to map pjproject log levels to Asterisk log levels + + Warnings and errors in the pjproject libraries are generally handled by + Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings + or errors so the messages emitted by pjproject directly are either superfluous + or misleading. A good exampe of this are the level-0 errors pjproject emits + when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't + consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS + client be treated any differently? + + A config file for res_pjproject has bene added (pjproject.conf) and a new + log_mappings object allows mapping pjproject levels to Asterisk levels + (or nothing). The defaults if no pjproject.conf file is found are the same + as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR, + 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG + + Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898 + +2016-02-18 10:55 +0000 [14886643c6] Alexei Gradinari + + * res_pjsip_outbound_publish: Fix processing 412 response + + When Asterisk receives a 412 (Conditional Request Failed) response + it has to recreate publish session. + There is bug in res_pjsip_outbound_publish.c + The function sip_outbound_publish_client_alloc is called with wrong object + while processing 412 (Conditional Request Failed) response. + This patch fixes it. + + ASTERISK-25229 #close + + Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359 + +2016-02-18 11:15 +0000 [8055d080cd] Mark Michelson + + * Fix failing threadpool_auto_increment test. + + The threadpool_auto_increment test fails infrequently for a couple of + reasons + * The threadpool listener was notified of fewer tasks being pushed than + were actually pushed + * The "was_empty" flag was set to an unexpected value. + + The problem is that the test pushes three tasks into the threadpool. + Test expects the threadpool to essentially gather those three tasks, and + then distribute those to the threadpool threads. It also expects that as + the tasks are pushed in, the threadpool listener is alerted immediately + that the tasks have been pushed. In reality, a task can be distributed + to the threadpool threads quicker than expected, meaning that the + threadpool has already emptied by the time each subsequent task is + pushed. In addition, the internal threadpool queue can be delayed so + that the threadpool listener is not alerted that a task has been pushed + even after the task has been executed. + + From the test's point of view, there's no way to be able to predict + exactly the order that task execution/listener notifications will occur, + and there is no way to know which listener notifications will indicate + that the threadpool was previously empty. + + For this reason, the test has been updated to only check the things it + can check. It ensures that all tasks get executed, that the threads go + idle after the tasks are executed, and that the listener is told the + proper number of tasks that were pushed. + + Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c + +2016-02-17 13:30 +0000 [30a49b8a6a] Richard Mudgett + + * cel.c: Fix mismatch in ast_cel_track_event() return type. + + The return type of ast_cel_track_event() is not large enough to return all + 64 potential bits of the event enable mask. Fortunately, the defined CEL + events do not really need all 64 bits and the return value is only used to + determine if the requested CEL event is enabled. + + * Made the ast_cel_track_event() return 0 or 1 only so the return value + can fit inside an int type instead of zero or a truncated 64 bit non-zero + value. + + Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c + +2016-02-16 23:37 +0000 [15aeb78c66] Rodrigo Ramírez Norambuena + + * app_queue: fix Calculate talktime when is first call answered + + Fix calculate of average time for talktime is wrong when is completed the + first call beacuse the time for talked would be that call. + + ASTERISK-25800 #close + + Change-Id: I94f79028935913cd9174b090b52bb300b91b9492 + +2016-02-16 16:37 +0000 [62282bb8ce] gtjoseph + + * res_odbc: Fix exports.in for missing symbols + + res_odbc.exports.in was missing a few symbols. + Changed to wildcards. + + Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c + +2016-02-16 12:20 +0000 [49203628f9] gtjoseph + + * res_statsd: Fix exports.in for missing symbols + + res_statsd.export.in was missing the _va variations of the log + functions causing Asterisk to crash in res_pjsip if OPTIONAL_API + wasn't enabled. + + ASTERISK-25727 #close + Reported-by: Gergely Dömsödi + + Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b + +2016-02-15 21:31 +0000 [4f08e9fb64] gtjoseph + + * res_pjsip_config_wizard: Add command to export primitive objects + + A new command (pjsip export config_wizard primitives) has been added that + will export all the pjsip objects it created to the console or a file + suitable for reuse in a pjsip.conf file. + + ASTERISK-24919 #close + Reported-by: Ray Crumrine + + Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b + +2016-02-15 15:37 +0000 [be811c4be1] gtjoseph + + * res_pjsip_caller_id: Fix segfault when replacing rpid or pai header + + If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid + or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify + the header added by the dialplan function. Since the header added by the + dialplan function is generic string, there are no virtual functions to parse + the uri and we get a segfault when we try. Since the modify, was really only + an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER + and recreate it. + + This raises a question for another time though: What should happen with + duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups + so if it's session supplement is loaded after res_pjsip_caller_id's (or any + other module that adds headers), there'll be dups in the message. + + ASTERISK-25337 #close + + Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa + +2016-02-15 13:08 +0000 [13b6c02945] Mark Michelson + + * Fix creation race of contact_status structures. + + It is possible when processing a SIP REGISTER request to have two + threads end up creating contact_status structures in sorcery. + contact_status is created using a "find or create" function. If two + threads call into this at the same time, each thread will fail to find + an existing contact_status, and so both will end up creating a new + contact status. + + During testing, we would see sporadic failures because the + PJSIP_CONTACT() dialplan function would operate on a different + contact_status than what had been updated by res_pjsip/pjsip_options. + + The fix here is two-fold: + 1) The "find or create" function for contact_status now has a lock + around the entire operation. This way, if two threads attempt the + operation simultaneously, the first to get there will create the object, + and the second will find the object created by the first thread. + + 2) res_sorcery_memory has had its create callback updated so that it + will not allow for objects with duplicate IDs to be created. + + Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97 + +2016-02-15 12:52 +0000 [5c400a0fed] Joshua Colp + + * res_pjsip_pubsub: Move where the subscription is stored to after initialized. + + A problem arose when testing the AMI subscription listing actions where it + was possible for a subscription that had not been fully initialized to be + listed. This was problematic as the underlying listing code would crash. + + This change makes it so the subscription tree is fully set up before it is + added to the list of subscriptions. This ensures that when the listing actions + get the subscription it is valid. + + ASTERISK-25738 #close + + Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48 + +2016-02-09 17:34 +0000 [b37555cc94] gtjoseph + + * res_pjsip: Refactor load_module/unload_module + + load_module was just too hairy with every step having to clean up all + previous steps on failure. + + Some of the pjproject init calls have now been moved to a separate + load_pjsip function and the unload_pjsip function was enhanced to clean + up everything if an error happened at any stage of the load process. + + In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns + and ast_threadpool_shutdowns were also corrected. + + Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302 + +2016-02-09 22:42 +0000 [c4d9f46878] Badalyan Vyacheslav + + * Resources/res_phoneprov: fix memory leak and heap-use-after-free + + * heap-use-after-free happens when we free "cfg" + but then use "value" which refers to it + + * A memory leak occurs because in some cases + it is not released "defaults" + + ASTERISK-25721 #close + Reported by: Badalyan Vyacheslav + Tested by: Badalyan Vyacheslav + + Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469 + +2016-02-11 11:21 +0000 [e5fd972d24] Etienne Lessard (license #6394) + + * func_iconv: Ensure output strings are properly terminated. + + ASTERISK-25272 #close + Reported by: Etienne Lessard + patches: + AST-25272.patch submitted by Etienne Lessard (license #6394) + + Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17 + +2016-02-10 16:16 +0000 [168c18737f] gtjoseph + + * res_pjsip: Handle pjsip_dlg_create_uas deprecation + + Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with + pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically + increments the lock on the returned dialog. To account for this, configure.ac + now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c + has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use + the original call or the new one. If the new one was used, the ref count is + decremented before returning. + + ASTERISK-25751 #close + Reported-by Josh Colp + + Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8 + +2016-02-09 20:13 +0000 [fd668670b5] Rodrigo Ramírez Norambuena + + * res_config_pgsql: Show error message in reload if not connected. + + Change-Id: I9290115a1aaadb589eb1d02eaeb502eec01b31fa + +2016-02-09 23:40 +0000 [a23d01e943] Badalyan Vyacheslav + + * Build: Added testing compiler to support the system sanitizes + + In older versions of the compiler was not sanitizes. + Compilers other than GCC can not support the Usan and TSAN + or have other options for *FLAGS. + + ASTERISK-25767 #close + Reported by: Badalyan Vyacheslav + Tested by: Badalyan Vyacheslav + + Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916 + +2016-02-09 20:57 +0000 [c7186c7f0a] Badalyan Vyacheslav + + * Build: Fix menuselect USAN conflicts + + USAN can be used together with other sanitizers. + + Reported by: Badalyan Vyacheslav + Tested by: Badalyan Vyacheslav + + Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f + +2016-02-09 14:21 +0000 [68643f83cd] Corey Farrell + + * Simplify and fix conditional in FD_SET. + + FD_SET contains a conditional statement to protect against buffer + overruns. The statement was overly complicated and prevented use + of the last array element of ast_fdset. We now just verify the fd + is less than ast_FDMAX. + + Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40 + +2016-02-09 07:11 +0000 [e40fddbeb5] Joshua Colp + + * tests/test_sorcery_memory_cache_thrash: Improve termination process. + + When terminating the threads thrashing a sorcery memory cache each + would be told to stop and then we would wait on them. During at + least one thrashing test this was problematic due to the specific + usage pattern in use. It would take some time for termination of the + thread to occur. + + This would occur due to contention between the threads retrieving + and the threads updating the cache. As the retrieving threads are + given priority it may be some time before the updating threads + are able to proceed. + + This change makes it so all threads are told to stop and then each + are joined to ensure they stop. This way all the threads should + stop at around the same time instead of waiting for one to stop, + the next to stop, then the next, and so on. As a result of this + the execution time for each thrash test is much closer to their + expected value than previously seen as well. + + Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827 +2016-01-29 17:56 +0000 [bbf3ace682] gtjoseph + + * res_pjsip: Fix infinite recursion when loading transports from realtime + + Attempting to load a transport from realtime was forcing asterisk into an + infinite recursion loop. The first thing transport_apply did was to do a + sorcery retrieve by id for an existing transport of the same name. For files, + this just returns the previous object from res_sorcery_config's internal + container, if any. For realtime, the res_sourcery_realtime driver looks in the + database and finds the existing row but now it has to rehydrate it into a + sorcery object which means calling... transport_apply. And so it goes. + + The main issue with loading from realtime (apart from the loop) was that + transport stores structures and pointers directly in the ast_sip_transport + structure instead of the separate ast_transport_state structure. This patch + separates those items into the ast_sip_transport_state structure. The pattern + is roughly the same as res_pjsip_outbound_registration. + + Although all current usages of ast_sip_transport and ast_sip_transport_state + were modified to use the new ast_sip_get_transport_state API, the original + items are left in ast_sip_transport and kept updated to maintain ABI + compatability for third-party modules. They are marked as deprecated and + noted that they're now in ast_sip_transport_state. + + ASTERISK-25606 #close + Reported-by: Martin Moučka + + Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19 + +2016-02-07 13:00 +0000 [72bf53eea5] Rodrigo Ramírez Norambuena + + * res_config_pgsql: Add message on cli failed command status + + In case failed of command "realtime show pgsql status" show a message the data + of connection to more clear information in error. + + Change-Id: Ia8e9e2400466606e7118f52a46e05df0719b6a29 + +2016-02-05 10:29 +0000 [b69729dde5] gtjoseph + + * chan_misdn: Fix a few issues causing compile errors + + Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98 + +2016-01-25 17:36 +0000 [1bc54aee80] Richard Mudgett + + * app_confbridge: Only use b_profile options from the conference. + + A user cannot set new bridge options after the conference is created by + the first user. Attempting to do so is documented as undefined behavior. + + This patch ensures that the bridge profile options used are from the + conference and not what a subsequent user may have tried to set. + + Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266 + +2016-02-04 16:17 +0000 [3b426a8b09] Mark Michelson + + * Check for OpenSSL defines before trying to use them. + + The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior + to OpenSSL version 1.0.1. A recent commit attempts to, by default, set + these options, which can cause problems on systems with older OpenSSL + installations. + + This commit adds a configure script check for those defines and will not + attempt to make use of those if they do not exist. We will print a + warning urging the user to upgrade their OpenSSL installation if those + defines are not present. + + Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d +2016-02-03 14:25 +0000 [9b13ab6a63] gtjoseph + + * pjsip/alembic: Add missing columns to system and registration + + ps_systems needed disable_tcp_switch + ps_registrations needed line and endpoint + + ASTERISK-25737 #close + + Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19 + +2016-02-04 11:39 +0000 [82e2938fa8] Mark Michelson + + * res_stasis_device_state: Fix refcounting error. + + Device state subscription lifetimes were governed by when the + subscription was established and unsubscribed from. However, it is + possible that at the time of unsubscription, there could be device state + events still in flight. When those device state events occur, the device + state callback could attempt to dereference a freed pointer. Crash. + + This change ensures that the lifetime of the device state subscription + does not end until the underlying stasis subscription has confirmed that + its final message has been sent. + + Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2 + +2016-01-27 10:44 +0000 [d83dba7099] Sean Bright + + * res_rtp_asterisk: Allow ICE host candidates to be overriden + + During ICE negotiation the IPs of the local interfaces are sent to the remote + peer as host candidates. In many cases Asterisk is behind a static one-to-one + NAT, so these host addresses will be internal IP addresses. + + To help in hiding the topology of the internal network, this patch adds the + ability to override the host candidates by matching them against a + user-defined list of replacements. + + Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f + +2016-02-03 12:05 +0000 [0de74fad55] Joshua Colp + + * AST-2016-001 http: Provide greater control of TLS and set modern defaults. + + This change exposes the configuration of various aspects of the TLS + support and sets the default to the modern standards. + + The TLS cipher is now set to the best values according to the + Mozilla OpSec team, different TLS versions can now be disabled, and + the cipher order can be forced to be that of the server instead of + the client. + + ASTERISK-24972 #close + + Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8 +2015-12-07 12:46 +0000 [e67b445e8d] Richard Mudgett + + * AST-2016-003 udptl.c: Fix uninitialized values. + + Sending UDPTL packets to Asterisk with the right amount of missing + sequence numbers and enough redundant 0-length IFP packets, can make + Asterisk crash. + + ASTERISK-25603 #close + Reported by: Walter Doekes + + ASTERISK-25742 #close + Reported by: Torrey Searle + + Change-Id: I97df8375041be986f3f266ac1946a538023a5255 +2015-09-28 17:07 +0000 [a877e0d94b] Richard Mudgett + + * AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow. + + Setting the sip.conf timert1 value to a value higher than 1245 can cause + an integer overflow and result in large retransmit timeout times. These + large timeout times hold system file descriptors hostage and can cause the + system to run out of file descriptors. + + NOTE: The default sip.conf timert1 value is 500 which does not expose the + vulnerability. + + * The overflow is now detected and the previous timeout time is + calculated. + + ASTERISK-25397 #close + Reported by: Alexander Traud + + Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290 +2016-02-03 14:07 +0000 [dcbedf9ab1] gtjoseph + + * logging: Remove/fix some message annoyances + + test_dlinklists doesn't need to NOTICE everyone that every macro worked. + + res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or + provider was registered. + + res_odbc was missing a newline at the end of one message. + + Change-Id: I6c06361518ef3711821795e535acd439782a995e + +2016-02-02 10:52 +0000 [6522361871] Alexei Gradinari License #5691 + + * res_sorcery_realtime: Fix regex regression. + + A regression was introduced where searching for realtime PJSIP objects + by regex by starting the regex with a leading "^" would cause no items + to be returned. + + This was due to a change which attempted to drop the requirement for a + leading "^" to be present due to how some CLI commands formulate their + regexes. However, the change, rather than simply eliminating the + requirement, caused any regexes that did begin with "^" to end up not + returning the expected results. + + This change fixes the problem by inspecting the regex and formulating + the realtime query differently depending on if it begins with "^". + + ASTERISK-25702 #close + Reported by Nic Colledge + + Patches: + realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691 + + Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693 + +2016-02-02 04:05 +0000 [2a6f18cd55] Karsten Wemheuer + + * res_xmpp: Does not connect in component mode + + The module res_xmpp does not accept usernames in the form used in component + mode (XEP-0114). In component mode there is no @something in the name. + In component mode the connection is now not dropped anymore. + + If the xmpp server sends out a "stream" tag before handshake is finished, + the connection gets dropped in res_xmpp. Now this tag will be ignored and + the connection will be established. + + After connecting there will be an exchange of presence states. This does + not work as expected in component mode. The responsible function + "xmpp_pak_presence" is left before the states get sent out. Sending + presence states in component mode is now moved to the top of the function. + + ASTERISK-25735 #close + + Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c +2016-02-01 13:04 +0000 [40da6434c1] gtjoseph + + * build_system: Fix some warnings highlighted by clang + + Fix some warnings found with clang. + + Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd + +2016-01-31 20:13 +0000 [52b29f9b4c] gtjoseph + + * pjsip/alembic: Fix definition of qualify_timeout + + A recent commit set qualify_timeout to Decimal which isn't supported. + This path corrects it to Float. + + Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf + +2016-01-29 07:39 +0000 [55a7367ad4] Stefan Engström + + * chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip. + + When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a) + AMI action: SIPnotify or b) cli command: sip notify , I expect + asterisk to include the same value for its own ip in both cases a) and b), + but it seems a) produces a contact header like Contact: + whereas b) produces a contact header like + . 0.0.0.0:8060 is my udpbindaddr in sip.conf + + My guess is that manager_sipnotify should call + ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does, + because after applying this patch, both cases a) and b) produce + the contact header that I expect: + + Reported by: Stefan Engström + Tested by: Stefan Engström + + Change-Id: I86af5e209db64aab82c25417de6c768fb645f476 +2016-01-28 12:44 +0000 [d2397f028f] Richard Mudgett + + * config_options.c: Fix warning message wording. + + Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4 + +2016-01-25 17:34 +0000 [af6b15976d] Richard Mudgett + + * app_confbridge.c: Replace inlined code with existing function. + + Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51 + +2016-01-25 16:05 +0000 [7932336a3d] Richard Mudgett + + * app_confbridge: Add ability to get the muted conference state. + + * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. + + * Added Muted header to AMI ConfbridgeListRooms action response list + events to indicate the muted conference state. + + * Added Muted column to CLI "confbridge list" output to indicate the muted + conference state and made the locked column a yes/no value instead of a + locked/unlocked value. + + ASTERISK-20987 + Reported by: hristo + + Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1 + +2016-01-26 17:59 +0000 [894045e7cf] Richard Mudgett + + * app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. + + Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7 + +2016-01-25 15:48 +0000 [12c93e8f81] Richard Mudgett + + * app_confbridge: Make non-admin users join a muted conference muted. + + ASTERISK-20987 #close + Reported by: hristo + + Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38 + +2016-01-27 13:08 +0000 [f19bf7a321] gtjoseph + + * res_pjsip: Add res_pjproject dependency to samples + + Since res_pjsip now depends on res_pjproject, this has been added to + basic-pbx modules.conf. + + Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d +2016-01-27 10:29 +0000 [c53903d447] gtjoseph + + * build_system: Prevent goals needing makeopts from running when it's missing + + The Makefile only optionally includes makeopts so when goals like uninstall that + dont depend on anything else are run after a distclean, rules like + 'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts + to remove everything in the root directory. + + Although there's a rule defined for makeopts which prints a message and does + an 'exit 1', since '-include makepopts' was specified (with the -), the exit + was ignored letting the rest of the rules run. + + This patch makes makeopts required unless the goal has the string 'clean' in it. + + ASTERISK-25730 #close + Reported-by: George Joseph + + Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7 + +2016-01-25 09:35 +0000 [1dfd104a27] Joshua Colp + + * config: Allow options to register when documentation is unavailable. + + The config options framework is strict in that configuration options must + be documented unless XML documentation support is not available. In + practice this is useful as it ensures documentation exists however in + off-nominal cases this can cause strange problems. + + If it is expected that a config option has a non-zero or non-empty + default value but the config option documentation is unavailable + this reasonable expectation will not be met. This can cause obscure + crashes and weirdness depending on how the code handles it. + + This change tweaks the behavior to ensure that the config option + is still allowed to register, apply default values, and be set when + devmode is not enabled. If devmode is enabled then the option can + NOT be set. + + This also does not remove the initial documentation error message that + is output on load when registering the configuration option. + + ASTERISK-25725 #close + + Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8 + +2016-01-25 10:23 +0000 [a706ad44e6] Mark Michelson + + * Stasis: Use custom structure when setting variables. + + A recent change to queue channel variable setting to the Stasis control + queue caused a regression. When setting channel variables, it is + possible to give a NULL channel variable value in order to unset the + variable (i.e. remove it from the channel variable list). The change + introduced a call to ast_variable_new(), which is not tolerant of NULL + channel variable values. + + This new change switches from using ast_variable to using a custom + channel variable struct that is lighter weight and NULL value-tolerant. + + Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d + +2016-01-25 16:56 +0000 [289daca9e8] Rusty Newton + + * sounds/Makefile: Incremented core and extra sounds versions to 1.5 + + Core and extra sounds 1.5 was recently released! The tarballs contain + change descriptions however I figure more people will see this one so + I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra + to Core for en, en_GB, fr and added for languages that didn't already + have Extra sound sets (it,ja,ru). + + In addition all of the English and Russian sounds have been completely + re-recorded. + + Sounds moved and added: + activated,added,all-circuits-busy-now,astcc-followed-by-pound + at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy + ,call-fwd-unconditional,calling,call-waiting,cancelled, + cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated + ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist + ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello + ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to + ,location,number,number-not-answering,num-was-successfully,one-moment-please + ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option + ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached + ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial + ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension + ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered + ,your + + There were also a few random fixes here and there to file names for a few + of the languages. + + ASTERISK-25068 #close + + Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3 +2016-01-25 16:51 +0000 [b073244c51] Mark Michelson + + * res_pjsip_pubsub: Prevent crash from AMI command on freed subscription. + + A test recently uncovered that running an ill-timed AMI command to show + inbound subscriptions could cause a crash since Asterisk will try to + operate on a freed subscription. + + The fix for this is to remove the subscription tree from the list of + subscriptions at the time that we are sending our final NOTIFY request + out. This way, as the subscription is in the process of dying, it is + inaccessible from AMI. + + Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23 + +2016-01-25 11:03 +0000 [830f8933c2] Corey Farrell + + * chan_sip: Fix buffer overrun in sip_sipredirect. + + sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer + of 256 characters. This patch reduces the copy to 255 characters to leave + room for the string null terminator. + + ASTERISK-25722 #close + + Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab + +2016-01-23 16:45 +0000 [f299dc0d76] Rodrigo Ramírez Norambuena + + * app_queue: Add Lastpause field of queue member + + Add time when started a the last pause for a queue member for + QueueMemberStatus ami event. + + Also show accumulate time in seconds when started a pause for a queue + member to CLI command 'queue show'. + + ASTERISK-16394 #close + + Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c + +2016-01-23 12:34 +0000 [8c664da0ff] Rodrigo Ramírez Norambuena + + * app_queue: fix some tab format + + Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c + +2016-01-23 11:41 +0000 [2fb45c7801] Rodrigo Ramírez Norambuena + + * cdr_pgsql.cl: REFACTOR Macro LENGTHEN_BUF + + Remove repeated code on macro of assigned buffer to SQL vars. + + Add table and connection name to log error message when is not possible + allocate memory. + + Change-Id: I1fbf37d286a032d38fdda72a9f736356956c9ffe + +2016-01-22 15:08 +0000 [959f7436cc] Mark Michelson + + * Stasis: Fix potential memory leak of control data. + + When queuing tasks onto the Stasis control queue, you can pass an + arbitrary data pointer and a function to free that data. All ARI + commands that use the Stasis control queue made the assumption that the + destructor function would be called in all paths, whether the task was + queued successfully or not. However, this was not correct. If a task was + queued onto a control structure that was already completed, the + allocated data would not be freed properly. + + This patch corrects this by making sure that all return paths call the + data destructor. + + Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb + +2016-01-21 10:58 +0000 [a45eacebf3] Mark Michelson + + * Stasis: Use control queue to prevent crash. + + A crash occurred when attempting to set a channel variable on a channel + that had already been hung up. This is because there is a small window + between when a control is grabbed and when the channel variable is set + that the channel can be hung up. + + The fix here is to queue the setting of the channel variable onto the + control queue. This way, the manipulation of the channel happens in a + thread where it is safe to be done. + + In this change, I also noticed that the setting of bridge roles on + channels was being done outside of the control queue, so I also changed + those operations to be done in the control queue. + + ASTERISK-25709 #close + Reported by Mark Michelson + + Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78 + +2016-01-22 11:48 +0000 [7866806fc3] Richard Mudgett + + * logger.c: Fix buffer overrun found by address sanitizer. + + The null terminator of the tail struct member was not being allocated + when no logger.conf config file is installed. + + ASTERISK-25714 #close + Reported by: Badalian Vyacheslav + + Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30 + +2015-12-23 15:07 +0000 [9714da7aa4] Mark Michelson + + * res_odbc: Remove connection management + + Asterisk by default will create a single database connection and share + it among all threads that attempt to access the database. In previous + versions of Asterisk, this was tolerable, because the most used channel + driver, chan_sip, mostly accessed the database from a single thread. + With PJSIP, however, many threads may be attempting to perform database + operations, and there is the potential for many more database accesses, + meaning the concurrency is a horrible bottleneck if only one connection + is shared. + + Asterisk has a connection pooling facility built into it, but the + implementation has flaws. For one, there is a strict limit on the number + of simultaneous connections that could be made to the database. Anything + beyond the maximum would result in a failed operation. Attempting to + predict what the maximum should be is nearly impossible even for someone + intimately familiar with Asterisk's threading model. In addition, use of + transactions in the dialplan can cause some severe bugs if connection + pooling is enabled. + + This commit seeks to fix the concurrency problem by removing all + connection management code from Asterisk and leaving that to the + underlying unixODBC code instead. Now, Asterisk does not share a single + connection, nor does it try to maintain a connection pool. Instead, all + Asterisk ever does is request a connection from unixODBC and allow + unixODBC to either allocate those connections or retrieve them from a + pool. + + Doing this has a bit of a ripple effect. For one, since connections are + not long-lived objects, several of the safeguards that previously + existed have been removed. We don't have to worry about trying to use a + connection that has gone stale. In every case, when we request a + connection, it has just been made and we don't need to perform any + sanity checks to be sure it's still active. + + Another major player affected by this change is transactions. + Transactions and their respective connections were so tightly coupled + that it was almost pornographic. This code change moves + transaction-related code to its own file separate from the core ODBC + functionality. This way, the core of ODBC does not even have to know + that transactions exist. + + In making this large change, I had to look at a lot of code and + understand it. When making this change, I discovered several places + where the behavior is definitely not ideal, but it seemed outside the + scope of this change to be fixing it. Instead, any place where I saw + some sort of room for improvement has had a XXX comment added explaining + what could be altered to improve it. + + Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf + +2016-01-22 11:18 +0000 [d3969d09ae] Rodrigo Ramírez Norambuena + + * app_queue.c: remove include for core_unreal.h not used in code. + + Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d + +2016-01-21 16:40 +0000 [5dde111719] Corey Farrell + + * Build System: Add support for checking alembic branches. + + * Add 'check-alembic' target to root Makefile. + * Create build_tools/make_check_alembic to do the actual checks. + + ASTERISK-25685 + + Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004 + +2016-01-19 18:20 +0000 [04078f43b5] Richard Mudgett + + * res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case. + + ASTERISK-25712 #close + Reported by: Richard Mudgett + + Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f + +2016-01-13 16:49 +0000 [5615db3714] Richard Mudgett + + * res_pjsip: Add CLI "pjsip dump endpt [details]" + + Dump the res_pjsip endpt internals. + + In non-developer mode we will not document or make easily accessible the + "details" option even though it is still available. The user has to know + it exists to use it. Presumably they would also be aware of the potential + crash warning below. + + Warning: PJPROJECT documents that the function used by this CLI command + may cause a crash when asking for details because it tries to access all + active memory pools. + + Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb + +2016-01-18 03:49 +0000 [b259ac95ac] Diederik de Groot + + * main/asterisk.c: ast_el_read_char + + Make sure buf[res] is not accessed at res=-1 (buffer underrun). + Address Sanitizer will complain about this quite loudly. + + ASTERISK-24801 #close + + Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9 + +2016-01-18 19:27 +0000 [dd5c063934] gtjoseph + + * res_pjproject: Add module providing pjproject logging and utils + + res_pjsip_log_forwarder has been renamed to res_pjproject + and enhanced as follows: + + As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch, + a new ast_pjproject_get_buildopt function has been added. It + allows the caller to get the value of one of the buildopts. + + The initial use case is retrieving the runtime value of + PJ_MAX_HOSTNAME to insure we don't send a hostname greater + than pjproject can handle. Since it can differ between + the version of pjproject that Asterisk was compiled against + and the version of pjproject that Asterisk is running against, + we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk + source code. + + Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e + +2016-01-18 17:16 +0000 [3b9cba4294] Matt Jordan + + * funcs/func_cdr: Correctly report high precision values for duration and billsec + + When CDRs were refactored, func_cdr's ability to report high precision values + for duration and billsec (the 'f' option) was broken. This was due to func_cdr + incorrectly interpreting the duration/billsec values provided by the CDR engine + in milliseconds, as opposed to seconds. Since the CDR engine only provides + duration and billsec in seconds, and does not expose either attribute with + sufficient precision to merely pass back the underlying value, this patch fixes + the bug by re-calculating duration and billsec with microsecond precision based + on the start/answer/end times on the CDR. + + ASTERISK-25179 #close + + Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841 + +2016-01-20 07:52 +0000 [479cc99acd] Rodrigo Ramírez Norambuena + + * README: Update year in copyright + + Change-Id: I56240f537fb3205672cdb2a74f0591ae7bb73dbc + +2016-01-19 17:15 +0000 [9fa76ba215] Joshua Colp + + * test_threadpool: Wait for each task to complete and fix memory leak. + + This change makes the thread_timeout_thrash unit test wait for + each task to complete. This fixes the problem where the test would + prematurely end when all threads were gone and a new one had to be + started to handle the last task. It also increases the thrasing as + it is now more likely for each task to encounter the above scenario. + + This also fixes a memory leak where the data for each task was not + being freed. + + ASTERISK-25611 #close + + Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6 + +2016-01-18 19:44 +0000 [c9f7269b2e] Richard Mudgett + + * taskprocessor.c: Increase CLI "core ping taskprocessor" timeout. + + Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029 + +2016-01-18 19:43 +0000 [6e2a867716] Richard Mudgett + + * taskprocessor.c: Fix some taskprocessor unrefs. + + You have to call ast_taskprocessor_unref() outside of the taskprocessor + implementation code. Taskprocessor use since v12 has become more + transient than just the singleton uses in earlier versions. + + Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb + +2016-01-19 14:16 +0000 [a4dcbdf50f] Richard Mudgett + + * Fix alembic branches on master. + + Change-Id: I64ed21fec50eb833641ca49d92184f6aaabd86e8 + +2016-01-05 17:12 +0000 [35a3e8cc7f] Corey Farrell + + * Refactor init_logger_chain locking. + + This removes logchannels locking from init_logger_chain, puts the + responsibility on the caller. Adds locking around the one call that was + missing it. + + ASTERISK-24833 + + Change-Id: I6cc42117338bf9575650a67bcb78ab1a33d7bad8 + +2016-01-18 22:10 +0000 [378fed4900] Rodrigo Ramírez Norambuena + + * app_queue: Fix preserved reason of pause when Asterisk is restared + + When the Asterisk is restared is not preseved reason paused of members. + This patch fixed this cases, retain data on astdb and set when Asterisk + is started. + + ASTERISK-25732 #close + + Report by: Rodrigo Ramírez Norambuena + + Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5 + +2016-01-18 19:01 +0000 [130aa1427e] gtjoseph + + * pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject + + Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b + +2016-01-16 13:18 +0000 [eaf2b5052e] Daniel Journo + + * Update version number in features.conf.sample + + Update the version number in the comments from Asterisk 12 to Asterisk 12+ + + Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b + +2016-01-13 15:58 +0000 [c60d6c0162] Daniel Journo + + * pjsip/alembic: Fix qualify_timeout column definition + + Corrects the qualify_timeout column type from Integer to Decimal + + ASTERISK-25686 #close + Reported-by: Marcelo Terres + + Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8 + +2016-01-15 19:52 +0000 [480ccfcc97] Corey Farrell + + * main/config: Clean config maps on shutdown. + + ASTERISK-25700 #close + + Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808 + +2016-01-14 14:42 +0000 [a5b38b604c] Kevin Harwell + + * bridge_basic: don't cache xferfailsound during an attended transfer + + The xferfailsound was read from the channel at the beginning of the transfer, + and that value is "cached" for the duration of the transfer. Therefore, changing + the xferfailsound on the channel using the FEATURE() dialplan function does + nothing once the transfer is under way. + + This makes it so the transfer code instead gets the xferfailsound configuration + options from the channel when it is actually going to be used. + + This patch also fixes a potential memory leak of the props object as well as + making sure the condition variable gets initialized before being destroyed. + + ASTERISK-25696 #close + + Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4 + +2015-07-10 10:37 +0000 [d36c4d0b01] Richard Mudgett + + * taskprocessor.c: Simplify ast_taskprocessor_get() return code. + + Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1 + +2016-01-13 18:20 +0000 [0a878020dc] Richard Mudgett + + * astmm.c: Add more stats to CLI "memory show" commands. + + * Add freed regions totals to allocations and summary. + + * Add totals for all allocations and not just the selected allocations. + + Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a + +2016-01-14 16:00 +0000 [84b30c5e18] Kevin Harwell + + * bridge_basic: don't play an attended transfer fail sound after target hangs up + + If the attended transfer destination answers (picks call up or goes to + voicemail) and then hangs up on the transferer then transferer hears the + fail sound. + + This patch makes it so the fail sound is not played when the transfer + destination/target hangs up after answering. + + ASTERISK-25697 #close + + Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded + +2016-01-14 14:36 +0000 [c7caee6c4b] Corey Farrell + + * Remove *.gcna / *.gcno files from added module sources. + + Asterisk uses a Makefile macro to associate additional sources with a + module. This macro is responsible for creating clean targets but + previously left behind *.gcna and *.gcno files. + + ASTERISK-25683 #close + Reported by yaron nahum + + Change-Id: Idc0823fe80a25c42cefae901fde875e9fc38d8ea + +2016-01-14 09:26 +0000 [68cad96ffd] Rusty Newton + + * func_channel: Add help text for undocumented CHANNEL function arguments + + Adding help text documentation for: + * hangupsource + * appname + * appdata + * exten + * context + * channame + * uniqueid + * linkedid + + ASTERISK-24097 #close + Reported by: Steven T. Wheeler + Tested by: Rusty Newton + + Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d + +2016-01-10 16:22 +0000 [8182146e85] Daniel Journo + + * pjsip: Add option global/regcontext + + Added new global option (regcontext) to pjsip. When set, Asterisk will + dynamically create and destroy a NoOp priority 1 extension + for a given endpoint who registers or unregisters with us. + + ASTERISK-25670 #close + Reported-by: Daniel Journo + + Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62 + +2016-01-12 11:14 +0000 [022423b98b] Joshua Colp + + * app: Queue hangup if channel is hung up during sub or macro execution. + + This issue was exposed when executing a connected line subroutine. + When connected or redirected subroutines or macros are executed it is + expected that the underlying applications and logic invoked are fast + and do not consume frames. In practice this constraint is not enforced + and if not adhered to will cause channels to continue when they shouldn't. + This is because each caller of the connected or redirected logic does not + check whether the channel has been hung up on return. As a result the + the hung up channel continues. + + This change makes it so when the API to execute a subroutine or + macro is invoked the channel is checked to determine if it has hung up. + If it has then a hangup is queued again so the caller will see it + and stop. + + ASTERISK-25690 #close + + Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea + +2016-01-13 07:20 +0000 [79a7321a47] Sean Bright + + * res_musiconhold: Prevent multiple simultaneous reloads. + + There are two ways in which the reload() function in res_musiconhold can be + called from the CLI: + + * module reload res_musiconhold.so + * moh reload + + In the former case, the module loader holds a lock that prevents multiple + concurrent calls, but in the latter there is no such protection. + + This patch changes the 'moh reload' CLI command to invoke the module loader + directly, rather than call reload() explicitly. + + ASTERISK-25687 #close + + Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c +2016-01-12 14:25 +0000 [1fffe71f77] Richard Mudgett + + * res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts". + + PJPROJECT has a function available to dump the compile time + options used when building the library. + + * Add CLI "pjsip show buildopts" command. + + * Update contrib/scripts/autosupport to get pjproject information. + + Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748 + +2016-01-12 10:36 +0000 [01c5e2a07e] Mark Michelson + + * res_sorcery_realtime: Remove leading ^ requirement. + + res_sorcery_realtime's search-by-regex callback performed a check to + ensure that the passed-in regex began with a caret (^). If it did not, + then no results would be returned. + + This callback only started to become used when "like" support was added + to PJSIP CLI commands. The CLI command for listing objects would pass an + empty regex ("") to the sorcery backend if no "like" statement was + present. For most sorcery backends, this resulted in returning all + objects. However, for realtime, this resulted in returning no objects. + + This commit seeks to fix the regression by removing the requirement from + res_sorcery_realtime for the passed-in-regex to begin with a caret. + + ASTERISK-25689 #close + Reported by Marcelo Terres + + Change-Id: I22b4dc5d7f3f11bb29ac2e42ef94682e9bab3b20 + +2016-01-07 11:57 +0000 [a41aab477a] gtjoseph + + * pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address + + On a system with multiple ip addresses in the same subnet, if a + transport is bound to a specific ip address and endpoint/media_address + is set, the SIP/SDP will have the correct address in all fields but + the rtp stream MAY still originate from one of the other ip addresses, + most probably the "primary" ip address. This happens because + res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with + the "all" ip address (0.0.0.0 or ::). + + The new option causes res_pjsip_sdp_rtp/create_rtp to call + ast_rtp_instance_new with the endpoint's media_address (if specified) + instead of the "all" address. This causes the packets to originate from + the specified address. + + ASTERISK-25632 + ASTERISK-25637 + Reported-by: Olivier Krief + Reported-by: Dan Journo + + Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88 + +2016-01-08 16:59 +0000 [7760029f19] Kevin Harwell + + * pbx: Deadlock between contexts container and context_merge locks + + Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5) + introduced the possibility of a deadlock. Due to the mentioned modifications + ast_change_hints now needs to keep both merge/delete and state callbacks from + occurring while it executes. Unfortunately, sometimes ast_change_hints can be + called with the contexts container locked. When this happens it's possible for + another thread to grab the context_merge_lock before the thread calling into + ast_change_hints does and then try to obtain the contexts container lock. This + of course causes a deadlock between the two threads. The thread calling into + ast_change_hints waits for the other thread to release context_merge_lock and + the other thread is waiting on that one to release the contexts container lock. + + Unfortunately, there is not a great way to fix this problem. When hints change, + the subsequent state callbacks cannot run at the same time as a merge/delete, + nor when the usual state callbacks do. This patch alleviates the problem by + having those particular callbacks (the ones run after a hint change) occur in a + serialized task. By moving the context_merge_lock to a task it can now safely be + attempted or held without a deadlock occurring. + + ASTERISK-25640 #close + Reported by: Krzysztof Trempala + + Change-Id: If2210ea241afd1585dc2594c16faff84579bf302 + +2016-01-10 17:08 +0000 [e9c2c1dc67] Corey Farrell + + * devicestate: Cleanup engine thread during graceful shutdown. + + ASTERISK-25681 #close + + Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1 + +2016-01-10 13:51 +0000 [90c0dcaee4] Corey Farrell + + * manager: Cleanup manager_channelvars during shutdown. + + ASTERISK-25680 #close + + Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446 + +2016-01-10 13:27 +0000 [a868a381f0] Corey Farrell + + * res_calendar: Cleanup scheduler context at unload. + + ASTERISK-25679 #close + + Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f + +2016-01-08 11:49 +0000 [a1c43022d2] Joshua Colp + + * res_rtp_asterisk: Revert DTLS negotiation changes. + + Due to locking issues within pjnath these changes are being + reverted until pjnath can be changed. + + ASTERISK-25645 + + Revert "res_rtp_asterisk.c: Fix DTLS negotiation delays." + + This reverts commit 24ae124e4f7310cfa64c187b944b2ffc060da28d. + + Change-Id: I2986cfb2c43dc14455c1bcaf92c3804f9da49705 + + Revert "res_rtp_asterisk: Resolve further timing issues with DTLS negotiation" + + This reverts commit 965a0eee46d24321f74c244e23c5a5f45e67e12b. + + Change-Id: Ie68fafde27dad4b03cb7a1e27ce2a8502c3f7bbe + +2016-01-09 17:57 +0000 [220ba979cf] gtjoseph + + * Revert "pjsip_location: Delete contact_status object when contact is deleted" + + This reverts commit 0a9941de9d24093b5ff44096d1d7406f29d11e45. + + Matt, + + This patch causes another problem and should not have been needed. + Before this patch, persistent_endpoint_contact_deleted_observer WAS + deleting the contact_status when ast_sip_location_delete_contact was + called. By deleting it yourself in ast_sip_location_delete_contact + it was gone before the observer could run and the observer therefore + was throwing an error and not sending stasis/AMI/statsd messages. + + So, I don't think this was the cause of your original issue. I also + had verified the contact AMI and statsd lifecycle and it was working. + I'll double check now though. + + ASTERISK-25675 + Reported-by: Daniel Journo + + Change-Id: Ib586a6b7f90acb641b0c410f659743ab90e84f1a + +2016-01-09 18:04 +0000 [26e0e113dc] Corey Farrell + + * pbx_dundi: Run cleanup on failed load. + + During failed startup of pbx_dundi no cleanup was performed. Add a call + to unload_module before returning AST_MODULE_LOAD_DECLINE. + + ASTERISK-25677 #close + + Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29 + +2016-01-09 13:28 +0000 [dc2c000fd5] Corey Farrell + + * res_crypto: Perform cleanup at shutdown. + + This change causes res_crypto to unregister CLI at shutdown while still + preventing the module from being unloaded. + + ASTERISK-25673 #close + + Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc + +2016-01-06 19:10 +0000 [0bca2a5c26] Richard Mudgett + + * res_pjsip: Create human friendly serializer names. + + PJSIP name formats: + pjsip/aor/- -- registrar thread pool serializer + pjsip/default- -- default thread pool serializer + pjsip/messaging -- messaging thread pool serializer + pjsip/outreg/- -- outbound registration thread pool + serializer + pjsip/pubsub/- -- pubsub thread pool serializer + pjsip/refer/- -- REFER thread pool serializer + pjsip/session/- -- session thread pool serializer + pjsip/websocket- -- websocket thread pool serializer + + Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084 + +2016-01-06 19:09 +0000 [f0f5fbbc01] Richard Mudgett + + * Sorcery: Create human friendly serializer names. + + Sorcery name formats: + sorcery/- -- Sorcery thread pool serializer + + Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47 + +2016-01-06 19:09 +0000 [b1c7ae9afc] Richard Mudgett + + * Stasis: Create human friendly taskprocessor/serializer names. + + Stasis name formats: + subm:- -- Stasis subscription mailbox task processor + subp:- -- Stasis subscription thread pool serializer + + Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd + +2016-01-07 16:15 +0000 [3e857bb347] Richard Mudgett + + * taskprocessor.c: New API for human friendly taskprocessor names. + + * Add new API call to get a sequence number for use in human friendly + taskprocessor names. + + * Add new API call to create a taskprocessor name in a given buffer and + append a sequence number. + + Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9 + +2016-01-06 17:19 +0000 [84c245d38c] Richard Mudgett + + * taskprocessor.c: Fix CLI "core show taskprocessors" output format. + + Update the CLI "core show taskprocessors" output format to not be + distorted because UUID names are longer than previously used taskprocessor + names. + + Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601 + +2016-01-07 21:07 +0000 [7d86979ea0] Richard Mudgett + + * taskprocessor.c: Fix CLI "core show taskprocessors" unref. + + Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5 + +2016-01-06 19:00 +0000 [1fb39aa8a0] Richard Mudgett + + * ccss.c: Replace space in taskprocessor name. + + The CLI "core ping taskprocessor" command does not work very + well with taskprocessor names that have spaces in them. You + have to put quotes around the name so using tab completion + becomes awkward. + + Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0 + +2016-01-07 20:44 +0000 [71bb7b9c40] Richard Mudgett + + * taskprocessor.c: Sort CLI "core show taskprocessors" output. + + Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e + +2016-01-05 16:54 +0000 [b025e1982f] Richard Mudgett + + * taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock. + + Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b + +2015-12-16 11:25 +0000 [c5e16fe33a] Mark Michelson + + * Alembic: Add PJSIP global keep_alive_interval. + + The keep_alive_interval option was added about a year ago, but no + alembic revision was created to add the appropriate column to the + database. + + This commit fixes the problem and adds the column. This was discovered + by running the testsuite with automatic conversion to realtime enabled. + + Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a + +2016-01-07 03:21 +0000 [6745cd6529] Diederik de Groot + + * include/asterisk/time.h: Renamed global declaration:tv + + Renamed global declaration:tv to dummy_tv_var_for_types, + which would oltherwise cause 'shadow' warnings when 'tv' + was declared as a local variable elsewhere. + + Added comment to note that dummy_tv_var_for_types is never + really exported and only used as a place holder. + + ASTERISK-25627 #close + + Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28 + +2016-01-07 15:37 +0000 [1afc8432dc] Mark Michelson + + * PJSIP: Prevent deadlock due to dialog/transaction lock inversion. + + A deadlock was observed where the monitor thread was stuck, therefore + resulting in no incoming SIP traffic being processed. + + The problem occurred when two 200 OK responses arrived in response to a + terminating NOTIFY request sent from Asterisk. The first 200 OK was + dispatched to a threadpool worker, who locked the corresponding + transaction. The second 200 OK arrived, resulting in the monitor thread + locking the dialog. At this point, the two threads are at odds, because + the monitor thread attempts to lock the transaction, and the threadpool + thread loops attempting to try to lock the dialog. + + In this case, the fix is to not have the monitor thread attempt to hold + both the dialog and transaction locks at the same time. Instead, we + release the dialog lock before attempting to lock the transaction. + + There have also been some debug messages added to the process in an + attempt to make it more clear what is going on in the process. + + ASTERISK-25668 #close + Reported by Mark Michelson + + Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a + +2016-01-07 09:39 +0000 [5d8c42c6d3] Corey Farrell + + * ast_format_cap_append_by_type: Resolve codec reference leak. + + This resolves a reference leak caused by ASTERISK-25535. The pointer + returned by ast_format_get_codec is saved so it can be released. + + ASTERISK-25664 #close + + Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec + +2016-01-07 03:33 +0000 [7856762f2f] Diederik de Groot + + * main: Use ast_strdup instead of strdup + + Fix compile error in main/utils.c because strdup was used in dummy_start + + Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793 + +2016-01-06 07:12 +0000 [64b2046f3d] Walter Doekes + + * Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts. + + The spandspflow2pcap.py creates pcap files from fax.log files, generated + through 'fax set debug on' when receiving a fax. An example fax.log is + included as spandspflow2pcap.log. + + The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a + recent version of SIPp. + + ASTERISK-25660 #close + + Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526 + +2016-01-04 04:26 +0000 [084563e136] Aaron An + + * cel/cel_radius: Fix wrong pointer. + + The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter + y not the address of y. + + I capture the radius UDP packet via tcpdump, and the AV pairs are not correct, + then i review the source code and compare it with cdr/cdr_radius.c. Fix it and + it works. + + ASTERISK-25647 #close + Reported by: Aaron An + Tested by: Aaron An + + Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0 + +2016-01-04 20:23 +0000 [36f1eaf0b5] Corey Farrell + + * main/pbx: Move hangup handler routines to pbx_hangup_handler.c. + + This is the sixth patch in a series meant to reduce the bulk of pbx.c. + This moves hangup handler management functions to their own source. + + Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104 + +2015-12-21 11:07 +0000 [90b06d1a3c] Martin Tomec + + * app_queue: Add member flag "in_call" to prevent reading wrong lastcall time + + Member lastcall time is updated later than member status. There was chance to + check wrapuptime for available member with wrong (old) lastcall time. + New boolean flag "in_call" is set to true right before connecting call, and + reset to false after update of lastcall time. Members with "in_call" set to true + are treat as unavailable. + + ASTERISK-19820 #close + + Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500 + +2016-01-04 19:46 +0000 [3507494b8a] Corey Farrell + + * main/pbx: Move dialplan application management routines to pbx_app.c. + + This is the sixth patch in a series meant to reduce the bulk of pbx.c. + This moves dialplan application management functions to their own source. + + Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c + +2016-01-04 18:20 +0000 [54a8f1a396] Corey Farrell + + * main/pbx: Move switch routines to pbx_switch.c. + + This is the fifth patch in a series meant to reduce the bulk of pbx.c. + This moves ast_switch functions to their own source. + + Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e + +2016-01-04 18:00 +0000 [c3c8b8e41d] Corey Farrell + + * main/pbx: Move timing routines to pbx_timing.c. + + This is the fourth patch in a series meant to reduce the bulk of pbx.c. + This moves pbx timing functions to their own source. + + Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6 + +2015-12-30 10:49 +0000 [6d18fe151c] gtjoseph + + * voicemail: Move app_voicemail / res_mwi_external conflict to runtime + + The menuselect conflict between app_voicemail and res_mwi_external + makes it hard to package 1 version of Asterisk. There no actual + build dependencies between the 2 so moving this check to runtime + seems like a better solution. + + The ast_vm_register and ast_vm_greeter_register functions in app.c + were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there + is already a voicemail module registered. The modules' load_module + functions were then modified to return DECLINE instead of -1 to the + loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, + the modules were incorrectly causing Asterisk to stop so this needed + to be cleaned up anyway. + + Now you can build both and use modules.conf to decide which voicemail + implementation to load. + + The default menuselect options still build app_voicemail and not + res_mwi_external but if both ARE built, res_mwi_external will load + first and become the voicemail provider unless modules.conf rules + prevent it. This is noted in CHANGES. + + Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247 + +2016-01-04 16:15 +0000 [5ee5c3739e] Corey Farrell + + * main/pbx: Move variable routines to pbx_variables.c. + + This is the third patch in a series meant to reduce the bulk of pbx.c. + This moves channel and global variable routines to their own source. + + Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6 + +2015-12-04 17:22 +0000 [f88b952093] Richard Mudgett + + * app_dial: Immediately exit dial if the caller is already hung up. + + If a caller hangs up before dial is executed within an AGI then the AGI + has likely eaten all queued frames before executing the dial in DeadAGI + mode. With the caller hung up and no pending frames from the caller's + read queue, dial would not know that the call has hung up until a called + channel answers. It is rather annoying to whoever just answered the + non-existent call. + + Dial should not continue execution in DeadAGI mode, hangup handlers, or + the h exten. + + * Added a check early in dial to abort dialing if the caller has hungup. + + ASTERISK-25307 #close + Reported by: David Cunningham + + Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418 + +2016-01-02 10:26 +0000 [e9dd16364e] Matt Jordan + + * main/cdr: Allow setting properties on a finalized CDR if it is the last one + + Prior to this patch, we explicitly disallowed setting any properties on a + finalized CDR. This seemed like a good idea at the time; in practice, it was + more restrictive. + + There are weird and strange scenarios where setting a property on a finalized + CDR is definitely wrong. For example, we may Fork a CDR, finalizing the + previous one, then change a property. In said case, the old CDR is supposed + to now be 'immutable' (so to speak), and should not be updated. From the + perspective of the code, a forked CDR that is finalized is just finalized. + Hence why we decided these should not be updated. + + In practice, it is much more common to want to set a property on a CDR in + the h extension or in a hangup handler. Disallowing a common scenario to make + an esoteric behaviour work isn't good. This patch fixes this by allowing + callers to set a property IF we are the last CDR in the chain. This preserves + the finalized CDR if it was forked, while allowing the more common case to + function. + + ASTERISK-25458 #close + + Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9 + +2016-01-02 10:23 +0000 [153547a9b1] Matt Jordan + + * main/cdr: Set the end time on a CDR if endbeforehexten is Yes + + Prior to this patch, the CDR engine attempted to set the end time on a CDR + that was executing hangup logic and with endbeforehexten set to Yes by + calling a function that inspects the properties on the Party A snapshot to + determine if we are ready to set the end time. That always failed. This is + because a Party A snapshot is not updated for CDRs that are executing hangup + logic with endbeforehexten=Yes. + + Instead of calling a function that looks at the Party A snapshot, we just + simply set the end time on the CDR. This is safe to call multiple times, and is + safe to call at this point as we know that (a) we are executing hangup logic, + and (b) we are supposed to set the end time at this point. + + ASTERISK-25458 + + Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3 + +2015-12-30 20:51 +0000 [f9bfc2450e] Corey Farrell + + * main/pbx: Move custom function routines to pbx_functions.c. + + This is the second patch in a series meant to reduce the bulk of pbx.c. + This moves custom function management routines to their own source. + + Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177 + +2016-01-01 05:25 +0000 [3fd528dddf] Rodrigo Ramírez Norambuena + + * Happy new year 2016. + + Change-Id: I22d3c90f6f27df82e915bbf81c1d91221f7a945e + +2015-12-13 13:09 +0000 [9cdf3ec19d] Matt Jordan + + * res_pjsip_history: Add a module that provides PJSIP history for debugging + + This patch adds a new module, res_pjsip_history, that provides a slightly + better way of debugging SIP message traffic on a busy Asterisk system. The + existing mechanisms all rely on passively dumping a SIP message to the CLI. + While this is perfectly fine for logging purposes and well controlled + environments, on many installations, the amount of SIP messages Asterisk + receives will quickly swamp the CLI. This makes it difficult to view/capture + those messages that you want to diagnose in real time. + + This patch provides another way of handling this. When enabled, the module + will store SIP message traffic in memory. This traffic can then be queried + at leisure. + + In order to make the querying useful, a CLI command has been implemented, + 'pjsip show history', that supports a basic expression syntax similar to + SQL or other query languages. A small number of useful fields have been + added in this initial patch; additional fields can easily be added in + later improvements. Those fields are: + - number: The entry index in the history + - timestamp: The time the message was recieved + - addr: The source/destination address of the message + - sip.msg.request.method: The request method + - sip.msg.call-id: The Call-ID header + + Note - this is a resurrection of the module initially proposed on Review Board + here: https://reviewboard.asterisk.org/r/4053/ + + Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36 + +2015-12-28 19:18 +0000 [5e67e51c6a] gtjoseph + + * main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c + + We joked about splitting pbx.c into multiple files but this first step was + fairly easy. All of the pbx_builtin dialplan applications have been moved + into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() + is called by asterisk.c just after load_pbx(). + + A few functions were renamed and are cross-exposed between the 2 source files. + + Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a + +2015-12-28 14:02 +0000 [a05bb258b1] Joshua Colp + + * test_time: Provide a timeout when waiting. + + The test_timezone_watch unit test is written to expect a + condition to be signaled when the inotify daemon thread runs. + There exists a small window where the test_timezone_watch + thread can signal the inotify daemon thread while it is not + reading on the underlying file descriptor. If this occurs + the test_timezone_watch thread will wait indefinitely for a + signal that will never arrive. + + This change adds a timeout to the condition so it will return + regardless after a period of time. + + Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390 + +2015-12-24 20:26 +0000 [96b32e0321] Matt Jordan + + * tests/test_stasis_endpoints: Remove expected duplicate events + + The cache_clear test was written to expect duplicate Stasis messages + sent from the technology endpoint to the all caching topic. This patch + fixes the test to no longer expect these duplicate messages. + + ASTERISK-25137 + + Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981 + +2015-12-24 22:19 +0000 [3bddcc0219] Dade Brandon + + * res_http_websocket.c: prevent avoidable disconnections caused by write errors + + Updated ast_websocket_write to encode the entire frame in to one + write operation, to ensure that we don't end up with a situation + where the websocket header has been sent, while the body can not + be written. + + Previous to August's patch in commit b9bd3c14, certain network + conditions could cause the header to be written, and then the + sub-sequent body to fail - which would cause the next successful + write to contain a new header, and a new body (resulting in + the peer receiving two headers - the second of which would be + read as part of the body for the first header). + + This was patched to have both write operations individually fail + by closing the websocket. + + In a case available to the submitter of this patch, the same + body which would consistently fail to write, would succeed + if written at the same time as the header. + + This update merges the two operations in to one, adds debug messages + indicating the reason for a websocket connection being closed during + a write operation, and clarifies some variable names for code legibility. + + Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598 + +2015-05-27 13:22 +0000 [22db16fa81] gtjoseph + + * endpoint/stasis: Eliminate duplicate events on endpoint status change + + When an endpoint is created, its messages are forwarded to both the tech + endpoint topic and the all endpoints topic. This is done so that various + parties interested in endpoint messages can subscribe to just the tech + endpoint and receive all messages associated with that particular technology, + as opposed to subscribing to the all endpoints topic. Unfortunately, when the + tech endpoint is created, it also forwards all of its messages to the all + topic. This results in duplicate messages whenever an endpoint publishes its + messages. + + This patch resolves the duplicate message issue by creating a new function + for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts + as a normal caching topic, save that it no longer forwards messages it receives + to the all endpoints topic. This allows it to act as an aggregation "sink", + while preserving the necessary caching behaviour. + + ASTERISK-25137 #close + Reported-by: Vitezslav Novy + + ASTERISK-25116 #close + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b + +2015-12-27 22:38 +0000 [6b08f01c60] Corey Farrell + + * Remove res_jabber file that was left behind. + + Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b + +2015-12-26 09:24 +0000 [d4b10cfb3e] Ward van Wanrooij + + * chan_sip: option 'notifyringing' change and doc fix + + In the sample sip.conf this is written with regard to notifyringing: + ;notifyringing = no ; Control whether subscriptions already INUSE get sent + RINGING when another call is sent (default: yes) + + However, this setting changes whether or not any RINGING indications are sent + to subscriptions. There is no separate configurable setting that allows + to control whether INUSE subscriptions also get sent RINGING. This is however + a useful option, to see (using BLF) if somebody else is able to handle an + incoming call or if everybody is busy. + + This patch corrects the documentation for notifyringing (so the documentation + matches the functionality) and make notifyringing a tri-state option, by adding + the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing = + notinuse, only subscriptions that are not INUSE are sent the RINGING signal. + + The default setting for notifyringing remains set to yes, so the default + behaviour is not affected. + + ASTERISK-25558 + + Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa + +2015-12-25 09:56 +0000 [6dc21bbf00] Dade Brandon + + * chan_sip.c: fix websocket_write_timeout default value + + websocket_write_timeout was not being set to its default value + during sip config reload, which meant that prior to this commit, + 1) the default value of 100 was not used, unless an invalid value + (or 1) was specified in sip.conf for websocket_write_timeout, and + 2) if the websocket_write_timeout directive was removed from sip.conf + without a full restart of asterisk, then the previous value would + continue to be used indefinitely. + + This essentially lead to a 0ms write timeout (the first write attempt + in ast_careful_fwrite must have succeeded) in websocket write requests + from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf. + + Changes to websocket_write_timeout still only apply to new websocket + sessions, after the sip reload -- timeouts on existing sessions are + not adjusted during sip reload. + + Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953 + +2015-12-23 17:40 +0000 [8eb5da0679] Richard Mudgett + + * bridge_basic.c: Fix GOTO_ON_BLINDXFR + + Use of GOTO_ON_BLINDXFR would not work at all. The target location would + never be executed by the transferring channel. + + * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with + valid context, exten, and priority parameters from the transferring + channel. + + * Renamed some feature_blind_transfer() local variables for clarity. + + ASTERISK-25641 #close + Reported by Dmitry Melekhov + + Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac + +2015-12-24 12:19 +0000 [2df4ad647c] Matt Jordan + + * res/res_pjsip_location: Delete contact_status object when contact is deleted + + In 450579e908, a change was made that removed the deletion of the + 'contact_status' object when a 'contact' object is deleted in sorcery. + This unfortunately means that the 'contact_status' object persists, even when + something has explicitly removed a contact. The result is that the state of + the contact will not be regenerated if that contact is re-created, and the + stale state will be reported/used for that contact. It also results in + no ContactStatusChanged events being generated for either ARI or AMI. + + This patch restores the deletion logic that was removed. Doing so now + results in the expected events being generated again. + + Change-Id: I28789a112e845072308b5b34522690e3faf58f07 + +2015-12-24 10:18 +0000 [b8876711f3] Kevin Harwell + + * res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef + + Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151 + +2015-12-20 21:33 +0000 [ca394161cf] Dade Brandon + + * app_amd: Correct maximum_number_of_words functionality & documentation + + - The maximum_number_of_words was previously documented as being + the number of words that when exceeded, would result in the AMD + application returning that the audio represents a machine. + + This was inconsistent with its actual functionality - it was + a number of words that when REACHED, would result in determination + as a machine. + + This update corrects the functionality to match the previously + documented functionality. This is a backwards incompatible change + in configuration file, and has been added to UPGRADE.txt as a result. + + The sample configuration file and application defaults have been updated + so that the default value is now 2, which reflects the same default + functionality as previous versions. + + - Update documentation for silence_threshold, which previously implied + that it was measuring time, rather than noise averages in the sample. + + - Update the comments in amd.conf.sample. + + ASTERISK-25639 #close + Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093 + +2015-12-17 19:05 +0000 [648ca2b1b8] Dade Brandon + + * res_rtp_asterisk: Resolve further timing issues with DTLS negotiation + + Resolves an edge case dtls negotiation delay for certain networks which + somehow manage to drop the rtcp side's packet when these are both sent + ast_rtp_remote_address_set, causing it to have to time-out and restart + the handshake. + + Move dtls pending bio flush in to it's own function, and call it from + ast_rtp_on_ice_complete, when we're rtp->ice, rather than when + ast_rtp_remote_address_set. + + Keep the existing flush from the recent change to res_rtp_remote_address_set + if ice is not being used. + + ASTERISK-25614 #close + Reported-by: XenCALL + Tested by: XenCALL + + Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5 + +2015-12-05 10:01 +0000 [902309fd04] Joshua Colp + + * res_sorcery_memory_cache: Add support for a full backend cache. + + This change introduces the configuration option 'full_backend_cache' + which changes the cache to be a full mirror of the backend instead + of a per-object cache. This allows all sorcery retrieval operations + to be carried out against it and is useful for object types which + are used in a "retrieve all" or "retrieve some" pattern. + + ASTERISK-25625 #close + + Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5 + +2015-12-17 10:25 +0000 [a2431f83ef] Joshua Colp + + * rtp_engine: Ignore empty filenames in DTLS configuration. + + When applying an empty DTLS configuration the filenames in the + configuration will be empty. This is actually valid to do and + each filename should simply be ignored. + + Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539 + +2015-12-17 08:10 +0000 [d2c8614122] Joshua Colp + + * chan_sip: Enable WebSocket support by default. + + Per the documentation the WebSocket support in chan_sip is + supposed to be enabled by default but is not. This change + corrects that. + + Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423 + +2015-12-14 12:04 +0000 [d17d9a9288] Joshua Colp + + * json: Audit ast_json_* usage for thread safety. + + The JSON library Asterisk uses, jansson, is not thread + safe for us in a few ways. To help with this wrappers for JSON + object reference count increasing and decreasing were added + which use a global lock to ensure they don't clobber over + each other. This does not extend to reference count manipulation + within the jansson library itself. This means you can't safely + use the object borrowing specifier (O) in ast_json_pack and + you can't share JSON instances between objects. + + This change removes uses of the O specifier and replaces them + with the o specifier and an explicit ast_json_ref. Some cases + of instance sharing have also been removed. + + ASTERISK-25601 #close + + Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1 + +2015-12-16 11:28 +0000 [cfb34adb83] Mark Michelson + + * Alembic: Increase column size of PJSIP AOR "contact". + + When running the PJSIP AMI "show_endpoint" test with automatic + conversion to realtime, the test would fail. This was because the AOR + "contact" column was sized at 40, and the configured contact was larger + than that. + + This commit increases the size of the contact column to 255 characters. + + Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1 + +2015-12-14 13:53 +0000 [32ec83f37f] server-pandora + + * res_rtp_asterisk.c: Fix DTLS negotiation delays. + + - Trigger pending DTLS packets to send out, once the RTP instance's remote + address is set. + - Avoids locking the DTLS structure unnecessarily by only doing this if + DTLS is passive. + - Add DTLS locks around the structurally sensitive calls in the SSL + portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock + inside of itself, and we're dealing with the SSL BIO in at least two + threads. + + WebRTC channels may receive a DTLS handshake before + ast_rtp_remote_address_set is called, which causes there to be a pending + response to send out. Previous to 1ad827, this was handled by calling + dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP + packet could trigger the pending handshake response. Since that was + rightfully removed, whenever the DTLS handshake is received before the + remote address is set, we would have to wait until another SSL packet + arrives. + + As of Chrome M47's optimizations to their handshake process, WebRTC + conversations between Chrome M47+ and Asterisk, where Asterisk is passive, + experience a 1 second delay without this patch, because the SSL handshake + is received before ICE negotation stores the remote_address, and the next + SSL packet isn't received until after a 1 second timeout in Chrome, which + causes a new handshake request. + + ASTERISK-25614 #close + + Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908 + +2015-12-08 13:04 +0000 [52ca6fb94a] sungtae kim + + * AMI: Fixed OriginateResponse message + + When the asterisk sending OriginateResponse message, + it doesn't set the "Uniqueid". + And it didn't support correct response message for + Application originate. + + ASTERISK-25624 #close + + Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d + +2015-12-14 15:25 +0000 [eccdf2250b] Richard Mudgett + + * Fix sscanf() format string type mismatch. + + ASTERISK-25615 + Reported by: George Joseph + + Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b + +2015-12-14 06:26 +0000 [3e7522533c] Carlos Oliva + + * app_queue: update RT members when the 1st call joins a queue with no agents + + If a call enters on a queue and the members on that queue are updated in + realtime (ex: using mysql inserting a new agent) the queue members are + never refreshed and the call will stay in the queue until other event occurs. + This happens only if this is the first call of the queue and there is no + agents servicing. + This patch prevent this issue, ensuring realtime members are updated if + there is one call in the queue and no available agents + + ASTERISK-25442 #close + + Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682 + +2015-12-13 13:13 +0000 [9a96a86e2d] Matt Jordan + + * main/utils: Don't emit an ERROR message if the read end of a pipe closes + + An ERROR or WARNING message should generally indicate that something has gone + wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not + in control of when the far end closes its reading on a file descriptor. If the + far end does close the file descriptor in an unclean fashion, this isn't a bug + or error in Asterisk, particularly when the situation can be gracefully + handled in Asterisk. + + Currently, when this happens, a user would see the following somewhat cryptic + ERROR message: + + "utils.c: write() returned error: Broken pipe" + + There's a few problems with this: + (1) It doesn't provide any context, other than 'something broke a pipe' + (2) As noted, it isn't actually an error in Asterisk + (3) It can get rather spammy if the thing breaking the pipe occurs often, such + as a FastAGI server + (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even + mask legitimate issues + + This patch changes ast_carefulwrite to only log an ERROR if we actually had one + that was reasonably under our control. For debugging purposes, we still emit + a debug message if we detect that the far side has stopped reading. + + Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566 + +2015-12-12 11:08 +0000 [3e6637feb5] gtjoseph + + * pjsip/config_transport: Check pjproject version at runtime for async ops + + pjproject < 2.5.0 will segfault on a tls transport if async_operations + is greater than 1. A runtime version check has been added to throw + an error if the version is < 2.5.0 and async_operations > 1. + + To assist in the check, a new api "ast_compare_versions" was added + to utils which compares 2 major.minor.patch.extra version strings. + + ASTERISK-25615 #close + + Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-12-10 11:44 +0000 [ceebdfce40] Jonathan Rose + + * chan_sip: Add TCP/TLS keepalive to TCP/TLS server + + Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously + this option was only being set on session sockets. + http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ + According to the link above, the SO_KEEPALIVE option is useful for knowing + when a TCP connected endpoint has severed communication without indicating + it or has become unreachable for some reason. Without this patch, keep + alive is not set on the socket listening for incoming TCP sessions and + in Komatsu's report this resulted in the thread listening for TCP becoming + stuck in a waiting state. + + ASTERISK-25364 #close + Reported by: Hiroaki Komatsu + + Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36 +2015-12-07 13:07 +0000 [fcaebb0e43] Corey Farrell + + * app_meetme: Set default value for audio_buffers. + + The default value was never set for audio_buffers, causing bad + audio quality. This ensures the default is always set. + + ASTERISK-25569 #close + + Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44 +2015-12-09 09:48 +0000 [5790700497] Tyler Cambron + + * res_chan_stats: Fix bug to send correct statistics to StatsD + + Fixed a bug that originally would show a negative number of + active calls occuring in Asterisk. A gauge is persistent so + incrementing and decrementing it results in a more consistent + performance. Also changed to the call to StatsD to use + ast_statsd_log_string() so that a "+" could be sent to StatsD. + + ASTERISK-25619 #close + + Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7 +2015-12-08 17:49 +0000 [a987434564] gtjoseph + + * res_pjsip: Add existence and readablity checks for tls related files + + Both transport and endpoint now check for the existence and readability + of tls certificate and key files before passing them on to pjproject. + This will cause the object to not load rather than waiting for pjproject + to discover that there's a problem when a session is attempted. + + NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located + in build_peer which is gigantic and I didn't want to disturb it. + Error messages will emit but it won't interrupt chan_sip loading. + + ASTERISK-25618 #close + + Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-12-02 12:42 +0000 [be693539c3] Eugene Voityuk + + * chan_sip.c: Start ICE negotiation when response is sent or received. + + The current logic for ICE negotiation starts it + when receiving an SDP with ICE candidates. This is + incorrect as ICE negotiation can only start when each + call party have at least one pair of local and remote + candidate. Starting ICE negotiation early would result + in negotiation failure and ultimately no audio. + + This change makes it so ICE negotiation is only started + when a response with SDP is received or when a response + with SDP is sent. + + ASTERISK-24146 + + Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca + +2015-12-08 01:57 +0000 [59a91c350a] Filip Jenicek + + * chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) + + Asterisk may crash when calling ast_channel_get_t38_state(c) + on a locked channel which is being hung up. + + ASTERISK-25609 #close + + Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b +2015-12-08 11:03 +0000 [28ab03fbf7] gtjoseph + + * res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls + + See ASTERISK-25615. + If the transport protocol is tls and async_operations > 1, pjproject + will segfault if more than one operation is attempted on the same socket. + Until this is fixed upstream, a check has been added to throw an error + if a tls transport config has async_operations set to > 1. + + ASTERISK-25615 + + Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-12-08 08:39 +0000 [55dd7125b3] Alexander Traud + + * codec_resample: Increase buffer for Opus Codec with FEC. + + ASTERISK-25599 #close + + Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e + +2015-12-08 03:46 +0000 [64f899e5f3] Alexander Traud + + * translate: Avoid a warning message when doing FEC within Opus Codec. + + ASTERISK-25616 #close + + Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319 + +2015-12-04 15:36 +0000 [65c8147952] Richard Mudgett + + * chan_sip: Fix crash involving the bogus peer during sip reload. + + A crash happens sometimes when performing a CLI "sip reload". The bogus + peer gets refreshed while it is in use by a new call which can cause the + crash. + + * Protected the global bogus peer object with an ao2 global object + container. + + ASTERISK-25610 #close + + Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed + +2015-11-13 07:58 +0000 [48c065e46d] Christof Lauber + + * chan_sip: Support parsing of Q.850 reason header in SIP BYE and CANCEL requests. + + Current support for reason header did work only in SIP responses. + According to RFC3336 the reason header might appear in any SIP request. + But it seems to make most sence in BYE and CANCEL so parasing is done + there too (if use_q850_reason=yes). + + Change-Id: Ib6be7b34c23a76d0e98dfd0816c89931000ac790 + +2015-12-06 16:35 +0000 [75c800eb28] Matt Jordan + + * Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" + + This reverts commit f42d22d3a1ca5c8ea73df99a50c6a28caa8f8749. + + Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks + in core_unreal/chan_local. Local channels attempt to reach across both their + peer and the peer's bridge to inspect T.38 state. Given the propensity of + Local channel chains, managing the locking situation in such a scenario is + practically infeasible. + + Change-Id: I932107387c13aad2c75a7a4c1e94197a9d6d8a51 + +2015-12-04 16:23 +0000 [4be231e82f] gtjoseph + + * res_pjsip/contacts/statsd: Make contact lifecycle events more consistent + + It will never be perfect or even pretty, mostly because of the differences + between static and dynamic contacts. + + Created: + + Can't use the contact or contact_status alloc functions + because the objects come and go regardless of the actual state. + + Can't use the contact_apply_handler, ast_sip_location_add_contact or + a sorcery created handler because they only get called for dynamic + contacts. Similarly, permanent_uri_handler only gets called for + static contacts. + + So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is + the only place it can go and not have duplicated code. Both + permanent_uri_handler and contact_apply_handler call find_or_create. + + Removed: + + Can't use the destructors for the same reason as above. The only + place to put this is in persistent_endpoint_contact_deleted_observer + which I believe is the "correct" place but even that will handle only + dynamic contacts. This doesn't called on shutdown however. There is + no hook to use for static contacts that may be removed because of a + config change while asterisk is in operation. + + I moved the cleanup of contact_status from ast_sip_location_delete_contact + to the handler as well. + + Status Change and RTT: + + Although they worked fine where they were (in update_contact_status) I + moved them to persistent_endpoint_contact_status_observer to make it + more consistent with removed. There was logic there already to detect + a state change. + + Finally, fixed a nit in permanent_uri_handler rmudgett reported + eralier. + + ASTERISK-25608 #close + + Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d + Reported-by: George Joseph + Tested-by: George Joseph + +2015-11-21 06:08 +0000 [63c6d39a3e] Alexander Traud + + * res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8. + + ASTERISK-25584 #close + + Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91 + +2015-11-28 08:46 +0000 [f42d22d3a1] Matt Jordan + + * bridges/bridge_t38: Add a bridging module for managing T.38 state + + When 4875e5ac32 was merged, it fixed several issues with a direct media bridge + transitioning to handling a T.38 fax. However, it uncovered a race condition + caused by the bridging core. When a channel involved in a T.38 fax leaves a + bridge, the frame queued by the channel driver that should inform the far side + that it is no longer in a T.38 fax may not make it across the bridge. The + bridging framework is *extremely* aggressive in tearing down the bridge, and + control frames that are currently in flight *may* get dropped. + + This patch adds a new module to the bridging framework, bridge_t38. This module + maintains some notion of the T.38 state for the two channels in a bridge. When + the bridge detects that it is being torn down or when one of the two channels + leaves, it informs the respective channel(s) that they should stop faxing. This + ensures that channels switch back to audio if they survive and are ejected out + of a bridge while faxing. + + ASTERISK-25582 + + Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0 + +2015-11-21 05:35 +0000 [dcc01bc0a7] Alexander Traud + + * res_format_attr_opus: Update to latest RFC 7587. + + Beside that, the format-attribute module sends only non-default values in the + line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore, + previously the parameter stereo was not parsed when being the first parameter. + + ASTERISK-25583 #close + + Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73 +2015-12-02 14:11 +0000 [69457b8d61] Jonathan Rose + + * Fix crash in audiohook translate to slin + + This patch fixes a crash which would occur when an audiohook was + applied to a channel using an audio codec that could not be translated + to signed linear (such as when using pass-through codecs like OPUS or + when the codec translator module for the format in use is not loaded). + + ASTERISK-25498 #close + Reported by: Ben Langfeld + + Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384 +2015-12-03 12:07 +0000 [5959186017] gtjoseph + + * res_pjsip: Use a MD5 hash for static Contact IDs + + When 90d9a70789 was merged, it mostly tested dynamic contacts created as + a result of registering a PJSIP endpoint. Contacts generated in this + fashion typically have a long alphanumeric string as their object identifier, + which maps reasonably well for StatsD. Unfortunately, this doesn't work in the + general case. StatsD treats both '.' and ':' characters as special characters. + In particular, having a ':' appear in the middle of a StatsD metric will + result in the metric being rejected. + + This causes some obvious issues with SIP URIs. + + The StatsD API should not be responsible for escaping the metric name passed + to it. The metric is treated as a single long string, and it would be + challenging to know what to escape in the string passed to the function. + Likewise, we don't want to escape the metric in PJSIP, as that involves + overhead that is wasted when either res_statsd isn't loaded or enabled. + + This patch takes an alternative approach. The Contact ID has been changed + to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the + aforementioned special characters, (b) can be done on Contact creation, + which has minimal impact on run-time performance, and (c) also conforms to an + earlier commit that changed the ID for dynamic contacts. + + The downside of this is that StatsD users will have to map SHA1 hashes back to + the Contacts that are emitting the statistics. To that end, the CLI commands + have been updated to include the first 10 characters of the MD5 hash, which + should be enough to match what is shown in Graphite (or some other StatsD + backend). + + ASTERISK-25595 #close + + Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2 + Reported-by: Matt Jordan + Tested-by: George Joseph + +2015-11-30 22:19 +0000 [bd265a90be] gtjoseph + + * res_pjsip: Update logging to show contact->uri in messages + + An earlier commit changed the id of dynamic contacts to contain + a hash instead of the uri. This patch updates status change + logging to show the aor/uri instead of the id. This required + adding the aor id to contact and contact_status and adding + uri to contact_status. The aor id gets added to contact and + contact_status in their allocators and the uri gets added to + contact_status in pjsip_options when the contact_status is + created or updated. + + ASTERISK-25598 #close + + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511 + +2015-12-01 16:11 +0000 [b5281b74e0] Jonathan Rose + + * Unset BRIDGEPEER when leaving a bridge + + Currently if a channel is transferred out of a bridge, the BRIDGEPEER + variable (also BRIDGEPVTCALLID) remain set even once the channel is + out of the bridge. This patch removes these variables when leaving + the bridge. + + ASTERISK-25600 #close + Reported by: Mark Michelson + + Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da + +2015-11-30 14:22 +0000 [59ba84e5cd] Richard Mudgett + + * res_sorcery_memory_cache.c: Fix off nominal ref leak. + + Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49 + +2015-11-30 16:42 +0000 [ef77439e39] Richard Mudgett + + * sched.c: Make not return a sched id of 0. + + According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0 + was never returned historically and several users incorrectly coded usage + of the returned sched ID assuming that 0 was invalid. + + ASTERISK-25476 + + Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20 + +2015-11-25 12:23 +0000 [145d10a5d0] Richard Mudgett + + * Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions) + + chan_sip.c: + * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to + ao2 conversion. + + * Initialize register scheduler ids earlier because of ASTOBJ to ao2 + conversion. + + chan_skinny.c: + * Fix more scheduler usage for the valid 0 id value. + + ASTERISK-25476 + + Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95 + +2015-11-24 12:44 +0000 [fa20729032] Richard Mudgett + + * Audit improper usage of scheduler exposed by 5c713fdf18f. + + channels/chan_iax2.c: + * Initialize struct chan_iax2_pvt scheduler ids earlier because of + iax2_destroy_helper(). + + channels/chan_sip.c: + channels/sip/config_parser.c: + * Fix initialization of scheduler id struct members. Some off nominal + paths had 0 as a scheduler id to be destroyed when it was never started. + + chan_skinny.c: + * Fix some scheduler id comparisons that excluded the valid 0 id. + + channel.c: + * Fix channel initialization of the video stream scheduler id. + + pbx_dundi.c: + * Fix channel initialization of the packet retransmission scheduler id. + + ASTERISK-25476 + + Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8 + +2015-12-01 07:55 +0000 [b24f2f4c2e] Alexander Traud + + * codec_resample: Increase buffer for Opus Codec. + + ASTERISK-25599 #close + + Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10 + +2015-11-30 11:13 +0000 [e5723d2776] gtjoseph + + * dns: Change lookup failures from LOG_ERROR to debug 1. + + dns.c and dns_system_resolver.c were spitting out errors for lookup + failures for things like not finding a SRV record even though + there was an A record. Those have been changed to debug messages. + Logging not finding ANY record is left to the higher level caller. + + Also, dns_system_resolver was using Windows line endings so I + converted them to Unix style. The actual log changes are on lines + 156 and 159. + + Change-Id: I65be16ea15304b96f9dcb4d289dbd3e2286fc094 + +2015-11-25 10:42 +0000 [270f7be54f] Alexander Traud + + * Build System: Support include-what-you-use. + + ASTERISK-25591 #close + + Change-Id: I8d3efa0826142ece9cbed2fd0d46f3b607fee6ae + +2015-11-08 23:49 +0000 [f2a84b500d] Rodrigo Ramírez Norambuena + + * app_queue: Show reason of pause on CLI + + Add value of pause reason when is paused on CLI command "queue show" + + ASTERISK-25581 #close + + Report by: Rodrigo Ramírez Norambuena + + Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864 + +2015-11-27 07:39 +0000 [7cb8f2f33e] Niklas Larsson + + * CHANGES: Fix a typo + + Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7 + +2015-11-25 15:26 +0000 [9014f1f4a5] Kevin Harwell + + * fastagi: record file closed after sending result + + The fastagi record-file testsuite test sometimes fails reporting an empty + recorded file. This was happening because Asterisk was sending the agi result + notification prior to actually closing the file and the data, being buffered, + had not been written to the file yet when the test attempts to check the file + size. + + This patch makes it so the record file stream is closed prior to sending the + agi result notification. + + ASTERISK-25593 #close + + Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde + +2015-11-25 13:29 +0000 [03759c5587] Walter Doekes + + * main: Slight refactor of main. Improve color situation. + + Several issues are addressed here: + - main() is large, and half of it is only used if we're not rasterisk; + fixed by spliting up the daemon part into a separate function. + - Call ast_term_init from rasterisk as well. + - Remove duplicate code reading/writing asterisk history file. + - Attempt to tackle background color issues and color changes that + occur. Tested by starting asterisk -c until the colors stopped + changing at odd locations. + - Remove unused term_prep() and term_prompt() functions. + + ASTERISK-25585 #close + + Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f + +2015-11-24 13:54 +0000 [91346b9fb7] David M. Lee + + * Fixed some typos + + Fixes some minor typos in the CHANGES file, plus an embarrasing typo in + the StatsD API. + + Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7 + +2015-11-24 13:07 +0000 [fb45130476] Corey Farrell + + * res_pjsip_notify: Fix CLI usage info + + The usage info for 'pjsip send notify' previously referenced the + chan_sip configuration sip_notify.conf. Fix this to reference + the correct configuration pjsip_notify.conf. + + ASTERISK-25590 #close + + Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea + +2015-11-18 09:43 +0000 [ee9c114747] Matt Jordan + + * res/res_endpoint_stats: Add module to emit endpoint StatsD statistics + + This patch adds a module that emits StatsD statistics about Asterisk + endpoints. This includes: + * A GAUGE statistic for endpoint states, tracking how many endpoints are in + a particular state. + * A GAUGE statistic for each endpoint, counting the number of channels + currently associated with an endpoint. + + ASTERISK-25572 + + Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305 +2015-11-23 14:27 +0000 [9ca652f1b9] Richard Mudgett + + * res_sorcery_realtime.c: Fix crash from NULL sorcery object type. + + If the sorcery object type is not found a NULL is returned. + Unfortunately, sorcery_realtime_filter_objectset() will crash after + complaining about not finding the object type and saying to expect errors. + + * Use ao2_cleanup() instead of ao2_ref() to prevent the crash. + + ASTERISK-25165 + Reported by Corey Farrell + + Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97 + +2015-11-18 10:07 +0000 [75d90a9951] Matt Jordan + + * res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts + + This patch adds the ability to send StatsD statistics related to the + state of PJSIP contacts. This includes: + * A GUAGE statistic measuring the count of contacts in a particular state. + This measures how many contacts are reachable, unreachable, etc. + * The RTT time for each contact, if those contacts are qualified. This + provides StatsD engines useful time-based data about each contact. + + ASTERISK-25571 + + Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c + +2015-11-13 10:34 +0000 [482f2fc5ff] Matt Jordan + + * res/res_pjsip_outbound_registration: Add registration statistics for StatsD + + This patch adds outbound registration statistics for StatsD. This includes + the following: + * A GUAGE metric for the overall count of outbound registrations. + * A GUAGE metric for each state an outbound registration can be in. As the + outbound registrations change state, the overall count of how many + outbound registrations are in the particular state is changed. + + These statistics are particularly useful for systems with a large number of + SIP trunks, and where measuring the change in state of the trunks is useful + for monitoring. + + ASTERISK-25571 + + Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37 + +2015-11-18 10:05 +0000 [97d7b344de] Matt Jordan + + * res_statsd: Add functions that support variable arguments + + Often, the metric names of statistics we are generating for StatsD have some + dynamic component to them. This can be the name of a particular resource, or + some internal status label in Asterisk. With the current set of functions, + callers of the statsd API must first build the metric name themselves, then + pass this to the API functions. This results in a large amount of boilerplate + code and usage of either fixed length static buffers or dynamic memory + allocation, neither of which is desireable. + + This patch adds two new functions to the StatsD API that support a printf + style format specifier for constructing the metric name. A dynamic string, + allocated in threadstorage, is used to build the metric name. This eases + the burden on users of the StatsD API. + + Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea + +2015-11-20 21:08 +0000 [726ee873a6] Matt Jordan + + * chan_pjsip: Handle T.38 faxes with direct media bridges + + When a channel is in a direct media bridge, a re-INVITE may arrive that forces + Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge + must change its technology to a simple bridge, and re-INVITE the media back + to Asterisk. + + Generally, this logic mostly already exists in Asterisk. However, prior to this + patch, there were a few bugs: + (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from + ever entering into a direct media bridge. This applies even when the only + media being passed over the channel is audio. This patch fixes this bug + by having the framehook specify that it defers caring about any frame type. + This allows the channels to enter into a direct media bridge, which will + be broken when a re-INVITE is received. + (2) When a re-INVITE is received, nothing instructed the bridging layer to + re-inspect the allowed bridging technology. This now occurs when either + a re-INVITE is received from a peer, or when a response is received from + the far end (that is, when the T.38 state changes to either + T38_PEER_REINVITE or T38_LOCAL_REINVITE). + (3) chan_pjsip needs to do a small amount of work to prevent a direct media + bridge from being chosen when a T.38 session is in progress. When a T.38 + session supplement has a t38 datastore - which is added when we detect + we should start thinking about T.38 on a channel - we now refuse a native + RTP bridge. + (4) When a BYE request is received, we don't terminate the T.38 session. If + the other side of a T.38 fax survives the hangup (due to the 'g' flag + in Dial, for example), we don't currently re-INVITE the media on the + other channel back to audio. This patch now has res_pjsip_t38 intercept + BYE requests and inform the far side that the T.38 session is terminated. + This naturally causes the correct re-INVITEs to be sent. + + ASTERISK-25582 + + Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb + +2015-10-22 09:44 +0000 [9315a93757] Matt Jordan + + * main/cli: Use proper string methods to check existence of context/exten/app + + Because the context, extension, and application are stored in stringfields, + checking for them being NULL doesn't work so well. This patch uses the + appropriate string library call, ast_strlen_zero, to see if there is a value + in the context/exten/app values. + + Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23 + +2015-11-20 21:07 +0000 [d2b141c79f] Matt Jordan + + * res/res_pjsip_t38: Add debug statements + + This patch adds some debug statements to res_pjsip_t38. These statements help + to determine which SDP negotiation callbacks are being executed, and, when + a particular callback exits, why a callback may not have applied its logic + to the local or remote SDP. + + Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77 + +2015-11-19 09:40 +0000 [1bca90fcbe] Matt Jordan + + * res/res_pjsip_outbound_registration: Apply configuration on object type load + + When Asterisk is configured to use a dynamic sorcery backend (such as + res_sorcery_astdb) with 'registration' objects, it will fail to create the + internal state objects associated with the registration objects on module + load. This is due to nothing actually querying for the specific objects + and calling their sorcery apply handler during module load. + + This patch fixes that by calling get_registrations in the sorcery observer's + object_type_loaded handler. Doing this causes the sorcery backends to be + asked for the current state of all registration objects, which causes the + apply handler to be called and the internal run-time state to be created. + + ASTERISK-25575 #close + + Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23 + +2015-11-11 06:29 +0000 [8ccb1d2bed] Alexander Traud + + * translate: Provide translation modules the result of SDP negotiation. + + Previously, a trancoding module did not have access to the joint but cached + format. Therefore, the module did not have access to the attributes negotiated + via SDP (line fmtp). Now, a translation module receives the joint format. + + ASTERISK-25545 #close + + Change-Id: Id6878a989b50573298dab115d3371ea369e1a718 + +2015-11-19 01:03 +0000 [92ea46ba94] Alexander Traud + + * res_format_attr_h264: Do not reset string buffer. + + When no parameter is present, Asterisk does not generate the line fmtp, as + expected. However, because a buffer was reset, even rtpmap and fmtp of previous + media codecs got removed. Now, Asterisk does not reset other codecs in case of + no parameter for H.264. + + ASTERISK-25573 #close + + Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286 + +2015-11-18 02:25 +0000 [8c14b91651] Alec Davis + + * app_bridgeaddchan: ability to barge into existing call + + To be able to barge into a call by dialling a prefix+extension that maps + to the extensions device. + + Senario is that DECT headset users may be away from their desks and need + to transfer the call, the goal is that from any phone they dial a prefix + then their extension and are added to the bridge that they are in, from + there they can drop the headset call, as it's also on the handset, + and transfer the caller. + + The dialplan would look like, where prefix=73, extension = 8512; + exten => _738512,1,BridgeAdd(SIP/cisco0001) + + ASTERISK-25551 #close + Reported By: Alec Davis + + Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540 + +2015-11-05 15:37 +0000 [05addf3d8f] Tyler Cambron + + * StatsD: Add sample rate compatibility + + Implemented support for the StatsD sample rate parameter, + which is a parameter for determining when to send computed + statistics to a client. + + Valid sample rate values are: + Less than or equal to 0.0 will never be sent. + Between 0.0 and 1.0 will randomly be sent. + Greater than or equal to 1.0 will always be sent. + + ASTERISK-25419 + Reported By: Ashley Sanders + + Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52 + +2015-11-17 14:53 +0000 [3dbaf696e9] Richard Mudgett + + * res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts. + + Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d + +2015-11-17 14:53 +0000 [eaf898ac88] Richard Mudgett + + * res_pjsip_outbound_registration.c: Fix 423 response handling. + + Receiving a 423 Interval Too Brief response after authentication for an + outbound registration attempt results in assuming that the registrar has + rejected the registration permanently. If there are no configured retries + for fatal responses then the outbound registration is stopped for that + endpoint. + + For registrations, PJSIP/PJPROJECT intercepts the handling of 423 + responses and does not include any authentication in the updated + registration request. When the updated request is challenged then the + Asterisk code assumes that we were challenged again because the peer + rejected the authentication we sent earlier. + + * Made registration challenges keep track of the CSeq number to determine + if the received challenge response was for the request we thought we sent. + If the response's CSeq number differs from the CSeq number we last sent + with authentication then authenticate again because it is a challenge to a + different request. + + Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09 + +2015-11-18 00:20 +0000 [4013f9d577] Alec Davis + + * app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! + + commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) + refer ASTERISK-24958 + + above commit removed ast_channel_lock(qe->chan); + but failed to remove corresponding ast_channel_unlock(qe->chan); + + ASTERISK-25561 #close + Reported Alec Davis + + Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a + +2015-11-16 16:10 +0000 [6919daab61] gtjoseph + + * dns: Fix pointer increment in dns_parse_answer_ex + + When dns_parse_answer_ex was iterating over the answers it + wasn't incrementing the answer pointer correctly after the first + answer. The result was that no answers after the first + were being returned. For results where multiple records should + have been sorted by priority, weight, etc., there was nothing + to sort so the only the first record was returned even if it + wouldn't have been the correct record based on the sort. + + ASTERISK-25565 #close + Reported-by: Daniel Tryba + Tested-by George Joseph + + Change-Id: I8622604fefdcd3c11e2c5609a6382e53b1467b0b + +2015-11-13 14:03 +0000 [ed13732188] Mark Michelson + + * Confbridge: Add a user timeout option + + This option adds the ability to specify a timeout, in seconds, for a + participant in a ConfBridge. When the user's timeout has been reached, + the user is ejected from the conference with the CONFBRIDGE_RESULT + channel variable set to "TIMEOUT". + + The rationale for this change is that there have been times where we + have seen channels get "stuck" in ConfBridge because a network issue + results in a SIP BYE not being received by Asterisk. While these + channels can be hung up manually via CLI/AMI/ARI, adding some sort of + automatic cleanup of the channels is a nice feature to have. + + ASTERISK-25549 #close + Reported by Mark Michelson + + Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98 + +2015-11-16 13:56 +0000 [a83e426e91] Matt Jordan + + * res/res_pjsip: Fix off nominal crash with requests that fail and have a timer + + When a request is sent using pjsip_endpt_send_request and fails, a condition + exists where the request wrapper, which is an AO2 object, may be de-ref'd + more times than it should. This occurs when the request's callback is called, + and, in the callback, the timer on the PJSIP heap is cancelled. When that + occurs, the request wrapper's lifetime is decremented. When + pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of + the request wrapper again, even though we've already cancelled the reference + associated with the timer. + + This patch checks the return result of pj_timer_heap_cancel_if_active before + removing the reference associated with the timer. We now only decrement it + in this case if a timer is cancelled as a result of the function call. + + Change-Id: I21332343a1a019c1117076f9bf2df27be2850102 + +2015-11-14 07:02 +0000 [a1fcf6f7b2] Joshua Colp + + * hashtab: Add NULL check when destroying iterator. + + The hashtab API is pretty NULL tolerant which has resulted + in remaining callers not doing much checks themselves. + Unfortunately the function to destroy an iterator does not + do a NULL check and will result in a crash if passed NULL. + This change fixes that. + + ASTERISK-25552 #close + + Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619 + +2015-11-13 14:32 +0000 [436023a322] Richard Mudgett + + * res_pjsip_rfc3326.c: Fix crash when channel goes away. + + If an authenticated incoming caller does not respond to our 200 OK INVITE + response with an ACK then PJSIP will hangup the call. Unfortunately, + there is a chance that the session's channel will go away between one use + of the channel pointer and another when building the BYE request because + the BYE is being built by the monitor thread and not the call's serializer + thread. + + * Added a check to ensure that the thread trying to add the Reason header + is the call's serializer thread. This ensures that the channel will not + go away on us. + + Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89 + +2015-11-13 14:19 +0000 [e8881e1770] Mark Michelson + + * Taskprocessors: Increase high-water mark + + In practical tests, we have seen certain taskprocessors, specifically + Stasis subscription taskprocessors, cross the recently-added high-water + mark and emit a warning. This high-water mark warning is only intended + to be emitted when things have tanked on the system and things are + heading south quickly. In the practical tests, the Stasis taskprocessors + sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in + any danger at all. + + As such, this ups the high-water mark to 500 tasks instead. It also + redefines the SIP threadpool request denial number to be a multiple of + the taskprocessor high-water mark. + + Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce + +2015-11-11 07:00 +0000 [fd23d423d8] Alexander Traud + + * format: Register format-attribute module with cached formats. + + In Asterisk 13, cached formats are created before their corresponding format- + attribute module is registered. Cached formats are involved when a local + extension is called. Therefore, ast_format_generate_sdp_fmtp did not work + on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264, + and format-attribute modules provided externally. + + ASTERISK-25160 #close + + Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354 + +2015-11-12 11:17 +0000 [40b58a5d2b] Mark Michelson + + * res_pjsip distributor: Don't send 503 response to responses. + + When the SIP threadpool is backed up with tasks, we send 503 responses + to ensure that we don't try to overload ourselves. The problem is that + we were not insuring that we were not trying to send a 503 to an + incoming SIP response. + + This change makes it so that we only send the 503 on incoming requests. + + Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404 + +2015-11-11 17:11 +0000 [264c74aa22] Mark Michelson + + * res_pjsip: Deny requests when threadpool queue is backed up. + + We have observed situations where the SIP threadpool may become + deadlocked. However, because incoming traffic is still arriving, the SIP + threadpool's queue can continue to grow, eventually running the system + out of memory. + + This change makes it so that incoming traffic gets rejected with a 503 + response if the queue is backed up too much. + + Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816 + +2015-11-12 06:24 +0000 [a159747660] Joshua Colp + + * format_cap: Don't append the 'none' format when appending all. + + When appending all formats of a type all the codecs are iterated + and added. This operation was incorrectly adding the ast_format_none + format which is special in that it is supposed to be used when no + format is present. It shouldn't be appended. + + ASTERISK-25535 + + Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c + +2015-11-11 04:16 +0000 [d982b99e71] Steve Davies + + * Further fixes to improper usage of scheduler + + When ASTERISK-25449 was closed, a number of scheduler issues mentioned in + the comments were missed. These have since beed raised in ASTERISK-25476 + and elsewhere. + + This patch attempts to collect all of the scheduler issues discovered so + far and address them sensibly. + + ASTERISK-25476 #close + + Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b + +2015-11-11 11:04 +0000 [2954354404] Joshua Colp + + * threadpool: Handle worker thread transitioning to dead when going active. + + This change adds handling of dead worker threads when moving them + to be active. When this happens the worker thread is removed from + both the active and idle threads container. If no threads are able + to be moved to active then the pool grows as configured. + + A unit test has also been added which thrashes the idle timeout + and thread activation to exploit any race conditions between the + two. + + ASTERISK-25546 #close + + Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143 + +2015-11-10 09:24 +0000 [525c7ab780] Alexander Traud + + * rtp_engine: Init a format-attribute module to its RFC defaults. + + Previously, format-attribute modules relied on an existing fmtp line in SDP + negotiation. However, fmtp is optional for several formats like the Opus Codec. + Now, the format-attribute module is called with an empty fmtp, which allows the + module to initialise itself to RFC defaults. Furthermore now, Asterisk is able + to differentiate between internally and externally created formats. + + ASTERISK-25537 #close + + Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52 + +2015-11-09 18:19 +0000 [be93036a4e] Corey Farrell + + * Remove ABI compatibility stub functions. + + ABI compatibility stubs existed for ast_app_separate_args and ast_verbose, + this is not needed in master. + + Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453 + +2015-11-10 07:51 +0000 [02a124eda5] Corey Farrell + + * Remove execute permission from dns_system_resolver.c + + Change-Id: I3185735db42064bab00d3e073aed703385a00bf4 + +2015-11-09 03:01 +0000 [cf79b62778] Alexander Traud + + * ast_format_cap_get_names: To display all formats, the buffer was increased. + + ASTERISK-25533 #close + + Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a + +2015-11-09 07:04 +0000 [e85f0c81af] Alexander Traud + + * ast_format_cap: Avoid format creation on module load, use cache instead. + + Since Asterisk 13, formats are immutable and cached. However while loading a + module like chan_sip, some formats were created instead using cached ones. + + ASTERISK-25535 #close + + Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b + +2015-11-06 07:54 +0000 [7dd8f89a50] Walter Doekes + + * func_callerid: Document that CALLERID(pres) is available. + + CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) + and CALLERID(name-pres). But for channel driver that don't make a + distinction between the two (e.g. SIP), it makes more sense to get/set + both at once. This change reveals the availability of CALLERID(pres), + CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and + REDIRECTING(from-pres). + + ASTERISK-25373 #close + + Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a +2015-11-06 07:52 +0000 [39daf9f066] Walter Doekes + + * docs: Fix a few typo's in app docs (more then, resourse). + + Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7 + +2015-11-06 14:19 +0000 [d82a4b098f] gtjoseph + + * dns: Use ntohl for ans->ttl in dns_parse_answer_ex + + dns_parse_answer_ex was not converting ans->ttl from network + by order to host byte order which was causing certain ttls + it to go negative. In turn this was causing answer edit checks + to fail. + + ASTERISK-25528 #close + Reported-by: Daniel Tryba + Tested-by: George Joseph + + Change-Id: I31505132d6321c46d2f39fd06c20ee808a864037 + +2015-11-06 07:36 +0000 [74e7333317] Walter Doekes + + * xmldoc: Improve xmldoc wrapping of 'core show ...' output. + + Previously, the wrapping did both lookahead and lookback, which, + together with color escape sequences, caused some lines to be wrapped + way earlier than other lines. This led to inconsistent output. + + This simplifies the wrapping code and makes it more sane: if maxcolumns + is hit, we simply jump back to the last space and wrap there. + + ASTERISK-25527 #close + + Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957 + +2015-11-06 06:57 +0000 [9d6e917349] Sean Bright (license #5060) + + * res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP. + + In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual + amount of channels is negotiated in-band. Therefore now, the Opus codec and its + attribute rtpmap are registered with two channels. + + ASTERISK-24779 #close + Reported by: PowerPBX + Tested by: Alexander Traud + patches: + asterisk-24779.patch submitted by Sean Bright (license #5060) + + Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b + +2015-11-03 16:19 +0000 [a2c2a8e1bb] Jonathan Rose + + * taskprocessor: Add high water mark warnings + + If a taskprocessor's queue grows large, this can indicate that there + may be a problem with tasks not leaving the processor or else that + the number of available task processors for a given type of task is + too low. This patch makes it so that if a taskprocessor's task queue + grows above 100 queued tasks that it will emit a warning message. + Warning messages are emitted only once per task processor. + + ASTERISK-25518 #close + Reported by: Jonathan Rose + + Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c + +2015-11-02 20:11 +0000 [cd5ae02812] Corey Farrell + + * Increase account code maximum length to 80. + + This increases the maximum length of account code's to match + extensions. This ensures it is always possible to set an + accountcode to ${EXTEN} without truncation. + + ASTERISK-23904 + Reported by: Ben Merrills + + Change-Id: If122602304ce03362722eb213a3111b32da5eeb9 + +2015-11-03 14:36 +0000 [379c041038] Tyler Cambron + + * StatsD: Add res_statsd compatibility + + Added a new api to res_statsd.c to allow it to receive a + character pointer for the value argument. This allows for a + '+' and a '-' to easily be sent with the value. + + ASTERISK-25419 + Reported By: Ashley Sanders + + Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611 + +2015-11-04 14:31 +0000 [9c293b5104] Matt Jordan + + * main/dial: Protect access to the format_cap structure of the requesting channel + + When a dial attempt is made that involves a requesting channel, we previously + were not: + a) Protecting access to the native format capabilities structure on the + requesting channel. That is inherently unsafe. + b) Reference bumping the lifetime of the format capabilities structure. + + In both cases, something else could sneak in, blow away the format + capabilities, and we'd be holding onto an invalid format_cap structure. When + the newly created channel attempts to construct its format capabilities, things + go poorly. + + This patch: + a) Ensures that we get a reference to the native format capabilities while + the requesting channel is locked + b) Holds a reference to the native format capabilities during the creation + of the new channel. + + ASTERISK-25522 #close + + Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f + +2015-10-30 22:57 +0000 [b0bf189908] Corey Farrell + + * Fix cli display of build options. + + A previous commit reduced the AST_BUILDOPTS compiler define to + only include options that affected ABI. This included some options + that were previously displayed by cli "core show settings". This + change corrects the CLI display while still restricting buildopts.h + to ABI effecting options only. + + ASTERISK-25434 #close + Reported by: Rusty Newton + + Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325 + +2015-11-03 10:58 +0000 [63e02b45c6] Matt Jordan + + * pjsip_configuration: On delete, remove the persistent version of an endpoint + + When an endpoint is deleted (such as through an API), the persistent endpoint + currently continues to lurk around. While this isn't harmful from a memory + consumption perspective - as all persistent endpoints are reclaimed on + shutdown - it does cause Stasis endpoint related operations to continue + to believe that the endpoint may or may not exist. + + This patch causes the persistent endpoint related to a PJSIP endpoint to be + destroyed if the PJSIP endpoint is deleted. + + Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb +2015-11-03 11:15 +0000 [d33a1682e3] Matt Jordan + + * res_pjsip/location: Destroy contact_status objects on contact deletion + + The contact_status Sorcery objects are currently not destroyed when a contact + is deleted. This causes the contact's last known RTT/status to be 'sticky' + when the contact itself may no longer exist. This patch causes the + contact_status objects associated with both dynamic and static contacts to + be destroyed if the AoR holding those contacts is also destroyed (or via + other paths where a contact may be deleted.) + + Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e + +2015-11-03 08:15 +0000 [e26a06c1da] Matt Jordan + + * main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field + + The JSON packing for the ContactStatusChange event forgot to include the + roundtrip_usec field. As a result, the field never showed up in any event, + even when the data was available. This patch corrects that error by properly + packing the JSON blob with the data. + + Change-Id: I8df80da659a44010afbd48f645967518ff5daa17 + +2015-11-02 20:24 +0000 [40574a2ea3] Corey Farrell + + * chan_sip: Allow websockets to be disabled. + + This patch adds a new setting "websockets_enabled" to sip.conf. + Setting this to false allows chan_sip to be used without causing + conflicts with res_pjsip_transport_websocket. + + ASTERISK-24106 #close + Reported by: Andrew Nagy + + Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7 + +2015-11-02 17:19 +0000 [f80a0ae49b] Mark Michelson + + * res_pjsip: Set threadpool max size default to 50. + + During a stress test of subscriptions, a huge blast of + subscription-related traffic resulted in the threadpool expanding to a + ridiculous number of threads. The balooning of threads resulted in an + increase of memory, which led to a crash due to being out of memory. + + An easy fix for the particular test was to limit the size of the + threadpool, thus reining in the amount of memory that would be used. It + was decided that there really is no downside to having a non-infinite + default value for the maximum size of the threadpool, so this change + introduces 50 threads as the maximum threadpool size for the SIP + threadpool. + + ASTERISK-25513 #close + Reported by John Bigelow + + Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be + +2015-10-29 15:25 +0000 [c5093b21ad] Tyler Cambron + + * StatsD: Send stuff to the StatsD server and test + + Added code to allow the StatsD dialplan application to + send data to the server specified in statsd.conf. + + ASTERISK-25419 + + Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922 + +2015-11-02 06:57 +0000 [014e3d426b] Matt Jordan + + * pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction + + When an AoR is created or destroyed dynamically, the scheduled OPTIONS + requests that qualify the contacts on the AoR are not necessarily started + or destroyed, particularly for persistent contacts created for that AoR. + This patch adds create/update/delete sorcery observers for an AoR, which + schedule/unschedule the qualifies as expected. + + Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d + +2015-10-30 13:22 +0000 [80cf4960ff] Matt Jordan + + * Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs + + This patch adds a rule for installing the Super Awesome Company based 'Basic + PBX' configuration files. As part of adding this rule, a bit of the content + that makes up installing the configuration files under the 'samples' target + was refactored into a make subroutine for usage by additional later config + make targets. + + Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404 +2015-10-29 08:28 +0000 [b522a5e30f] Joshua Colp + + * res_pjsip_pubsub: Fix assertion when UAS dialog creation fails. + + When compiled with assertions enabled one will occur when destroying + the subscription tree when UAS dialog creation fails. This is because + the code assumes that a dialog will always exist on a subscription + tree when in reality during this specific scenario it won't. + + This change makes it so a dialog is not removed from the subscription + tree if it is not present. + + ASTERISK-25505 #close + + Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee + +2015-10-08 11:50 +0000 [fdfd0fb488] Tyler Cambron + + * StatsD: Add user input validation to the application + + Added code to accept user input and validate it before + allowing it to be sent to the StatsD server. + + ASTERISK-25419 + Reported By: Ashley Sanders + + Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3 + +2015-10-26 11:42 +0000 [d343a25173] Alexander Traud + + * chan_sip: Do not send all codecs on INVITE. + + Since version 13, Asterisk sent all allowed codecs as callee, even when the + caller did not request/support them. In case of dynamic RTP payloads, this led + to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the + intersection between the requested and the supported codecs is send again. + + ASTERISK-24543 #close + + Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287 + +2015-10-19 07:11 +0000 [88f3dbaec9] Rodrigo Ramírez Norambuena + + * install_prereq: Update repositories before install on Debian systems + + When to install packages the indexed local is more old of the + version of software on the repository they have been upgraded by security + update then get the package will give 404 not found. + + The patch prevent by update local index to repository for aptitude before + install. + + ASTERISK-25495 #close + + Reporte by: Rodrigo Ramírez Norambuena + + Change-Id: I645959e553aac542805ced394cac2dca964051fa + +2015-10-24 13:08 +0000 [4328d320c2] gtjoseph + + * build: GCC 5.1.x catches some new const, array bounds and missing paren issues + + Fixed 1 issue in each of the affected files. + + ASTERISK-25494 #close + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77 + +2015-10-20 16:02 +0000 [a8aee0bbdb] gtjoseph + + * res_pjsip: Add "like" processing to pjsip list and show commands + + Add the ability to filter output from pjsip list and show commands + using the "like" predicate like chan_sip. + + For endpoints, aors, auths, registrations, identifyies and transports, + the modification was a simple change of an ast_sorcery_retrieve_by_fields + call to ast_sorcery_retrieve_by_regex. For channels and contacts a + little more work had to be done because neither of those objects are + true sorcery objects. That was just removing the non-matching object + from the final container. Of course, a little extra plumbing in the + common pjsip_cli code was needed to parse the "like" and pass the regex + to the get_container callbacks. + + Some of the get_container code in res_pjsip_endpoint_identifier was also + refactored for simplicity. + + ASTERISK-25477 #close + Reported by: Bryant Zimmerman + Tested by: George Joseph + + Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1 + +2015-10-21 12:22 +0000 [691c0e0b31] Kevin Harwell + + * res_pjsip_outbound_registration: registration stops due to fatal 4xx response + + During outbound registration it is possible to receive a fatal (any permanent/ + non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due + to a problem with the registrar itself. Upon receiving the failure response + Asterisk terminates outbound registration for the given endpoint. + + This patch adds an option, 'fatal_retry_interval', that when set continues + outbound registration at the given interval up to 'max_retries' upon receiving + a fatal response. + + ASTERISK-25485 #close + + Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2 + +2015-10-22 17:07 +0000 [5dd9e1938a] Mark Michelson + + * format_cap: Detect vector allocation failures. + + A crash was seen on a system that ran out of memory due to Asterisk not + checking for vector allocation failures in format_cap.c. With this + change, if either of the AST_VECTOR_INIT calls fail, we will return a + value indicating failure. + + Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8 + +2015-10-02 15:32 +0000 [7f9823ff57] Mark Michelson + + * res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog. + + A certain situation can result in our attempting to send a NOTIFY on a + destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but + that subscriber has dropped off the network. We end up retransmitting + that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY + transaction. When the pjsip evsub code is told that the transaction has + been terminated, it responds in kind by alerting us that the + subscription has been terminated, destroying the subscription, and then + removing its reference to the dialog, thus destroying the dialog. + + The problem is that when we get told that the subscription is being + terminated, we detect that we have not sent a terminating NOTIFY + request, so we queue up such a NOTIFY to be sent out. By the time that + queued NOTIFY gets sent, the dialog has been destroyed, so attempting to + send that NOTIFY can result in a crash. + + The fix being introduced here is actually a reintroduction of something + the pubsub code used to employ. We hold a reference to the dialog and + wait to decrement our reference to the dialog until our subscription + tree object is destroyed. This way, we can send messages on the dialog + even if the PJSIP evsub code wants to terminate earlier than we would + like. + + In doing this, some NULL checks for subscription tree dialogs have been + removed since NULL dialogs are no longer actually possible. + + Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5 + +2015-09-29 14:53 +0000 [e9e4bc9ece] Mark Michelson + + * res_pjsip_pubsub: Ensure dialog lock balance. + + When sending a NOTIFY, we lock the dialog and then unlock the dialog + when finished. A recent change made it so that the subscription tree's + dialog pointer will be set NULL when sending the final NOTIFY request + out. This means that when we attempt to unlock the dialog, we pass a + NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog + remains locked after we think we have unlocked it. When a response to + the NOTIFY arrives, the monitor thread attempts to lock the dialog, but + it cannot because we never released the dialog lock. This results in + Asterisk being unable to process incoming SIP traffic any longer. + + The fix in this patch is to use a local pointer to save off the pointer + value of the subscription tree's dialog when locking and unlocking the + dialog. This way, if the subscription tree's dialog pointer is NULLed + out, the local pointer will still have point to the proper place and the + dialog lock will be unlocked as we expect. + + Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a + +2015-09-28 16:36 +0000 [b96267f7a3] Mark Michelson + + * res_pjsip_pubsub: Prevent crashes on final NOTIFY. + + The SIP dialog is removed from the subscription tree when the final + NOTIFY is sent. However, after the final NOTIFY is sent, the persistence + update function still attempts to access the cseq from the dialog, + resulting in a crash. + + This fix removes the subscription persistence at the same time that the + dialog is removed from the subscription tree. This way, there is no + attempt to update persistence when the subscription is being destroyed. + + Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb + +2015-09-17 17:28 +0000 [386cd7b2b0] Mark Michelson + + * res_pjsip_pubsub: Remove serializer when sending final NOTIFY. + + There have been crashes seen where a taskprocessor's listener is NULL + unexpectedly. + + Looking at backtraces, the problem was specifically seen in PJSIP + serializers. + + Subscriptions make the mistake of removing a serializer from a dialog + during subscription tree destruction. Since subscription trees are + reference-counted, guaranteeing the circumstances behind the destruction + are not possible. This makes it so that the dialog serializer can be + removed while not holding the dialog lock. This makes it possible for + the distributor to get a pointer to the dialog serializer and have that + serializer get freed out from under it. + + The fix for this is to remove the serializer from a subscription dialog + when sending the final NOTIFY. This guarantees that the serializer is + removed with the dialog lock held. By doing this, we guarantee that if + the distributor gains access to the dialog's serializer, it will not be + possible for the serializer to get freed by another thread. + + Change-Id: I21f5dac33529f65cec45679bdace60670800ff66 + +2015-09-02 09:14 +0000 [0b63d011c9] Mark Michelson + + * res_pjsip_pubsub: Fix crash on destruction of empty subscription tree. + + If an old persistent subscription is recreated but then immediately + destroyed because it is out of date, the subscription tree will have no + leaf subscriptions on it. This was resulting in a crash when attempting + to destroy the subscription tree. + + A simple NULL check fixes this problem. + + Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac + +2015-09-01 15:47 +0000 [ac0194dad6] Mark Michelson + + * res_pjsip_pubsub: Solidify lifetime and ownership of objects. + + There have been crashes and general instability seen in the pubsub code, + so this patch introduces three changes to increase the stability. + + First, the ownership model for subscriptions has been modified. Due to + RLS, subscriptions are stored in memory as a tree structure. Prior to my + patch, the PJSIP subscription was the owner of the subscription tree. + When the PJSIP subscription told us that it was terminating, we started + destroying the subscription tree along with all of the individual leaf + subscriptions that belong to the tree. The problem with this model is + that the two actors in play here, the PJSIP subscription and the + individual leaf subscriptions, need to have joint ownership of the + subscription tree. So now, the PJSIP subscription and the individual + leaf subscriptions each have a reference to the subscription tree. This + way, we will not actually free memory until no players are left that + care. The PJSIP subscription is a bigger stakeholder, in that if the + PJSIP subscription's reference to the subscription tree is removed, the + subscription tree instructs the leaf subscriptions to shut down and drop + their references to the subscription tree when possible. The individual + leaf subscriptions, upon being told to shut down, can drop their stasis + subscriptions or whatever they use to learn of new state, and then drop + their reference to the subscription tree once they are ready to die. + + Second, the lifetime of a PJSIP subscription's reference to our + subscription tree has been altered. As I learned from doing a deep dive, + the PJSIP evsub code can tell Asterisk multiple times that the + subscription has been terminated, and not all of these times + are especially helpful. I have altered the message flow that we use for + SIP subscriptions such that we will always drop the PJSIP subscription's + reference to the subscription tree when we send the NOTIFY that + terminates a SIP subscription. This also means that we will now queue + NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so + that we can have predictable state changes from the PJSIP evsub code. + + Third, the synchronization of operations has been improved. PJSIP can + call into our code from a serializer thread (e.g. upon receiving an + incoming request) or from the monitor thread (e.g. when a subscription + times out). Because of this, there is the possibility of competing + threads stepping on each other. PJSIP attempts to do some + synchronization on its own by always keeping the dialog lock held when + it calls into us. However, since we end up pushing tasks into the + serializer, the result was that serialized operations were not grabbing + the dialog lock and could, as a result, step on something that was being + attempted by a different thread. Now we ensure that serialized + operations grab the dialog lock, then check for extenuating + circumstances, then proceed with their operation if they can. + + Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5 + +2015-10-19 15:28 +0000 [1ce62b2545] Richard Mudgett + + * strings.c: Fix __ast_str_helper() to always return a terminated string. + + Users of functions which call __ast_str_helper() such as the ones listed + below are likely to not check the return value for failure so ensuring + that the string is always nil terminated is a good safety measure. + + ast_str_set_va() + ast_str_append_va() + ast_str_set() + ast_str_append() + + Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07 + +2015-10-19 15:27 +0000 [a04d946eaa] Richard Mudgett + + * Add missing failure checks to ast_str_set_va() callers. + + Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f + +2015-10-21 11:44 +0000 [64c172deba] Joshua Colp + + * res_pjsip: Move URI validation to use time. + + In a realtime based system with a limited number of threadpool threads + it is possible for a deadlock to occur. This happens when permanent + endpoint state is updated, which will cause database queries to be done. + These queries may result in URI validation being done which is done + synchronously using a PJSIP thread. If all PJSIP threads are in use + processing traffic they themselves may be blocked waiting to get the + permanent endpoint container lock when identifying an endpoint. + + This change moves URI validation to occur at use time instead of + configuration time. While this comes at a cost of not seeing a problem + until you use it it does solve the underlying deadlock problem. + + ASTERISK-25486 #close + + Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a + +2015-10-21 08:08 +0000 [f9cbac7321] Alexander Traud + + * format: Update the maximum packetization time for iLBC 30. + + In September 2006, the maximum packetization time (ptime) were set to such a + low value, packetization was disabled for many codecs actually. This was fixed + for many codecs but not for iLBC 30. This enables packetization for iLBC which + can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf. + + ASTERISK-7803 + + Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12 +2015-10-21 09:51 +0000 [f3b2b3d1b3] Alexander Traud + + * chan_sip: Fix autoframing=yes. + + With Asterisk 13, the structures ast_format and ast_codec changed. Because of + that, the paketization timing (framing) of the RTP channel moved away from the + formats/codecs. In the course of that change, the ptime of the callee was not + honored anymore, when the optional autoframing was enabled. + + ASTERISK-25484 #close + + Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4 + +2015-10-20 22:24 +0000 [b425850f8b] Matt Jordan + + * rest-api-templates: Wikify error code response reasons + + Error response code descriptions may contain wiki markup that need to be + escaped. Without this patch, Confluence will reject the document being sent + and the responsible script will raise an exception. + + Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5 + +2015-10-20 12:06 +0000 [7be6194d6f] Matt Jordan + + * funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function + + When ab803ec342 was committed, it accidentally forgot to actually *add* the + HOLD_INTERCEPT function. This highlights two interesting points: + * Gerrit forces you to put the patch as it is going to into the repo up for + review, which Review Board did not. Yay Gerrit. + * No one apparently bothered to use this feature, or else they don't know about + it. I'm going to go with the latter explanation. + + ASTERISK-24922 + + Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396 + +2015-10-19 14:14 +0000 [77780790e0] Jonh Wendell + + * main/cdr: Allow modules to modify CDR fields before dispatching them + + This patch adds the functions + + ast_cdr_modifier_register() + ast_cdr_modifier_unregister() + + That work much like ast_cdr_register() and ast_cdr_unregister(). + + Modules registered will be given a chance to modify (or to do whatever + they want) CDR fields just before they are passed to registered engines. + + Thus, for instance, if a module change the "userfield" field of a CDR, + the modified value will be passed to every registered CDR backend for + logging. + + ASTERISK-25479 #close + + Change-Id: If11d8fd19ef89b1a66ecacf1201e10fcf86ccd56 +2015-10-19 19:59 +0000 [b9bd249a85] Matt Jordan + + * contrib/scripts/autosupport: Update for Asterisk 13 + + This patch adds some minor tweaks for autosupport to update it for Asterisk 13. + This includes: + * Finally removing most references to Zaptel + * Adding support for some additional 'core' commands, and fixing nomenclature + that generally hasn't been used for some time + * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels + + Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 + (cherry picked from commit 9fc9777fa34753fb38991d42d8dbed516e907ca2) + +2015-10-18 18:22 +0000 [92fa8d1e0e] Rodrigo Ramírez Norambuena + + * app_queue: Added reason pause of member + + In app_queue added value Paused Reason on QueueMemberStatus when a member + on queue is paused and the reason was set. + + ASTERISK-25480 #close + Reporte by: Rodrigo Ramírez Norambuena + + Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e + +2015-10-16 22:01 +0000 [b19860c03a] Corey Farrell + + * res_ari_events: Fix memory leak in mustache template. + + ASTERISK-25308 fixed a memory leak in res_ari_events.c, but + this file is regenerated by a template and the template was + not fixed. + + Change-Id: Ied4c6deae89d21f87f9cf99676b1d055aa83b38b + +2015-10-14 14:15 +0000 [d799bcf361] mdu113 + + * res_config_pgsql.c: Fix deadlock loading realtime configuration. + + On v13, loading several thousand PJSIP endpoints on Asterisk start causes + a deadlock most of the time. + + Thanks to mdu113 for discovering that there was a call to pgsql_exec() not + protected by the pgsql_lock reentrancy lock. + + {quote} + I believe a code path exists that attempts to use pgsql connection without + locking pgsql_lock. I believe what happens during that deadlock that I + see is two concurrent threads are both attempting to send query to pgsql, + one of the thread is using a code path without locking pgsql_lock. If + they managed to send queries at the same time, it seems postgres ignores + one of the queries and replies only to the one of them. If it happens so + that the thread holding the lock didn't receive the reply it will wait for + it (and hold the lock) forever (or at least for very long time), thus + completely blocking all access to db. + {quote} + + * Added missing reentrancy locking around pgsql_exec() in find_table(). + + * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion + between the psql_tables list lock and the pgsql_lock. + + ASTERISK-25455 #close + Reported by: mdu113 + Patches: + res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113 + + Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2 + +2015-10-13 14:13 +0000 [13229037d1] Olle Johansson (License 5267) + + * channels/chan_sip: Set cause code to 44 on RTP timeout + + To quote Olle: + + "When issuing a hangup due to RTP timeouts the cause code is not set. I have + selected 44 based on Cisco's implementation..." + + ASTERISK-25135 #close + Reported by: Olle Johansson + patches: + rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) + + Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc + +2015-10-12 11:21 +0000 [984f100dab] Richard Mudgett + + * config.c: Fix off-nominal memory leak. + + Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 + +2015-10-12 11:20 +0000 [9951255775] Richard Mudgett + + * config.c: Fix potential memory corruption after [section](+). + + The memory corruption could happen if the [section](+) is the last section + in the file with trailing comments. In this case process_text_line() has + left *last_cat is set to newcat and newcat is destroyed. + + Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93 + +2015-10-12 11:21 +0000 [c1ed11ee31] Richard Mudgett + + * config.c: Fix #include after [section](+). + + An #include right after a [section](+) would associate any variable + assignments before a new section in the #include with the wrong section. + + * Fix section association by setting the current section to the appended + section. + + * Fix '+' and '!' section flag interaction corner case depending upon + which flag came first. If the '!' came first then it would be ignored. + If the '!' came after then it would affect the appended section. The '!' + will now no longer be ignored. + + ASTERISK-25461 #close + Reported by: Sean Pimental + + Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3 + +2015-10-10 15:20 +0000 [a12eb89ea4] Ivan Poddubny + + * Build: Add menuselect options for using compiler sanitizers + + This patch adds menuselect options for building Asterisk with + various sanitizers provided by gcc and clang. + + When one of *SANITIZER flags is set in menuselect, the appropriate + option is added to CFLAGS ad LDFLAGS for the build. + + Information on sanitizers in the project wiki: + https://github.com/google/sanitizers/wiki + + GCC Manual: + https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html + + Clang Compiler User's Manual: + http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation + + ASTERISK-24718 #close + Reported by: Badalian Vyacheslav + + Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0 + +2015-10-08 16:43 +0000 [ca030845ff] Richard Mudgett + + * configure: Fix check for libunbound to require v1.5.0 as minimum. + + Versions of libunbound before v1.4.21 do not compile with Asterisk. + However, since v1.4.21 has a configure script bug that fails to detect the + ldns library (which is fixed in v1.4.22) and v1.4.22 is not an easily + detectable version we will require v1.5.0 as a minimum version of the + library to work with Asterisk. + + ASTERISK-25108 #close + Reported by: Richard Mudgett + + Change-Id: Ieb228bfb01467573fc121c7356a9dde27128894d + +2015-10-08 11:50 +0000 [2fe9f09705] Tyler Cambron + + * StatsD: Write skeleton Asterisk application + + Wrote the skeleton framework for the Asterisk StatsD dialplan + application. This includes a load function, unload function, a + callback for execution, and XML documentation. + + ASTERISK-25419 + Reported By: Ashley Sanders + + Change-Id: I9597730e134c6e82c8a55ef4d5334b62dd473363 + +2015-10-06 18:01 +0000 [34d7fa6c4a] Richard Mudgett + + * res_pjsip: Fix deadlock when sending out-of-dialog requests. + + The struct send_request_wrapper has a pjsip lock associated with it that + is created non-recursive. There is a code path for the struct + send_request_wrapper lock that will attempt to lock it recursively. The + reporter's deadlock showed that the thread calling endpt_send_request() + deadlocked itself right after the wrapper object got created. + + Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited + MWI NOTIFY messages can hit this deadlock. + + * Replaced the struct send_request_wrapper pjsip lock with the mutex lock + that can come with an ao2 object since all of Asterisk's mutexes are + recursive. Benefits include removal of code maintaining the pjsip + non-recursive lock since ao2 objects already know how to maintain their + own lock and the lock will show up in the CLI "core show locks" output. + + ASTERISK-25435 #close + Reported by: Dmitriy Serov + + Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d + +2015-10-06 11:05 +0000 [cc131832aa] Stefan Engström + + * res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending + + In ast_rtp_read, the value of the variable 'mark' which we try to assign to a + frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate + it to 0 or 1. + + ASTERISK-25451 #close + Change-Id: I53bdf5c026041730184a6a809009c028549ce626 + +2015-10-07 01:24 +0000 [c944263e36] Ivan Poddubny + + * func_presencestate: Return "not_set" when no data is set in AstDB + + Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not + exist, i.e. when a new CustomPresence is added in the dialplan. + + ASTERISK-25400 #close + Reported by: Andrew Nagy + + Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a + +2015-10-06 20:43 +0000 [4bf395e81e] Matt Jordan + + * res/res_rtp_asterisk: Fix assignment after ao2 decrement + + When we decide we will no longer schedule an RTCP write, we remove the + reference to the RTP instance, then assign -1 to the stored scheduler ID + in case something else comes along and wants to see if anything is scheduled. + + That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to + fix the regression introduced by 3cf0f29310, this improper assignment on a + potentially destroyed object started getting tripped on the build agents. + + Frankly, this should have been crashing a lot more often earlier. I can only + assume that the timing was changed just enough by both changes to start + actually hitting this problem. + + As it is, simply moving the assignment prior to the ao2 deference is sufficient + to keep the RTP instance from being referenced when it is very, truly, + aboslutely dead. + + (Note that it is still good practice to assign -1 to the scheduler ID when we + know we won't be scheduling it again, as the ao2 deref *may* not always destroy + the ao2 object.) + + ASTERISK-25449 + + Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7 + +2015-10-06 12:40 +0000 [3ec9cf7d6a] Florian Sauerteig + + * chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers. + + If a Via header containes an IPv6 address and a port number is ommitted, + as it is the standard port, we now leave the port empty and to not set it + to the value after the first colon of the IPv6 address. + + ASTERISK-25443 #close + + Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70 + +2015-10-05 16:53 +0000 [8fe9350b68] Richard Mudgett + + * chan_pjsip: Fix crash on reINVITE before initial INVITE completes. + + Apparently some endpoints attempt to send a reINVITE before completing the + initial INVITE transaction. In this case PJSIP responds appropriately to + the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip + is using the initial INVITE transaction state to determine if an INVITE is + the initial INVITE or a reINVITE. Since the initial INVITE transaction + has not been confirmed yet chan_pjsip thinks the reINVITE is an initial + INVITE and starts another PBX thread on the channel. The extra PBX thread + ensures that hilarity ensues. + + * Fix checks for a reINVITE on incoming requests to look for the presence + of a to-tag instead of the initial INVITE transaction state. + + * Made caller_id_incoming_request() determine what to do if there is a + channel on the session or not. After a channel is created it is too late + to just store the new party id on the session because the session's party + id has already been copied to the channel's caller id. + + ASTERISK-25404 #close + Reported by: Chet Stevens + + Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be + +2015-10-05 21:34 +0000 [8cb614fe20] Matt Jordan + + * Fix improper usage of scheduler exposed by 5c713fdf18f + + When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of + '0' returned. While this was valid per the documentation for the API, it was + apparently never returned previously. As a result, several users of the + scheduler API viewed the result as being invalid, causing them to reschedule + already scheduled items or otherwise fail in interesting ways. + + This patch corrects the users such that they view '0' as valid, and a returned + ID of -1 as being invalid. + + Note that the failing HEP RTCP tests now pass with this patch. These tests + failed due to a duplicate scheduling of the RTCP transmissions. + + ASTERISK-25449 #close + + Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39 +2015-08-26 16:58 +0000 [c6b0d60264] Debian Amtelco + + * chan_pjsip: Add Referred-By header to the PJSIP REFER packet. + + Some systems require the REFER packet to include a Referred-By header. + If the channel variable SIPREFERREDBYHDR is set, it passes that value as the + Referred-By header value. Otherwise, it adds the current dialog’s local info. + + Reported by: Dan Cropp + Tested by: Dan Cropp + + Change-Id: I3d17912ce548667edf53cb549e88a25475eda245 + +2015-10-03 06:27 +0000 [89dec7675d] Ivan Poddubny + + * manager: Fix GetConfigJSON returning invalid JSON + + When GetConfigJSON was introduced back in 1.6, it returned each + section as an array of strings: ["key=value", "key2=value2"]. + Afterwards, it was changed a few times and became + ["key": "value", "key2": "value2"], which is not a correct JSON. + This patch fixes that by constructing a JSON object {} instead of + an array []. + + Also, the keys "istemplate" and "tempates" that are used to + indicate templates and their inherited categories are now wrapped in + quotes. + + ASTERISK-25391 #close + Reported by: Bojan Nemčić + + Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8 + +2015-09-30 17:28 +0000 [1b80dbeb60] Richard Mudgett + + * res_sorcery_memory_cache.c: Fix deadlock with scheduler. + + A deadlock can happen when a sorcery object is being expired from the + memory cache when at the same time another object is being placed into the + memory cache. There are a couple other variations on this theme that + could cause the deadlock. Basically if an object is being expired from + the sorcery memory cache at the same time as another thread tries to + update the next object expiration timer the deadlock can happen. + + * Add a deadlock avoidance loop in expire_objects_from_cache() to check if + someone is trying to remove the scheduler callback from the scheduler. + + ASTERISK-25441 #close + + Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc + +2015-10-01 14:30 +0000 [9c1ca287a4] Richard Mudgett + + * res_sorcery_memory_cache.c: Replace inline code with function. + + Make sorcery_memory_cache_close() call remove_all_from_cache() instead of + partially inlining it. + + ASTERISK-25441 + + Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c + +2015-10-01 14:27 +0000 [6554a3b25e] Richard Mudgett + + * res_sorcery_memory_cache.c: Shutdown in a less crash potential order. + + Basically you should shutdown in the opposite order of how you setup since + later setup pieces likely depend on earlier setup pieces. e.g., + Registering your external API with the rest of the system should be the + last thing setup and the first thing unregistered during shutdown. + + Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e + +2015-09-30 17:27 +0000 [359394cc29] Richard Mudgett + + * res_sorcery_memory_cache.c: Misc tweaks. + + Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160 + +2015-09-30 17:27 +0000 [7942d1c2ff] Richard Mudgett + + * res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK. + + Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c + +2015-09-30 13:42 +0000 [9f229d6a49] Joshua Colp + + * res_rtp_asterisk: Move "Set role" warning to be debug. + + In practice the set_role API callback can be invoked even + when no ICE is present on an RTP instance. This can occur + if ICE has not been enabled on it. + + ASTERISK-25438 #close + + Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69 + +2015-09-28 15:31 +0000 [9bc7386b7c] Richard Mudgett + + * sched.c: Add warning about negative time interval request. + + Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc + +2015-09-25 18:37 +0000 [12feec0bf7] Richard Mudgett + + * res/ari/config.c: Fix user sort compare function. + + Made use the ao2 sort compare template function and OBJ_SEARCH_xxx + identifiers. + + Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c + +2015-09-25 17:26 +0000 [3f4fa245e5] Richard Mudgett + + * res/ari/config.c: Optimize conf_alloc() object init. + + * Now conf_alloc() has more off nominal error checking. + + * Eliminated RAII_VAR() use in conf_alloc(). + + * Eliminated a dubius shortcut when destroying cfg->general in + conf_destructor() that would cause a crash if cfg->general failed to get + allocated. + + * Add some ACO registration section comments. + + Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a + +2015-09-25 16:48 +0000 [aa00df62ee] Richard Mudgett + + * res/ari/config.c: Fix conf_alloc() object init. + + Need to finish initializing the string fields in the ao2 object before + putting any default strings into them. + + ASTERISK-25383 #close + Reported by: yaron nahum + + Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84 + +2015-09-21 07:26 +0000 [2d7a4a3357] Matt Jordan + + * main/logger: Add log formatters and JSON structured logs + + When Asterisk is part of a larger distributed system, log files are often + gathered using tools (such as logstash) that prefer to consume information + and have it rendered using other tools (such as Kibana) that prefer a + structured format, e.g., JSON. This patch adds support for JSON formatted + logs by adding support for an optional log format specifier in Asterisk's + logging subsystem. By adding a format specifier of '[json]': + + full => [json]debug,verbose,notice,warning,error + + Log messages will be output to the 'full' channel in the following + format: + + { + "hostname": Hostname or name specified in asterisk.conf + "timestamp": Date/Time + "identifiers": { + "lwp": Thread ID, + "callid": Call Identifier + } + "logmsg": { + "location": { + "filename": Name of the file that generated the log statement + "function": Function that generated the log statement + "line": Line number that called the logging function + } + "level": Log level, e.g., DEBUG, VERBOSE, etc. + "message": Actual text of the log message + } + } + + ASTERISK-25425 #close + + Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238 + +2015-09-27 20:45 +0000 [9402f80726] Matt Jordan + + * res/res_stasis: Fix accidental subscription to 'all' bridge topic + + When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a + NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to + all bridges. Unfortunately, the res_stasis control loop did not check that + a bridge changing on a channel's control object was actually also non-NULL. + As a result, app_subscribe_bridge will be called with a NULL bridge when a + channel leaves a bridge. This causes a new subscription to be made to the + bridge. If an application has also subscribed to the bridge, the application + will now have two subscriptions: + (1) The explicit one created by the app + (2) The implicit one accidentally created by the control structure + + As a result, the 'BridgeDestroyed' event can be sent multiple times. This + patch corrects the control loop such that it only subscribes an application + to a new bridge if the bridge pointer is non-NULL. + + ASTERISK-24870 + + Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f + +2015-09-04 13:51 +0000 [d6472d96b3] Scott Griepentrog + + * Scripts: check file versions of Asterisk and dependencies + + To help in diagnosing mismatched modules and libraries, this + script scans for version, repository, and source information + and reports what is found. + + ASTERISK-25376 #close + Reported by: Ashley Sanders + + Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6 + +2015-09-24 14:56 +0000 [7c7a7ddd27] Richard Mudgett + + * app_queue.c: Force COLP update if outgoing channel name changed. + + * When a call is answered and the outgoing channel name has changed then + force a connected line update because the channel is no longer the same. + The channel was masqueraded into by another channel. This is usually + because of a call pickup. + + Note: Forwarded calls are handled in a controlled manner so the original + channel name is replaced with the forwarded channel. + + ASTERISK-25423 #close + Reported by: John Hardin + + Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172 + +2015-09-24 14:20 +0000 [145608bd81] Richard Mudgett + + * app_queue.c: Factor out a connected line update routine. + + Replace inlined code with update_connected_line_from_peer(). + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3 + +2015-09-24 13:27 +0000 [1d394774b2] Richard Mudgett + + * app_dial.c: Make 'A' option pass COLP updates. + + While the 'A' option is playing the announcement file allow the caller and + peer to exchange COLP update frames. + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9 + +2015-09-24 12:59 +0000 [680b76eb25] Richard Mudgett + + * app_dial.c: Force COLP update if outgoing channel name changed. + + * When a call is answered and the outgoing channel name has changed then + force a connected line update because the channel is no longer the same. + The channel was masqueraded into by another channel. This is usually + because of a call pickup. + + Note: Forwarded calls are handled in a controlled manner so the original + channel name is replaced with the forwarded channel. + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c + +2015-09-24 12:37 +0000 [fdf0bcb04a] Richard Mudgett + + * app_dial.c: Factor out a connected line update routine. + + Replace inlined code with update_connected_line_from_peer(). + + ASTERISK-25423 + Reported by: John Hardin + + Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091 + +2015-09-23 17:41 +0000 [c285879845] Richard Mudgett + + * app_dial.c: Remove some no-op code. + + Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54 + +2015-09-23 14:02 +0000 [3eefa07a39] Mark Michelson + + * logger: Prevent duplicate dynamic channels from being added. + + There was a problem observed where the "logger add channel" CLI command + would allow for a channel with the same name to be added multiple times. + This would result in each message being written out to the same file + multiple times. + + The problem was due to the difference in how logger channel filenames + are stored versus the format they are allowed to be presented when they + are added. For instance, if adding the logger channel "foo" through the + CLI, the result would be a logger channel with the file name + /var/log/asterisk/foo being stored. So when trying to add another "foo" + channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily + add the duplicate channel. + + The fix presented here is to introduce two new methods in the logger + code: + * make_filename(): given a logger channel name, this creates the + filename for that logger channel. + * find_logchannel(): given a logger channel name, this calls + make_filename() and then traverses the list of logchannels in order + to find a match. + + This change has made use of make_filename() and find_logchannel() + throughout to more consistently behave. + + ASTERISK-25305 #close + Reported by Mark Michelson + + Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36 + +2015-09-24 14:49 +0000 [f42084be09] Mark Michelson + + * Do not swallow frames on channels leaving bridges. + + When leaving a bridge, indications on a channel could be swallowed by + the internal indication logic because it appears that the channel is on + its way to be hung up anyway. One such situation where this is + detrimental is when channels on hold are redirected out of a bridge. The + AST_CONTROL_UNHOLD indication from the bridging code is swallowed, + leaving the channel in question to still appear to be on hold. + + The fix here is to modify the logic inside ast_indicate_data() to not + drop the indication if the channel is simply leaving a bridge. This way, + channels on hold redirected out of a bridge revert to their expected "in + use" state after the redirection. + + ASTERISK-25418 #close + Reported by Mark Michelson + + Change-Id: If6115204dfa0551c050974ee138fabd15f978949 + +2015-09-22 17:08 +0000 [06f4f80a63] Richard Mudgett + + * app_page.c: Fix crash when forwarding with a predial handler. + + Page uses the async method of dialing with the dial API. When a call gets + forwarded there is no calling channel available. If the predial handler + was set then the calling channel could not be put into auto-service + for the forwarded call because it doesn't exist. A crash is the result. + + * Moved the callee predial parameter string processing to before the + string is passed to the dial API rather than having the dial API do it. + There are a few benefits do doing this. The first is the predial + parameter string processing doesn't need to be done for each channel + called by the dial API. The second is in async mode and the forwarded + channel is to have the predial handler executed on it then the + non-existent calling channel does not need to be present to process the + predial parameter string. + + * Don't start auto-service on a non-existent calling channel to execute + the predial handler when the dial API is in async mode and forwarding a + call. + + ASTERISK-25384 #close + Reported by: Chet Stevens + + Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981 + +2015-09-04 12:25 +0000 [b99a705262] Matt Jordan + + * ARI: Add the ability to subscribe to all events + + This patch adds the ability to subscribe to all events. There are two possible + ways to accomplish this: + (1) On initial WebSocket connection. This patch adds a new query parameter, + 'subscribeAll'. If present and True, Asterisk will subscribe the + applications to all ARI events. + (2) Via the applications resource. When subscribing in this manner, an ARI + client should merely specify a blank resource name, i.e., 'channels:' + instead of 'channels:12354'. This will subscribe the application to all + resources of the 'channels' type. + + ASTERISK-24870 #close + + Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6 + +2015-09-21 18:06 +0000 [c74101509d] Kevin Harwell + + * app_record: RECORDED_FILE variable not being populated + + The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. + This patch makes it so the variable is always set to the filename. + + ASTERISK-25410 #close + + Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653 + +2015-09-21 08:16 +0000 [a29cf45c76] Elazar Broad + + * core/logging: Fix logging to more than one syslog channel + + Currently, Asterisk will log to the last configured syslog + channel in logger.conf. This is due to the fact that the + final call to openlog() supersedes all of the previous calls. + This commit removes the call to openlog() and passes the + facility to ast_log_vsyslog(), along with utilizing the + LOG_MAKEPRI macro to ensure that the message is routed to + the correct facility and with the correct priority. + + ASTERISK-25407 #close + Reported by: Elazar Broad + Tested by: Elazar Broad + + Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2 +2015-09-04 12:24 +0000 [47813cc51c] Matt Jordan + + * res/res_stasis_device_state: Allow for subscribing to 'all' device state + + This patch adds support for subscribing to all device state changes. This is + done either by subscribing to an empty device, e.g., 'eventSource=deviceState:', + or by the WebSocket connection specifying that it wants all state in the + system. + + ASTERISK-24870 + + Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32 + +2015-09-03 21:19 +0000 [5206aa9d30] Matt Jordan + + * ARI: Add events for Contact and Peer Status changes + + This patch adds support for receiving events regarding Peer status changes + and Contact status changes. This is particularly useful in scenarios where + we are subscribed to all endpoints and channels, where we often want to know + more about the state of channel technology specific items than a single + endpoint's state. + + ASTERISK-24870 + + Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9 + +2015-09-19 12:49 +0000 [9200ad03a3] Alexander Traud + + * astfd: Adds a timestamp for each entry. + + Now with menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", a timestamp is + shown with each file descriptor. This helps to debug leaked UDP/TCP ports on + long-lived servers, for example. + + ASTERISK-25405 #close + + Change-Id: I968339e5155a512eba1032a5263f1ec8b5e1f80b + +2015-09-16 08:22 +0000 [42a897c4c3] Joshua Colp + + * pbx: Update device and presence state when changing a hint extension. + + When changing a hint extension without removing the hint first the + device state and presence state is not updated. This causes the state + of the hint to be that of the previous extension and not the current + one. This state is kept until a state change occurs as a result of + something (presence state change, device state change). + + This change updates the hint with the current device and presence + state of the new extension when it is changed. Any state callbacks + which may have been added before the hint extension is changed are + also informed of the new device and presence state if either have + changed. + + ASTERISK-25394 #close + + Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f + +2015-09-17 16:34 +0000 [d9723d242a] Scott Griepentrog + + * CHAOS: avoid crash if string create fails + + Validate string buffer allocation before using them. + + ASTERISK-25323 + + Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0 + +2015-09-11 01:52 +0000 [99aa7cb26e] Rodrigo Ramírez Norambuena + + * dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF + + Remove repeated code on macro of assigned buffer to SQL vars + + Change-Id: Icb19ad013124498e172ea1d0b29ccd0ed17deef0 + +2015-09-17 04:52 +0000 [e4df271a3e] Walter Doekes + + * chan_sip: Fix From header truncation for extremely long CALLERID(name). + + The CALLERID(num) and CALLERID(name) and other info are placed into the + `char from[256]` in initreqprep. If the name was too long, the addr-spec + and params wouldn't fit. + + Code is moved around so the addr-spec with params is placed there first, + and then fitting in as much of the display-name as possible. + + ASTERISK-25396 #close + + Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260 + +2015-09-17 16:59 +0000 [e1927915bc] Richard Mudgett + + * CHAOS: res_pjsip_diversion avoid crash if allocation fails + + Validate ast_malloc buffer returned before using it in + set_redirecting_value(). + + ASTERISK-25323 + + Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253 + +2015-09-17 16:47 +0000 [729a4325da] Kevin Harwell + + * app_queue: AgentComplete event has wrong reason + + When a queued caller transfers an agent to another extension sometimes the + raised AgentComplete event has a reason of "caller" and sometimes "transfer". + Since a transfer has taken place this should always be transfer. This occurs + because sometimes the stasis hangup event arrives before the transfer event + thus writing a different reason out. + + With this patch, when a hangup event is received during a transfer it will + check to see if the channel that is hanging up is part of a transfer. If so + it will return and let the subsequently received transfer event handler take + care of the cleanup. + + ASTERISK-25399 #close + + Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d + +2015-09-17 13:09 +0000 [87f04d5acf] Scott Griepentrog + + * PJSIP: avoid crash when getting rtp peer + + Although unlikely, if the tech private is returned as + a NULL, chan_pjsip_get_rtp_peer() would crash. + + ASTERISK-25323 + + Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a + +2015-09-17 11:31 +0000 [63ede41227] Kevin Harwell + + * app_queue: Crash when transferring + + During some transfer scenarios involving queues Asterisk would sometimes + crash when trying to obtain a channel snapshot (could happen on caller or + member channels). This occurred because the underlying channel had already + disappeared when trying to obtain the latest snapshot. + + This patch adds a reference to both the member and caller channels that + extends to the lifetime of the queue'd call, thus making sure the channels + will always exist when retrieving the latest snapshots. + + ASTERISK-25185 #close + Reported by: Etienne Lessard + + Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128 + +2015-09-16 17:36 +0000 [e47396721f] Mark Michelson + + * res_pjsip_pubsub: Eliminate race during initial NOTIFY. + + There is a slim chance of a race condition occurring where two threads + can both attempt to manipulate the same area. + + Thread A can be handling an incoming initial SUBSCRIBE request. Thread A + lets the specific subscription handler know that the subscription has + been established. + + At this point, Thread B may detect a state change on the subscribed + resource and queue up a notification task on Thread C, the subscription + serializer thread. + + Now Thread A attempts to generate the initial NOTIFY request to send to + the subscriber at the same time that Thread C attempts to generate a + state change NOTIFY request to send to the subscriber. + + The result is that Threads A and C can step on the same memory area, + resulting in a crash. The crash has been observed as happening when + attempting to allocate more space to hold the body for the NOTIFY. + + The solution presented here is to queue the subscription establishment + and initial NOTIFY generation onto the subscription serializer thread + (Thread C in the above scenario). This way, there is no way that a state + change notification can occur before the initial NOTIFY is sent, and if + there is a quick succession of NOTIFYs, we can guarantee that the two + NOTIFY requests will be sent in succession. + + Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815 + +2015-08-28 15:42 +0000 [077adf48b8] Alexander Traud + + * translate: Fix transcoding while different in frame size. + + When Asterisk translates between codecs, each with a different frame size (for + example between iLBC 30 and Speex-WB), too large frames were created by + ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame + length, creating several frames when necessary. Affects all transcoding modules + which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex. + + ASTERISK-25353 #close + + Change-Id: I2e229569d73191d66a4e43fef35432db24000212 + +2015-09-10 17:19 +0000 [0a74c80300] Mark Michelson + + * scheduler: Use queue for allocating sched IDs. + + It has been observed that on long-running busy systems, a scheduler + context can eventually hit INT_MAX for its assigned IDs and end up + overflowing into a very low negative number. When this occurs, this can + result in odd behaviors, because a negative return is interpreted by + callers as being a failure. However, the item actually was successfully + scheduled. The result may be that a freed item remains in the scheduler, + resulting in a crash at some point in the future. + + The scheduler can overflow because every time that an item is added to + the scheduler, a counter is bumped and that counter's current value is + assigned as the new item's ID. + + This patch introduces a new method for assigning scheduler IDs. Instead + of assigning from a counter, a queue of available IDs is maintained. + When assigning a new ID, an ID is pulled from the queue. When a + scheduler item is released, its ID is pushed back onto the queue. This + way, IDs may be reused when they become available, and the growth of ID + numbers is directly related to concurrent activity within a scheduler + context rather than the uptime of the system. + + Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2 + +2015-09-03 21:15 +0000 [45cf79665c] Matt Jordan + + * main/config_options: Check for existance of internal object before derefing + + Asterisk can load and register an object type while still having an invalid + sorcery mapping. This can cause an issue when a creation call is invoked. + For example, mis-configuring PJSIP's endpoint identifier by IP address mapping + in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a + subsequent ARI invocation to create the object will cause a crash, as the + internal type may not be registered as sorcery expects. + + Merely checking for a NULL pointer here solves the issue. + + Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac + +2015-08-21 21:50 +0000 [34aa96bef4] Rodrigo Ramírez Norambuena + + * chan_sip.c: Validation on module reload + + Change validation on reload module because now used the cli function for + reload. The sip_reload() function never fail and ever return NULL for this + reason on reload() now use the call the sip_reload() and return + AST_MODULE_LOAD_SUCCESS. + + This problem is dectected on reload by PUT method on ARI, getting always + 404 http code when the module is reloaded. + + ASTERISK-25325 #close + Reporte by: Rodrigo Ramírez Norambuena + + Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb + +2015-08-21 17:39 +0000 [69824fdfbf] Richard Mudgett + + * res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used. + + Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093 + +2015-09-09 12:24 +0000 [2526659432] Richard Mudgett + + * res_pjsip_pubsub.c: Add some notification comments. + + Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20 + +2015-08-21 18:01 +0000 [9b290dfe2f] Richard Mudgett + + * res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response. + + We should not try to send a SIP response message because we may be + restoring a persistent subscription where we are not responding to a SIP + request. + + Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec + +2015-08-21 17:40 +0000 [73eb132012] Richard Mudgett + + * res_pjsip_pubsub.c: Fix off-nominal memory leak. + + Fix off-nominal visited vector leak in build_resource_tree(). + + Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c + +2015-08-21 15:26 +0000 [2b30fc2b2d] Richard Mudgett + + * res_pjsip_pubsub.c: Fix one byte buffer overrun error. + + ast_sip_pubsub_register_body_generator() did not account for the null + terminator set by sprintf() in the allocated output buffer. + + Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2 + +2015-08-21 15:25 +0000 [08a182c8e6] Richard Mudgett + + * res_pjsip_pubsub.c: Use ast_alloca() instead of alloca(). + + Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3 + +2015-08-21 11:04 +0000 [61f30db877] Richard Mudgett + + * res_pjsip_pubsub.c: Add missing error return in load_module(). + + Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc + +2015-08-21 11:03 +0000 [b8f07527b2] Richard Mudgett + + * res_pjsip/location.c: Use the builtin ao2_callback() match function instead. + + Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f + +2015-09-10 09:49 +0000 [f1a2e82d49] Mark Michelson + + * res_pjsip: Copy default_from_user to avoid crash. + + The default_from_user retrieval function was pulling the + default_from_user from the global configuration struct in an unsafe way. + If using a database as a backend configuration store, the global + configuration struct is short-lived, so grabbing a pointer from it + results in referencing freed memory. + + The fix here is to copy the default_from_user value out of the global + configuration struct. + + Thanks go to John Hardin for discovering this problem and proposing the + patch on which this fix is based. + + ASTERISK-25390 #close + Reported by Mark Michelson + + Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c + +2015-09-10 08:39 +0000 [bd71dcd1da] Matt Jordan + + * res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route + + We will only rewrite the Contact header if there is no Record-Route header in + the received request. If a malfunctioning proxy places a Record-Route header + into a REGISTER request, we will decide that we shouldn't update the IP/port + in the Contact header, and we will end up storing a contact with an AoR that + contains the NAT'd IP address. + + While it is nice to have the proxy *not* send a Record-Route in a REGISTER + request, it's also a good idea to not process the header in a non-dialog + message. This patch updates the code to explicitly ignore the Record-Route + header in REGISTER requests. + + ASTERISK-25387 #close + + Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f + +2015-09-09 16:46 +0000 [5bd363010e] Alexander Anikin + + * chan_ooh323: Add ProgressIndicator IE with inband info available + + Add ProgressIndicator IE with inband info present to Progress and + Alerting Q.931 message + + ASTERISK-25227 #close + Reported by: Alexandr Dranchuk + + Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203 +2015-09-08 10:35 +0000 [fcea6910f6] Scott Griepentrog + + * pjsip: avoid possible crash req_caps allocation failure + + Make certain that the pjsip session has not failed to + allocate the format capabilities structure, which can + otherwise cause a crash when referenced. + + ASTERISK-25323 + + Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750 + +2015-09-04 16:33 +0000 [8e5ed27a16] David M. Lee + + * res_rtp_asterisk: Add more ICE debugging + + In working through a recent ICE negotiation bug, I found the debug + logging in res_rtp_asterisk to be lacking. This patch adds a number of + debug and warning statements that were helpful. + + Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80 +2015-09-08 07:21 +0000 [3628e380b8] Joshua Colp + + * res_pjsip: Use hash for contact object identity instead of Contact URI. + + In the wild it is possible for Contact URIs to be quite long as + parameters can exist on them. This can present a problem when storing + them in the AstDB as the URI is used as part of the object name and + there is a fixed length limit for the AstDB. This will cause + the contact to not get stored. + + This change uses the MD5 hash of the Contact URI as part of the + object name instead. This has a fixed length which is guaranteed + to not exceed the AstDB length limit. + + ASTERISK-25295 #close + + Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02 + +2015-09-07 13:19 +0000 [d2106c0b21] Alexander Anikin + + * chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy + + Call ast_rtp_instance_stop on ooh323_destroy to free resources + allocated by rtp instance + + ASTERISK-25299 #close + Report by: Alexandr Dranchuk + + Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f + +2015-09-07 11:15 +0000 [ef3358d0c0] Matt Jordan + + * res/res_pjsip: Purge contacts when an AoR is deleted + + When an AoR is deleted by an external mechanism, such as through ARI, we + currently do not remove dynamic contacts that were created for that AoR as a + result of a received REGISTER request. As a result, re-creating the AoR will + cause the dynamic contact to be interpreted as a persistent contact, leading + to some rather strange state being created for the contacts/endpoints. + + This patch adds a sorcery observer for the 'aor' object. When a delete is + issued on the underlying sorcery object, the observer is called, and all + contacts created and persisted in sorcery for that AoR are also removed. Note + that we don't want to perform this action when an AO2 object that is an AoR is + destroyed, as the AoR can still exist in the backing storage (and we would + thus be removing valid contacts from an AoR that still "exists".) + + ASTERISK-25381 #close + + Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328 + +2015-09-05 14:58 +0000 [86b02228f5] Matt Jordan + + * channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id + + This patch adds a new option to the CHANNEL function that allows for the + extraction of the SIP call-id. It is used in conjunction with the 'pjsip' + option, and will return the Call-ID of the INVITE request that established + the PJSIP channel. + + ASTERISK-25352 + + Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a + +2015-09-04 16:06 +0000 [27c89053b0] David M. Lee + + * Fix when remote candidates exceed PJ_ICE_MAX_CAND + + We were passing the wrong count into pj_ice_sess_create_check_list(), + causing the create to fail if we ever received more than PJ_ICE_MAX_CAND + candidates. + + Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378 + +2015-09-04 14:40 +0000 [993ae9a669] Mark Michelson + + * res_pjsip: Change default from user value. + + When Asterisk sends an outbound SIP request, if there is no direct + reason to place a specific value for the username in the From header, + Asterisk would generate a UUID. For example, this would happen when + sending outbound OPTIONS requests when qualifying or when sending + outbound INVITE requests when originating (if no explicit caller ID were + provided). The issue is that some SIP providers reject these sorts of + requests with a "Name too long" error response. + + This patch aims to fix this by changing the default outbound username in + From headers to "asterisk". This value can be overridden by changing the + default_from_user option in the global options if desired. + + ASTERISK-25377 #close + Reported by Mark Michelson + + Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190 + +2015-09-03 14:07 +0000 [7d981b787c] Jonathan Rose + + * ParkAndAnnounce: Add variable inheritance + + In Asterisk 11, the announcer channel would receive channel variables + from the channel being parked by means of normal channel inheritance. + This functionality was lost during the big res_parking project in + Asterisk 12. This patch restores that functionality. + + ASTERISK-25369 #close + Review: https://gerrit.asterisk.org/#/c/1180/ + + Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e + +2015-09-04 09:26 +0000 [7691035312] Scott Griepentrog + + * endpoint snapshot: avoid second cleanup on alloc failure + + In ast_endpoint_snapshot_create(), a failure to init the + string fields results in two attempts to ao2_cleanup the + same pointer. Removed RAII_VAR to eliminate problem. + + ASTERISK-25375 #close + Reported by: Scott Griepentrog + + Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979 + +2015-09-04 05:33 +0000 [be31747db8] Martin Tomec + + * res/pjsip: Mark WSS transport as secure + + Pjsip is refusing to use unsecure transport with "sips" in url. + WSS should be considered as secure transport. + + ASTERISK-24602 #comment Partially fixed by setting WSS as secure + + Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353 + +2015-09-01 10:16 +0000 [fbdb42c9fc] Guido Falsi + + * Core/General: Add #ifdef needed on FreeBSD. + + pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD + too. + + ASTERISK-25310 #close + Reported by: Guido Falsi + + Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4 +2015-09-02 17:26 +0000 [c15d8cc0ed] Mark Michelson + + * res_pjsip: Fix contact refleak on stateful responses. + + When sending a stateful response, creation of the transaction can fail, + most commonly because we are trying to create a transaction from a + retransmitted request. When creation of the transaction fails, we end up + leaking a reference to a contact that was bumped when the response was + created. + + This patch adds the missing deref and fixes the reference leak. + + Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07 + +2015-09-02 12:41 +0000 [b51cf1e712] Joshua Colp + + * pbx: Fix crash when issuing "core show hints" with long pattern match. + + When issuing the "core show hints" CLI command a combination of both + the hint extension and context is created. This uses a fixed size + buffer expecting that the extension will not exceed maximum extension + length. When the extension is actually a pattern match this constraint + does not hold true, and the extension may exceed the maximum extension + length. In this case extra characters are written past the end of the + fixed size buffer. + + This change makes it so the construction of the combined hint extension + and context can not exceed the size of the buffer. + + ASTERISK-25367 #close + + Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499 + +2015-09-01 09:05 +0000 [beb568e51c] Mark Michelson + + * res_pjsip_pubsub: re-re-fix persistent subscription storage. + + A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as + a means of writing an appropriate packet to persistent storage. While + this partially solved the issue, it had its own problems. + pjsip_msg_print will always add a Content-Length header to the message + it prints. Frequent restarts of Asterisk can result in persistent + subscriptions being written with five or more Content-Length headers. In + addition, sometimes some apparent corruption of individual headers could + be seen. + + This aims to fix the problem by not running a parsed message through an + interpreter but rather by taking the raw message and saving it. The + logic for what to save is going to be different depending on whether a + SUBSCRIBE was received from the wire or if it was pulled from + persistence. When receiving a packet from the wire, when using a + streaming transport, the rdata->pkt_info.packet may contain multiple SIP + messages or fragments. However, the rdata->msg_info.msg_buf will always + contain the current SIP message to be processed. When pulling from + persistence, though, the rdata->msg_info.msg_buf will be NULL since no + transport actually handled the packet. However, since we know that we + will always ever pull one SIP message from persistence, we are free to + save directly from rdata->pkt_info.packet instead. + + ASTERISK-25365 #close + Reported by Mark Michelson + + Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b + +2015-08-29 10:36 +0000 [fc4d4f5379] Joshua Colp + + * taskprocessor: Fix race condition between unreferencing and finding. + + When unreferencing a taskprocessor its reference count is checked + to determine if it should be unlinked from the taskprocessors + container and its listener shut down. In between the time when the + reference count is checked and unlinking it is possible for + another thread to jump in, find it, and get a reference to it. If + the thread then uses the taskprocessor it may find that it is not + in the state it expects. + + This change locks the taskprocessors container during almost the + entire unreference operation to ensure that any other thread which + may attempt to find the taskprocessor has to wait. + + ASTERISK-25295 + + Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c + +2015-08-28 20:22 +0000 [bb38010c67] Joshua Colp + + * res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items. + + The keepalive support in res_pjsip_sdp_rtp currently assumes + that a stream will only be negotiated once. This is false. + If the stream is replaced and later added back it can be + negotiated again causing multiple keepalive scheduled items + to exist. This change explicitly deletes the existing + keepalive scheduled item before adding the new one. + + The res_pjsip_sdp_rtp module also does not stop RTP + keepalives or timeout timer if the stream has been + replaced. This change adds a callback to the session media + interface to allow a media stream to be stopped without + the resources being destroyed. This allows the scheduled + items and RTP to be stopped when the stream no longer + exists. + + ASTERISK-25356 #close + + Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de + +2015-08-28 19:57 +0000 [c036e50fbe] Joshua Colp + + * sched: ast_sched_del may return prematurely due to spurious wakeup + + When deleting a scheduled item if the item in question is currently + executing the ast_sched_del function waits until it has completed. + This is accomplished using ast_cond_wait. Unfortunately the + ast_cond_wait function can suffer from spurious wakeups so the + predicate needs to be checked after it returns to make sure it has + really woken up as a result of being signaled. + + This change adds a loop around the ast_cond_wait to make sure that + it only exits when the executing task has really completed. + + ASTERISK-25355 #close + + Change-Id: I51198270eb0b637c956c61aa409f46283432be61 + +2015-08-27 12:26 +0000 [229b95d253] Joshua Colp + + * res_pjsip_session: Don't invoke session supplements twice for BYE requests. + + When a BYE request is received the PJSIP invite session implementation + creates and sends a 200 OK response before we are aware of it. This + causes the INVITE session state callback to be called into and ultimately + the session supplements run on the BYE request. Once this response has + been sent the normal transaction state callback is invoked which + invokes the session supplements on the BYE request again. This can + be problematic in particular with res_pjsip_rfc3326 as it may + attempt to update the hangup cause code on the channel while it is + in the process of being hung up. + + This change makes it so the session supplements are only invoked + once by the INVITE session state callback. + + ASTERISK-25318 #close + + Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a + +2015-08-26 15:26 +0000 [6bfa14bdad] Scott Griepentrog + + * Chaos: handle failed allocation in get_media_encryption_type + + If the ast_strndup() call fails to allocate a copy of the + transport string for parsing, fail gracefully. + + ASTERISK-25323 + Reported by: Scott Griepentrog + + Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28 + +2015-08-26 14:25 +0000 [490db8ba94] Scott Griepentrog + + * Chaos: make hangup NULL tolerant + + In chan_pjsip_new, if allocation of the pvt + structure fails, ast_hangup is called. But + it was written to assume pvt was valid, and + this change corrects that. + + ASTERISK-25323 + Reported by: Scott Griepentrog + + Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87 +2015-08-26 05:40 +0000 [d03d09aad3] Joshua Colp + + * chan_sip: Allow call pickup to set the hangup cause. + + The call pickup implementation in chan_sip currently sets the channel + hangup cause to "normal clearing" if call pickup is successfully + performed. This action overwrites the "answered elsewhere" hangup cause + set by the call pickup code and can result in the SIP device in + question showing a missed call when it should not. + + This change sets the hangup cause to "normal clearing" as a + default initially but allows the call pickup to change it as + needed. + + ASTERISK-25346 #close + + Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff + +2015-08-25 07:17 +0000 [d013ecf748] Joshua Colp + + * res_pjsip: Add common ast_sip_get_host_ip API. + + Modules commonly used the pj_gethostip function for retrieving the + IP address of the host. This function does not cache the result and may + result in a DNS lookup occurring, or additional work. If the DNS + server is unreachable or network issues arise this can cause the + pj_gethostip function to block for a period of time. + + This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string + function which does the same thing but caches the host IP address at + module load time. This results in no additional work being done each + time the local host IP address is needed. + + ASTERISK-25342 #close + + Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e + +2015-08-24 06:21 +0000 [98d089fb9a] Joshua Colp + + * bridge: Kick channel from bridge if hung up during action. + + When executing an action in a bridge it is possible for the + channel to be hung up without the bridge becoming aware of it. + This is most easily reproducible by hanging up when the bridge + is streaming DTMF due to a feature timeout. This change makes + it so after action execution the channel is checked to determine + if it has been hung up and if it has it is kicked from the bridge. + + ASTERISK-25341 #close + + Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062 + +2015-08-24 11:04 +0000 [a408369bac] Joshua Colp + + * res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced + + When recreating a subscription it is possible for a freed sub_tree + to be referenced when the initial NOTIFY fails to be created. + + Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788 + +2015-08-23 18:26 +0000 [3af34441eb] Matt Jordan + + * res_pjsip/pjsip_configuration: Disregard empty auth values + + When an endpoint is backed by a non-static conf file backend (such as + the AstDB or Realtime), the 'auth' object may be returned as being an + empty string. Currently, res_pjsip will interpret that as being a valid + auth object, and will attempt to authenticate inbound requests. This + isn't desired; is an auth value is empty (which the name of an auth + object cannot be), we should instead interpret that as being an invalid + auth object and skip it. + + ASTERISK-25339 #close + + Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7 + +2015-08-21 23:37 +0000 [89003ea320] Rodrigo Ramírez Norambuena + + * README*: Remove trailing whitespace + + Change-Id: I18b7d75187548a9ed55b4f258d21aaaf29d08874 + +2015-07-28 13:47 +0000 [857923d9c7] Richard Mudgett + + * chan_sip.c: Set preferred rx payload type mapping on incoming offers. + + ASTERISK-25166 + Reported by: Kevin Harwell + + ASTERISK-17410 + Reported by: Boris Fox + + Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e + +2015-07-24 18:46 +0000 [d643b206c6] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Set preferred rx payload type mapping on incoming offers. + + ASTERISK-25166 + Reported by: Kevin Harwell + + ASTERISK-17410 + Reported by: Boris Fox + + Change-Id: I97ecebc1ab9b5654fb918bf1f4c98c956b852369 + +2015-07-27 19:19 +0000 [f7df3e1a01] Richard Mudgett + + * rtp_engine.c: Get current or create a needed rx payload type mapping. + + * Make ast_rtp_codecs_payload_code() get the current mapping or create a + rx payload type mapping. + + ASTERISK-25166 + Reported by: Kevin Harwell + + ASTERISK-17410 + Reported by: Boris Fox + + Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56 + +2015-07-27 19:15 +0000 [38854a9f7b] Richard Mudgett + + * rtp_engine.c: Extract rtp_codecs_payload_replace_rx(). + + ASTERISK-25166 + Reported by: Kevin Harwell + + ASTERISK-17410 + Reported by: Boris Fox + + Change-Id: I34e23bf5b084c8570f9c3e6ccd19b95fe85af239 + +2015-07-23 19:24 +0000 [1a549ed134] Richard Mudgett + + * rtp_engine.c: Initial split of payload types into rx and tx mappings. + + There are numerous problems with the current implementation of the RTP + payload type mapping in Asterisk. It uses only one mapping structure to + associate payload types to codecs. The single mapping is overkill if all + of the payload type values are well known values. Dynamic payload type + mappings do not work as well with the single mapping because RFC3264 + allows each side of the link to negotiate different dynamic mappings for + what they want to receive. Not only could you have the same codec mapped + for sending and receiving on different payload types you could wind up + with the same payload type mapped to different codecs for each direction. + + 1) An independent payload type mapping is needed for sending and + receiving. + + 2) The receive mapping needs to keep track of previous mappings because of + the slack to when negotiation happens and current packets in flight using + the old mapping arrive. + + 3) The transmit mapping only needs to keep track of the current negotiated + values since we are sending the packets and know when the switchover takes + place. + + * Needed to create ast_rtp_codecs_payload_code_tx() and make some callers + use the new function because ast_rtp_codecs_payload_code() was used for + mappings in both directions. + + * Needed to create ast_rtp_codecs_payloads_xover() for cases where we need + to pass preferred codec mappings to the peer channel for early media + bridging or when we need to prefer the offered mapping that RFC3264 says + we SHOULD use. + + * ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are + the only new public functions created. All the others were only used for + the tx or rx mapping direction so the function doxygen now reflects which + direction the function operates. + + * chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing + that makes no sense when processing an incoming SDP. We would be wiping + out any mappings that we set for the possible outgoing SDP we sent + earlier. + + ASTERISK-25166 + Reported by: Kevin Harwell + + ASTERISK-17410 + Reported by: Boris Fox + + Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac + +2015-08-19 12:10 +0000 [21d419e4fc] Richard Mudgett + + * ari/ari_websockets.c: Fix ast_debug parameter type mismatch. + + This is a type mismatch fix of the debugging commit + c63316eec10e1990a88bf4712238d6deb375bfa9 made to find out why + a testsuite test was failing only on one of the continuous + integration build agents. + + Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75 + +2015-08-19 10:30 +0000 [53e2a6a829] Scott Griepentrog + + * contrib: script install_prereq should install sqlite3 + + Asterisk needs the sqlite 3 library, which is package + sqlite-devel in CentOS. By adding this package to the + script, a problem with configure failing is resolved. + + ASTERISK-25331 #close + Reported by: Kevin Harwell + + Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec + +2015-08-18 15:07 +0000 [03eb6cbc10] Richard Mudgett + + * res_ari_events: Fix shutdown ref leak. + + ASTERISK-25308 #close + Reported by: Joshua Colp + + Change-Id: I592785bf70ff4b63d00e535b482f40da8e82a082 + +2015-08-18 14:24 +0000 [e1e7e205bc] Richard Mudgett + + * res_http_websocket.c: Add missing unref on an off nominal path. + + Change-Id: I228df6adecd4cb450d03e09e9a38c86bb566e811 + +2015-08-18 16:06 +0000 [59253a2262] Richard Mudgett + + * res_http_websocket.c: Fix some off nominal path cleanup. + + * Remove extraneous unlock on off-nominal path. + * Add missing HTTP error reply. + + Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b + +2015-08-18 14:46 +0000 [1f0a9f8a76] Richard Mudgett + + * res_ari.c: Add missing off nominal unlock and remove a RAII_VAR(). + + Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf + +2015-08-14 12:55 +0000 [9fb4a96e15] Richard Mudgett + + * app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'. + + Setting the 'paused' and 'ringinuse' options on a queue member using the + dialplan function QUEUE_MEMBER did not behave the same way as the + equivalent dialplan applications or AMI actions. + + * Made queue_function_mem_write() call the set_member_paused() and + set_member_value() for the 'paused' and 'ringinuse' options respectively. + A beneficial side effect is that the queue name is now optional and sets + the value in all queues the interface is a member. + + * Update QUEUE_MEMBER XML documentation. + + * Fix error checking in QUEUE_MEMBER() write. + + ASTERISK-25215 #close + Reported by: Lorne Gaetz + + Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb + +2015-08-17 16:41 +0000 [87b22969a4] Richard Mudgett + + * app_queue.c: Extract some functions for simpler code. + + * Extract set_queue_member_pause() from set_member_paused() for simpler + and more consistent code. + + * Extract set_queue_member_ringinuse() from + set_member_ringinuse_help_members() for simpler code. + + Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306 + +2015-08-17 13:34 +0000 [5cf98e2459] Richard Mudgett + + * app_queue.c: Fix error checking in QUEUE_MEMBER() read. + + Change-Id: I7294e13d27875851c2f4ef6818adba507509d224 + +2015-08-17 11:00 +0000 [178e1adffb] Scott Griepentrog + + * CHAOS: prevent sorcery object with null id + + When allocating a sorcery object, fail if the + id value was not allocated. + + ASTERISK-25323 + Reported by: Scott Griepentrog + + Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e + +2015-08-14 15:46 +0000 [5a85711568] Mark Michelson + + * res_pjsip_sdp_rtp: Restore removed NULL check. + + When sending an RTP keepalive, we need to be sure we're not dealing with + a NULL RTP instance. There had been a NULL check, but the commit that + added the rtp_timeout and rtp_hold_timeout options removed the NULL + check. + + Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64 + +2015-08-13 12:30 +0000 [7c4cb8618d] Richard Mudgett + + * audiohook.c: Simplify variable usage in audiohook_read_frame_both(). + + Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c + +2015-08-13 12:22 +0000 [bb37473234] Richard Mudgett + + * audiohook.c: Fix MixMonitor crash when using the r() or t() options. + + The built frame format in audiohook_read_frame_both() is now set to a + signed linear format before the rx and tx frames are duplicated instead of + only for the mixed audio frame duplication. + + ASTERISK-25322 #close + Reported by Sean Pimental + + Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538 + +2015-08-12 12:59 +0000 [43bdddfc26] Kevin Harwell + + * chan_sip.c: wrong peer searched in sip_report_security_event + + In chan_sip, after handling an incoming invite a security event is raised + describing authorization (success, failure, etc...). However, it was doing + a lookup of the peer by extension. This is fine for register messages, but + in the case of an invite it may search and find the wrong peer, or a non + existent one (for instance, in the case of call pickup). Also, if the peers + are configured through realtime this may cause an unnecessary database lookup + when caching is enabled. + + This patch makes it so that sip_report_security_event searches by IP address + when looking for a peer instead of by extension after an invite is processed. + + ASTERISK-25320 #close + + Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4 +2015-08-13 05:26 +0000 [495dfb24b7] Joshua Colp + + * res_http_websocket: When shutting down a session don't close closed socket + + Due to the use of ast_websocket_close in session termination it is + possible for the underlying socket to already be closed when the + session is terminated. This occurs when the close frame is attempted + to be written out but fails. + + Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b +2015-08-11 05:24 +0000 [7e65be4ecd] Joshua Colp + + * res_http_websocket: Forcefully terminate on write errors. + + The res_http_websocket module will currently attempt to close + the WebSocket connection if fatal cases occur, such as when + attempting to write out data and being unable to. When the + fatal cases occur the code attempts to write a WebSocket close + frame out to have the remote side close the connection. If + writing this fails then the connection is not terminated. + + This change forcefully terminates the connection if the + WebSocket is to be closed but is unable to send the close frame. + + ASTERISK-25312 #close + + Change-Id: I10973086671cc192a76424060d9ec8e688602845 + +2015-08-09 18:42 +0000 [a87e2dd254] Matt Jordan + + * res/res_format_attr_silk: Expose format attributes to other modules + + This patch adds the .get callback to the format attribute module, such + that the Asterisk core or other third party modules can query for the + negotiated format attributes. + + Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c + +2015-08-10 13:43 +0000 [87c92d2aee] Richard Mudgett + + * chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. + + Pressing DTMF digits on a phone to go out on a DAHDI channel can result in + the digit not being recognized or even heard by the peer. + + Phone -> Asterisk -> DAHDI/channel + + Turns out the DAHDI behavior with DTMF generation (and any other generated + tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When + Asterisk requests to start sending DTMF then DAHDI waits until its write + buffer is empty before generating any samples for the DTMF tones. When + Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI + immediately stops generating the DTMF samples. As a result, the more + samples there are in the DAHDI write buffer the shorter the time DTMF + actually gets sent on the wire. If there are more samples in the write + buffer than the time DTMF is supposed to be sent then no DTMF gets sent on + the wire. With the "buffers=12,half" setting and each buffer representing + 20 ms of samples then the DAHDI write buffer is going to contain around + 120 ms of samples. For DTMF to be recognized by the peer the actual sent + DTMF duration needs to be a minimum of 40 ms. Therefore, the intended + duration needs to be a minimum of 160 ms for the peer to receive the + minimum DTMF digit duration to recognize it. + + A simple and effective solution to work around the DAHDI behavior is for + Asterisk to flush the DAHDI write buffer when sending DTMF so the full + duration of DTMF is actually sent on the wire. When someone is going to + send DTMF they are not likely to be talking before sending the tones so + the flushed write samples are expected to just contain silence. + + * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting + to send a DTMF digit. + + ASTERISK-25315 #close + Reported by John Hardin + + Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a + +2015-08-05 14:21 +0000 [b9b957d4e9] Richard Mudgett + + * chan_dahdi.c: Lock private struct for ast_write(). + + There is a window of opportunity for DTMF to not go out if an audio frame + is in the process of being written to DAHDI while another thread starts + sending DTMF. The thread sending the audio frame could be past the + currently dialing check before being preempted by another thread starting + a DTMF generation request. When the thread sending the audio frame + resumes it will then cause DAHDI to stop the DTMF tone generation. The + result is no DTMF goes out. + + * Made dahdi_write() lock the private struct before writing to the DAHDI + file descriptor. + + ASTERISK-25315 + Reported by John Hardin + + Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb + +2015-08-10 18:23 +0000 [f3f5b45d57] Richard Mudgett + + * res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. + + If the saved SUBSCRIBE message is not parseable for whatever reason then + Asterisk could crash when libpjsip tries to parse the message and adds an + error message to the parse error list. + + * Made ast_sip_create_rdata() initialize the parse error rdata list. The + list is checked after parsing to see that it remains empty for the + function to return successful. + + ASTERISK-25306 + Reported by Mark Michelson + + Change-Id: Ie0677f69f707503b1a37df18723bd59418085256 + +2015-08-10 07:40 +0000 [991d4da1eb] Alexander Traud + + * chan_sip: Fix negotiation of iLBC 30. + + iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is + supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, + only iLBC 30 is negotiated now. + + ASTERISK-25309 #close + + Change-Id: I92d724600a183eec3114da0ac607b994b1a793da + +2015-08-09 17:56 +0000 [e188192ad1] Matt Jordan + + * main/format: Add an API call for retrieving format attributes + + Some codecs that may be a third party library to Asterisk need to have + knowledge of the format attributes that were negotiated. Unfortunately, + when the great format migration of Asterisk 13 occurred, that ability + was lost. + + This patch adds an API call, ast_format_attribute_get, to the core + format API, along with updates to the unit test to check the new API + call. A new callback is also now available for format attribute modules, + such that they can provide the format attribute values they manage. + + Note that the API returns a void *. This is done as the format attribute + modules themselves may store format attributes in any particular manner + they like. Care should be taken by consumers of the API to check the + return value before casting and dereferencing. Consumers will obviously + need to have a priori knowledge of the type of the format attribute as + well. + + Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3 + +2015-08-07 22:11 +0000 [d5f0c27122] David M. Lee + + * Replace htobe64 with htonll + + We don't have a compatability function to fill in a missing htobe64; but + we already have one for the identical htonll. + + Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac + +2015-07-24 17:04 +0000 [40caf0ad9b] David M. Lee + + * Replaces clock_gettime() with ast_tsnow() + + clock_gettime() is, unfortunately, not portable. But I did like that + over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we + usually do when we want a timespec and all we have is ast_tvnow(). + + This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a + timespec. If clock_gettime() is available, it will use that. Otherwise + ast_tsnow() falls back to using ast_tvnow(). + + Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e + +2015-08-07 14:20 +0000 [12e6f5ac01] Scott Emidy + + * ARI: Retrieve existing log channels + + An http request can be sent to get the existing Asterisk logs. + + The command "curl -v -u user:pass -X GET 'http://localhost:8088 + /ari/asterisk/logging'" can be run in the terminal to access the + newly implemented functionality. + + * Retrieve all existing log channels + + ASTERISK-25252 + + Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808 + +2015-08-07 11:14 +0000 [b91ca7ba49] Scott Emidy + + * ARI: Creating log channels + + An http request can be sent to create a log channel + in Asterisk. + + The command "curl -v -u user:pass -X POST + 'http://localhost:088/ari/asterisk/logging/mylog? + configuration=notice,warning'" can be run in the terminal + to access the newly implemented functionality for ARI. + + * Ability to create log channels using ARI + + ASTERISK-25252 + + Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782 + +2015-08-06 15:18 +0000 [f19c4930c2] Scott Emidy + + * ARI: Deleting log channels + + An http request can be sent to delete a log channel + in Asterisk. + + The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 + /ari/asterisk/logging/mylog'" can be run in the terminal + to access the newly implemented functionally for ARI. + + * Able to delete log channels using ARI + + ASTERISK-25252 + + Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6 + +2015-08-06 12:48 +0000 [382334cc06] Mark Michelson + + * res_pjsip_pubsub: More accurately persist packet. + + The pjsip_rx_data structure has a pkt_info.packet field on it that is + the packet that was read from the transport. For datagram transports, + the packet read from the transport will correspond to the SIP message + that arrived. For streamed transports, however, it is possible to read + multiple SIP messages in one packet. + + In a recent case, Asterisk crashed on a system where TCP was being used. + This is because at some point, a read from the TCP socket resulted in a + 200 OK response as well as an incoming SUBSCRIBE request being stored in + rdata->pkt_info.packet. When the SUBSCRIBE was processed, the + combination 200 OK and SUBSCRIBE was saved in persistent storage. Later, + a restart of Asterisk resulted in the crash because the persistent + subscription recreation code ended up building the 200 OK response + instead of a SUBSCRIBE request, and we attempted to access + request-specific data. + + The fix here is to use the pjsip_msg_print() function in order to + persist SUBSCRIBE requests. This way, rather than using the raw socket + data, we use the parsed SIP message that PJSIP has given us. If we + receive multiple SIP messages from a single read, we will be sure only + to save off the relevant SIP message. There also is a safeguard put in + place to make sure that if we do end up reconstructing a SIP response, + it will not cause a crash. + + ASTERISK-25306 #close + Reported by Mark Michelson + + Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2 + +2015-08-04 16:12 +0000 [4b6c657a82] Joshua Colp + + * res_pjsip: Ensure sanitized XML is NULL terminated. + + The ast_sip_sanitize_xml function is used to sanitize + a string for placement into XML. This is done by examining + an input string and then appending values to an output + buffer. The function used by its implementation, strncat, + has specific behavior that was not taken into account. + If the size of the input string exceeded the available + output buffer size it was possible for the sanitization + function to write past the output buffer itself causing + a crash. The crash would either occur because it was + writing into memory it shouldn't be or because the resulting + string was not NULL terminated. + + This change keeps count of how much remaining space is + available in the output buffer for text and only allows + strncat to use that amount. + + Since this was exposed by the res_pjsip_pidf_digium_body_supplement + module attempting to send a large message the maximum allowed + message size has also been increased in it. + + A unit test has also been added which confirms that the + ast_sip_sanitize_xml function is providing NULL terminated + output even when the input length exceeds the output + buffer size. + + ASTERISK-25304 #close + + Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302 + +2015-08-05 05:23 +0000 [7351d33a1f] Joshua Colp + + * res_rtp_asterisk: Don't leak temporary key when enabling PFS. + + A change recently went in which enabled perfect forward secrecy for + DTLS in res_rtp_asterisk. This was accomplished two different ways + depending on the availability of a feature in OpenSSL. The fallback + method created a temporary instance of a key but did not free it. + This change fixes that. + + ASTERISK-25265 + + Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396 +2015-08-04 09:47 +0000 [c63316eec1] Mark Michelson + + * res_http_websocket: Debug write lengths. + + Commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee attempted to fix a + test failure observed on 32 bit test agents by ensuring that a cast from + a 32 bit unsigned integer to a 64 bit unsigned integer was happening in + a predictable place. As it turns out, this did not cause test runs to + succeed. + + This commit adds several redundant debug messages that print the payload + lengths of websocket frames. The idea here is that this commit will not + cause tests to succeed for the faulty test agent, but we might deduce + where the fault lies more easily this way by observing at what point the + expected value (537) changes to some ungangly huge number. + + If you are wondering why something like this is being committed to the + branch, keep in mind that in commit + 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee I noted that the observed test + failures only happen when automated tests are run. Attempts to run the + tests by hand manually on the test agent result in the tests passing. + + Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d + +2015-08-03 11:06 +0000 [35a98161df] Mark Michelson + + * res_http_websocket: Avoid passing strlen() to ast_websocket_write(). + + We have seen a rash of test failures on a 32-bit build agent. Commit + 48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where + we were not encoding a 64-bit value correctly over the wire. This + commit, however, did not solve the test failures. + + In the failing tests, ARI is attempting to send a 537 byte text frame + over a websocket. When sending a frame this small, 16 bits are all that + is required in order to encode the payload length on the websocket + frame. However, ast_websocket_write() thinks that the payload length is + greater than 65535 and therefore writes out a 64 bit payload length. + Inspecting this payload length, the lower 32 bits are exactly what we + would expect it to be, 537 in hex. The upper 32 bits, are junk values + that are not expected to be there. + + In the failure, we are passing the result of strlen() to a function that + expects a uint64_t parameter to be passed in. strlen() returns a size_t, + which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit + unsigned value to somewhere where a 64-bit unsigned value is expected + would cause no problems. In fact, in manual runs of failing tests, this + works just fine. However, ast_websocket_write() uses the Asterisk + optional API, which means that rather than a simple function call, there + are a series of macros that are used for its declaration and + implementation. These macros may be causing some sort of error to occur + when converting from a 32 bit quantity to a 64 bit quantity. + + This commit changes the logic by making existing ast_websocket_write() + calls use ast_websocket_write_string() instead. Within + ast_websocket_write_string(), the 64-bit converted strlen is saved in a + local variable, and that variable is passed to ast_websocket_write() + instead. + + Note that this commit message is full of speculation rather than + certainty. This is because the observed test failures, while always + present in automated test runs, never occur when tests are manually + attempted on the same test agent. The idea behind this commit is to fix + a theoretical issue by performing changes that should, at the least, + cause no harm. If it turns out that this change does not fix the failing + tests, then this commit should be reverted. + + Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67 + +2015-07-29 14:17 +0000 [1f02d20da4] Benjamin Ford + + * ARI: Rotate log channels. + + An http request can be sent to rotate a specified log channel. + If the channel does not exist, an error response will be + returned. + + The command "curl -v -u user:pass -X PUT 'http://localhost:8088 + /ari/asterisk/logging/logChannelName/rotate'" can be run in the + terminal to access this new functionality. + + * Added the ability to rotate log files through ARI + + ASTERISK-25252 + + Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01 + +2015-07-31 11:27 +0000 [fe804b09b3] Ashley Sanders + + * ARI: Channels added to Stasis application during WebSocket creation ... + + Prior to ASTERISK-24988, the WebSocket handshake was resolved before Stasis + applications were registered. This was done such that the WebSocket would be + ready when an application is registered. However, by creating the WebSocket + first, the client had the ability to make requests for the Stasis application + it thought had been created with the initial handshake request. The inevitable + conclusion of this scenario was the cart being put before the horse. + + ASTERISK-24988 resolved half of the problem by ensuring that the applications + were created and registered with Stasis prior to completing the handshake + with the client. While this meant that Stasis was ready when the client + received the green-light from Asterisk, it also meant that the WebSocket was + not yet ready for Stasis to dispatch messages. + + This patch introduces a message queuing mechanism for delaying messages from + Stasis applications while the WebSocket is being constructed. When the ARI + event processor receives the message from the WebSocket that it is being + created, the event processor instantiates an event session which contains a + message queue. It then tries to create and register the requested applications + with Stasis. Messages that are dispatched from Stasis between this point and + the point at which the event processor is notified the WebSocket is ready, are + stashed in the queue. Once the WebSocket has been built, the queue's messages + are dispatched in the order in which they were originally received and the + queue is concurrently cleared. + + ASTERISK-25181 #close + Reported By: Matt Jordan + + Change-Id: Iafef7b85a2e0bf78c114db4c87ffc3d16d671a17 + +2015-07-29 12:58 +0000 [86034227ca] Mark Michelson + + * dns_core: Allow zero-length DNS responses. + + A testsuite test recently failed due to a crash that occurred in the DNS + core. The problem was that the test could not resolve an address, did + not set a result on the DNS query, and then indicated the query was + completed. The DNS core does not handle the case of a query with no + result gracefully, and so there is a crash. + + This changeset makes the DNS system resolver set a result with a + zero-length answer in the case that a DNS resolution failure occurs + early. The DNS core now also will accept such a response without + treating it as invalid input. A unit test was updated to no longer treat + setting a zero-length response as off-nominal. + + Change-Id: Ie56641e22debdaa61459e1c9a042e23b78affbf6 + +2015-07-29 13:49 +0000 [f49bef08a2] Richard Mudgett + + * rtp_engine.c: Fix performance issue with several channel drivers that use RTP. + + ast_rtp_codecs_get_payload() gets called once or twice for every received + RTP frame so it would be nice to not allocate an ao2 object to then have + it destroyed shortly thereafter. The ao2 object gets allocated only if + the payload type is not set by the channel driver as a negotiated value. + The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. + + * Made static_RTP_PT[] an array of ao2 objects that + ast_rtp_codecs_get_payload() can return instead of an array of structs + that must be copied into a created ao2 object. + + ASTERISK-25296 #close + Reported by: Richard Mudgett + + Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0 + +2015-07-29 17:00 +0000 [33a465249b] Richard Mudgett + + * res_rtp_asterisk.c: Fix off-nominal crash potential. + + ASTERISK-25296 + Reported by: Richard Mudgett + + Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b + +2015-07-29 13:48 +0000 [5f925d48b7] Richard Mudgett + + * rtp_engine.c: Must protect mime_types_len with mime_types_lock. + + Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e + +2015-07-24 18:38 +0000 [ba7dd38470] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Fixup some whitespace. + + Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973 + +2015-07-24 18:42 +0000 [3751bf0971] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list. + + Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2 + +2015-07-27 19:10 +0000 [e2d5d4db35] Richard Mudgett + + * rtp_engine.h: No sense allowing payload types larger than RFC allows. + + * Tweaked add_static_payload() to not use magic numbers. + + Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b + +2015-07-23 14:04 +0000 [bc1eae55cb] Richard Mudgett + + * rtp_engine.c: Minor tweaks. + + * Fix off nominial ref leak of new_type in + ast_rtp_codecs_payloads_set_m_type(). + + * No need to lock static_RTP_PT_lock in + ast_rtp_codecs_payloads_set_m_type() and + ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type + parameter sanity check. + + * No need to create ast_rtp_payload_type ao2 objects with a lock since the + lock is not used. + + Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4 + +2015-07-17 16:23 +0000 [d122c1e50b] Richard Mudgett + + * chan_sip.c: Tweak glue->update_peer() parameter nil value. + + Change glue->update_peer() parameter from 0 to NULL to better indicate it + is a pointer. + + Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd + +2015-07-23 12:41 +0000 [d12dc97fc9] Richard Mudgett + + * rtp_engine.h: Misc comment fixes. + + Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43 + +2015-07-30 17:05 +0000 [077c58cd5c] Richard Mudgett + + * res_pjsip_session.c: Fix crashes seen when call cancelled. + + Two testsuite tests crashed in the same place as a result of an INVITE + being CANCELed. + + tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified + tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp + + The session pointer is no longer in the inv->mod_data[session_module.id] + location because the INVITE transaction has reached the terminated state. + + ASTERISK-25297 #close + Reported by: Richard Mudgett + + Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427 + +2015-07-29 14:35 +0000 [5fcd1bc556] Mark Michelson + + * res_http_websocket: Properly encode 64 bit payload + + A test agent was continuously failing all ARI tests when run against + Asterisk 13. As it turns out, the reason for this is that on those test + runs, for some reason we decided to use the super extended 64 bit + payload length for websocket text frames instead of the extended 16 bit + payload length. For 64-bit payloads, the expected byte order over the + network is + + 7, 6, 5, 4, 3, 2, 1, 0 + + However, we were sending the payload as + + 3, 2, 1, 0, 7, 6, 5, 4 + + This meant that we were saying to expect an absolutely MASSIVE payload + to arrive. Since we did not follow through on this expected payload + size, the client would sit patiently waiting for the rest of the payload + to arrive until the test would time out. + + With this change, we use the htobe64() function instead of htonl() so + that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. + + Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a + +2015-07-29 12:23 +0000 [8fb8988fd4] Mark Michelson + + * Add a test event for inband ringing. + + This event is necessary for the bridge_wait_e_options test to be able to + confirm that ringing is being played on the local channel that runs the + BridgeWait() application with the e(r) option. + + ASTERISK-25292 #close + Reported by Kevin Harwell + + Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e + +2015-07-28 05:33 +0000 [1d081ec970] Mark Duncan + + * res/res_rtp_asterisk: Add ECDH support + + This will add ECDH support to Asterisk. It will + detect auto ECDH support in OpenSSL + (1.0.2b and above) during ./configure. If this is + available, it will use it, + otherwise it will fall back to prime256v1 (this + behavior is consistent with + other projects such as Apache and nginx). + + This fixes WebRTC being broken in Firefox 38+ due + to Firefox now only supporting + ciphers with perfect forward secrecy. + + ASTERISK-25265 #close + + Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b + +2015-07-16 12:16 +0000 [687597ca8c] Jonathan Rose + + * holding_bridge: ensure moh participants get frames + + Currently, if a blank musiconhold.conf is used, musiconhold will fail + to start for a channel going into a holding bridge with an anticipation + of getting music on hold. That being the case, no frames will be written + to the channel and that can pose a problem for blind transfers in PJSIP + which may rely on frames being written to get past the REFER framehook. + This patch makes holding bridges start a silence generator if starting + music on hold fails and makes it so that if no music on hold functions + are installed that the ast_moh_start function will report a failure so + that consumers of that function will be able to respond appropriately. + + ASTERISK-25271 #close + + Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99 + (cherry picked from commit 8458b8d441c2f4143ff135163ff3da4f88fe14c8) + +2015-07-18 11:16 +0000 [309dd2a409] Joshua Colp + + * pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. + + This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' + endpoint options. These allow the channel to be hung up if RTP + is not received from the remote endpoint for a specified number of + seconds. + + ASTERISK-25259 #close + + Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9 + +2015-07-24 09:46 +0000 [a0c31c7a05] Mark Michelson + + * res_pjsip: Add rtp_keepalive to sample config file. + + Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19 + +2015-07-23 13:11 +0000 [d97bed46b7] Mark Michelson + + * Local channels: Alternate solution to ringback problem. + + Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a + specific scenario involving local channels and a native local RTP bridge + could result in ringback still being heard on a calling channel even + after the call is bridged. + + That commit caused many tests in the testsuite to fail with alarming + consequences, such as not sending DialBegin and DialEnd events, and + giving incorrect hangup causes during calls. + + This commit reverts the previous commit and implements and alternate + solution. This new solution involves only passing AST_CONTROL_RINGING + frames across local channels if the local channel is in AST_STATE_RING. + Otherwise, the frame does not traverse the local channels. By doing + this, we can ensure that a playtones generator does not get started on + the calling channel but rather is started on the local channel on which + the ringing frame was initially indicated. + + ASTERISK-25250 #close + Reported by Etienne Lessard + + Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39 + +2015-07-22 12:24 +0000 [1cc99ba8b6] Joshua Colp + + * audiohook: Use manipulated frame instead of dropping it. + + Previous changes to sample rate support in audiohooks accidentally + removed code responsible for allowing the manipulate audiohooks + to work. Without this code the manipulated frame would be dropped + and not used. This change restores it. + + ASTERISK-25253 #close + + Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13 + +2015-07-22 09:46 +0000 [0b7148e262] Mark Michelson + + * Local channels: Do not block control -1 payloads. + + Control frames with a -1 payload are used as a special signal to stop + playtones generators on channels. This indication is sent both by + app_dial as well as by ast_answer() when a call is answered in case any + tones were being generated on a calling channel. + + This control frame type was made to stop traversing local channel pairs + as an optimization, because it was thought that it was unnecessary to + send these indications, and allowing such unnecessary control frames to + traverse the local channels would cause the local channels to optimize + away less quickly. + + As it turns out, through some special magic dialplan code, it is + possible to have a tones being played on a non-local channel, and it is + important for the local channel to convey that the tones should be + stopped. The result of having tones continue to be played on the + non-local channel is that the tones play even once the channel has been + bridged. By not blocking the -1 control frame type, we can ensure that + this situation does not happen. + + ASTERISK-25250 #close + Reported by Etienne Lessard + + Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815 + +2015-07-22 05:16 +0000 [e5fe8d40c8] Joshua Colp + + * audiohook: Read the correct number of samples based on audiohook format. + + Due to changes in audiohooks to support different sample rates the + underlying storage of samples is in the format of the audiohook + itself and not of the format being requested. This means that if a + channel is using G722 the samples stored will be at 16kHz. If + something subsequently reads from the audiohook at a format which + is not the same sample rate as the audiohook the number of samples + needs to be adjusted. + + Given the following example: + 1. Channel writing into audiohook at 16kHz (as it is using G722). + 2. Chanspy reading from audiohook at 8kHz. + + The original code would read 160 samples from the audiohook for + each 20ms of audio. This is incorrect. Since the audio in the + audiohook is at 16kHz the actual number needing to be read is 320. + Failure to read this much would cause the audiohook to reset + itself constantly as the buffer became full. + + This change adjusts the requested number of samples by determining + the duration of audio requested and then calculating how many + samples that would be in the audiohook format. + + ASTERISK-25247 #close + + Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d + +2015-07-20 15:59 +0000 [293c9f6894] Elazar Broad + + * cdr/cdr_adaptive_odbc.c: Fix quoted identifier usage when inserting CDR records + + Commit a24ce38 added support for the use of quoted indentifiers when inserting + CDR records into the database. However, the if statement logic responsible for + determining whether to use those identifiers is reversed, resulting in a + reference to the quoted identifier character buffer which will be null, hence + null terminating the SQL query, resulting in a truncated statement which + fails to execute. + + ASTERISK-25263 #close + Reported by: Elazar Broad + Tested by: Elazar Broad + + Change-Id: I40da47309b67cc1572207b1515dcc08ec9b1f644 +2015-07-20 12:39 +0000 [d02196448b] Rusty Newton + + * Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c + + * In sip.conf.sample fix sentence where we said that WS or WSS are supported + transports for use in an outbound register definition. They are not + supported in that case. + * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used + to enable CDR on a channel. + + ASTERISK-24867 #close + Reported by: Rusty Newton + + ASTERISK-24853 #close + Reported by: PSDK + + Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca + +2015-07-09 14:17 +0000 [2b42264e66] Mark Michelson + + * res_pjsip: Add rtp_keepalive endpoint option. + + This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the + chan_sip option, this specifies an interval, in seconds, at which we + will send RTP comfort noise frames. This can be useful for keeping RTP + sessions alive as well as keeping NAT associations alive during lulls. + + ASTERISK-25242 #close + Reported by Mark Michelson + + Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d + +2015-07-16 09:13 +0000 [8b503f2a10] Michael Cargile + + * res/res_musiconhold: Add a warning when MOH does not exist + + Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b + +2015-07-19 09:11 +0000 [9475dc9492] Matt Jordan + + * res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf + + Misconfiguring sorcery.conf with a 'config' wizard with no extra data + will currently crash Asterisk on startup, as the wizard requires a comma + delineated list to parse. This patch updates res_sorcery_config to check + for the presence of the data before it starts manipulating it. + + Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847 + +2015-07-16 09:46 +0000 [649460aa44] Joshua Colp + + * chan_pjsip: Don't change formats when frame of unsupported format is received. + + Receipt of an RTP packet currently causes the formats on an PJSIP channel to + change to the format of the RTP packet. In some off-nominal cases it's possible + for this to be a format that has not been configured or negotiated. This change + makes it so only formats explicitly configured on the endpoint are allowed. + + ASTERISK-25258 #close + + Change-Id: If93d641fb6418a285928839300d7854cab8c1020 + +2015-07-14 16:55 +0000 [4a875e8082] Richard Mudgett + + * pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable. + + ASTERISK-25256 #close + Reported by: Richard Mudgett + + Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3 + +2015-07-17 04:59 +0000 [7908ae4934] Patric Marschall + + * sig_pri.h: force_restart_unavailable_chans in wrong scope + + In channels/sig_pri.h, struct sig_pri_span, the field + force_restart_unavailable_chans is only defined if + + #if defined(HAVE_PRI_MCID) is true. + + All other occurences of force_restart_unavailable_chans are outside of the + + #if defined(HAVE_PRI_MCID) + endif + + scope. + + ASTERISK-25257 #close + Reported by: Patric Marschall + + Change-Id: I071de89cc2cd0d85927a013036e235851f672549 + +2015-07-08 16:39 +0000 [254d07b15b] Matt Jordan + + * ARI: Add support for push configuration of dynamic object + + This patch adds support for push configuration of dynamic, i.e., + sorcery, objects in Asterisk. It adds three new REST API calls to the + 'asterisk' resource: + * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current + object given its ID. This returns back a list of ConfigTuples, which + define the fields and their present values that make up the object. + * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an + object. A body may be passed with the request that contains fields to + populate in the object. The same format as what is retrieved using + the GET operation is used for the body, save that we specify that the + list of fields to update are contained in the "fields" attribute. + * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic + object from its backing storage. + + Note that the success/failure of these operations is somewhat + configuration dependent, i.e., you must be using a sorcery wizard that + supports the operation in question. If a sorcery wizard does not support + the create or delete mechanisms, then the REST API call will fail with a + 403 forbidden. + + ASTERISK-25238 #close + + Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c + +2015-07-15 15:40 +0000 [b34c4528ab] Richard Mudgett + + * strings.h: Fix issues with escape string functions. + + Fixes for issues with the ASTERISK-24934 patch. + + * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is + an empty string. If it were an empty string the functions returned NULL + as if there were a memory allocation failure. This failure caused the AMI + VarSet event to not get posted if the new value was an empty string. + + * Fixed dest buffer overwrite potential in ast_escape() and + ast_escape_c(). If the dest buffer size is smaller than the space needed + by the escaped s parameter string then the dest buffer would be written + beyond the end by the nul string terminator. The num parameter was really + the dest buffer size parameter so I renamed it to size. + + * Made nul terminate the dest buffer if the source string parameter s was + an empty string in ast_escape() and ast_escape_c(). + + * Updated ast_escape() and ast_escape_c() doxygen function description + comments to reflect reality. + + * Added some more unit test cases to /main/strings/escape to cover the + empty source string issues. + + ASTERISK-25255 #close + Reported by: Richard Mudgett + + Change-Id: Id77fc704600ebcce81615c1200296f74de254104 + +2015-07-14 14:29 +0000 [097c15ac51] Richard Mudgett + + * parking_applications.c: Fix ast_verb() line terminator. + + Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775 + +2015-07-14 14:36 +0000 [8b620c555b] Richard Mudgett + + * res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park. + + setup_park_common_datastore() was assuming that a non-NULL string returned + for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty + strings. Things got crashy as a result. + + * Made setup_park_common_datastore() treat the channel variable values the + same whether they are NULL or empty for ATTENDEDTRANSFER and + BLINDTRANSFER. + + ASTERISK-25254 #close + Reported by: Richard Mudgett + + Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2 + +2015-07-10 18:01 +0000 [4af24ec74b] Richard Mudgett + + * res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). + + Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb + +2015-07-10 10:42 +0000 [71b3bcf5e0] Richard Mudgett + + * res_pjsip_session.c: Add some helpful comments and minor tweaks. + + Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743 + +2015-07-10 10:43 +0000 [53c91737a5] Richard Mudgett + + * res_pjsip_session.c: Fix off nominal crash potential in debug message. + + Change-Id: I09928297927ee85f7655289acee3a586816466bc + +2015-07-15 10:31 +0000 [eff6a88a88] Matt Jordan + + * apps/app_dictate: Fix typo in attribution + + Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian + (GameGamer43) for pointing that out. + + Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106 + +2015-07-15 10:28 +0000 [e01d93e092] Benjamin Ford + + * ARI: Fixed unload mode for unload module. + + Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM, + which would unload a module even if it was in use. + + * Changed unload mode to proper mode + + ASTERISK-25173 + + Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533 + +2015-07-10 18:17 +0000 [1b666549f3] Richard Mudgett + + * res_pjsip_session.c: Fix crash on call disconnect. + + The crash fix for ASTERISK-25183 backported some code from master to try + to make sure that a BYE response is processed by the same serializer used + by the BYE request. The identified race condition causing that backport + was the BYE request code had not finished processing after sending the BYE + before the BYE response came in for processing under a different thread. + Unfortunately, there is still a race condition. Now the race condition is + between destroying the call session's serializer in + ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a + reference to the serializer for a BYE response. Even worse, the new race + condition is a design limitation of the taskprocessor implementation that + didn't matter in versions before v12. Back then, taskprocessors were only + destroyed when a module unloaded. Now res_pjsip can destroy them when a + call ends. + + However, as noted on the ASTERISK-25183 commit, + session_inv_on_state_changed() is disassociating the dialog from the + session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED. + This is a tad too soon because our BYE request transaction has not + completed yet. + + * Split session_end() that is called by session_inv_on_state_changed() to + hold off session destruction until the BYE transaction timeout occurs or a + failed initial INVITE transaction timeout occurs in + session_inv_on_tsx_state_changed(). + + ASTERISK-25201 #close + Reported by: Matt Jordan + + Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961 + +2015-07-14 13:12 +0000 [9d458b8311] Benjamin Ford + + * ARI: Added new functionality to reload a single module. + + An http request can be sent to reload an Asterisk module. If the + module can not be reloaded or is not already loaded, an error + response will be returned. + + The command "curl -v -u user:pass -X PUT 'http://localhost:8088 + /ari/asterisk/modules/{moduleName}'" (or something similar, based + on configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Asterisk modules can be reloaded through http requests + + ASTERISK-25173 + + Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1 + +2015-07-14 08:55 +0000 [f64f1c2772] Benjamin Ford + + * ARI: Added new functionality to unload a single module. + + An http request can be sent to unload an Asterisk module. If the + module can not be unloaded or is already unloaded, an error response + will be returned. + + The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 + /ari/asterisk/modules/{moduleName}'" (or something similar, depending + on configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Asterisk modules can be unloaded through http requests + + ASTERISK-25173 + + Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57 + +2015-07-13 16:00 +0000 [aa5707b889] Benjamin Ford + + * ARI: Added new functionality to load a single module. + + An http request can be sent to load an Asterisk module. If the + module can not be loaded or is loaded already, an error response + will be returned. + + The command curl -v -u user:pass -X POST 'http://localhost:8088/ari + /asterisk/modules/{moduleName}'" (or something similar, depending on + configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Asterisk modules can be loaded through http requests + + ASTERISK-25173 + + Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33 + +2015-07-13 10:54 +0000 [6a764db370] Benjamin Ford + + * ARI: Added new functionality to get information on a single module. + + An http request can be sent to retrieve information on a single + module, including the resource name, description, use count, status, + and support level. + + The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari + /asterisk/modules/{moduleName}'" (or something similar, depending on + configuration) can be run in the terminal to access this new + functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Information on a single module can now be retrieved + + ASTERISK-25173 + + Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463 + +2015-07-08 14:56 +0000 [c855523519] Kevin Harwell + + * bridge.c: Fixed race condition during attended transfer + + During an attended transfer a thread is started that handles imparting the + bridge channel. From the start of the thread to when the bridge channel is + ready exists a gap that can potentially cause problems (for instance, the + channel being swapped is hung up before the replacement channel enters the + bridge thus stopping the transfer). This patch adds a condition that waits + for the impart thread to get to a point of acceptable readiness before + allowing the initiating thread to continue. + + ASTERISK-24782 + Reported by: John Bigelow + + Change-Id: I08fe33a2560da924e676df55b181e46fca604577 + +2015-05-13 16:22 +0000 [ef82190804] Matt Jordan + + * media cache: Add CLI commands + + This patch adds five CLI commands for the media cache: + * 'media cache show all' - display a summary of all items in the media + cache. + * 'media cache show ' - display detailed information about a + single item in the media cache. + * 'media cache delete ' - remove an item from the media cache, and + inform the bucket backend for the URI scheme to remove the item as + well. + * 'media cache refresh ' - refresh a URI. If the item does not + exist in the media cache, the bucket backend will pull down the media + associated with the URI and create the item in the cache. + * 'media cache create ' - create an item in the media cache from + some local media storage. Note that the bucket backend for the URI + scheme must still permit the item creation. + + Change-Id: Id1c5707a3b8e2d96b56e4691a46a936cd171f4ae + +2015-01-29 08:38 +0000 [3ea0d38396] Matt Jordan + + * media cache: Add a core API and facade for a backend agnostic media cache + + This patch adds a new API to the Asterisk core that acts as a media + cache. The core API itself is mostly a thin wrapper around some bucket + API provided implementation that itself acts as the mechanism of + retrieval for media. The media cache API in the core provides the + following: + * A very thin in-memory cache of the active bucket_file items. Unlike a + more traditional cache, it provides no expiration mechanisms. Most + queries that hit the in-memory cache will also call into the bucket + implementations as well. The bucket implementations are responsible + for determining whether or not the active record is active and valid. + This makes sense for the most likely implementation of a media cache + backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of + whether or not a record is truly active; as such, the in-memory cache + in the core has to defer to it. + * The ability to create new items in the media cache from local + resources. This allows for re-creation of items in the cache on + restart. + * Synchronization of items in the media cache to the AstDB. This + also includes various pieces of important metadata. + + The API provides sufficient access that higher level APIs, such as the + file or app APIs, do not have to worry about the semantics of the bucket + APIs when needing to playback a resource. + + In addition, this patch provides unit tests for the media cache API. The + unit tests use a fake bucket backend to verify correctness. + + Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e + +2015-07-11 20:25 +0000 [887945d410] Matt Jordan + + * main/bucket: Add a callback function for ast_bucket_file objects + + This patch adds a new function to the bucket API for ast_bucket_file + objects, ast_bucket_file_metadata_callback. It will call ao2_callback on + the ast_bucket_file's ao2_container of metadata, calling the provided + ao2_callback_fn callback on each piece of metadata associated with the + file. + + This is particularly useful when a bucket backend has added metadata, + and a higher level API wants to be aware of/access said metadata, + without knowing for sure what the key is. + + Change-Id: I96f6757717f47b650df91a437f7df16406227466 + +2015-07-08 16:28 +0000 [458715d088] Matt Jordan + + * main/sorcery: Don't fail object set creation from JSON if field fails + + Some individual fields may fail their conversion due to their default + values being invalid for their custom handlers. In particular, + configuration values that depend on others being enabled (and thus have + an empty default value) are notorious for tripping this routine up. An + example of this are any of the DTLS options for endpoints. Any of the + DTLS options will fail to be applied (as DTLS is not enabled), causing + the entire object set to be aborted. + + This patch makes it so that we log a debug message when skipping a + field, and rumble on anyway. + + ASTERISK-25238 + + Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76 + +2015-07-08 16:21 +0000 [6ed58014f5] Matt Jordan + + * main/format_cap: Parse capabilities generated by ast_format_cap_get_names + + We have a strange relationship between the parsing of format + capabilities from a string and their representation as a string. We + expect the format capabilities to be expressed as a string in the + following format: + + allow = !all,ulaw,alaw + disallow = g722 + + While we would generate the string representation of those formats as: + + allow = (ulaw|alaw) + disallow = (ulaw|alaw|g729...) + + When the configuration framework needs to store values as a string, it + generates the format capabilities using the second representation; this + representation however cannot be parsed when the entry is rehydrated. + This patch fixes that by updating + ast_format_cap_update_by_allow_disallow to parse an entry as if it were + in the generated format if it has a leading '(' and a trailing ')'. + + ASTERISK-25238 + + Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca + +2015-07-08 16:38 +0000 [e64e586900] Matt Jordan + + * res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails + + Having a debug message tell us that we attempted to look up an item but + failed is nice in circumstances when it isn't clear if the wizard was + queried correctly or not. + + Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7 + +2015-07-08 16:37 +0000 [7c14dfdc61] Matt Jordan + + * res/res_pjsip_outbound_registration: Fix WARNING message + + Newlines are nice. + + Change-Id: Icf0d915db02882e47cd9077ed9009f5d44140d42 + +2015-07-08 16:35 +0000 [3e286e6b51] Matt Jordan + + * res_pjsip/configuration: Fix a variety of default value problems + + This patch fixes some bad default value handling in the following + settings: + + * The 'message_context' and 'accountcode' settings are not mandatory. As + such, we can allow their stringfield values to be empty. + * The 'media_encryption' setting applies a default value of 'none' to + the setting, which it then can't parse or understand. Since the value + is documented to be 'no', this will now apply that as the default + value. + + Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83 + +2015-07-08 16:32 +0000 [ffadb5f1de] Matt Jordan + + * main/sorcery: Provide log messages when a wizard does not support an operation + + If a sorcery wizard does not support one of the 'optional' CRUD + operations (namely the CUD), log a WARNING message so we are aware of + why the operation failed. This also removes an assert in this case, as + the CUD operation may have been triggered by an external system, in + which case it is not a programming error but a configuration error. + + Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53 + +2015-06-27 17:53 +0000 [5266796432] Matt Jordan + + * tests/test_devicestate: Add additional tests for the device state API + + This patch adds more tests that exercise the device state API. This includes: + + * Tests that cover adding a device state provider, as well as deleting a + device state provider. This also verifies that you cannot add an + already added device state provider, and cannot delete an already + deleted device state provider. + * A test that covers changing device state and receiving said updates + from a device state subscriber. This also covers hitting both the + device state cache as well as a custom device state provider. + * A test that covers converting device state to channel state and device + state values to a string representation and back. + * A test that covers obtaining device state from an active channel and a + channel driver that provides its own device state. + + Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d + +2015-06-27 17:51 +0000 [f77e688f20] Matt Jordan + + * main/devicestate: Prevent duplicate registration of device state providers + + Currently, the device state provider API will allow you to register a + device state provider with the same case insensitive name more than + once. This could cause strange issues, as the duplicate device state + providers will not be queried when a device's state has to be polled. + This patch updates the API such that a device state provider with the + same name as one that has already registered will be rejected. + + Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2 + +2015-06-26 10:57 +0000 [1b7760a8aa] Benjamin Ford + + * ARI: Added new functionality to get all module information. + + An http request can be sent to retrieve a list of all existing modules, + including the resource name, description, use count, status, and + support level. + + The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/ + asterisk/modules" (or something similar, depending on configuration) + can be run in the terminal to access this new functionality. + + For more information, see: + https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource + + * Added new ARI functionality + * Information on modules can now be retrieved + + Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0 + +2015-07-09 09:18 +0000 [4a25d55416] Joshua Colp + + * bridge_native_rtp.c: Don't start native RTP bridging after attended transfer. + + The bridge_native_rtp module adds a frame hook to channels which are in + a native RTP bridge. This frame hook is used to intercept when a hold + or unhold frame traverses the bridge so native RTP can be stopped or + started as appropriate. This is expected but exposes a specific bug + when attended transfers are involved. + + Upon completion of an attended transfer an unhold frame is queued up + to take one of the channels involved off hold. After this is done + the channel is moved between bridges. + + When the frame hook is involved in this case for the unhold it + releases the channel lock and acquires the bridge lock. This + allows the bridge core to step in and move the channel + (potentially changing the bridging techology) from another thread. + Once completed the bridge lock is released by the bridge core. + The frame hook is then able to acquire the bridge lock and + wrongfully starts native RTP again, despite the channel no longer + being in the bridge or needing to start native RTP. In fact at + this point the frame hook is no longer attached to the channel. + + This change makes it so the native RTP bridge data is available to + the frame hook when it is invoked. Whether the frame hook has + been detached or not is stored on the native RTP bridge data and + is checked by the frame hook before starting or stopping native + RTP bridging. If the frame hook has been detached it does nothing. + + ASTERISK-25240 #close + + Change-Id: I13a73186a05f4e5a764f81e5cd0ccec1ed1891d2 + +2015-07-08 04:21 +0000 [9276415f65] Joshua Colp + + * res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used. + + This change fixes a bug where the DTLS timeout timer would be + initialized to 0 if DTLS was not used for an RTP session. + + ASTERISK-25103 + + Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac + +2015-07-07 15:03 +0000 [3cdfd39af7] Ashley Sanders + + * DNS: Create a system-level DNS resolver + + Prior to this patch, the DNS core present in master had no default system-level + resolver implementation. Therefore, it was not possible for the DNS core to + perform resolutions unless the libunbound library was installed and the + res_resolver_unbound module was loaded. + + This patch introduces a system-level DNS resolver implementation that will + register itself with the lowest consideration priority available (to ensure + that it is to be used only as a last resort). The resolver relies on low-level + DNS search functions to perform a rudimentary DNS search based on a provided + query and then supplies the search results to the DNS core. + + ASTERISK-25146 #close + Reported By: Joshua Colp + + Change-Id: I3b36ea17b889a98df4f8d80d50bb7ee175afa077 + +2015-07-01 07:55 +0000 [5717340ab3] Joshua Colp + + * res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context. + + This change moves logic for setting up the DTLS SSL contexts to + when the SDP is done being processed instead of when ICE negotiation + completes. It also stops handshakes from being initiated when we + are acting as a server. + + Manipulating the SSL context when ICE negotiation has completed + is problematic as the SSL context is not protected and if acting + as a client the remote side may have started DTLS negotiation + already. + + The retransmission timeout timer code has also been split up + and simplified some. Both RTP and RTCP now have their own timers + and the points at which the timer is stopped and started is now + more specific. When a packet is sent the timer is started. When + a response is received but before it is processed the timer is + stopped. This provides a guarantee that the timeout is not + occurring while the response is processed. + + ASTERISK-22805 #close + ASTERISK-24550 #close + ASTERISK-24651 #close + ASTERISK-24832 #close + ASTERISK-25103 #close + ASTERISK-25127 #close + + Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91 + +2015-06-26 18:48 +0000 [189841ddb7] Richard Mudgett + + * res_pjsip_mwi.c: Fix MWI subscription memory corruption crash. + + MWI subscriptions can crash or corrupt memory when using the subscription + datastore to access the MWI subscription object because the datastore is + not holding a reference to the object. + + * Give the subscription datastore a ref to the MWI subscription object. + It is unfortunate that the ref causes a circular ref chain that must be + explicitly broken to allow the memory to get released. The loop is broken + when the subscription is shutdown and if the subscription setup fails. + + ASTERISK-25168 #close + Reported by: Carl Fortin + + Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f + +2015-07-02 14:51 +0000 [7cd99be534] Richard Mudgett + + * PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error. + + When res_pjsip body generator modules were generating XML or XPIDF + response bodies, there was a chance that the generated body would be the + exact size of the supplied buffer. Adding the nul string terminator would + then write beyond the end of the buffer and potentially corrupt memory. + + * Fix MALLOC_DEBUG high fence violations caused by adding a nul string + terminator on the end of a buffer for XML or XPIDF response bodies. + + * Made calls to pj_xml_print() safer if the XML prolog is requested. Due + to a bug in pjproject, the return value could be -1 _or_ + AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough. + + * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the + return value of pj_xml_print() when the supplied buffer is not large + enough. + + ASTERISK-25168 + Reported by: Carl Fortin + + Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de + +2015-06-26 10:36 +0000 [792ed7ce93] Richard Mudgett + + * PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences. + + When a caller calls a FAX number and then hangs up right after the call is + answered then the T.38 re-INVITE automatic reject timer may still be + running after the channel goes away. + + * Added session NULL channel checks on the code paths that get executed by + t38_automatic_reject() to prevent a crash when the T.38 re-INVITE + automatic reject timer expires. + + ASTERISK-25168 + Reported by: Carl Fortin + + Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403 + +2015-06-30 11:17 +0000 [030e8339dd] Richard Mudgett + + * res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str(). + + Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907 + +2015-06-30 11:14 +0000 [453d7b8d69] Richard Mudgett + + * res_pjsip_mwi.c: Eliminate a simple RAII_VAR. + + Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da + +2015-06-30 11:11 +0000 [786c6d42ef] Richard Mudgett + + * res_pjsip_mwi.c: Fix mid-line log message line breaks. + + * Add create_mwi_subscriptions_for_endpoint() doxygen comment. + + Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2 + +2015-06-26 16:10 +0000 [1b91094edd] Richard Mudgett + + * res_pjsip_t38.c: Fix always false if test. + + Calling t38_change_state() sets the t38 state so it makes little sense to + then check the state right after the call for something else. + + * Made the code in t38_interpret_parameters() reject or exit T.38 mode as + intended but not implemented. + + Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2 + +2015-06-30 15:19 +0000 [74135c8efa] Kevin Harwell + + * res_pjsip: Failover when server is not available + + Previously Asterisk did not properly failover to the next resolved DNS + address when a endpoint could not be reached. With this patch, and while + using res_pjsip, SIP requests (both in/out of dialog) now attempt to use + the next address in the list of resolved addresses until a proper response + is received or no more addresses are left. + + ASTERISK-25076 #close + Reported by: Joshua Colp + + Change-Id: Ief14f4ebd82474881f72f4538f4577f30af2a764 + +2015-07-06 09:24 +0000 [38a3c27a09] Joshua Colp + + * res_sorcery_memory_cache: Execute stale unit test last. + + In Jenkins there is currently a sporadic test failure of a + variable number of sorcery memory cache unit tests. I have not + been able to reproduce this on the build agents themselves or + on my development machine. + + My working theory is that the stale unit test is causing a + sorcery instance to persist longer than expected, causing subsequent + tests to fail when setting up and initializing the next + sorcery instance. + + To see if this is the case this change moves the stale unit test + to execute last so no subsequent unit tests can have issues + initializing their sorcery instance. + + Change-Id: Ifd6550a949613be774b75fa5db12c02110f82c4a + +2015-06-20 13:54 +0000 [ef8d3f6506] Matt Jordan + + * bucket: Add clone/staleness operations for ast_bucket/ast_bucket_file + + This patch enhances the bucket API in two ways. + + First, since ast_bucket and ast_bucket_file instances are immutable, a 'clone' + operation has been added that provides a 'clone' of an existing + ast_bucket/ast_bucket_file object. Note that this makes use of the + ast_sorcery_copy operation, along with the copy callback handler on the + "bucket" and "file" object types for the bucket sorcery instance. + + Second, there is a need for the bucket API to ask a wizard if an object + is stale. This is particularly useful with the upcoming media cache + enhancements, where we want to ask the backing data storage if the + object we are currently operating on has known updates. This patch adds + API calls for ast_bucket and ast_bucket_file objects, which callback + into their respective sorcery wizards via the sorcery API. + + Unit tests have also been added to cover the respective + ast_bucket/ast_bucket_file clone and staleness operations. + + Change-Id: Ib0240ba915ece313f1678a085a716021d75d6b4a + +2015-07-04 10:03 +0000 [b178f8701b] Matt Jordan + + * sorcery: Add support for object staleness + + This patch enhances the sorcery API to allow for sorcery wizards to + determine if an object is stale. This includes the following: + + * Sorcery objects now have a timestamp that is set on creation. Since + sorcery objects are immutable, this can be used by sorcery wizards to + determine if an object is stale. + + * A new API call has been added, ast_sorcery_is_stale. This API call + queries the wizards associated with the object, calling a new callback + function 'is_stale'. Note that if a wizard does not support the new + callback, objects are always assumed to not be stale. + + * Unit tests have been added that cover the new API call. + + Change-Id: Ica93c6a4e8a06c0376ea43e00cf702920b806064 + +2015-07-04 18:22 +0000 [f35a4b8525] Joshua Colp + + * res/res_http_websocket: Don't send HTTP response fragmented. + + This change makes it so that when accepting a WebSocket + connection the HTTP response is sent as one packet instead of + fragmented. Browsers don't like it when you send it fragmented. + + ASTERISK-25103 + + Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69 + +2015-06-27 18:47 +0000 [2c17515f3c] Matt Jordan + + * Makefile: Remove coverage files on 'make clean' + + This patch updates a variety of Makefiles in Asterisk's build system to + remove .gcda and .gcno files when 'make clean' is executed. These files + are generated when '--enable-coverage' is passed to the Asterisk + configure script. + + Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602 + +2015-07-02 09:08 +0000 [34323f9f95] Walter Doekes + + * chan_sip: Fix early call pickup channel leak. + + When handle_invite_replaces() was called, and either ast_bridge_impart() + failed or there was no bridge (because the channel we're picking up was + still ringing), chan_sip would leak a channel. + + Thanks Matt and Corey for checking the bridge path. + + ASTERISK-25226 #close + + Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8 + +2015-07-01 16:04 +0000 [ef74ccb18d] Matt Jordan + + * sorcery/realtime: Add a bit of debug and warning messages for bad configs + + When a mapping does not exist between a sorcery.conf defined object and + a realtime mapping in extconf, currently, the user will receive a slew + of ERROR messages that don't really tell what is happening. Some ERROR + messages may even be misleading, as they occur after the sorcery API has + already given up on the attempt to load and create the sorcery object. + + This patch adds a bit of debug and a useful WARNING message for when a + wizard's open callback fails for a particular object type. In the bad + configurations that resulted in this patch, this provided a 'root cause' + WARNING message that pointed in the right direction of the configuration + problem. + + Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b + +2015-07-02 06:54 +0000 [f18436642b] Joshua Colp + + * dns: Fix crash when invoking cancel in DNS recurring unit test. + + The recurring unit test expects the user data on a DNS query + created as a result of a recurring DNS query to be the recurring + structure itself. This is true, mostly. When invoking the user + provided callback this user data is changed to the user provided + data. This presents a race condition where the data may or may + not point to the recurring data. + + This change simplifies the callback of the user provided callback + by creating a new query and populating it with the expected values. + This leaves the recurring DNS query alone and fixes the race + condition. This is more in line with how the API should be used + overall. + + ASTERISK-25222 #close + + Change-Id: I10fb6deec025dff097157e7ec17e6e4921778478 + +2015-07-02 06:19 +0000 [6fbb58c7f7] Walter Doekes + + * chan_mgcp: Don't call close on fd -1. + + ASTERISK-25220 #close + + Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3 + +2015-07-02 06:10 +0000 [13a318bbb1] Walter Doekes + + * rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format. + + When running valgrind on Asterisk, it complained about: + + ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304) + ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...) + ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292) + ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437) + + The code in question is a struct assignment, which may be performed by + memcpy as a compiler optimization. It is changed to only copy the struct + contents if source and destination are different. + + ASTERISK-25219 #close + + Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a + +2015-07-02 05:16 +0000 [40274e3652] Walter Doekes + + * astfd: Fix buffer overflow in DEBUG_FD_LEAKS. + + If DEBUG_FD_LEAKS was used and more file descriptors than the default of + 1024 were available, some DEBUG_FD_LEAKS-patched functions would + overwrite memory past the fixed-size (1024) fdleaks buffer. + + This change: + - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe + - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024 + - stores pointers to constants instead of copying the contents + - reorders the fdleaks struct for possibly tighter packing + - adds a tiny bit of documentation + + ASTERISK-25212 #close + + Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5 + +2015-07-02 04:57 +0000 [3fab8212e3] Walter Doekes + + * res_timing: Don't close FD 0 when out of open files. + + This fixes so a failure to get a timer file descriptor does not cascade + to closing FD 0. + + On error, both res_timing_kqueue and res_timing_timerfd would call the + destructor before setting the file handle. The file handle had been + initialized to 0, causing FD 0 to be closed. This in turn, resulted in + floods of "CLI>" messages and an unusable terminal. + + ASTERISK-19277 #close + Reported by: Barry Chern + + For the master branch, this was already fixed. This patch only ensures + that we do not attempt to close a negative file descriptor. + + Change-Id: I147d7e33726c6e5a2751928d56561494f5800350 + +2015-07-01 17:25 +0000 [41610df8d5] Richard Mudgett + + * chan_vpb.cc: Fix compiler warning Jenkins found. + + Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0 + +2015-07-01 13:34 +0000 [537df26f9c] Scott Griepentrog + + * Channel alert pipe: improve diagnostic error return + + When a frame is queued on a channel, any failure in + ast_channel_alert_write is logged along with errno. + + This change improves the diagnostic message through + aligning the errno value with actual failure cases. + + ASTERISK-25224 + Reported by: Andrey Biglari + + Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b + +2015-06-29 12:45 +0000 [58d18324f0] Mark Michelson + + * res_sorcery_realtime: Fix leak of sorcery object type. + + This prevents a leak of a sorcery object type when realtime sorcery + objects are retrieved by fields or when multiple objects are retrieved. + + The extent of this leak is that sorcery object types would be leaked. + These are allocated whenever an object type is registered with sorcery, + meaning that on module shutdown, these objects would be leaked. This + could be problematic if many reloads were performed, but it is not as + severe as if every sorcery object retrieved from realtime were being + leaked. + + ASTERISK-25165 #close + Reported by Corey Farrell + + Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8 +2015-06-26 22:02 +0000 [80d97290bb] Matt Jordan + + * res/res_corosync: Always decline module load, instead of failing + + Returns a 'failure' from the module load routine indicates to Asterisk + that it should abort loading completely. This is rarely - in fact, + really, never - a good option. Aborting load of Asterisk from a dynamic + module implies that the core, and the rest of the dynamic modules, don't + matter: we should abandon all processing. + + res_corosync is really not that important. + + This patch updates the module such that, if it fails to load, it + politely declines (emitting ERROR messages along the way), and allows + Asterisk to continue to function. + + Note that this issue was keeping Asterisk unit tests from running on + certain build agents. + + Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f + +2015-06-26 20:38 +0000 [892cc5625f] Matt Jordan + + * main/pbx: Resolve case sensitivity regression in PBX hints + + When 8297136f was merged for ASTERISK-25040, a regression was introduced + surrounding the case sensitivity of device names within hints. + Previously, device names - such as 'sip/foo' - were compared in a case + insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After + that patch, only the case sensitive name would match, i.e., 'SIP/foo'. + As a result, some dialplan hints stopped working. + + This patch re-introduces case insensitive matching for device names in + hints. + + ASTERISK-25040 + + ASTERISK-25202 #close + + Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c + (cherry picked from commit 96bbcf495a1da9e607d9b04a44b5c4f49e83cc03) + +2015-06-26 16:12 +0000 [e18b22a806] Mark Michelson + + * res_pjsip_nat: Adjust when contact should be rewritten. + + A previous change made the contact only get rewritten if the dialog's + route set was not marked frozen. Unfortunately, while the intent of this + is correct, the dialog's route set actually gets marked as frozen + earlier than expected, especially for UAS dialogs. + + Instead, the idea is that the contact needs to not be rewritten if there + is a pre-existing route set on the dialog. This is now accomplished by + checking the dialog's route set list instead of checking if the route + set is frozen. + + Doing this causes some broken tests to begin passing again. + + ASTERISK-25196 + Reported by Mark Michelson + + Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e + +2015-06-19 18:27 +0000 [99b1aa6d26] Richard Mudgett + + * res_pjsip_outbound_registration.c: Add a serializer shutdown group. + + The client_state objects contain a serializer used to send the outbound + REGISTER messages. Once all those message transactions are complete then + the module can shutdown. + + ASTERISK-24907 #close + Reported by: Kevin Harwell + + Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547 + +2015-06-26 10:41 +0000 [f536e9b59c] Mark Michelson + + * res_pjsip_refer: Prevent sending duplicate headers. + + res_pjsip_refer will attempt to add Referred-By or Replaces headers to + outbound INVITEs at times. If the INVITE gets challenged for + authentication, then we will resend the INVITE. Prior to this patch, the + Referred-By or Replaces header would be re-added to the outbound INVITE, + resulting in duplicated headers. + + ASTERISK-25204 #close + Reported by Mark Michelson + + Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d + +2015-06-23 14:34 +0000 [c2d48a2a28] Richard Mudgett + + * AMI: Add Linkedid to the standard channel snapshot AMI event headers. + + ASTERISK-25189 #close + Reported by: John Hardin + + Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac +2015-06-23 17:43 +0000 [700606a659] Mark Michelson + + * res_pjsip_nat: Rewrite route set when required. + + When performing some provider testing, the rewrite_contact option was + interfering with proper construction of a route set when sending an ACK + after receiving a 200 OK response to an INVITE. + + The initial INVITE was sent to address sip:foo. The 200 OK had a Contact + header with URI sip:bar. In addition, the 200 OK had Record-Route + headers for sip:baz and sip:foo, in that order. Since the Record-Route + headers had the lr parameter, the result should have been: + + * Set R-URI of the ACK to sip:bar. + * Add Route headers for sip:foo and sip:baz, in that order. + + However, the rewrite_contact option resulted in our rewriting the + Contact header on the 200 OK to sip:foo. The result was: + + * R-URI remained sip:foo. + * We added Route headers for sip:foo and sip:baz, in that order. + + The result was that sip:bar was not indicated in the ACK at all, so the + far end never received our ACK. The call eventually dropped. + + The intention of rewrite_contact is to rewrite the most immediate + destination of our SIP request to be the same address on which we + received a request or response. In the case of processing a SIP response + with Record-Route headers, this means that instead of rewriting the + Contact header, we should instead rewrite the bottom-most Record-Route + header. In the case of processing a SIP request with Record-Route + headers, this means we rewrite the top-most Record-route header. + Like when we rewrite the Contact header, we also ensure to update + the dialog's route set if it exists. + + ASTERISK-25196 #close + Reported by Mark Michelson + + Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f +2015-06-19 16:16 +0000 [af4ae3095e] Richard Mudgett + + * threadpool, res_pjsip: Add serializer group shutdown API calls. + + A module trying to unload needs to wait for all serializers it creates and + uses to complete processing before unloading. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059 + +2015-06-16 15:06 +0000 [4c133d81cd] Richard Mudgett + + * res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs + + * handle_client_state_destruction() must always be passed a ref to + client_state because it will always unref client_state. + handle_registration_response() was not passing a client_state ref. + + * Made the final un-REGISTER message get sent normally using the pjproject + register control structure in handle_client_state_destruction(). The + previous code attempted to short circuit the response handling for the + module to unload. That doesn't work for a couple reasons. One, + pjsip_regc_send() may call the registered callback before it returns and + unbalance the client_state ref count. Two, the registered callback + handles any authentication for the un-REGISTER message. + + * Made the distinction between internal registration state and external + registration status with sip_outbound_registration_status_str(). This is + necessary to avoid altering documented AMI messages with internal + changes. + + * Removed references to client_state->client outside of the serializer + thread. When handle_client_state_destruction() destroys the pjproject + register control structure that memory is freed and cannot be referenced + anymore. These accesses were to provide information for debug and + off-nominal warning messages. + + * In sip_outbound_registration_timer_cb() you should not access entry->id + after unrefing client_state because the passed in entry is normally + pointing to the timer entry in the client_state object. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f + +2015-06-15 15:28 +0000 [dc63377c60] Richard Mudgett + + * res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API + + The sorcery pjsip 'registration' config object needs to be destroyed on + module unload. Otherwise, a reload of res_pjsip could try to use + callbacks for a previously unloaded instance of the module provided by + ast_sorcery_object_register() or one of the variants. Also, if + res_pjsip_outbound_registration were subsequently reloaded, the sorcery + config field objects would be registered in sorcery twice. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697 + +2015-06-15 15:28 +0000 [9ec8a0f3cc] Richard Mudgett + + * sorcery: Add ast_sorcery_object_unregister() API call. + + Find and unlink the specified sorcery object type to complement + ast_sorcery_object_register(). Without this function you cannot + completely unload individual modules that use sorcery for configuration. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88 + +2015-06-15 13:38 +0000 [77ff7325a2] Richard Mudgett + + * res_pjsip_outbound_registration.c: Reorder load_module() and unload_module(). + + It is best if the loading code creates and initializes the module's + infrastructure before letting the system know of its existence. The + unloading code needs to reverse the actions of the loading code and in the + reverse order. + + ASTERISK-24907 + Reported by: Kevin Harwell + + Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4 + +2015-06-25 06:42 +0000 [8d6cf667dc] Joshua Colp + + * channel: Remove ignore of answer on non-outgoing channels. + + Due to the way that channels can now be moved around inside of + Asterisk it is possible for the outgoing flag of a channel to get + cleared before it has been answered. This results in the bridge + not receiving notification that the outgoing leg has been answered. + + This most easily exhibits itself with DTMF based blond transfers. + Since the answer of the outgoing leg is ignored the other party + continues to receive both a locally generated ringing and the + media stream of the outgoing leg upon its answer. This results + in no media being heard. + + This change removes the ignore of the answer and allows it + to pass through. + + ASTERISK-25171 #close + + Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e + +2015-06-24 14:30 +0000 [daaa551c92] Richard Mudgett + + * test.c: Add unit test registration checks for summary and description. + + Added checks when a unit test is registered to see that the summary and + description strings do not end with a new-line '\n' for consistency. + + The check generates a warning message and will cause the + /main/test/registrations unit test to fail. + + * Updated struct ast_test_info member doxygen comments. + + Change-Id: I295909b6bc013ed9b6882e85c05287082497534d + +2015-06-24 16:39 +0000 [71a4d1a033] Richard Mudgett + + * Unit tests: Fix more unit test description strings. + + Analyzing the code shows that the unit test summary and description + strings should not end with a new-line character. Where these strings are + used in the code a new-line is provided for output. + + Change-Id: I2f4f37988ec363c8d1c5077a2fc8ca841c5cd30c + +2015-06-24 14:39 +0000 [9c6d72e30d] Richard Mudgett + + * Unit tests: Fix unit test description strings. + + Analyzing the code shows that the unit test summary and description + strings should not end with a new-line character. Where these strings are + used in the code a new-line is provided for output. + + Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba + +2015-06-24 16:37 +0000 [a0c2d2089d] Richard Mudgett + + * DNS unit tests: Fix extraneous description string commas. + + Change-Id: Icf5f13c8e1c2c92a4473bb573ed2dd856ce1b64e + +2015-06-23 11:21 +0000 [3b2b004d69] Joshua Colp + + * app_dial: Hold reference to calling channel formats when dialing outbound. + + Currently when requesting a channel the native formats of the + calling channel are provided to the core for usage when dialing + the outbound channel. This occurs without holding the channel lock + or keeping a reference to the formats. This is problematic as + the channel driver may end up changing the formats during this time. + In the case of chan_sip this happens when an SDP negotiation + completes. + + This change makes it so app_dial keeps a reference to the native + formats of the calling channel which guarantees that they will + remain valid for the period of time needed. + + ASTERISK-25172 #close + + Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db + +2015-06-17 16:23 +0000 [af66b0f3f7] Richard Mudgett + + * res_pjsip_outbound_registration.c: Add missing line endings to CLI commands + + Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7 + +2015-06-12 14:29 +0000 [3f0708e5fe] Richard Mudgett + + * res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage. + + Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e + +2015-06-12 13:33 +0000 [9ceb848242] Richard Mudgett + + * res_pjsip_outbound_registration.c: Misc code cleanups. + + * Break some long lines. + + * Fix doxygen comment. + + Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305 + +2015-06-22 15:11 +0000 [44c3c392e3] Kevin Harwell + + * bridge.c: Hangup attended transfer target if bridged + + After completing an attended transfer the transfer target channel was not being + hung up after leaving the bridge. Added an explicit softhangup to hangup said + channel, but only if it was previously bridged. + + ASTERISK-24782 #close + Reported by: John Bigelow + + Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada + +2015-06-17 05:04 +0000 [7846f73432] Joshua Colp + + * res_pjsip_mwi: Set up unsolicited MWI upon registration. + + The res_pjsip_mwi previously required a reload to set up the proper + subscriptions to allow unsolicited MWI to work. This change + makes it so the act of registering will also cause this to occur. + This is particularly useful if realtime is involved as no reload + needs to occur within Asterisk to cause the MWI information + to get sent. + + ASTERISK-25180 #close + + Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252 + +2015-06-22 13:57 +0000 [096b27d9d2] Richard Mudgett + + * res_pjsip_outbound_registration.c: Fix whitespace conflict potential. + + Change-Id: I82e6e388e3688aebe0783f16c9e0800a747584b5 + +2015-06-22 09:26 +0000 [1ad9a6b6b6] Alexander Traud (License 6520) + + * chan_sip: Reload peer without its old capabilities. + + On reload, previously allowed codecs were not removed. Therefore, it was not + possible to remove codecs while Asterisk was running. Furthermore, newly added + codecs got appended behind the previous codecs. Therefore, it was not possible + to add a codec with a priority of #1. This change removes the old capabilities + before the current ones are added. + + ASTERISK-25182 #close + Reported by: Alexander Traud + patches: + asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) + + Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802 + +2015-06-20 19:38 +0000 [5caefc98a1] Joshua Colp + + * chan_sip: Destroy peers without holding peers container lock. + + Due to the use of stasis_unsubscribe_and_join in the peer destructor + it is possible for a deadlock to occur when an event callback is + occurring at the same time. + + This happens because the peer may be destroyed while holding the + peers container lock. If this occurs the event callback will never + be able to acquire the container lock and the unsubscribe will + never complete. + + This change makes it so the peers that have been removed from the + peers container are not destroyed with the container lock held. + + ASTERISK-25163 #close + + Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33 + +2015-06-18 13:16 +0000 [d7a1e84a1e] Mark Michelson + + * Resolve race conditions involving Stasis bridges. + + This resolves two observed race conditions. + + First, a bit of background on what the Stasis application does: + + 1a Creates a stasis_app_control structure. This structure is linked into + a global container and can be looked up using a channel's unique ID. + 2a Puts the channel in an event loop. The event loop can exit either + because the stasis_app_control structure has been marked done, or + because of some other factor, such as a hangup. In the event loop, the + stasis_app_control determines if any specific ARI commands need to be + run on the channel and will run them from this thread. + 3a Checks if the channel is bridged. If the channel is bridged, then + ast_bridge_depart() is called since channels that are added to Stasis + bridges are always imparted as departable. + 4a Unlink the stasis_app_control from the container. + + When an ARI command is received by Asterisk, the following occurs + 1b A thread is spawned to handle the HTTP request + 2b The stasis_app_control(s) that corresponds to the channel(s) in the + request is/are retrieved. If the stasis_app_control cannot be + retrieved, then it is assumed that the channel in question has exited + the Stasis app or perhaps was never in Stasis in the first place. + 3b A command is queued onto the stasis_app_control, and the channel's + event loop thread is signaled to run the command. + 4b While most ARI commands do nothing further, some, such as adding or + removing channels from a bridge, will block until the command they + issued has been completed by the channel's event loop. + + The first race condition that is solved by this patch involves a crash + that can occur due to faulty detection of the channel's bridged status + in step 3a. What can happen is that in step 2a, the event loop may run + the ast_bridge_impart() function to asynchronously place the channel + into a bridge, then immediately exit the event loop because the channel + has hung up. In step 3a, we would detect that the channel was not + bridged and would not call ast_bridge_depart(). The reason that the + channel did not appear to be bridged was that the depart_thread that is + spawned by ast_bridge_impart() had not yet started. That is the thread + where the channel is marked as being bridged. Since we did not call + ast_bridge_depart(), the Stasis application would exit, and then the + channel would be destroyed Then the depart_thread would start up and + try to manipulate the destroyed channel, causing a crash. + + The fix for this is to switch from using ast_channel_is_bridged() to + checking the NULLity of ast_channel_internal_bridge_channel() to + determine if ast_bridge_depart() needs to be called. The channel's + internal bridge_channel is set when ast_bridge_impart() is called and + is NULLed by the call to ast_bridge_depart(). If the channel's internal + bridge_channel is non-NULL, then the channel must have been imparted + into the bridge and needs to be departed, even if the actual bridging + operation has not yet started. By departing the channel when necessary, + the thread that is running the Stasis application will block until the + bridge gives the okay that the depart_thread has exited. + + The second race condition that is solved by this patch involves a leak + of HTTP handler threads. The problem was that step 2b would successfully + retrieve a stasis_app_control structure. Then step 2a would exit the + channel from the event loop due to a hangup. Steps 3a and 4a would + execute, and then finally steps 3b and 4b would. The problem is that at + step 4b, when attempting to add a channel to a bridge, the thread would + block forever since the channel would never execute the queued command + since it was finished with the event loop. This meant that the HTTP + handling thread would be leaked, along with any references that thread + may have owned (in my case, I was seeing bridges leaked). + + The fix for this is to hone in better on when the channel has exited the + event loop. The stasis_app_control structure has an is_done field that + is now set at each point where the channel may exit the event loop. If + step 2b retrieves a valid stasis_app_control structure but the control + is marked as done, then the attempted operation exits immediately since + there will be nothing to service the attempted command. + + ASTERISK-25091 #close + Reported by Ilya Trikoz + + Change-Id: If66265b73b4c9f8f58599124d777fedc54576628 +2015-06-17 07:00 +0000 [9668a1acb5] Joshua Colp + + * res_sorcery_memory_cache: Remove 'prefetch' option. + + To prevent confusion I am removing the prefetch option until such + time as it is implemented. All other functionality, however, has + been implemented. + + ASTERISK-25067 + + Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895 + +2015-06-16 11:13 +0000 [59552c2d08] Mark Michelson + + * Parking: Add documentation for AMI ParkedCallSwap event. + + This event was added some time ago in order to clarify when a channel + took the place of another channel in a parking lot. However, there was + no XML documentation added for the event. This patch adds the XML + documentation. + + ASTERISK-24900 #close + Reported by Rusty Newton + + Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac +2015-06-15 16:40 +0000 [ea9d5f155e] Corey Farrell + + * func_pjsip_aor: Fix leaked contact from iterator. + + ASTERISK-25162 #close + + Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e + +2015-06-12 16:58 +0000 [93ac45d3bd] Kevin Harwell + + * res_pjsip: Add option to force G.726 to be treated as AAL2 packed. + + Some phones send g.726 audio packed for AAL2, which differs from what is + recommended by RFC 3351. If Asterisk receives audio formatted as such when + negotiating g.726 then it sounds a bit distorted. Added an option to + res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 + AAL2 packed. + + ASTERISK-25158 #close + Reported by: Steve Pitts + + Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615 + +2015-06-14 19:48 +0000 [15c2208701] Matt Jordan + + * main/cdr: Carry over the disable flag when 'disable all' is specified + + The CDR_PROP function (as well as the NoCDR application) set the + 'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This + flag is supposed to be applied to all CDRs that are currently in the + chain, as well as all CDRs that may be created in the future. Currently, + however, the flag is only applied to the existing CDRs in the chain; new + CDRs do not receive the 'disable all' flag. In particular, this affects + parallel dials, which generate new CDRs for each pair of channels in + the dial attempt. + + This patch carries over the 'disable all' flag when it is specified on a + CDR and a new CDR is generated for the chain. + + ASTERISK-24344 #close + + Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e +2015-06-12 14:28 +0000 [b8bc15286f] Matt Jordan + + * main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines + + When a parallel dial occurs, a new CDR will be created for each dial + attempt that is made. In most circumstances, the act of creating each + CDR in the chain will include a step that updates the Party A snapshot, + which causes the context/extension of the Party A to be copied onto the + CDR object. + + However, when the Party A is in a subroutine, we explicitly do *not* + copy the context/extension onto the CDR. This prevents the Macro or + GoSub routine name from blowing away the context/extension that the + channel was originally executing in. For the original CDR, this is not a + problem: the original CDR already recorded the last known 'good' state + of the channel just prior to it going into the subroutine. However, for + newly generated CDRs in a chain, there is no context/extension set on + them. Since we are in a subroutine, we will never set the Party A's + context/extension on the CDR, and we end up with a CDR with no + destination recorded on it. + + This patch updates the creation of a chained CDR such that it copies + over the original CDR's context/extension. This is the last known "good" + state of the CDR, and is a reasonable starting point for the newly + generated CDR. In the case where we are not in a subroutine, subsequent + code will update the location of the CDR from the Party A information; + in the case where we are in a subroutine, the context/extension on the + original CDR is the correct information. + + ASTERISK-24443 #close + + Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a + +2015-06-11 08:18 +0000 [19f60d9412] Damian Ivereigh + + * chan_sip.c: Update dialog fromtag after request with auth + + If a client sends and INVITE which is 401 rejected, then subsequently + sends a new INVITE with the auth info and uses a different fromtag + from the first INVITE, Asterisk will accept the new INVITE as part of + the original dialog - match_req_to_dialog() specifically ignores the + fromtag. However it does not update the stored dialog with the new + fromtag. + + This results in Asterisk being unable to match future packets that are + part of this dialog (such as the ACK to the OK or the OK to the BYE), + and the call is dropped. + + This problem was originally found when using an NEC-i SV8100-GE (NEC SIP + Card). + + * After a successful match of a packet to the dialog, if the packet is + not a SIP_RESPONSE, authentication is present and the fromtags are + different, the stored fromtag is updated with the one from the recent + INVITE. + + ASTERISK-25154 #close + Reported by: Damian Ivereigh + Tested by: Damian Ivereigh + + Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e + +2015-06-11 18:52 +0000 [bb00b26f35] Matt Jordan + + * chan_pjsip: Set the context and extension on the channel when created + + Prior to this patch, chan_pjsip was failing to pass the endpoint's + context and the desired extension to the ast_channel_alloc_* routine. + This caused a new channel snapshot to be issued without a context and + extension, which can cause some reporting issues for users of AMI, CEL, + and other APIs. The channel driver would later set the context and + extension on the channel such that the channel would start in the + correct location in the dialplan, but the information reported in the + initial event would be incorrect. + + This patch modifies the channel driver such that it now passes the + context and extension directly into the allocation routine. This + provides the information in the new channel snapshot published over + Stasis. + + ASTERISK-25156 #close + Reported by: cloos + + Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e + +2015-06-10 18:28 +0000 [7230ee2efe] Joshua Colp + + * bridge: When performing a blonde transfer update connected line information. + + When performing a blonde transfer the code uses the old masquerade + mechanism to move a channel around. As a result of this certain information, + such as connected line, is moved between the channels involved. Upon + completion of the move a frame is queued which is supposed to update the + connected line information on the channel. This does not occur as the + code considers it a redundant update since the masquerade operation + updated the channel (but did not inform it of the new connected line + information). The code also does not queue a connected line update + to be handled by the thread handling the channel. Without this any + other channel that may be loosely involved does not know it is + talking to a different caller. + + This change does the following to resolve this: + + 1. The indicated connected line information is cleared upon + completion of the masquerade operation when doing a blonde transfer. + This prevents the connected line update from being considered + redundant. + + 2. A connected line update frame is now queued upon the completion + of the masquerade operation so any other channel loosely involved + knows that there is a different caller. + + ASTERISK-25157 #close + Reported by: Joshua Colp + + Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20 + +2015-06-11 14:39 +0000 [a657ab12f9] Richard Mudgett + + * app_directory: Fix crash when using the alias option 'a'. + + The voicemail.conf mailbox key/value pair is defined as: + =[[,[,[,[,]]]]] + Where all fields in the value including the field values are optional. + + Since the parsing code for the mailbox key/value pair is sloppy, this + patch tightens the parsing for the directory information. + + * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' + respectively in search_directory_sub(). Those names make more sense. + + * Made sure that search_directory_sub() is dealing with the voicemail.conf + mailbox options field if it even exists when looking for the 'hidefromdir' + and 'alias' options. + + * Fix crash if a voicemail.conf mailbox is just + =, when the 'a' option is used. If there were no + fields after the name then the 'options' pointer was not checked for NULL. + + * Fix users.conf alias processing if the 'a' option is used. The wrong + variable was used. + + ASTERISK-25087 #close + Reported by: Chet Stevens + + Change-Id: I86052ea77307beddddba5279824d39dc0d593374 + +2015-06-05 15:37 +0000 [30cd559345] Richard Mudgett + + * DNS: Need to use the same serializer for a pjproject SIP transaction. + + All send/receive processing for a SIP transaction needs to be done under + the same threadpool serializer to prevent reentrancy problems inside + pjproject when using an external DNS resolver to process messages for the + transaction. + + * Add threadpool API call to get the current serializer associated with + the worker thread. + + * Pick a serializer from a pool of default serializers if the caller of + res_pjsip.c:ast_sip_push_task() does not provide one. + + This is a simple way to ensure that all outgoing SIP request messages are + processed under a serializer. Otherwise, any place where a pushed task is + done that would result in an outgoing out-of-dialog request would need to + be modified to supply a serializer. Serializers from the default + serializer pool are picked in a round robin sequence for simplicity. + + A side effect is that the default serializer pool will limit the growth of + the thread pool from random tasks. This is not necessarily a bad thing. + + * Made pjsip_resolver.c use the requesting thread's serializer to execute + the async callback. + + * Made pjsip_distributor.c save the thread's serializer name on the + outgoing request tdata struct so the response can be processed under the + same serializer. + + ASTERISK-25115 #close + Reported by: John Bigelow + + Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a + +2015-06-05 12:16 +0000 [b23f33e7e5] Richard Mudgett + + * DNS: Fix some corner cases. + + * Fix query_set destruction before we are done kicking the queries off. + + * Fixed no queries requested handling. + + * Add empty queries request unit test. + + * Added missing allocation check in ast_dns_query_set_add(). + + * Made initial pjsip resolving query vector slightly larger. + + ASTERISK-25115 + Reported by: John Bigelow + + Change-Id: Ie8be8347d0992e93946d72b6e7b1299727b038f2 + +2015-06-10 17:51 +0000 [ae589da466] Richard Mudgett + + * DNS: Remove trailing newline from summary and descriptions. + + Those trailing newlines mess up test formatting. + + Change-Id: I5e3f3a55b82c9d7acb9661201d4993d1958f1185 + +2015-06-05 11:43 +0000 [83bc9d366d] Richard Mudgett + + * pjsip_resolver.c: Fix debug code to only execute at acceptable debug level. + + Change-Id: I1716c93d6e097ad28128ecb9e806aac7a4180c8a + +2015-06-05 11:41 +0000 [6d49dccd85] Richard Mudgett + + * DNS: Fix doxygen comments. + + Change-Id: Icafea3fb4ea64ac027561b23cbfe2b17997dc549 + +2015-06-09 15:31 +0000 [b705c09dbb] Richard Mudgett + + * res_pjsip.h: Fix some doxygen comments. + + Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976 + +2015-06-05 13:46 +0000 [aa8479778e] Richard Mudgett + + * taskprocessor.c: Remove extra unref from off-nominal path. + + Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60 + +2015-05-31 12:37 +0000 [07f5f45e5a] Ivan Poddubny + + * res_pjsip_transport_websocket: Fix use-after-free bugs. + + This patch fixes use-after-free bugs caught by AddressSanitizer. + + 1. PJSIP transport manager may decide to destroy transport on its own. + For example, when the contact registered via websocket has not renewed + its registration in time. The transport was destoyed, but the websocket + listener thread was still active until the socket closes, and then tried + to call transport_shutdown on transport that has been freed. + + Also, the transport destructor accessed wstransport->rdata.tp_info.pool + right after freeing memory that contained wstransport itself. + + This patch converts transport to an ao2 object, allowing it to be + refcounted, so that it is available until both websocket listener and + pjsip transport manager are finished with it. + + 2. The websocket listener deletes the last reference on websocket session + when the tcp connection is closed, and it gets destroyed, but + the transport manager may still use it, for example when disconnect + happens in the middle of a SIP transaction. + + A new reference to websocket session has been added that is released + with the transport to prevent this. + + ASTERISK-25096 #close + Reported by: Josh Kitchens + + ASTERISK-24963 #close + Reported by: Badalian Vyacheslav + + Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b + +2015-06-09 13:41 +0000 [f897f36721] ibercom + + * weakref attribute detection broken with gcc 4.6 and higher + + GCC 4.7 Manual: + http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html + + weakref ("target") + + A weak reference is an alias that does not by itself require a definition + to be given for the target symbol. + + ASTERISK-22559 #close + Reported by: Ibercom + + Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf + +2015-06-08 10:09 +0000 [80621ce3c5] Corey Farrell + + * Fix unsafe uses of ast_context pointers. + + Although ast_context_find, ast_context_find_or_create and + ast_context_destroy perform locking of the contexts table, + any context pointer can become invalid at any time that the + contexts table is unlocked. This change adds locking around + all complete operations involving these functions. + + Places where ast_context_find was followed by ast_context_destroy + have been replaced with calls ast_context_destroy_by_name. + + ASTERISK-25094 #close + Reported by: Corey Farrell + + Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa + +2015-06-08 09:44 +0000 [53c1126090] Kevin Harwell + + * AMI: Escape string values. + + So this issue is a bit complicated. Since it is possible to pass values to AMI + that contain a '\r\n' (or other similar sequences) these values need to be + escaped. One way to solve this is to escape the values and then pass the escaped + values to the AMI variable parameter string building function. However, this + puts the onus on the pre-build function to escape all string values. This + potentially requires a fair amount of changes along with a lot of string + allocations/freeing for all values. + + Surely there is a way to push this complexity down a level into the string + building function itself? This of course is possible, but ends up requiring a + way to distinguish between strings that need to be escaped and those that don't. + The best way to handle this is by introducing a new format specifier in the + format string. For instance a %s (no escape) and %S (escape). However, that is + a bit weird and unexpected. + + So faced with those possibilities this patch implements a limited version of the + first option. Instead of attempting to escape all string values this patch only + escapes those values that make sense. This approach limits the number of changes + and doesn't suffer from the odd format specifier problem. + + ASTERISK-24934 #close + Reported by: warren smith + + Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0 + +2015-06-02 15:07 +0000 [9fca378b36] David M. Lee + + * Fixes for OS X + + * Add some type casting so tv_usec can really be a long, instead of + some strange platform specific type. + + * Add some .dylib style files to .gitignore. + + * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer + versions of GCC, when compiling the Homebrew formula for Asterisk, + are not properly passing the -Xlinker options to the linker. Given + that -Wl, does exactly the [same thing][], and does it properly, this + patch changes the -Xlinker options to use -Wl, instead. + + [reasons unknown]: http://bit.ly/1SUbEYx + [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html + + Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd + +2015-06-04 07:14 +0000 [d463bac574] ibercom + + * CLI: Cosmetic issue - core show uptime + + Show uptime information ends with an unnecessary space. + + Now NEEDCOMMA is better defined. + + Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1 + +2015-06-04 13:11 +0000 [128fe4cee8] Joshua Colp + + * res_sorcery_memory_cache: Implement expire_on_reload option. + + This change implements the expire_on_reload option for memory caches. + If enabled and a reload is performed all objects within the cache + will be expired and the cache emptied. + + ASTERISK-25067 + Reported by: Matt Jordan + + Change-Id: Id46aa1957d660556700e689e195eed57c989b85e + +2015-06-02 10:20 +0000 [028edae82e] Joshua Colp + + * test_sorcery_memory_cache_thrash: Add unit tests for thrashing the memory cache. + + This change adds a CLI command which can perform memory cache thrashing as well + as unit tests which perform thrashing under the following configurations: + + 1. Low number of unique objects that go stale after 1 second + 2. Low number of unique objects that expire after 1 second + 3. Low number of unique objects which are constantly updated + 4. Large number of unique objects which exceed a defined cache size + 5. Large number of unique objects which exceed a defined cache size + that also expire and go stale rapidly + 6. Large number of unique objects which expire and go stale rapidly + 7. Large number of unique objects + + For all of the above there are a large number of threads constantly + attempting to retrieve random objects and each test runs for a few + seconds. + + ASTERISK-25067 + Reported by: Matt Jordan + + Change-Id: I8c8ceff977332c80ed4a31f10d694d48552b2f78 + +2015-06-04 05:33 +0000 [19de2bbc5f] Joshua Colp + + * res_sorcery_memory_cache: Add test event when a refresh occurs. + + This change adds a testsuite event for when a refresh occurs. + This is useful as it provides a guaranteed mechanism of knowing when + it has occurred instead of waiting an arbitrary amount of time. + + ASTERISK-25067 + Reported by: Matt Jordan + + Change-Id: Iaa6b8d2d6bab7f99ee08e1c8908b8272a8987e65 + +2015-06-03 20:12 +0000 [6737ded058] Rodrigo Ramírez Norambuena + + * install_prereq: Check if is installed aptitude otherwise to install. + + If in Debian or system based, dont have aptitude installed the script do + nothing. This patch checked if aptitude installed, if not installed. + + Also, if execute script with all packages installed yet, the script not show + nothing and return exit 1 because the command 'grep' get nothing from pipe from + 'awk'. + + ASTERISK-25113 #close + Reported By: Rodrigo Ramírez Norambuena + + Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f + +2015-06-03 17:41 +0000 [92ccffd9e6] Mark Michelson + + * res_pjsip: Prevent access of NULL channels. + + It is possible to receive incoming requests or responses after the channel + on an ast_sip_session has been destroyed and NULLed out. Handlers of these + sorts of requests or responses need to be prepared for the possibility + that the channel is NULL or else they could cause a crash. + + While several places have been amended to deal with NULL channels, there + were still a couple of places that needed updating. + + res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to + return early if there is no channel on the session. + + res_pjsip_session.c: When handling a 302 response, we need to stop the + redirecting attempt if there is no channel on the session. + + ASTERISK-25148 #close + reported by Mark Michelson + + Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9 + +2015-06-03 13:17 +0000 [d355ee7ff3] gtjoseph + + * res_pjsip/location: Fix ref leak in contact_apply_handler + + contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status + to force the creation of a contact_status object whenever a new + contact is added but it didn't unref the returned object. + + Added an ao2_cleanup(status) to plug the leak. + + ASTERISK-25141 + + Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 + Reported-by: Corey Farrell + +2015-06-02 13:02 +0000 [6d8dc9bb5c] Richard Mudgett + + * res_pjsip: Remove outgoing authentication code no longer needed. + + Associated with ASTERISK-25131 + + Change-Id: Iefa3b2066cfd8b108a90d2dd4a64d92c3a195d33 + +2015-06-02 12:55 +0000 [00a47ffc7e] Richard Mudgett + + * res_pjsip_session: Fix cherry pick to master compile error. + + ASTERISK-25131 + Reported by: Richard Mudgett + + Change-Id: I87c9c96ae4a8fe2bc8a0ddea6958a2ad9cefd8e3 + +2015-06-02 12:27 +0000 [9472bbaa95] Joerg Sonnenberger + + * Remove const cast from leaf functions. + + app_control_register_rule and app_control_unregister_rule lock/unlock + the queue, which is a mutating operation according to the + ao2_lock/_unlock prototype. Depending on the specific (implicit) casts + in SCOPED_LOCK and RAII_VAR, the compiler may warn or not. As the only + callers of those functions do not have the const, get consistent results + by just dropping it. + + Change-Id: Ib9e6296155a39bc5d627142a3828180c3cfe8fbb + +2015-06-02 11:35 +0000 [5f712e82ac] Joerg Sonnenberger + + * tcptls.c: Don't use OpenSSL functions when no SSL support is present. + + Change-Id: I68a85a7fcbdb282140ff333c6274b6763d5f82a3 +2015-06-01 12:08 +0000 [2cd40c2bd7] Rodrigo Ramírez Norambuena + + * cdr/cdr_csv.c: Set file name for csv master to the module when (re)loaded. + + Compute the location for the csv master file when the module is + loaded or reload. Before it was calculated every time a log + entry was written. + + Change-Id: I3ed9f6a8f965308099db70b71128f43d4d3f5585 +2015-05-26 13:56 +0000 [5cdcae5240] Richard Mudgett + + * res_pjsip_session: Fix in-dialog authentication. + + When the remote peer requires authentication for in-dialog requests then + re-INVITEs to the peer cause the call to be disconnected and other + in-dialog requests to the peer like MESSAGE just don't go through. + + * Made session_inv_on_tsx_state_changed() handle in-dialog authentication + for re-INVITEs and other methods. Initial INVITEs cannot be handled here + because the INVITE transaction must be restarted earlier. + + * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in + preparation for removing the file. The generic outbound authentication + code did not work as well as anticipated. + + * Created outbound_invite_auth() to only handle initial outbound INVITEs. + Re-INVITEs cannot be handled here. The re-INVITE transaction is still in + progress and the PJSIP library cannot handle the overlapping INVITE + transactions. Other method types should not be handled here as this code + only works on outgoing calls and we need to handle incoming and outgoing + calls. + + ASTERISK-25131 #close + Reported by: Richard Mudgett + + Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0 + +2015-05-30 20:22 +0000 [9f1939ee27] Corey Farrell + + * pjsip_configuration: Fix leak in persistent_endpoint_update_state. + + The loop to find the first available contact of an endpoint grabbed + contact from the iterator, then checked for offline state. This + caused the first contact after the state was found to leak a reference. + + ASTERISK-25141 + + Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08 + +2015-05-31 11:33 +0000 [0a5f8c0d73] Ivan Poddubny + + * Fix buffer overflow in slin sample frames generation. + + The length of frames retured by sample functions was twice as large as + real, what caused global buffer overflow caught by AddressSanitizer. + + ASTERISK-24717 #close + Reported by: Badalian Vyacheslav + + Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6 + +2015-05-29 16:19 +0000 [bef000dd7c] gtjoseph + + * res_pjsip/location: Fix memory leak in permanent_uri_handler + + When permanent_uri_handler was creating the contact status + object for each contact, it wasn't unreffing it at the + end of the loop. + + ASTERISK-25141 #close + Reported-by: Corey Farrell + + Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12 + +2015-05-29 14:52 +0000 [82716410a4] gtjoseph + + * Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change" + + This reverts commit 6fca75bb628dfff2ab112e80b0228cf3ac0b8a05. + + Change-Id: Ifee026cc63e22c5ac5717c37867a9f036373ae5a + +2015-05-26 07:34 +0000 [dfc45254d1] Joshua Colp + + * res_sorcery_memory_cache: Add CLI commands and AMI actions. + + This change adds the following CLI commands and AMI actions: + + sorcery memory cache show + sorcery memory cache dump + sorcery memory cache expire + sorcery memory cache stale + + SorceryMemoryCacheExpire + SorceryMemoryCacheExpireObject + SorceryMemoryCacheStale + SorceryMemoryCacheStaleObject + + These allow both examination and manipulation of sorcery memory + caches from external sources. + + Cached objects can be explicitly expired from a cache or marked + as stale. If expired they are immediately removed. If marked as + stale they will be background refreshed when next retrieved. + + ASTERISK-25067 + Reported by Matt Jordan + + Change-Id: I68e03cfd8c34b5e07f4b6ee4fd93a3f4a00a3d9e + +2015-05-27 13:22 +0000 [6fca75bb62] gtjoseph + + * endpoint/stasis: Eliminate duplicate events on endpoint status change + + When an endpoint was created, it's messages were being forwarded to + both the tech endpoint topic and the all endpoints topic. Since + the tech topic was also forwarded to all, this was resulting in + duplicate messages whenever an endpoint published. This patch + causes the endpoint to only forward to the tech topic and lets + the tech topic forward to all. + + To accomplish this, the existing stasis_cp_single_create function + (which both creates and forwards) was cloned and split into 2 + functions, one that creates the topic and one that sets up the + forwarding. This allows endpoint_internal_create to create + the topic from the endpoint_all cache without forwarding it there, + then allows it to do the forward to the tech's topic. + + ASTERISK-25137 #close + Reported-by: Vitezslav Novy + ASTERISK-25116 #close + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c + +2015-05-26 13:01 +0000 [2e54e7227c] Mark Michelson + + * res_sorcery_memory_cache: Add support for refreshing stale objects. + + This change introduces a check of object_lifetime_stale when retrieving + cached objects. If the amount of time the object has been in the cache + exceeds the lifetime, then a task is scheduled to update the cached + object based on an object retrieved from other sorcery wizards instead. + + To prevent the cached object from being retrieved during a refresh, + thread-local storage is used to mark the thread as being a stale object + update. This results in the cache returning no object, leading to + sorcery querying other wizards for the object instead. + + A test has been added for stale objects as well. This test ensures that + stale objects are retrieved the same as freshly-cached objects. The test + also ensures that after an object is stale, changes in the backend are + reflected in the cache, to include if the object has been deleted from + the backend. + + ASTERISK-25067 + Reported by Matt Jordan + + Change-Id: I9bd7c049adf6939bfe2899f393c2bfbbf412d217 +2015-05-21 17:21 +0000 [b8ac683822] gtjoseph + + * res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes + + Add a new ContactStatus AMI event. + Publish the following status/state changes: + Created + Removed + Reachable + Unreachable + Unknown + + Contact URI, new status/state, aor and endpoint names, and the + last qualify rtt result are included in the event. + + ASTERISK-25114 #close + + Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e + Reported-by: George Joseph + Tested-by: George Joseph + +2015-05-07 11:18 +0000 [95b186a174] Rodrigo Ramírez Norambuena + + * res/res_config_pgsql.c: Use PQescapeStringConn for escaping names. + + Use function PQescapeStringConn for escaping the name of the table and + schema instead of doing it manually. + + ASTERISK-25132 #close + Reported By: Rodrigo Ramírez Norambuena + + Change-Id: I302a263f7210d20925f14716b508b081998b7608 + +2015-05-26 07:44 +0000 [a7af6bca3c] Joshua Colp + + * sorcery: Fix cache creation callback. + + The cache creation callback function expects to receive a sorcery_details + structure and not just a standalone object. + + Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450 + +2015-05-24 13:47 +0000 [23a798fecc] Ivan Poddubny + + * Astobj2: Correctly treat hash_fn returning INT_MIN + + The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0. + However, abs(INT_MIN) = INT_MIN and is still negative, as well as + abs(INT_MIN) % num_buckets, and as a result this led to a crash. + + One way to trigger the bug is using host=::80 or 0.0.0.128 in peer + configuration section in chan_sip or chan_iax. + + This patch takes the remainder before applying abs, so that bucket + number is always in range. + + ASTERISK-25100 #close + Reported by: Mark Petersen + + Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899 +2015-05-23 04:36 +0000 [70d54ab6c4] Ivan Poddubny + + * res_pjsip_transport_websocket: Fix crash on receiving large SIP packets + + Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves + truncated before passing to pjsip_tpmgr_receive_packet, but the length + was passed unaltered, thus causing memory corruption and segfault. + + ASTERISK-25122 #close + + Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab + +2015-05-22 21:50 +0000 [50044fdc15] Corey Farrell + + * Stasis: Fix unsafe use of stasis_unsubscribe in modules. + + Many uses of stasis_unsubscribe in modules can be reached through unload. + These have been switched to stasis_unsubscribe_and_join. + + Some subscription callbacks do nothing, for these I've created a noop + callback function in stasis.c. This is used by some modules that monitor + MWI topics in order to enable cache, since the callback does not become + invalid after dlclose it is safe to use stasis_unsubscribe on these, even + during module unload. + + ASTERISK-25121 #close + + Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c + +2015-05-22 16:52 +0000 [5a1f2a5884] Corey Farrell + + * Astobj2: Run weakproxy subscription callbacks in reverse order. + + Modify ao2_weakproxy_subscribe so each new subscription is added + to the head of the list. This ensures that when other objects + are allocated and use a subscription to the weakproxy for cleanup, + cleanup will occur in the correct order. + + ASTERISK-25120 #close + + Change-Id: Ie0476f08ec21330de1b3f5a2dd3d9eb683df3d3d + +2015-05-22 12:22 +0000 [f66c41e668] Matt Jordan + + * res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS + + In addition to specifying lists of 'presence' and 'message-summary', + users can also create lists of type 'dialog'. These should be treated in + the same fashion as 'presence'. + + Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e + +2015-05-22 12:18 +0000 [ad7192a8fd] Matt Jordan + + * res/res_pjsip_exten_state: Fix confusing NOTICE message + + When a SUBSCRIBE request is made to a dialplan hint that doesn't exist, + the current NOTICE message informing users of this swaps the context and + extension parameters. This can cause a bit of confusion. + + Thanks to CptBurger in #asterisk for helping to point this out. + + Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43 + +2015-05-17 20:36 +0000 [9cffcca5f9] Matt Jordan + + * res/ari: Register Stasis application on WebSocket attempt + + Prior to this patch, when a WebSocket connection is made, ARI would not + be informed of the connection until after the WebSocket layer had + accepted the connection. This created a brief race condition where the + ARI client would be notified that it was connected, a channel would be + sent into the Stasis dialplan application, but ARI would not yet have + registered the Stasis application presented in the HTTP request that + established the WebSocket. + + This patch resolves this issue by doing the following: + * When a WebSocket attempt is made, a callback is made into the ARI + application layer, which verifies and registers the apps presented in + the HTTP request. Because we do not yet have a WebSocket, we cannot + have an event session for the corresponding applications. Some + defensive checks were thus added to make the application objects + tolerant to a NULL event session. + * When a WebSocket connection is made, the registered application is + updated with the newly created event session that wraps the WebSocket + connection. + + ASTERISK-24988 #close + Reported by: Joshua Colp + + Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636 + +2015-05-20 11:11 +0000 [29ef6571cb] gtjoseph + + * res_pjsip: Refactor endpt_send_transaction (qualify_timeout) + + This patch refactors the transaction timeout processing to eliminate + calling the lower level public pjsip functions and reverts to calling + pjsip_endpt_send_request again. This is the result of me noticing + a possible incompatibility with pjproject-2.4 which was causing + contact status flapping. + + The original version of this feature used the lower level calls to + get access to the tsx structure in order to cancel the transaction + when our own timer expires. Since we no longer have that access, + if our own timer expires before the pjsip timer, we call the callbacks + and just let the pjsip transaction take it's own course. When the + transaction ends, it discovers the callbacks have already been run + and just cleans itself up. + + A few messages in pjsip_configuration were also added/cleaned up. + + ASTERISK-25105 #close + + Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e + Reported-by: George Joseph + Tested-by: George Joseph +2015-05-20 17:35 +0000 [81d375baad] Joshua Colp + + * res_sorcery_memory_cache: Add support for object_lifetime_maximum. + + This makes the "object_lifetime_maximum" option operational. + + On the addition of an object to an empty memory cache a scheduled + task is created which, when invoked, expires objects from the cache + which have exceeded their lifetime. If more objects have been added + the remaining life of the oldest object is used to schedule the + next invocation of the scheduled task. + + If the oldest object is removed from the cache before it can be + expired automatically the scheduled task is cancelled, if possible, + and the lifetime of the next oldest is used to schedule the task. + + If during these two operations no additional objects exist in the + cache then no task is scheduled. + + An additional unit test has been added which verifies this + functionality. + + ASTERISK-25067 + Reported by: Matt Jordan + + Change-Id: I87409674674a508e7717ee20739ca15cec6ba7b6 + +2015-05-20 00:45 +0000 [9e2a582d2d] demon-ru + + * res_pjsip_outbound_registration: Check request URI for line. + + When an inbound call is received the To header is checked + for the "line" option. Some remote servers will place this + in the request URI instead. This adds an additional check for + the option in the request URI. + + ASTERISK-25072 #close + Reported by: Dmitriy Serov + + Change-Id: Id4e44debbb80baad623b914a88574371575353c8 + +2015-05-20 15:19 +0000 [071b3d43cb] Mark Michelson + + * res_sorcery_memory_cache: Add support for maximum_objects. + + This makes the "maximum_objects" option operational. + + A heap has been added alongside the hash table in the cache. When + objects are added to the cache, they are also added to the heap. + Similarly, when objects are removed from the cache, they are removed + from the heap. + + The heap's use comes into play when an item is to be added to a "full" + cache. When the cache is full, the oldest item is removed from the + cache, using the heap to determine the oldest item. + + A unit test has been added that verifies that the maximum_objects option + works as expected and that the oldest object is removed from the cache + when an object beyond the maximum is added. + + ASTERISK-25067 #close + Reported by Matt Jordan + + Change-Id: I490658830e9c4cbf0b3051e4cdc4913cf9f1b73a + +2015-05-16 17:02 +0000 [f2cc766d81] Joshua Colp + + * res_sorcery_memory_cache: Add basic module implementation. + + This change adds a basic res_sorcery_memory_cache module which implements + configuration option parsing, configuration file parsing for threading, + sorcery interface implementation, and unit tests. + + Objects can be added, updated, deleted, and retrieved from the memory + cache. Automatic expiration and stale handling will be added in the + future. + + Note that unit tests exist within the module itself in case the + threading done as a result of expiration results in asynchronous + actions (which it likely will). Providing access and a notification + mechanism for an external test module would be complicated and + not worth it. + + ASTERISK-25067 #close + Reported by: Matt Jordan + + Change-Id: Id8a6a357ef5a83d466f81eee56a67d13eeb118b9 + +2015-05-21 17:51 +0000 [36e5402885] Corey Farrell + + * res_mwi_external_ami: Use module version of AMI registration. + + Use ast_manager_register_xml for res_mwi_external_ami manager + actions. This ensures the module is held open while any of + the actions are being run. + + ASTERISK-25117 #close + Reported by: Corey Farrell + + Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7 + +2015-05-21 13:05 +0000 [3e2a994c71] Matt Jordan + + * ARI: Update version to 1.7.0 + + This patch updates the version of ARI to 1.7.0 to reflect the backwards + compatible changes that will be introduced in 13.4.0. + + Change-Id: I6c36e6144da426412f25828a868e4df916bff60a + (cherry picked from commit 9d8a462356a938eea82e8424242d89a682495b57) + +2015-05-20 20:53 +0000 [d067847695] Corey Farrell + + * Logger: Reset defaults before processing config. + + Reset options to default values before reloading config. This ensures + that if a setting is removed or commented out of the configuration file + it is unset on reload. + + ASTERISK-25112 #close + Reported by: Corey Farrell + + Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd + +2015-05-20 19:05 +0000 [31f0d78d7b] gtjoseph + + * app_playback: Suppress warnings on playback if channel hung up + + If a channel hangs up while an audio file is playing, there's + no need to clutter up the logs with a warning so suppress it + if ast_check_hangup returns true. + + Also, change warning to debug/2 in file.c if writing a frame + fails. Same reasoning. + + Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 + Reported-by: George Joseph + Tested-by: George Joseph + +2015-04-20 16:00 +0000 [83ff268b9e] Yousf Ateya + + * chan_iax2: Prevent deadlock between hangup and sending lagrq/ping + + channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/ + send_ping. This deadlock happens because the scheduled task send_lagrq(or + send_ping) starts execution after the call hangup procedure starts but before + it deletes the tasks in the scheduler. + + The solution is to delete scheduled lagrq (and ping) task asynchronously + (i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will + be called in a new context (doesn't have callno locked). + + This commit also cleans up the procedure of sending LAGRQ and PING. + + main/sched.c: Do not assert when deleting non existant entry from scheduler. + This assert seems to be the reason for a lot of awkward code to avoid it. + + ASTERISK-24983 #close + Reported by: Y Ateya + + Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c + +2015-05-14 15:21 +0000 [7bf88eb60d] Kevin Harwell + + * audiohook.c: Difference in read/write rates caused continuous buffer resets + + Currently, everytime a sample rate change occurs (on read or write) the + associated factory buffers are reset. If the requested sample rate on a + read differed from that of a write then the buffers are continually reset + on every read and write. This has the side effect of emptying the buffer, + thus there being no data to read and then write to a file in the case of + call recording. + + This patch fixes it so that an audiohook_list's rate always maintains the + maximum sample rate among hooks and formats. Audiohook sample rates are + only overwritten by this value when slin native compatibility is turned on. + Also, the audiohook sample rate can only overwrite the list's sample rate + when its rate is greater than that of the list or if compatibility is + turned off. This keeps the rate from constantly switching/resetting. + + ASTERISK-24944 #close + Reported by: Ronald Raikes + + Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f + +2015-05-13 09:55 +0000 [5ce54ed74a] Matt Jordan + + * res/res_http_websocket: Add a pre-session established callback + + This patch updates http_websocket and its corresponding implementation + with a pre-session established callback. This callback allows for + WebSocket server consumers to be notified when a WebSocket connection is + attempted, but before we accept it. Consumers can choose to reject the + connection, if their application specific logic allows for it. + + As a result, this patch pulls out the previously private + websocket_protocol struct and makes it public, as + ast_websocket_protocol. In order to preserve backwards compatibility + with existing modules, the existing APIs were left as-is, and new APIs + were added for the creation of the ast_websocket_protocol as well as for + adding a sub-protocol to a WebSocket server. + + In particular, the following new API calls were added: + * ast_websocket_add_protocol2 - add a protocol to the core WebSocket + server + * ast_websocket_server_add_protocol2 - add a protocol to a specific + WebSocket server + * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. + Consumers can populate this with whatever callbacks they wish to + support, then add it to the core server or a specified server. + + ASTERISK-24988 + Reported by: Joshua Colp + + Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2 + +2015-05-20 12:55 +0000 [ddb7cbef8e] John Bigelow + + * res/res_resolver_unbound.c: Add missing include of signal.h + + ASTERISK-25110 #close + Reported by: John Bigelow + + Change-Id: I99a9d93f066f265357b647b8e99a75e45da5a39f + +2015-05-06 21:18 +0000 [9c3c7797e5] Rodrigo Ramírez Norambuena + + * cel, cdr: Assigned separator for column name and values. + + Use a separator string between column names and values for SQL sentences + instead of evaluating the separator to use each time. + + This change adds a space after the comma in constructing SQL sentences. + Before the SQL was created like "INSERT INTO cdr(calldate,clid,dst" + without spaces between column name and values. + + The files applied this change are cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, + cel/cel_odbc.c + + ASTERISK-25109 #close + Reported By: Rodrigo Ramírez Norambuena + + Change-Id: Ia5a1a161f5e26e1643703b30f8cc9cf0860cc7ea + +2015-05-17 07:15 +0000 [d8698b7f3f] Matt Jordan + + * doxygen: Fix doxygen errors + + This patch fixes a number of errors and warning messages in the doxygen + log. Specifically, it addresses: + * A number of files incorrectly places a '\brief' tag immediately after + a '\file' tag. Doing so emits a warning, as '\file' takes an optional + argument specifying which file the doxygen comment is for. As '\brief' + is not a file, doxygen was unamused. + * A grouping of Stasis Topics and Messages in rtp_engine.h was + incorrectly terminated. We now correctly terminate the grouping, which + prevents members of rtp_engine.h from showing up in the wrong group. + * Group indicators which are not part of the Stasis Topics and Messages + group were removed. Group indicators without an \addtogroup or + \ingroup have no meaning. + + Change-Id: Ia1415ffec6767e27233ae1cae5ed5970de5656d4 + +2015-05-19 13:01 +0000 [d2e998cd68] Corey Edwards + + * main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits + + ASTERISK-24887 #close + Reported by: Makoto Dei + Tested by: tensai + + Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf +2015-05-14 22:05 +0000 [17129d2c29] snuffy + + * chan_pjsip: Fix crash during off-nominal when no endpoint specified. + + Add missing return -1 when no endpoint name is specified. + + ASTERISK-25086 #close + Reported by: snuffy + + Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e +2015-05-14 18:01 +0000 [5d93928175] gtjoseph + + * res_pjsip_config_wizard/config: Fix template processing + + The config wizard was always pulling the first occurrence of + a variable from an ast_variable list but this gets the template + value from the list instead of any overridden value. This patch + creates ast_variable_find_last_in_list() in config.c and updates + res_pjsip_config_wizard to use it instead of + ast_variable_find_in_list. Now the overridden values, where they + exist, are used instead of template variables. + + Updated test_config to test the new API. + + ASTERISK-25089 #close + + Reported-by: George Joseph + Tested-by: George Joseph + Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4 + +2015-05-15 01:54 +0000 [e48d29054f] snuffy + + * cdr: Fix 'core show channel' CDR variable truncation. + + When the new Bridging API was implemented, the workspace variable + changed to a malloc'd string, causing sizeof() to always be 8 (char). + + Revert back to stored on stack string for workspace. + + ASTERISK-25090 #close + + Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7 +2015-05-10 09:55 +0000 [8f3f414d8c] Alexander Traud (License 6520) + + * tcptls: Enable multiple TLS certificate chains (RSA+ECC+DSA) for server socket. + + When a client connects to a server via SSL/TLS, the server commonly utilizes an + RSA key-pair. However, other such algorithms exist (i.e. DSA and ECDSA), and if + the server socket is configured with a certificate for either one of those, it + would lose its compatibility with RSA-only clients. + + Now, the server socket can be configured with up to one RSA, ECDSA and DSA key + each. For example, if a client is not compatible with SHA-2 hashed certificates + like Nokia mobile phones, the server socket still can use RSA/SHA-1 for legacy + clients and ECDSA/SHA-2 for everyone else. + + ASTERISK-24815 #close + Reported by: Alexander Traud + patches: + tls_rsa_ecc_dsa.patch uploaded by Alexander Traud (License 6520) + + Change-Id: Iada5e00d326db5ef86e0af7069b4dfa1b979da9a + +2015-05-14 17:12 +0000 [2415a14ce9] Maciej Szmigiero + + * Add X.509 subject alternative name support to TLS certificate + verification. + + This way one X.509 certificate can be used for hosts that + can be reached under multiple DNS names or for multiple hosts. + + Signed-off-by: Maciej Szmigiero + + ASTERISK-25063 #close + + Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f + +2015-05-13 15:41 +0000 [3e89f01b55] Jonathan Rose + + * Message.c: Clear message channel frames on cleanup + + The message channel is a special channel that doesn't actually process frames. + However, certain actions can cause frames to be placed in the channel's read + queue including the Hangup application which is called on the channel after + each message is processed. Since the channel will continually be reused for + many messages, it's necessary to flush these frames at some point. + + ASTERISK-25083 #close + Reported by: Jonathan Rose + + Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f + +2015-05-14 00:06 +0000 [0a46d43b9c] Corey Farrell + + * Fix potential crash after unload of func_periodic_hook or test_message. + + These modules save a pointer to the context they create on load, and + use that pointer to destroy the context at unload. It is not safe + to save this pointer, it is replaced during load of pbx_config, + pbx_lua or pbx_ael. + + This change causes the modules to pass NULL to ast_context_destroy, + a safer way to perform the unregistration since it does not use + a pointer that could become invalid. + + ASTERISK-25085 #close + Reported by: Corey Farrell + + Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835 + +2015-05-12 08:58 +0000 [478fb4a388] Corey Farrell + + * MALLOC_DEBUG: Replace WRAP_LIBC_MALLOC with ASTMM_LIBC. + + There are 3 ways that calls directly to standard allocator functions can + be dealt with: + 1. Block their use, cause them to generate an error. This is the default. + 2. Replace them with the Asterisk equivalent function calls. + 3. Leave them alone. + + This change allows one of these 3 options to be selected by any source. + The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT, + or ASTMM_IGNORE to use option 1, 2 or 3 respectively. Normally ASTMM_BLOCK + is the correct option, so it is default when ASTMM_LIBC is not defined. + In some cases when building 3rd party code it is desirable to have it use + Asterisk functions, without changing the whole source - ASTMM_REDIRECT + accomplishes this. When using 3rd party libraries sometimes a static + inline function will make use of malloc or free. In these cases it may + be unsafe to replace the allocator in the header, as it's possible the + memory could be freed by the library using standard allocators. For + those cases ASTMM_IGNORE is needed. + + Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7 + +2015-05-05 19:49 +0000 [eec010829a] Rodrigo Ramírez Norambuena + + * AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. + + Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723 +2015-05-06 05:28 +0000 [46bb8449e8] Rodrigo Ramírez Norambuena + + * cel/cel_pgsql.c: Use the 'SEP' macro when appending a column name + + When appending a column name to the sql buffer, the predicate, "if first is + non-null, use empty string; else, use comma", is identical to the 'SEP' macro + definition. Since they are the same, this patch replaces the redundant + predicate statement with the 'SEP' macro. + + Change-Id: Ib8b6138b06a48381723108a05ab8752cb8700509 +2015-05-12 17:45 +0000 [0d97d7cb94] Jonathan Rose + + * app_voicemail: fix moving when old messages full + + When completing voicemail playback of a message in the 'INBOX', the + message gets moved to the 'Old' messages folder. Without this patch, if + the 'Old' folder is already at its set limit, then the 'INBOX' message will + simply be deleted. With this patch, the flag to delete the message will be + removed if the save_to_folder function indicates that the message could + not be moved due to a full folder. + + ASTERISK-25082 #close + Reported by: Jonathan Rose + Review: https://gerrit.asterisk.org/#/c/448/ + + Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f +2015-05-12 17:34 +0000 [0bb0d4a603] Richard Mudgett + + * chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision. + + If an ISDN call is hungup by both sides at the same time a crash could + happen. + + * Added missing NULL checks for the owner channel after calling + pri_queue_pvt_cause_data() in two places. Code after those calls need to + check the owner channel pointer for NULL before use because + pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the + owner and the owner may get hung up. + + ASTERISK-21893 #close + Reported by: Alexandr Gordeev + + Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a + +2015-05-06 08:31 +0000 [57386dcb67] Corey Farrell + + * Allow command-line options to override asterisk.conf. + + Previous versions of Asterisk processed command-line options before + processing asterisk.conf. This meant that if an option was set in + asterisk.conf, it could not be overridden with the equivelent command + line option. This change causes Asterisk to process the command-line + twice. First it processes options that are needed to load asterisk.conf, + then it processes the remaining options after the config is read. + + This changes the function of -X slightly. Previously using -X without + disabling execincludes in asterisk.conf caused #exec to be usable in any + config. Now -X only enables #exec for the load of asterisk.conf, if it + is wanted in the rest of the system it must be enabled with execincludes + in asterisk.conf. Updated 'asterisk -h' and 'man asterisk' to reflect + the limited function of -X. + + ASTERISK-25042 #close + Reported by: Corey Farrell + + Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7 + +2015-05-05 15:32 +0000 [52407088f8] gtjoseph + + * sorcery: Add API to insert/remove a wizard to/from an object type's list + + Currently you can 'apply' a wizard to an object type but the wizard + always goes at the end of the object type's wizard list. This patch + adds a new ast_sorcery_insert_wizard_mapping function that allows + you to insert a wizard anyplace in the list. I.E. You could + add a caching wizard to an object type and place it before all + wizards. + + ast_sorcery_get_wizard_mapping_count and + ast_sorcery_get_wizard_mapping were added to allow examination + of the mapping list. + + ast_sorcery_remove_mapping was added to remove a mapping by name. + + As part of this patch, the object type's wizard list was converted + from an ao2_container to an AST_VECTOR_RW. + + A new test was added to test_sorcery for this capability. + + ASTERISK-25044 #close + + Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57 + +2015-05-12 01:31 +0000 [cc853dcf90] Corey Farrell + + * Fix processing of asterisk.conf debug=yes. + + The code which reads asterisk.conf supports processing the debug + option with ast_true, but ast_true returns -1. This causes debug + to still be off, convert to 1 so debug will be on as requested. + + ASTERISK-25042 + Reported by: Corey Farrell + + Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6 + +2015-05-10 02:26 +0000 [c624e4bae1] Sebastian Kemper + + * General: Fix recent menuselect-related cross compile regression + + MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the + target platform. But menuselect is to be run on the build system, so + BUILD_CC needs to be used instead - like it was in the past, before the + recent changes (https://reviewboard.asterisk.org/r/4370/). This is the + patch for ASTERISK-25074. + + ASTERISK-25074 #close + Reported by: Sebastian Kemper + Tested by: Sebastian Kemper + + Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8 +2015-05-01 12:22 +0000 [e6daafb8a6] Rodrigo Ramírez Norambuena + + * cdr_pgsql, cel_pgsql: Store maximum buffer size to prevent reallocation + + The code previously used a fixed size of 512 for the SQL + queries. Depending on the size this may require it to grow. + + This change makes it so if the buffer size does grow the size + is stored and next time the buffer will be large enough. + + Change-Id: I55385899f1c06dee47e4274c2d21538037b2d895 +2015-05-09 16:58 +0000 [87d8b36755] gtjoseph + + * vector: Add REMOVE, ADD_SORTED and RESET macros + + Based on feedback from Corey Farrell and Y Ateya, a few new + macros have been added... + + AST_VECTOR_REMOVE which takes a parameter to indicate if + order should be preserved. + + AST_VECTOR_ADD_SORTED which adds an element to + a sorted vector. + + AST_VECTOR_RESET which cleans all elements from the vector + leaving the storage intact. + + Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14 + +2015-05-11 07:07 +0000 [e6ebddd9ae] Ivan Poddubny + + * pbx/pbx_spool: Fix issue when call files were executed too early + + pbx_spool used to delete/move the call file upon successful outgoing + call completion, but did not delete it from in-memory list of files + (dirlist, used only when compiled with inotify/kqueue support). + That resulted in an extra attempt to process that filename after + retrytime seconds. + Then, if a new file with the same name appears that is scheduled + in future further than the completed one plus its retrytime, + then it gets executed earlier than expected. + + This patch fixes remove_from_queue function to also remove the entry + from the dirlist. + + ASTERISK-17069 #close + Reported by: Jeremy Kister + + ASTERISK-24442 #close + Reported by: tootai + + Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b + +2015-05-01 23:43 +0000 [c61b146238] Rodrigo Ramírez Norambuena + + * cdr_pgsql: Use PQescapeStringConn for escaping names. + + Use function PQescapeStringConn for escaping the name + of the table and schema instead of doing it manually. + + Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599 +2015-05-10 07:37 +0000 [2ab5d22c0d] Yousf Ateya + + * res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS. + + First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC + https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values. + + Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31 + +2015-05-10 08:36 +0000 [f82bd76e3c] Joshua Colp + + * dns_srv: Fix SRV sorting when records with priority zero exist with non-zero. + + The DNS SRV sorting code currently has an issue when records with a priority + of zero exist with records of a non-zero priority. This occurs because the + sorting code considers zero to mean unset when in reality is a valid + value. If the current priority is zero it will get replaced with any remaining + record that has a priority of non-zero, until no records of those exist after + which the records of priority zero are handled. + + This change makes it so that the priority of the first remaining record is + the current starting priority. There is also a small optimization to prevent + iterating records when the starting priority is already zero. + + Change-Id: I103511f35b50428f770bd4db3ffef70fb6f82d35 + +2015-05-08 18:01 +0000 [1503d0c14c] Alexandre Fournier + + * res_config_mysql: Fix broken column type checking + + MySQL configuration engine contains a bug in require_mysql(). This + function is used for column type checking in tables. This bug only + affects DATETIME, DATE and FLOAT types. + + It came from mixing the first condition (switch-case-like + if/then/else), to check the expected column type, with the second + condition, to check the actual column type against the expected column + type. Both conditions must be checked separately in order to avoid the + execution of the wrong block. + + ASTERISK-18252 #comment This patch might fix the issue + Reported by: Gareth Blades + + ASTERISK-25041 #close + Reported by: Alexandre Fournier + Tested by: Alexandre Fournier + + Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa + +2015-05-08 14:47 +0000 [5e361e1476] Rusty Newton + + * configs/basic-pbx: Modified main IVR to play new Allison prompt. + + The main IVR was playing demo-congrats. I've switched it over to the + basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt + has Allison prompting the user with the actual IVR menu. + + ASTERISK-24892 #close + + Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d + +2015-05-08 12:30 +0000 [2d4dc0c963] Corey Farrell + + * Fix error's produced by astmm.h when standard allocators are used. + + astmm.h includes defines that are meant to cause error's when standard + allocators (malloc, calloc, free, etc) are used. It actually only + causes a warning, which is not always caught on certain sources. In + modules this unknown symbol is not detected until runtime, where the + module fails to load. This modifies the define's so that using one + of the blocked functions will cause a compile error regardless of + CFLAGS. + + Moved spandsp header includes to before asterisk.h so the static inline + functions can continue using malloc and free. Although these functions + are never called and optimized away, the updated replacement macro's + would still cause a failure. + + Change-Id: I532640aca0913ba9da3b18c04a0f010ca1715af5 + +2015-05-08 10:39 +0000 [63c71c9f4a] Sean Bright + + * res_rtp_asterisk: Issue ERROR if res_srtp is not found. + + While trying to get WebRTC working with chan_pjsip, I was running + into the following error: + + Attempted to set an invalid DTLS-SRTP configuration on RTP + instance... + + Josh helpfully pointed out that res_srtp.so might not be loaded, and + sure enough, it wasn't. This patch adds a ERROR indiciating as much + to hopefully help others having a similar problem. + + Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f + +2015-05-07 17:49 +0000 [60bf9ed91a] Rusty Newton + + * sounds: Add Swedish sounds to Makefile and XML + + Added the necessary lines to the Makefile and sounds.xml so we'll have the + Swedish sounds in all available formats in menuselect. + + See also: Swedish sounds were added into the core sounds release 1.4.27. + + ASTERISK-24744 #close + + Reported by: Tove Hjelm + Tested by: Rusty Newton + + Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908 + +2015-05-08 10:30 +0000 [f93b3a22d6] Corey Farrell + + * Fix crash in codec_lpc10 when MALLOC_DEBUG is enabled. + + This switches codecs/lpc10/lpcini.c back to including "asterisk.h" + instead of . lpcini.c allocates memory that is freed by + codec_lpc10.c, so it is important to use MALLOC_DEBUG allocator. + Added #define WRAP_LIBC_MALLOC to the start of the source to prevent + runtime symbol link error's. + + Change-Id: I74f63fd09fdeb673ee7753122c3bb4722ab6e1ac + +2015-05-07 14:54 +0000 [cf637f2510] gtjoseph + + * doc: Make progdocs play nice with git + + Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in + + Changed /Makefile to copy asterisk-ng-doxygen.in to + asterisk-ng-doxygen then modify it with version instead of + modifying asterisk-ng-doxygen directly. Updated clean + targets as well. + + Updated /.gitignore and doc/.gitignore. + + Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622 + +2015-05-04 14:43 +0000 [b885f719bf] Ivan Poddubny + + * contrib/editors: Fix vim syntax highlighting of comments in config files + + * Added a lookbehind to one-line comment matcher to skip escaped + semicolons. + * Added support for block comments. + + Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7 + +2015-05-06 13:24 +0000 [e33682cae2] Joshua Colp + + * res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination + + The res_pjsip_exten_state module currently has a race condition between + processing the extension state callback from the PBX core and processing + the subscription shutdown callback from res_pjsip_pubsub. There is currently + no synchronization between the two. This can present a problem as while + the SIP subscription will remain valid the tree it points to may not. + This is in particular a problem as a task to send a NOTIFY may get queued + which will try to use the tree that may no longer be valid. + + This change does the following to fix this problem: + + 1. All access to the subscription tree is done within the task that + sends the NOTIFY to ensure that no other thread is modifying or + destroying the tree. This task executes on the serializer for the + subscriptions. + + 2. A reference to the subscription serializer is kept to ensure it + remains valid for the lifetime of the extension state subscription. + + 3. The NOTIFY task has been changed so it will no longer attempt + to send a NOTIFY if the subscription has already been terminated. + + ASTERISK-25057 #close + Reported by: Matt Jordan + + Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643 + +2015-05-05 20:22 +0000 [c886be5df2] gtjoseph + + * vector: Additional enhancements and fixes + + After using the new vector stuff for real I found... + + A bug in AST_VECTOR_INSERT_AT that could cause a seg fault. + + The callbacks needed to be closer to ao2_callback in behavior + WRT to CMP_MATCH and CMP_STOP behavior and the ability to return + a vector of matched entries. + + A pre-existing issue with APPEND and REPLACE was also fixed. + + I also added a new macro to test.h that acts like ast_test_validate + but also accepts a return code variable and a cleanup label. As well + as printing the error, it sets the rc variable to AST_TEST_FAIL and + does a goto to the specified label on error. I had a local version + of this in test_vector so I just moved it. + + ASTERISK-25045 + + Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc + +2015-05-06 17:37 +0000 [1f5db1c7e3] Kevin Harwell + + * res_stasis_snoop: Spying on a single direction continually increases CPU + + Creating a snoop channel in ARI and spying only on a single direction (in or + out) results in CPU utilization continually increasing until the CPU is fully + consumed. This occurs because frames are being put in the opposing direction's + slin factory queue, but not being removed. + + Fixed the problem by always reading and disposing of frames from the opposite + queue of the direction selected. + + ASTERISK-24938 #closes + + Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60 + +2015-05-06 16:00 +0000 [7103b374ef] Richard Mudgett + + * chan_dahdi: Improve force_restart_unavailable_chans option description. + + ASTERISK-25034 + Reported by: Richard Mudgett + + Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30 + +2015-05-06 04:32 +0000 [d2e2271874] Joshua Colp + + * manager: Fix build due to missing variable usage. + + Change-Id: I26d4d2cb9cee924632ff59ef0b30a7e6a1e2b00d + +2015-05-04 20:11 +0000 [6b40bbf5bb] Rodrigo Ramírez Norambuena + + * main/manager.c: Bugfix sort action_manager by alphabetically + + Fix the alphabetic order added on ast_manager_register_struct. The order + for struct manager_action added is not working, this change fixes the + problem. + + Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b + +2015-05-05 18:17 +0000 [6c4d1c3223] Richard Mudgett + + * features: Fix crash when transferee hangs up during DTMF attended transfer. + + A crash happens with this sequence of steps: + 1) Party A is connected to party B. + 2) Party B starts a DTMF attended transfer. + 3) Party A hangs up while party B is dialing party C. + + When party A hangs up the bridge that party A and party B are in is + dissolved and party B is kicked out of the bridge. When party B finishes + dialing party C he attempts to move to the new bridge with party C. Since + party B is no longer in a bridge the attempted move dereferences a NULL + bridge_channel pointer and crashes. + + * Made the hold(), unhold(), ringing(), and the bridge_move() functions + tolerant of the channel not being in a bridge. The assertion that party B + is always in a bridge is not true if the bridged peer of party B hangs up + and dissolves the bridge. Being tolerant of not being in a bridge allows + the peer hangup stimulus to be processed by the FSM. + + * Made the bridge_move() function return void since where the return value + for a failed move was checked generated a FSM coding ERROR message for a + normal off-nominal condition. + + * Eliminated most uses of RAII_VAR in bridge_basic.c. + + ASTERISK-25003 #close + Reported by: Artem Volodin + + Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada + +2015-05-05 14:48 +0000 [90bfc02e84] Ivan Poddubny + + * app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter + + This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3 + parameters: position, original position and waiting time. + + ASTERISK-25038 #close + Reported by: Etienne Lessard + + Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618 + +2015-05-05 13:34 +0000 [bebf0b9b27] Joshua Colp + + * chan_unistim: Fix build failure due to ACL changes. + + Change-Id: I57081045c72b9fcf12d5c84493278f9272c31b32 + +2015-05-05 11:35 +0000 [247fef6653] Alexander Traud (License 6520) + + * tcptls: Avoiding ERR_remove_state in OpenSSL. + + ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by + ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were + called by SSL_load_error_strings already and got removed. These changes allow + OpenSSL forks like BoringSSL to be used with Asterisk. + + ASTERISK-25043 #close + Reported by: Alexander Traud + patches: + asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520) + + Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629 +2015-05-05 09:47 +0000 [c541923ac3] Corey Farrell + + * res_ari_bridges: Add missing dependencies. + + Missed this module in the previous commit. res_ari_bridges uses symbols + from res_stasis_playback and res_stasis_recording. + + ASTERISK-25027 #close + Reported by: Corey Farrell + + Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f + +2015-05-05 09:27 +0000 [8a3e93a349] Corey Farrell + + * pbx_config: Register manager actions with module version of macro. + + Switch manager actions in pbx_config to use the registration macro that + passes the module pointer, allowing pbx_config reference to be bumped + while the manager actions run. + + ASTERISK-25061 #close + Reported by: Corey Farrell + + Change-Id: I422c50dd74814616ac10c5e9c6598a0b1bc2c44e + +2015-05-01 22:14 +0000 [cb79b8ab80] Rodrigo Ramírez Norambuena + + * cel_pgsql: Add support for setting schema + + Add feature to set optional schema parameter on configuration file via + 'schema' setting. + + Fix query to get columns from table while considering schema. If in + the database there exists two tables with same name in distinct schemas + it will return an error when inserting record. + + ASTERISK-24967 #close + + Change-Id: I691fd2cbc277fcba10e615f5884f8de5d8152f2c + +2015-05-04 12:16 +0000 [11f650c6ac] Joshua Colp + + * stasis: Fix dial masquerade datastore lifetime + + A recent change went into Asterisk which added reference counts to the + channels stored in a dial masquerade datastore. Unfortunately this + included a reference to the caller in a dialing operation. While all + of the dialed targets have the datastore removed from them upon dialing + completion this did not occur for the caller, causing it to have a + reference to itself that could go never go away (as it depended on + the destruction of the datastore which only happened when the channel + was destroyed). This resulted in the caller channel remaining on the + system despite it having hung up. + + This change does the following to fix this issue: + + 1. The dial masquerade datastore is now removed from the caller upon + dialing completion, just like the dialed targets. + 2. Upon destruction of the caller all the dialed targets are also + removed from the dial masquerade datastore (just in case). + 3. The reference to the caller has been removed as it should not be + possible for the datastore to now be valid/useful after the lifetime + of the caller has ended. + + ASTERISK-25025 #close + + Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f + +2015-04-21 17:27 +0000 [a24ce38e5e] Rodrigo Ramírez Norambuena + + * cdr_adaptive_odbc: Add ability to set character for quoted identifiers. + + Added the ability to set the character to quote identifiers. This + allows adding the character at the start and end of table and column + names. This setting is configurable for cdr_adaptive_odbc via the + quoted_identifiers in configuration file cdr_adaptive_odbc.conf. + + ASTERISK-25006 + + Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8 + +2015-05-04 22:57 +0000 [39cf642d40] Rodrigo Ramírez Norambuena + + * cdr: standardizes tab for options of AST_MODULE_INFO + + Change-Id: I3c6de30b4859717873100092a7c06e206713a301 + +2015-05-04 16:41 +0000 [df6c1d755f] Corey Farrell + + * CLI: Enable automatic references to modules. + + * Pass module to ast_cli_register and ast_cli_register_multiple. + * Add a module reference before executing any CLI callback, remove + the reference when complete. + + ASTERISK-25049 #close + Reported by: Corey Farrell + + Change-Id: I7aafc7c9f2b912918f28fe51d51e9e8a755750e3 + +2015-05-04 14:26 +0000 [a8bfa9e104] Corey Farrell + + * Modules: Make ast_module_info->self available to auxiliary sources. + + ast_module_info->self is often needed to register items with the core. Many + modules have ad-hoc code to make this pointer available to auxiliary sources. + This change updates the module build process to make the needed information + available to all sources in a module. + + ASTERISK-25056 #close + Reported by: Corey Farrell + + Change-Id: I18c8cd58fbcb1b708425f6757becaeca9fa91815 + +2015-05-01 19:25 +0000 [6d5941297b] gtjoseph + + * vector: Traversal, retrieval, insert and locking enhancements + + Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really + does replace not insert. The few users of AST_VECTOR_INSERT were + refactored. Because these are macros, there should be no ABI + compatibility issues. + + Added AST_VECTOR_INSERT_AT that actually inserts an element into the + vector at a specific index pushing existing elements to the right. + + Added AST_VECTOR_GET_CMP that can retrieve from the vector based + on a user-provided compare function. + + Added AST_VECTOR_CALLBACK function that will execute a function + for each element in the vector. Similar to ao2_callback and + ao2_callback_data functions although the vector callback can take + a variable number of arguments. This should allow easy migration + to a vector where a container might be too heavy. + + Added read/write locked vector and lock manipulation macros. + + Added unit tests. + + ASTERISK-25045 #close + + Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0 + +2015-05-03 13:55 +0000 [4f4aaa0c30] Corey Farrell + + * main/test.c: Add test to verify there were no registration errors. + + This adds a test that will fail if any test failed to register. Also fail + if any test registration produced a warning about missing a leading or + trailing slash. + + ASTERISK-25053 #close + Reported by: Corey Farrell + + Change-Id: I93e50b8fcbcfa7f1f5b41b2c44a51685c09529c3 + +2015-04-21 11:52 +0000 [ebe371357e] Martin Tomec + + * res_odbc: Use negative connection cache for all connections + + Apply the negative connection cache setting to all connections, + even those that are not pooled. This ensures that the connection + will not be re-established before the negative connection cache + time is met. + + ASTERISK-22708 #close + + Change-Id: I431cc2e8584ab0b6908b3523d0a0e18c9a527271 +2015-05-03 21:03 +0000 [981084f08c] Corey Farrell + + * Format Interfaces: Prevent unload except by shutdown. + + Format interfaces cannot be unregistered, so the modules that provide them + need to be held open except by shutdown. + + ASTERISK-25054 #close + Reported by: Corey Farrell + + Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3 + +2015-05-03 20:28 +0000 [75c0aa6979] Matt Jordan + + * contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update + + The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755 + failed to add ENUM support for Postgres databases. This requires a + specific import from the sqlalchemy.dialects.postgresql package. This + patch corrects this error, which allows for Postgres update scripts to + be generated. + + ASTERISK-24706 + + Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015 + +2015-05-03 13:36 +0000 [1368dae773] Corey Farrell + + * main/presencestate.c: Add trailing slash to test category. + + ASTERISK-25053 + Reported by: Corey Farrell + + Change-Id: I8c0375dd0818747b2d2e1ceaea87bfbeb2daf8d4 + +2015-04-20 13:03 +0000 [305ce3defd] Diederik de Groot + + * Update configure.ac/Makefile for clang + + Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which + checks compiler requirements for RAII: + gcc: -fnested-functions support + clang: -fblocks (and if required -lBlocksRuntime) + The original check was implemented in configure.ac and now has it's + own file. This function also sets C_COMPILER_FAMILY to either gcc or + clang for use by makefile + + Created autoconf/ast_check_strsep_array_bounds.m4 (contains + AST_CHECK_STRSEP_ARRAY_BOUNDS): + which checks if clang is able to handle the optimized strsep & strcmp + functions (linux). If not, the standard libc implementation should be + used instead. Clang + the optimized macro's work with: + strsep(char *, char []), but not with strsepo(char *, char *). + Instead of replacing all the occurences throughout the source code, + not using the optimized macro version seemed easier + + See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h': + llvm-comment: Normally, this array-bounds warning are suppressed for + macros, so that unused paths like the one that accesses __s1[3] are + not warned about. But if you preprocess manually, and feed the + result to another instance of clang, it will warn about all the + possible forks of this particular if statement. Instead of switching + of this optimization, another solution would be to run the preproces- + sing step with -frewrite-includes, which should preserve enough + information so that clang should still be able to suppress the diag- + nostic at the compile step later on. + + See also "https://llvm.org/bugs/show_bug.cgi?id=20144" + See also "https://llvm.org/bugs/show_bug.cgi?id=11536" + + Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning + suppressions: + -Wno-unused-value + -Wno-parentheses-equality + In an earlier review (reviewboard: 4550 and 4554), they were deemed a + nuisace and less than benefitial. + + configure.ac: + Added AST_CHECK_RAII() see earlier + Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier + Removed moved content + + ASTERISK-24917 + Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb + +2015-04-28 04:49 +0000 [8886b724ae] Rodrigo Ramírez Norambuena + + * cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8 + + This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR + columns added in Asterisk 1.8. The columns are: + * peeraccount + * linkedid + * sequence + When enabled, the columns in the database entry will be populated with the data + from the CDR. + + ASTERISK-24976 #close + + Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b +2015-05-03 04:39 +0000 [94532b2c22] Rodrigo Ramírez Norambuena + + * main/asterisk.c: Update Asterisk copyright year + + Change-Id: I5e75d7f7e2c096d74edd9e8735268a894f4b93ab + +2015-05-03 04:09 +0000 [2ed5e6a9ba] Rodrigo Ramírez Norambuena + + * utils: Remove trailing whitespace + + Change-Id: I4644f43a6a1ca9b5130cd2a6746772b888eb4f7a + +2015-05-02 18:58 +0000 [c3ec5da156] Corey Farrell + + * Remove unneeded uses of optional_api providers. + + A few cases exist where headers of optional_api provders are included but + not needed. This causes unneeded calls to ast_optional_api_use. + + * Don't include optional_api.h from sip_api.h. + * Move 'struct ast_channel_monitor' to channel.h. + * Don't include monitor.h from chan_sip.c, channel.c or features.c. + + The move of struct ast_channel_monitor is needed since channel.c depends on + it. This has no effect on users of monitor.h since channel.h is included + from monitor.h. + + ASTERISK-25051 #close + Reported by: Corey Farrell + + Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478 + +2015-05-02 02:15 +0000 [44bbdbe3a4] Corey Farrell + + * res_pjsip_dlg_options: Fix MODULEINFO section. + + Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options. + This extra space prevented any of the dependencies from being seen by + menuselect, so building with default options would fail if PJSIP was + not installed. + + This also makes the tool that extracts information for menuselect + tolerant of multiple spaces in the future. + + ASTERISK-25033 #close + Reported by: Peter Whisker + + Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698 + +2015-05-01 19:50 +0000 [e4f0a55f7f] D Tucny + + * term: send proper reset sequence when black background is forced + + When using the force black background command-line option or configuration + option an invalid reset sequence is sent following a coloured output item + in the CLI, the result is that the colour is not 'turned off' and continues + until the next non-default coloured text output. + + A reset sequence is already defined in term.c, but the ast_term_reset + function doesn't use it, instead building it's own invalid sequence and + returning that. + + This patch changes that behaviour, removing the building of a reset sequence + and instead using the pre-built constant 'enddata' which is a suitable reset + sequence for this purpose. + + ASTERISK-24896 #close + Reported by: Dan Tucny + + Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43 +2015-05-01 13:22 +0000 [8f3cee1258] Corey Farrell + + * Astobj2: Fix initialization order of refdebug and AO2_DEBUG. + + This ensures that refdebug is initialized before AO2_DEBUG if + both are enabled, since AO2_DEBUG allocates a container. + + This change also makes AO2_DEBUG initialization critical, a + failure will abort Asterisk startup. This is needed since + the failure would be caused by reg_containers allocation + failure, and that would result in a segmentation fault by + ao2_container_register later in startup. + + ASTERISK-25048 #close + Reported by: Corey Farrell + + Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244 + +2015-04-29 14:49 +0000 [7ac28be04b] Matt Jordan + + * main/pbx: Improve performance of dialplan reloads with a large number of hints + + The PBX core maintains two hash tables for hints: a container of the + actual hints (hints), along with a container of devices that are watching that + hint (hintdevices). When a dialplan reload occurs, each hint in the hints + container is destroyed; this requires a lookup in the container of devices to + find the device => hint mapping object. In the current code, this performs an + ao2_callback, iterating over each of the device to hint objects in the + hintdevices container. For a large number of hints, this is extremely + expensive: dialplan reloads with 20000 hints could take several minutes + in just this phase. + + This patch improves the performance of this step in the dialplan reloads + by caching which devices are watching a hint on the hint object itself. + Since we don't want to create a circular reference, we just cache the + name of the device. This allows us to perform a smarter ao2_callback on + the hintdevices container during hint removal, hashing on the name of the + device and returning an iterator to the matching names. The overall + performance improvement is rather large, taking this step down to a number of + seconds as opposed to minutes. + + In addition, this patch also registers the hint containers in the PBX + core with the astobj2 library. This allows for reasonable debugging to + hash collisions in those containers. + + ASTERISK-25040 #close + Reported by: Matt Jordan + + Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360 + +2015-04-30 15:54 +0000 [6b208d8c3b] Corey Farrell + + * Sample Configs: Fix syntax error in pjsip.conf + + The sample pjsip.conf has a few comment lines that are missing the + semicolons at the start of the comment, causing the config to fail + load. + + Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352 + +2015-04-30 15:20 +0000 [dc23204aca] Mark Michelson + + * Prevent potential crash on blond transfer. + + Scenario: + Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects + the incoming call (or some other immediate circumstance causes Carol not + to answer the call) + + What occurs in this case is that when the bridge between Alice and Bob + breaks, Alice is told to masquerade into Bob's channel that had placed + the call to Carol. The actual masquerade goes down without a hitch. + However, a channel fixup callback that attempts to publish dial events + over Stasis has a crash. The reason for this crash is that the datastore + on Bob's channel that placed the outbound call to Carol only had a bare + pointer to Carol's channel. Since Carol rejected the incoming call, + Carol's channel has been hung up and freed, meaning accessing her + channel results in a crash. + + The fix here is simple. The dial fixup code has been altered to hold + references to the involved channels and to drop those references when + freeing data. + + ASTERISK-25025 #close + Reported by Chet Stevens + + Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197 + +2015-04-30 14:40 +0000 [47fa2ad10b] Corey Farrell + + * Build System: Fix issue with addons moduleinfo. + + The build system now scans additional sources when generating + moduleinfo for menuselect. Unfortunately the extra sources + for format_mp3 only exist if downloaded. + + Use the Makefile macro 'wildcard' to allow moduleinfo generator + to ignore sources that do not exist. + + Change-Id: I596604713b7345ce994f32197f8f6bfd9bcf4170 + +2015-04-30 13:42 +0000 [bb6ddb3dc8] Joshua Colp + + * res_ari_device_states: Fix dependency on res_stasis_device_state. + + The res_ari_device_states module depends on res_stasis_device_state, + not res_stasis_device_states. + + Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de + +2015-04-28 17:00 +0000 [11ffcf662f] Mark Michelson + + * Restrict functionality when ACLs are misconfigured. + + This patch has two main purposes: + + 1) Improve warning messages when ACLs are configured improperly. + 2) Prevent misconfigured ACLs from allowing potentially unwanted + traffic. + + To acomplish point (2) in most cases, whatever configuration object that + the ACL belonged to was not allowed to load. + + The one exception is res_pjsip_acl. In that case, ACLs are their own + configuration object. Furthermore, the module loading code has no + indication that a ACL configuration had a failure. So the tactic taken + here is to create an ACL that just blocks everything. + + ASTERISK-24969 + Reported by Corey Farrell + + Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae + +2015-04-29 14:29 +0000 [03c51cf525] Richard Mudgett + + * chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. + + Some telco switches occasionally ignore ISDN RESTART requests. The fix + for ASTERISK-19608 added an escape clause for B channels in the restarting + state if the telco ignores a RESTART request. If the telco fails to + acknowledge the RESTART then Asterisk will assume the telco acknowledged + the RESTART on the second call attempt requesting the B channel by the + telco. The escape clause is good for dealing with RESTART requests in + general but it does cause the next call for the restarting B channel to be + rejected if the telco insists the call must go on that B channel. + + chan_dahdi doesn't really need to issue a RESTART request in response to + receiving a cause 44 (Requested channel not available) code. Sending the + RESTART in such a situation is not required (nor prohibited) by the + standards. I think chan_dahdi does this for historical reasons to deal + with buggy peers to get channels unstuck in a similar fashion as the + chan_dahdi.conf resetinterval option. + + * Add the chan_dahdi.conf force_restart_unavailable_chans compatability + option that when disabled will prevent chan_dahdi from trying to RESTART + the channel in response to a cause 44 code. + + ASTERISK-25034 #close + Reported by: Richard Mudgett + + Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65 +2015-04-29 21:54 +0000 [556653d937] Rodrigo Ramírez Norambuena + + * cdr/cdr_csv.c: Refactor, function to write content of csv file. + + Create a function for write content of CDR on csv files. Before used same + code for write two distinct files (account and master cdr) instead use a + function for thats. + + Reduced to one lock when files are written. + + Change-Id: Idce707f4c108083252e0aeb948f421d924953e65 + +2015-04-30 06:04 +0000 [80aa9aee5d] Joshua Colp + + * res_pjsip_outbound_registration: Fix double unref on error return. + + When the PJSIP pjsip_regc_send function is invoked and an error + status returned the caller currently decrements the reference count + of the client state that it just incremented, assuming the + registration callback would not have been invoked. In practice + this is not correct. If the failure happens after the transaction + has been set up the callback will still be invoked. This will + cause the reference count to be incorrectly decremented twice, once + by the registration callback and second by the caller of + pjsip_regc_send. + + This change makes it so that whether the callback is invoked or + not is known by the caller of pjsip_regc_send. Depending on + this it can know whether it is responsible for decrementing the + reference count of the client state or not. + + ASTERISK-25037 #close + Reported by: Joshua Colp + + Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de + +2015-04-30 02:07 +0000 [7ff3b2d479] Rodrigo Ramírez Norambuena + + * include/asterisk/channel.h: Fix typo + + Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3 + +2015-04-29 16:15 +0000 [39d3e1ef6e] Matt Jordan + + * main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8 + + The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS + structures in the RTP engine. However, when a 'core reload' is issued, a + double free of the memory pointed to by the char *'s in the DTLS + configuration struct can occur, as ast_rtp_dtls_cfg_free does not set + the pointers to NULL when they are freed. + + This patch sets those pointers to NULL, preventing a second call to + ast_rtp_dtls_cfg_free from corrupting memory. + + ASTERISK-25022 + + Change-Id: I820471e6070a37e3c26f760118c86770e12f6115 + +2015-04-29 13:05 +0000 [5d0c182885] Kevin Harwell + + * res_fax: allow 2400 transmission rate according to v.27ter standard + + A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so + a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits + per second. This reverts all or some of those patches since according to the + v.27ter standard a rate of 2400 bits per second is also supported. + + One of the original patches also added 9600 bits per second support for v.27. + This patch also removes that since v.27ter only supports 2400/4800 bits per + second. + + Also, since Asterisk specifically supports v.27ter the enum was renamed to + better reflect this. + + ASTERISK-24955 #close + Reported by: Matt Jordan + + Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733 + +2015-04-28 23:35 +0000 [c9c03998cc] Corey Farrell + + * Astobj2: Add ao2_weakproxy_ref_object function. + + This function allows code to run ao2_ref against the real + object associated with a weakproxy. It is useful when + all of the following conditions are true: + * You have a pointer to weakproxy. + * You do not have or need a pointer to the real object. + * You need to ensure the real object exists and is not + destroyed during a process. + + In this case it's wasteful to store a pointer to the real + object just for the sake of releasing it later. + + Change-Id: I38a319b83314de75be74207a8771aab269bcca46 + +2015-04-27 16:13 +0000 [4f1db2070d] Mark Michelson + + * res_pjsip_outbound_registration: Don't fail on delayed processing. + + Odd behaviors have been observed during outbound registrations. The most + common problem witnessed has been one where a request with + authentication credentials cannot be created after receiving a 401 + response. Other behaviors include apparently processing an incorrect SIP + response. + + Inspecting the code led to an apparent issue with regards to how we + handle transactions in outbound registration code. When a response to a + REGISTER arrives, we save a pointer to the transaction and then push a + task onto the registration serializer. Between the time that we save the + pointer and push the task, it's possible for the transaction to be + destroyed due to a timeout. It's also possible for the address to be + reused by the transaction layer for a new transaction. + + To allow for authentication of a REGISTER request to be authenticated + after the transaction has timed out, we now hold a reference to the + original REGISTER request instead of the transaction. The function for + creating a request with authentication has been altered to take the + original request instead of the transaction where the original request + was sent. + + ASTERISK-25020 + Reported by Mark Michelson + + Change-Id: I756c19ab05ada5d0503175db9676acf87c686d0a +2015-04-29 10:46 +0000 [ed5715eb39] Joshua Colp + + * res_sorcery_config: Fix build issue due to syntax error. + + Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac + +2015-04-29 06:46 +0000 [f226bd6f60] Corey Farrell + + * ARI: Fix missing dependencies. + + ARI modules that are generated by 'make ari-stubs' are all dependent on + res_ari_model. Additionally some of the same modules depend on one or more + res_stasis_* modules. + + ASTERISK-25027 #close + Reported by: Corey Farrell + + Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153 + +2015-04-29 06:26 +0000 [881844297a] Corey Farrell + + * res_pjsip: Remove incorrect MODULEINFO from presence_xml.c. + + Remove incorrect MODULEINFO block and unneeded header includes + from presence_xml.c. + + ASTERISK-25027 + Reported by: Corey Farrell + + Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb + +2015-04-29 06:17 +0000 [c232ff3af0] Corey Farrell + + * Git Migration: Create doc/rest-api when needed. + + Create the directory './doc/rest-api' at the start of 'make ari-stubs' + to prevent an error when documentation is generated. The directory is + also added to git ignores. + + ASTERISK-25027 + Reported by: Corey Farrell + + Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b + +2015-04-29 03:03 +0000 [5d997ecc83] Corey Farrell + + * Build System: Prevent unneeded changes to asterisk/buildopts.h. + + * Add AST_DEVMODE to BUILDOPTS + * Use BUILDOPTS to generate AST_BUILDOPT_SUM. + * Remove loop that defined AST_MODULE_* + + These changes ensure that only ABI effecting options are considered for + AST_BUILDOPT_SUM. This also reduces unneeded full system rebuilds caused + by enabling or disabling one module that another is dependent on. + + ASTERISK-25028 #close + Reported by: Corey Farrell + + Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2 + +2015-04-29 00:02 +0000 [55a780d211] Corey Farrell + + * Git Conversion: Switch Non-C files to ASTERISK_REGISTER_FILE. + + This switches files used to generate other sources to use the new + ASTERISK_REGISTER_FILE macro. + + ASTERISK-25026 #close + Reported by: Corey Farrell + + Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740 + +2015-04-28 13:28 +0000 [5ebfed8ef3] Yousf Ateya + + * chan_iax2: Ensure that IAX flags are 64 bits. + + Flags are 64 bits. Without LLU suffix the value of 1<<31 is negative. + Although it doesn't have an effect on the current implementation, it will + be problem if more flags are added. + + Change-Id: Ic290c81cfbbbf062872392d99d3322932cc49487 +2015-04-28 00:29 +0000 [46cf643c75] Ashley Sanders + + * chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR + Sections Exist in pjsip.conf + + This patch modifies the current loading strategy of the pjsip configuration. If + duplicate sections (e.g. sections containing the same [id/type]) are defined in + [pjsip.conf], the loader will consider the configuration for the given type as + invalid when the duplicate section is encountered. The entire configuration + (including what was previously loaded) for the duplicate [id/type] sections + will be rejected and destroyed, an error message is logged and the load + processing for the given stops. + + ASTERISK-24996 + Reported By: Ashley Sanders + + Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef +2015-04-28 11:50 +0000 [0bbe2c35cf] Richard Mudgett + + * chan_vpb: Fix compile error due to use of ASTERISK_FILE_VERSION. + + Change-Id: I51179e2a83937423676da522b766f1126de4059e +2015-04-27 14:44 +0000 [f47fed2e12] Mark Michelson + + * res_pjsip_outbound_registration: Add debugging messages. + + When problems occur regarding outbound registrations, it currently + is difficult to debug. Most off-nominal paths had warning messages, + but sometimes we want to know what's going on before hitting the + off-nominal path. This patch adds lots of debugging output that + should give a clearer picture of what is happening with regards + to outbound registrations. + + ASTERISK-25020 + Reported by Mark Michelson + + Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45 + +2015-04-28 05:38 +0000 [5e96584829] Steve Davies + + * res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS + + ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. + The resources are linked into a table, but the original alloc refs + are never released. ast_strdup leak in rtp_engine.c. If + ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, + a pointer to an alloc'd string is overwritten before the string is free'd. + + ASTERISK-25022 + Reported by: one47 + + Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b + +2015-04-28 04:28 +0000 [d6a2d92353] Rodrigo Ramírez Norambuena + + * cdr/cdr_csv.c: Add missing space after comma. + + Change-Id: I3866a20019b1a3a2f10fe36640053929330b0fcb + +2015-04-27 22:01 +0000 [542bfee881] Rodrigo Ramírez Norambuena + + * CHANGES: Add missing spaces. + + Change-Id: I534ea0f22759e3633585dfa9b145b4a284efe67f + +2015-04-17 02:16 +0000 [5c1d07baf0] Corey Farrell + + * Astobj2: Allow reference debugging to be enabled/disabled by config. + + * The REF_DEBUG compiler flag no longer has any effect on code that uses + Astobj2. It is used to determine if reference debugging is enabled by + default. Reference debugging can be enabled or disabled in asterisk.conf. + * Caller information is provided in logger errors for ao2 bad magic numbers. + * Optimizes AO2 by merging internal functions with the public counterpart. + This was possible now that we no longer require a dual ABI. + + ASTERISK-24974 #close + Reported by: Corey Farrell + + Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1 + +2015-04-27 12:11 +0000 [356568dc7f] gtjoseph + + * res_pjsip: Fix SEGV on pending-qualify contacts + + Permanent contacts that hadn't been qualified yet were missing + their contact_status entries causing SEGVs when running CLI + commands. + + This patch makes sure that contact_statuses are created for + both dynamic and permanent contacts when they are created. + It also adds checks in the CLI code to make sure there's a + contact_status, just in case. + + ASTERISK-25018 #close + Reported-by: Ivan Poddubny + Tested-by: Ivan Poddubny + Tested-by: George Joseph + + Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029 + +2015-04-15 18:55 +0000 [358080e86e] Rodrigo Ramírez Norambuena + + * cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version + + Add new column to INSERT new columns added in cdr 1.8 version. The columns are: + * peeraccount + * linkedid + * sequence + This feature is configurable in cdr_odbc.conf using a new configuration + option, 'newcdrcolumns'. + + ASTERISK-24976 #close + + Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127 +2015-04-26 17:21 +0000 [d7f4788341] Matt Jordan + + * channels/chan_skinny: Fix compilation error introduced in f8e21a1adf + + A typo in commit f8e21a1adf resulted in a compilation error in + chan_skinny. This patch fixes the typo. + + ASTERISK-24917 + + Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c + +2015-04-23 17:29 +0000 [9f65ea482e] Kevin Harwell + + * app_confbridge: Default the template option to a compatible default profile. + + Confbridge dynamic profiles did not have a default profile unless you + explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a + template was not set prior to the bridge being created then some + options were left with no default values set. This patch makes it so + the default templates are set to the default bridge and user profiles. + + ASTERISK-24749 #close + Reported by: philippebolduc + + Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a + +2015-04-23 07:31 +0000 [cafdb7a049] Olle E. Johansson + + * CREDITS: Update credits for Olle Johansson + + Change-Id: I8f3d0a6c3f1075a1f7d8308593394611a96749de +2015-04-24 09:17 +0000 [bd61c9300c] Mark Michelson + + * res_pjsip_outbound_authenticator: Increase CSeq on authed requests. + + The way PJSIP generates an authenticated request is to use a previous + request as a template. This means that the authenticated request will + have the same Call-ID, From header (including tag), and CSeq as the + original request. PJSIP generates a new branch on the Via header to + indicate that this is a new transaction, though. + + There are some SIP implementations, though, that do not notice the + change in the branch and therefore will match the authed request to the + original request's transaction. Since the CSeq is the same, the server + will repeat the response it sent to the original request. + + This patch aids interoperability by increasing the CSeq of the authed + request by one. + + ASTERISK-24845 #close + Reported by: Carl Fortin + Tested by: Carl Fortin + + Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01 + +2015-04-22 04:17 +0000 [f8e21a1adf] Diederik de Groot + + * Clang: Fix some more tautological-compare warnings. + + clang can warn about a so called tautological-compare, when it finds + comparisons which are logically always true, and are therefor deemed + unnecessary. + + Exanple: + unsigned int x = 4; + if (x > 0) // x is always going to be bigger than 0 + + Enum Case: + Each enumeration is its own type. Enums are an integer type but they + do not have to be *signed*. C leaves it up to the compiler as an + implementation option what to consider the integer type of a particu- + lar enumeration is. Gcc treats an enum without negative values as + an int while clang treats this enum as an unsigned int. + + rmudgett & mmichelson: cast the enum to (unsigned int) in assert. + The cast does have an effect. For gcc, which seems to treat all enums + as int, the cast to unsigned int will eliminate the possibility of + negative values being allowed. For clang, which seems to treat enums + without any negative members as unsigned int, the cast will have no + effect. If for some reason in the future a negative value is ever + added to the enum the assert will still catch the negative value. + + ASTERISK-24917 + Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62 + +2015-04-20 13:06 +0000 [1e74793061] Diederik de Groot + + * Example script for scan-build (the llvm static analyzer) + + - Added Pre-amble (Options / Flags / Usage Example / GNU License) + - Extended Configurability + - Made Executable + + ASTERISK-24917 + Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8 + +2015-04-23 12:54 +0000 [89a3fc0572] Mark Michelson + + * res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX. + + When Asterisk originates a channel to an application, the channel is + hung up once the application finishes executing. When the application + in question is SendFax, the Asterisk PJSIP code will attempt to reinvite + the T.38 session to audio after the FAX completes. The hangup of the + channel happens in the midst of this reinvite transaction. In most + circumstances, this works out okay because the BYE is delayed until the + reinvite transaction can complete. + + However, if the reinvite that Asterisk sends receives a 401/407 + response, then Asterisk's attempt to re-send the reinvite with + authentication will fail. This is because the session supplement in + res_pjsip_t38 makes the assumption that the channel on the session will + always be non-NULL. Since the channel has been hung up, though, the + channel is now NULL. Attempting to operate on the channel causes a + crash. + + This patch fixes the issue by ensuring that the channel on the session + is not NULL before attempting to mess with the T.38 framehook. + + This patch also contains some corrections for comments that were + incorrect and really confused me when I first started looking at the + code. + + ASTERISK-25004 #close + Reported by Mark Michelson + + Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0 + +2015-04-23 09:16 +0000 [75666ad7c6] gtjoseph + + * res_pjsip: Validate that contact uris start with sip: or sips: + + Currently we use pjsip_parse_hdr to validate contact uris but it + appears that it allows uris without a scheme if there's a port + supplied. I.E myexample.com will fail but myexample.com:5060 will + pass even though it has no scheme. This causes SEGVs later on + whenever the uri is used. + + To prevent this, permanent_contact_validate has been updated to check + that the scheme is either 'sip' or 'sips'. + + 2 uses of possibly-null endpoint have also been fixed in + create_out_of_dialog_request. + + ASTERISK-24999 + + Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 + Reported-by: Brad Latus + +2015-04-23 08:00 +0000 [ca7193167e] Diederik de Groot + + * Clang: change previous tautological-compare fixes. + + clang can warn about a so called tautological-compare, when it finds + comparisons which are logically always true, and are therefor deemed + unnecessary. + + Exanple: + unsigned int x = 4; + if (x > 0) // x is always going to be bigger than 0 + + Enum Case: + Each enumeration is its own type. Enums are an integer type but they + do not have to be *signed*. C leaves it up to the compiler as an + implementation option what to consider the integer type of a particu- + lar enumeration is. Gcc treats an enum without negative values as + an int while clang treats this enum as an unsigned int. + + rmudgett & mmichelson: cast the enum to (unsigned int) in assert. + The cast does have an effect. For gcc, which seems to treat all enums + as int, the cast to unsigned int will eliminate the possibility of + negative values being allowed. For clang, which seems to treat enums + without any negative members as unsigned int, the cast will have no + effect. If for some reason in the future a negative value is ever + added to the enum the assert will still catch the negative value. + + ASTERISK-24917 + + Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a + +2015-04-22 16:22 +0000 [cc77440deb] gtjoseph + + * res_corosync: Add check for config file before calling corosync apis + + On some systems, res_corosync isn't compatible with the installed version of + corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE, + and Asterisk terminates. The work around has been to remember to add + res_corosync as a noload in modules.conf. A better solution though is to have + res_corosync check for its config file before attempting to call corosync apis + and return LOAD_DECLINE if there's no config file. This lets Asterisk loading + continue. + + If you have a res_corosync.conf file and res_corosync fails, you get the same + behavior as today and the fatal error tells you something is wrong with the + install. + + ASTERISK-24998 + + Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889 +2015-04-22 15:17 +0000 [c231c85ea4] Corey Farrell + + * Astobj2: Ensure all calls to __adjust_lock pass a valid object. + + __adjust_lock doesn't check for invalid objects, and doesn't have an + appropriate return value for invalid objects. Most callers of + __adjust_lock pass objects that have already been confirmed valid, + this change adds checks before the remaining calls. + + ASTERISK-24997 #close + Reported by: Corey Farrell + + Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f + +2015-04-22 16:32 +0000 [0722e11f26] gtjoseph + + * .gitignore: Add .gcno and .gcda + + Products of --enable-coverage + + Change-Id: Ie20882d64b60692e2c941ea8872ab82a86ce77a3 + +2015-04-22 11:28 +0000 [7216e3c608] Joshua Colp + + * dns: Make query sets hold on to queries for their lifetime. + + The query set documentation states that upon completion queries can be + retrieved for the lifetime of the query set. This is a reasonable + expectation but does not currently occur. This was originally done + to resolve a circular reference between queries and query sets, but + in practice the query can be kept. + + This change makes it so a query does not have a reference to the + query set until it begins resolving. It also makes it so that the + reference is given up upon the query being completed. This allows + the queries to remain for the lifetime of the query set. As the + query set on the query is only useful to the query set functionality + and only for the lifetime that the query is resolving this is safe + to do. + + ASTERISK-24994 #close + Reported by: Joshua Colp + + Change-Id: I54e09c0cb45475896654e7835394524e816d1aa0 + +2015-04-20 13:01 +0000 [09c7c678a3] Diederik de Groot + + * Fix/Update clang-RAII macro implementation + + - When you need to refer to 'variable XXX' outside a block, it needs + to be declared as '__block XXX', otherwise it will not be available with- + in the block, making updating that variable hard to do, and ast_free + lead to issues. + + - Removed the #error message + because it creates complications when compiling external projects + against asterisk For example when using a different compiler than the + one used to compile asterisk. The warning/error should be generated + during the configure process not the compilation process + + ASTERISK-24917 + Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2 + +2015-04-14 14:04 +0000 [190fa4f333] Joshua Colp + + * res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers. + + Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon + a mailbox state change (such as a new message being left, or one being deleted). + In practice this is not sufficient to keep clients aware of the current MWI status. + + This change makes the module send unsolicited MWI NOTIFY on startup so that + clients are guaranteed to have the most up to date MWI information. It also makes + clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware + of the current MWI status they receive it. + + ASTERISK-24982 #close + Reported by: Joshua Colp + + Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58 + +2015-04-21 17:45 +0000 [2a36bb5d9a] Rodrigo Ramírez Norambuena + + * CHANGES remove tab space + + Change-Id: I6b43e43474bf6fb77b8227eadb036036f8e90521 + +2015-04-21 15:17 +0000 [5757d2d30d] Corey Farrell + + * Check for ao2_alloc failure in __ast_channel_internal_alloc. + + Fix a crash that could occur in __ast_channel_internal_alloc if + ao2_alloc fails. + + ASTERISK-24991 #close + + Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90 + +2015-04-20 14:30 +0000 [6331be0638] Mark Michelson + + * res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. + + When SUBSCRIBE dialogs were established, we never associated + the endpoint that created the subscription with the dialog + we end up creating. In most cases, this ended up not causing + any problems. + + The actual bug that was observed was that when a device that + was behind NAT established a subscription with Asterisk, Asterisk + would end up sending in-dialog NOTIFY requests to the device's + private IP addres instead of the public address of the NAT router. + + When Asterisk receives the initial SUBSCRIBE from the device, + res_pjsip_nat rewrites the contact to the public address on which the + SUBSCRIBE was received. This allows for the dialog to have its target + address set to the proper public address. Asterisk then would send a 200 + OK response to the SUBSCRIBE, then a NOTIFY with the initial + subscription state. The device would then send a 200 OK response to + Asterisk's NOTIFY. + + Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat + did not rewrite the address in the Contact header. Then, when the PJSIP + dialog layer processed the 200 OK, PJSIP would perform a comparison + between the IP address in the Contact header and its saved target + address for the dialog. Since they differed, PJSIP would update the + target dialog address to be the address in the Contact header. From this + point, if Asterisk needed to send a NOTIFY to the device, the result was + that the NOTIFY would be sent to the private address that the device + placed in the Contact header. + + The reason why res_pjsip_nat did not rewrite the address when it + received the 200 OK response was that it could not associate the + incoming response with a configured endpoint. This is because on a + response, the only way to associate the response to an endpoint is by + finding the dialog that the response is associated with and then finding + the endpoint that is associated with that dialog. We do not perform + endpoint lookups on responses. res_pjsip_pubsub skipped the step of + associating the endpoint with the dialog we created, so res_pjsip_nat + could not find the associated endpoint and therefore couldn't rewrite + the contact. + + This commit message is like 50x longer than the actual fix. + + ASTERISK 24981 #close + Reported by Mark Michelson + + Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd +2015-04-16 22:34 +0000 [2f418c052e] Gareth Palmer + + * New AMI Command Output Format + + This change modifies how the the output from a CLI command is sent + to a client over AMI. + + Output from the CLI command is now sent as a series of zero-or-more + Output: headers. + + Additionally, commands that fail to execute (eg: no such command, + invalid syntax etc.) now cause an Error response instead of Success. + + If the command executed successfully, but the manager unable to + provide the output the reason will be included in the Message: + header. Otherwise it will contain 'Command output follows'. + + Depends on a new version of starpy (> 1.0.2) that supports the new + output format. + + See pull-request https://github.com/asterisk/starpy/pull/34 + + ASTERISK-24730 + + Change-Id: I6718d95490f0a6b3f171c1a5cdad9207f9a44888 +2015-04-20 18:00 +0000 [614f506690] Richard Mudgett + + * chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels. + + The chan_dahdi channel driver is a very old driver. The ability for it to + support ISDN was added well after the initial analog support. Setting the + softhangup flags is a carry over from the original analog code. The + driver was not updated to call ast_queue_hangup() which will post the AMI + HangupRequest event. + + * Changed sig_pri.c to call ast_queue_hangup() instead of setting the + softhangup flag when the remote party initiates a hangup. + + ASTERISK-24895 #close + Reported by: Andrew Zherdin + + Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325 + +2015-04-20 13:40 +0000 [bff3064578] Rodrigo Ramírez Norambuena + + * cdr/cdr_adaptive_odbc.c: Refactor concatenate columns name. + + The concatenate for columns name to INSERT INTO is always the same. It is + possible to do it on one line. + + ASTERISK-24980 + + Change-Id: Ib8bb53c42535378581d4ef729cc5ebbb22b067ac +2015-04-20 09:53 +0000 [06ba1e59cb] gtjoseph + + * pjsip_options: Fix format specifier for int64_t rtt. + + Contact status rtt is an int64_t and needs the PRId64 macro to + properly create the format specifier on 32-bit systems. + + Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7 + +2015-04-18 13:36 +0000 [298faf7c50] gtjoseph + + * pjsip_options: Fix non-qualified contacts showing as unavailable + + The "Add qualify_timeout processing and eventing" patch introduced + an issue where contacts that had qualify_frequency set to 0 were + showing Unavailable instead Unknown. This patch checks for + qualify_frequency=0 and create an "Unknown" contact_status + with an RTT = 0. + + Previously, the lack of contact_status implied Unknown but since + we're now changing endpoint state based on contact_status, I've + had to add new UNKNOWN status so that changes could trigger the + appropriate contact_status observers. + + ASTERISK-24977: #close + + Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7 + +2015-04-19 15:49 +0000 [8e903b17ea] Matt Jordan + + * main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple + + When a PBX registrar is unloaded, it will fail to remove its extension from + the context root_table if a dialplan application used by that extension is + still loaded. This can be the case for AGI, which can be unloaded after several + of the standard PBX providers. Often, this is harmless; however, if the + extension's priorities are removed during the failed unloading *and* the + dialplan application later unregisters, it leaves a ticking timebomb for the + next PBX provider that attempts to iterate over the extensions. When that + occurs, the peer_table pointer on the extension will already be set to NULL. + The current code does not check to see if the pointer is NULL before passing + it to a hashtab function this is not NULL tolerant. + + Since it is possible for the peer_table to be NULL when we normally would not + expect that to be the case, the solution in this patch is to simply skip over + processing an extension's priorities if peer_table is NULL. + + Prior to this patch, the tests/pbx/callerid_match test would crash during + module unload. With this patch, the test no longer crashes after running. + + ASTERISK-24774 #close + Reported by: Corey Farrell + + Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40 + +2015-04-17 18:05 +0000 [1269dd06bc] Richard Mudgett + + * res_fax: Fix latent bug exposed by ASTERISK-24841 changes. + + Three fax related tests started failing as a result of changes made for + ASTERISK-24841: + tests/fax/pjsip/gateway_t38_g711 + tests/fax/sip/gateway_mix1 + tests/fax/sip/gateway_mix3 + + Historically, ast_channel_make_compatible() did nothing if the channels + were already "compatible" even if they had a sub-optimal translation path + already setup. With the changes from ASTERISK-24841 this is no longer + true in order to allow the best translation paths to always be picked. In + res_fax.c:fax_gateway_framehook() code manually setup the channels to go + through slin and then called ast_channel_make_compatible(). With the + previous version of ast_channel_make_compatible() this was always a + no-operation. + + * Remove call to ast_channel_make_compatible() in fax_gateway_framehook() + that now undoes what was just setup when the framehook is attached. + + * Fixed locking around saving the channel formats in + fax_gateway_framehook() to ensure that the formats that are saved are + consistent. + + * Fix copy pasta errors in fax_gateway_framehook() that confuses read and + write when dealing with saved channel formats. + + ASTERISK-24841 + Reported by: Matt Jordan + + Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d + +2015-04-17 16:19 +0000 [c1d44ff043] Corey Farrell + + * Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled. + + When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be + called as a function. This causes a compile error with raw threadstorage as + it uses NULL for cleanup. This fix uses a macro that provides NULL when + DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);" + with "{};" when DEBUG_THREADLOCALS is enabled. + + ASTERISK-24975 #close + Reported by: Ashley Sanders + + Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402 + +2015-04-15 10:38 +0000 [aae45acbda] Mark Michelson + + * Detect potential forwarding loops based on count. + + A potential problem that can arise is the following: + + * Bob's phone is programmed to automatically forward to Carol. + * Carol's phone is programmed to automatically forward to Bob. + * Alice calls Bob. + + If left unchecked, this results in an endless loops of call forwards + that would eventually result in some sort of fiery crash. + + Asterisk's method of solving this issue was to track which interfaces + had been dialed. If a destination were dialed a second time, then + the attempt to call that destination would fail since a loop was + detected. + + The problem with this method is that call forwarding has evolved. Some + SIP phones allow for a user to manually forward an incoming call to an + ad-hoc destination. This can mean that: + + * There are legitimate use cases where a device may be dialed multiple + times, or + * There can be human error when forwarding calls. + + This change removes the old method of detecting forwarding loops in + favor of keeping a count of the number of destinations a channel has + dialed on a particular branch of a call. If the number exceeds the + set number of max forwards, then the call fails. This approach has + the following advantages over the old: + + * It is much simpler. + * It can detect loops involving local channels. + * It is user configurable. + + The only disadvantage it has is that in the case where there is a + legitimate forwarding loop present, it takes longer to detect it. + However, the forwarding loop is still properly detected and the + call is cleaned up as it should be. + + Address review feedback on gerrit. + + * Correct "mfgium" to "Digium" + * Decrement max forwards by one in the case where allocation of the + max forwards datastore is required. + * Remove irrelevant code change from pjsip_global_headers.c + + ASTERISK-24958 #close + + Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23 +2015-04-16 10:51 +0000 [56a2baa21d] Kevin Harwell + + * bridge.c: NULL app causes crash during attended transfer + + Due to a race condition there was a chance that during an attended transfer the + channel's application would return NULL. This, of course, would cause a crash + when attempting to access the memory. This patch retrieves the channel's app + at an earlier time in processing in hopes that the app name is available. + However, if it is not then "unknown" is used instead. Since some string value + is now always present the crash can no longer occur. + + ASTERISK-24869 #close + Reported by: viniciusfontes + Review: https://gerrit.asterisk.org/#/c/133/ + + Change-Id: I5134b84c4524906d8148817719d76ffb306488ac + +2015-04-11 17:04 +0000 [c6ed681638] gtjoseph + + * res_pjsip: Add global option to limit the maximum time for initial qualifies + + Currently when Asterisk starts initial qualifies of contacts are spread out + randomly between 0 and qualify_timeout to prevent network and system overload. + If a contact's qualify_frequency is 5 minutes however, that contact may be + unavailable to accept calls for the entire 5 minutes after startup. So while + staggering the initial qualifies is a good idea, basing the time on + qualify_timeout could leave contacts unavailable for too long. + + This patch adds a new global parameter "max_initial_qualify_time" that sets the + maximum time for the initial qualifies. This way you could make sure that all + your contacts are initialy, randomly qualified within say 30 seconds but still + have the contact's ongoing qualifies at a 5 minute interval. + + If max_initial_qualify_time is > 0, the formula is initial_interval = + min(max_initial_interval, qualify_timeout * random(). If not set, + qualify_timeout is used. + + The default is "0" (disabled). + + ASTERISK-24863 #close + + Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 + Tested-by: George Joseph + +2015-04-16 13:20 +0000 [664d3263e4] Scott Griepentrog + + * res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced + + This change makes the send_notify of the sub_tree + not happen when the sub_tree has been deleted due + to the notify call failing, which avoids a crash. + + ASTERISK-24970 #close + + Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf +2015-04-11 16:56 +0000 [51886c68dc] gtjoseph + + * pjsip_options: Add qualify_timeout processing and eventing + + This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the + discussion at + http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html + + The basic issues are that changes in contact status don't cause events to be + emitted for the associated endpoint. Only dynamic contact add/delete actions + update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds + which is a long time. + + This patch makes use of the new transaction timeout feature in r4585 and + provides the following capabilities... + + 1. A new aor/contact variable 'qualify_timeout' has been added that allows the + user to specify the maximum time in milliseconds to wait for a response to an + OPTIONS message. The default is 3000ms. When the timer expires, the contact is + marked unavailable. + + 2. Contact status changes are now propagated up to the endpoint as follows... + When any contact is 'Available', the endpoint is marked as 'Reachable'. When + all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The + existing endpoint events are generated appropriately. + + ASTERISK-24863 #close + + Change-Id: Id0ce0528e58014da1324856ea537e7765466044a + Tested-by: Dmitriy Serov + Tested-by: George Joseph + +2015-04-11 16:39 +0000 [ab6382cafd] gtjoseph + + * res_pjsip: Refactor endpt_send_request to include transaction timeout + + This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the + discussion at + http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html + + Since we currently have no control over pjproject transaction timeout, this + patch pulls the pjsip_endpt_send_request function out of pjproject and into + res_pjsip/endpt_send_transaction in order to implement that capability. + + Now when the transaction is initiated, we also schedule our own pj_timer with + our own desired timeout. + + If the transaction completes before either timeout, pjproject cancels its timer, + and calls our tsx callback where we cancel our timer and run the app callback. + + If the pjproject timer times out first, pjproject calls our tsx callback where + we cancel our timer and run the app callback. + + If our timer times out first, we terminate the transaction which causes + pjproject to cancel its timer and call our tsx callback where we run the app + callback. + + Regardless of the scenario, pjproject is calling the tsx callback inside the + group_lock and there are checks in the callback to make sure it doesn't run + twice. + + As part of this patch ast_sip_send_out_of_dialog_request was created to replace + its similarly named private function. It takes a new timeout argument in + milliseconds (<= 0 to disable the timeout). + + ASTERISK-24863 #close + Reported-by: George Joseph + Tested-by: George Joseph + + Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747 +2015-04-15 16:08 +0000 [043c38f6de] gtjoseph + + * More .gitignore updates + + Added .pyc and .sha1 to the top-level .gitignore. + + Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a + Tested-by: George Joseph + +2015-04-14 02:36 +0000 [abf10a1d4c] Corey Farrell + + * Build System: Enable use of ~/.asterisk.makeopts and /etc/asterisk.makeopts. + + The Makefile claims that you can set default menuselect options by creating + ~/.asterisk.makeopts or /etc/asterisk.makeopts, but they are never read. + The rule for menuselect.makeopts is only allowed to run if the active target + is 'menuselect', but the menuselect target doesn't depend on + menuselect.makeopts. A dot (wildcard character) was added so the rule will + be active for the targets that cause it to run: nmenuselect, cmenuselect, + and gmenuselect. + + ASTERISK-13271 #close + Reported by: John Nemeth + + Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2 +2015-04-13 08:47 +0000 [a3cec44a0a] Joshua Colp + + * res_pjsip: Add external PJSIP resolver implementation using core DNS API. + + This change adds the following: + + 1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked. + 2. Unit tests for the query set implementation. + 3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups. + + For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV + are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A, + with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit + transport has been provided. Configured transports on the system are taken into account to + eliminate resolved addresses which have no hope of completing. + + ASTERISK-24947 #close + Reported by: Joshua Colp + + Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e + +2015-04-14 13:16 +0000 [33a319ae73] Rodrigo Ramírez Norambuena + + * cel_pgsql: Fix name string for log on unable allocate memory. + + The LOG_ERROR has reference to CDR instead of CEL for LENGTHEN_BUF1 and + LENGTHEN_BUF2. + + ASTERISK-24965 #close + Reported by: Rodrigo Ramirez Norambuena + + Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744 +2015-04-14 15:59 +0000 [f89481e39c] Corey Farrell + + * test_astobj2_weaken: Fix source file registration. + + Update test_astobj2_weaken to use the new AST_REGISTER_FILE macro. + + Change-Id: Ieedadf16610f2e042f393e0501a36447cd07f83d + +2015-04-13 05:28 +0000 [62508d6891] Corey Farrell + + * Build System: Create Makefile macro MOD_ADD_SOURCE. + + This new macro allows a single line to add all additional + sources to a module. This helps prevent modules from + missing steps, and makes future changes easier since + they can be made in a single place. + + ASTERISK-24960 #close + Reported by: Corey Farrell + + Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b + +2015-04-12 09:08 +0000 [23a180cade] Rodrigo Ramírez Norambuena + + * cdr_pgsql: Fix CLI "cdr show pgsql status" command. + + The command always showed the usage information. + + * Fix the error in command validation for CLI_SHOWUSAGE. + + ASTERISK-24959 #close + Reported by: Rodrigo Ramirez Norambuena + + Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5 +2015-04-13 19:06 +0000 [bf46ef35ca] gtjoseph + + * .gitignore updates for master/13 + + Added products of ./bootstrap + + Added nmenuselect and gmenuselect to menuselect/ + + Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35 + +2015-04-13 06:52 +0000 [62e95065d6] Corey Farrell + + * AMI: Fix improper handling of lines that are exactly 1025 bytes long. + + When AMI receives a line that is 1025 bytes long, it sends two error + messages. Copy the last byte in the buffer to the first postiion, + set the length to 1. + + ASTERISK-20524 #close + Reported by: David M. Lee + + Change-Id: Ifda403e2713b59582c715229814fd64a0733c5ea + +2015-04-12 03:22 +0000 [cb6bf3094e] Corey Farrell + + * astobj2: Add support for weakproxy objects. + + This implements "weak" references. The weakproxy object is a real ao2 with + normal reference counting of its own. When a weakproxy is pointed to a normal + object they hold references to each other. The normal object is automatically + freed when a single reference remains (the weakproxy). The weakproxy also + supports subscriptions that will notify callbacks when it does not point + to any real object. + + ASTERISK-24936 #close + Reported by: Corey Farrell + + Change-Id: Ib9f73c02262488d314d9d9d62f58165b9ec43c67 + +2015-04-13 14:41 +0000 [a573b77f78] David M. Lee + + * Fixing extconf compile + + During the mass code deletion for clang support, a stray backslash was + left behind that was causing utils to fail to compile. + + Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1 + +2015-04-13 09:54 +0000 [3f9aa29945] Matt Jordan + + * build_tools/make_version: Update version parsing for Git migration + + External systems - such as the Asterisk Test Suite - require knowledge of the + upstream branch. Unfortunately, after moving to Git, the Asterisk version + currently consists of only a 'GIT" prefix followed by an object blob, + e.g., GIT-as08d7. This makes it difficult for such systems to know what + features are available in a particular check out of Asterisk. + + This patch fixes this by hardcoding the branch in a variable in the + make_version script. Since the mainline branches are not changed often - + typically only once a year - this is a reasonable approach to solving + the problem, and is more reliable than parsing the output of 'git branch + -vv'. Branches that track off of an upstream primary branch will then get the + benefit of knowing which mainline branch they are currently based off + of. + + ASTERISK-24954 #close + + Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799 + +2015-04-13 05:57 +0000 [fbc8ddfe63] Corey Farrell + + * Optional API: Fix handling of sources that are both provider and user. + + OPTIONAL_API has conditionals to define AST_OPTIONAL_API and + AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined. + Unfortunately this is inside the include protection block, so only the + first status of AST_API_MODULE is respected. For example res_monitor + is an optional API provider, but uses func_periodic_hook. This makes + func_periodic_hook non-optional to res_monitor. + + This changes optional_api.h so that AST_OPTIONAL_API and + AST_OPTIONAL_API_ATTR is redefined every time the header is included. + + ASTERISK-17608 #close + Reported by: Warren Selby + + Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679 + +2015-04-11 21:38 +0000 [4a58261694] Matt Jordan + + * git migration: Refactor the ASTERISK_FILE_VERSION macro + + Git does not support the ability to replace a token with a version + string during check-in. While it does have support for replacing a + token on clone, this is somewhat sub-optimal: the token is replaced + with the object hash, which is not particularly easy for human + consumption. What's more, in practice, the source file version was often + not terribly useful. Generally, when triaging bugs, the overall version + of Asterisk is far more useful than an individual SVN version of a file. As a + result, this patch removes Asterisk's support for showing source file + versions. + + Specifically, it does the following: + + * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and + remove passing the version in with the macro. Other facilities + than 'core show file version' make use of the file names, such as + setting a debug level only on a specific file. As such, the act of + registering source files with the Asterisk core still has use. The + macro rename now reflects the new macro purpose. + + * main/asterisk: + - Refactor the file_version structure to reflect that it no longer + tracks a version field. + - Remove the "core show file version" CLI command. Without the file + version, it is no longer useful. + - Remove the ast_file_version_find function. The file version is no + longer tracked. + - Rename ast_register_file_version/ast_unregister_file_version to + ast_register_file/ast_unregister_file, respectively. + + * main/manager: Remove value from the Version key of the ModuleCheck + Action. The actual key itself has not been removed, as doing so would + absolutely constitute a backwards incompatible change. However, since + the file version is no longer tracked, there is no need to attempt to + include it in the Version key. + + * UPGRADE: Add notes for: + - Modification to the ModuleCheck AMI Action + - Removal of the "core show file version" CLI command + + Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e + +2015-04-12 06:12 +0000 [5d34bce635] Corey Farrell + + * main/editline: Add .gitignore. + + This patch adds a .gitignore for main/editline to ignore all build results. + + Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d + +2015-04-11 23:22 +0000 [d6605b3c10] Matt Jordan + + * .gitignore: Ignore tarballs (*.gz) + + This patch updates the root .gitignore file to ignore files with a .gz + extension. This will cause git to ignore downloaded sound tarballs in + the the sounds/ directory. + + Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 + +2015-04-11 13:20 +0000 [b35e184d41] gtjoseph + + * Add .gitignore and .gitreview files + + Add the .gitignore and .gitreview files to the asterisk repo. + + NB: You can add local ignores to the .git/info/exclude file + without having to do a commit. + + Common ignore patterns are in the top-level .gitignore file. + Subdirectory-specific ignore patterns are in their own .gitignore + files. + + Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 + Tested-by: George Joseph + +2015-04-11 10:27 +0000 [356b770632] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix various warnings for tests + + This patch fixes a variety of clang compiler warnings for unit tests. This + includes autological comparison issues, ignored return values, and + interestingly enough, one embedded function. Fun! + + Review: https://reviewboard.asterisk.org/r/4555 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4555.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434705 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434706 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-11 10:11 +0000 [5f181bcccd] Juergen Spies (License 6698) + + * res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram + + Prior to this patch, the far_max_datagram value on the UDPTL structure would + remain -1 if the remote endpoint fails to provide the SDP media attribute + T38FaxMaxDatagram. This can result in the INVITE request being rejected. With + this patch, we will now properly initialize the value with either the default + value or with the value provided by pjsip.conf's t38_udptl_maxdatagram + parameter. + + Review: https://reviewboard.asterisk.org/r/4589 + + ASTERISK-24928 #close + Reported by: Juergen Spies + Tested by: Juergen Spies + patches: + pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698) + ........ + + Merged revisions 434688 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 18:37 +0000 [c499cabf53] Richard Mudgett + + * chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. + + With this patch, chan_pjsip/res_pjsip now sets the native formats to the + codecs negotiated by a call. + + * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native + formats to include all the negotiated audio codecs instead of only the + initial preferred audio codec and later the currently received audio + codec. + + * The audio frame handling in channel.c:ast_read() is more streamlined and + will automatically adjust to changes in received frame formats. The new + policy is to remove translation and pass the new frame format to the + receiver except if the translation was to a signed linear format. A more + long winded version is commented in ast_read() along with some caveats. + + * The audio frame handling in channel.c:ast_write() is more streamlined + and will automatically adjust any needed translation to changes in the + frame formats sent. Frame formats sent can change for many reasons such + as a recording is being played back or the bridged peer changed the format + it sends. Since it is a normal expectation that sent formats can change, + the codec mismatch warning message is demoted to a debug message. + + * Removed the short circuit check in + channel.c:ast_channel_make_compatible_helper(). Two party bridges need to + make channels compatible with each other. However, transfers and moving + channels among bridges can result in otherwise compatible channels having + sub-optimal translation paths if the make compatible check is short + circuited. A result of forcing the reevaluation of channel compatibility + is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc + options take effect consistently now. It is unfortunate that these two + options are enabled by default and negate some of the benefits to the + changes in channel.c:ast_read() by forcing translation through signed + linear on a two party bridge. + + * Improved the softmix bridge technology to better control the translation + of frames to the bridge. All of the incoming translation is now normally + handled by ast_read() instead of splitting any translation steps between + ast_read() and the slin factory. If any frame comes in with an unexpected + format then the translation path in ast_read() is updated for the next + frame and the slin factory handles the current frame translation. + + This is the final patch in a series of patches aimed at improving + translation path choices. The other patches are on the following reviews: + https://reviewboard.asterisk.org/r/4600/ + https://reviewboard.asterisk.org/r/4605/ + + ASTERISK-24841 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4609/ + ........ + + Merged revisions 434671 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 16:06 +0000 [66f3fd0028] Kevin Harwell + + * chan_sip: make progressinband default to no + + After the "progressinband" value setting of "never" was updated to never send a + 183 this separated its use from the "no" value. Since "never" was the default, + but most users probably expect "no" this patch updates the default for the + "progressinband" setting to "no." + + ASTERISK-24835 #close + Reported by: Andrew Nagy + Review: https://reviewboard.asterisk.org/r/4606/ + ........ + + Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 12:56 +0000 [8bae18ab93] yaron nahum (License 6676) + + * res_pjsip: Add an 'auto' option for DTMF Mode + + This patch adds support for automatically detecting the type of DTMF that a + PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', + the channel created for an endpoint will attempt to determine if RFC 4733 + DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type + for the channel will be set to inband. + + Review: https://reviewboard.asterisk.org/r/4438 + + ASTERISK-24706 #close + Reported by: yaron nahum + patches: + yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) + ........ + + Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 12:00 +0000 [f69e46de25] gtjoseph + + * res_pjsip_config_wizard: Cleanup load unload + + While investigating other unload issues I realized that the load/unload process + for the config wizard was pretty ugly so I've refactored it as follows... + + When the res_pjsip sorcery instance is created the config_wizard bumps it's own + module reference to prevent it from unloading while the sorcery instance is + still active. When res_pjsip unloads and it's sorcery instance is destroyed, + the config wizard unrefs itself which then allows itself to unload cleanly. + Since the config wizard now can't load after res_pjsip or unload before it + (which should have been the correct behavior all along), I was able to remove + the chunks of code in both load_module and unload_module that handled that case. + + Ran the testsuite tests to insure there were no functional changes and REF_DEBUG + to insure that Asterisk was shutting down cleanly with no FRACKs or leaks. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4610/ + ........ + + Merged revisions 434619 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 11:38 +0000 [6f1a7fe05f] Richard Mudgett + + * bridge_softmix.c,channel.c: Minor code simplification and cleanup. + + * Made code easier to follow in bridge_softmix.c:analyse_softmix_stats() + and made some debug messages more helpful. + + * Made some debug and warning messages more helpful in + channel.c:set_format(). + + Review: https://reviewboard.asterisk.org/r/4607/ + ........ + + Merged revisions 434617 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434618 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 11:32 +0000 [0b805cb875] Richard Mudgett + + * translate.c: Only select audio codecs to determine the best translation choice. + + Given a source capability of h264 and ulaw, a destination capability of + h264 and g722 then ast_translator_best_choice() would pick h264 as the + best choice even though h264 is a video codec and Asterisk only supports + translation of audio codecs. When the audio starts flowing, there are + warnings about a codec mismatch when the channel tries to write a frame to + the peer. + + * Made ast_translator_best_choice() only select audio codecs. + + * Restore a check in channel.c:set_format() lost after v1.8 to prevent + trying to set a non-audio codec. + + This is an intermediate patch for a series of patches aimed at improving + translation path choices for ASTERISK-24841. + + This patch is a complete enough fix for ASTERISK-21777 as the v11 version + of ast_translator_best_choice() does the same thing. However, chan_sip.c + still somehow tries to call ast_codec_choose() which then calls + ast_best_codec() with a capability set that doesn't contain any audio + formats for the incoming call. The remaining warning message seems to be + a benign transient. + + ASTERISK-21777 #close + Reported by: Nick Ruggles + + ASTERISK-24380 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4605/ + ........ + + Merged revisions 434614 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434615 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 09:56 +0000 [894153b8b1] Matt Jordan + + * res/ari: Fix model validation for ChannelHold event + + When the ChannelHold event was added, the 'musicclass' parameter was + erroneously removed. This caused the ChannelHold events to be rejected as + they failed model validation. This patch updates the Swagger schema such that + it now properly reflects the event that is being created. + + Hooray for tests that catch things like this. + ........ + + Merged revisions 434597 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 08:32 +0000 [02a0a4d65f] Joshua Colp + + * dns: Fix build when TEST_FRAMEWORK is not defined. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 07:40 +0000 [80c443bea4] Y Ateya (License 6693) + + * channels/chan_iax2: Improve POKE expiration time calculation for lossy networks + + POKE is used to check for peer availability; however, in networks with packet + loss, the current calculations may result in POKE expiration times that are too + short. This patch alters the expiration/retry time logic to take into account + the last known qualify round trip time, as opposed to always using a static + value for each peer. + + Review: https://reviewboard.asterisk.org/r/4536 + + ASTERISK-22352 #close + Reported by: Frederic Van Espen + + ASTERISK-24894 #close + Reported by: Y Ateya + patches: + poke_noanswer_duration.diff submitted by Y Ateya (License 6693) + ........ + + Merged revisions 434564 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434565 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-10 07:23 +0000 [b3d01f1fbf] Y Ateya (License 6693) + + * channels/chan_iax2: Add a configuration parameter for call token expiration + + This patch adds a new configuration parameter, 'calltokenexpiration', that + controls how long before an authentication call token is expired. The default + maintains the RFC specified 10 seconds. Setting it to a higher value may be + useful in lossy networks. + + Review: https://reviewboard.asterisk.org/r/4588 + + ASTERISK-24939 #close + Reported by: Y Ateya + patches: + ctoken_configuration.diff submitted by Y Ateya (License 6693) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 18:12 +0000 [ed6b6e3c03] gtjoseph + + * res_pjsip_phoneprov_provider: Fix reference leak on unload + + res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to + a missing OBJ_NODATA in an ao2_callback in load_users(). Rather than adding the + OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator. + This plugged the leak but exposed an unload order issue between + res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip. + + res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip. + Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it + unloads, it's objects are still in the sorcery instance. When res_pjsip + unloads, it destroys all its objects including res_pjsip_phoneprov_provider's. + The phoneprov destructor then attempts to unregister the extension from + res_phoneprov but because res_phoneprov is already cleaned up, its users + container is gone and we get a FRACK. + + Simple solution, check for the NULL users container before attempting to remove + the entry. Duh. + + Ran tests/res_phoneprov/res_phoneprov_provider. No leaks in + res_pjsip_phoneprov_provider and no FRACKs. + + Reported-by: Corey Farrell + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4608/ + ASTERISK-24935 #close + ........ + + Merged revisions 434545 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434547 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 18:08 +0000 [9a63ada03a] gtjoseph + + * loader/main: Don't set ast_fully_booted until deferred reloads are processed + + Until we have a true module management facility it's sometimes necessary for one + module to force a reload on another before its own load is complete. If + Asterisk isn't fully booted yet, these reloads are deferred. The problem is + that asterisk reports fully booted before processing the deferred reloads which + means Asterisk really isn't quite ready when it says it is. + + This patch moves the report of fully booted after the processing of the deferred + reloads is complete. + + Since the pjsip stack has the most number of related modules, I ran the + channels/pjsip testsuite to make sure there aren't any issues. All tests + passed. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4604/ + ........ + + Merged revisions 434544 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 17:07 +0000 [520b9f2174] Kevin Harwell + + * res_pjsip: add CLI command to show global and system configuration + + Added a new CLI command for res_pjsip that shows both global and system + configuration settings: pjsip show settings + + ASTERISK-24918 #close + Reported by: Scott Griepentrog + Review: https://reviewboard.asterisk.org/r/4597/ + ........ + + Merged revisions 434527 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 11:09 +0000 [b2b1f24af6] Richard Mudgett + + * chan_iax2.c: Fix ref leak in iax2_request(). + + * Increased warning message format capability string buffer size in + iax2_request(). + + Review: https://reviewboard.asterisk.org/r/4601/ + ........ + + Merged revisions 434510 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434511 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 11:05 +0000 [459171be12] Richard Mudgett + + * bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible(). + + Review: https://reviewboard.asterisk.org/r/4601/ + ........ + + Merged revisions 434508 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 10:43 +0000 [3ef0a17b1f] yaron nahum (License 6676) + + * res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests + + This patch adds a new session supplement that handles in-dialog OPTIONS + requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup + for the OPTIONS request would already have been done by the time the + session supplement receives the inbound request. + + ASTERISK-24862 #close + Reported by: yaron nahum + patches: + res_pjsip_dlg_options.c submitted by yaron nahum (License 6676) + ........ + + Merged revisions 434506 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 09:58 +0000 [c08ebc6eeb] Mark Michelson + + * Reduce duplication of common DNS code. + + The NAPTR and SRV branches were worked on independently and + resulted in some code being duplicated in each. Since both + have been merged into trunk now, this patch reduces the + duplication by factoring out common code into its own + source files. + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-09 07:57 +0000 [ea0098724e] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix autological comparisons + + This fixes autological comparison warnings in the following: + * chan_skinny: letohl may return a signed or unsigned value, depending on the + macro chosen + * func_curl: Provide a specific cast to CURLoption to prevent mismatch + * cel: Fix enum comparisons where the enum can never be negative + * enum: Fix comparison of return result of dn_expand, which returns a signed + int value + * event: Fix enum comparisons where the enum can never be negative + * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be + negative + * presencestate: Use the actual enum value for INVALID state + * security_events: Fix enum comparisons where the enum can never be negative + * udptl: Don't bother to check if the return value from encode_length is less + than 0, as it returns an unsigned int + * translate: Since the parameters are unsigned int, don't bother checking + to see if they are negative. The cast to unsigned int would already blow + past the matrix bounds. + * res_pjsip_exten_state: Use a temporary value to cache the return of + ast_hint_presence_state + * res_stasis_playback: Fix enum comparisons where the enum can never be + negative + * res_stasis_recording: Add an enum value for the case where the recording + operation is in error; fix enum comparisons + * resource_bridges: Use enum value as opposed to -1 + * resource_channels: Use enum value as opposed to -1 + + Review: https://reviewboard.asterisk.org/r/4533 + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4533.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 21:05 +0000 [2201e27340] Stefan Engström (License 6691) + + * apps/app_queue: Prevent possible crash when evaluating queue penalty rules + + Although it only occurred once, a crash occurred when a queue attempted to + evaluate a queue penalty rule that appeared to have already been destroyed. + In many locations in app_queue, a test is done to see if qe->pr is NULL; + however, when we dispose of a queue's penalty rules, we don't set the pointer + to NULL after free'ing it. This patch does that to prevent any dangling + pointers from lingering on the queue object. + + Review: https://reviewboard.asterisk.org/r/4522 + + ASTERISK-23319 #close + Reported by: Vadim + patches: + rb4552.patch submitted by Stefan Engström (License 6691) + ........ + + Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434449 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 13:32 +0000 [a759714101] Jonathan Rose + + * res_pjsip_t38: Fix FAX failures when using PJSIP with authentication + + Without this patch, if a PJSIP endpoint with udptl enabled and authentication + set attempted to use sendFax, the FAX session would fail during setup. This + was because the invite issued in response to being auth challenged would cause + the PJSIP channel performing the FAX to receive a second T38 framehook and + this would cause frames to be consumed in an inappropriate manner. + + ASTERISK-24933 #close + Reported by: Jonathan Rose + Review: https://reviewboard.asterisk.org/r/4577/ + ........ + + Merged revisions 434425 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 13:20 +0000 [09df34d880] Richard Mudgett + + * Bridging: Eliminate the unnecessary make channel compatible with bridge operation. + + When a channel enters the bridging system it is first made compatible with + the bridge and then the bridge technology makes the channel compatible + with the technology. For all but the DAHDI native and softmix bridge + technologies the make channel compatible with the bridge step is an + effective noop because the other technologies allow all audio formats. + For the DAHDI native bridge technology it doesn't matter because it is not + an initial bridge technology and chan_dahdi allows only one native format + per channel. For the softmix bridge technology, it is a noop at best and + harmful at worst because the wrong translation path could be setup if the + channel's native formats allow more than one audio format. + + This is an intermediate patch for a series of patches aimed at improving + translation path choices. + + * Removed code dealing with the unnecessary step of making the channel + compatible with the bridge. + + ASTERISK-24841 + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4600/ + ........ + + Merged revisions 434424 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 11:49 +0000 [8ec9a82b9a] Maciej Szmigiero (license 6085) + + * Security/tcptls: MitM Attack potential from certificate with NULL byte in CN. + + When registering to a SIP server with TLS, Asterisk will accept CA signed + certificates with a common name that was signed for a domain other than the + one requested if it contains a null character in the common name portion of + the cert. This patch fixes that by checking that the common name length + matches the the length of the content we actually read from the common name + segment. Some certificate authorities automatically sign CA requests when + the requesting CN isn't already taken, so an attacker could potentially + register a CN with something like www.google.com\x00www.secretlyevil.net + and have their certificate signed and Asterisk would accept that certificate + as though it had been for www.google.com - this is a security fix and is + noted in AST-2015-003. + + ASTERISK-24847 #close + Reported by: Maciej Szmigiero + Patches: + asterisk-null-in-cn.patch submitted by mhej (license 6085) + ........ + + Merged revisions 434337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 434338 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434384 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434385 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 11:31 +0000 [2bd9e008a7] Richard Mudgett + + * format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear(). + ........ + + Merged revisions 434357 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 07:02 +0000 [3f54af689f] Matt Jordan + + * chan_iax2: Fix compilation issue due to funky merge + + Don't mix declarations and code! + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 07:00 +0000 [a9b6a62461] Jaco Kroon (License 5671) + + * chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno] + + This patch fixes an access to the peer callnumber that is unprotected by a + corresponding mutex. The peer->callno value can be changed by multiple threads, + and all data inside the iaxs array must be procted by a corresponding lock + of iaxsl. + + The patch moves the unprotected access to a location where the mutex is + safely obtained. + + Review: https://reviewboard.asterisk.org/r/4599/ + + ASTERISK-21211 #close + Reported by: Jaco Kroon + patches: + asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671) + ........ + + Merged revisions 434291 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434292 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 06:54 +0000 [477536ef25] Valentin Vidić (License 6697) + + * chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled + + When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will + attempt to handle both IPv4 and IPv6 addresses, although the information will + be stored in a struct with an AF_INET6 address type. However, the current + NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly. + This patch adds an additional check for the mapped address case, allowing + the NAT code to handle clients even when the address is IPv6. + + Review: https://reviewboard.asterisk.org/r/4563/ + + ASTERISK-18032 #close + Reported by: Christoph Timm + patches: + nat_with_ipv6.diff submitted by Valentin Vidić (License 6697) + ........ + + Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434289 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 06:45 +0000 [b8fa8aa775] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix pointer-bool-converesion warnings + + This patch fixes several warnings pointed out by the clang compiler. + * chan_pjsip: Removed check for data->text, as it will always be non-NULL. + * app_minivm: Fixed evaluation of etemplate->locale, which will always + evaluate to 'true'. This patch changes the evaluation to use + ast_strlen_zero. + * app_queue: + - Fixed evaluation of qe->parent->monfmt, which always evaluates to + true. Instead, we just check to see if the dereferenced pointer + evaluates to true. + - Fixed evaluation of mem->state_interface, wrapping it with a call to + ast_strlen_zero. + * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero. + + Review: https://reviewboard.asterisk.org/r/4541 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4541.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-08 06:35 +0000 [016fba12e2] Rodrigo Ramirez Norambuena (License 6577) + + * cel_pgsl: Add support for GMT timestamps + + This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps + to be logged in GMT. + + Review: https://reviewboard.asterisk.org/r/4571/ + + ASTERISK-23186 #close + Reported by: Rodrigo Ramirez Norambuena + patches: + cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577) + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 14:40 +0000 [d923ec80b9] Scott Griepentrog + + * pjsip: resolve compatibility problem with ast_sip_session + + A change in r430179 inserted a variable near the top of a + structure caused a problem when running DPMA in a version + of Asterisk compiled across the change. This patch moves + the new variable to the end of the structure, eliminating + the problem. + + Review: https://reviewboard.asterisk.org/r/4574/ + ........ + + Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13 + ........ + + Merged revisions 434261 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 11:42 +0000 [153c4044e4] Kevin Harwell + + * bridge.c: Hangup attended transfer target after it has been swapped out + + After completing an attended transfer the transfer target channel (the one that + gets swapped out) was not being hung up after leaving the bridge. This resulted + in a channel possibly being left around. Added an explicit softhangup for the + channel in question after the transfer is successfully completed in order to + make sure the channel is hung up. + + ASTERISK-24782 #close + Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/4575/ + ........ + + Merged revisions 434240 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 10:34 +0000 [1eba6abae5] Mark Michelson + + * Do not queue message requests that we do not respond to. + + If we receive a MESSAGE request that we cannot send a response + to, we should not send the incoming MESSAGE to the dialplan. + + This commit should help the bouncing message_retrans test to + pass consistently. + ........ + + Merged revisions 434218 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-07 10:22 +0000 [c2f50ba6f4] Matt Jordan + + * ARI: Add the ability to intercept hold and raise an event + + For some applications - such as SLA - a phone pressing hold should not behave + in the fashion that the Asterisk core would like it to. Instead, the hold + action has some application specific behaviour associated with it - such as + disconnecting the channel that initiated the hold; only playing MoH to channels + in the bridge if the channels are of a particular type, etc. + + One way of accomplishing this is to use a framehook to intercept the + hold/unhold frames, raise an event, and eat the frame. Tasty. This patch + accomplishes that using a new dialplan function, HOLD_INTERCEPT. + + In addition, some general cleanup of raising hold/unhold Stasis messages was + done, including removing some RAII_VAR usage. + + Review: https://reviewboard.asterisk.org/r/4549/ + + ASTERISK-24922 #close + ........ + + Merged revisions 434216 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 21:10 +0000 [af4d802773] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix sometimes-initialized warning in func_math + + This patch fixes a bug in a unit test in func_math where a variable could be + passed to ast_free that wasn't allocated. This patch corrects the issue and + ensures that we only attempt to free a variable if we previously allocated + it. + + Review: https://reviewboard.asterisk.org/r/4552 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4552.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434190 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434191 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 21:03 +0000 [c1cfe3fae2] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix non-literal-null-conversion warnings + + Clang will flag errors when a char pointer is set to '\0', as opposed to a + value that the char pointer points to. This patch fixes this warning + in a variety of locations. + + Review: https://reviewboard.asterisk.org/r/4551 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4551.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434187 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434188 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434189 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 16:54 +0000 [79fb8c32a6] Mark Michelson + + * Uncomment test case. + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434170 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 16:13 +0000 [fc314cb43f] Mark Michelson + + * Add missing DNS NAPTR test file. + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434154 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 14:23 +0000 [87d7c90e4e] Kevin Harwell + + * res_pjsip: config option 'timers' can't be set to 'no' + + When setting the configuration option 'timers' equal to 'no' the bit flag was + not properly negated. This patch clears all associated flags and only sets the + specified one. pjsip will handle any necessary flag combinations. Also went + ahead and did similar for the '100rel' option. + + ASTERISK-24910 #close + Reported by: Ray Crumrine + Review: https://reviewboard.asterisk.org/r/4582/ + ........ + + Merged revisions 434131 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 14:04 +0000 [e48f2e7897] gtjoseph + + * build: Fixes for gcc 5 compilation + + These are fixes for compilation under gcc 5.0... + + chan_sip.c: In parse_request needed to make 'lim' unsigned. + inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 + inline semantics (same as clang). + ccss.c: In ast_cc_set_parm, needed to fix weird comparison. + dsp.c: Needed to work around a possible compiler bug. It was throwing + an array-bounds error but neither + sgriepentrog, rmudgett nor I could figure out why. + manager.c: In action_atxfer, needed to correct an array allocation. + + This patch will go to 11, 13, trunk. + + Review: https://reviewboard.asterisk.org/r/4581/ + Reported-by: Jeffrey Ollie + Tested-by: George Joseph + ASTERISK-24932 #close + ........ + + Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434114 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 13:18 +0000 [0543879228] Diederik de Groot (License 6600) + + * clang compiler warnings: Remove large chunks of unused code from extconf + + This patch fixes a warning caught by clang, in which it detected that large + chunks of extconf were unused. Frankly, I wish we could pretend that all of + extconf was unused, but alas, that is not yet the case. + + A few extraneous functions in the parking tests were removed as well, for + the same reason. + + Review: https://reviewboard.asterisk.org/r/4553 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4553.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434093 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434097 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 13:03 +0000 [e309a91e2d] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config + + This patch fixes a warning caught by clang, in which a char pointer could be + assigned to before it was initialized. The patch re-organizes the code to + ensure that the pointer is always initialized, even on off nominal paths. + + Review: https://reviewboard.asterisk.org/r/4529 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4529.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434090 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434091 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 12:52 +0000 [ed3cf8761b] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix format specified in framehook + + This patch fixes an invalid format specifier used in the formatting of an + ERROR message in the framehook code. The format specifier specifies a + type of 'unsigned short', but the argument passed to it is of type 'int'. + The patch changes the format specifier to 'i'. + + Review: https://reviewboard.asterisk.org/r/4540 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4535.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 434087 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 434088 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 12:05 +0000 [0a26602b8c] Mark Michelson + + * Merge NAPTR support into trunk. + + This adds NAPTR record allocation and sorting, as well as + unit tests that verify that NAPTR records are parsed and + sorted correctly. + + Review: https://reviewboard.asterisk.org/r/4542 + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434068 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 11:02 +0000 [edf9da4365] Mark Michelson + + * Ensure that a non-zero sample rate is returned for all formats. + + Versions of Asterisk prior to 12 defaulted to 8000 as a sample rate + if one was not provided by a format. In Asterisk 13, this was removed. + The result was that some calculations which involve dividing by the + sample rate resulted in dividing by 0. The fix being put in place + here is to have the same default fallback that was present in previous + versions of Asterisk. + + Asterisk-24914 #close + Reported by Marcello Ceschia + ........ + + Merged revisions 434046 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434047 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 10:17 +0000 [ffd7319df3] Corey Farrell + + * res_pjsip_phoneprov_provider: Revert 433996 / 433997. + + res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then + ignoring the return. OBJ_NODATA flag was to prevent a reference leak, but + this caused the module to FRACK on unload. Revert change until this can + be investigated further. + + ASTERISK-24935 + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4578/ + ........ + + Merged revisions 434025 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-06 09:51 +0000 [53af579d4c] Mark Michelson (license #5049) + + * ParkedCall: Don't allow dialplan fallthrough after retrieving parked call. + + This is a change to align behavior with that of Asterisk 11 and previous versions. + In those versions, if a parked call were retrieved, and the call ended, the parked + call retriever would be hung up after the ParkedCall application ran. Prior to this + patch, in Asterisk 13, the same situation would result in the parked call retriever + falling through to additional priorities in the extension where the ParkedCall + application was called. With this patch, the behavior between Asterisk 11 and 13 + aligns. + + ASTERISK-24899 #close + Reported by Malcolm Davenport + Patches: + ASTERISK-24899.patch uploaded by Mark Michelson(license #5049) + ........ + + Merged revisions 434022 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-05 07:55 +0000 [e6f0410028] Corey Farrell + + * res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator. + + res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then + ignoring the return. Added OBJ_NODATA flag to prevent a reference leak. + + ASTERISK-24935 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4578/ + ........ + + Merged revisions 433996 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-03 16:54 +0000 [3439487a81] Mark Michelson + + * res_pjsip_messaging: Serialize outbound SIP MESSAGEs + + Outbound SIP MESSAGEs had the potential to be sent out + of order from how they were specified in a set of + dialplan steps. + + This change creates a serializer for sending outbound + MESSAGE requests on. This ensures that the MESSAGEs are + sent by Asterisk in the same order that they were sent + from the dialplan. + + ASTERISK-24937 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4579 + ........ + + Merged revisions 433968 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433969 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-02 09:56 +0000 [6e5efe04bd] Scott Griepentrog + + * pjsip: resolve compatibility problem with ast_sip_session + + A change in r430179 inserted a variable near the top of a + structure caused a problem when running DPMA in a version + of Asterisk compiled across the change. This patch moves + the new variable to the end of the structure, eliminating + the problem. + + Review: https://reviewboard.asterisk.org/r/4574/ + ........ + + Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-02 05:38 +0000 [154ba47766] Corey Farrell + + * Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS. + + DEBUG_CHAOS was marked as conflicting with MALLOC_DEBUG, but + for this to work correctly MALLOC_DEBUG must also be marked + as conflicting with DEBUG_CHAOS. + + Review: https://reviewboard.asterisk.org/r/4557/ + ........ + + Merged revisions 433923 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-01 11:30 +0000 [a217d2d1db] Ashley Sanders + + * stasis: set a channel variable on websocket disconnect error + + Resolve compile errors caused by r433863 by fixing the + documentation xml to comply with the schema. + ........ + + Merged revisions 433888 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-01 11:27 +0000 [39824e3d01] Joshua Colp + + * dns: Add support for SRV record parsing and sorting. + + This change adds support for parsing SRV records and consuming their values + in an easy fashion. It also adds automatic sorting of SRV records according + to RFC 2782. + + Tests have also been included which cover parsing, sorting, and off-nominal + cases where the record is corrupted. + + ASTERISK-24931 #close + Reported by: Joshua Colp + + Review: https://reviewboard.asterisk.org/r/4528/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-04-01 08:35 +0000 [da13d15425] Mark Michelson + + * stasis: set a channel variable on websocket disconnect error + + Resolve compile errors caused by r433839 by included the missing + header file, pbx.h. + ........ + + Merged revisions 433863 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 17:49 +0000 [06578ef407] Ashley Sanders + + * stasis: set a channel variable on websocket disconnect error + + When an error occurs while writing to a web socket, the web socket is + disconnected and the event is logged. A side-effect of this, however, is that + any application on the other side waiting for a response from Stasis is left + hanging indefinitely (as there is no mechanism presently available for + notifying interested parties about web socket error states in Stasis). + + To remedy this scenario, this patch introduces a new channel variable: + STASISSTATUS. + + The possible values for STASISSTATUS are: + SUCCESS - The channel has exited Stasis without any failures + FAILED - Something caused Stasis to croak. Some (not all) possible + reasons for this: + - The app registry is not instantiated; + - The app requested is not registered; + - The app requested is not active; + - Stasis couldn't send a start message + + ASTERISK-24802 + Reported By: Kevin Harwell + Review: https://reviewboard.asterisk.org/r/4519/ + ........ + + Merged revisions 433839 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433845 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 12:04 +0000 [2d28fa678e] Richard Mudgett + + * chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos. + + Fix misplaced parentheses in original fabs() expression. + ........ + + Merged revisions 433816 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433817 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-31 06:55 +0000 [076fc12afb] Corey Farrell + + * Blocked revisions 433795 + + ........ + Re-add _ast_mem_backtrace_buffer variable for ABI compatibility. + + Modules built prior to commit of r4502 expect to link at runtime + to the variable _ast_mem_backtrace_buffer. This change re-adds + the variable to the C file only. + + Review: https://reviewboard.asterisk.org/r/4558/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433796 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-30 06:43 +0000 [8d12288d8a] Corey Farrell + + * Fix an ABI compatibility issue with ast_log_safe for modules. + + Binary modules are sometimes built against the latest release of + Asterisk in each branch, and need to be compatible with all + releases of that branch. This change ensures that utils.h only + uses ast_log_safe from the core. For modules and utilities ast_log + is used instead. + + Review: https://reviewboard.asterisk.org/r/4548/ + ........ + + Merged revisions 433772 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433773 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 21:45 +0000 [7bc2345fb1] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix -Wabsolute-value warnings + + This patch fixes several warnings caught by clang - in this case, usage of the + abs function on non-integer values. This patch uses labs and fabs, as + appropriate, in the various affected files. + + Review: https://reviewboard.asterisk.org/r/4525 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4525.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433750 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 21:39 +0000 [ce59fabd5c] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix invalid enum conversion + + This patch fixes some invalid enum conversion warnings caught by clang. In + particular: + * chan_sip: Several functions mixed usage of the st_refresher_param + enum and st_refresher enum. This patch corrects the functions to use the + right enum. + * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state. + * strings: Fixed incorrect usage of AO2 flags with strings container. + * res_stasis: Change a return enumeration to stasis_app_user_event_res. + + Review: https://reviewboard.asterisk.org/r/4535 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4535.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433747 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433748 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 21:29 +0000 [61577cbee6] Matt Jordan + + * main/stdtime/localtime: Fix warning introduced in r433720 + + The patch in r433720 caused a warning to be kicked back by gcc. It occurred + due to this check in unistd.h: + + if (__nbytes > __bos0 (__buf)) + return __read_chk_warn (__fd, __buf, __nbytes, __bos0 (__buf)); + + That is, if __nbytes is greater than the result of GCC's built-in object size + for the struct, we'll kick back a warning. + + As it turns out, this is because there is an error in the code in the patch. + We are passing the address of the pointer to the struct, not iev, which is a + pointer to the struct. Hence, the number of bytes is probably going to be lot + larger than the number of bytes that make up a pointer! This patch changes + the code just read from the pointer to the struct - which fixes the warning. + + ASTERISK-24917 + ........ + + Merged revisions 433743 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433744 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 20:57 +0000 [072734692e] Diederik de Groot (License 6600) + + * clang compiler warnings: Ignore -Wunused-command-line-argument + + Asterisk's build system has a tendency to pass include directives for libraries + to everything compiled within a particular group of source files. This means + we pass the header for libxml2 to things that don't necessarily need it. As a + result, we ignore this particular warning. + + Review: https://reviewboard.asterisk.org/r/4545/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4545.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433720 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433721 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-29 20:53 +0000 [1cf949c489] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix warning for -Wgnu-variable-sized-type-not-at-end + + This patch fixes a warning caught by clang, wherein a variable sized struct is + not located at the end of a struct. While the code in question actually + expected this, this is a good warning to watch for. Hence, this patch refactors + the code in question to not have two variable length elements in the same + struct. + + Review: https://reviewboard.asterisk.org/r/4530/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4530.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433717 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433718 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:56 +0000 [d2776d4d45] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix a variety of "unused" warnings + + This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable + errors caught by clang. Specifically: + + * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], + qsmp_cmd_usage[] + * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" + * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" + * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" + * funcs/func_env.c:729: Fixed ast_str_append_substr. + * main/editline/np/strlcat.c: removed unused rcsid variable + * main/editline/np/strlcpy.c: removed unused rcsid variable + * main/security_events.c: removed unused TIMESTAMP_STR_LEN + * utils/conf2ael.c: removed unused cfextension_states + * utils/extconf.c: removed unused cfextension_states + + Review: https://reviewboard.asterisk.org/r/4526 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4526.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433694 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:48 +0000 [cb7b6bc4be] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix -Wself-assign + + Assigning a variable to itself isn't super useful. However, the WAV format + modules make use of this in order to perform byte endian checks. This patch + works around the warning by only performing the self assignment if we are + going to do more than just assign it to ourselves. Which is odd, but true. + + Review: https://reviewboard.asterisk.org/r/4544/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4544.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433690 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433691 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:41 +0000 [e9520dbe0d] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix -Wparantheses-equality warnings + + Clang will treat ((a == b)) as a warning, as it reasonably expects that the + developer may have intended to write (a == b) or ((a = b)). This patch cleans + up all instances where equality, not assignment, was intended between two + parantheses. + + Review: https://reviewboard.asterisk.org/r/4531/ + + ASTERISK-24917 + Repoted by: dkdegroot + patches: + rb4531.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433688 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:33 +0000 [fd50e5bfb5] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix -Wbitfield-constant-conversion warning + + In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by + clang, as it will truncate the -1 to a 1 implicitly. + + Instead, we just assign the value a '1'. + + Review: https://reviewboard.asterisk.org/r/4537/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4537.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433683 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433684 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433686 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:32 +0000 [c747b3b12a] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix -Winitializer-overrides + + This patch fixes clange compiler warnings for initializer overrides. + Specifically: + + res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration + value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing + those enum values, we therefore initialize the value twice to two different + values, "tlsv1" and "default". This patch changes it to just initialize + the index in the array to "tlsv1". + + Review: https://reviewboard.asterisk.org/r/4539/ + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4539.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433682 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-28 07:20 +0000 [d6173cd1d0] Diederik de Groot (License 6600) + + * clang compiler warnings: Fix -Wunused-function; make inline function static + + This patch fixes clang compilers warnings for unused functions. Specifically: + * channels/chan_iax2: removed user_ref function + * main/dsp.c: removed goertzel_update function + * main/config.c: made variable_list_switch static + + Review: https://reviewboard.asterisk.org/r/4527 + + ASTERISK-24917 + Reported by: dkdegroot + patches: + rb4527.patch submitted by dkdegroot (License 6600) + ........ + + Merged revisions 433678 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433680 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433681 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 17:26 +0000 [b56592e3ae] Jonathan Rose + + * SAC: Add conferencing extensions and configuration + + Review: https://reviewboard.asterisk.org/r/4504/ + ........ + + Merged revisions 433656 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 16:21 +0000 [c21e2e45a8] Rusty Newton + + * configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2 + + Example configuration files for a "basic PBX" deployment for the fictitious + Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/ + and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company + + Patch 4488 includes all functionality needed for SAC's outside connectivity + and some externally accessed features, as well as outbound dialing. + + Reported by: Malcolm Davenport + Tested by: Rusty Newton + + Review: https://reviewboard.asterisk.org/r/4488/ + ........ + + Merged revisions 433624 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 16:06 +0000 [2659e48d9d] Richard Mudgett + + * res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage. + + * Converted the contact_autoexpire container to use the ao2 template hash + and cmp functions. Also made use the OBJ_SEARCH_xxx names instead of the + deprecated names. + + * Eliminates several unnecessary uses of RAII_VAR(). + + Review: https://reviewboard.asterisk.org/r/4524/ + ........ + + Merged revisions 433622 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 15:46 +0000 [0b62e41654] Mark Michelson + + * Add stateful PJSIP response API call, and use it for out-of-dialog responses. + + Asterisk had an issue where retransmissions of MESSAGE requests resulted in + Asterisk processing the retransmission as if it were a new MESSAGE request. + + This patch fixes the issue by creating a transaction in PJSIP on the incoming + request. This way, if a retransmission arrives, the PJSIP transaction layer + will resend the response and Asterisk will not ever see the retransmission. + + ASTERISK-24920 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4532/ + ........ + + Merged revisions 433619 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 15:23 +0000 [a18da4eaf2] Richard Mudgett + + * res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown. + + Contact expiration object refs were leaked when the module was unloaded. + + * Made empty the scheduler of entries before destroying it to release the + object ref held by the scheduler entry. + + Review: https://reviewboard.asterisk.org/r/4523/ + ........ + + Merged revisions 433596 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 12:58 +0000 [cb1c639817] Richard Mudgett + + * Add missing file. ASTERISK-24781 + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 09:41 +0000 [a024af1156] Justin T. Gibbs (License 6692) + + * res/res_timing_kqueue: Update the module to conform to current timer API + + This patch updates the kqueue timing module to conform to current timer API. + + This fixes issues with using the kqueue timing source on Asterisk 13 on + FreeBSD 10. These issues include: + + - Remove support for kevent64(). The values used to support Asterisk timers + fit within 32bits and so can be handled on all platforms via kevent(). + + - Provide debug logging for, but do not track, unacked events. This matches + the behavior of all other timer implementations. + + - Implement continuous mode by triggering and leaving active, a user event. + This ensures that the file descriptor for the timer returns immediately from + poll(), without placing the load of a high speed timer on the kernel. + + - In kqueue_timer_get_max_rate(), don't overstate the capability of the timer. + On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer + type kqueue supports for timers. + + - In kqueue_timer_get_event(), assume the caller woke up from poll() and just + return the mode the timer is currently in. This matches all other timer + implementations. + + - Adjust the test code now that unacked events are not tracked. + + Review: https://reviewboard.asterisk.org/r/4465/ + + ASTERISK-24857 #close + Reported by: scsiguy + Tested by: Ed Hynan + patches: + rb4465.patch submitted by scsiguy (License 6692) + ........ + + Merged revisions 433574 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 07:27 +0000 [10458d2878] Corey Farrell + + * Fix link error for utils/aelparse. + + Use the standard ast_log instead of ast_log_safe for STANDALONE programs. + + Review: https://reviewboard.asterisk.org/r/4538/ + ........ + + Merged revisions 433549 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433550 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-27 02:12 +0000 [28e3bd0af7] Corey Farrell + + * Improved and portable ast_log recursion avoidance + + This introduces a new logger routine ast_log_safe. This routine should be + used for all error messages in code that can be run as a result of ast_log. + ast_log_safe does nothing if run recursively. All error logging in + astobj2.c, strings.c and utils.h have been switched to ast_log_safe. + + This required adding support for raw threadstorage. This provides direct + access to the void* pointer in threadstorage. In ast_log_safe, NULL is used + to signify that this thread is not already running ast_log_safe, (void*)1 when + it is already running. This was done since it's critical that ast_log_safe + do nothing that could log during recursion checking. + + ASTERISK-24155 #close + Reported by: Timo Teräs + Review: https://reviewboard.asterisk.org/r/4502/ + ........ + + Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433523 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 18:09 +0000 [554eb74516] Corey Farrell + + * Fix compile errors caused by r4500 / r4501. + + * Add ast_register_cleanup to utils/clicompat.c to deal with + any utils that copy sources from main. + * Asterisk 13+: remove unused variables from core_local.c. + + Review: https://reviewboard.asterisk.org/r/4534/ + ........ + + Merged revisions 433499 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433500 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 17:24 +0000 [3ddd92902a] Corey Farrell + + * Replace most uses of ast_register_atexit with ast_register_cleanup. + + Since 'core stop now' and 'core restart now' do not stop modules, + it is unsafe for most of the core to run cleanups. Originally all + cleanups used ast_register_atexit, and were only changed when it + was shown to be unsafe. ast_register_atexit is now used only when + absolutely required to prevent corruption and close child processes. + + Exceptions that need to use ast_register_atexit: + * CDR: Flush records. + * res_musiconhold: Kill external applications. + * AstDB: Close the DB. + * canary_exit: Kill canary process. + + ASTERISK-24142 #close + Reported by: David Brillert + + ASTERISK-24683 #close + Reported by: Peter Katzmann + + ASTERISK-24805 #close + Reported by: Badalian Vyacheslav + + ASTERISK-24881 #close + Reported by: Corey Farrell + + Review: https://reviewboard.asterisk.org/r/4500/ + Review: https://reviewboard.asterisk.org/r/4501/ + ........ + + Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 12:47 +0000 [d7fc85e69d] Corey Farrell + + * res_pjsip: Enable unload of all modules at shutdown. + + * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes + caused by running PJSIP functions from non-PJSIP threads. + * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing + crashes in some cases. In theory pj_shutdown() should take care of this. + * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at + shutdown. + * Resolve leaked config global in res_pjsip_notify. + * Unregister pubsub pjsip service module. + * Implement cleanup for res_pjsip_session. + + ASTERISK-24731 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4498/ + ........ + + Merged revisions 433469 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-26 12:13 +0000 [ab674f67b5] Kevin Harwell + + * app_confbridge: file playback blocks dtmf + + Attempting to execute DTMF in a confbridge while file playback (prompt, + announcement, etc) is occurring is not allowed. You have to wait until + the sound file has completed before entering DTMF. This patch fixes it + so that app_confbridge now monitors for dtmf key presses during menu + driven file playback. If a key is pressed playback stops and it executes + the matched menu option. + + ASTERISK-24864 #close + Reported by: Steve Pitts + Review: https://reviewboard.asterisk.org/r/4510/ + ........ + + Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433446 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-25 13:37 +0000 [e953d15223] Richard Mudgett + + * A couple minor cleanup tweaks. + + * In res/res_sorcery_realtime.c: Broke long line. + + * In main/bucket.c: Eliminated unnecessary NULL check as + ast_sorcery_unref() is NULL tolerant and set the global object to NULL + after unref in the system shutdown bucket_cleanup(). + ........ + + Merged revisions 433420 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-25 10:31 +0000 [47156aab92] Simon Arlott (License 5756) + + * res_xmpp: Buddies are always auto-registered when processing the roster + + Due to a quirk in the configuration handling of res_xmpp, the 'autoregister' + setting was never actually processed. This was due to not properly copying + over the global settings to the client settings when applying the + configuration to the run-time object. + + Review: https://reviewboard.asterisk.org/r/4496/ + + ASTERISK-14233 + ASTERISK-24780 #close + Reported by: Simon Arlott + patches: + asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756) + ........ + + Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433396 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433397 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-25 07:32 +0000 [abf3e40902] Joshua Colp + + * dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. + + This change adds an abstracted core DNS API which resembles the API described + here[1]. The API provides a pluggable mechanism for resolvers and also a + consistent view for records. Both synchronous and asynchronous queries are + supported. + + This change also adds a res_resolver_unbound module which uses the libunbound + library to provide resolution. + + Unit tests have also been written for all of the above to confirm the API and + functionality. + + ASTERISK-24834 #close + Reported by: Matt Jordan + + ASTERISK-24836 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4474/ + Review: https://reviewboard.asterisk.org/r/4512/ + + [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-24 14:41 +0000 [4c2fc5b811] Richard Mudgett + + * chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. + + Incoming PJSIP call legs that have not been answered yet send unnecessary + "180 Ringing" or "183 Progress" messages every time a connected line + update happens. If the outgoing channel is also PJSIP then the incoming + channel will always send a "180 Ringing" or "183 Progress" message when + the outgoing channel sends the INVITE. + + Consequences of these unnecessary messages: + + * The caller can start hearing ringback before the far end even gets the + call. + + * Many phones tend to grab the first connected line information and refuse + to update the display if it changes. The first information is not likely + to be correct if the call goes to an endpoint not under the control of the + first Asterisk box. + + When connected line first went into Asterisk in v1.8, chan_sip received an + undocumented option "rpid_immediate" that defaults to disabled. When + enabled, the option immediately passes connected line update information + to the caller in "180 Ringing" or "183 Progress" messages as described + above. + + * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or + "183 Progress" messages. The default is "no" to disable sending the + unnecessary messages. + + ASTERISK-24781 #close + Reported by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4473/ + ........ + + Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-22 19:05 +0000 [60f01520e7] snuffy (License 5024) + + * Fix compilations errors on 64-bit OpenBSD systems + + In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to + (long) when printing members of certain time structs. + + Review: https://reviewboard.asterisk.org/r/4507 + + ASTERISK-24879 #close + Reported by: snuffy + Tested by: snuffy + patches: + openbsd-time64.diff uploaded by snuffy (License 5024) + ........ + + Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433269 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-22 18:11 +0000 [66670f02e6] snuffy (License 5024) + + * Fix compilation issues for OpenBSD + + This patch addresses compilation issues for OpenBSD. Specifically, it + addresses: + * It allows including in asterisk.c + * Provides a needed (size_t) cast in xmldoc.c + + In 13+, it also addresses a conditional inclusion in loader.c. + + Review: https://reviewboard.asterisk.org/r/4506 + + ASTERISK-24880 #close + Reported by: snuffy + Tested by: snuffy + patches: + misc-openbsd.diff uploaded by snuffy (License 5024) + ........ + + Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433247 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-20 14:54 +0000 [7e097bce86] Richard Mudgett + + * Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks. + + Valgrind found some memory leaks associated with + ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending + responses to OPTIONS requests, processing MESSAGE requests, and + res_pjsip supplements implementing the incoming_request callback. + + * Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in + res/res_pjsip.c:supplement_on_rx_request(), + res/res_pjsip/pjsip_options.c:send_options_response(), + res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and + res/res_pjsip_messaging.c:send_response(). + + * Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in + res/res_pjsip_nat.c:nat_on_rx_message(). + + * Fixed inconsistent but benign return value in + res/res_pjsip/pjsip_options.c:options_on_rx_request(). + + Review: https://reviewboard.asterisk.org/r/4511/ + ........ + + Merged revisions 433222 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-20 13:27 +0000 [148e8799fe] Richard Mudgett + + * res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively. + + Valgrind found a memory leak and invalid access. + + * Fix invalid access by sscanf() being fed a non-nul terminated string of + digits in res/res_pjsip_sdp_rtp.c:get_codecs(). + + * Fix memory leak in main/sorcery.c:sorcery_object_field_destructor(). + + * Fix potential NULL pointer dereference in + main/xmldoc.c:xmldoc_get_syntax_config_option(). + + Review: https://reviewboard.asterisk.org/r/4513/ + ........ + + Merged revisions 433199 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-19 14:20 +0000 [627cc16a8d] Matt Jordan + + * funcs/func_env: Fix regression caused in FILE read operation + + When r432935 was merged, it did correctly fix a situation where a FILE read + operation on the middle of a file buffer would not read the requested length + in the parameters passed to the FILE function. Unfortunately, it would also + allow the FILE function to append more bytes than what was available in the + buffer if the length exceeded the end of the buffer length. + + This patch takes the minimum of the remaining bytes in the buffer along with + the calculated length to append provided by the original patch, and uses + that as the length to append in the return result. This patch also updates + the unit tests with the scenarios that were originally pointed out in + ASTERISK-21765 that the original implementation treated incorrectly. + + ASTERISK-21765 + ........ + + Merged revisions 433173 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433174 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-19 10:27 +0000 [79a81fed59] Kevin Harwell + + * alemebic scripts: endpoint identifier order option + + The script was added in 13, but when committed to trunk it caused a branch to + occur due to some trunk only alemebic changes. This fixes it so that the new + 'add_pjsip_endpoint_identifier_order script points to the correct down revision. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-19 05:21 +0000 [3aa0a869c2] Corey Farrell + + * logger: Apply default console logging when configuration cannot be loaded. + + When logger.conf is missing or invalid enable console logging and display + an error message. + + ASTERISK-24817 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4497/ + ........ + + Merged revisions 433122 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433126 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-19 04:57 +0000 [d486659502] Corey Farrell + + * chan_sip: Simplify dialog/peer references, improve REF_DEBUG output. + + * Replace functions for ref/undef of dialogs and peers with macro's + to call ao2_t_bump/ao2_t_cleanup. + * Enable passthough of REF_DEBUG caller information to sip_alloc and + find_call. + + ASTERISK-24882 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4189/ + ........ + + Merged revisions 433115 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-19 04:46 +0000 [2c83ac4364] Corey Farrell + + * chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout. + + Release the scheduler reference to the dialog for reinvite timeout during + dialog_unlink_all. + + ASTERISK-24876 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4491/ + ........ + + Merged revisions 433112 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433113 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 21:42 +0000 [e0ea490a11] Richard Mudgett + + * res_pjsip_session: Fix off-nominal extra unref of session. + ........ + + Merged revisions 433088 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 17:15 +0000 [8c65c9167e] Scott Griepentrog + + * Various: bugfixes found via chaos + + Using DEBUG_CHAOS several instances of a null + pointer crash, and one uninitialized variable + were uncovered and fixed. Also added details + on why Asterisk failed to initialize. + + Review: https://reviewboard.asterisk.org/r/4468/ + ........ + + Merged revisions 433064 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433065 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 17:03 +0000 [f25b265329] Scott Griepentrog + + * core: Introduce chaos into memory allocations + + Locate potential crashes by exercising seldom + used code paths. This patch introduces a new + define DEBUG_CHAOS, and mechanism to randomly + return an error condition from functions that + will seldom do so. Functions that handle the + allocation of memory get the first treatment. + + Review: https://reviewboard.asterisk.org/r/4463/ + ........ + + Merged revisions 433060 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 17:03 +0000 [62cf2a2c02] Scott Griepentrog + + * Reverting accidental ci of wrong change in r433061 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 17:00 +0000 [cb6c7eecfd] Scott Griepentrog + + * various: cleanup issues found during leak hunt + + In this collection of small patches to prevent + Valgrind errors are: fixes for reference leaks + in config hooks, evaluating a parameter beyond + bounds, and accessing a structure after a lock + where it could have been already free'd. + + Review: https://reviewboard.asterisk.org/r/4407/ + ........ + + Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 16:52 +0000 [c41dd32b94] Richard Mudgett + + * Audit ast_sockaddr_resolve() usage for memory leaks. + + Valgrind found some memory leaks associated with ast_sockaddr_resolve(). + Most of the leaks had already been fixed by earlier memory leak hunt + patches. This patch performs an audit of ast_sockaddr_resolve() and found + one more. + + * Fix ast_sockaddr_resolve() memory leak in + apps/app_externalivr.c:app_exec(). + + * Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs + parameter for safety so the pointer will never be uninitialized on return. + The same goes for res/res_pjsip_acl.c:extract_contact_addr(). + + * Made functions that call ast_sockaddr_resolve() with RAII_VAR() + controlling the addrs variable use ast_free instead of ast_free_ptr to + provide better MALLOC_DEBUG information. + + Review: https://reviewboard.asterisk.org/r/4509/ + ........ + + Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 433057 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 13:35 +0000 [803a916334] Kevin Harwell + + * res_pjsip: Allow configuration of endpoint identifier query order + + Updated some documentation stating that endpoint identifiers registered without + a name are place at the front of the lookup list. Also renamed register method + 'ast_sip_register_endpoint_identifier_by_name' to + 'ast_sip_register_endpoint_identifier_with_name' + + ASTERISK-24840 + Reported by: Mark Michelson + ........ + + Merged revisions 433031 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 13:22 +0000 [aef7278af6] Kevin Harwell + + * res_pjsip: Allow configuration of endpoint identifier query order + + This patch fixes previously reverted code that caused binary incompatibility + problems with some modules. And like the original patch it makes sure that + no matter what order the endpoint identifier modules were loaded, priority is + given based on the ones specified in the new global 'endpoint_identifier_order' + option. + + ASTERISK-24840 + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4489/ + ........ + + Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-17 11:11 +0000 [259e833e88] Richard Mudgett + + * res_pjsip: Add reason comment. + ........ + + Merged revisions 433005 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 21:29 +0000 [e89f83b3ad] Matt Jordan + + * main/frame: Don't report empty disallow values as an error + + In realtime, it is normal to have a database with both 'allow' and 'disallow' + columns in the schema. It is perfectly valid to have an 'allow' value of + '!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files, + you can't *not* provide the disallow value. Thus, the empty disallow value + causes a spurious WARNING message, which is kind of annoying. + + This patch makes it so that a 'disallow' value with no ... value ... is + ignored. Granted, you can still screw this up as well, as technically + specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no + 'allow' value in your database. But really, why would you do that? WHY? + + ASTERISK-16779 #close + Reported by: Atis Lezdins + ........ + + Merged revisions 432970 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432971 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 21:01 +0000 [0d52907d2b] Joshua Colp + + * func_curl: Don't hold exclusive lock when performing HTTP request. + + This code originally kept a lock held when performing the HTTP + request to ensure that the options provided to curl remain valid. + This doesn't seem to be necessary these days and holding the lock + caused requests to happen sequentially instead of in parallel. + + ASTERISK-18708 #close + Reported by: Dave Cabot + ........ + + Merged revisions 432948 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432949 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 20:53 +0000 [ac1214d9d4] Jan Juergens (License 6538) + + * apps/app_sms: Add an option to prevent SMS content from being logged + + In some countries, privacy laws specify that SMS content cannot be saved by a + provider. This patch adds a new option to the SMS application, 'n', which + prevents the SMS content from being written to the SMS log. + + ASTERISK-22591 #close + Reported by: Jan Juergens + patches: + DisableSmsContentLoggingByParam.patch uploaded by Jan Juergens (License 6538) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 20:37 +0000 [b3fa35786f] Joshua Colp + + * core: Fix tab completion of "core set debug channel" CLI command. + + The "core set debug channel" CLI command mistakenly had source filenames + added to its tab completion. This occurred because the CLI generator fell back + to the "core set debug" command which permits setting debug at a source + filename level. + + ASTERISK-21038 #close + Reported by: Richard Kenner + ........ + + Merged revisions 432944 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432945 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 20:22 +0000 [b4cc056067] Di-Shi Sun (License 5076) + + * FILE: fix retrieval of file contents when offset is specified + + The loop that reads in a file was not correctly using the offset when + determining what bytes to append to the output. This patch corrects + the logic such that the correct portion of the file is extracted when an + offset is specified. + + ASTERISK-21765 + Reported by: John Zhong + Tested by: Matt Jordan, Di-Shi Sun + patches: + file_read_390821.patch uploaded by Di-Shi Sun (License 5076) + ........ + + Merged revisions 432935 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432938 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 19:24 +0000 [dc752f515b] Matt Jordan + + * apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation + + This patch corrects the documentation for the AMD application. Specifically: + * It documents the maximum_word_length option, which limits the maximum allowed + length of a single utterance. + * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH + was documented as MAXWORDS, while MAXWORDS was undocumented. + + Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues. + + ASTERISK-19470 #close + Reported by: Frank DiGennaro + ........ + + Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432920 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 12:06 +0000 [c52adca396] Richard Mudgett + + * chan_pjsip: AMI action PJSIPShowEndpoint closes AMI connection on error. + + Also fixed similar problem with AMI action PJSIPShowEndpoints. + + ASTERISK-24872 #close + Reported by: Dmitriy Serov + + Review: https://reviewboard.asterisk.org/r/4487/ + ........ + + Merged revisions 432894 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 11:37 +0000 [636d82f4d8] Richard Mudgett + + * chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. + + The res_pjsip modules were manually checking both name and number + presentation values when there is a function that determines the combined + presentation for a party ID struct. The function takes into account if + the name or number components are valid while the manual code rarely + checked if the data was even valid. + + * Made use ast_party_id_presentation() rather than manually checking party + ID presentation values. + + * Ensure that set_id_from_pai() and set_id_from_rpid() will not return + presentation values other than what is pulled out of the SIP headers. It + is best if the code doesn't assume that AST_PRES_ALLOWED and + AST_PRES_USER_NUMBER_UNSCREENED are zero. + + * Fixed copy paste error in add_privacy_params() dealing with RPID + privacy. + + * Pulled the id->number.valid test from add_privacy_header() and + add_privacy_params() up into the parent function add_id_headers() to skip + adding PAI/RPID headers earlier. + + * Made update_connected_line_information() not send out connected line + updates if the connected line number is invalid. Lower level code would + not add the party ID information and thus the sent message would be + unnecessary. + + * Eliminated RAII_VAR usage in send_direct_media_request(). + + Review: https://reviewboard.asterisk.org/r/4472/ + ........ + + Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-13 09:55 +0000 [d42c6adb1a] Kevin Harwell + + * Revert - res_pjsip: Allow configuration of endpoint identifier query order + + Due to a break in binary compatibility with some other modules these changes + are being reverted until the issue can be resolved. + + ASTERISK-24840 + Reported by: Mark Michelson + ........ + + Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-12 21:10 +0000 [f2c21ead1f] Corey Farrell + + * Logger: Fix MALLOC_DEBUG build error. + + Revision 432834 introduced a build error when MALLOC_DEBUG + is used. Switch callid threadstorage to simple + AST_THREADSTORAGE since we no longer need custom cleanup. + + Reported by: Corey Farrell + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-12 20:12 +0000 [c08fd275bf] Corey Farrell + + * Logger: Convert 'struct ast_callid' to unsigned int. + + Switch logger callid's from AO2 objects to simple integers. + This helps in two ways. Copying integers is faster than + referencing AO2 objects, so this will result in a small + reduction in logger overhead. This also erases the possibility + of an infinate loop caused by an invalid callid in + threadstorage. + + ASTERISK-24833 #comment Committed callid conversion to trunk. + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4466/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-12 07:58 +0000 [38ee441ea7] Matt Jordan + + * main/audiohook: Update internal sample rate on reads + + When an audiohook is created (which is used by the various Spy applications + and Snoop channel in Asterisk 13+), it initially is given a sample rate of + 8kHz. It is expected, however, that this rate may change based on the media + that passes through the audiohook. However, the read/write operations on the + audiohook behave very differently. + + When a frame is written to the audiohook, the format of the frame is checked + against the internal sample rate. If the rate of the format does not match + the internal sample rate, the internal sample rate is updated and a new SLIN + format is chosen based on that sample rate. This works just fine. + + When a frame is read, however, we do something quite different. If the format + rate matches the internal sample rate, all is fine. However, if the rates + don't match, the audiohook attempts to "fix up" the number of samples that + were requested. This can result in some seriously large number of samples + being requested from the read/write factories. + + Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of + audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)). + However, if the audiohook is still expecting an internal sample rate of 8000, + we'll attempt to "fix up" the requested samples to: + + samples_converted = samples * (ast_format_get_sample_rate(format) / + (float) audiohook->hook_internal_samp_rate); + + which is: + + 92160 = 3840 * (192000 / 8000) + + This results in us attempting to read 92160 samples from our factories, as + opposed to the 3840 that we actually wanted. On a 64-bit machine, this + miraculously survives - despite allocating up to two buffers of length 92160 + on the stack. The 32-bit machines aren't quite so lucky. Even in the case where + this works, we will either (a) get way more samples than we wanted; or (b) get + about 3840 samples, assuming the timing is pretty good on the machine. + + Either way, the calculation being performed is wrong, based on the API users + expectations. + + My first inclination was to allocate the buffers on the heap. As it is, + however, there's at least two drawbacks with doing this: + (1) It's a bit complicated, as the size of the buffers may change during the + lifetime of the audiohook (ew). + (2) The stack is faster (yay); the heap is slower (boo). + + Since our calculation is flat out wrong in the first place, this patch fixes + this issue by instead updating the internal sample rate based on the format + passed into the read operation. This causes us to read the correct number of + samples, and has the added benefit of setting the audihook with the right + SLIN format. + + Note that this issue was caught by the Asterisk Test Suite as a result of + r432195 in the 13 branch. Because this issue is also theoretically possible + in Asterisk 11, the change is being made here as well. + + Review: https://reviewboard.asterisk.org/r/4475/ + ........ + + Merged revisions 432810 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432811 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432812 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-12 07:40 +0000 [29304d10a0] Diederik de Groot (License 6600) + + * Add support for the clang compiler; update RAII_VAR to use BlocksRuntime + + RAII_VAR, which is used extensively in Asterisk to manage reference counted + resources, uses a GCC extension to automatically invoke a cleanup function + when a variable loses scope. While this functionality is incredibly useful + and has prevented a large number of memory leaks, it also prevents Asterisk + from being compiled with clang. + + This patch updates the RAII_VAR macro such that it can be compiled with clang. + It makes use of the BlocksRuntime, which allows for a closure to be created + that performs the actual cleanup. + + Note that this does not attempt to address the numerous warnings that the clang + compiler catches in Asterisk. + + Much thanks for this patch goes to: + * The folks on StackOverflow who asked this question and Leushenko for + providing the answer that formed the basis of this code: + http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang + * Diederik de Groot, who has been extremely patient in working on getting this + patch into Asterisk. + + Review: https://reviewboard.asterisk.org/r/4370/ + + ASTERISK-24133 + ASTERISK-23666 + ASTERISK-20399 + ASTERISK-20850 #close + Reported by: Diederik de Groot + patches: + RAII_CLANG.patch uploaded by Diederik de Groot (License 6600) + ........ + + Merged revisions 432807 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432808 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-11 11:39 +0000 [4115e327ac] Richard Mudgett + + * res_pjsip: Move internal init/destroy prototypes to private header file. + + Done as a separate commit from a finding in + https://reviewboard.asterisk.org/r/4467/ + ........ + + Merged revisions 432787 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432788 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-11 10:26 +0000 [89b65f5dda] Richard Mudgett + + * res_pjsip: Fix pjsip.conf type=global object default value handling. + + When a type=global section is not defined in pjsip.conf the global + defaults are not applied. As a result the mandatory Max-Forwards header + is not added to SIP messages for res_pjsip/chan_pjsip. + + The handling of pjsip.conf type=global objects has several problems: + + 1) If the global object is missing the defaults are not applied. + + 2) If the global object is missing the default_outbound_endpoint's default + value is not returned by ast_sip_global_default_outbound_endpoint(). + + 3) Defines are needed so default values only need to be changed in one + place. + + * Added a sorcery instance observer callback to check if there were any + type=global sections loaded. If there were more than one then issue an + error message. If there were none then apply the global defaults. + + * Fixed ast_sip_global_default_outbound_endpoint() to return the + documented default when no type=global object is defined. + + * Made defines for the global default values. + + * Increased the default_useragent[] size because SVN version strings can + get lengthy and 128 characters may not be enough. + + * Fixed an off-nominal code path ref leak in global_alloc() if the string + fields fail to initialize. + + * Eliminated RAII_VAR in get_global_cfg() and + ast_sip_global_default_outbound_endpoint(). + + ASTERISK-24807 #close + Reported by: Anatoli + + Review: https://reviewboard.asterisk.org/r/4467/ + ........ + + Merged revisions 432766 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432767 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-11 10:22 +0000 [185d2e082a] Richard Mudgett + + * res_pjsip: Fixed invalid empty Server and User-Agent SIP headers. + + Setting pjsip.conf useragent to an empty string results in an empty SIP + header being sent. + + * Made not add an empty SIP header item to the global SIP headers list. + + Review: https://reviewboard.asterisk.org/r/4467/ + ........ + + Merged revisions 432764 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-10 18:09 +0000 [2889f074a0] Joshua Colp + + * core: Don't create snapshots with locks. + + Snapshots are immutable and are never changed. Allocating them + with a lock is wasteful. + + Review: https://reviewboard.asterisk.org/r/4469/ + ........ + + Merged revisions 432742 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-10 16:33 +0000 [15d266bf85] Javier Acosta (License 6690) + + * res/res_config_odbc: Fix improper escaping of backslashes with MySQL + + When escaping backslashes with MySQL, the proper way to escape the characters + in a LIKE clause is to escape the '\' four times, i.e., '\\\\'. To quote the + MySQL manual: + + "Because MySQL uses C escape syntax in strings (for example, “\n” to represent + a newline character), you must double any “\” that you use in LIKE strings. + For example, to search for “\n”, specify it as “\\n”. To search for “\”, + specify it as “\\\\”; this is because the backslashes are stripped once by the + parser and again when the pattern match is made, leaving a single backslash to + be matched against." + + ASTERISK-24808 #close + Reported by: Javier Acosta + patches: + res_config_odbc.diff uploaded by Javier Acosta (License 6690) + ........ + + Merged revisions 432720 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432721 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-10 13:13 +0000 [ab6e2c93f3] Graham Barnett (License 6685) + + * app_voicemail: Fix crash with IMAP backends when greetings aren't present + + When an IMAP backend is in use and greetings are set to be used, but aren't + present for a user in their IMAP folder, Asterisk will crash. This occurs + due to the mailstream being set to the 'greetings' folder and being left + in that particular state, regardless of the success/failure of the attempt + to access the folder the mailstream points to. Later access of the mailstream + assumes that it points to the 'INBOX' (or some other folder), resulting in + either a crash (if the greetings folder didn't exist and the mailstream is + invalid) or an inability to read messages from the 'INBOX' folder. + + This patch restores the mailstream to its correct state after accessing the + greetings. This fixes the crash, and sets the mailstream to the state that + VoiceMailMain expects. + + Note that while ASTERISK-23390 also contained a patch for this issue, the + patch on ASTERISK-24786 is the one being merged here. + + Review: https://reviewboard.asterisk.org/r/4459/ + + ASTERISK-23390 #close + Reported by: Ben Smithurst + + ASTERISK-24786 #close + Reported by: Graham Barnett + Tested by: Graham Barnett + patches: + app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685) + ........ + + Merged revisions 432695 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432696 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-10 13:05 +0000 [79e9b37ad0] Ed Hynan (Licnese 6680) + + * localtime: Fix file descriptor leak on kqueue(2) systems + + The localtime management in the Asterisk core contains a thread that watches + for changes in the local timezone. On systems where the directory containing + /etc/localtime is modified frequently, the thread monitoring the changes will + be woken up to determine if any changes in timezone have occurred. When using + kqueue(2), this can cause a leak of file descriptors due to some improper + management of resources. + + This patch updates the kqueue(2) handling in localtime, such that is no longer + leaks resources. + + Review: https://reviewboard.asterisk.org/r/4450/ + + ASTERISK-24739 #close + Reported by: Ed Hynan + patches: + 11.15.0-u.diff uploaded by Ed Hynan (Licnese 6680) + 11.7.0-u.diff uploaded by Ed Hynan (License 6680) + svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License 6680) + ........ + + Merged revisions 432691 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432693 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-10 11:08 +0000 [e7ee83ea90] Richard Mudgett + + * res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER. + + A race condition happened between initiating a transfer and requesting + that a dialog termination be delayed. Occasionally, the transferrer + channels would exit the bridge and hangup before the dialog termination + delay was requested. + + * Made request dialog termination delay before initiating the transfer + action. If the transfer fails then cancel the delayed dialog termination + request. + + ASTERISK-24755 #close + Reported by: John Bigelow + + Review: https://reviewboard.asterisk.org/r/4460/ + ........ + + Merged revisions 432668 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-09 11:13 +0000 [1ce529d30e] Kevin Harwell + + * res_pjsip: allow configuration of endpoint identifier query order + + It's possible to have a scenario that will create a conflict between endpoint + identifiers. For instance an incoming call could be identified by two different + endpoint identifiers and the one chosen depended upon which identifier module + loaded first. This of course causes problems when, for example, the incoming + call is expected to be identified by username, but instead is identified by ip. + This patch adds a new 'global' option to res_pjsip called + 'endpoint_identifier_order'. It is a comma separated list of endpoint + identifier names that specifies the order by which identifiers are processed + and checked. + + ASTERISK-24840 #close + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4455/ + ........ + + Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-07 19:47 +0000 [a5f80f1781] Joshua Colp + + * res_rtp_asterisk: Fix wrongful use of USE_PJPROJECT define. + + As pjproject is now used as a shared library a different define, + HAVE_PJPROJECT, is used to specify if pjproject is present. + + ASTERISK-24830 #close + Reported by: Stefan Engström + ........ + + Merged revisions 432614 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-06 16:59 +0000 [affcf1d766] Richard Mudgett + + * res_pjsip_refer: Make safely get the context for a blind transfer. + + Made safely get the TRANSFER_CONTEXT channel value while the channel is + locked in refer_incoming_attended_request() and + refer_incoming_blind_request(). The pointer returned by + pbx_builtin_getvar_helper() is only valid while the channel is locked. + ........ + + Merged revisions 432594 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-06 16:18 +0000 [090ab1735b] Richard Mudgett + + * res_pjsip_refer: Made refer_attended_alloc() not create the ao2 object with a lock. + + The lock is unused. + ........ + + Merged revisions 432574 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-06 15:38 +0000 [b85cb7ea1b] Jonathan Rose + + * app: Add functions to swap voicemail function table for testing purposes + ........ + + Merged revisions 432556 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-06 14:24 +0000 [c7cc1b3059] Richard Mudgett + + * chan_dahdi/sig_analog: Fix distinctive ring detection to suck less. + + The distinctive ring feature interferes with detecting Caller ID and + appears to have been broken for years. What happens is if you have a + ring-ring cadence as used in the UK you get too many DAHDI events for the + distinctive ring pattern array and Caller ID detection is aborted. I + think when Zapata/DAHDI added the ring begin event it broke distinctive + ring. More events happen than before and the code does no filtering of + which event times are recorded in the pattern array. + + * Made distinctive ring only record the ringt count when the ring ends + instead of on just any DAHDI event. Distinctive ring can be ring, + ring-ring, ring-ring-ring, or different ring durations for the up to three + rings. + + * Fixed the distinctive ring detection enable (chan_dahdi.conf option + usedistinctiveringdetection) to be per port instead of somewhat per port + and somewhat global. This has been broken since v1.8. + + * Fixed using the default distinctive ring context when the detected + pattern does not match any configured dringX patterns. The default + context did not get set when the previous call was a matched distinctive + ring pattern and the current call is not matched. This has been broken + since v1.8. + + * Made distinctive ring have no effect on Caller ID detection when it is + disabled. Caller ID detection just monitors for 10 seconds before giving + up. + + * Fixed leak of struct callerid_state memory when a polarity reversal + during Caller ID detection causes the incoming call to be aborted. + + DAHDI-1143 + AST-1545 + ASTERISK-24825 #close + Reported by: Richard Mudgett + + ASTERISK-17588 + Reported by: Daniel Flounders + + Review: https://reviewboard.asterisk.org/r/4444/ + ........ + + Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432534 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-06 13:34 +0000 [f1ab2c5e8b] Richard Mudgett + + * chan_sip: Fix realtime locking inversion when poking a just built peer. + + When a realtime peer is built it can cause a locking inversion when the + just built peer is poked. If the CLI command "sip show channels" is + periodically executed then a deadlock can happen because of the locking + inversion. + + * Push the peer poke off onto the scheduler thread to avoid the locking + inversion of the just built realtime peer. + + AST-1540 + ASTERISK-24838 #close + Reported by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4454/ + ........ + + Merged revisions 432526 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432528 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-05 10:40 +0000 [5c3e33b3ca] gtjoseph + + * app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE. + + There is a leftover "assert" in app_voicemail/__messagecount that references + variables that don't exist. This causes the compile to fail when + --enable-dev-mode and IMAP_STORAGE are selected. + + This patch removes the assert. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4461/ + ........ + + Merged revisions 432484 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432485 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-04 12:55 +0000 [41ba8fd7c0] Matt Jordan + + * translate: Prevent invalid memory accesses on fast shutdown + + When a 'core restart now' or 'core stop now' is executed and a channel is + currently in a media operation, the translator matrix can be destroyed while a + channel is currently blocked on getting the best translation choice + (see ast_translator_best_choice). When the channel gets the mutex, the + translation matrix now has invalid memory, and Asterisk crashes. + + This patch does two things: + (1) We now only clean up the translation matrix on a graceful shutdown. In that + case, there are no channels, and so there is no risk of this occurring. + (2) We also now set the __matrix and __indextable to NULL. In some initial + backtraces when this occurred, it looked as if there was a memory corruption + occurring, and it wasn't until we determined that something had restarted + Asterisk that the issue became clear. By setting these to NULL on shutdown, + it becomes a bit easier to determine why a crash is occurring. + + Note that we could litter the code with NULL checks on the __matrix, but the + act of making the translation matrix cleaned up on shutdown should preclude + this issue from occurring in the first place, and this part of the code needs + to be as fast as possible. + + Review: https://reviewboard.asterisk.org/r/4457/ + ........ + + Merged revisions 432453 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-03-02 13:15 +0000 [278ea2f468] Matt Jordan + + * res/res_pjsip_sdp_rtp: Revert portion of r432195 + + Unfortunately, while initial testing with ConfBridge did not reproduce the + audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing + did show that bridge_softmix and/or ConfBridge has a severe problem bridging + two or more participants at different sampling rates. Sometimes, it even picks + odd sampling rates that cause hideous audio problems. + + This patch backs out the offending portion of the code until the issues in + the affected bridging modules can be more properly analyzed. + + ASTERISK-24841 + ........ + + Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-27 12:31 +0000 [9e841e4fb6] Richard Mudgett + + * ARI: Fix crash if integer values used in JSON payload 'variables' object. + + Sending the following ARI commands caused Asterisk to crash if the JSON + body 'variables' object passes values of types other than strings. + + POST /ari/channels + POST /ari/channels/{channelid} + PUT /ari/endpoints/sendMessage + PUT /ari/endpoints/{tech}/{resource}/sendMessage + + * Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(), + ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and + ast_ari_endpoints_send_message_to_endpoint(). + + ASTERISK-24751 #close + Reported by: jeffrey putnam + + Review: https://reviewboard.asterisk.org/r/4447/ + ........ + + Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-26 12:53 +0000 [d79670b269] Scott Griepentrog + + * Dial API: add self destruct option when complete + + This patch adds a self-destruction option to the + dial api. The usefulness of this is mostly when + using async mode to spawn a separate thread used + to handle the new call, while the calling thread + is allowed to go on about other business. + + The only alternative to this option would be the + calling thread spawning a new thread, or hanging + around itself waiting to destroy the dial struct + after completion. + + Example of use (minus error checking): + + struct ast_dial *dial = ast_dial_create(); + + ast_dial_append(dial, "PJSIP", "200", NULL); + + ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo"); + ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL); + + ast_dial_run(dial, NULL, 1); + + The dial_run call will return almost immediately + after spawning the new thread to run and monitor + the dial. If the call is answered, it is placed + into the echo app. When completed, it will call + ast_dial_destroy() on the dial structure. + + Note that any allocations made to pass values to + ast_dial_set_user_data() or dial options must be + free'd in a state callback function on any of: + AST_DIAL_RESULT_UNASWERED, + AST_DIAL_RESULT_ANSWERED, + AST_DIAL_RESULT_HANGUP, or + AST_DIAL_RESULT_TIMEOUT. + + Review: https://reviewboard.asterisk.org/r/4443/ + ........ + + Merged revisions 432385 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432386 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-26 11:12 +0000 [d04fbb0f9d] Kevin Harwell + + * app_chanspy, channel: fix frame leaks + + Fixed a couple of frame leaks that were found during testing. + + ASTERISK-24828 #close + Reported by: John Hardin + Review: https://reviewboard.asterisk.org/r/4445/ + ........ + + Merged revisions 432362 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432363 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 22:58 +0000 [8a16c2f0c2] Matt Jordan + + * make: Remove 'res_features' from libraries to link against with cygwin/mingw32 + + Both the apps and channels Makefiles still listed 'res_features' as modules to + link against when compiling for cygwin or mingw32. This module hasn't existed + for quite some time. + + ASTERISK-18105 #close + Reported by: feyfre + ........ + + Merged revisions 432341 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432342 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 21:03 +0000 [3725173b9e] Makoto Dei (License 5027) + + * channels/chan_sip: Don't send a BYE after final response when PBX thread fails + + When Asterisk fails to start a PBX thread for a new channel - for example, when + the maxcalls setting in asterisk.conf is exceeded - we currently send a final + response, and then attempt to send a BYE request to the UA. Since that's all + sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt + such that we don't get stuck sending BYE requests to something that does not + want it. + + Note that this patch is a slight modification of the one on ASTERISK-15434. + For clarity, it explicitly calls sipalreadygone with the calls to transmit a + final response. + + ASTERISK-21845 + ASTERISK-15434 #close + Reported by: Makoto Dei + Tested by: Matt Jordan + patches: + sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027) + ........ + + Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432321 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432322 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 17:49 +0000 [e484140aed] Rusty Newton + + * configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1 + + Example configuration files for a "basic PBX" deployment for the fictitious + Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4379/ + and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company + + Reported by: Malcolm Davenport + Tested by: Rusty Newton + + Review: https://reviewboard.asterisk.org/r/4379/ + ........ + + Merged revisions 432301 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 17:09 +0000 [ced84d7e62] Matt Jordan + + * configure: Promote SQLite3 "not installed" warning to error + + Since Asterisk won't build without the library, not having it is definitely + an error. Thanks to Kyle Kurz for pointing this out. + ........ + + Merged revisions 432280 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432281 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 17:05 +0000 [4b63da7f7d] Matt Jordan + + * channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario + + When we receive an SDP as part of an offer/answer for a peer/friend has been + configured to require encryption, and that SDP offer/answer failed to provide + acceptable crypto attributes, we currently issue a WARNING that uses the phrase + "we" and "requested". In this case, both of those terms are ambiguous - the + user will probably think "we" is Asterisk (it most likely isn't) and it may + not be a "request", so much as an SDP that was received in some fashion. + + This patch makes the WARNING messages slightly less bad and a bit more + accurate as well. + + ASTERISK-23214 #close + Reported by: Rusty Newton + ........ + + Merged revisions 432277 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432278 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 15:42 +0000 [d68012d1a3] Olle Johansson (License 5267) + + * channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI + + Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would + be rejected if those crypto attributes contained either a key lifetime or a + MKI parameter. While from a theoretical point of view this was defensible - + Asterisk does not support key lifetimes or multiple crypto keys - from a + practical point of view, this is quite a problem. A large number of endpoints + offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually + have to support anything more than a single key or refresh the key. + In reality, this is (so far as we've seen) always the case. + + This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8 + branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters + in the following fashion: + + > The Lingon branch now handle lifetime and MKI parameters. + > + > We only accept lifetimes up to max for the crypto and higher than 10 hours + > for packetization of 20 ms (50 pps). + > + > We only handle MKI with index 1. + > + > We do not really bother with counting packets and reinviting at end of + > lifetime, so the min of 10 hours kind of takes care of most calls. If there + > are longer ones, we rely on the other side for re-invites. + > + > It's still not perfect, but I personally think this is an improvement. A + > configuration option for minimum lifetime accepted could be added. + + When the patch was ported forward, I decided against adding a configuration + option as Olle's handling was more than sufficient for every case I've seen + come through the issue tracker or through interoperability testing. We can + revisit that decision if it proves to be false. + + A few small other tweaks were made to the surrounding code to reduce + indentation and provide better type safety for the 'tag' parameter. + + Review: https://reviewboard.asterisk.org/r/4419/ + Review: https://reviewboard.asterisk.org/r/4418/ + + ASTERISK-17721 #close + Reported by: Terry Wilson + + ASTERISK-17899 #close + Reported by: Dwayne Hubbard + patches: + lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267) + + ASTERISK-20233 + Reported by: tootai + + ASTERISK-22748 + Reported by: Alejandro Mejia + ........ + + Merged revisions 432239 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432258 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-25 14:47 +0000 [ff642289f4] David M. Lee + + * Increase WebSocket frame size and improve large read handling + + Some WebSocket applications, like [chan_respoke][], require a larger + frame size than the default 8k; this patch bumps the default to 16k. + This patch also fixes some problems exacerbated by large frames. + + The sanity counter was decremented on every fread attempt in + ws_safe_read(), regardless of whether data was read from the socket or + not. For large frames, this could result in loss of sanity prior to + reading the entire frame. (16k frame / 1448 bytes per segment = 12 + segments). + + This patch changes the sanity counter so that it only decrements when + fread() doesn't read any bytes. This more closely matches the original + intention of ws_safe_read(), given that the error message is + "Websocket seems unresponsive". + + This patch also properly logs EOF conditions, so disconnects are no + longer confused with unresponsive connections. + + [chan_respoke]: https://github.com/respoke/chan_respoke + + Review: https://reviewboard.asterisk.org/r/4431/ + ........ + + Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432237 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-24 17:00 +0000 [57525c3cf2] Richard Mudgett + + * config.h: Use real parameter names for ast_variable_new() define. + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-24 16:14 +0000 [8574c4d197] Matt Jordan + + * channels/chan_sip: Fix crash when transmitting packet after thread shutdown + + When the monitor thread is stopped, its pthread ID is set to a specific value + (AST_PTHREADT_STOP) so that later portions of the code can determine whether + or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit + failed to check for that value, checking instead only for AST_PTHREAD_STOP. + Passing the invalid yet very specific value to pthread_kill causes a crash. + + This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that + it doesn't attempt to poke the thread if the thread has already been stopped. + + ASTERISK-24800 #close + Reported by: JoshE + ........ + + Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432199 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-24 16:00 +0000 [a528dfc9a7] Matt Jordan + + * ARI/PJSIP: Apply requesting channel's format cap to created channels + + This patch addresses the following problems: + * ari/resource_channels: In ARI, we currently create a format capability + structure of SLIN and apply it to the new channel being created. This was + originally done when the PBX core was used to create the channel, as there + was a condition where a newly created channel could be created without any + formats. Unfortunately, now that the Dial API is being used, this has two + drawbacks: + (a) SLIN, while it will ensure audio will flows, can cause a lot of + needless transcodings to occur, particularly when a Local channel is + created to the dialplan. When no format capabilities are available, the + Dial API handles this better by handing all audio formats to the requsted + channels. As such, we defer to that API to provide the format + capabilities. + (b) If a channel (requester) is causing this channel to be created, we + currently don't use its format capabilities as we are passing in our own. + However, the Dial API will use the requester channel's formats if none + are passed into it, and the requester channel exists and has format + capabilities. This is the "best" scenario, as it is the most likely to + create a media path that minimizes transcoding. + Fixing this simply entails removing the providing of the format capabilities + structure to the Dial API. + + * chan_pjsip: Rather than blindly picking the first format in the format + capability structure - which actually *can* be a video or text format - we + select an audio format, and only pick the first format if that fails. That + minimizes the weird scenario where we attempt to transcode between video/audio. + + * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. + Since ast_request already limits us down to one format capability once the + format capabilities are passed along, there's no reason to squelch it here. + + * channel: Fixed a comment. The reason we have to minimize our requested + format capabilities down to a single format is due to Asterisk's inability + to convey the format to be used back "up" a channel chain. Consider the + following: + + PJSIP/A => L;1 <=> L;2 => PJSIP/B + g,u,a g,u,a g,u,a u + + That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials + PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local + channel has inherited those format capabilities down the line; PJSIP/B + supports only ulaw. According to these format capabilities, ulaw is + acceptable and should be selected across all the channels, and no + transcoding should occur. However, there is no way to convey this: when L;2 + and PJSIP/B are put into a bridge, we will select ulaw, but that is not + conveyed to PJSIP/A and L;1. Thus, we end up with: + + PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B + g g X u u + + Which causes g722 to be written to PJSIP/B. + + Even if we can convey the 'ulaw' choice back up the chain (which through + some severe hacking in Local channels was accomplished), such that the chain + looks like: + + PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B + u u u u + + We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back + with only 'ulaw'. This results in all the channel structures being set up + correctly, but PJSIP/A *still* sending g722 and causing the chain to fall + apart. + + There's a lot of difficulty just in setting this up, as there are numerous + race conditions in the act of bridging, and no clean mechanism to pass the + selected format backwards down an established channel chain. As such, the + best that can be done at this point in time is clarifying the comment. + + Review: https://reviewboard.asterisk.org/r/4434/ + + ASTERISK-24812 #close + Reported by: Matt Jordan + ........ + + Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-24 12:38 +0000 [91733b5d15] Kevin Harwell + + * bridge_softmix: G.729 codec license held + + When more than one call using the same codec type enters into a softmix bridge + and no audio is present for a channel the bridge optimizes the out frame by + using the same one for all channels with the same codec type. Unfortunately, + when that number (channels with same codec type) dropped to <= 1 the codec + was not dereferenced. At least not until all parties left the bridge. Thus in + the case of G.729 the license was not released. This patch ensures that the + codec is dereferenced immediately when the optimization no longer applies. + + ASTERISK-24797 #close + Reported by: Luke Hulsey + Review: https://reviewboard.asterisk.org/r/4429/ + ........ + + Merged revisions 432174 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432175 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-21 14:48 +0000 [bedf51b2ce] Joshua Colp + + * res_ari_channels: Return a 404 response when a requested channel variable does not exist. + + This change makes it so that if a channel variable is requested and it does not exist + a 404 response will be returned instead of an allocation failed response. This makes + it easier to debug and figure out what is going on for a user. + + ASTERISK-24677 #close + Reported by: Joshua Colp + ........ + + Merged revisions 432154 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-21 13:28 +0000 [87b7060f36] Joshua Colp + + * res_pjsip_registrar: Add Expires header to 200 OK if present in REGISTER. + + Some implementations don't pay attention to the expires for individual contacts. + In this case they may consider the lack of an Expires header in the 200 OK as + unregistered. This change makes it so if an Expires header is present in the REGISTER + we will add one in the 200 OK. + + ASTERISK-24785 #close + Reported by: Ross Beer + ........ + + Merged revisions 432136 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-21 12:53 +0000 [283bb15c16] Joshua Colp + + * res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid. + + ASTERISK-24499 #close + Reported by: Rusty Newton + ........ + + Merged revisions 432118 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-21 11:36 +0000 [b3c1ad5d73] Graham Barnett (License 6685) + + * apps/app_voicemail: Demote an ERROR message to a WARNING message + + When using IMAP voicemail with FreePBX, you will often get ERROR messages + complaining about not being able to find a mailbox. This is due to how FreePBX + handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this + a configuration error, as in any other system it would be indicative of + someone misconfiguring their system. + + Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch + demotes the message so that system administrators can hopefully reduce some + of the noise in their log files. + + Note that in the original patch this was made into a NOTICE, but that's a + too forgiving. + + ASTERISK-24790 #close + Reported by: Graham Barnett + patches: + app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685) + ........ + + Merged revisions 432098 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432099 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-21 08:06 +0000 [2ea7ccbf70] Joshua Colp + + * http: Add missing html tag to 'httpstatus' functionality. + + ASTERISK-24724 #close + Reported by: Ashley Sanders + ........ + + Merged revisions 432078 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432079 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-20 20:58 +0000 [e66b874f5d] Corey Farrell + + * Allow shutdown to unload modules that register bucket scheme's or codec's. + + * Change __ast_module_shutdown_ref to be NULL safe (11+). + * Allow modules that call ast_bucket_scheme_register or ast_codec_register + to be unloaded during graceful shutdown only (13+ only). + + ASTERISK-24796 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4428/ + ........ + + Merged revisions 432058 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432059 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-20 20:51 +0000 [bb71672a47] Corey Farrell + + * main/asterisk.c: Reverse #if statement in listener() to fix code folding. + + listener() opens the same code block in two places (#if and #else). This + confuses some folding editors causing it to think that an extra code block + was opened. Folding in 'geany' causes all code after listener() to be + folded as if it were part of that procedure. + + ASTERISK-24813 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4435/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-20 20:47 +0000 [ce50fa314a] Corey Farrell + + * asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers. + + Add a couple of missing closing brackets / parenthesis. + + ASTERISK-24814 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4436/ + ........ + + Merged revisions 432054 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432055 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432056 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-20 11:55 +0000 [bb06603d5f] Richard Mudgett + + * chan_dahdi/sig_analog: Put log message strings on one line. + + With the log messages on one line, you can search for the log message seen + in the log and expect to find it. + ........ + + Merged revisions 432032 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432034 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-20 11:53 +0000 [340818ad12] Matt Hoskins (license 6688) + + * ASTERISK-24811: Add ast_sorcery_apply_config() to res_pjsip_publish_asterisk. + + Matt Hoskins reported that res_pjsip_publish_asterisk wouldn't pull config from + realtime. Turns out it was just missing a call ast_sorcery_apply_config(). + + res_pjsip_acl was missing it as well, so I added it. The other pjsip modules + looked OK. + + ASTERISK-24811 #close + Reported-by: Matt Hoskins + Tested-by: George Joseph + Tested-by: Matt Hoskins + patches: + res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins (license 6688) + + Review: https://reviewboard.asterisk.org/r/4433/ + ........ + + Merged revisions 432033 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432035 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-20 09:47 +0000 [4dab71831f] Graham Barnett (License 6685) + + * apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange + + When interfacing with Microsoft Exchange, custom headers will be returned as + all lower case. Currently, the IMAP header code will fail to parse the returned + custom headers, as it will be performing a case sensitive comparison. This can + cause playback of messages to fail, as needed information - such as origtime - + will not be present. + + This patch updates app_voicemail's header parsing code to perform a case + insensitive lookup for the requested custom headers. Since the headers are + specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be + unique in an IMAP message, this should cause no issues with other systems. + + ASTERISK-24787 #close + Reported by: Graham Barnett + patches: + app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685) + ........ + + Merged revisions 432012 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 432013 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-19 15:26 +0000 [05cc6d6d55] Richard Mudgett + + * chan_dahdi: Remove some dead code. + ........ + + Merged revisions 431992 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431993 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-19 12:26 +0000 [252aee4228] Richard Mudgett + + * ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association. + + Processing an AOC-E event that does not or no longer has a channel + association causes a crash. + + The problem with posting AOC events to the channel topic is that AOC-E + events don't always have a channel association and posting the event to + the all channels topic is just wrong. AOC-E events do however have their + own charging association method to refer to the agreement with the + charging entity. + + * Changed the AOC events to post to the AMI manager topic instead of the + channel topics. If a channel is associated with the event then channel + snapshot information is supplied with the AMI event. + + * Eliminated RAII_VAR() usage in aoc_to_ami() and ast_aoc_manager_event(). + + This patch supercedes the patch on Review: https://reviewboard.asterisk.org/r/4427/ + + ASTERISK-22670 #close + Reported by: klaus3000 + + ASTERISK-24689 #close + Reported by: Marcel Manz + + ASTERISK-24740 #close + Reported by: Panos Gkikakis + + Review: https://reviewboard.asterisk.org/r/4430/ + ........ + + Merged revisions 431974 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431975 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-19 11:37 +0000 [6992b2e8fa] Richard Mudgett + + * res_pjsip_refer: Handle INVITE with Replaces failure after answer. + + * Fixed hangup handling of the session->channel after answer if the + ast_channel_move() or ast_bridge_impart() fails. We are still the thread + controlling the session->channel so we need to call ast_hangup() to kill + the channel. + + * Fixed debug messages in refer_incoming_invite_request() referencing + incorrect channnels on success. Code comments now say why the + session->channel cannot be used. + + Review: https://reviewboard.asterisk.org/r/4422/ + ........ + + Merged revisions 431956 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-19 09:28 +0000 [e3fd826cdb] Alexander Traud (License 6520) + + * tcptls: Handle new OpenSSL compile time option to disable SSLv3 + + Some distributions are going to disable SSLv3 at compile time. This option can + be checked using the directive OPENSSL_NO_SSL3_METHOD. This patch updates the + TCP/TLS handling in Asterisk to look for that directive before attempting to + use the SSLv3 specific methods. + + ASTERISK-24799 #close + Reported by: Alexander Traud + patches: + no-ssl3-method.patch uploaded by Alexander Traud (License 6520) + ........ + + Merged revisions 431936 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431937 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-18 20:03 +0000 [a4774ceaa5] Corey Farrell + + * Create work around for scheduler leaks during shutdown. + + * Added ast_sched_clean_by_callback for cleanup of scheduled events + that have not yet fired. + * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2. + Cleanup of replace_callno events is only run 11, since it no longer + releases any references or allocations in 13+. + + ASTERISK-24451 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4425/ + ........ + + Merged revisions 431916 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431917 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431918 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-17 09:34 +0000 [09bfe4b208] Richard Mudgett + + * res_pjsip_refer: Fix crash from a REFER and BYE collision. + + Analyzing a one-off crash on a busy system showed that processing a REFER + request had a NULL session channel pointer. The only way I can think of + that could cause this is if an outgoing BYE transaction overlapped the + incoming REFER transaction in a collision. Asterisk sends a BYE while the + phone sends a REFER to complete an attended transfer. + + * Made check the session channel pointer before processing an incoming + REFER request in res_pjsip_refer. + + * Fixed similar crash potential for res_pjsip supplement incoming request + processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, + res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER + messages. + + * Made res_pjsip_messaging respond to a message body too large with a 413 + instead of ignoring it. + + ASTERISK-24700 #close + Reported by: Zane Conkle + + Review: https://reviewboard.asterisk.org/r/4417/ + ........ + + Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431899 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-16 15:29 +0000 [d808eace5c] Matt Jordan + + * res/res_rtp_asterisk: Fix crash in debug from RTCP reports without report block + + When RTCP debugging was enabled, an RTCP report without a report block would + cause a crash. This was due to the verbose output not checking to see if the + report_block pointer was NULl before dereferencing it. + + This patch adds the necessary check to prevent printing any verbose output + if the far side hasn't provided us the information they should have. + + ASTERISK-24791 #close + Reported by: JoshE + Tested by: JoshE + ........ + + Merged revisions 431879 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-15 13:01 +0000 [55eb8fc068] Joshua Colp + + * pjsip: Remove "contact" type from pjsip.conf.sample + + The "contact" object is not meant to be configured from the pjsip.conf + configuration file. It is meant to be created as a result of a registration + and stored elsewhere. + + ASTERISK-24085 #close + Reported by: Rusty Newton + ........ + + Merged revisions 431860 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431861 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-15 12:00 +0000 [55709bc1f7] Joshua Colp + + * install_prereq: Tweak flags when configuring pjproject. + + This change does two things: + 1. Disables debugging so assertions which can return an error do, + instead of asserting. + 2. Enables IPv6 support. + + ASTERISK-24632 #close + Reported by: Rusty Newton + ........ + + Merged revisions 431843 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-15 11:43 +0000 [e78dd39885] Joshua Colp + + * res_sorcery_config: Improve object lookup times. + + The res_sorcery_config module currently uses a fixed bucket + size of 53. This means that depending on the number of objects + you either end up with excess buckets or a lot of collisions. + Due to the way that res_sorcery_config is implemented it's actually + possible to make the bucket size dynamic based on the number of + objects. This is due to the fact that each loading of the config file + produces a new container and does not modify the existing one. + This change uses the number of expected objects and finds a prime + number near it. In practice depending on the number of objects this + can speed up lookups anywhere from 2X to 15X. This change also removes + the lock from the container as it is not needed. + + Review: https://reviewboard.asterisk.org/r/4423/ + ........ + + Merged revisions 431841 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-15 10:01 +0000 [e6fe69b76c] Joshua Colp + + * res_pjsip: Add "pjsip show version" CLI command. + + When debugging things it can be useful to know absolutely what + version of pjproject res_pjsip is running against. This change + adds a "pjsip show version" CLI command which can be used to + query for this. + + ASTERISK-24685 #close + Reported by: Joshua Colp + + Review: https://reviewboard.asterisk.org/r/4424/ + ........ + + Merged revisions 431824 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431825 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-15 06:41 +0000 [17f9e0cacc] Matthias Urlichs (license 5508) + + * res_timing_pthread: Fix leaky pipes. + + During some refactoring the way private information for timers + was stored was changed. As a result of this the action which normally + removed the timer upon closure in res_timing_pthread was also removed + causing the timer to remain after it should using up resources. + This change ensures that the timer is removed upon closure. + + ASTERISK-24768 #close + Reported by: Matthias Urlichs + patches: + timer.patch submitted by Matthias Urlichs (license 5508) + ........ + + Merged revisions 431807 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-14 18:33 +0000 [d1bd8b091b] Matt Jordan + + * apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes + + The Test Event for MIXMONITOR_END - which signals that a MixMonitor has + completed - technically fired before the filestream was closed. If a test + used this to trigger a condition to verify that the file was written, it + could result in a race condition where the file size would not be what the + test expected. + + Luckily, no tests were using this (although they should have been). Since the + test event needed to be moved after the point where the MixMonitor autochan has + been destroyed, the test event no longer emits the channel name. Luckily, + nothing needs it. + ........ + + Merged revisions 431788 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431789 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-14 13:46 +0000 [455a98a2f8] Joshua Colp + + * sorcery: Output an error message if a wizard is specified for an object type and it isn't found. + + ASTERISK-24612 #close + Reported by: Joshua Colp + ........ + + Merged revisions 431771 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-14 12:31 +0000 [fae6bf8ace] Joshua Colp + + * res_pjsip_exten_state: Improve log message when a subscription is attempted to a non-existent extension. + + ASTERISK-24716 #close + Reported by: Rusty Newton + ........ + + Merged revisions 431754 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431755 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-14 12:21 +0000 [cc96e4a7ef] Joshua Colp + + * Multiple revisions 431751-431752 + + ........ + r431751 | file | 2015-02-14 14:19:07 -0400 (Sat, 14 Feb 2015) | 5 lines + + chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type. + + ASTERISK-24771 #close + Reported by: Niklas Larsson + ........ + r431752 | file | 2015-02-14 14:20:27 -0400 (Sat, 14 Feb 2015) | 2 lines + + 'information' ends with an 'n'. + ........ + + Merged revisions 431751-431752 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-13 11:24 +0000 [f00ebf0a2d] Richard Mudgett + + * res_pjsip_session: Fix double re-INVITE collision crash. + + A multi-asterisk box setup with direct media enabled would occasionally + crash when two re-INVITE collisions on a call leg happen in a row. + + The re-INVITE logic only had one timer struct to defer the re-INVITE. + When the second collision happens the timer struct is overwritten and put + into the timer heap again. Resources for the first timer are leaked and + the heap has two positions occupied by the same timer struct. Now the + heap ordering is potentially corrupted, the timer will fire twice, and any + resources allocated for the second timer will be released twice. + + * The solution is to put the collided re-INVITE into the delayed requests + queue with all the other delayed requests and cherry pick the next request + that can come off the queue when an event happens. + + * Changed to put delayed BYE requests at the head of the delayed queue. + There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE + has been requested. + + * Made the start of a BYE request flush the delayed requests queue to + prevent a delayed request from overlapping the BYE transaction. I saw a + few cases where a delayed re-INVITE got started after the BYE transaction + started. + + * Changed the delayed_request struct to use an enum instead of a string + for the request method. Cherry picking the queue is easier with an enum + than string comparisons and the compiler can warn if a switch statement + does not cover all defined enum values. + + * Improved the debug output to give more information. It helps to know + which channel is involved with an endpoint. Trunks can have many channels + associated with the endpoint at the same time. + + ASTERISK-24727 #close + Reported by: Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4414/ + ........ + + Merged revisions 431734 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-12 14:34 +0000 [29f66b0429] Matt Jordan + + * ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app + + This patch adds a new feature to ARI to redirect a channel to another server, + and fixes a few bugs in PJSIP's handling of the Transfer dialplan + application/ARI redirect capability. + + *New Feature* + A new operation has been added to the ARI channels resource, redirect. With + this, a channel in a Stasis application can be redirected to another endpoint + of the same underlying channel technology. + + *Bug fixes* + In the process of writing this new feature, two bugs were fixed in the PJSIP + stack: + (1) The existing .transfer channel callback had the limitation that it could + only transfer channels to a SIP URI, i.e., you had to pass + 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is + still supported, it is somewhat unintuitive - particularly in a world full + of endpoints. As such, we now also support specifying the PJSIP endpoint to + transfer to. + (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by + updating its Contact header. Alas, that resulted in the forwarding + destination set by the dialplan application/ARI resource/whatever being + rewritten with very incorrect information. Hence, we now don't bother + updating an outgoing response if it is a 302. Since this took a looong time + to find, some additional debug statements have been added to those modules + that update the Contact headers. + + Review: https://reviewboard.asterisk.org/r/4316/ + + ASTERISK-24015 #close + Reported by: Private Name + + ASTERISK-24703 #close + Reported by: Matt Jordan + ........ + + Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-11 12:03 +0000 [9d081ed06c] Kevin Harwell + + * res_pjsip: dtls_handler causes Asterisk to crash + + There have been a couple of times where a crash occurred in the dtls_handler + section of the code for res_pjsip. Unfortunately, in working this issue the + problem was unable to be reproduced. After looking at the backtraces and + through the code the current best guess as to why this happened might be due + to a reentrance problem and the strtok function. So, the current fix is to + convert the strtok function into the reentrant version of the function, + strtok_r. + + ASTERISK-24741 #close + Reported by: Zane Conkle + Review: https://reviewboard.asterisk.org/r/4409/ + ........ + + Merged revisions 431698 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-11 11:45 +0000 [cc85e55d88] Kevin Harwell + + * ari_websockets: removed extra check on websocket session read + + When merging the websocket timeout issue (ASTERISK-24701) an extra, almost + duplicate, check was left in the code that should not have been. This removes + it. + + ASTERISK-24701 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4412/ + ........ + + Merged revisions 431693 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431695 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-11 11:39 +0000 [e2d3215b83] Richard Mudgett + + * HTTP: Stop accepting requests on final system shutdown. + + There are three CLI commands to stop and restart Asterisk each. + + 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. + New channels are prevented while the shutdown request is pending. + + 2) core stop/restart gracefully - Stop or restart Asterisk when there are + no calls remaining in the system. New channels are prevented while the + shutdown request is pending. + + 3) core stop/restart when convenient - Stop or restart Asterisk when there + are no calls in the system. New calls are not prevented while the + shutdown request is pending. + + ARI has made stopping/restarting Asterisk more problematic. While a + shutdown request is pending it is desirable to continue to process ARI + HTTP requests for current calls. To handle the current calls while a + shutdown request is pending, a new committed to shutdown phase is needed + so ARI applications can deal with the calls until the system is fully + committed to shutdown. + + * Added a new shutdown committed phase so ARI applications can deal with + calls until the final committed to shutdown phase is reached. + + * Made refuse new HTTP requests when the system has reached the final + system shutdown phase. Starting anything while the system is actively + releasing resources and unloading modules is not a good thing. + + * Split the bridging framework shutdown to not cleanup the global bridging + containers when shutting down in a hurry. This is similar to how other + modules prevent crashes on rapid system shutdown. + + * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and + ast_shutting_down(). You should not have to include channel.h just to + access these system functions. + + ASTERISK-24752 #close + Reported by: Matthew Jordan + + Review: https://reviewboard.asterisk.org/r/4399/ + ........ + + Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-11 11:13 +0000 [5a17ed7a38] Richard Miller (License 5685) + + * channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB + + When a SIP device that has its registration stored in RealTime unregisters, + the entry for that device is updated with blank values, i.e., "", indicating + that it is no longer registered. Unfortunately, one of those values that is + 'blanked' is the device's port. If the column type for the port is not a + string datatype (the recommended type is integer), an ODBC or database error + will be thrown. MariaDB does not coerce empty strings to a valid integer value. + + This patch updates the query run from chan_sip such that it replaces the port + value with a value of '0', as opposed to a blank value. This is the value that + other database backends coerce the empty string ("") to already, and the + handling of reading a RealTime registration value from a backend already + anticipates receiving a port of '0' from the backends. + + ASTERISK-24772 #close + Reported by: Richard Miller + patches: + chan_sip.diff uploaded by Richard Miller (License 5685) + ........ + + Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431674 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-11 11:03 +0000 [8cc50b1ebc] Corey Farrell + + * Enable REF_DEBUG for ast_module_ref / ast_module_unref. + + Add ast_module_shutdown_ref for use by modules that can + only be unloaded during graceful shutdown. + + When REF_DEBUG is enabled: + * Add an empty ao2 object to struct ast_module. + * Allocate ao2 object when the module is loaded. + * Perform an ao2_ref in each place where mod->usecount is manipulated. + * ao2_cleanup on module unload. + + ASTERISK-24479 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4141/ + ........ + + Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431663 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-11 10:52 +0000 [137c4b0778] Kevin Harwell + + * res_http_websocket: websocket write timeout fails to fully disconnect + + When writing to a websocket if a timeout occurred the underlying socket did not + get closed/disconnected. This patch makes sure the websocket gets disconnected + on a write timeout. Also a notice is logged stating that the websocket was + disconnected. + + ASTERISK-24701 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4412/ + ........ + + Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431670 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-10 17:17 +0000 [49161d8df8] gtjoseph + + * res_pjsip_config_wizard: Add ability to auto-create hints. + + Looking at the Super Awesome Company sample reminded me that creating hints is + just plain gruntwork. So you can now have the pjsip conifg wizard auto-create + them for you. + + Specifying 'hint_exten' in the wizard will create + 'exten => ,hint/PJSIP/' + in whatever is specified for 'hint_context'. + + Specifying 'hint_application' in the wizard will create + 'exten => ,1,' + in whatever is specified for 'hint_context'. + + The default for 'hint_context' is the endpoint's context. + There's no default for 'hint_application'. If not specified, no app is added. + There's no default for 'hint_exten'. If not specified, neither the hint itself + nor the application will be created. + + Some may think this is the slippery slope to users.conf but hints are a basic + necessity for phones unlike voicemail, manager, etc that users.conf creates. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/4383/ + ........ + + Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-08 21:12 +0000 [858e825568] Ben Merrills (License 6678) + + * res/ari/resource_channels: Add missing 'no_answer' reason to DELETE /channels + + One of the canonical reasons for hanging up a channel is because the far end + failed to answer - or because someone else answered, and we want to get rid of + this channel. This patch adds the missing value to the 'reason' query parameter + for the DELETE /channels operation. + + Review: https://reviewboard.asterisk.org/r/4400 + + ASTERISK-24745 #close + Reported by: Ben Merrills + patches: + add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678) + ........ + + Merged revisions 431622 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-08 20:35 +0000 [17247daae6] ibercom (License 6599) + + * res/res_odbc: Remove unneeded queries when determining if a table exists + + This patch modifies the ast_odbc_find_table function such that it only performs + a lookup of the requested table if the table is not already known. Prior to + this patch, a queries would be executed against the database even if the table + was already known and cached. + + Review: https://reviewboard.asterisk.org/r/4405/ + + ASTERISK-24742 #close + Reported by: ibercom + patches: + patch.diff uploaded by ibercom (License 6599) + ........ + + Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431618 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-08 11:24 +0000 [2ebe811d80] Matt Jordan + + * res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDP + + When an SDP is created for an outgoing request/response, the ICE candidates + obtained from the RTP instance are currently leaked. This causes the ao2 + container that holds the candidates to never properly be reclaimed when the + RTP instance is destroyed. + + This patch properly decrements the ICE candidates' container if it is + successfully obtained. + + ASTERISK-24769 #close + Reported by: Matt Jordan + ........ + + Merged revisions 431600 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431601 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-06 15:26 +0000 [7ca1a0da04] Scott Griepentrog + + * various: cleanup issues found during leak hunt + + In this collection of small patches to prevent + Valgrind errors are: fixes for reference leaks + in config hooks, evaluating a parameter beyond + bounds, and accessing a structure after a lock + where it could have been already free'd. + + Review: https://reviewboard.asterisk.org/r/4407/ + ........ + + Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431584 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-03 19:27 +0000 [a79c920aa1] Joshua Colp + + * res_pjsip_keepalive: Don't crash if PJSIP module is not loaded. + ........ + + Merged revisions 431555 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-02-03 18:59 +0000 [03ce56d6c5] Joshua Colp + + * sorcery: Don't try to load object types which haven't been defined. + + The act of defining wizards for an object type in sorcery.conf will + create a minimal object type. This can cause a problem when a module + has multiple sorcery instances (which all get the wizards from sorcery.conf + applied) but the sorcery instances do not all contain full information + about the object types. Upon loading errors will occur stating that + the objects can not be created. This is confusing and is actually + perfectly fine. + + This change makes it so that only object types which have been fully + defined will be loaded. + + ASTERISK-24748 #close + Reported by: Joshua Colp + ........ + + Merged revisions 431538 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-31 10:28 +0000 [14a57782a6] Joshua Colp + + * res_format_attr_h264: Fix crash when determining joint capability. + + The res_format_attr_h264 module currently incorrectly attempts to + copy SPS and PPS information from the wrong attribute. This change + fixes that. + + ASTERISK-24616 #close + Reported by: Yura Kocyuba + + Review: https://reviewboard.asterisk.org/r/4392/ + ........ + + Merged revisions 431521 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-30 11:49 +0000 [23bb5f6a73] Richard Mudgett + + * app_agent_pool: Fix initial module load agent device state reporting. + + When the app_agent_pool module initially loads there is a race condition + between the thread loading agents.conf and the device state internal + processing thread. If the device state internal processing thread handles + the agent creation state updates before the thread that loaded agents.conf + registers the device state provider callback then the cached agent state + is "Invalid". When a consumer module like app_queue asks for the agent state + it gets the cached "Invalid" state instead of the real state from the provider. + + * Moved loading the agents.conf configuration to the last thing setup by + app_agent_pool in load_module(). Now the device state provider callback + is registered before the config is loaded so the agent creation state + updates are guaranteed to get the initial device state. + + * Removed some now redundant config cleanup on error in load_config(). + + * Added lock protection when accessing the device state in + agent_pvt_devstate_get() and eliminated the RAII_VAR() usage. + + ASTERISK-24737 #close + Reported by: Steve Pitts + + Review: https://reviewboard.asterisk.org/r/4390/ + ........ + + Merged revisions 431492 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-30 11:41 +0000 [5c9f1b3f51] Kevin Harwell + + * res_pjsip_outbound_publish: eventually crashes when no response is ever received + + When Asterisk attempts to send SIP outbound publish information and no response + is ever received (no 200 okay, 412, 423) the system eventually crashes. A + response is never received because the system Asterisk is attempting to send + publish information to is not available. The underlying pjsip framework attempts + to send publish information. After several attempts it calls back into the + Asterisk outbound publish code. At this point if the "client->queue" is empty + Asterisk attempts to schedule a refresh which utilizes "rdata" and since no + response was received the given "rdata" struture is NULL. Attempting to + dereference a NULL object of course results in a crash. + + The fix here removes the dependency on rdata for schedule_publish_refresh. + Instead param->expiration is now passed to it as this is set to -1 if no + response is received. Also added a notification when no response is received. + + ASTERISK-24635 #close + Reported by: Marco Paland + Review: https://reviewboard.asterisk.org/r/4384/ + ........ + + Merged revisions 431490 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431491 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-30 11:21 +0000 [6a76740b83] Ashley Sanders + + * HTTP: For httpd server, need option to define server name for security purposes + + Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default. + + ASTERISK-24316 #close + Reported By: Andrew Nagy + Review: https://reviewboard.asterisk.org/r/4374/ + ........ + + Merged revisions 431471 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-30 10:49 +0000 [bd0bdf1e41] Mark Michelson + + * Fix some memory leaks. + + These memory leaks were found and fixed by John Hardin. I'm just + committing them for him. + + ASTERISK-24736 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4389 + ........ + + Merged revisions 431468 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-29 17:03 +0000 [388d691f34] Scott Griepentrog + + * stasis transfer: fix stasis bridge push race part two + + When swapping a Local channel in place of one already + in a bridge (to complete a bridge attended transfer), + the channel that was swapped out can actually be hung + up before the stasis bridge push callback executes on + the independant transfer thread. This results in the + stasis app loop dropping out and removing the control + that has the the app name which the local replacement + channel needs so it can re-enter stasis. + + To avoid this race condition a new push_peek callback + has been added, and called from the ast_bridge_impart + thread before it launches the independant thread that + will complete the transfer. Now the stasis push_peek + callback can copy the stasis app name before the swap + channel can hang up. + + ASTERISK-24649 + Review: https://reviewboard.asterisk.org/r/4382/ + ........ + + Merged revisions 431450 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-29 15:20 +0000 [f61c80a8f7] Mark Michelson + + * Allow disabling of 100rel support on PJSIP endpoints. + + Due to an inversion error, setting 100rel=no would not actually + change the current value of the setting (which defaulted to "yes"). + With this fix, the inversion is corrected. + ........ + + Merged revisions 431420 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-29 15:02 +0000 [034798e37e] Mark Michelson + + * Use SIPS URIs in Contact headers when appropriate. + + RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific + scenarios when we are required to use SIPS URIs in Contact + headers. Asterisk's non-compliance with this could actually + cause calls to get dropped when communicating with clients + that are strict about checking the Contact header. + + Both of the SIP stacks in Asterisk suffered from this issue. + This changeset corrects the behavior in res_pjsip/chan_pjsip.c + + Review: https://reviewboard.asterisk.org/r/4345 + ........ + + Merged revisions 431426 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-29 14:54 +0000 [fe76d4829f] Mark Michelson + + * Use SIPS URIs in Contact headers when appropriate. + + RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific + scenarios when we are required to use SIPS URIs in Contact + headers. Asterisk's non-compliance with this could actually + cause calls to get dropped when communicating with clients + that are strict about checking the Contact header. + + Both of the SIP stacks in Asterisk suffered from this issue. + This changeset corrects the behavior in chan_sip. + + ASTERISK-24646 #close + Reported by Stephan Eisvogel + + Review: https://reviewboard.asterisk.org/r/4346 + ........ + + Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431424 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-29 10:47 +0000 [8357ffab9c] gtjoseph + + * res_pjsip_exten_state: Reduce log clutter... change a WARNING to a VERBOSE/2 + + Reduce log clutter by changing the "Watcher for hint %s (removed|deactivated)" + message from WARNING to VERBOSE/2. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4387/ + ........ + + Merged revisions 431403 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-29 06:09 +0000 [9893ba7ffb] Joshua Colp + + * res_rtp_asterisk: Fix DTLS when used with OpenSSL 1.0.1k + + A recent security fix for OpenSSL broke DTLS negotiation for many + applications. This was caused by read ahead not being enabled when it + should be. While a commit has gone into OpenSSL to force read ahead + on for DTLS it may take some time for a release to be made and the + change to be present in distributions (if at all). As enabling read + ahead is a simple one line change this commit does that and fixes + the issue. + + ASTERISK-24711 #close + Reported by: Jared Biel + ........ + + Merged revisions 431384 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431385 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431386 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-28 11:42 +0000 [b3ff43a4e8] Mark Michelson + + * Fix file descriptor leak in RTP code. + + SIP requests that offered codecs incompatible with configured values + could result in the allocation of RTP and RTCP ports that would not get + reclaimed later. + + ASTERISK-24666 #close + Reported by Y Ateya + + Review: https://reviewboard.asterisk.org/r/4323 + + AST-2015-001 + ........ + + Merged revisions 431300 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 431303 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-28 11:34 +0000 [3cccfac399] Mark Michelson + + * Multiple revisions 431297-431298 + + ........ + r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines + + Mitigate possible HTTP injection attacks using CURL() function in Asterisk. + + CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection + can be performed given properly-crafted URLs. + + Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may + get cURL URLs from user input or remote sources, we have made a patch to Asterisk + to prevent such HTTP injection attacks from originating from Asterisk. + + ASTERISK-24676 #close + Reported by Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4364 + + AST-2015-002 + ........ + r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines + + Fix compilation error from previous patch. + ........ + + Merged revisions 431297-431298 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431299 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 431301 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-28 06:19 +0000 [f080ca6536] Sean Bright + + * media formats: update res_format_attr_opus & silk + + In r419044, we changed how formats were handled, but the return value + of the format_parse_sdp_fmtp functions in res_format_attr_opus and + res_format_attr_silk were not updated, causing calls to fail. Ran + into this when getting codec_opus working with Asterisk 13. + + Once the return value was corrected, we were crashing in opus_getjoint + because of NULL format attributes. I've fixed this as well in this + patch. + + Review: https://reviewboard.asterisk.org/r/4371/ + ........ + + Merged revisions 431267 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 22:29 +0000 [69e107b24e] Richard Mudgett + + * res_pjsip_outbound_registration: Fix reload race condition. + + Performing a CLI "module reload" command when there are new pjsip.conf + registration objects defined frequently failed to load them correctly. + + What happens is a race condition between res_pjsip pushing its reload into + an asynchronous task processor task and the thread that does the rest of + the reloads when it gets to reloading the res_pjsip_outbound_registration + module. A similar race condition happens between a reload and the CLI/AMI + show registrations commands. The reload updates the current_states + container and the CLI/AMI commands call get_registrations() which builds a + new current_states container. + + * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() + instead of ast_sip_push_task() to eliminate two threads processing config + reloads at the same time. + + * Made get_registrations() not replace the global current_states container + so the CLI/AMI show registrations command cannot interfere with reloading. + You could never add/remove objects in the container without the + possibility of the container being replaced out from under you by + get_registrations(). + + * Added a registration loaded sorcery instance observer to purge any dead + registration objects since get_registrations() cannot do this job anymore. + The struct ast_sorcery_instance_observer callbacks must be used because + the callback happens inline with the load process. The struct + ast_sorcery_observer callbacks are pushed to a different thread. + + * Added some global current_states NULL pointer checks in case the + container disappears because of unload_module(). + + * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded + callbacks guaranteed to be called before any struct + ast_sorcery_observer.loaded callbacks will be called. + + * Moved the check for non-reloadable objects to before the sorcery + instance loading callbacks happen to short circuit unnecessary work. + Previously with non-reloadable objects, the sorcery instance + loading/loaded callbacks would always happen, the individual wizard + loading/loaded would be prevented, and the non-reloadable type logging + message would be logged for each associated wizard. + + ASTERISK-24729 #close + Review: https://reviewboard.asterisk.org/r/4381/ + ........ + + Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 16:58 +0000 [c7591ef6bc] Kevin Harwell + + * tcptls: Bad file descriptor error when reloading chan_sip + + While running through some scenarios using chan_sip and tcp a problem would + occur that resulted in a flood of bad file descriptor messages on the cli: + + tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor + + The message is received because the underlying socket has been closed, so is + valid. This is probably happening because unloading of chan_sip is not atomic. + That however is outside the scope of this patch. This patch simply stops the + logging of multiple occurrences of that message. + + ASTERISK-24728 #close + Reported by: Thomas Thompson + Review: https://reviewboard.asterisk.org/r/4380/ + ........ + + Merged revisions 431218 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431219 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 13:31 +0000 [e826cb8a26] Jonathan Rose + + * Manager: Fix Manager Action ModuleLoad to give correct response when reloading + + Prior to this patch, ModuleLoad would respond with an error indicating that + the requested module wasn't found in spite of finding and reloading the + module. + + Review: https://reviewboard.asterisk.org/r/4373/ + ASTERISK-24721 #close + ........ + + Merged revisions 431153 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 13:22 +0000 [3b0f03ef7b] Kevin Harwell + + * chan_sip: stale nonce causes failure + + When refreshing (with a small expiration) a registration that was sent to + chan_sip the nonce would be considered stale and reject the registration. + What was happening was that the initial registration's "dialog" still existed + in the dialogs container and upon refresh the dialog match algorithm would + choose that as the "dialog" instead of the newly created one. This occurred + because the algorithm did not check to see if the from tag matched if + authentication info was available after the 401. So, it ended up assuming + the original "dialog" was a match and stopped the search. The old "dialog" + of course had an old nonce, thus the stale nonce message. + + This fix attempts to leave the original functionality alone except in the case + of a REGISTER. If a REGISTER is received if searches for an existing "dialog" + matching only on the callid. If the expires value is low enough it will reuse + dialog that is there, otherwise it will create a new one. + + ASTERISK-24715 #close + Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/4367/ + ........ + + Merged revisions 431187 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431194 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 13:12 +0000 [e62bd46511] Corey Farrell (license 5909) + + * res_pjsip: make it unloadable (take 2) + + Due to the original patch causing memory corruptions it was removed until the + problem could be resolved. This patch is the original patch plus some added + locking around stasis router subcription that was needed to avoid the memory + corruption. + + Description of the original problem and patch (still applicable): + + The res_pjsip module was previously unloadable. With this patch it can now + be unloaded. + + This patch is based off the original patch on the issue (listed below) by Corey + Farrell with a few modifications. Namely, removed a few changes not required to + make the module unloadable and also fixed a bug that would cause asterisk to + crash on unloading. + + This patch is the first step (should hopefully be followed by another/others at + some point) in allowing res_pjsip and the modules that depend on it to be + unloadable. At this time, res_pjsip and some of the modules that depend on + res_pjsip cannot be unloaded without causing problems of some sort. + + The goal of this patch is to get res_pjsip and only res_pjsip to be able to + unload successfully and/or shutdown without incident (crashes, leaks, etc...). + Other dependent modules may still cause problems on unload. + + Basically made sure, with the patch applied, that res_pjsip (with no other + dependent modules loaded) could be succesfully unloaded and Asterisk could + shutdown without any leaks or crashes that pertained directly to res_pjsip. + + ASTERISK-24485 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4363/ + patches: + pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) + ........ + + Merged revisions 431179 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 11:48 +0000 [94eebd5ba5] Richard Mudgett + + * app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel. + + Starting and stopping conference recording more than once causes the + recording channels to be leaked. For v13 the channels also show up in the + CLI "core show channels" output. + + * Reworked and simplified the recording channel code to use + ast_bridge_impart() instead of managing the recording thread in the + ConfBridge code. The recording channel's ref handling easily falls into + place and other off nominal code paths get handled better as a result. + + ASTERISK-24719 #close + Reported by: John Bigelow + + Review: https://reviewboard.asterisk.org/r/4368/ + Review: https://reviewboard.asterisk.org/r/4369/ + ........ + + Merged revisions 431135 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431160 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 11:34 +0000 [a43d24a9d3] Joshua Colp + + * bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media. + + This change fixes two issues: + + 1. During a swap operation bridging added the new channel before having the swap channel + leave. This was not handled in bridge_native_rtp and could result in a channel not getting + reinvited back to Asterisk. After this change the swap channel will leave first and the + new channel will then join. + + 2. If a re-invite was received after a session had been established any upstream elements + (such as bridge_native_rtp) were not notified that they may want to re-evaluate things. + After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs + and upstream can react. + + AST-1524 #close + + Review: https://reviewboard.asterisk.org/r/4378/ + ........ + + Merged revisions 431157 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 11:21 +0000 [fb8a2e0399] Matt Jordan + + * ARI: Improve wiki documentation + + This patch improves the documentation of ARI on the wiki. Specifically, it + addresses the following: + * Allowed values and allowed ranges weren't documented. This was particularly + frustrating, as Asterisk would reject query parameters with disallowed values + - but we didn't tell anyone what the allowed values were. + * The /play/id operation on /channels and /bridges failed to document all of + the added media resource types. + * Documentation for creating a channel into a Stasis application failed to + note when it occurred, and that creating a channel into Stasis conflicts with + creating a channel into the dialplan. + * Some other minor tweaks in the mustache templates, including italicizing the + parameter type, putting the default value on its own sub-bullet, and some + other nicities. + + Review: https://reviewboard.asterisk.org/r/4351 + ........ + + Merged revisions 431145 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 11:16 +0000 [aa8fd7d1b9] Matt Jordan + + * app_confbridge: Restore user's menu name to CLI output of 'confbridge list' + + When issuing a 'confbridge list XXXX' CLI command, the resulting output no + longer displays the menu associated with a ConfBridge participant. + + The issue was caused by ASTERISK-22760. When that patch was done, it removed + the copying of the menu name associated with the user from the actual user + profile. + + This patch fixes the issue by copying the menu name over to the user profile + when the menu hooks are applied to the user. Since that function now does a + little bit more than just apply the hooks, the name of the function has been + changed to cover the copying of the menu name over as well. + + In addition, there is a disparity between the menu name length as it is stored + on the conf_menu structure and the confbridge_user structure; this patch makes + the lengths match so that a strcpy can be used. + + Review: https://reviewboard.asterisk.org/r/4372/ + + ASTERISK-24723 #close + Reported by: Steve Pitts + ........ + + Merged revisions 431134 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-27 05:47 +0000 [2504f97b01] Joshua Colp + + * res_parking: Fix crash due to race condition when unloading. + + There is currently a race condition when unloading the res_parking + module. Depending on the will of the universe the subscription + invocation may occur AFTER the module is unloaded. This is because + the module does NOT use stasis_unsubscribe_and_join when terminating + the subscription. It merely uses stasis_unsubscribe. + + This change makes it use stasis_unsubscribe_and_join which is documented + for usage in this exact scenario. + + AST-1520 #close + + Review: https://reviewboard.asterisk.org/r/4375/ + ........ + + Merged revisions 431114 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-26 08:50 +0000 [965777ccfc] David M. Lee + + * Various fixes for OS X + + This patch addresses compilation errors on OS X. It's been a while, so + there's quite a few things. + + * Fixed __attribute__ decls in route.h to be portable. + * Fixed htonll and ntohll to work when they are defined as macros. + * Replaced sem_t usage with our ast_sem wrapper. + * Added ast_sem_timedwait to our ast_sem wrapper. + * Fixed some GCC 4.9 warnings using sig*set() functions. + * Fixed some format strings for portability. + * Fixed compilation issues with res_timing_kqueue (although tests still fail + on OS X). + * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue + on OS X). + + ASTERISK-24539 #close + Reported by: George Joseph + + ASTERISK-24544 #close + Reported by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4327/ + ........ + + Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-25 07:43 +0000 [a8ae5a7bcb] Matt Jordan + + * dynamic realtime: Updates fail to work due to update fields being passed over + + When a crash was fixed due to usage of the REALTIME function in r423003, a + regression was introduced into ast_update2_realtime where the update fields + passed to the function would be skipped and the lookup field processed twice. + + The use of this function is a bit interesting: A variable argument list is + used with two sentinel values - the first marks the end of the lookup + fields/values; the second marks the end of the update fields/values. + Unfortunately, ast_update2_realtime parses over the lookup fields twice, as + opposed to parsing over the update fields. This causes the lookups to succeed, + but the updates itself to have no effect. + + Note that the most common instance of this problem occurred in app_voicemail + during the updating of a mailbox password. + + Thanks to the issue reporter, Paddy Grice, for pointing out the problem. + + Review: https://reviewboard.asterisk.org/r/4356/ + + ASTERISK-24231 + + ASTERISK-24626 #close + Reported by: Paddy Grice + ........ + + Merged revisions 431072 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431073 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 14:17 +0000 [b69b0d12ee] Richard Mudgett + + * app_confbridge: Shorten CBRec channel names to CBRec/- + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431055 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 14:14 +0000 [c780223507] Richard Mudgett + + * app_confbridge: Make CBRec channel names more unique. + + Channel names should be different from other channels in the system while + the channel exists. + + * Use a sequence number for CBRec channels instead of a random number + because the same random number could be picked again for the next CBRec + channel. + ........ + + Merged revisions 431052 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 13:51 +0000 [b38be992b1] Richard Mudgett + + * app_confbridge: Whitespace + + Because there is sometimes no sence to any whitespace. + ........ + + Merged revisions 431049 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 431050 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 12:46 +0000 [89610adda5] David M. Lee + + * Add depend on pjproject to res_pjsip_config_wizard.c + ........ + + Merged revisions 431030 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 09:21 +0000 [ca02121ef7] Kevin Harwell + + * Investigate and fix memory leaks in Asterisk + + Fixed memory leaks that were found in Asterisk. + + ASTERISK-24693 #close + Reported by: Kevin Harwell + Review: https://reviewboard.asterisk.org/r/4347/ + ........ + + Merged revisions 430999 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 09:13 +0000 [49cbfa7de6] Walter Doekes + + * Fix typo's (retrieve, specified, address). + ........ + + Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-23 08:39 +0000 [874cb5615d] HZMI8gkCvPpom0tM (License 6658) + + * chan_sip: Case insensitive comparison of "defaultuser" parameter. + + All the other configuration options are case insensitive, so this one + should be too. + + ASTERISK-24355 #close + Reported by: HZMI8gkCvPpom0tM + patches: + ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658) + ........ + + Merged revisions 430993 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430994 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430995 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-22 13:30 +0000 [9bff4eeca3] Richard Mudgett + + * Bridge core: Pass a ref with the swap channel when joining a bridge. + + When code imparts a channel into a bridge to swap with another channel, a + ref needs to be held on the swap channel to ensure that it cannot + dissapear before finding it in the bridge. + + * The ast_bridge_join() swap channel parameter now always steals a ref for + the swap channel. This is the only change to the bridge framework's + public API semantics. + + * bridge_channel_internal_join() now requires the bridge_channel->swap + channel to pass in a ref. + + ASTERISK-24649 + Reported by: John Bigelow + + Review: https://reviewboard.asterisk.org/r/4354/ + ........ + + Merged revisions 430975 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430976 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-22 13:14 +0000 [e67ca431ee] Richard Mudgett + + * res_pjsip_outbound_registration.c: Minor code cleanup. + + * Add an allocation failure check and assert in + sip_outbound_registration_response_cb(). + + * Made sip_outbound_registration_state_destroy() handle partially created + state objects from sip_outbound_registration_state_alloc(). + + Review: https://reviewboard.asterisk.org/r/4366/ + ........ + + Merged revisions 430957 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-22 12:10 +0000 [49f405fe4c] Scott Griepentrog + + * stasis transfer: fix a race condition on stasis bridge push + + After a bridge transfer completes where a local replacement + channel is used, a stasis transfer message with the details + of the transfer is sent. This is processed by stasis which + then sets the stasis app name and replaced channel snapshot + on the replacement channel. + + However, since a separate thread was already started to run + stasis on the new replacement channel, a race was on to see + if the message processing would be completed before the app + name was needed, otherwise the channel would be hung up. + + This change moves the calls used to set the stasis app name + and the replace snapshot to the bridge_stasis_push function + callback from the bridge transfer logic, allowing the steps + to be completed earlier and more deterministically, and the + race elimianted. + + NOTE: the swap channel parameter to bridge_stasis_push (and + thus all bridge push callbacks) must always be present when + performing a swap with another channel. + + ASTERISK-24649 #close + Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/4341/ + ........ + + Merged revisions 430939 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-22 08:23 +0000 [7fcc9ce8bc] Gareth Palmer (License 5169) + + * apps/app_voicemail: Trigger MWI notification with MixMonitor m() option + + The MixMonitor m() option allows a recording to be pushed to a specific + voicemail mailbox. If the message is delivered to the mailbox's INBOX, however, + no MWI notification is currently raised. + + This patch corrects the issue by properly calling notify_new_state from the + msg_create_from_file function. This will cause MWI to be triggered if the + message was placed in the mailbox's INBOX. + + ASTERISK-24709 #close + Reported by: Gareth Palmer + patches: + app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169) + ........ + + Merged revisions 430920 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430921 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430922 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-21 15:57 +0000 [38738a7316] Richard Mudgett + + * res_pjsip_outbound_registration.c: Move unref to a better place. + + Move an unconditional unref of client_state so it doesn't look like it + could be used after the last ref has destroyed it. + ........ + + Merged revisions 430902 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-21 07:36 +0000 [5835bf7a7f] Matt Jordan + + * channels/chan_sip: Fix registration leak during reload + + When the SIP registrations were migrated to using ao2 in what was then trunk, + the explicit destruction of the registrations on module reload was removed and + not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the + issue reporter, on ASTERISK-24673 confirmed that the reference in the + registry_list container was being leaked. + + Since the purpose of cleanup_all_regs is to prep a registration for + destruction, this function now calls an ao2_callback function callback with the + OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations. + This cleans up each registration, and also removes it from the registration + container registry_list. + + Review: https://reviewboard.asterisk.org/r/4355/ + + ASTERISK-24640 #close + Reported by: Max Man + + ASTERISK-24673 #close + Reported by: Stefan Engström + Tested by: Stefan Engström + ........ + + Merged revisions 430864 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-21 07:27 +0000 [958a41a884] Matt Jordan + + * AMI: Add documentation for the missing Cdr/CEL events. + + This patch adds AMI event documentation for the Cdr and CEL AMI events. + + Note that while these events do share fields with each other and with other + channel related events, they do not contain all of the fields in a standard + channel snapshot, nor is the description of the fields identical. As such, + the patch opts for documentation for each field, for each event. + + Review: https://reviewboard.asterisk.org/r/4350/ + + ASTERISK-24671 #close + Reported by: Dan Jenkins + ........ + + Merged revisions 430862 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-21 07:12 +0000 [4740ef50f4] Matt Jordan + + * apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values + + The Dial application has some interesting options with the mid-call Macro (M) + and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific + values, the Dial application will take some action upon the channels involved + in the dial operation (such as hanging up a particular party, etc.) The Dial + application ensures that a Stasis message is published in the event that + MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so + that there is a corresponding DialEnd event published in AMI/ARI for the + DialBegin event that preceeded it. + + A bug exists where that same DialEnd event will be published on Stasis even if + the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial + application cares about. This causes two DialEnd events to be published - one + with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all + sorts of wrong. + + This patch fixes the bug by ensuring that we only publish a DialEnd message to + Stasis if the Dial application's mid-call Macro/GoSub returns something that + Dial cares about. + + Review: https://reviewboard.asterisk.org/r/4336 + + ASTERISK-24682 #close + Reported by: Matt Jordan + ........ + + Merged revisions 430842 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-21 07:06 +0000 [228fdb3f4e] Matt Jordan + + * main/rtp_engine: Format NTP timestamps as unsigned longs + + When the RTCP reports are created, the NTP timestamps are stored as strings, + as JSON does not have an integer type long enough to store the value. However, + on 32-bit systems, a signed long may overflow for some portion of the + timestamp. + + This patch corrects the overflow by formatting the timestamps as unsigned + longs. + ........ + + Merged revisions 430840 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-20 11:15 +0000 [804ab70f9d] Ashley Sanders + + * ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied. + + Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk. + + ASTERISK-24560 #close + Reported By: Kinsey Moore + + Review: https://reviewboard.asterisk.org/r/4349/ + ........ + + Merged revisions 430818 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-20 10:59 +0000 [e4738a59eb] Richard Mudgett + + * CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge. + + Calling ast_channel_bridge_peer() cannot be done while holding any channel + locks. The reported issue hit the deadlock in chan_iax2, but an audit of + the ast_channel_bridge_peer() calls found three more locations where the + same deadlock can occur. + + * Made CHANNEL(peer), res_fax, and the SNMP agent not call + ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I + had to rework the logic to not hold the channel lock. + + * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done + for legacy reasons that no longer apply. + + * Removed the iax.conf forcejitterbuffer option. It is now always enabled + when the jitterbuffer option is enabled. If you put a jitter buffer on a + channel it will be on the channel. + + ASTERISK-24600 #close + Reported by: Jeff Collell + + Review: https://reviewboard.asterisk.org/r/4342/ + ........ + + Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-19 20:41 +0000 [14b8e03dad] Ben Klang (License 5876) + + * contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts + + On Debian based systems, the install_prereq tool uses a search command on + Debian that results in selecting both 64-bit and 32-bit packages. Besides the + waste of disk space, this can actually cause aptitude use 100% of memory on a + VM with 1GB of RAM as it tried to work out all of the 32-bit package + dependencies. + + This patch filters out the 32-bit packages on a 64-bit machine, and leaves + 32-bit machines alone. + + ASTERISK-24048 #close + Reported by: Ben Klang + Tested by: Ben Klang, Matt Jordan + patches: + install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876) + ........ + + Merged revisions 430798 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430799 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430800 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-19 20:33 +0000 [112bf1597e] LEI FU (License 6640) + + * app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend + + When using ODBC or IMAP storage, temporary files created on the file system + must be disposed of using the DISPOSE macro. The DELETE macro will map to a + deletion function for the backend storage, but does not clean up any local + files created as a result of the operation. + + When using voicemail with the operator and review options enabled, pressing + 0 to enter the menu, followed by 1 to save the message, followed by any + other DTMF press to delete the message, will result in the temporary file + lingering on the file system. + + This patch properly calls DISPOSE after the DELETE. This causes the local + file to be disposed of. + + ASTERISK-24288 #close + Reported by: LEI FU + patches: + voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640) + ........ + + Merged revisions 430795 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430796 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-19 12:15 +0000 [7dc784ffa9] Mark Michelson + + * Call extension state callbacks at hint creation. + + When a hint gets created, any subsequent device or presence + state changes result in extension status events getting sent + out to interested parties. However, at the time of hint creation, + no such event gets sent out, so watchers of extension state are + potentially left in the dark until the first state change after + hint creation. + + Patch contributed by John Hardin (License #6512) + ........ + + Merged revisions 430776 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-19 07:19 +0000 [e43912f3f3] Joshua Colp + + * res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions. + + The first thing this patch fixes is UAS dialogs. Previously if a transport was + configured on an endpoint and an inbound session was created there was no guarantee + that requests sent on the dialog would use the correct transport and address + information. This has now been fixed so an explicitly configured transport + is taken into account. + + The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed + module attempts to determine what transport a message should go out on and what + addressing information should go into the message itself. In a scenario where + multiple transports exist bound to the same IP address but a different port the + code would incorrectly alter the transport and change the message to the wrong + transport. This change makes the res_pjsip_multihomed module smarter so it will + only change the transport and address information in the message when it is + possible and makes sense. + + ASTERISK-24615 #close + Reported by: David Justl + + Review: https://reviewboard.asterisk.org/r/4331/ + ........ + + Merged revisions 430755 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-16 18:35 +0000 [07e2a48ab1] Kevin Harwell + + * REVERTING res_pjsip: make it unloadable + + Due to the original patch causing memory corruptions the patch is + being removed until the problem can be resolved. + ........ + + Merged revisions 430734 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-16 16:14 +0000 [1111944afb] Mark Michelson + + * Change PJProject version requirement for ca_list_path transport option in CHANGES file. + ........ + + Merged revisions 430716 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-16 16:13 +0000 [831acba826] Mark Michelson + + * Fix problem where a hung channel could occur on a failed blind transfer. + + Different clients react differently to being told that a blind transfer + has failed. Some will simply send a BYE and be done with it. Others will + attempt to reinvite themselves back onto the call. + + In the latter case, we were creating a new channel and then leaving it to + sit forever doing nothing. With this code change, that new channel will + not be created and the dialog with the transferring channel will be cleaned + up properly. + + ASTERISK-24624 #close + Reported by Zane Conkle + + Review: https://reviewboard.asterisk.org/r/4339 + ........ + + Merged revisions 430714 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430715 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-16 15:46 +0000 [023fa0f9e8] cloos (License #5956) + + * Add support for the ca_list_path option for PJSIP transports. + + This allows for a path to be specified that has a collection of CA + certificates in it. + + ASTERISK-24575 #close + Reported by cloos + Patches: + pj-ca-path-trunk.diff uploaded by cloos (License #5956) + + Review: https://reviewboard.asterisk.org/r/4344 + ........ + + Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-15 11:36 +0000 [a8ea2f9287] Richard Mudgett + + * res_fax.c, res_fax_spandsp.c: Remove redundant locking. + + When FAX was developed, apparently the faxregistry.container used to be a + linked list that was converted to an ao2 container. Some of the + replacement ao2 container operations still had explicit lock/unlocks + around them. + + Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the + channel even though the routine did not lock the channel and other code + paths in the routine do not unlock the channel. + + Review: https://reviewboard.asterisk.org/r/4340/ + ........ + + Merged revisions 430687 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-15 11:28 +0000 [9b1c36d3fa] Richard Mudgett + + * res_fax.c, res_fax_spandsp.c: Fix some curlies on the end of function definitions. + ........ + + Merged revisions 430685 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430686 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-15 06:10 +0000 [1e605d950b] Joshua Colp + + * res_pjsip_outbound_registration: Fix race condition when reloading and listing registrations. + + Due to the split of outbound registration state from configuration it is possible during + a reload for a "pjsip show registrations" CLI command to be executed which gets an older + snapshot of the configuration. This configuration may include outbound registrations which + have been removed due to a reload operation occurring at the same time. The code for + printing the outbound registration did not take this into account but now it does. + + AST-1506 #close + + Review: https://reviewboard.asterisk.org/r/4338/ + ........ + + Merged revisions 430664 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-14 20:19 +0000 [f11fb76205] abelbeck (License 5903) + + * configure: If cross-compiling, assume we have working semaphores + + The Asterisk 13 configure.ac checks for HAS_WORKING_SEMAPHORE but does not have + an option for cross-compiling so it fails with an exit. Since we're cross- + compiling, we can't exactly go looking for the header. The semaphore.h header + is relatively common: + * It's part of the POSIX standard + * It's part of GNU C Library + As such, we assume that it will be present when cross-compiling. + + As such, this patch defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling + is detected. + + If you're cross-compiling to a platform that doesn't support this, then make + sure you re-define this to 0. + + ASTERISK-24663 #close + Reported by: abelbeck + patches: + asterisk-13-anonymous-semaphores.patch uploaded by abelbeck (License 5903) + ........ + + Merged revisions 430646 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-14 17:15 +0000 [49542a794b] Corey Farrell (license 5909) + + * res_pjsip: make it unloadable + + The res_pjsip module was previously unloadable. With this patch it can now + be unloaded. + + This patch is based off the original patch on the issue (listed below) by Corey + Farrell with a few modifications. Namely, removed a few changes not required to + make the module unloadable and also fixed a bug that would cause asterisk to + crash on unloading. + + This patch is the first step (should hopefully be followed by another/others at + some point) in allowing res_pjsip and the modules that depend on it to be + unloadable. At this time, res_pjsip and some of the modules that depend on + res_pjsip cannot be unloaded without causing problems of some sort. + + The goal of this patch is to get res_pjsip and only res_pjsip to be able to + unload successfully and/or shutdown without incident (crashes, leaks, etc...). + Other dependent modules may still cause problems on unload. + + Basically made sure, with the patch applied, that res_pjsip (with no other + dependent modules loaded) could be succesfully unloaded and Asterisk could + shutdown without any leaks or crashes that pertained directly to res_pjsip. + + ASTERISK-24485 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4311/ + patches: + pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) + ........ + + Merged revisions 430628 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-14 14:39 +0000 [67234b3ee2] Mark Michelson + + * Prevent slow graceful shutdown when outbound publications never started. + + The code was missing the case for explicitly destroying an outbound publication + when Asterisk had never actually published anything. The result was that Asterisk + would hang for a while on a graceful shutdown. + + With this change, the case is taken into account, and on a graceful shutdown, these + publications are destroyed without the need to actually send a PUBLISH request. + + ASTERISK-24655 #close + Reported by Kevin Harwell + + Review: https://reviewboard.asterisk.org/r/4325 + ........ + + Merged revisions 430608 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430609 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-14 09:40 +0000 [3eec8e4c44] Diederik de Groot (License 6600) + + * build_tools/mkpkgconfig: Fix Cflags concatenation error in asterisk.pc + + The mkpkgconfig script incorrectly concatenates Cflags options together. As an + example, the following: + Cflags: -I/usr/include/libxml2 -g3 + + Is instead generated as: + Cflags: -I/usr/include/libxml2-g3 + + This patch corrects the generation of Cflags in mkpkgconfig such that the + Cflags options are output correctly. + + Review: https://reviewboard.asterisk.org/r/3707/ + + ASTERISK-23991 #close + Reported by: Diederik de Groot + patches: + fix_mkpkgconfig.diff uploaded by Diederik de Groot (License 6600) + ........ + + Merged revisions 430589 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430590 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-13 12:17 +0000 [1780de95e4] Richard Mudgett + + * app_macro: Don't restore the calling location on a channel redirect. + + v11: If a channel redirect to a macro exten of a macro that is active + happens, the redirect location doesn't get executed. Instead the original + macro location is restored and gets reexecuted. + + v13: An additional effect happens if a parked call times out to an + extension in the macro that parked the call then the macro is reexecuted + instead of the expected park return location. + + * Made not restore the macro calling location on an + AST_SOFTHANGUP_ASYNCGOTO. + + * Increased the locked channel range when setting up the macro execution + environment to cover things that should be done while the channel is + locked. + + * Removed unnecessary NULL tests before calling ast_free() in + _macro_exec(). + + ASTERISK-23850 #close + Reported by: Andrew Nagy + + Review: https://reviewboard.asterisk.org/r/4292/ + ........ + + Merged revisions 430564 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430565 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430567 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-13 06:09 +0000 [0e631a541d] Joshua Colp + + * chan_pjsip: Add configure check for 'pjsip_get_dest_info' function. + + The 'pjsip_get_dest_info' function is used to determine if the signaling transport + of the dialog is secure or not. This function was added in PJSIP 2.3 and does not + exist in earlier versions. + + This configure check allows Asterisk to build and run with older versions at the + loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of + this argument will require upgrading to PJSIP 2.3. + + ASTERISK-24665 #close + Reported by: Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4329/ + ........ + + Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-12 13:13 +0000 [4dd6b6ff59] Richard Mudgett + + * AMI: Revert non-backwards compatible changes from earlier commit. + + * Reverted the change to astman_send_listack() to not use the listflag + parameter and always set the value to "Start" so the start capitalization + is consistent. Unfortunately changing the case of a returned value is not + a backward compatible change so for now FAXSessions is going to have to + remain inconsistent with all of the other AMI list actions. + + * Reverted the minor protocol error fix in action_getconfig() when no + requested categories are found. Each line needs to be formatted as + "Header: text". + + Caught by the testsuite. + + ASTERISK-24049 + ........ + + Merged revisions 430528 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-12 12:28 +0000 [aa7e06f797] Niklas Larsson (License 5068) + + * configs/samples/features.conf.sample: Document attended transfer DTMF options + + The sample config was missing the configuration options for DTMF attended + transfer completion scenarios. The configuration options 'atxferabort', + 'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the + appropriate configuration file. + + ASTERISK-24678 #close + Reported by: Niklas Larsson + patches: + features.conf.sample.diff uploaded by Niklas Larsson (License 5068) + ........ + + Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-12 12:09 +0000 [c7ea108e02] Richard Mudgett + + * Revert -r430452 It needs to be redone for the next major AMI version change instead. + + ASTERISK-24049 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-12 12:01 +0000 [9065488ddd] Michael L. Young (license 5026) + + * main/syslog: Allow dynamic logs, such as security events, to log to the syslog + + The security event log uses a dynamic log level (SECURITY) that is registered + with the Asterisk logging core. Unfortunately, the syslog would ignore log + statements that had a dynamic log level associated with them. Because the + syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic + log entries sent to the syslog as logs with a level of NOTICE. + + ASTERISK-20744 #close + Reported by: Michael Keuter + Tested by: Michael L. Young, Jacek Konieczny + patches: + asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026) + ........ + + Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430507 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-12 09:18 +0000 [b38acbce6e] Kristian Hogh (License 6639) + + * funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyed + + When the channel datastore associated with the usage of CURLOPT on a specific + channel is freed, the underlying structure holding the list of options is not + disposed of. This patch properly frees the structure in the datastore .destroy + callback. + + ASTERISK-24672 #close + Reported by: Kristian Hogh + patches: + func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639) + ........ + + Merged revisions 430487 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430488 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430489 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-09 16:09 +0000 [fba836cc02] Scott Griepentrog + + * sip_to_pjsip: improve ability to parse input files + + General improvements to SIP to PJSIP conversion utility: + + 1) track default section of input file to allow parsing + an include file that doesn't specify a [section] + + 2) informatively handle case of assignment without [section] + + 3) correctly handle getting sections from included files + - [section]'s are inherited by included file + + 4) provide null string as default transport bind ip + + 5) gracefully handle missing portions of registration string + + 6) denote steps of operation during conversion and confirm + top level files as a convenience + + ASTERISK-24474 #close + Review: https://reviewboard.asterisk.org/r/4280/ + Reported by: John Kiniston + ........ + + Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-09 15:45 +0000 [5b30938394] Scott Griepentrog + + * app_bridge: return to the next dialplan priority + + When app_bridge grabs a channel and puts it into + a bridge, the channel should then continue where + it left off in the dialplan after the bridge has + ended. Although it stores the current dialplan + location as an after bridge goto on the channel, + it was executing the same priority again instead + of going to the next priority. By swapping the + "specific" version of bridge_set_after_goto with + bridge_set_after_go_on, the next priority in the + dialplan is executed instead. + + ASTERISK-24637 #close + Review: https://reviewboard.asterisk.org/r/4322/ + Reported by: John Bigelow + ........ + + Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-09 12:53 +0000 [ef34a05f21] Richard Mudgett + + * AMI: Remove no longer used parameter from astman_send_listack(). + + Follow-up issue to -r430435 from reviewboard review. + + ASTERISK-24049 + Review: https://reviewboard.asterisk.org/r/4315/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-09 12:16 +0000 [52a7cdb101] Richard Mudgett + + * AMI: Make AMI actions that generate event lists consistent. + + * Made the following AMI actions use list API calls for consistency: + Agents + BridgeInfo + BridgeList + BridgeTechnologyList + ConfbridgeLIst + ConfbridgeLIstRooms + CoreShowChannels + DAHDIShowChannels + DBGet + DeviceStateList + ExtensionStateList + FAXSessions + Hangup + IAXpeerlist + IAXpeers + IAXregistry + MeetmeList + MeetmeListRooms + MWIGet + ParkedCalls + Parkinglots + PJSIPShowEndpoint + PJSIPShowEndpoints + PJSIPShowRegistrationsInbound + PJSIPShowRegistrationsOutbound + PJSIPShowResourceLists + PJSIPShowSubscriptionsInbound + PJSIPShowSubscriptionsOutbound + PresenceStateList + PRIShowSpans + QueueStatus + QueueSummary + ShowDialPlan + SIPpeers + SIPpeerstatus + SIPshowregistry + SKINNYdevices + SKINNYlines + Status + VoicemailUsersList + + * Incremented the AMI version to 2.7.0. + + * Changed astman_send_listack() to not use the listflag parameter and + always set the value to "Start" so the start capitalization is consistent. + i.e., The FAXSessions used "Start" while the rest of the system used + "start". The corresponding complete event always used "Complete". + + * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the + AMI ActionID for all of its list events. + + * Fixed off-nominal AMI protocol error in manager_bridge_info(), + manager_parking_status_single_lot(), and + manager_parking_status_all_lots(). Use of astman_send_error() after + responding to the original AMI action request violates the action response + pattern by sending two responses. + + * Fixed minor protocol error in action_getconfig() when no requested + categories are found. Each line needs to be formatted as "Header: text". + + * Fixed off-nominal memory leak in manager_build_parked_call_string(). + + * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). + + ASTERISK-24049 #close + Reported by: Jonathan Rose + + Review: https://reviewboard.asterisk.org/r/4315/ + ........ + + Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-09 08:53 +0000 [77ee23210d] Kinsey Moore + + * res_fax: Add T.38 negotiation timeout option + + This change makes the T.38 negotiation timeout configurable via + 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously + hard coded to be 5000 milliseconds. + + This change also handles T.38 switch failures by aborting the fax since + in the case where this can happen, both sides have agreed to switch to + T.38 and Asterisk is unable to do so. + + Review: https://reviewboard.asterisk.org/r/4320/ + ........ + + Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-08 15:41 +0000 [8786fe13a4] gtjoseph + + * res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown + + If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't + survive. If you do a 'core (shutdown|restart) now' or asterisk terminates for + some reason, they do. Here's why... + + When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to + subscribers for each subscription. This not only tells the subscribers that the + dialog/state machine is done, it also frees the last reference to the + subscription tree which causes the persistent subscription to get deleted from + astdb. When asterisk restarts, nothing's left. Just preventing the delete from + astdb doesn't work because we already told the subscriber to terminate the + dialog so we can't restart it even if it was still in astdb. Everything works + OK if asterisk terminates unexpectedly because we never send the 'terminated' + message so on restart, the subscription is still in astdb and the subscriber is + none the wiser. + + This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for + persistent connections. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4318/ + ........ + + Merged revisions 430397 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430398 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-08 15:38 +0000 [c55f86c69d] gtjoseph + + * res_pjsip_outbound_registration: Fix reference leak. + + Every time a registration started, sip_outbound_registration_response_cb bumps + the ref count on client_state then pushes a handle_registration_response task. + handle_registration_response never unreffed it though. So every time a + registration goes out, the ref count goes up by one. + + This patch adds the unreffs to handle_registration_response. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4303/ + ........ + + Merged revisions 430395 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430396 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-08 11:51 +0000 [030facce94] gtjoseph + + * res_pjsip_outbound_registration: Fix several reload issues + + There are 2 issues with reloading registrations... + + 1. The 'can_reuse_registration' test wasn't considering the intervals or + expiration in its determination of whether a registration changed or not so if + you changed any of the intervals or the expiration and reloaded, the object + would get reloaded but the actual timers wouldn't change. + can_reuse_registration now does a sorcery diff on the old and new objects + instead of discretely testing certain fields. Now if you change expiration for + instance, and reload, the timer is updated and re-registration will occur on the + new value. + + 2. If you mung up your password on an outbound registration you get a permanent + failure. If you fix the password (on the outbound_auth object) and reload, + nothing tells outbound_registration to try again because the registration itself + didn't change. This patch adds an observer on the "auth" object type and if any + auth changes, existing registration states are searched and those in a + REJECTED_PERMANENT state are retried. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4304/ + ........ + + Merged revisions 430373 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430374 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 15:26 +0000 [f8c4909eb7] Kinsey Moore + + * ARI: Allow usage of ASYNCGOTO with Stasis() + + When the AMI Redirect action is used with a channel bridged inside + Stasis() and not running a pbx, the channel is hung up instead of + proceeding to the desired location in dialplan. This change allows + such channels to be Redirected properly by detecting the operation + used by Redirect (ASYNCGOTO) and using the code already established + for functionality of the ARI channel continue operation. + + ASTERISK-24591 #close + Review: https://reviewboard.asterisk.org/r/4271/ + ........ + + Merged revisions 430355 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 12:54 +0000 [7f836c1c15] Mark Michelson + + * Add the ability to continue and originate using priority labels. + + With this patch, the following two ARI commands + + POST /channels + POST /channels/{id}/continue + + Accept a new parameter, label, that can be used to continue to or originate + to a priority label in the dialplan. + + Because this is adding a new parameter to ARI commands, the API version of + ARI has been bumped from 1.6.0 to 1.7.0. + + This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks! + + ASTERISK-24412 #close + Reported by Nir Simionovich + + Review: https://reviewboard.asterisk.org/r/4285 + ........ + + Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 12:17 +0000 [e83853eebc] gtjoseph + + * res_pjsip_exten_state: Change 'does not exist' warning to notice + + The 'new_subscribe: Extension <> does not exist or has no associated hint' + is a config issue and doesn't need to clutter up logs with warnings. + Changed to notice. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4307/ + ........ + + Merged revisions 430319 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 12:15 +0000 [8cde7443c2] gtjoseph + + * res_pjsip_mwi: Change "MWI Subscription failed" message from warning to notice + + The "MWI Subscription failed" message means the client is trying to subscribe + to a mailbox that doesn't exist. There's no need to clutter up logs with + warnings for a client misconfiguration so I changed it to a notice. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4306/ + ........ + + Merged revisions 430317 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 11:54 +0000 [685f7ef924] gtjoseph + + * func_config: Add ability to retrieve specific occurrence of a variable + + I guess nobody uses templates with AST_CONFIG because today if you have a + context that inherits from a template and you call AST_CONFIG on the context, + you'll get the value from the template even if you've overridden it in the + context. This is because AST_CONFIG only gets the first occurrence which is + always from the template. + + This patch adds an optional 'index' parameter to AST_CONFIG which lets you + specify the exact occurrence to retrieve, or '-1' to retrieve the last. + The default behavior is the current behavior. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4313/ + ........ + + Merged revisions 430315 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 11:45 +0000 [464647d8f8] Mark Michelson + + * Fix ability to perform a remote attended transfer with PJSIP. + + This fix has two parts: + + * Corrected an error message to properly state that external_replaces is an extension. The + error message also prints what dialplan context the external_replaces extension was being + looked for in. + * Corrected the printing of the Replaces: header in an INVITE request. We were duplicating + "Replaces: " in the header. + + ASTERISK-24376 #close + Reported by Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4296 + ........ + + Merged revisions 430313 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-07 10:56 +0000 [56de48107f] gtjoseph + + * config: Add option to NOT preserve effective context when changing a template + + Let's say you have a template T with variable VAR1 = ON and you have a + context C(T) that doesn't specify VAR1. If you read C, the effective value + of VAR1 is ON. Now you change T VAR1 to OFF and call + ast_config_text_file_save. The current behavior is that the file gets + re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this + is a bug. It's preserving the effective state of C even though I didn't + specify C/VAR1 in th first place. I believe the behavior should be that if + I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should + continue to follow the inherited state. Now, if I DID explicitly specify + C/VAR1, the it should be preserved even if the template changes. + + Even though I think the existing behavior is a bug, it's been that way forever + so I'm not changing it. Instead, I've created ast_config_text_file_save2() + that takes a bitmask of flags, one of which is to preserve the effective context + (the current behavior). The original ast_config_text_file_save calls *2 with + the preserve flag. If you want the new behavior, call *2 directly without a + flag. + + I've also updated Manager UpdateConfig with a new parameter + 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior + with UpdateConfig, set 'PreserveEffectiveContext: no'. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4297/ + ........ + + Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-06 21:01 +0000 [0c5234f12a] Kinsey Moore + + * Fix dev-mode build on recent gcc + ........ + + Merged revisions 430274 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430275 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-06 16:46 +0000 [220df246d9] Matt Jordan + + * Blocked revisions 430252 + + ........ + contrib/ast-db-manage: Correct down_revision path for user_eq_phone + + When the user_eq_phone patch was backported to 13, it referenced the downward + revision that the PJSIP optimistic encryption option also references. This + creates a multi-path upgrade Exception when generating the SQL files. + + This patch corrects this in the 13 branch. Note that trunk, which already + contained both of these features, is unaffected by this problem. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-06 11:53 +0000 [8b5bde3e5a] gtjoseph + + * res_pjsip_mwi: Change warning to notice + + When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi, + if a contact hasn't registered yet, res_pjsip_mwi spits out a warning. + This is a perfectly normal situation though and doesn't require something + as serious as a warning. It's also self correcting. The device will start + getting mwi as soon as it registers. + + This patch changes the warning to a notice. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4314/ + ........ + + Merged revisions 430227 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-06 11:49 +0000 [5f60ebc004] gtjoseph + + * bridge_native_rtp: Change local/remote message from debug/2 to verb/4 + + Change the "Locally bridged"/"Remotely bridged" messages from dbg/2 to verb/4. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4300/ + ........ + + Merged revisions 430225 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430226 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-06 11:43 +0000 [fb3c8e3424] gtjoseph + + * outbound_registration: Add 'pjsip send register' and update 'send unregister' + + The current behavior of 'pjsip send unregister' is to send the unregister + (REGISTER with 0 exp) but let the next scheduled register proceed normally. + I don't think that's a good idea. If you unregister, it should stay + unregistered until you decide to start registrations again. So this patch + just adds a cancel_registration call to the current unregister_task to + cancel the timer. + + Of course, now you need a way to start registration again so I've added + a 'pjsip send register' command that unregisters and cancels any existing + registration (the same as send unregister), then sends an immediate + registration and starts the timer back up again. + + Both changes also ripple to AMI. There's a new PJSIPRegister command. + + There's no harm in calling either command repeatedly. They don't care + about the actual state. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4301/ + ........ + + Merged revisions 430223 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-06 11:29 +0000 [7dc0c88fc6] gtjoseph + + * pjsip cli: Fix sorting of contacts for 'pjsip list contacts' + + For some reason I was using a hash container instead of a list to gather the + contacts for 'pjsip list/show contacts' so even though I had a sort function, + the output wasn't sorted. This patch just changes the hash container to a + list container and the contacts now appear sorted in the CLI. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4305/ + ........ + + Merged revisions 430221 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-05 16:50 +0000 [0b8fbf9238] Scott Griepentrog + + * bridge: avoid leaking channel during blond transfer pt2 + + A blond transfer to a failed destination, when followed + by a recall attempt, lead to a leak of the reference to + the destination channel. In addition to correcting the + regression on the previous attempt (r429826) this fixes + the leak and two additional reference leaks on failures + of bridge_import. + + ASTERISK-24513 #close + Review: https://reviewboard.asterisk.org/r/4302/ + ........ + + Merged revisions 430199 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 430200 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-05 11:57 +0000 [e0bd2ca104] Joshua Colp + + * pjsip: Document addition of 'PJSIP_AOR' and 'PJSIP_CONTACT' in CHANGES file. + ........ + + Merged revisions 430181 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430182 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2015-01-05 11:53 +0000 [f7cf988a82] Joshua Colp + + * pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions. + + The PJSIP_AOR dialplan function allows inspection of configured AORs including + what contacts are currently bound to them. + + The PJSIP_CONTACT dialplan function allows inspection of contacts in existence. + These can include both externally added (by way of registration) or permanent + ones. + + ASTERISK-24341 + Reported by: xrobau + + Review: https://reviewboard.asterisk.org/r/4308/ + ........ + + Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-31 12:54 +0000 [8d059c3808] Scott Griepentrog + + * rtp_engine: keep payload types in correct range + + In r428708 additional codecs were added including + a payload type of 128 which is outside of nominal + range of 0-127. This change moves changes 128 to + 96 to avoid causing a pjsip assertion when making + a call to an endpoint configured with allow=all. + + ASTERISK-24367 #close + Review: https://reviewboard.asterisk.org/r/4286/ + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430164 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-29 07:14 +0000 [cb6a737359] Kinsey Moore + + * PJSIP: Update transport method documentation + + This updates the documentation for the 'method' configuration option to + be more verbose about the behaviors of values 'unspecified' and + 'default'. They do exactly the same thing which is to select the + default as defined by PJSIP which is currently TLSv1. + + Review: https://reviewboard.asterisk.org/r/4264/ + ........ + + Merged revisions 430145 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-24 15:28 +0000 [91becf952a] Kevin Harwell + + * app_queue: Update sample conf documenation + + Updated the queues.conf.sample file to explicitly state which channel queue + variables are propagated to. + + ASTERISK-24267 + Reported by: Mitch Claborn + ........ + + Merged revisions 430126 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430127 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430128 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-24 10:59 +0000 [3a73c6c90e] Matt Jordan + + * main/pbx.c: Fix double lock of contexts lock introduced by r429967 + + We only need to hold the context_merge_lock once. Locking it twice will make + many other parts of Asterisk very sad. + + ASTERISK-24641 #close + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-23 17:19 +0000 [7ea4156a5e] gtjoseph + + * pjsip_options: Fix continued qualifies after endpoint/aor deletion + + If you remove an endpoint/aor from pjsip.conf then do a core reload, + qualifies will continue even though the object are gone. This happens + because nothing clears out the qualify tasks. + + This patch unschedules all existing qualify tasks before scheduling + new ones on reload. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4290/ + ........ + + Merged revisions 430064 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-23 17:16 +0000 [62d1dba271] gtjoseph + + * test_astobj2: Fix warning for missing trailing slash in category + + This patch adds a trailing slash to the category for this test. + No more warning. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4295/ + ........ + + Merged revisions 430059 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-22 15:20 +0000 [1c0604e905] Richard Mudgett + + * DTMF atxfer: Setup recall channels as if the transferee initiated the call. + + After the initial DTMF atxfer call attempt to the transfer target fails to + answer during a blonde transfer, the recall callback channels do not get + setup with information from the initial transferrer channel. As a result, + the recall callback to the transferrer does not have callid, channel + variables, datastores, accountcode, peeraccount, COLP, and CLID setup. A + similar situation happens with the recall callback to the transfer target + but it is less visible. The recall callback to the transfer target does + not have callid, channel variables, datastores, accountcode, peeraccount, + and COLP setup. + + * Added missing information to the recall callback channels before + initiating the call. callid, channel variables, datastores, accountcode, + peeraccount, COLP, and CLID + + * Set callid of the transferrer channel on the DTMF atxfer controller + thread attended_transfer_monitor_thread(). + + * Added missing channel unlocks and props unref to off nominal paths in + attended_transfer_properties_alloc(). + + ASTERISK-23841 #close + Reported by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4259/ + ........ + + Merged revisions 430034 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-22 14:25 +0000 [7d954f4cb1] Richard Mudgett + + * Fix compilation since the patch for ASTERISK-24363 went in. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-22 14:08 +0000 [bbd9ff122e] Richard Mudgett + + * queue_log: Post QUEUESTART entry when Asterisk fully boots. + + The QUEUESTART log entry has historically acted like a fully booted event + for the queue_log file. When the QUEUESTART entry was posted to the log + was broken by the change made by ASTERISK-15863. + + * Made post the QUEUESTART queue_log entry when Asterisk fully boots. + This restores the intent of that log entry and happens after realtime has + had a chance to load. + + AST-1444 #close + Reported by: Denis Martinez + + Review: https://reviewboard.asterisk.org/r/4282/ + ........ + + Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 430010 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-22 09:40 +0000 [264a50c52a] Karsten Wemheuer (License 5930) + + * chan_sip: Send CANCEL via original INVITE destination even after UPDATE request + + Given the following scenario: + * Three SIP phones (A, B, C), all communicating via a proxy with Asterisk + * A call is established between A and B. B performs a SIP attended transfer of + A to C. B sets the call on hold (A is hearing MOH) and dials the extension of + C. While phone C is ringing, B transfers the call (that is, what we typically + call a 'blond transfer'). + * When the transfer completes, A hears the ringing of phone C, while B is idle. + + In the SIP messaging for the above scenario, a REFER request is sent to + transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an + UPDATE request to phone C to update party information. This update is sent + directly to phone C, not through the intervening proxy. This has the unfortunate + side effect of providing route information, which is then set on the sip_pvt + structure for C. If someone (e.g. B) is trying to get the call back (through a + directed pickup), Asterisk will send a CANCEL request to C. However, since we + have now updated the route set, the CANCEL request will be sent directly to C + and not through the proxy. The phone ignores this CANCEL according to RFC3261 + (Section 9.1). + + This patch updates reqprep such that the route is not updated if an UPDATE + request is being sent while the INVITE state is INV_PROCEEDING or + INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent + to the correct location. + + Review: https://reviewboard.asterisk.org/r/4279 + + ASTERISK-24628 #close + Reported by: Karsten Wemheuer + patches: + issue.patch uploaded by Karsten Wemheuer (License 5930) + ........ + + Merged revisions 429982 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429983 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-22 08:33 +0000 [0c38276d6e] Gareth Palmer (License 5169) + + * presencestate: Allow channel drivers to provide presence state information + + This patch adds the ability for channel drivers to supply presence information + in a similar manner to device state. The patch does not provide any channel + driver implementations, but it does provide the core infrastructure necessary + for channel drivers to provide such information. + + The core handles multiple providers of presence state information. Ordering + of presence state is as follows: + INVALID < NOT_SET < AVAILABLE < UNAVAILABLE < CHAT < AWAY < XA < DND + + Each provider can trump the previous if it provides a presence state that + supercedes a previous one. + + Review: https://reviewboard.asterisk.org/r/4050 + + ASTERISK-24363 #close + Reported by: Gareth Palmer + patches: + chan_presencestate-428146.patch uploaded by Gareth Palmer (License 5169) + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-22 06:16 +0000 [2afeadcc84] Matt Jordan + + * app_confbridge: Fix build error caused by XML validation errors + + Summaries can't contain XML nodes, as they are defined to contain only text + data. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-21 20:35 +0000 [b79a4a464f] Gareth Palmer (License 5169) + + * app_confbridge: Add the ability to pass options/command to MixMonitor + + This patch adds the ability to pass options and a command to MixMontor when + recording a conference using ConfBridge. + + New options are - + + * record_options: Options to MixMontor, eg: m(), W() etc. + * record_command: The command to execute when recording is over. + * record_file_timestamp: Append the start time to the file name. + + These options can also be used with the CONFBRIDGE function, e.g., + Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME})) + + Review: https://reviewboard.asterisk.org/r/4023 + + ASTERISK-24351 #close + Reported by: Gareth Palmer + patches: + record_command-428838.patch uploaded by Gareth Palmer (License 5169) + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-21 18:17 +0000 [b137a92aef] gtjoseph + + * res_pjsip_phoneprovi_provider: Fix reload + + Reloading wasn't working correctly because on a reload, the sorcery apply + handler was never being called for unchanged users. So, instead of using + an apply handler, I'm now iterating over all users. Works much more reliably. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4288/ + ........ + + Merged revisions 429914 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-20 14:57 +0000 [ba403e83bd] Joshua Colp + + * acl: Fix reloading of configuration if configuration file does not exist at startup. + + The named ACL code incorrectly destroyed the config options information if loading + of the configuration file failed at startup. This would result in reloading + also failing even if a valid configuration file was put in place. + + ASTERISK-23733 #close + Reported by: Richard Kenner + ........ + + Merged revisions 429893 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429894 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-19 14:56 +0000 [54bd1c9683] Richard Mudgett + + * res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write(). + + This won't fix the reported issue but it is an incorrect use of sizeof. + + ASTERISK-24566 + Reported by: Badalian Vyacheslav + ........ + + Merged revisions 429867 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429868 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-19 11:34 +0000 [b508b3474e] Richard Mudgett + + * chan_dahdi: Don't ignore setvar when using configuration section scheme. + + When the configuration section scheme of chan_dahdi.conf is used (keyword + dahdichan instead of channel) all setvar= options are completely ignored. + No variable defined this way appears in the created DAHDI channels. + + * Move the clearing of setvar values to after the deferred processing of + dahdichan. + + AST-1378 #close + Reported by: Guenther Kelleter + Patch by: Guenther Kelleter + ........ + + Merged revisions 429825 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429829 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-19 11:27 +0000 [07d1012383] Scott Griepentrog + + * bridge: avoid leaking channel during blond transfer + + After a blond transfer (start attended and hang up) + to a destination that also hangs up without answer, + the Local;1 channel was leaked and would show up on + core show channels. This was happening because the + attended state blond_nonfinal_enter() resetting the + props->transfer_target to null while releasing it's + own reference, which would later prevent props from + releasing another reference during destruction. The + change made here is simply to not assign the target + to NULL. + + ASTERISK-24513 #close + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4262/ + ........ + + Merged revisions 429826 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429827 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-18 16:40 +0000 [2cbfafa8c1] Richard Mudgett + + * chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5. + + ASTERISK-24337 #close + Reported by: Rusty Newton + ........ + + Merged revisions 429804 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429805 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429806 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-18 14:09 +0000 [eacbb4ceb5] Richard Mudgett + + * chan_dahdi: Populate CALLERID(ani2) for incoming calls in featdmf signaling mode. + + For the featdmf signaling mode the incoming MF Caller-ID information is + formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}# + + Rather than discarding the ani2 digits, populate the CALLERID(ani2) value + with what is received instead. + + AST-1368 #close + Reported by: Denis Martinez + Patches: + extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett + ........ + + Merged revisions 429783 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429784 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-18 09:55 +0000 [546a54574f] Kevin Harwell + + * res_pjsip_sdp_rtp: wrong bridge chosen when the DTMF mode is not compatible + + A native rtp bridge was being chosen (it shouldn't have been) when using two + pjsip channels with incompatible DTMF modes. This patch sets the rtp instance + property, AST_RTP_PROPERTY_DTMF, for the appropriate DTMF mode(s) for pjsip. + It was not being set before, meaning all DTMF modes for pjsip were being treated + as compatible, thus native bridging would be chosen as the bridge type when it + shouldn't have been. + + ASTERISK-24459 #close + Reported by: Yaniv Simhi + Review: https://reviewboard.asterisk.org/r/4265/ + ........ + + Merged revisions 429763 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-18 09:40 +0000 [2f3e5b494a] Mark Michelson + + * Prevent potential infinite outbound authentication loops in registration. + + Prior to this patch, Asterisk would always respond to 401 responses to + registration attempts by trying to provide a registration with authentication + credentials. Even if subsequent attempts were rejected with 401 responses, + Asterisk would continue this behavior. If authentication credentials were + incorrect, this could continue forever. + + With this patch, we keep track of whether we have attempted authentication + on an outbound registration attempt. If we already have, we don not try + again until the next attempt. This prevents the infinite loop scenario. + + Review: https://reviewboard.asterisk.org/r/4273 + ........ + + Merged revisions 429761 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-18 09:18 +0000 [2b1f2b5c1f] Mark Michelson + + * Prevent possible race condition on dual redirect of channels in the same bridge. + + The AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent bridges from + prematurely acting on orphaned channels in bridges. The problem with the AMI + redirect action was that it was setting this flag on channels based on the presence + of a PBX, not whether the channel was in a bridge. Whether a channel has a PBX + is irrelevant, so the condition has been altered to check if the channel is in a + bridge. + + ASTERISK-24536 #close + Reported by Niklas Larsson + + Review: https://reviewboard.asterisk.org/r/4268 + ........ + + Merged revisions 429741 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-18 08:50 +0000 [cc1405bd38] Mark Michelson + + * Ensure the correct value is returned for CHANNEL(pjsip, secure) + + Prior to this patch, we were using the PJSIP dialog's secure flag + to determine if a secure transport was being used. Unfortunately, + the dialog's secure flag was only set if a SIPS URI were in use, + as required by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested + in is not dialog security, but transport security. This code change + switches to a model where we use the dialog's target URI to determine + what transport would be used to communicate, and then check if that + transport is secure. + + AST-1450 #close + Reported by John Bigelow + + Review: https://reviewboard.asterisk.org/r/4277 + ........ + + Merged revisions 429739 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429740 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-17 18:11 +0000 [18b5a336ef] gtjoseph + + * res_pjsip_config_wizard: fix unload SEGV + + If certain pjsip modules aren't loaded, the wizard causes a SEGV + when it unloads. Added a check for the presense of the object + type wizard before trying to clean it up. + + Tested-by: George Joseph + ........ + + Merged revisions 429719 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-17 17:06 +0000 [c4360796f7] gtjoseph + + * res_pjsip_config_wizard: Change FILEUNCHANGED config_load2 flag determination + + The module now applies the FILEUNCHANGED flag when both reloaded is + specified AND there's no last_config for the object type. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4276/ + ........ + + Merged revisions 429699 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-17 04:23 +0000 [8b6ecc449c] Walter Doekes + + * Fix printf problems with high ascii characters after r413586 (1.8). + + In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. + Those fixes included things like: + + -out += sprintf(out, "%%%02X", (unsigned char) *ptr); + +out += sprintf(out, "%%%02X", (unsigned) *ptr); + + That works for low ascii characters, but for the high range that yields + e.g. FFFFFFC3 when C3 is expected. + + This changeset: + - fixes those casts to use the 'hh' unsigned char modifier instead + - consistently uses %02x instead of %2.2x (or other non-standard usage) + - adds a few 'h' modifiers in various places + - fixes a 'replcaes' typo + - dev/urandon typo (in 13+ patch) + + Review: https://reviewboard.asterisk.org/r/4263/ + + ASTERISK-24619 #close + Reported by: Stefan27 (on IRC) + ........ + + Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429675 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-16 11:53 +0000 [c4cc668ba9] gtjoseph + + * res_pjsip_config_wizard: fix test breakage + + Fix test breakage caused by not checking for res_pjsip before + calling ast_sip_get_sorcery. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4269/ + ........ + + Merged revisions 429653 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429654 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-16 10:39 +0000 [58095d2486] Andreas Steinmetz (license 6523) + + * chan_sip: Allow T.38 switch-over when SRTP is in use. + + Previously when SRTP was enabled on a channel it was not possible + to switch to T.38 as no crypto attributes would be present. + + This change makes it so it is now possible. If a T.38 re-invite + comes in SRTP is terminated since in practice you can't encrypt + a UDPTL stream. Now... if we were doing T.38 over RTP (which + does exist) then we'd have a chance but almost nobody does that so + here we are. + + ASTERISK-24449 #close + Reported by: Andreas Steinmetz + patches: + udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523) + ........ + + Merged revisions 429632 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429633 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429634 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-16 09:44 +0000 [b5182a6795] Joshua Colp + + * res_pjsip_t38: Fix T.38 failure when peer reinvites immediately. + + If a remote endpoint reinvites to T.38 immediately the state machine + will go into a peer reinvite state. If a T.38 capable application + (such as ReceiveFax) queries it will receive this state. Normally + the application will then indicate so that the channel driver will + queue up the T.38 offer previously received. Once it receives this + offer the application will act normally and negotiate. + + The res_pjsip_t38 module incorrectly partially squashed this indication. + This would cause the application to think the request had failed when + in reality it had actually worked. + + This change makes it so that no T.38 control frames (or indications) + are squashed. + ........ + + Merged revisions 429612 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-15 11:08 +0000 [39b54a21dc] gtjoseph + + * res_pjsip_config_wizard: Allow streamlined config of common pjsip scenarios + + res_pjsip_config_wizard + ------------------ + * This is a new module that adds streamlined configuration capability for + chan_pjsip. It's targetted at users who have lots of basic configuration + scenarios like 'phone' or 'agent' or 'trunk'. Additional information + can be found in the sample configuration file at + config/samples/pjsip_wizard.conf.sample. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4190/ + ........ + + Merged revisions 429592 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-15 09:48 +0000 [53e5b377a0] Mark Michelson + + * Activate persistent subscriptions when they are recreated. + + Prior to this change, recreating persistent subscriptions would + create the subscription but would not activate it. This led to subscriptions + being listed in the "NULL" state by diagnostics and not sending NOTIFYs + when expected. + + Review: https://reviewboard.asterisk.org/r/4261 + ........ + + Merged revisions 429571 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 17:57 +0000 [6472568bc6] gtjoseph + + * loader: Move definition of ast_module_reload from _private.h to module.h + + No functionality change. Just move the definition of ast_module_reload + from _private.h to module.h so it can be public. + + Also removed the include of _private.h from manager.c since ast_module_load + was the only reason for including it. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4251/ + ........ + + Merged revisions 429542 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 17:49 +0000 [308c1b41dd] Richard Mudgett + + * DEBUG_THREADS: Fix regression and lock tracking initialization problems. + + This patch started with David Lee's patch at + https://reviewboard.asterisk.org/r/2826/ and includes a regression fix + introduced by the ASTERISK-22455 patch. + + The initialization of a mutex's lock tracking structure was not protected + in a critical section. This is fine for any mutex that is explicitly + initialized, but a static mutex may have its lock tracking double + initialized if multiple threads attempt the first lock simultaneously. + + * Added a global mutex to properly serialize initialization of the lock + tracking structure. The painful global lock can be mitigated by adding a + double checked lock flag as discussed on the original review request. + + * Defer lock tracking initialization until first use. + + * Don't be "helpful" and initialize an uninitialized lock when + DEBUG_THREADS is enabled. Debug code is not supposed to fix or change + normal code behavior. We don't need a lock initialization race that would + force a re-setup of lock tracking. Lock tracking already handles + initialization on first use. + + * Properly handle allocation failures of the lock tracking structure. + + * No need to initialize tracking data in __ast_pthread_mutex_destroy() + just to turn around and destroy it. + + + The regression introduced by ASTERISK-22455 is the result of manipulating + a pthread_mutex_t struct outside of the pthread library code. The + pthread_mutex_t struct seems to have a global linked list pointer member + that can get changed by other threads. Therefore, saving and restoring + the contents of a pthread_mutex_t struct is a bad thing. + + Thanks to Thomas Airmont for finding this obscure regression. + + * Don't overwrite the struct ast_lock_track.reentr_mutex member to restore + tracking data in __ast_cond_wait() and __ast_cond_timedwait(). The + pthread_mutex_t struct must be treated as a read-only opaque variable. + + + Miscellaneous other items fixed by this patch: + + * Match ast_suspend_lock_info() with ast_restore_lock_info() in + __ast_cond_timedwait(). + + * Made some uninitialized lock sanity checks return EINVAL and try a + DO_THREAD_CRASH. + + * Fix bad canlog initialization expressions. + + ASTERISK-24614 #close + Reported by: Thomas Airmont + + Review: https://reviewboard.asterisk.org/r/4247/ + Review: https://reviewboard.asterisk.org/r/2826/ + ........ + + Merged revisions 429539 from http://svn.asterisk.org/svn/asterisk/branches/11 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 16:54 +0000 [901221ffae] Matt Jordan + + * res/res_agi: Make Verbose message for 'stream file' match other playbacks + + The Verbose message displayed when a file is played back via 'stream file' + was formatted differently than other playbacks: + * It didn't include the channel name + * It didn't include the channel language + It does, however, include the playback offset as well as any escape digits. + That information was kept; however, this patch updates the formatting to more + closely match the Verbose messages displayed when a file is played back by + 'control stream file', Playback, ControlPlayback, or any other file playback + operation. + ........ + + Merged revisions 429519 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 11:01 +0000 [8d325be503] Joshua Colp + + * media: Fix crash when determining sample count of a frame during shutdown. + + When shutting down Asterisk the codecs are cleaned up. As a result anything + attempting to get a codec based on ID or details will find that no codec + exists. This currently occurs when determining the sample count of a frame. + This code did not take this situation into account. + + This change fixes this by getting the codec directly from the format and + eliminates the lookup. This is both faster and also provides a guarantee + that the codec will exist and will be valid. + + ASTERISK-24604 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4260/ + ........ + + Merged revisions 429497 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 09:31 +0000 [72499dc697] Kevin Harwell + + * chan_pjsip: Race between channel answer and bridge setup when using direct media + + When direct media is enabled and a pjsip channel is answered a race would occur + between the handling of the answer and bridge setup. Sometimes the media + negotiation would take place after the native bridge was setup. This resulted + in a NULL media address, which in turn resulted in Asterisk using its address + as the remote media address when sending a reinvite. This patch makes the + chan_pjsip answer handler synchronous thus alleviating the race condition (the + bridge won't start setting things up until after it returns). + + ASTERISK-24563 #close + Reported by: Steve Pitts + Review: https://reviewboard.asterisk.org/r/4257/ + ........ + + Merged revisions 429477 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429478 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 09:03 +0000 [2e6d2b1484] David M. Lee + + * Fix crash for sorcery misconfigs + + res_pjsip_outbound_publish was missing the CHECK_PJSIP_MODULE_LOADED() + call in load_module, and would crash with a segfault if res_pjsip + declined to load. + + Review: https://reviewboard.asterisk.org/r/4258/ + ........ + + Merged revisions 429457 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 08:12 +0000 [a6cf13f2e9] Kinsey Moore + + * PJSIP: Allow use of 'inactive' streams for hold + + This allows use of the 'inactive' stream direction identifier to be + used for hold where 'sendonly' is normally used. Some Seimens phones + use 'inactive' and this change allows music on hold to operate + properly. + + Review: https://reviewboard.asterisk.org/r/4252/ + Reported by: Steve Pitts + ........ + + Merged revisions 429432 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429433 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429434 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 08:04 +0000 [b99770d4fe] Kinsey Moore + + * Sorcery: Log when old config remains in use + + This adds a log message notifying the user that a stale configuration + is in place upon reload when a config object fails to load. This + situation can end up causing confusion when the object failed to load + but exists from a previous config load especially when the old config + is significantly different from the new config. + + Review: https://reviewboard.asterisk.org/r/4250/ + Reported by: Thomas Thompson + ........ + + Merged revisions 429429 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429430 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429431 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 07:06 +0000 [74d43977cf] Joshua Colp + + * res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress. + + Given the scenario where a PJSIP channel is in a native RTP bridge with direct + media and the channel is then hung up the code will currently re-INVITE the channel + back to Asterisk and send a BYE at the same time. Many SIP implementations dislike + this greatly. + + This change makes it so that if a re-INVITE transaction is in progress the BYE + is queued to occur after the completion of the transaction (be it through normal + means or a timeout). + + Review: https://reviewboard.asterisk.org/r/4248/ + ........ + + Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-12 06:32 +0000 [8d384f3825] Joshua Colp + + * res_pjsip_session: Fix issue where a declined media stream in a re-INVITE would fail SDP negotiation. + + In the past the SDP negotiation within res_pjsip_session was made more tolerant of + certain situations. The only case where SDP negotiation will fail is when a major + error occurs during negotiation. Receiving an already declined media stream is + not considered a major error. + + When producing the local SDP the logic took this into account so on the initial INVITE + the declined media stream did not cause an SDP negotiation failure. Unfortunately + the logic for handling media streams with a handler did not mirror this logic and + considered an already declined media stream an error and thus failed the SDP + negotiation. + + This change makes the logic between both situations match so only under major + errors will the SDP negotiation fail. + + ASTERISK-24607 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4254/ + ........ + + Merged revisions 429407 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-11 14:32 +0000 [63d3f0af95] Kevin Harwell + + * ARI/AMI: Include language in standard channel snapshot output + + The CHANGES verbiage for the "language" addition had been put under the wrong + release. This moves it to be under 13.1 to 13.2 changes. + + ASTERISK-24553 + Reported by: Matt Jordan + ........ + + Merged revisions 429387 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-11 07:53 +0000 [d64b9904fd] Kinsey Moore + + * Stasis: Update unittest for channel snapshots + + This adjusts the unit test for channel snapshots to take the new + language key into account. + ........ + + Merged revisions 429352 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429353 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-10 09:43 +0000 [e890f9f653] Kevin Harwell + + * ARI/AMI: Include language in standard channel snapshot output + + Adding information about including "language" in the standard channel snapshot + output to the CHANGES file. Note the actual source changes have already been + previously committed. + + ASTERISK-24553 + Reported by: Matt Jordan + ........ + + Merged revisions 429325 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429326 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-10 07:35 +0000 [03c94ef761] Joshua Colp + + * res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero. + + Frames with a payload length of 0 were incorrectly handled in res_http_websocket. + Provided a frame with a payload had been received prior it was possible for a double + free to occur. The realloc operation would succeed (thus freeing the payload) but be + treated as an error. When the session was then torn down the payload would be + freed again causing a crash. The read function now takes this into account. + + This change also fixes assumptions made by users of res_http_websocket. There is no + guarantee that a frame received from it will be NULL terminated. + + ASTERISK-24472 #close + Reported by: Badalian Vyacheslav + + Review: https://reviewboard.asterisk.org/r/4220/ + Review: https://reviewboard.asterisk.org/r/4219/ + ........ + + Merged revisions 429270 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429272 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429273 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-10 07:16 +0000 [0cba439c4d] Kinsey Moore + + * PJSIP: Fix assert on initial mass qualify + + This fixes the MWI test regressions caused by r429127 and ensures that + contacts have non-zero qualify_frequency before attempting scheduling. + ........ + + Merged revisions 429245 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429246 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-09 14:47 +0000 [8fe45f0f0a] Scott Griepentrog + + * core: avoid possible asterisk -r crash from long id + + When connecting to the remote console, an id string + is first provided that consts of the hostname, pid, + and version. This is parsed by the remote instance + using a buffer that may be too short, and can allow + a buffer overrun because it is not terminated. This + patch adds termination and a larger buffer. + + Review: https://reviewboard.asterisk.org/r/4182/ + ........ + + Merged revisions 429223 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-09 14:20 +0000 [d673209abc] Kevin Harwell + + * ARI/AMI: Include language in standard channel snapshot output + + The channel "language" was already part of a channel snapshot, however is was + not sent out over AMI or ARI. This patch makes it so the channel "language" is + included in the appropriate AMI or ARI events. + + ASTERISK-24553 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4245/ + ........ + + Merged revisions 429204 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429206 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429209 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-09 14:03 +0000 [c17cef1c38] Kevin Harwell + + * Direct Media calls within private network sometimes get one way audio + + When endpoints with direct_media enabled, behind a firewall (Asterisk on a + separate network) and were bridged sometimes Asterisk would send the ip + address of the firewall in the sdp to one of the phones in the reinvite + resulting in one way audio. When sending the reinvite Asterisk will retrieve + the media address from the associated rtp instance, but if frames were being + read this can be overwritten with another address (in this case the + firewall's). This patch ensures that Asterisk uses the original device + address when using direct media. + + ASTERISK-24563 + Reported by: Steve Pitts + Review: https://reviewboard.asterisk.org/r/4216/ + ........ + + Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429196 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-09 12:36 +0000 [7844266e21] Kevin Harwell + + * res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard + + When using a non-default sorcery wizard (in this instance realtime) for outbound + publishes Asterisk will crash after a stack overflow occurs due to the code + infinitely recursing. The fix entails removing the outbound publish state + dependency from the outbound publish sorcery object and instead keeping an in + memory container that can be used to lookup the state when needed. + + ASTERISK-24514 #close + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4178/ + ........ + + Merged revisions 429175 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429176 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-09 09:45 +0000 [60ab564ad2] Joshua Colp + + * ari: Add support for specifying an originator channel when originating. + + If an originator channel is specified when originating a channel the linked ID + of it will be applied to the newly originated outgoing channel. This allows + an association to be made between the two so it is known that the originator + has dialed the originated channel. + + ASTERISK-24552 #close + Reported by: Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4243/ + ........ + + Merged revisions 429153 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429154 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-09 08:01 +0000 [b6e18cae5c] Kinsey Moore + + * PJSIP: Stagger outbound qualifies + + This change staggers initiation of outbound qualify (OPTIONS) attempts + to reduce instantaneous server load and prevent network congestion. + + Review: https://reviewboard.asterisk.org/r/4246/ + ASTERISK-24342 #close + Reported by: Richard Mudgett + ........ + + Merged revisions 429127 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429128 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-08 10:54 +0000 [fe6cbf455a] Matt Jordan + + * AMI/ARI: Update version to 2.6.0/1.6.0 respectively for new features + + AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per + semantic versioning, that warrants a bump in the minor version number, as it + reflects a backwards compatible change. Hence, this commit. + ........ + + Merged revisions 429091 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-08 10:43 +0000 [bba1763f47] Mark Michelson + + * Fix a crash that would occur when receiving a 491 response to a reinvite. + + The reviewboard description does a fine job of summarizing this, so here it is: + + A reporter discovered that Asterisk would crash when attempting to retransmit + a reinvite that had previously received a 491 response. The crash occurred + because a pjsip_tx_data structure was being saved for reuse, but its reference + count was not being increased. The result was that the pjsip_tx_data was being + freed before we were actually done with it. When we attempted to re-use the + structure when re-sending the reinvite, Asterisk would crash. + + The fix implemented here is not to try holding onto the pjsip_tx_data at all. + Instead, when we reschedule sending the reinvite, we create a brand new + pjsip_tx_data and send that instead. Because of this change, there is no need + for an ast_sip_session_delayed_request structure to have a pjsip_tx_data on + it any more. So any code referencing its use has been removed. + + When this initial fix was introduced, I encountered a second crash when + processing a subsequent 200 OK on a rescheduled reinvite. The reason was + that when rescheduling the reinvite, we gave the wrong location for a + response callback. This has been fixed in this patch as well. + + ASTERISK-24556 #close + Reported by Abhay Gupta + + Review: https://reviewboard.asterisk.org/r/4233 + ........ + + Merged revisions 429089 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-08 10:24 +0000 [fe7671fee6] Mark Michelson + + * Add new AMI and ARI events for connected line changes on a channel. + + The AMI event is called NewConnectedLine and the ARI event is called + ChannelConnectedLine. + + ASTERISK-24554 #close + Reported by Matt Jordan + + Review: https://reviewboard.asterisk.org/r/4231 + ........ + + Merged revisions 429064 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-08 09:45 +0000 [4bb556a847] Kinsey Moore + + * Stasis: Fix StasisStart/End order and missing events + + This corrects several bugs that currently exist in the stasis + application code. + + * After a masquerade, the resulting channels have channel topics that + do not match their uniqueids + ** Masquerades now swap channel topics appropriately + * StasisStart and StasisEnd messages are leaked to observer + applications due to being published on channel topics + ** StasisStart and StasisEnd publishing is now properly restricted + to controlling apps via app topics + * Race conditions exist where StasisStart and StasisEnd messages due to + a masquerade may be received out of order due to being published on + different topics + ** These messages are now published directly on the app topic so this + is now a non-issue + * StasisEnds are sometimes missing when sent due to masquerades and + bridge swaps into and out of Stasis() + ** This was due to StasisEnd processing adjusting message-sent flags + after Stasis() had already exited and Stasis() had been re-entered + ** This was corrected by adjusting these flags prior to sending the + message while the initial Stasis() application was still shutting + down + + Review: https://reviewboard.asterisk.org/r/4213/ + ASTERISK-24537 #close + Reported by: Matt DiMeo + ........ + + Merged revisions 429061 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429062 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-06 12:16 +0000 [49aa87e17c] Nuno Borges (License 6116) + + * res/res_monitor: Reset in/out sample counts on Monitor start + + When repeatedly starting/stopping a Monitor on a channel, the accumulated + in/out sample counts are never reset to 0. This can cause inadvertent jumps + in the recordings, as the code in the channel core will determine incorrectly + that a jump in the recorded file position should occur. Setting the sample + counts to 0 simply reflects the initial state a Monitor should be in when it + is started, as this is the initial count that would be on the channels at that + time. + + ASTERISK-24573 #close + Reported by: Nuno Borges + patches: + 24573.patch uploaded by Nuno Borges (License 6116) + ........ + + Merged revisions 429031 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429032 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429033 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-06 11:36 +0000 [0cdb71aae9] Nuno Borges (License 6116) + + * apps/app_meetme: Apply default values on initial load with no config file + + When the app_meetme module is loaded without its configuration file, the + module settings aren't initialized. In particular, this impacts the use + of logging realtime members. This patch guarantees that we always set the + default module settings on initial load. + + Review: https://reviewboard.asterisk.org/r/4242/ + + ASTERISK-24572 #close + Reported by: Nuno Borges + patches: + 24572.patch uploaded by Nuno Borges (License 6116) + ........ + + Merged revisions 429027 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 429028 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429029 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-05 11:08 +0000 [d04445c24a] gtjoseph + + * sorcery: Add additional observer capabilities. + + Add new global, instance and wizard observers. + instance_created + wizard_registered + wizard_unregistered + instance_destroying + instance_loading + instance_loaded + wizard_mapped + object_type_registered + object_type_loading + object_type_loaded + wizard_loading + wizard_loaded + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4215/ + ........ + + Merged revisions 428999 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 429000 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429001 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-04 11:13 +0000 [19992844be] Matt Jordan + + * main/test: Fix compilation issue on 32-bit systems + + On a 32-bit system, a type of intmax_t will result in a compilation warning + when formatted as a 'long int'. Use the format specifier of %jd (which was + what was used originally in manager.c) to format the JSON extracted integer + on both 32-/64-bit systems. + ........ + + Merged revisions 428972 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428973 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-04 09:48 +0000 [343a83d7d8] Matt Jordan + + * main/test: Fix race condition between AMI topic and Test Suite topic + + This patch fixes a race condition between the raising of test AMI events (which + drive many tests in the Asterisk Test Suite) and other AMI events. Prior to + this patch, the Stasis messages published to the test topic were not forwarded + to the AMI topic. Instead, the code in manager had a dedicated handler for test + messages that was independent of the topics forwarded to the AMI topic. This + results in no synchronization between the test messages and the rest of the + Stasis messages published out over AMI. In some test with very tight timing + constraints, this can result in out of order messages and spurious test + failures. Properly forwarding the Test Suite topic to the AMI topic ensures + that the messages are synchronized properly. + + This patch does that, and moves the message handling to the Stasis definition + of the Test Suite message in test.c as well. + + Review: https://reviewboard.asterisk.org/r/4221/ + ........ + + Merged revisions 428945 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428946 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-03 14:59 +0000 [7cb2c446b4] Matt Jordan + + * tests/test_cel: Add test_cel_attended_transfer_bridges_link to racey tests + + Despite failing less often, the ordering of the ATTENDEDTRANSFER event and the + BRIDGE_EXIT event for the Alice and David channels is not defined. This makes + the test still fail. + ........ + + Merged revisions 428918 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428919 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-03 13:49 +0000 [7475e1c948] Matt Jordan + + * tests/test_cel: Fix CEL unit test failures caused by attended transfer changes + + When the publication of attended transfer messages were pushed to another + thread, some subtle race conditions were introduced with the CEL unit tests. + This patch fixes one of them, and pushes the other to ASTERISK-22367, which + already exists to fix another bouncy CEL unit test. + + In particular, this patch fixes the test_cel_attended_transfer_bridges_link + test, and defers the test_cel_attended_transfer_bridges_swap test to the + aforementioned JIRA issue. + + ASTERISK-22367 + ........ + + Merged revisions 428891 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428892 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-03 10:45 +0000 [6d4ef7ddf4] David Duncan Ross Palmer (License 6660) + + * apps/app_voicemail: Fix crash with IMAP when streams are opened simultaneously + + The UW IMAP library is instrinsically not thread-safe, and relies upon higher + level applications to guarantee thread safety. For the most part, this is + provided by the vms object, which provides locking for individual streams. + Unfortunately, this is not sufficient for calls to mail_open which create the + IMAP stream. mail_open can, on some systems, call into a UW IMAP specific + function for determining the address of a system based on a hostname, + ip_nametoaddr. + + In the ip6_unix implementation of this function, static variables are used + to hold parsing buffers. This can cause a crash if multiple threads attempt + to convert a hostname to an address at the same time. Locking on a single + mail stream is not sufficient to prevent simultaneous access to these static + variables. + + In the IMAP library, this function can be called from the mail_open and + imap_status functions. As the imap_status function is not used by + app_voicemail, locking on access to mail_open is sufficient to prevent + any mangling of the buffers. + + Review: https://reviewboard.asterisk.org/r/4188/ + + ASTERISK-24516 #close + Reported by: David Duncan Ross Palmer + Tested by: David Duncan Ross Palmer + patches: + ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660) + ........ + + Merged revisions 428863 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428864 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428865 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-02 15:54 +0000 [63cbd28999] gtjoseph + + * CHANGES: Add item for new 'pjsip show identif(y|ies) commands + + Tested-by: George Joseph + ........ + + Merged revisions 428836 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428837 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-02 13:04 +0000 [dd00e80cbe] Matt Jordan + + * tests/test_stasis: Resolve compilation issues from Asterisk 12 merge + + When merging the changes up stream in r428687, I missed the fact that the + signature for stasis_message_type_create was changed. This patch fixes + the compilation issues introduced by that merge. + ........ + + Merged revisions 428815 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-02 11:10 +0000 [08636aadec] Birger Harzenetter (License 5870) + + * pbx/pbx_loopback: Speed up switches by avoiding unneeded lookups + + This patch makes a small rearrangement to only do dialplan lookups during + loopback switches if the pattern matches. Prior to this patch, the dialplan + lookups were always performed, even when the result would be discarded. + Dialplan lookups can be very costly if remote switches - like DUNDi - are + present. In those cases extension matching is sped up considerably, making + the issue of lost digits more manageable. + + As collateral damage, 6 trailing spaces were killed. + + Review: https://reviewboard.asterisk.org/r/4211 + + ASTERISK-24577 #close + Reported by: Birger Harzenetter + patches: + ast-loopback.patch uploaded by Birger Harzenetter (License 5870) + ........ + + Merged revisions 428787 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428788 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428789 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-02 06:21 +0000 [0c1aaa7da5] Joshua Colp + + * res_pjsip_refer: Fix issue where native bridge may not occur upon completion of a transfer. + + There are two methods within res_pjsip_refer for keeping track of the state of a transfer. + The first is a framehook which looks at frames passing by to determine the state. The second + subscribes to know when the channel joins a bridge. In the case when the channel joins the + bridge the framehook is *NOT* removed and this prevents the native RTP bridging technology + from getting used. + + This change gets the channel and if it still exists remove the framehook. + + Review: https://reviewboard.asterisk.org/r/4218/ + ........ + + Merged revisions 428760 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428761 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-01 18:38 +0000 [f128ff61ab] gtjoseph + + * config: Create ast_variable_find_in_list() + + Add + const char *ast_variable_find_in_list(const struct ast_variable *list, + const char *variable); + + ast_variable_find() requires a config category to search whereas + ast_variable_find_in_list() just needs the root list element which is + useful if you don't have a category. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4217/ + ........ + + Merged revisions 428733 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428734 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-01 18:31 +0000 [f418f25c44] gtjoseph + + * res_pjsip_endpoint_identifier_ip: Add 'show identify(ies)' cli commands + + While troubleshooting other things I realized there were no pjsip cli + commands for identify. This patch adds them. It also also fixes a + reference leak when a 'show endpoint' displayed identifies and properly + sets the return code if load_module can't allocate a cli formatter structure. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4212/ + ........ + + Merged revisions 428725 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428731 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428732 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-01 12:51 +0000 [4ff6bd831f] Joshua Colp + + * rtp_engine: Add support for transporting signed linear at 12kHz, 24kHz, 32kHz, 44kHz, 48kHz, 96kHz, and 192kHz over RTP. + + This change adds mappings in the RTP engine layer for the remaining signed linear formats. + + ASTERISK-24274 #close + Reported by: Frankie Chin + + Review: https://reviewboard.asterisk.org/r/4093/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-01 11:59 +0000 [1106e8fd0f] Matt Jordan + + * main/stasis: Allow subscriptions to use a threadpool for message delivery + + Prior to this patch, all Stasis subscriptions would receive a dedicated + thread for servicing published messages. In contrast, prior to r400178 + (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions + shared a thread pool. It was discovered during some initial work on Stasis + that, for a low subscription count with high message throughput, the + threadpool was not as performant as simply having a dedicated thread per + subscriber. + + For situations where a subscriber receives a substantial number of messages + and is always present, the model of having a dedicated thread per subscriber + makes sense. While we still have plenty of subscriptions that would follow + this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into + the following two categories: + * Large number of subscriptions, specifically those tied to endpoints/peers. + * Low number of messages. Some subscriptions exist specifically to coordinate + a single message - the subscription is created, a message is published, the + delivery is synchronized, and the subscription is destroyed. + In both of the latter two cases, creating a dedicated thread is wasteful (and + in the case of a large number of peers/endpoints, harmful). In those cases, + having shared delivery threads is far more performant. + + This patch adds the ability of a subscriber to Stasis to choose whether or not + their messages are dispatched on a dedicated thread or on a threadpool. The + threadpool is configurable through stasis.conf. + + Review: https://reviewboard.asterisk.org/r/4193 + + ASTERISK-24533 #close + Reported by: xrobau + Tested by: xrobau + ........ + + Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-01 07:41 +0000 [ef9ca8bc32] Ben Smithurst (license 6529) + + * app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording. + + The Record dialplan function trims 1/4 of a second from the end of recordings in case + they are terminated because of DTMF. When hanging up, however, you don't want this to happen. + This change makes it so on hangup this does not occur. + + ASTERISK-24530 #close + Reported by: Ben Smithurst + patches: + app_record_v2.diff submitted by Ben Smithurst (license 6529) + + Review: https://reviewboard.asterisk.org/r/4201/ + ........ + + Merged revisions 428653 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428654 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428655 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-12-01 07:08 +0000 [7db3d1642b] snuffy (license 5024) + + * channel: Extend size of buffer for codecs in "core show channeltype" CLI command. + + The static buffer for codecs when invoking the "core show channeltype" CLI command + did not have enough room for all codecs. This has been extended so it does. + + ASTERISK-24542 #close + Reported by: snuffy + patches: + channeltype-tech.diff submitted by snuffy (license 5024) + + Review: https://reviewboard.asterisk.org/r/4204/ + ........ + + Merged revisions 428632 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428633 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-24 14:39 +0000 [3e08619faf] Richard Mudgett + + * test_channel_feature_hooks.c: Fix unit test for DTMF hooks. + + Fix the failing /channels/features/test_features_channel_dtmf unit test. + + DTMF emulation does not work without a stream of packets to prod the + emulation code. + + Review: https://reviewboard.asterisk.org/r/4199/ + ........ + + Merged revisions 428604 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428605 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-24 14:32 +0000 [c38ffca9a1] Richard Mudgett + + * DTMF hooks: Leaving channels need to push any collected digits into the bridge. + + Any partially collected DTMF digits for a DTMF hook need to be pushed into + the bridge when a channel leaves the bridging system as if there were a + timeout. + + Review: https://reviewboard.asterisk.org/r/4199/ + ........ + + Merged revisions 428601 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428602 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-21 13:16 +0000 [3576ae47f4] Richard Mudgett + + * manager: Fix could not extend string messages. + + When shutting down Asterisk that has an active AMI connection, you get + several "failed to extend from %d to %d" messages because use of the + EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the + event. + + * Created MAX_AUTH_PERM_STRING to use when creating stack based struct + ast_str variables used with the authority_to_str() and + user_authority_to_str() functions instead of a variety of magic numbers + that could be too small. + + * Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so + it will not attempt to add all permission level strings. + + Review: https://reviewboard.asterisk.org/r/4200/ + ........ + + Merged revisions 428570 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428571 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428572 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-21 11:49 +0000 [4394e0431c] gtjoseph + + * sorcery: Make is_object_field_registered handle field names that are regexes. + + As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime + was tossing database fields that didn't have an exact match to a sorcery + registered field. This broke the ability to use regexes as field names which + manifested itself as a failure of res_pjsip_phoneprov_provider which uses + this capability. It also broke handling of fields that start with '@' in + realtime but I don't think anyone noticed. + + This patch does the following... + * Modifies ast_sorcery_fields_register to pre-compile the name regex. + * Modifies ast_sorcery_is_object_field_registered to test the regex if it + exists instead of doing an exact strcmp. + * Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work + with realtime. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4185/ + ........ + + Merged revisions 428543 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428544 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-21 07:59 +0000 [d663e045f5] Olle Johansson + + * sip.conf.sample - note that media_address does not change listen address, just the SDP + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 20:17 +0000 [2be984fb11] Matt Jordan + + * main/bridge_basic: Fix features regressions introduced by r428165 + + In r428165, two bugs were introduced: + + * Prior to entering the features retry loop, the buffer that holds the + collected digits is wiped. However, this inadvertently wipes out the + first collected digit on the first pass through, which is obtained + in ast_stream_and_wait. This caused all of the features tests to fail. + * If ast_app_dtget returns a hangup (-1), the loop would retry incorrectly. + If we detect a hangup, we have to stop trying the feature. + + This patch fixes both issues. + + Review: https://reviewboard.asterisk.org/r/4196/ + ........ + + Merged revisions 428505 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 10:37 +0000 [2f78fde10f] Matt Jordan (License #6283) + + * Fix error with mixed address family ACLs. + + Prior to this commit, the address family of the first item in an ACL + was used to compare all incoming traffic. This could lead to traffic + of other IP address families bypassing ACLs. + + ASTERISK-24469 #close + + Reported by Matt Jordan + Patches: + ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283) + + AST-2014-012 + ........ + + Merged revisions 428402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 428417 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428422 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428425 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428426 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 10:35 +0000 [2486b48cec] Gareth Palmer (license 5169) + + * AST-2014-018 - func_db: DB Dialplan function permission escalation via AMI. + + The DB dialplan function when executed from an external protocol (for instance + AMI), could result in a privilege escalation. + + Asterisk now inhibits the DB function from being executed from an external + interface if the live_dangerously option is set to no. + + ASTERISK-24534 + Reported by: Gareth Palmer + patches: submitted by Gareth Palmer (license 5169) + ........ + + Merged revisions 428331 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 428363 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428409 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428413 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428418 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 10:25 +0000 [2f97486d43] Jonathan Rose + + * PJSIP ACLs: Fix ACLs not loading on startup and apply/acl issues on contact + + The biggest problem this patch fixes is that ACLs weren't previously being + loaded when the res_pjsip_acl module was loaded. Yikes. In addition, the + ACL options contact_permit and contact_acl were effectively interpreted as + contact_deny and this patch fixes that as well. + + AST-1418 #close + Reported by: Thomas Thompson + Review: https://reviewboard.asterisk.org/r/4120/ + + ASTERISK-24531 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4171/ + ........ + + Merged revisions 428333 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428343 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 09:57 +0000 [a389f2d7a0] Kevin Harwell + + * AST-2014-017 - app_confbridge: permission escalation/ class authorization. + + Confbridge dialplan function permission escalation via AMI and inappropriate + class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan + function when executed from an external protocol (for instance AMI), could + result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord” + could also be used to execute arbitrary system commands without first checking + for system access. The AMI “ConfbridgeStopRecord” has also been updated to + only run under a system authorization. + + Asterisk now inhibits the CONFBRIDGE function from being executed from an + external interface if the live_dangerously option is set to no. Also, the + “ConfbridgeStartRecord” AMI action is now only allowed to execute under a + user with system level access. + + ASTERISK-24490 + Reported by: Gareth Palmer + ........ + + Merged revisions 428332 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428334 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428339 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428342 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 08:56 +0000 [1c88ca9d31] Joshua Colp + + * AST-2014-016: Fix crash when receiving an in-dialog INVITE with Replaces in res_pjsip_refer. + + The implementation of INVITE with Replaces in res_pjsip_refer did not expect them to + occur in-dialog. As a result it would incorrectly attempt to hang up a channel it + thought was under its control. In reality the channel would be under the control of + another thread. When the other thread accessed the channel it would be accessing freed + memory and could crash. + + This change makes res_pjsip_refer not act on an in-dialog INVITE with Replaces. + + ASTERISK-24528 #close + Reported by: Joshua Colp + ........ + + Merged revisions 428304 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428305 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-20 08:49 +0000 [d25eda5fb2] Joshua Colp + + * AST-2014-015: Fix race condition in chan_pjsip when sending responses after a CANCEL has been received. + + Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may + be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK) + are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted. + + This change makes it so that these responses are not sent on disconnected sessions. + + ASTERISK-24471 #close + Reported by: yaron nahum + ........ + + Merged revisions 428301 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428302 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-19 13:32 +0000 [57c6f89bf0] Corey Farrell + + * stringfields: Fix bug in ast_string_fields_copy. + + ast_string_fields_copy relies on the fact that + __ast_string_field_release_active never previously + zeroed pool->used, so keeping the existing pointer + was "ok". Now that existing pools can be reset to + 'empty', it is important to set each field to + __ast_string_field_empty after releasing the memory. + + ASTERISK-24535 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4186/ + ........ + + Merged revisions 428272 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428273 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-19 11:22 +0000 [a7c9f4c668] Richard Mudgett + + * ast_str: Fix improper member access to struct ast_str members. + + Accessing members of struct ast_str outside of the string manipulation API + routines is invalid since struct ast_str is supposed to be treated as + opaque. + + Review: https://reviewboard.asterisk.org/r/4194/ + ........ + + Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428246 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-19 06:50 +0000 [7f8b7ace72] Joshua Colp + + * res_pjsip_sdp_rtp: Add support for optimistic SRTP. + + Optimistic SRTP is the ability to enable SRTP but not have it be + a fatal requirement. If SRTP can be used it will be, if not it won't be. + This gives you a better chance of using it without having your sessions + fail when it can't be. + + Encrypt all the things! + + Review: https://reviewboard.asterisk.org/r/3992/ + ........ + + Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-19 06:45 +0000 [b2e766a6b7] Joshua Colp + + * alembic: Fix alembic migration for 'moh_passthrough' option in res_pjsip. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-19 05:51 +0000 [3119c3737f] Joshua Colp + + * res_pjsip_refer: Ensure Refer-To is NULL terminated and parse it as a URI. + + There is no guarantee that when we get a Refer-To that it will be NULL terminated. + As the URI parsing function requires it to be we now NULL terminate it. + + Additionally parsing the Refer-To as a 'To' header is needless and it can + simply be done as a URI. This also fixes a problem where certain Refer-To headers + would not be parsed as a 'To' header causing the REFER to fail. + + ASTERISK-24508 #close + Reported by: Beppo Mazzucato + + Review: https://reviewboard.asterisk.org/r/4187/ + ........ + + Merged revisions 428195 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428196 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-18 13:12 +0000 [a94efa239c] Richard Mudgett + + * parking_tests.c: Add missing newline on a unit test message. + ........ + + Merged revisions 428168 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428169 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428170 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-17 10:58 +0000 [2e750db120] Mark Michelson + + * Allow for transferer to retry when dialing an invalid extension. + + This allows for a configurable number of attempts for a transferer + to dial an extension to transfer the call to. For Asterisk 13, the + default values are such that upgrading between versions will not + cause a behaivour change. For trunk, though, the defaults will be + changed to be more user-friendly. + + Review: https://reviewboard.asterisk.org/r/4167 + ........ + + Merged revisions 428145 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-17 10:02 +0000 [4cea5fd4ba] Corey Farrell + + * chan_sip: Fix theoretical leak of p->refer. + + If transmit_refer is called when p->refer is already allocated, + it leaks the previous allocation. Updated code to always free + previous allocation during a new allocation. Also instead of + checking if we have a previous allocation, always create a + clean record. + + ASTERISK-15242 #close + Reported by: David Woolley + Review: https://reviewboard.asterisk.org/r/4160/ + ........ + + Merged revisions 428117 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428118 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428119 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-17 09:27 +0000 [948af7fd79] Matt Jordan + + * apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves + + When r428077 was made for ASTERISK-24522, it failed to take into account users + who are neither wait_marked nor end_marked. These users are *also* supposed to + hear the 'leader has left the conference' message. Granted, this behaviour is + a bit odd; however, that is how it used to work... and behaviour changes are + not good. + + This patch ensures that if there are any 'normal' users present when the last + marked user leaves the conference, the message will still be played to them. + + Note that this regression was caught by the Asterisk Test Suite's + confbridge_nominal test, which has a quirky combination of users. + ........ + + Merged revisions 428113 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428114 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428115 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-16 21:08 +0000 [fc2279afea] Matt Jordan + + * app_confbridge: Don't play leader leaving prompt if no one will hear it + + Consider the following: + - A marked user in a conference + - One or more end_marked only users in the conference + + When the marked users leaves, we will be in the conf_state_multi_marked state. + This currently will traverse the users, kicking out any who have the end_marked + flags. When they are kicked, a full ast_bridge_remove is immediately called on + the channels. At this time, we also unilaterally set the need_prompt flag. + + When the need_prompt flag is set, we then playback a sound to the bridge + informing everyone that the leader has left; however, no one is left in the + bridge. This causes some odd behaviour for the end_marked users - they are + stuck waiting for the bridge to be unlocked. This results in them waiting for + 5 or 6 seconds of dead air before hearing that they've been kicked. + + Unfortunately, we do have to keep the bridge locked while we're playing back + the 'leader-has-left' prompt. If there are any wait_marked users in the + conference, this behaviour can't be easily changed - but we do make the case + of the end_marked users better with this patch. + + Review: https://reviewboard.asterisk.org/r/4184/ + + ASTERISK-24522 #close + Reported by: Matt Jordan + ........ + + Merged revisions 428077 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 428078 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428079 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-16 15:13 +0000 [656601d8c4] Joshua Colp + + * chan_pjsip: Remove AOR check when dialing and one is specified. + + The AOR value may contain the name of an AOR or a full SIP URI. + Checking if the AOR exists can't be done as a result of this. + ........ + + Merged revisions 428051 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428052 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-16 06:12 +0000 [bc02cbabd9] Joshua Colp + + * chan_sip: Fix bug where DTLS configuration from general would copy dtlsenable. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-15 15:52 +0000 [6993743b1f] Etienne Lessard (License 6394) + + * cel/cel_odbc: Provide microsecond precision in 'eventtime' column when possible + + This patch adds microsecond precision when inserting a CEL record into a table + with an "eventtime" column of type timestamp, instead of second precision. The + documentation (configs/cel_odbc.conf.sample) was already saying that the + eventtime column included microseconds precision, but that was not the case. + + Also, without this patch, if you had a table with an "eventtime" column of + type varchar, you had millisecond precision. With this patch, you also get + microsecond precision in this case. + + Review: https://reviewboard.asterisk.org/r/3980 + + ASTERISK-24283 #close + Reported by: Etienne Lessard + patches: + cel_odbc_time_precision.patch uploaded by Etienne Lessard (License 6394) + ........ + + Merged revisions 427952 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427953 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427954 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-15 15:36 +0000 [ece61f5ed1] Joshua Colp + + * chan_pjsip: Add additional log message when an AOR is specified when dialing and it does not exist. + + ASTERISK-24499 #close + Reported by: Rusty Newton + ........ + + Merged revisions 428007 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 428008 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-15 13:01 +0000 [49e63a191d] Joshua Colp + + * chan_motif / chan_pjsip: Fix incorrect "No such module" messages when reloading. + + For chan_motif the direct return value of the underlying config options framework + was passed back. This can relay various states which the module loader would not + interpet as success. It has been changed so only on errors will it report back + an error. + + For chan_pjsip the code implemented a dummy reload function which always + returned an error. This has been removed as all configuration is held within + res_pjsip instead. + + ASTERISK-23651 #close + Reported by: Rusty Newton + ........ + + Merged revisions 427981 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427982 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-15 12:29 +0000 [9d2882d274] Joshua Colp + + * res_pjsip: Enforce requirements for session timer minimum expiration period and normal expiration period. + + This change enforces the requirements in PJSIP for session timer configuration. The minimum + expiration period must be 90 seconds or higher and the normal expiration period can not + be lower than the minimum expiration period. If either of these were done the code would + assert at session setup time. + + ASTERISK-24336 #close + Reported by: Leon Rowland + ........ + + Merged revisions 427978 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427979 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-15 10:31 +0000 [d0523b4b3c] Michael K. (license 6621) + + * chan_sip: Add support for setting DTLS configuration in the general section. + + Configuration of DTLS in the general section will be applied to any users + or peers. If configuration exists at their level it overrides the general + section values. + + ASTERISK-24128 #close + Reported by: Michael K. + patches: + dtls_default_settings.patch submitted by Michael K. (license 6621) + + Review: https://reviewboard.asterisk.org/r/3867/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-14 15:51 +0000 [3268544907] Matt Jordan + + * tests/test_cel: Unlock bridge on off nominal paths + + If the test fails due to memory allocation errors, we may as well attempt to + unlock the bridge on the way out. + ........ + + Merged revisions 427927 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-14 12:12 +0000 [df2090b931] Jonathan Rose + + * Documentation: Revise explanation of cdr.conf option 'Unanswered' + + ASTERISK-24279 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4109/ + ........ + + Merged revisions 427901 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427902 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-14 09:52 +0000 [ba811ae1c3] Scott Griepentrog + + * stun: correct attribute string padding to match rfc + + When sending the USERNAME attribute in an RTP STUN + response, the implementation in append_attr_string + passed the actual length, instead of padding it up + to a multiple of four bytes as required by the RFC + 3489. This change adds separate variables for the + string and padded attributed lengths, and performs + padding correctly. + + Reported by: Thomas Arimont + Review: https://reviewboard.asterisk.org/r/4139/ + ........ + + Merged revisions 427874 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427875 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427876 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-14 09:28 +0000 [2d9471ab1f] Mark Michelson + + * Fix race condition that could result in ARI transfer messages not being sent. + + From reviewboard: + + "During blind transfer testing, it was noticed that tests were failing + occasionally because the ARI blind transfer event was not being sent. + After investigating, I detected a race condition in the blind transfer + code. When blind transferring a single channel, the actual transfer + operation (i.e. removing the transferee from the bridge and directing + them to the proper dialplan location) is queued onto the transferee + bridge channel. After queuing the transfer operation, the blind transfer + Stasis message is published. At the time of publication, snapshots of + the channels and bridge involved are created. The ARI subscriber to the + blind transfer Stasis message then attempts to determine if the bridge + or any of the involved channels are subscribed to by ARI applications. + If so, then the blind transfer message is sent to the applications. The + way that the ARI blind transfer message handler works is to first see + if the transferer channel is subscribed to. If not, then iterate over + all the channel IDs in the bridge snapshot and determine if any of + those are subscribed to. In the test we were running, the lone + transferee channel was subscribed to, so an ARI event should have been + sent to our application. Occasionally, though, the bridge snapshot did + not have any channels IDs on it at all. Why? + + The problem is that since the blind transfer operation is handled by a + separate thread, it is possible that the transfer will have completed and + the channels removed from the bridge before we publish the blind transfer + Stasis message. Since the blind transfer has completed, the bridge on + which the transfer occurred no longer has any channels on it, so the + resulting bridge snapshot has no channels on it. Through investigation of + the code, I found that attended transfers can have this issue too for the + case where a transferee is transferred to an application." + + The fix employed here is to decouple the creation of snapshots for the transfer + messages from the publication of the transfer messages. This way, snapshots + can be created to reflect what they are at the time of the transfer operation. + + Review: https://reviewboard.asterisk.org/r/4135 + ........ + + Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-14 08:56 +0000 [737b811749] Joshua Colp + + * app_confbridge: Play "leader has left" sound even when musiconhold is enabled. + + Currently if the leader of a conference bridge leaves any participant + that has musiconhold enabled will not hear the "leader has left" sound. + This is because musiconhold is started and THEN the sound is played. + + This change makes it so that the sound is played and THEN musiconhold + is started. This provides a better experience for users as they may not + have known previously why they went back to musiconhold. + + Review: https://reviewboard.asterisk.org/r/4177/ + ........ + + Merged revisions 427844 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427845 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427846 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427847 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-14 08:40 +0000 [2454505d5a] Mark Michelson + + * Fix race condition where duplicated requests may be handled by multiple threads. + + This is the Asterisk 13 version of the patch. The main difference is in the pubsub + code since it was completely refactored between Asterisk 12 and 13. + + Review: https://reviewboard.asterisk.org/r/4175 + ........ + + Merged revisions 427841 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-13 16:26 +0000 [49b7a1cbaf] Kevin Harwell + + * res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash + + When using a non-default sorcery wizard (in this instance realtime) for + outbound registrations and after adding in an appropriate call to + ast_sorcery_apply_config() (since it is missing) Asterisk will crash after + a stack overflow occurs due to the code infinitely recursing. The fix entails + removing the outbound registration state dependency from the outbound + registration sorcery object and instead keeping an in memory container that + can be used to lookup the state when needed. + + ASTERISK-24514 + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4164/ + ........ + + Merged revisions 427814 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427815 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427823 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-13 09:46 +0000 [74e706878b] Kinsey Moore + + * Stasis: Fix StasisEnd message ordering + + This change corrects message ordering in cases where a channel-related + message can be received after a Stasis/ARI application has received the + StasisEnd message. The StasisEnd message was being passed to + applications directly without waiting for the channel topic to empty. + + As a result of this fix, other bugs were also identified and fixed: + * StasisStart messages were also being sent directly to apps and are + now routed through the stasis message bus properly + * Masquerade monitor datastores were being removed at the incorrect + time in some cases and were causing StasisEnd messages to not be sent + * General refactoring where necessary for the above + * Unsubscription on StasisEnd timing changes to prevent additional + messages from following the StasisEnd when they shouldn't + + A channel sanitization function pointer was added to reduce processing + and AO2 lookups. + + Review: https://reviewboard.asterisk.org/r/4163/ + ASTERISK-24501 #close + Reported by: Matt Jordan + ........ + + Merged revisions 427788 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427789 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-12 18:23 +0000 [cc4c396647] Matt Jordan + + * main/rtp_engine: Fix crash when processing more than one RTCP report info block + + Asterisk - in res_rtp_asterisk - only understands a single RTCP report info + block. When the RTCP information was refactored in the RTP Engine to be pushed + over the Stasis message bus, I put in the hooks into the engine to handle + multiple RTCP report info blocks, in the hope that a future RTP implementation + would be able to provide that data. Unfortunately, res_rtp_asterisk has a + tendency to "lie": + (1) It will send RTCP reports with a reception_report_count greater than 1 + (which is pulled directly from the RTCP packet itself, so that part is + correct) + (2) It will only provide a single report block + + When the rtp_engine goes to convert this to a JSON blob, hilarity ensues as it + looks for a report block that doesn't exist. + + This patch updates the rtp_engine to be a bit more skeptical about what it is + presented with. While this could also be fixed in res_rtp_asterisk, this patch + prefers to fix it in the engine for two reasons: + (1) The engine is designed to work with multiple RTP implementation, and hence + having it be more robust is a good thing (tm) + (2) res_rtp_asterisk's handling of RTCP information is "fun". It should report + the correct reception_report_count; ideally it should also be giving us all + of the blocks - but it is *definitely* not designed to do that. Going down + that road is a non-trivial effort. + + Review: https://reviewboard.asterisk.org/r/4158/ + + ASTERISK-24489 #close + Reported by: Gregory Malsack + Tested by: Gregory Malsack + + ASTERISK-24498 #close + Reported by: Beppo Mazzucato + Tested by: Beppo Maazucato + ........ + + Merged revisions 427762 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427763 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-12 14:40 +0000 [ec1a7654f3] Corey Farrell + + * Fix leak in AMI Action Bridge + + Add missing reference cleanup for newly created bridge. + + ASTERISK-24281 + Reported by: Stefan Engström + Review: https://reviewboard.asterisk.org/r/4154/ + ........ + + Merged revisions 427736 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427737 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-12 10:13 +0000 [dbb8f0a935] Joshua Colp + + * pbx: Fix off-nominal case where a freed extension may still be used. + + If during the operation of adding an extension a priority is added but + fails it is possible for the extension to be freed but still exist in + the PBX core. If this occurs subsequent lookups may try to access the + extension and end up in freed memory. + + This change removes the extension from the PBX core when the priority + addition fails and then frees the extension. + + ASTERISK-24444 #close + Reported by: Leandro Dardini + + Review: https://reviewboard.asterisk.org/r/4162/ + ........ + + Merged revisions 427709 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427710 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427711 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427712 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-12 07:47 +0000 [9f89b83269] Corey Farrell + + * Fix compiler error when using ./configure --enable-dev-mode --enable-coverage + + When DONT_OPTIMIZE is enabled with dev-mode, it causes a shadow compilation + to be done with output to /dev/null. This can cause errors with coverage + when GCC attempts to write to /dev/null.gcno. This change disables + coverage for the shadow compilation. + + ASTERISK-24502 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4151/ + ........ + + Merged revisions 427682 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427683 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427684 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-09 02:01 +0000 [21c41e4542] Corey Farrell + + * manager: Fix HTTP connection reference leaks. + + Fix reference leak that happens if (session && !blastaway). + + ASTERISK-24505 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4153/ + ........ + + Merged revisions 427641 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427642 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427643 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-08 18:38 +0000 [f4392c4b6d] Xavier Hienne (License 6657) + + * channels/chan_mgcp: Fix regression which causes gateways to be skipped + + In r227276, a while loop was turned into a for loop. Unfortunately, a portion + of the while loop was left in the code such that, when a static gateway is + encountered in the list of MGCP gateways, the next gateway would be skipped. + At best, we would simply flip past a gateway; at worst, this could lead to a + crash. + + ASTERISK-24500 #close + Reported by: Xavier Hienne + patches: + chan_mgcp.patch uploaded by Xavier Hienne (License 6657) + ........ + + Merged revisions 427613 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427614 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427615 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-08 18:26 +0000 [d773f9d03e] Dmitriy Bubnov (License 6651),Dmitry Bubnov (License 6651) + + * addons/chan_mobile: Increase buffer size of UCS2 encoded SMS messages + + When UCS2 character encoding is used, one symbol in national language can be + expanded to 4 bytes. The current buffer used for receiving message in + do_monitor_phone is 256 bytes, which is not large enough for incoming messages. + + For example: + * AT+CMGR phone response prefix + '+CMGR: "REC UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes + * SMS body with UCS2 encoding (max) - 280 bytes + * AT+CMGR phone response suffix '\r\n\r\nOK\r\n' - 8 bytes + * Terminating null character - 1 byte + + This results in a needed buffer size of 349 bytes. Hence, this patch opts for a + 350 byte buffer. + + ASTERISK-24468 #close + Reported by: Dmitriy Bubnov + patches: + chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651) + chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651) + ........ + + Merged revisions 427607 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427610 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427611 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427612 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-08 18:14 +0000 [08d773532b] abelbeck (License 5903) + + * app_voicemail: Fix enhancement that allowed multiple recipients in To: header + + An issue existed in r420577, which added multiple recipients to voicemail + emails. The patch, when looking at the intended recipients, looked ahead for + the '|' character inside a while loop which already had pulled out the + appropriate field parsing on the '|' character. This would cause it to skip + the recipients. + + This patch fixes it such that it relies completely on the while loop to parse + through the e-mail fields. + + Note that the original author of the patch looked at this fix and approved it. + + ASTERISK-24250 #close + Reported by: abelbeck + patches: + voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903) + ........ + + Merged revisions 427585 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-08 18:04 +0000 [9a1ab5d548] Matt Jordan + + * bridge_native_rtp: Fix T.38 issues with remote bridges + + After r425242 the fax/sip/directmedia_reinvite_t38 test started failing due to + the surviving channel not being re-INVITEd back from T.38 to audio. This patch + fixes that bug - a deeper explanation of what happened follows. + + When two RTP channels are in a native bridge, the bridging layer will + investigate each via the get_rtp_info glue callback. This callback returns the + native bridge preference of the channel *at that moment in time* (that part is + key). At different points during the bridging, the native bridging layer will + inform the RTP capable channels of the status of the bridge via the update_peer + glue callback. + + In a T.38 scenario with audio direct media, the sequence of events will often + look like the following: + * SIP/A and SIP/B both have audio and enter a native bridge. + * Asterisk re-INVITEs audio between SIP/A and SIP/B directly (via an + update_peer callback). + * SIP/A sends a re-INVITE to T.38, which causes Asterisk to send a re-INVITE + to T.38 to SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack + receives UDPTL packets in Asterisk from both endpoints. From the perspective + of the channels, we are now in a local bridge for T.38, even though we are + technically still in a remote bridge in bridge_native_rtp. (YAY!) + * When one side hangs up, bridge_native_rtp is told to stop bridging. It then + re-evaluates the channels and asks them how they are bridged - and since + T.38 is enabled, they reply with a Local bridge (which is correct), but is + wrong because the audio portion is still technically in a remote bridge. + * Asterisk releases the surviving channel, whose audio is *not* re-INVITED + back to Asterisk as bridge_native_rtp incorrectly assumes that it was in a + local bridge. + + Ironically, prior to r425242, this used to work mostly due to a fluke in the + bridging layer. + + The purpose of the get_rtp_info callback shouldn't be modified: it should tell + the bridging layer what kind of bridge the channel prefers at that moment in + time. If you have T.38 enabled, that *must* be a local bridge, as the UDPTPL + stack must be in the media path. As such, this patch does not modify that + part of the code. + + However, we have to tell the channels to re-evaluate themselves when they come + out of a native bridge, since we can no longer trust the get_rtp_info callbacks + when the native bridge is being stopped. Something else may have changed in the + channels, and they may now be lying to us. As such, this patch makes it so that + we unilaterally tell the channels that they are no longer bridged via the + update_peer callback. This is actually what the channels expect anyway: code in + both chan_sip and chan_pjsip's callbacks look at the T.38 state and - if they + were in T.38 - send a re-INVITE to get the audio back to Asterisk. + + Review: https://reviewboard.asterisk.org/r/4157/ + ........ + + Merged revisions 427582 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427583 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427584 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-08 12:20 +0000 [d4fd0774f4] Corey Farrell + + * chan_console: Fix reference leaks to pvt. + + Fix a bunch of calls to get_active_pvt + where the reference is never released. + + ASTERISK-24504 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4152/ + ........ + + Merged revisions 427554 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427555 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427557 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-06 13:26 +0000 [7571bae5ab] Richard Mudgett + + * app_agent_pool: Made agent alert interruptable by DTMF. + + Made agent able to interrupt the alerting beep playback with DTMF. Any + digit can interrupt if the call does not need to be acknowledged. Only + the first digit of the acknowledgement can interrupt if the call needs to + be acknowledged. The agent interrupting the alerting playback builds on + the ASTERISK-24447 patch because it knows what digit interrupted the + playback and needs to be able to pass that digit to the DTMF hook digit + collection code. + + ASTERISK-24257 #close + Reported by: Steve Pitts + + Review: https://reviewboard.asterisk.org/r/4123/ + ........ + + Merged revisions 427508 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427512 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-06 13:12 +0000 [a68baad74f] Richard Mudgett + + * Bridge DTMF hooks: Made audio pass from the bridge while waiting for more matching digits. + + * Made collecting DTMF digits for the DTMF feature hooks pass frames from + the bridge. + + * Made collecting DTMF digits possible by other bridge hooks if there is a + need. + + ASTERISK-24447 #close + Reported by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4123/ + ........ + + Merged revisions 427493 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427494 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-06 12:21 +0000 [47074f4bfd] Joshua Colp + + * res_pjsip: Ensure in-dialog responses have an endpoint associated. + + When handling incoming messages we determine if it is associated with + a dialog. If so we use that to determine what serializer and endpoint + to use for the message. Previously this would pass the endpoint to the + endpoint lookup module to actually place the endpoint completely on the + message. For in-dialog responses, however, this did not occur as + dialog processing took over and the endpoint lookup did not occur. + + This change just places the endpoint in the expected spot immediately + instead of relying on the endpoint lookup module. In-dialog responses + thus have the expected endpoint. + + AST-1459 #close + + Review: https://reviewboard.asterisk.org/r/4146/ + ........ + + Merged revisions 427490 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427491 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427492 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-06 06:15 +0000 [4d80f223af] Corey Farrell + + * main/file.c: fix possible extra ast_module_unref to format modules. + + fn_wrapper only adds a reference to the format's module if the file + was able to be opened. If not this causes an unmatched + ast_module_unref in filestream_destructor. Move ast_module_ref to + get_stream. + + ASTERISK-24492 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4149/ + ........ + + Merged revisions 427464 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427465 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427466 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-06 03:24 +0000 [c46664305a] Corey Farrell + + * res_hep: fix major leak that occurs when config is missing or enabled=no. + + Add missing unreference in hepv3_send_packet. + + ASTERISK-24491 #close + Reported by: Zane Conkle + Tested by: Zane Conkle + Review: https://reviewboard.asterisk.org/r/4150/ + ........ + + Merged revisions 427400 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427405 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-06 03:18 +0000 [7e2369310c] Corey Farrell + + * Fix unintential memory retention in stringfields. + + * Fix missing / unreachable calls to __ast_string_field_release_active. + * Reset pool->used to zero when the current pool->active reaches zero. + + ASTERISK-24307 #close + Reported by: Etienne Lessard + Tested by: ibercom, Etienne Lessard + Review: https://reviewboard.asterisk.org/r/4114/ + ........ + + Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 427381 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427382 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427384 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-05 20:41 +0000 [362dde2229] gtjoseph + + * test_strings: Remove string tests that exercise asserts. + + Since unit tests are run with DO_CRASH, those tests were causing + the test to fail. + + Tested-by: George Joseph + ........ + + Merged revisions 427354 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427355 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427356 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-05 13:53 +0000 [69f29e627f] Mark Michelson + + * Make the disable_tcp_switch PJSIP system object enabled by default. + + Testing has shown repeatedly that PJSIP's default behavior of switching + automatically to TCP for large messages can cause issues. The most common + issues are that devices that we are communicating with do not handle the + switch to TCP gracefully, thus causing situations such as broken calls or + broken subscriptions. Now, in order to have this behavior happen, you must + opt into it. The sample file has been updated to warn that enabling the + TCP switch behavior may cause issues for you, so use at your own risk. + ........ + + Merged revisions 427334 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-05 06:19 +0000 [b06078880b] Joshua Colp + + * res_pjsip_multihomed: Add logging during startup to aid debugging if local DNS is misbehaving. + + This change adds a bit of logging so if the local DNS is misbehaving it is easier + to track down what is going on and where Asterisk may be hanging. + + ASTERISK-24438 #close + Reported by: Melissa Shepherd + + Review: https://reviewboard.asterisk.org/r/4148/ + ........ + + Merged revisions 427300 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427303 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 18:17 +0000 [d5de94201e] gtjoseph + + * config: Make text_file_save and 'dialplan save' escape semicolons in values. + + When a config file is read, an unescaped semicolon signals comments which are + stripped from the value before it's stored. Escaped semicolons are then + unescaped and become part of the value. Both of these behaviors are normal + and expected. When the config is serialized either by 'dialplan save' or + AMI/UpdateConfig however, the now unescaped semicolons are written as-is. + If you actually reload the file just saved, the unescaped semicolons are + now treated as start of comments. + + Since true comments are stripped on read, any semicolons in + ast_variable.value must have been escaped originally. This patch + re-escapes semicolons in ast_variable.values before they're written to + file either by 'dialplan save' or config/ast_config_text_file_save which + is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting + issues nearby in pbx_config.c + + Tested-by: George Joseph + ASTERISK-20127 #close + + Review: https://reviewboard.asterisk.org/r/4132/ + ........ + + Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427276 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 16:51 +0000 [c77a71ad2f] Joshua Colp + + * res_pjsip: Apply the 'user_eq_phone' setting to the To header as well. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 16:31 +0000 [5e43d68717] Joshua Colp + + * res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 14:49 +0000 [bd42a09d7f] gtjoseph + + * config: BUG: Restore ability for non-templ to be used as base objs in config. + + My recent refactor of config.c accidentally removed the capability for an + object to inherit from a non-template object. + + This patch restores the capability to inherit from both template and + non-template objects. + + Tested-by: George Joseph + Reported-by: Scott Griepentrog + ASTERISK-24487 #close + + Review: https://reviewboard.asterisk.org/r/4147/ + ........ + + Merged revisions 427227 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427228 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 13:46 +0000 [97e1c7f3a9] Corey Farrell + + * func_talkdetect: Fix stasis message leak in audiohook callback. + + ASTERISK-24482 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4142/ + ........ + + Merged revisions 427203 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427204 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427205 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 13:33 +0000 [9f2874639d] Corey Farrell + + * res_http_websockets: Fix extra unref of module + + In websocket_add_protocol_internal is used to add the "echo" + protocol, but ast_websocket_remove_protocol is used to remove + it. This causes an extra call to ast_module_unref. + + ASTERISK-24480 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4140/ + ........ + + Merged revisions 427200 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427201 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 08:11 +0000 [bdc35c77b9] Corey Farrell + + * Fix crash caused by merge error on review 4138 + + When merging from 12 to 13 there were conflicts, + I mistakenly had the loop run ast_closestream(others[0]) + when it should be ast_closestream(others[x]). + ........ + + Merged revisions 427181 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427182 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-04 06:03 +0000 [d159885e50] Joshua Colp + + * res_pjsip_outbound_registration: Add virtual line support. + + Virtual line support establishes a relationship between messages + related to an outbound registration and a local endpoint. This is + accomplished by attaching a parameter to the Contact of the outbound + registration and looking for it on any received requests. If the + parameter exists and can be matched to an outbound registration + the configured endpoint is associated with the request. + + Review: https://reviewboard.asterisk.org/r/2964/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-03 12:22 +0000 [33f0251b6c] Richard Mudgett + + * res_pjsip: Add disable_tcp_switch option. + + When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In + some circumstances (on some networks), this can cause some issues with + messages not getting sent to the correct destination - and can also cause + connections to get dropped due to quirks in pjproject deciding to + terminate TCP connections with no messages. + + While fixing the routing/messaging issues is important, having a + configuration option in Asterisk that tells pjproject to not switch over + to TCP would be useful. That way, if some glitch is discovered on some + other network/site, we can at least disable the behavior until a fix is + put into place. + + AFS-197 #close + + Review: https://reviewboard.asterisk.org/r/4137/ + ........ + + Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427130 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-03 09:03 +0000 [b9aeff9580] Joshua Colp + + * chan_pjsip: Update CHANGES file to include 'moh_passthrough' setting + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-03 08:45 +0000 [ac091d4184] Joshua Colp + + * chan_pjsip: Add support for passing hold and unhold requests through. + + This change adds an option, moh_passthrough, that when enabled will pass + hold and unhold requests through using a SIP re-invite. When placing on + hold a re-invite with sendonly will be sent and when taking off hold a + re-invite with sendrecv will be sent. This allows remote servers to handle + the musiconhold instead of the local Asterisk instance being responsible. + + Review: https://reviewboard.asterisk.org/r/4103/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-02 20:36 +0000 [285be15aaf] Corey Farrell + + * Fix compile error caused by review 4138 + + There is no procedure called ast_closeframe, fix code to use + ast_closestream. + + Reported By: Matt Jordan + ........ + + Merged revisions 427087 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427088 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427089 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-02 02:13 +0000 [509c04ef38] Corey Farrell + + * Fix ast_writestream leaks + + Fix cleanup in __ast_play_and_record where others[x] may be leaked. + This was caught where prepend != NULL && outmsg != NULL, once + realfile[x] == NULL any further others[x] would be leaked. A cleanup + block was also added for prepend != NULL && outmsg == NULL. + + 11+: Fix leak of ast_writestream recording_fs in + app_voicemail:leave_voicemail. + + ASTERISK-24476 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4138/ + ........ + + Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427026 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-02 01:40 +0000 [85c1822a9d] Corey Farrell + + * func_jitterbuffer: fix frame leaks. + + Fix code paths where it is possible for frames to leak. + Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. + + ASTERISK-22409 #related + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4128/ + ........ + + Merged revisions 427019 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 427020 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 427021 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-11-01 20:01 +0000 [5db1c978e3] Matt Jordan + + * res/res_stasis: Fix crash on module unload while performing operation + + When the res_stasis module is unloaded, it will dispose of the apps_registry + container. This is a problem if an ARI operation is in flight that attempts + to use the registry, as the shutdown occurs in a separate thread. This patch + adds some sanity checks to the various routines that access the registry which + cause the operations to fail if the apps_registry does not exist. + + Crash caught by the Asterisk Test Suite. + ........ + + Merged revisions 426995 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426996 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-31 11:52 +0000 [4219c40775] Tzafrir Cohen + + * install init.d files on GNU/kFreeBSD + + Review: https://reviewboard.asterisk.org/r/4118/ + ........ + + Merged revisions 426926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426927 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426933 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426934 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426935 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-31 11:41 +0000 [28173ddf05] Scott Griepentrog + + * pjsip: clarify tls cert and key file usage + + A question arose as to whether a .pem file + could be provided in place of the .crt and + .key files in a PJSIP TLS configuration. I + tested this and discovered that although a + cert will be read from the pem file, a key + will not, and thus the priv_key_file entry + is still required. This update to the fine + documentation clarifies the option usage. + + AST-1448 #close + Review: https://reviewboard.asterisk.org/r/4129/ + Reported by: John Bigelow + ........ + + Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426930 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-31 11:24 +0000 [f59db388a7] John Bigelow (License 5091) + + * pjsip: Handle outbound unregister correctly + + This updates the status of the outbound registration + to reflect when it has been unregistered. Since the + registration is unregistered but is not stopped, the + registration schedule remains active as before. The + patch also updates the documentation of both the AMI + and CLI commands. + + ASTERISK-24411 #close + Review: https://reviewboard.asterisk.org/r/4119/ + Reported by: John Bigelow + patches: + unregister-patch1.txt uploaded by John Bigelow (License 5091) + ........ + + Merged revisions 426923 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426924 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 22:26 +0000 [d88282af40] Matt Jordan + + * channels/sip/reqresp_parser: Fix unit tests for r426594 + + When r426594 was made, it did not take into account a unit test that verified + that the function properly populated the unsupported buffer. The function + would previously memset the buffer if it detected it had any contents; since + this function can now be called iteratively on successive headers, the unit + tests would now fail. This patch updates the unit tests to reset the buffer + themselves between successive calls, and updates the documentation of the + function to note that this is now required. + ........ + + Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426860 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426863 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426865 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 22:09 +0000 [bf684b63a3] Corey Farrell + + * REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts + + This change ensures refcounter.py is installed to a place where it + can be found by the Asterisk testsuite if REF_DEBUG is enabled. + + ASTERISK-24432 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4094/ + ........ + + Merged revisions 426830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426831 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426832 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426833 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 18:56 +0000 [e4374a3abe] Corey Farrell + + * app_queue: fix a couple leaks to struct call_queue in set_member_value + + set_member_value has a couple leaks to references in the variable q + found through testsuite tests/queues/set_penalty. Also remove the + REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible + with the updated REF_DEBUG code. + + ASTERISK-24466 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4125/ + ........ + + Merged revisions 426805 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426806 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426807 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 18:45 +0000 [ced81afff2] Corey Farrell + + * audiohooks: Clean references to formats + + Cleanup references to in_translate[x].format and + out_translate[x].format in ast_audiohook_detach_list. + + ASTERISK-24465 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4124/ + ........ + + Merged revisions 426803 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 16:14 +0000 [a537e314d1] Kevin Harwell + + * res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash + + Currently, it is possible for some subscriptions to get into a NULL state. When + this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a + device is subscribed for extension state then the associated subscription state + object can't be located. The code then attempts to dereference a NULL object. + Added a NULL check to avoid the problem. + + Reported by: John Bigelow + ........ + + Merged revisions 426779 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426780 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426781 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 12:18 +0000 [cd52456ea1] Kevin Harwell + + * res_pjsip: incorrect qualify statistics after disabling for contact + + When removing the qualify_frequency from an AoR or a contact the statistics + shown when issuing "pjsip show aors" from the CLI are incorrect. This patch + deletes the contact's status object from sorcery, disassociating it from the + contact, if the qualify_freqency is removed from configuration. + + ASTERISK-24462 #close + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4116/ + ........ + + Merged revisions 426755 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426757 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426761 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 04:21 +0000 [5d8d90c402] Walter Doekes + + * app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE. + + In update_messages_by_imapuser(), messages were appended to a finite + array which resulted in a crash when an IMAP mailbox contained more + than 256 entries. This memory is now dynamically increased as needed. + + Observe that this patch adds a bunch of XXX's to questionable code. See + the review (url below) for more information. + + ASTERISK-24190 #close + Reported by: Nick Adams + Tested by: Nick Adams + + Review: https://reviewboard.asterisk.org/r/4126/ + ........ + + Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426692 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426696 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426702 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-30 01:15 +0000 [c866ced76b] Igor Goncharovskiy + + * + Add additional checks for NULL pointers to fix several crashes reported. + + ASTERISK-24304 #close + Reported by: dhanapathy sathya + ........ + + Merged revisions 426666 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426667 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426668 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-29 20:59 +0000 [0ddc3bde24] Olle Johansson (License 5267) + + * channels/chan_sip: Add improved support for 4xx error codes + + This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER + response handling. This helps interoperability in a number of scenarios. + + Review: https://reviewboard.asterisk.org/r/3437 + + patches: + rb3437.patch uploaded by oej (License 5267) + ........ + + Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426600 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426601 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426602 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-29 20:48 +0000 [ff83ff564c] Olle Johansson (License 5267) + + * channels/chan_sip: Support mutltiple Supported and Required headers + + A SIP request may contain multiple Supported: and Required: headers. Currently, + chan_sip only parses the first Supported/Required header it finds. This patch + adds support for multiple Supported/Required headers for INVITE requests. + + Review: https://reviewboard.asterisk.org/r/2478 + + ASTERISK-21721 #close + Reported by: Olle Johansson + patches: + rb2478.patch uploaded by oej (License 5267) + ........ + + Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426595 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426596 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426597 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-29 08:02 +0000 [8a69aedd17] Tzafrir Cohen + + * Fix building chan_phone on big endian systems + + A left over from the formats conversion (Corey Farrell). + + ASTERISK-24458 #close + Review: https://reviewboard.asterisk.org/r/4117/ + + ........ + + Merged revisions 426570 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 16:35 +0000 [0ed8aebda9] Richard Mudgett + + * bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config(). + + The feature_automonitor() and feature_automixmonitor() functions were not + locking the channel around ast_get_chan_features_general_config(). + Accessing the channel datastore list without the channel locked is a good + way to corrupt the list or follow the pointer chain into oblivion. + ........ + + Merged revisions 426531 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426552 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 16:10 +0000 [7205d76d7d] Corey Farrell + + * res_fax: Resolve T38 gateway frame leak. + + When frames are translated by a fax gateway they need to be freed. The + existing call to ast_frfree was unreachable. This change reorganizes + fax_gateway_framehook to ensure that ast_frfree is called when needed. + + ASTERISK-24457 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4115/ + ........ + + Merged revisions 426527 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426528 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426529 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 15:44 +0000 [67e496c275] Corey Farrell + + * manager: Unsubscribe from acl_change_sub at shutdown. + + ASTERISK-24453 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4110/ + ........ + + Merged revisions 426524 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426525 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 13:09 +0000 [1fe22c411d] Malcolm Davenport + + * ASTERISK-23512, correct inaccurate comment in manager.conf.sample + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 11:41 +0000 [8e9f593e3a] Matt Jordan + + * main/bridge: Destroy features struct on off nominal path during bridge impart + + When a channel is imparted to a bridge, the invocation of the function may + provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, + the caller must assume that ownership has passed to the function, as in all + paths the function destroys the struct prior to returning (as its purpose is + to configure the behavior of the channel while in the bridge). On one off + nominal path - where the channel already has a PBX thread - the struct was not + being destroyed. + + This patch fixes that glitch. + + ASTERISK-24437 #close + Reported by: Scott Griepentrog + ........ + + Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426432 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426433 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 09:59 +0000 [f4b4d42630] Matt Jordan + + * main/manager: Fix typo in AMI event documentation of "OriginateResponse" + + The parameter name is "Response", not "Resonse". + + ASTERISK-24430 #close + Reported by: Dafi Ni + ........ + + Merged revisions 426366 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426367 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426368 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 09:57 +0000 [68d9872f58] Malcolm Davenport + + * ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 08:13 +0000 [684b8762a9] Malcolm Davenport + + * ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 06:22 +0000 [2290393273] Corey Farrell + + * app_queue: Cleanup ao2_iterator + + Clean ao2_iterator, resolving reference leak to queue members. + + ASTERISK-24454 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4111/ + ........ + + Merged revisions 426255 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426260 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426266 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-28 06:12 +0000 [ab16f46139] Corey Farrell + + * func_cdr: Fix CDR_PROP payload leak + + Remove duplicate allocation of payload, preventing leak. + + ASTERISK-24455 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4113/ + ........ + + Merged revisions 426252 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-27 12:55 +0000 [ef8cdd40e5] Sean Bright + + * configure: Add autoconf check for libopus. + + Because opus transcoding support cannot be included in the standard Asterisk + distribution, a few codec_opus implementations have popped up. To make it + easier for people to drop in opus support in their own installations, this + patch adds configure checks for libopus. + + Review: https://reviewboard.asterisk.org/r/4106/ + ........ + + Merged revisions 426234 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-26 21:47 +0000 [5a17878085] Matt Jordan + + * res/res_http_websocket: Fix minor nits found by wdoekes on r409681 + + When Moises committed the fixes for WSS (which was a great patch), wdoekes had + a few style nits that were on the review that got missed. This patch resolves + what I *think* were all of the ones that were still on the review. + + Thanks to both moy for the patch, and wdoekes for the reviews. + + Review: https://reviewboard.asterisk.org/r/3248/ + ........ + + Merged revisions 426209 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426210 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426211 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-26 21:27 +0000 [62bee9b327] Matt Jordan + + * res/res_phoneprov: Fix crash on shutdown caused by container cleanup + + In res_phoneprov, unloading the module first destroys the http_routes + container, followed by the users. However, users may have a route in + the http_routes container; the validity of this container is not checked + in the users destructor. Hence, we hit an assert as the container has already + been set to NULL. + + This patch does two things: + (1) It adds a sanity check in the user destructor (because why not) + (2) It switches the order of destruction, so that users are disposed of prior + to the HTTP routes they may hold a reference to. + + Note that this crash was caught by the Test Suite (go go testing!) + ........ + + Merged revisions 426174 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426176 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426179 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-26 20:47 +0000 [130a3fcd7f] Matt Jordan + + * res/res_srtp: Fix include issue for libsrtp 1.5.0 + + In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including + srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this + header file has been provided by the library since 2006, so this is a + relatively benign change. + + ASTERISK-24436 #close + Reported by: Patrick Laimbock + ........ + + Merged revisions 426140 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 426141 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 426142 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 426143 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-24 10:32 +0000 [c084728690] Jonathan Rose + + * Documentation: Improve documentation for ExtensionStatus AMI events + + Review: https://reviewboard.asterisk.org/r/4085/ + ........ + + Merged revisions 426120 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-22 16:41 +0000 [c4d7e7e270] Shaun Ruffell + + * codec_dahdi: Cannot use struct ast_translator.core_{src,src}_codec. + + This fixes a Segmentation fault introduced in r419044 "media formats: re-architect + handling of media for performance improvements". + + The problem is that codec_dahdi was using core_src_codec and core_dst_codec in the + ast_translator structure when these fields were never set. Now instead of trying to map + the new core codec descriptions to the way DAHDI defines different codecs, we will store + the DAHDI specific formats in 'struct translator' directly so we can refer to them without + mapping. + + This also allows us to remove the "global_format_map" structure, since we can now query + the list of translators directly to make sure we do not ever register a DAHDI based + translator for a specific path more than once and eliminate the need to keep the list and + the map in sync. + + ASTERISK-24435 #close + Reported by: Marian Koniuszko + + Review: https://reviewboard.asterisk.org/r/4105/ + ........ + + Merged revisions 426097 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-21 13:04 +0000 [2165868be7] Richard Mudgett + + * translage.c: Fix regression when generating translation path strings. + + Fix the AMI Status action read and write translation path strings from + growing for each channel in the status event list by reseting the ast + string given to ast_translate_path_to_str() to fill in the given + translation path. + ........ + + Merged revisions 426079 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426080 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-20 09:20 +0000 [dad0334cf1] abelbeck (License 5903),Matt Jordan (License 6283) + + * AST-2014-011: Fix POODLE security issues + + There are two aspects to the vulnerability: + (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use + TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the + TCP/TLS core, which should be done as an improvement at a latter date. + (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, + will default to the OpenSSL SSLv23_method. This method allows for all + ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by + forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. + This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, + and explicitly disables SSLv2/SSLv3 if using SSLv23_method. + + For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is + explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or + SSLv3. + + Much thanks to abelbeck for reporting the vulnerability and providing a patch + for the res_jabber/res_xmpp modules. + + Review: https://reviewboard.asterisk.org/r/4096/ + + ASTERISK-24425 #close + Reported by: abelbeck + Tested by: abelbeck, opsmonitor, gtjoseph + patches: + asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) + asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) + AST-2014-011-1.8.diff uploaded by mjordan (License 6283) + AST-2014-011-11.diff uploaded by mjordan (License 6283) + ........ + + Merged revisions 425987 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425991 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-19 12:09 +0000 [5e10e369b1] gtjoseph + + * build: Force -fsigned-char on platforms where the default for char is unsigned + + gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and + SPARC default to 'signed char'. This is only an issue in the rare cases where + negative values are assigned to a 'char' but this this patch insures + compatibility by detecting platforms that default to 'unsigned' and adding an + '-fsigned-char' flag to _ASTCFLAGS. + + If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh + and ./configure to regenerate the build files. You shouldn't have to do this + for Intel or SPARC. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4091/ + ........ + + Merged revisions 425964 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425965 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425966 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-18 23:03 +0000 [404b6ab3ab] Matt Jordan + + * res/res_pjsip_sdp_rtp: Revert 425924 + + This patch for r425924 introduced a bug, wherein sending an INVITE request + with no SDP would cause Asterisk to not send an SDP Offer in the 200 + OK. The current structure of res_pjsip_sdp_rtp is a bit hard to deal with + to fix this, as create_outgoing_sdp has no knowledge of whether or not it is + creating an SDP as a new Offer or an Answer. This is something of an oversight + in the callback definition, as the caller of it does have this information. + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-18 19:56 +0000 [b263c8bdae] Matt Jordan + + * res/res_pjsip_sdp_rtp: Remove left over reference to override_prefs + + The usage of the local override_prefs variable in create_outgoing_sdp_stream + was previously to track an override format preference set by PJSIP_MEDIA_OFFER. + Now, however, that function simply sets the joint capabilities structure, + session->req_caps. During the media format rework, the override_prefs was + instead used to check if there were any formats in session->req_caps. + + However, this usage isn't useful in create_outgoing_sdp_stream. + session->req_caps contains the negotiated formats for *all* streams, not just + the current one being created. Thus, so long as any stream of any type has + provided a format, override_prefs will be non-zero. Hence, its usage in + checking whether or not we should look at the formats on the endpoint or + the joint capabilities is generally useless. + + There's only two things useful to check: + (1) Does the endpoint have a format for the media type? + (2) Did we negotiate a format for the media type? + + If either of those is a 'no', then we must kill the media stream. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-17 17:45 +0000 [b8f687f27c] Jonathan Rose + + * Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modules + + AST-1432 #close + Reported by: John Bigelow + ........ + + Merged revisions 425905 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-17 08:35 +0000 [8f58592252] Matt Jordan + + * res_pjsip_session/res_pjsip_sdp_rtp: Be more tolerant of offers + + When an inbound SDP offer is received, Asterisk currently makes a few + incorrection assumptions: + + (1) If the offer contains more than a single audio/video stream, Asterisk will + reject the entire stream with a 488. This is an overly strict response; + generally, Asterisk should accept the media streams that it can accept and + decline the others. + (2) If the offer contains a declined media stream, Asterisk will attempt to + process it anyway. This can result in attempting to match format + capabilities on a declined media stream, leading to a 488. Asterisk should + simply ignore declined media streams. + (3) Asterisk will currently attempt to handle offers with AVPF with + use_avpf=No/AVP with use_avpf=Yes. This mismatch results in invalid SDP + answers being sent in response. If there is a mismatch between the media + type being offered and the configuration, Asterisk must reject the offer + with a 488. + + This patch does the following: + * Asterisk will accept SDP offers with at least one media stream that it can + use. Some WARNING messages have been dropped to NOTICEs as a result. + * Asterisk will not accept an offer with a media type that doesn't match its + configuration. + * Asterisk will ignore declined media streams properly. + + #SIPit31 + + Review: https://reviewboard.asterisk.org/r/4063/ + + ASTERISK-24122 #close + Reported by: James Van Vleet + + ASTERISK-24381 #close + Reported by: Matt Jordan + ........ + + Merged revisions 425868 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425879 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-17 08:17 +0000 [0d0e38a0e1] Joshua Colp + + * res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports. + + This change adds a module which is configurable using the keep_alive_interval setting in the + global section that will send a CRLF keep alive to all active connection-oriented transports at + the provided interval. This is useful because it can help keep connections open through NATs. + This functionality also exists within PJSIP but can not be controlled at runtime and requires + recompiling it. + + Review: https://reviewboard.asterisk.org/r/4084/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425825 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-17 08:11 +0000 [86eea19c8f] Damian Ivereigh (License 6632) + + * channels/chan_sip: Respect outboundproxy setting when sending qualify requests + + The outboundproxy setting is currently ignored when sending OPTIONS requests + as a result of the qualify setting. This means that if an Asterisk server is + unable to send the packet directly to a peer, it is unable to qualify any + non-inbound registered peer (e.g. a peer SIP Trunk). + + This patch grabs the outboundproxy information for a peer when a qualify + attempt is being constructed and, if it finds the information, uses it + when sending the OPTIONS request. + + Review: https://reviewboard.asterisk.org/r/3948 + + ASTERISK-24063 #close + Reported by: Damian Ivereigh + patches: + outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632) + ........ + + Merged revisions 425818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425819 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425820 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425821 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425822 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-17 06:30 +0000 [7144c739e9] Joshua Colp + + * res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable. + + This change adds a configuration option which adds a 'user=phone' parameter if the user + portion of the request URI or the From URI is determined to be a number. + + Review: https://reviewboard.asterisk.org/r/4073/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 21:49 +0000 [f91cb1207c] Richard Mudgett + + * AMI: Add missing VarSet events when a channel inherits variables. + + There should be AMI VarSet events when channel variables are inherited by + an outgoing channel. Also local;2 should generate VarSet events when it + gets all of its channel variables from channel local;1. + + ASTERISK-24415 #close + Reported by: Richard Mudgett + Patches: + jira_asterisk_24415_v12.patch (license #5621) patch uploaded by Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4074/ + ........ + + Merged revisions 425782 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425783 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 21:01 +0000 [df59a71b83] Matt Jordan + + * bridge_native_rtp: Fix audio issues when moving from remote bridge to softmix + + When a native RTP bridge that is remotely bridging its participants switches + to a softmix bridge, it may not properly re-INVITE the media for one or both + participants back to Asterisk. This is due to the current bridge_native_rtp + code only re-INVITEs if it believes the channel will survive the bridge + operation. Currently, that code is failing, as it expects the channels to + have a soft hangup flag set on it indicating that a redirect has occurred + or that the channel is going to leave the bridge. (The code did not take into + account a smart bridge operation). + + This patch also renames a few things to be more reflective of the underlying + types. + + Review: https://reviewboard.asterisk.org/r/3997/ + + ASTERISK-24327 #close + ........ + + Merged revisions 425760 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425761 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 20:46 +0000 [2ccbdd2624] Matt Jordan + + * test_cel: Update pickup test to expect CANCEL instead of ANSWSER + + The CEL pickup test previously looked for a disposition of ANSWER between the + original caller/peer when the call is picked up. This is actually incorrect: + the disposition should, at the very least, not be ANSWER as the call was + never ANSWERed. The disposition is now CANCEL; this patch updates the test + accordingly. + ........ + + Merged revisions 425757 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425758 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 16:21 +0000 [873d956144] Matt Jordan + + * main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size' + + When refactoring CDRs to use the configuration framework, a 'whoops' was + introduced where the CDR batch size was used when rescheduling a batch, + as opposed to the time duration. This patch corrects that obvious mistake. + + ASTERISK-24426 #close + Reported by: Shane Blaser + ........ + + Merged revisions 425735 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425736 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 12:32 +0000 [c2ec5f0f6f] gtjoseph + + * config: Fix inf loop using ast_category_browse and ast_variable_retrieve + + Fix infinite loop when calling ast_variable_retrieve inside an + ast_category_browse loop when there is more than 1 category with + the same name. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4089/ + ........ + + Merged revisions 425713 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425714 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425715 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 11:32 +0000 [86a4ce4957] Kinsey Moore + + * PJSIP: Enforce module load dependencies + + This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub + have loaded properly before attempting to load any modules that depend + on them since the module loader system is not currently capable of + resolving module dependencies on its own. + + ASTERISK-24312 #close + Reported by: Dafi Ni + Review: https://reviewboard.asterisk.org/r/4062/ + ........ + + Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-16 01:22 +0000 [a770ca168d] Igor Goncharovskiy + + * + Fix loss of voice after second call drops (on a second line) in case using multiple lines on unistim phones. There is regression was introduced in r391379. + + Reported by: Rustam Khankishyiev + (closes issue ASTERISK-23846) + ........ + + Merged revisions 425667 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425668 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425669 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-15 20:26 +0000 [bfee1b4bc5] Joshua Colp + + * res_rtp_asterisk: Fix a bug where ICE state would get reset when it shouldn't. + + In the case where the ICE negotiation had not yet started current state would + get wiped when it shouldn't. + + This also removes channel binding as in practice this does not work well with + other implementations. + ........ + + Merged revisions 425644 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425645 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425646 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-15 14:39 +0000 [28c11fff78] Richard Mudgett + + * chan_motif: Cleanup jingle_tech.capabilities only once. + ........ + + Merged revisions 425627 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-15 14:17 +0000 [3d58066de9] Jonathan Rose + + * parking_tests: Fix assertions and possibly crashes in res_parking unit tests + + Assertions were caused by attempting to play music on hold to a channel with + no formats. Parking unit test channels were given formats and a technology so + that they would be able to pretend to read/write frames. + + ASTERISK-24413 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/4075/ + ........ + + Merged revisions 425611 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425613 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-15 05:03 +0000 [90c98d384b] Alexandr Anikin + + * chan_ooh323: fix rtptimeout general value checking + + correct condition to check rtptimeout in [general] config section + + ASTERISK-24393 #close + Reported by: Dmitry Melekhov + Tested by: Dmitry Melekhov + Patches: + ASTERISK-24393.patch + ........ + + Merged revisions 425547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425548 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425589 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425590 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-14 15:48 +0000 [104fca5001] gtjoseph + + * config: Fix SEGV in unit test with MALLOC_DEBUG + + With MALLOC_DEBUG the /main/config config_basic_ops test was causing a + SEGV while doing an ast_category_delete in an ast_category_browse loop. + Apparently this never worked but was also never tested. I removed the + test, added 2 notes to config.h indicating that it's not supported and + added a few lines of code to ast_category_delete to prevent the SEGV + should someone attempt it in the future. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4078/ + ........ + + Merged revisions 425525 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425526 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-14 14:12 +0000 [87b5006ff0] Jonathan Rose + + * Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute + + Tasks that were marked for pending deletion in the scheduler would be moved to + the cache for later reuse, but after being recycled the deleted mark wouldn't + be removed resulting in fresh tasks being deleted without reason... and + immediately moved back into the cache where they could be reused again. This + could cause horrendous things to happen in just about anything that used a + scheduler. + + ASTERISK-24321 #close + Reported by: Steve Pitts + Review: https://reviewboard.asterisk.org/r/4071/ + ........ + + Merged revisions 425503 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425504 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-14 13:13 +0000 [527b58aeb7] gtjoseph + + * res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup + + Based on feedback from Richard, I created an accessor for + res_phoneprov/ast_phoneprov_std_variable_lookup and added + load priority to AST_MODULE_INFO. + + Tested-by: George Joseph + Tested-by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4076/ + ........ + + Merged revisions 425480 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425481 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-14 11:47 +0000 [fbb19db0c8] Corey Farrell + + * res_fax: Fix reference leak caused by gateway sessions + + Fax gateway session objects can be re-used, causing the + same gateway session to be added to faxregistry.container + more than once. This change causes fax_session_new to + remove the reserved session from the container before + it's id is changed, ensuring it's possible for the + session to be freed. + + ASTERISK-24392 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4049/ + ........ + + Merged revisions 425457 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425458 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425459 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-14 11:43 +0000 [c61b66e107] Richard Mudgett + + * stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2) + + Masquerades into and out of channels that are involved in a dial operation + don't create the expected dial end event. The missing dial end event goes + against the model for things like CDRs and generating Dial end manager + actions and such. + + There are four cases: + + 1) A channel masquerades into the caller channel. The case happens when + performing a blonde transfer using the channel driver's protocol. + + 2) A channel masquerades into a callee channel. The case happens when + performing a directed call pickup. + + 3) The caller channel masquerades out of dial. The case happens when + using the Bridge application on the caller channel. + + 4) A callee channel masquerades out of dial. The case happens when using + the Bridge application on a peer channel. + + As it turned out, all four cases need to be handled instead of just the + first one. + + ASTERISK-24237 + Reported by: Richard Mudgett + + ASTERISK-24394 #close + Reported by: Richard Mudgett + + Review: https://reviewboard.asterisk.org/r/4066/ + ........ + + Merged revisions 425430 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425455 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425456 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-14 11:20 +0000 [01bdc80475] Corey Farrell + + * res_fax: Resolve module reference leak caused by reserved sessions + + Remove reference to module providing reserved session after + adding a reference to the final module. This re-reference + is done to ensure that module references are correct even + if the final session selects a different module than the + reserved session. + + ASTERISK-18923 #close + Reported by: Grigoriy Puzankin + Review: https://reviewboard.asterisk.org/r/4048/ + ........ + + Merged revisions 425405 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425407 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425411 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425415 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-13 11:12 +0000 [c7e6b6ba3d] gtjoseph + + * manager/config: Support templates and non-unique category names via AMI + + This patch provides the capability to manipulate templates and categories + with non-unique names via AMI. + + Summary of changes: + + GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list + of name_regex=value_regex expressions which will cause only categories whose + variables match all expressions to be considered. The special variable name + TEMPLATES can be used to control whether templates are included. Passing + 'include' as the value will include templates along with normal categories. + Passing 'restrict' as the value will restrict the operation to ONLY templates. + Not specifying a TEMPLATES expression results in the current default behavior + which is to not include templates. + + UpdateConfig: NewCat now includes options for allowing duplicate category + names, indicating if the category should be created as a template, and + specifying templates the category should inherit from. The rest of the + actions now accept a filter string as defined above. If there are non-unique + category names, you can now update specific ones based on variable values. + + To facilitate the new capabilities in manager, corresponding changes had to be + made to config, most notably the addition of filter criteria to many of the + APIs. In some cases it was easy to change the references to use the new + prototype but others would have required touching too many files for this + patch so a wrapper with the original prototype was created. Macros couldn't + be used in this case because it would break binary compatibility with modules + such as res_digium_phone that are linked to real symbols. + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4033/ + ........ + + Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-12 16:09 +0000 [8d6f1d763c] Joshua Colp + + * res_rtp_asterisk: Make the ICE transport check case insensitive as some implementations use 'udp'. + ........ + + Merged revisions 425360 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425361 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425362 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-12 03:17 +0000 [9e72c74db5] Walter Doekes + + * chan_sip: Fix so asterisk won't send reINVITE after a BYE. + + After a reINVITE glare situation, Asterisk would re-send the reINVITE + even though the call had been hung up in the mean time. This patch + unschedules the reinvite when handling the BYE. + + ASTERISK-22791 #close + Reported by: Paolo Compagnini + Tested by: Paolo Compagnini + + Review: https://reviewboard.asterisk.org/r/4056/ + (testcase is in review r4055) + ........ + + Merged revisions 425296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425297 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425298 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425299 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-12 02:57 +0000 [c0ac874106] Walter Doekes + + * build: Relax badshell tilde test to allow for ~ in middle of DESTDIR. + + The main Makefile has a target test called 'badshell' that tests if + DESTDIR does not happen to have an an-expanded tilde (~). This might + be the case if you run: make install DESTDIR=~/somewhere/ + + That test also disallowed valid tildes in directory names. The test is + now changed to only trigger on a tilde at the start of the path. + + ASTERISK-13797 #close + Reported by: Tzafrir Cohen + + Review: https://reviewboard.asterisk.org/r/4064/ + ........ + + Merged revisions 425291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425292 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425293 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425294 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-12 02:47 +0000 [2a03efdbae] Walter Doekes + + * res_calendar_ews: Relax neon version check to work with 0.30 too. + + Allow res_calendar_ews to work not only with libneon-0.29 but also + with 0.30. + + ASTERISK-24325 #close + Reported by: Tzafrir Cohen + + Review: https://reviewboard.asterisk.org/r/4068/ + ........ + + Merged revisions 425286 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425287 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425288 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425289 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-11 16:09 +0000 [6a3c11c75b] gtjoseph + + * res_phoneprov: Cleanup module load error handling + + Tested module load/reload interaction between res_phoneprov and + res_pjsip_phoneprov_provider in cases where res_phoneprov didn't + load correctly (usually misconfiguration or missing phoneprov.conf) + + Tested-by: George Joseph + + Review: https://reviewboard.asterisk.org/r/4069/ + ........ + + Merged revisions 425264 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425265 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-10 15:48 +0000 [98d5b7090d] Joshua Colp + + * bridge: During a smart bridge operation provide a more complete bridge to the old technology. + + When a smart bridge operation occurs and a bridge transitions from one + technology to another the old technology is provided the channels formerly + in it and told that they are leaving. Unfortunately the bridge provided + along with them is incomplete. The bridge, despite there being channels in it, + contains none. This forces technology implementations to have additional + logic when channels are leaving or to store their own duplicated + state. + + This change makes the bridge more complete so it contains the expected + channels. Now that the bridge is complete special logic within + bridge_native_rtp is no longer needed and has been removed. + + Review: https://reviewboard.asterisk.org/r/4057/ + ........ + + Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425243 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-10 09:31 +0000 [c3ff212cae] Matt Jordan + + * res/res_phoneprov: Bail on registration if res_phoneprov didn't load + + If res_phoneprov failed to fully load (due to not being configured), the + providers container will be NULL. If a module attempts to register a phone + provisioning provider, it should check for the presence of the container. + If there is no providers container, it should return an error. + + This patch makes the ast_phoneprov_provider_register function do that... + otherwise this would be a silly commit message. + ........ + + Merged revisions 425220 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425221 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-10 09:24 +0000 [c46100ad5f] Joshua Colp + + * res_pjsip_phoneprov_provider: Add missing dependency on pjproject. + ........ + + Merged revisions 425216 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425217 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-10 08:03 +0000 [37b5f52da7] Kinsey Moore + + * CallerID: Fix parsing regression + + This fixes a regression in callerid parsing introduced when another bug + was fixed. This bug occurred when the name was composed entirely of + DTMF keys and quoted without a number section (<>). + + ASTERISK-24406 #close + Reported by: Etienne Lessard + Tested by: Etienne Lessard + Patches: + callerid_fix.diff uploaded by Kinsey Moore + Review: https://reviewboard.asterisk.org/r/4067/ + ........ + + Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425153 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425154 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425155 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-10 07:10 +0000 [0ef680cff0] Joshua Colp + + * res_pjsip_nat: Place source port into rport of responses if 'force_rport' is on. + + When the 'force_rport' option is enabled the behavior should be the same + as if the remote side placed rport into the message themselves. Therefore + any responses we send should include the source port of the request in the + rport of the Via header. + + #SIPit31 + + ASTERISK-24387 #close + Reported by: Matt Jordan + ........ + + Merged revisions 425131 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425132 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-10 02:34 +0000 [d3f525fd8f] Torrey Searle (License #5334),Nitesh Bansal (License #6418) + + * chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE. + + If a device re-INVITEs at the same time as the dialog is hung up, and + if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would + fail to destroy the dialog after a while. This resulted in (most + prominently) file handle leaks. + + (Patch reindented by me.) + + ASTERISK-20784 #close + ASTERISK-15879 #close + Reported by: Torrey Searle, Nitesh Bansal + Patches: + reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334) + patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418) + + Reviewboard: https://reviewboard.asterisk.org/r/4052/ + (testcase can be found at r4051) + ........ + + Merged revisions 425068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 425069 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 425070 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425071 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 18:37 +0000 [aef63118da] gtjoseph + + * res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTOR + + endpoint->inbound_auths was changed to a vector in 13 and I + committed the 12 patch instead of the 13 patch. + + Tested-by: George Joseph + ........ + + Merged revisions 425052 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 16:39 +0000 [6fc4df7279] Kevin Harwell + + * res_rtp_asterisk: Crash if no candidates received for component + + When starting ice if there is not at least one remote ice candidate with an RTP + component asterisk will crash. This is due to an assertion in pjnath as it + expects at least one candidate with an RTP component. Added a check to make + sure at least one candidate contains an RTP component and at least one candidate + has an RTCP component. + + ASTERISK-24383 #close + Review: https://reviewboard.asterisk.org/r/4039/ + ........ + + Merged revisions 425031 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 15:55 +0000 [c6837c236f] gtjoseph + + * res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov + + This module allows res_pjsip to integrate with res_phoneprov. It handles + the pjsip 'phoneprov' object type. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/3976/ + ........ + + Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 425008 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 13:44 +0000 [3a187aa14a] Matt Jordan + + * res/res_phoneprov: Don't cancel Asterisk load on module load failure + ........ + + Merged revisions 424985 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424986 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 12:46 +0000 [cc595f7353] gtjoseph + + * res_phoneprov: Refactor phoneprov to allow pluggable config providers + + This patch makes res_phoneprov more modular so other modules (like pjsip) + can provide configuration information instead of res_phoneprov relying solely + on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API + is now exposed which allows config providers to register themselves, set + defaults (server profile, etc) and add user extensions. + + * ast_phoneprov_provider_register registers the provider and provides callbacks + for loading default settings and loading users. + * ast_phoneprov_provider_unregister clears the defaults and users. + * ast_phoneprov_add_extension should be called once for each user/extension + by the provider's load_users callback to add them. + * ast_phoneprov_delete_extension deletes one extension. + * ast_phoneprov_delete_extensions deletes all extensions for the provider. + + Tested-by: George Joseph + Review: https://reviewboard.asterisk.org/r/3970/ + ........ + + Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 11:38 +0000 [0f50e8856b] Richard Mudgett + + * cdr.c: Make turning on CDR debug a one step process instead of two. + + Now "cdr set debug on" doesn't also require "core set verbose 1" to see + CDR debug output. + ........ + + Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424942 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424943 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-09 03:10 +0000 [d0255c4a46] Michael Myles (License #6626) + + * safe_asterisk: Don't automatically exceed MAXFILES value of 2^20. + + On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided + by two can exceed the per-process file limit of 2^20. This patch + ensures the value is capped. + + (Patch cleaned up by me.) + + ASTERISK-24011 #close + Reported by: Michael Myles + Patches: + safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626) + ........ + + Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 424878 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424879 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424880 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-08 13:47 +0000 [8b0089ea1d] Joshua Colp + + * res_rtp_asterisk: Allow only UDP ICE candidates. + + The underlying library, pjnath, that res_rtp_asterisk uses for ICE + support does not have support for ICE-TCP. As candidates are + passed through directly to it this can cause error messages to occur + when it receives something unexpected (such as a TCP candidate). + This change merely ignores all non-UDP candidates so they never + reach pjnath. + + ASTERISK-24326 #close + Reported by: Joshua Colp + ........ + + Merged revisions 424852 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424853 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424854 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-08 13:24 +0000 [5e50638539] Kinsey Moore + + * Stasis: Relegate log message to dev-mode + + This error message primarily applies to development tasks and will now + only show up when dev-mode is enabled via configure. + ........ + + Merged revisions 424850 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-08 09:54 +0000 [3dfc485e35] Kinsey Moore + + * Indexer: Format message types may not exist + + In Asterisk 13+, any given message type is not guaranteed to exist even + if Asterisk comes up correctly since creation of the message type could + be declined. The indexer should not prevent Asterisk from starting + under these conditions. + ........ + + Merged revisions 424833 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-07 15:33 +0000 [d8bbf1ec1d] Kinsey Moore + + * Stasis: Only log errors for non-declined types + + When message type creation is declined via stasis.conf, certain + operations log errors assuming that the declined type is being used + before initialization or after destruction. These error messages get + quite spammy for oft used message types and should not be logged in the + first place since the message type is validly NULL. + + Reported by: Matt DiMeo + ........ + + Merged revisions 424769 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-07 13:34 +0000 [f7225da08a] Joshua Colp + + * data: Properly access formats in capabilities structure when adding codecs. + + Formats within a capabilities structure are addressed starting at 0, not 1. + Assuming 1 causes it to exceed an array. + + ASTERISK-24389 #close + Reported by: Kevin Harwell + ........ + + Merged revisions 424752 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-07 12:44 +0000 [a9011106b6] Matt Jordan + + * res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter + + Prior to this patch, the auth_reject_permanent parameter was not initialized on + the registration client state, leading to the parameter being disabled + regardless of the value specified in pjsip.conf. + + This patch initialized the setting on the registration client state to the + provided configuration value. + + ASTERISK-24398 #close + ........ + + Merged revisions 424730 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424731 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424732 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-07 09:09 +0000 [523da7d1b3] Matt Jordan + + * res/res_pjsip_pubsub: Fix typo in WARNING message + ........ + + Merged revisions 424713 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424714 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-06 13:39 +0000 [39bd5b7a70] Peter Katzmann (License 5968) + + * message: Don't close an AMI connection on SendMessage action error + + If SendMessage encounters an error (such as incorrect input provided to the + action), it will currently return -1. Actions should only return -1 if the + connection to the AMI client should be closed. In this case, SendMessage + causing the client to disconnect is inappropriate. + + This patch causes the action to return 0, which simply causes the action to + fail. + + Review: https://reviewboard.asterisk.org/r/4024 + + ASTERISK-24354 #close + Reported by: Peter Katzmann + patches: + sendMessage.patch uploaded by Peter Katzmann (License 5968) + ........ + + Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424692 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-06 10:41 +0000 [c384532aa4] Richard Mudgett + + * features.c: Fix lingering channel ref while Bridge() application is active. + + Using the Bridge application to bridge a channel that is executing an + applicaiton such as Wait results in a lingering Surrogate channel in the + CLI "core show channels" output even though it has already hungup. + + * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has + been put into the bridge. + + * Eliminated bridge_exec()'s use of RAII_VAR(). + + ASTERISK-24224 #close + Reported by: Mark Michelson + + Review: https://reviewboard.asterisk.org/r/4041/ + ........ + + Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424669 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-06 07:39 +0000 [3a87f32dc0] Matt Jordan + + * sdp_srtp: Add new lines to some WARNING messages + ........ + + Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424647 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-05 20:01 +0000 [cce3d99ec8] Matt Jordan + + * res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint + + An OPTIONS request that is sent to Asterisk but not to a specific endpoint is + currently sent a 404 in response. This is because, not surprisingly, an empty + extension is never going to be found in the dialplan. + + This patch makes it so that we only attempt to look up the endpoint in the + dialplan if it is specified in the OPTIONS request URI. + + #SIPit31 + + ASTERISK-24370 #close + Reported by: Matt Jordan + ........ + + Merged revisions 424624 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424625 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424626 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-05 19:53 +0000 [c013916869] Matt Jordan + + * pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels + + Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health. + It will treat the channels as a PJSIP channel, eventually hitting an ao2 error, + FRACKing on assertion error, and quite likely crashing. + + This patch adds checks to the read/write callbacks that ensure that the channel + technology is of type 'PJSIP' before attempting to operate on the channel. + + #SIPit31 + + ASTERISK-24382 #close + Reported by: Matt Jordan + ........ + + Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424622 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-05 19:31 +0000 [45b7b474ac] Matt Jordan + + * res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message + + When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable + transport, it is possible (although it shouldn't occur) for pjproject to pass + up an rdata object with a NULL msg in the msg_info. Needless to say, things + that attempt to dereference this are in for a rough ride. + + In particular, this caused crashes in three different locations, all of which + are 'low level' enough to intercept an rdata object early in processing: + + (1) res_pjsip_logger + (2) res_hep_pjsip + (3) res_pjsip/distributor + + Anything that can intercept an rdata object before res_pjsip/distributor should + be defensive when looking at the received packet. + + #SIPit31 + + ASTERISK-24369 #close + Reported by: Matt Jordan + ........ + + Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424619 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-05 19:13 +0000 [f27f41a288] Matt Jordan + + * res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions + + A subscription that has been persisted can - for various reasons - fail to be + re-created on startup. This patch resolves a number of crashes that occurred + when a subscription cannot be re-created on several off-nominal paths. + + #SIPit31 + + ASTERISK-24368 #close + Reported by: Matt Jordan + ........ + + Merged revisions 424601 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-04 19:49 +0000 [9611ef4f1e] Corey Farrell + + * Release AMI connections on shutdown. + + ASTERISK-24378 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4037/ + ........ + + Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424580 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-04 19:15 +0000 [1b0902caa4] Corey Farrell + + * chan_motif: Correct last commit to use ao2_cleanup to free format cap + + This fix applies to 13 and trunk. + + ASTERISK-24384 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4043/ + ........ + + Merged revisions 424554 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-04 19:02 +0000 [0cea12b9e8] Corey Farrell + + * chan_motif: Release format capabilities and config on module load error + + ASTERISK-24384 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4043/ + ........ + + Merged revisions 424550 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424551 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424552 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 16:58 +0000 [24ded9d9eb] Richard Mudgett + + * res_pjsip: Fix XML typo and update CHANGES. + + ASTERISK-24199 + ........ + + Merged revisions 424528 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424529 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 14:42 +0000 [70301b0438] Richard Mudgett + + * audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. + + Adding a mixmonitor to a channel causes the bridge to change technologies + from native to simple_bridge so the call can be recorded. However, when + the mixmonitor is stopped the bridge does not switch back to the native + technology. + + * Added unbridge requests to reevaluate the bridge when a channel + audiohook is removed. + + * Moved the unbridge request into ast_audiohook_attach() ensure that the + bridge reevaluates whenever an audiohook is attached. This simplified the + mixmonitor and chan_spy start code as well. + + * Added defensive code to stop_mixmonitor_full() in case additional + arguments are ever added to the StopMixMonitor application. + + * Made ast_framehook_detach() not do an unbridge request if the framehook + does not exist. + + * Made ast_framehook_list_fixup() do an unbridge request if there are any + framehooks. Also simplified the loop. + + ASTERISK-24195 #close + Reported by: Jonathan Rose + + Review: https://reviewboard.asterisk.org/r/4046/ + ........ + + Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424507 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 13:54 +0000 [cc11a78869] Kristian Hogh + + * app_queue: Add dialplan function to get the channel name at the specified position in a queue. + + The QUEUE_GET_CHANNEL function returns the caller's channel name at the + specified position in a queue. + + QUEUE_GET_CHANNEL([,]) + + The queue position parameter defaults to 1 if not specified. + + Noop(${QUEUE_GET_CHANNEL(queuename, 2)}) + "SIP/peer-00000002", if queue exist and have at least 2 callers + + Noop(${QUEUE_GET_CHANNEL(queuename, 1)}) + Noop(${QUEUE_GET_CHANNEL(queuename)}) + "SIP/peer-00000000", if queue exist and have at least 1 caller + + ASTERISK-24365 #close + Reported by: Kristian Hogh + Patches: + queue_get_firstchannel.patch (license #6639) patch uploaded by Kristian Hogh + rb4035.patch (license #6639) patch uploaded by Kristian Hogh + Patch morphed from QUEUE_GET_FIRSTCHANEL to the more general QUEUE_GET_CHANNEL + on reviewbord. + + Review: https://reviewboard.asterisk.org/r/4035/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 12:47 +0000 [0165c5f95a] Richard Mudgett + + * chan_pjsip: Fix deadlock when masquerading PJSIP channels. + + Performing a directed call pickup resulted in a deadlock when PJSIP + channels were involved. + + A masquerade needs to hold onto the channel locks while it swaps channel + information between the two channels involved in the masquerade. With + PJSIP channels, the fixup routine needed to push a fixup task onto the + PJSIP channel's serializer. Unfortunately, if the serializer was also + processing a task that needed to lock the channel, you get deadlock. + + * Added a new control frame that is used to notify the channels that a + masquerade is about to start and when it has completed. + + * Added the ability to query taskprocessors if the current thread is the + taskprocessor thread. + + * Added the ability to suspend/unsuspend the PJSIP serializer thread so a + masquerade could fixup the PJSIP channel without using the serializer. + + ASTERISK-24356 #close + Reported by: rmudgett + + Review: https://reviewboard.asterisk.org/r/4034/ + ........ + + Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 10:55 +0000 [4967478d18] gtjoseph + + * sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function + + When you call ast_sorcery_create() you don't necessarily know which wizard is + going to be invoked. If it happens to be a wizard like 'config' that doesn't + have a 'create' virtual function you get a segfault in the + sorcery_wizard_create callback. This patch catches the null function pointer, + does an ast_assert, and logs an error. + + Review: https://reviewboard.asterisk.org/r/4044/ + ........ + + Merged revisions 424447 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424448 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 08:59 +0000 [b1f8eba178] Kinsey Moore + + * PJSIP: Restore functional default for callerid_privacy + + The pjsip config option default fixups from r424263 altered the + functional default from "allowed_not_screened" to "allowed". This + change restores the functional default value when none is provided. + ........ + + Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424427 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424428 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-03 08:33 +0000 [4246652603] Kinsey Moore + + * Manager: Add missing fields and documentation for CoreShowChannels + + This corrects some issues introduced in the responses to the + CoreShowChannels AMI command as well as adding documentation for the + responses. The command in Asterisk 12 was missing the following fields: + Duration, Application, ApplicationData, and BridgedChannel and + BridgedUniqueID (replaced with BridgeId). + + ASTERISK-24262 #close + Reported by: Mitch Claborn + Review: https://reviewboard.asterisk.org/r/4040/ + ........ + + Merged revisions 424423 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424424 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-02 16:55 +0000 [2b0777c017] Richard Mudgett + + * res_pjsip: Make transport cipher option accept a comma separated list of cipher names. + + Improvements to the res_pjsip transport cipher option. + + * Made the cipher option accept a comma separated list of OpenSSL cipher + names. Users of realtime will be glad if they have more than one name to + list. + + * Added the CLI command 'pjsip list ciphers' so a user can know what + OpenSSL names are available for the cipher option. + + * Updated the cipher option online XML documentation to specify what is + expected for the value. + + * Updated pjsip.conf.sample to not indicate that ALL is acceptable since + ALL does not imply a preference order for the ciphers and PJSIP does not + simply pass the string to OpenSSL for interpretation. + + ASTERISK-24199 #close + Reported by: Joshua Colp + + Review: https://reviewboard.asterisk.org/r/4018/ + ........ + + Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424394 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424395 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-02 15:23 +0000 [b15cd42b5b] Jonathan Rose + + * Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing' + + The 'outgoing' value was left off of the enumerator when first creating the + column. This patch adds it, and should gracefully upgrade keeping the existing + data in tact. + + ASTERISK-23781 #close + Reported by: Stephen More + Review: https://reviewboard.asterisk.org/r/4013/ + ........ + + Merged revisions 424372 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424373 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-02 10:33 +0000 [2f570094b7] Jonathan Rose + + * chan_pjsip: Fix an assertion for channels that lack formats on creation + + ASTERISK-24222 #close + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4017/ + ........ + + Merged revisions 424333 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-02 08:36 +0000 [aa5458d6ab] Scott Griepentrog + + * res_pjsip: document use of rewrite_contact in sample conf + + Without setting rewrite_contact, an invite to an endpoint + behind NAT will not reach it - unless the endpoint itself + uses STUN or TURN to discover it's public URI. Thus, the + use of this should be in the sample documentation. + + Review: https://reviewboard.asterisk.org/r/4036/ + ........ + + Merged revisions 424337 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424338 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 15:37 +0000 [a752ca00bd] Corey Farrell + + * res_hep: Release allocation reference to configuration. + + ASTERISK-24362 #close + Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4026/ + ........ + + Merged revisions 424312 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424313 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 11:39 +0000 [adba2a8d7f] Joshua Colp + + * res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. + + During the latest update to DTLS-SRTP support the ability to configure + the hash used for fingerprints was added. This gave us two supported ones: + SHA-1 and SHA-256. The default was accordingly updated to SHA-256. + Unfortunately this configuration ability was not exposed within res_pjsip. + This change adds a dtls_fingerprint option that controls it. + + #SIPit31 + ........ + + Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424291 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 11:20 +0000 [9233b1cf44] Joshua Colp + + * res_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session. + + #SIPit31 + ........ + + Merged revisions 424287 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424288 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 07:28 +0000 [4d2c7c23f8] Kinsey Moore + + * PJSIP: Handle defaults properly + + This updates the code behind PJSIP configuration options with custom + handlers to deal with the assigned default values properly where it + makes sense and adjusting the default value where it doesn't. Before + applying this patch, there were several cases where the default value + for an option would prevent that config section from loading properly. + + Reported by: Thomas Thompson + Review: https://reviewboard.asterisk.org/r/4019/ + ........ + + Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424266 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424267 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 07:15 +0000 [122cc050d0] Kinsey Moore + + * PJSIP: Force transport on contact rewrite + + If contact rewriting is enabled but the contact differs in transport + from what is actually being used, messages after the initial INVITE + transaction can be sent to an incorrect transport/port combination. In + the case where this bug occurred the remote party never received a BYE + since it was sent to the remote party's TCP port over UDP. + + Review: https://reviewboard.asterisk.org/r/4032/ + ........ + + Merged revisions 424244 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424245 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424246 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 05:10 +0000 [c3a7524457] ibercom (License #6599) + + * chan_sip: Simplify some unref code by removing unlink_peer_from_tables. + + ASTERISK-22945 #related + Reported by: ibercom + Patches: + asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599) + ........ + + Merged revisions 424181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 424182 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424183 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424184 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-01 04:55 +0000 [841d978a30] ibercom (License #6599) + + * chan_sip: Remove excess ref of realtime peer before sip_poke_peer. + + The peer is referenced at the end of sip_poke_peer, it should not get + an extra ref before the call to sip_poke_peer. This fixes a memory + leak. + + ASTERISK-22945 #close + Reported by: ibercom + Tested by: Yuriy Gorlichenko + Patches: + asterisk11.patch uploaded by ibercom (License #6599) + + Review: https://reviewboard.asterisk.org/r/4031/ + ........ + + Merged revisions 424176 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 424177 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424178 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424179 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-30 06:42 +0000 [d7c29885ad] Joshua Colp + + * res_pjsip_sdp_rtp: Don't place an extra whitespace before 'rport' and don't put IPv6 addresses in brackets. + + #SIPit31 + ........ + + Merged revisions 424155 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424156 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-30 06:36 +0000 [3641ebcf96] Joshua Colp + + * res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP. + + This change fixes an issue where ICE candidates put into the SDP did not contain + the 'raddr' and 'rport' information for server reflexive and relay candidates. + + #SIPit31 + ........ + + Merged revisions 424151 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424152 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424153 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424154 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-29 17:00 +0000 [27396a6b59] gtjoseph + + * pjsip_cli: Suppress header print on error or no objects + + If there's an error on the pjsip command line or there are no objects, don't + print the column headers. + + ASTERISK-24350 #close + Reported-by: Brad Latus + Tested-by: George Joseph + Tested-by: Brad Latus + + Review: https://reviewboard.asterisk.org/r/4025/ + ........ + + Merged revisions 424128 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424129 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-29 16:32 +0000 [b56dfb78c5] Walter Doekes + + * autosupport: Fix bashism. + + '==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a + 'case' works better there. + + Originally committed in r375059 and r375060 on 2012-10-16 21:13:08. + + ASTERISK-20567 #close + Reported by: Tzafrir Cohen + ........ + + Merged revisions 424117 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 424125 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424126 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-29 16:18 +0000 [270932635d] Richard Mudgett + + * Simplify UUID generation in several places. + + Replace code using ast_uuid_generate() with simpler and faster code using + ast_uuid_generate_str(). The new code avoids a malloc(), free(), and + copy. + ........ + + Merged revisions 424103 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424105 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-29 15:28 +0000 [9d2bc0675a] Richard Mudgett + + * threadpool.c: Minor cleanup fixes. + + * Fix threadpool_alloc() prototype. + + * Add missing off-nominal NULL check of pool in threadpool_alloc(). + + * searializer_create() does not need to create the object with a lock as + the lock is not used. + ........ + + Merged revisions 424096 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424097 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-27 12:29 +0000 [2eef53c465] Joshua Colp + + * res_pjsip_session: Reduce SDP size by removing duplicate connection lines. + + Due to the architecture of how media streams are handled each individual + handler adds connection details (IP address) for it. The first media stream + is then used as the top level SDP connection line. In practice each + line ends up being the same so to reduce the SDP size stream-level connection + information is also added to the SDP if it differs from the top level SDP + connection line. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-27 07:44 +0000 [76744543b4] Joshua Colp + + * res_pjsip_session: Add additional checks for delaying session refreshes. + + There are certain situations which no checks existed for which need to prevent + session refreshes. This includes sending a session refresh with SDP before SDP + negotiation has completed and sending a session refresh before the dialog itself + has been established. Checks for these have been added. + + Additionally COLP related UPDATEs were including SDP when it is not needed. + + Review: https://reviewboard.asterisk.org/r/4008/ + ........ + + Merged revisions 424056 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 424057 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-26 10:51 +0000 [3c1804eb0d] Richard Mudgett + + * format_mp3: Made the get script conditionally apply patch if not already there. + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-26 10:43 +0000 [e0abb82ab8] Walter Doekes + + * core: Ouch, forgot to undo a test free() in r423978. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-26 10:28 +0000 [d07b9af24b] Jeremy Laine + + * res_fax: Fix out of bounds error in update_modem_bits(). + + ASTERISK-24357 #close + Reported by: Jeremy Laine + Patches: + res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine + Modified patch to not use magic numbers. + ........ + + Merged revisions 423979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423983 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423987 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423992 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424016 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-26 09:41 +0000 [37179a2b1f] Walter Doekes + + * core: Don't allow free to mean ast_free (and malloc, etc..). + + This gets rid of most old libc free/malloc/realloc and replaces them + with ast_free and friends. When compiling with MALLOC_DEBUG you'll + notice it when you're mistakenly using one of the libc variants. For + the legacy cases you can define WRAP_LIBC_MALLOC before including + asterisk.h. + + Even better would be if the errors were also enabled when compiling + without MALLOC_DEBUG, but that's a slightly more invasive header + file change. + + Those compiling addons/format_mp3 will need to rerun + ./contrib/scripts/get_mp3_source.sh. + + ASTERISK-24348 #related + Review: https://reviewboard.asterisk.org/r/4015/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-26 03:26 +0000 [b8c1130ed1] Jeremy Lainé (License #6561) + + * docs: Escape unescaped minus sign in asterisk.8 manpage. + + ASTERISK-23768 #close + Reported by: Jeremy Lainé + Patches: + escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561) + ........ + + Merged revisions 423915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423916 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423917 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423918 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-25 16:03 +0000 [fa0c33ebc1] Richard Mudgett + + * res_pjsip.c: Add missing off nominal cleanup in ast_sip_push_task_synchronous(). + + * Made memset the std struct in ast_sip_push_task_synchronous() because if + DEBUG_THREADS is enabled then uninitialized lock tracking data is used. + ........ + + Merged revisions 423894 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423895 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-25 15:49 +0000 [d172d84fe1] Kristian Høgh (License #6639) + + * musiconhold: Add preferchannelclass=no option to prefer app class. + + The new option 'preferchannelclass' is added to musiconhold.conf. If yes + (the default) the CHANNEL(musicclass) is preferred when choosing the + hold music. If it is no, the class suggested by the application that + calls the MoH (e.g. the Queue() app) gets preferred (new behaviour). + + This way you set a different hold-music from the Queue-music by setting + both the CHANNEL(musicclass) and the queue-context musicclass. + + ASTERISK-24276 #close + Reported by: Kristian Høgh + Patches: + app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639) + + Review: https://reviewboard.asterisk.org/r/4010/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-24 13:35 +0000 [68077634fe] Richard Mudgett + + * pjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request. + + The crash on the issues is a result of an invalid transport configuration + change when asterisk is restarted. The attempt to send the qualify + request fails and we cleaned up. However, the callback is also called + which results in a double unref of the objects involved. + + * Put a wrapper around pjsip_endpt_send_request() to detect when the + passed in callback is called because of an error so callers can know to + not cleanup. + + * Made send_request_cb() able to handle repeated challenges (Up to 10). + + * Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding + it. The sched entry will no longer self stop and must be externally + stopped. + + * Added REF_DEBUG description tags to struct sched_data in + pjsip_options.c. + + * Fix some off-nominal ref leaks in schedule_qualify(), + qualify_and_schedule(). + + * Reordered pjsip_options.c module start/stop code to cleanup better on + error. + + ASTERISK-24295 #close + Reported by: Rogger Padilla + + Review: https://reviewboard.asterisk.org/r/3954/ + ........ + + Merged revisions 423866 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423867 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-24 03:55 +0000 [39fada4dc9] Walter Doekes + + * chan_sip: Unref outbound proxy structure on dialog/pvt destruction. + + Make sure outbound proxy refs are always unreffed on dialog destruction. + + Review: https://reviewboard.asterisk.org/r/4016/ + ........ + + Merged revisions 423800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423801 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423802 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423803 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-23 09:36 +0000 [a89964a510] Mark Michelson + + * Make CDR and CEL unit tests less FRACKy. + + Prior to this commit, CDR and CEL tests were expected to trigger + FRACKs (i.e. assertions) due to the fact that the channels they + create have no formats on them. Some code was independently added + recently that attempts to prevent FRACKs from occurring by failing + early when attempting to set up translation paths if one or both + channels support no formats. Unfortunately, this attempt to be helpful + made the CDR and CEL tests go from simply FRACKing to outright + failing and in some cases, failing so badly as to crash Asterisk. + + This commit seeks to correct past mistakes by adding the ulaw format + to channels created by the CDR and CEL unit tests. This makes setting + up translation paths succeed, eliminates previously-seen FRACKs, and + ultimately causes the unit tests to succeed again. + + Review: https://reviewboard.asterisk.org/r/4014 + ........ + + Merged revisions 423783 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-22 14:49 +0000 [593455621b] Torrey Searle (License #5334) + + * chan_sip: On INVITE retransmission, don't add an extra 503 response. + + INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is + retransmitted, asterisk would generate a 503 in addition to the 486. + + Thanks Torrey Searle for providing a working regression test. + + ASTERISK-24335 #close + + Review: https://reviewboard.asterisk.org/r/4003/ + Patches: + retrans_486_invite.patch uploaded by Torrey Searle (License #5334) + ........ + + Merged revisions 423720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423721 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423722 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423723 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423724 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-22 12:42 +0000 [63a4da4a0d] Walter Doekes + + * cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. + + r421600 conflicted with r155763. + + ASTERISK-24348 #close + ........ + + Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423658 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423659 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423660 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423661 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-20 20:16 +0000 [64a9e5f001] Matt Jordan + + * main/channel: Unlock channel in off-nominal path + + In r423414 (13) / r423415 (trunk), an API call that determines if a format + capability structure is empty was added. This returns true if the format + capability structure is completely empty or "none". A check for this was added + in channel.c's set_format call. Unfortunately, when this check was true, it + returned from the function while still holding the channel lock. This caused + the CDR unit tests - which have a tendency to create channels with no formats - + to deadlock. Whoops. + + This patch unlocks the channel on the off-nominal path. + ........ + + Merged revisions 423641 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423642 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-20 18:55 +0000 [9bf039346a] Matt Jordan + + * rest-api/api-docs/events.json: Remove non-compliant 'extends' attribute + + Prior to the release of Swagger 1.2, the attribute 'extends' was being + promoted as a possible way to show that a particular object extends an existing + object. Instead, the Swagger specification went with the 'subTypes' attribute + in the base object. This patch removes the unsupported attribute; the object + that the offending objects proposed to extend already lists them in its + 'subTypes' attribute. + + ASTERISK-24300 #close + Reported by: Bradley Watkins + ........ + + Merged revisions 423620 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423621 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-20 18:41 +0000 [de6e467db7] Matt Jordan + + * rest-api/api-docs: Correct basePath in resources to match top resources file + + The resources.json file that defines the resource JSON files used with ARI + references a basePath of 'http://localhost:8088/ari'. This does not match what + is defined in the resource files themselves, 'http://localhost:8088/stasis'. + The correct base path is the one that includes 'ari' in the URL; this patch + updates the various resource JSON files to have the correct basePath. + + ASTERISK-24339 #close + Reported by: Bradley Watkins + ........ + + Merged revisions 423617 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423618 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-19 14:51 +0000 [354fff327d] Joshua Colp + + * res_pjsip_notify: Fix crash on unload/load and don't say the module doesn't exist on reload. + + When unloading the module did not unregister the CLI commands causing a crash upon + load when they were registered again. + + When reloading the module the return value from the config options framework was not + checked to determine if an error occurred or not. This caused a message to be output + saying the module did not exist when reloading if no changes were present. + + AST-1433 #close + AST-1434 #close + ........ + + Merged revisions 423579 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423580 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-19 12:16 +0000 [ec0313c411] Richard Mudgett + + * res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated. + + Outgoing PJSIP calls can result in non-negotiated formats listed in the + channel's native formats if video formats are listed in the endpoint's + configuration. The resulting call could then use a non-negotiated format + resulting in one way audio. + + * Simplified the update of session->req_caps in set_caps(). Why do + something in five steps when only one is needed? + + AFS-162 #close + + Review: https://reviewboard.asterisk.org/r/4000/ + ........ + + Merged revisions 423561 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-19 10:54 +0000 [6dae345674] Jonathan Rose + + * Stasis_channels: Resolve unfinished Dials when doing masquerades + + Masquerades into channels that are in the dialing state don't end their dial + and this goes against the model for things like CDRs and generating Dial end + manager actions and such. + + ASTERISK-24237 #close + Reported by: Richard Mudgett + Review: https://reviewboard.asterisk.org/r/3990/ + ........ + + Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423530 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-19 10:11 +0000 [7e602175ff] Jonathan Rose + + * chan_iax2: Fix a crash when using chan_iax2 jitterbuffer settings + + Caused by format changes in Asterisk 13 + + ASTERISK-24265 #close + Reported by: Dafi Ni + Review: https://reviewboard.asterisk.org/r/3999/ + ........ + + Merged revisions 423524 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-19 07:50 +0000 [7f2623a26f] Kinsey Moore + + * PJSIP: Prevent T38 framehook being put on wrong channel + + This change gives framehooks a reverse-direction masquerade callback in + addition to chan_fixup_cb similar to the callback added to datastores + to handle the same situation. The new callback provides the same + parameters as the fixup callback, but is called on the new channel's + framehooks before moving framehooks from the old channel to the new + channel. This gives the framehooks an oppurtunity to decide whether + they should remain on the new channel or be removed. + + This new callback is used to prevent the PJSIP T.38 framehook from + remaining on a masqueraded channel if the new channel is not also a + PJSIP channel. This was causing a crash when a local channel was + masqueraded into a PJSIP channel and the framehook was executed on the + local channel since the channel's tech private data was not structured + as expected. + + Review: https://reviewboard.asterisk.org/r/4001/ + ........ + + Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423504 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 14:31 +0000 [40e033a6b6] Sean Bright + + * res_pjsip: Don't require a password when doing userpass authentication. + + An empty password is valid for username/password authentication so we should + allow password to be empty/not supplied. + + Review: https://reviewboard.asterisk.org/r/3988 + ........ + + Merged revisions 423481 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423482 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423483 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 14:23 +0000 [ad8ef9175a] gtjoseph + + * utils: Create ast_strsep function that ignores separators inside quotes + + This function acts like strsep with three exceptions... + * The separator is a single character instead of a string. + * Separators inside quotes are treated literally instead of like separators. + * You can elect to have leading and trailing whitespace and quotes + stripped from the result and have '\' sequences unescaped. + + Like strsep, ast_strsep maintains no internal state and you can call it + recursively using different separators on the same storage. + + Also like strsep, for consistent results, consecutive separators are not + collapsed so you may get an empty string as a valid result. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3989/ + ........ + + Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423478 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 13:56 +0000 [de72f3edbc] Mark Michelson + + * Add subscription state test events. + + These are needed for a set of batched notification RLS tests that are + about to be committed to the testsuite. + + Review: https://reviewboard.asterisk.org/r/3967 + ........ + + Merged revisions 423462 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 12:22 +0000 [ac46240b62] Jonathan Rose + + * res_pjsip_endpoint_identifier_ip: Fix parsing of match value with CIDR + + Also fixes comma separates match lists + + ASTERISK-24290 #close + Reported by: Ray Crumrine + Review: https://reviewboard.asterisk.org/r/3995/ + ........ + + Merged revisions 423417 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423425 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 12:10 +0000 [02cf1835e3] Richard Mudgett + + * bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent. + + * Clarified some read/write format comments. + + * Fixed a doxygen tag typo. + ........ + + Merged revisions 423423 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 11:56 +0000 [a7add3a257] Richard Mudgett + + * astobj2.c/refcounter.py: Fix to deal with invalid object refs. + + * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 + object ref/unref is attempted. This is similar to the + constructor/destructor lines. + + * Fixed refcounter.py to handle skewed objects that have + constructor/destructor states. + + * Made refcounter.py highlight the invalid ao2 object refs by putting them + in their own section of the processed output file. + + * Made refcounter.py highlight unreffing an object by more than one that + results in a negative ref count and the object being destroyed. The + abnormally destroyed object is reported in the invalid and finalized + object sections of the output. + + Review: https://reviewboard.asterisk.org/r/3971/ + ........ + + Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423418 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 11:38 +0000 [fa6313ad29] Mark Michelson + + * Add API call to determine if format capability structure is "empty". + + Empty here means that there are no formats in the format_cap structure + or the only format in it is the "none" format. + + I've added calls to check the emptiness of a format_cap in a few places + in order to short-circuit operations that would otherwise be pointless + as well as to prevent some assertions from being triggered in cases + where channels with no formats are used. + ........ + + Merged revisions 423414 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 11:24 +0000 [389db2b720] Mark Michelson (License #5049) + + * res_fax_spandsp: Properly handle cleanup before starting FAXes. + + If faxing fails at a very early stage, then it is possible for + us to pass a NULL t30 state pointer to spandsp, which spandsp + is none too pleased with. + + This patch ensures that we pass the correct pointer to spandsp + in the situation where we have not yet set our local t30 state + pointer. + + ASTERISK-24301 #close + Reported by Matt Jordan + Patches: + ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049) + ........ + + Merged revisions 423360 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423365 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423372 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 11:09 +0000 [79eac1ffca] Mark Michelson + + * res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies. + + res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE + arrives. + * It checks that there is a subscription handler for the Event + * It checks that there are body generators for the types in the Accept header + + The problem is, there's nothing that ensures that these two things will + actually mesh with each other. For instance, Asterisk will accept a subscription + to MWI that accepts pidf+xml bodies. That doesn't make sense. + + With this commit, we add some type information to the mix. Subscription + handlers state they generate data of type X, and body generators state + that they consume data of type X. This way, Asterisk doesn't end up in + some hilariously mismatched situation like the one in the previous paragraph. + + ASTERISK-24136 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/3877 + Review: https://reviewboard.asterisk.org/r/3878 + ........ + + Merged revisions 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423348 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423350 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 10:14 +0000 [126334a7aa] gtjoseph + + * res_pjsip: ami: Fix error in AMI output when an endpoint has no transport + + When no transport is associated to an endpoint, the AMI output for + PJSIPShowEndpoint indicates an error instead of silently ignoring the + missing transport. + + This patch causes the error to appear only if a transport was specified + on the endpoint and the transport doesn't exist. It also fixes an issue + with counting the objects that were actually found. + + ASTERISK-24161 #close + ASTERISK-24331 #close + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3998/ + ........ + + Merged revisions 423282 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423284 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 10:07 +0000 [b89491e39c] David M. Lee + + * Only install dahdi_span_config_hook if DAHDI is enabled + + This patch changes the install to only install the hook script if + DAHDI is enabled. It also adds the script to the uninstall task, and + moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the + _MAKEOPTS variables and their comment. + + This allows installs which specify a --prefix to work normally, as + long as they don't enable DAHDI. + + Review: https://reviewboard.asterisk.org/r/3972/ + ........ + + Merged revisions 423281 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-18 09:46 +0000 [d120e40309] gtjoseph + + * config: bug: Fix SEGV in ast_category_insert when matching category isn't found + + If you call ast_category_insert with a match category that doesn't exist, the + list traverse runs out of 'next' categories and you get a SEGV. This patch + adds check for the end-of-list condition and changes the signature to return + an int for success/failure indication instead of a void. + + The only consumer of this function is manager and it was also changed to use + the return value. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3993/ + ........ + + Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423277 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423278 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423279 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-17 13:06 +0000 [8839ba3727] Joshua Colp + + * res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered. + ........ + + Merged revisions 423253 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423254 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423255 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-16 17:46 +0000 [fcc09fd0de] Matt Jordan + + * pbx/Makefile: Revert r423237 + + This patch was supposed to go into a team branch, but I was a bit fast on the + gun before 'svn switch' had apparently moved the target branch over. + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-16 17:42 +0000 [712b4195ef] Matt Jordan + + * Add some pbx python stuff + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423237 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-16 16:06 +0000 [618b46d8f0] Joshua Colp + + * Multiple revisions 423209,423212 + + ........ + r423209 | file | 2014-09-16 17:35:34 -0300 (Tue, 16 Sep 2014) | 8 lines + + res_rtp_asterisk: Fix building when pjproject is not used. + ........ + + Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + r423212 | file | 2014-09-16 18:03:59 -0300 (Tue, 16 Sep 2014) | 10 lines + + res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. + + Side note: I need a vacation. + ........ + + Merged revisions 423210 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423211 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423209,423212 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-16 11:33 +0000 [662b687dbe] Scott Griepentrog + + * Voicemail: get correct duration when copying file to vm + + Changes made during format improvements resulted in the + recording to voicemail option 'm' of the MixMonitor app + writing a zero length duration in the msgXXXX.txt file. + + This change introduces a new function ast_ratestream(), + which provides the sample rate of the format associated + with the stream, and updates the app_voicemail function + for ast_app_copy_recording_to_vm to calculate the right + duration. + + Review: https://reviewboard.asterisk.org/r/3996/ + ASTERISK-24328 #close + ........ + + Merged revisions 423192 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-16 07:12 +0000 [ceedf44edd] Joshua Colp + + * res_pjsip_session: Fix usage of wrong memory pool when creating local SDP. + ........ + + Merged revisions 423172 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423173 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-16 06:12 +0000 [e977425bc8] Joshua Colp + + * res_rtp_asterisk: Fix a myriad of TURN client issues. + + 1. The number of file descriptors an ioqueue instance can handle is fixed, so we + now spawn the required number to handle the load. + 2. Our transport identifiers were exceeding the range supported by pjnath. + 3. The TURN client did not set up client binding causing needless bandwidth usage. + 4. The code no longer updates address information on each packet. + 5. STUN traffic was getting looped back to Asterisk instead of going through the + TURN server. + 6. Synchronization now ensures things are completely setup or destroyed. + 7. Logging now reflects the target the TURN server is sending to/receiving from + on our behalf. + + ASTERISK-23577 #close + Reported by: Jay Jideliov + + ASTERISK-23634 #close + Reported by: Roman Skvirsky + + Review: https://reviewboard.asterisk.org/r/3982/ + ........ + + Merged revisions 423150 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423151 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423152 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423153 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-15 05:50 +0000 [77834b72d3] Zogot, cleaned up by me. + + * contrib: Fix verifyi typo in alembic DB script ps_transport table. + + Reported by: Zogot (on IRC) + Patches: + tmp.diff uploaded by Zogot, cleaned up by me. + ........ + + Merged revisions 423128 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423129 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-14 10:54 +0000 [a62fedf0cb] Walter Doekes + + * chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. + + Document it in sip.conf. + + ASTERISK-24249 #close + Reported by: Avinash Mohod + + Review: https://reviewboard.asterisk.org/r/3926/ + ........ + + Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 423067 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 423068 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 423069 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-14 10:41 +0000 [9c1f34c7e9] Walter Doekes + + * musiconhold: Add sort=randstart, and deprecate old stuff. + + - adds sort=randstart (next to sort=, sort=random, sort=alpha) + - combines duplicate moh option parsing code into a single function + - adds deprecationwarnings for application=r to sort randomly + - adds deprecationwarnings for random=yes to sort randomly + - removes invisible code that was supposed to stay until 1.8 + + The sort=randstart works like sort=alpha, except we start at a random + position. + + Review: https://reviewboard.asterisk.org/r/3991/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423065 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-12 12:42 +0000 [02295456ef] Joshua Colp + + * chan_rtp: Add unicast RTP support. + + This module supports sending both unicast and multicast RTP + to a specified target. Multicast functionality is the same as + chan_multicast_rtp was. In the case of unicast a specific + IP address and port can be specified, along with optional RTP + engine and format in the form of: + + UnicastRTP/:// + + This can be useful for sending a copy of a media stream to + another application for processing. + + Review: https://reviewboard.asterisk.org/r/3981/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-12 11:19 +0000 [dd6bdede7d] Jonathan Rose + + * Realtime: Fix a bug that caused realtime destroy command to crash + + Also has could affect with anything that goes through ast_destroy_realtime. + If a CLI user used the command 'realtime destroy ' with only a single + column/value pair, Asterisk would crash when trying to create a variable list + from a NULL value. + + ASTERISK-24231 #close + Reported by: Niklas Larsson + Review: https://reviewboard.asterisk.org/r/3985/ + ........ + + Merged revisions 422984 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422985 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-11 17:17 +0000 [c212a71f0b] Mark Michelson + + * Remove undocumented default behavior of ast_play_and_record_full acceptdtmf. + + ast_play_and_record_full() has a parameter called "acceptdtmf" that is a + string of acceptable DTMF digits that may be pressed by a caller to end + and accept the recording. + + ARI uses this function in order to perform recording, and it provides + options for what is passed as acceptdtmf to ast_play_and_record_full(). + By default, ARI passes an empty string, with the intention that no DTMF + can be used to end the recording. + + The problem is that ast_play_and_record_full() attempts to be "helpful" + by setting "#" as the acceptdtmf if an empty string or NULL pointer + has been passed in. With ARI, this results in unexpected behavior + occurring if you have attempted to intercept "#" yourself in order + to perform some other manipulation of the live recording. + + This change removes the "helpful" behavior by no longer accepting + "#" as a default acceptdtmf if none is specified by the caller of + ast_play_and_record_full(). This makes the ARI scenario work as + expected. + + The other callers of ast_play_and_record_full() are app_voicemail + and app_minivm, and in both cases, they pass an explicit "#" to + ast_play_and_record_full() as acceptdtmf, so they are unaffected + by this change. + ........ + + Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422965 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-10 11:07 +0000 [93894d53c4] gtjoseph + + * config: bug: fix truncation of included config files on permissions error + + ast_config_text_file_save() currently truncates include files as they + are processed. If a subsequent include file or the main config file has + a permissions error that prevents writing, earlier include files are left + truncated resulting in a frantic search for backups. + + This patch causes ast_config_text_file_save to check for write access + on all files before it truncates any of them. + + Will be applied 1.8 > trunk. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3986/ + ........ + + Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422903 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422904 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422905 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-10 11:00 +0000 [7bd3287a11] Sean Bright + + * pjsip/config_auth.c: Add missing whitespace to log messages. + + The errors generated when validating 'auth' settings are missing a space which + makes the messages a little confusing. + ........ + + Merged revisions 422899 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422901 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-09 15:15 +0000 [a47873168a] Richard Mudgett + + * Update CHANGES for CHANNEL(onhold). + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422885 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-09 15:11 +0000 [51f082af34] Rusty Newton + + * Sounds/BuildSystem: Modifications to include new releases and Japanese language. + + Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15 + sound prompt releases, plus the new Japanese core sound prompts contributed + by QLOOG. + + ASTERISK-23324 + Reported by: Kevin McCoy + Tested by: Rusty Newton + ........ + + Merged revisions 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422790 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422791 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422883 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422884 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-09 11:14 +0000 [9183416fe2] Richard Mudgett + + * func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel. + + It would be useful to get the current hold status of a channel. + + Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for + the hold status of a channel. + + ASTERISK-24038 + Reported by: Matt Jordan + + AFS-113 #close + Reported by: Mark Michelson + + Review: https://reviewboard.asterisk.org/r/3983/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-08 13:04 +0000 [baf99dffac] Mark Michelson + + * Add note about configuring list_items on a single line. + ........ + + Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-08 12:53 +0000 [5ad0edacb6] Mark Michelson + + * Add sample configuration for resource lists. + + On review /r/3977, it was recommended to note in the + sample configuration about the size limitation for + resource lists. However, since there was no section in + the sample configuration at all for resource list + subscriptions, I decided to make a separate commit + where I have added the necessary sample configuration + as well as the size limitation warning. + ........ + + Merged revisions 422853 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422854 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-08 12:35 +0000 [c6bc44f700] Mark Michelson + + * Pre-allocate transmission data buffer for RLS NOTIFY requests. + + PJSIP, unless a constant is modified at compilation time, limits + SIP requests to 4000 bytes. Full-state RLS notifications can easily + exceed this limit with moderately small lists. + + This changeset allows for Asterisk to work around this size limit by + performing its own allocation of the transmission data buffer. This + way, Asterisk can allocate a buffer that exceeds the built-in maximum. + + We still impose our own limit of 64000 bytes, mainly because making + allocations larger than that is a bit absurd. + + ASTERISK-24181 #close + Reported by Mark Michelson + + Review: https://reviewboard.asterisk.org/r/3977 + ........ + + Merged revisions 422851 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-08 10:58 +0000 [ef5f7a0e32] Jonathan Rose + + * res_pjsip_pubsub: Check supported headers for eventlist when subscribing to + resource list + + https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan + According to the off-nominal plan, if evenlist support is not specified in a + SUBSCRIBE's supported header(s), that subscription should be rejected with an + error. + + ASTERISK-23871 + Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header + ........ + + Merged revisions 422836 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-06 17:50 +0000 [71acca4de2] Matt Jordan + + * main/cdr: Copy over location information during a fork + + When a CDR is forked, a new CDR is created and appended to the CDR chain for + the Party A. The forked CDR starts life off as a clone of the last + non-finalized for the particular Party A. In the past, merely copying over + the snapshots for Party A/Party B would be sufficient. However, as the CDRs + now contain cached information from Party A - specifically application/data, + context, and extension - we need to copy that over during a fork as well. + + Huzzah for unit tests catching this when the context/extension were derived + from a cached value on the CDR instead of on Party A. + ........ + + Merged revisions 422769 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422770 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-06 17:22 +0000 [e4591f98b1] Matt Jordan + + * main/rtp_engine: Format NTP timestamps as unsigned ints + + On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as + opposed to long ints. When the RTP engine formats these as strings, it was + previously formatting them as signed integers, which can result in some + odd negative timestamp values (particularly on 32-bit systems). This patch + formats the values as unsigned long integers. + ........ + + Merged revisions 422766 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422767 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422768 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-06 14:13 +0000 [fd8010de2b] Joshua Colp + + * res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream. + ........ + + Merged revisions 422746 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422747 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422748 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 17:04 +0000 [d42b116925] Matt Jordan + + * main/cdrs: Preserve context/extension when executing a Macro or GoSub + + The context/extension in a CDR is generally considered the destination of a + call. When looking at a 2-party call CDR, users will typically be presented + with the following: + + context exten channel dest_channel app data + default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 + + However, if the Dial actually takes place in a Macro, the current behaviour + in 12 will result in the following CDR: + + context exten channel dest_channel app data + macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 + + The same is true of a GoSub: + + context exten channel dest_channel app data + subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 + + This generally makes the context/exten fields less than useful. + + It isn't hard to preserve these values in the CDR state machine; however, we + need to have something that informs us when a channel is executing a + subroutine. Prior to this patch, there isn't anything that does this. + + This patch solves this problem by adding a new channel flag, + AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a + Macro or a GoSub. The CDR engine looks for this value when updating a Party A + snapshot; if the flag is present, we don't override the context/exten on the + main CDR object. In a funny quirk, executing a hangup handler must *not* abide + by this logic, as the endbeforehexten logic assumes that the user wants to see + data that occurs in hangup logic, which includes those subroutines. Since + those execute outside of a typical Dial operation (and will typically have + their own dedicated CDR anyway), this is unlikely to cause any heartburn. + + Review: https://reviewboard.asterisk.org/r/3962/ + + ASTERISK-24254 #close + Reported by: tm1000, Tony Lewis + Tested by: Tony Lewis + ........ + + Merged revisions 422718 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422719 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 16:56 +0000 [4499eb05d8] Matt Jordan + + * main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios + + This patch fixes an issue where CDRs would get stuck generating an infinite + number of CDRs, eventually crashing Asterisk (and consuming a lot of memory + along the way). + + When a channel enters into a multi-party bridge, the CDR engine creates + mappings of each participant to each other participant, picking the 'A' party + as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob, + Charlie, Denise), we would have something like: + + Alice => Bob + Alice => Charlie + Alice => Denise + Bob => Charlie + Bob => Denise + Charlie => Denise + + This works fine when participants enter the bridge a single time. + + When a participant leaves a bridge, the CDRs for that channel are transitioned + to a finalized state. + + The bug occurs if Bob rejoins. When the CDR engine creates mappings between the + channels, it walks through all the participants currently in the bridge, and + realizes that no one in the bridge can create a CDR with the channel (Bob). + As such it creates a new CDR for the candidate and appends it to that + candidate's chain. Unfortunately, on this particular code path, it doesn't + stop traversing the candidate's chain. Since we just added ourselves to the + chain, this causes the loop to keep going, constantly adding new CDRs. + + This patch makes it so the engine bails when it creates a CDR match in this + case. + + Review: https://reviewboard.asterisk.org/r/3964/ + + ASTERISK-24241 #close + Reported by: Deepak Singh Rawat + Tested by: Deepak Singh Rawat + + ASTERISK-24208 + Reported by: Frankie Chin + ........ + + Merged revisions 422715 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422716 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 15:38 +0000 [025bd1bf3f] Richard Mudgett + + * func_channel.c: Add missing locking to some CHANNEL() requests. + + * The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and + audiowriteformat now need locking since the media format rework when + accessing the channel's format pointers. + + * Increased the buffer size for CHANNEL() audionativeformat and + videonativeformat output strings since the allow=all can be a lengthy + list. + + * Tweaked the CHANNEL() XML documentation for secure_bridge_signaling, + secure_bridge_media, and state. + + * Ensured the output buffer is initialized for secure_bridge_signaling and + secure_bridge_media. + + * Made use the locked_copy_string() macro instead of inlining it for trace + and checkhangup. + ........ + + Merged revisions 422700 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 15:22 +0000 [85878c4dd8] Jonathan Rose + + * Dial API: Add a dial option to indicate the dialed channel will replace dialer + + Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes. + + Review: https://reviewboard.asterisk.org/r/3968/ + ........ + + Merged revisions 422684 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 14:39 +0000 [e19017fc00] Jonathan Rose + + * Call IDs: Fix appearance of call ID in core show channels when NULL + + NULL call IDs were meant to appear as '(none)' but instead were showing + the contents of an uninitialized character buffer. + + ASTERISK-24223 + Review: https://reviewboard.asterisk.org/r/3979/ + ........ + + Merged revisions 422664 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422665 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 12:45 +0000 [5a1de68b9a] Richard Mudgett + + * devicestate.c: Minor tweaks + + * In ast_state_chan2dev() use ARRAY_LEN() instead of a sentinel value in + chan2dev[]. + + * Fix some comments in chan_iax2.c. + ........ + + Merged revisions 422661 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-05 08:29 +0000 [2362d88a18] Kinsey Moore + + * Menuselect: Fix incorrect enabling on failed deps + + This corrects a situation where menuselect can incorrectly enable a + module by default that has defaultenabled set to "no" and has + failed/non-selected dependencies. The bug is due to an inverted test + when checking for whether the given module should be set to enabled by + default on load. + + Review: https://reviewboard.asterisk.org/r/3975/ + Reported by: John Bigelow + ........ + + Merged revisions 422646 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-04 17:05 +0000 [af75e45da1] Jonathan Rose + + * Manager: Require read permission for SYSTEM in order to send FullyBooted + + Review: https://reviewboard.asterisk.org/r/3969/ + ........ + + Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422625 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422626 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422631 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422632 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-03 09:05 +0000 [3cd36d0e10] Joshua Colp + + * res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI. + + The code for changing the Contact header wrongly assumed that the Contact + would always contain a URI. This is incorrect. + + ASTERISK-24271 + Reported by: Dafi Ni + ........ + + Merged revisions 422557 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422558 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-02 15:29 +0000 [1b64f353f1] Mark Michelson + + * Resolve race condition where channels enter dialplan application before media has been negotiated. + + Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, + an AST_CONTROL_ANSWER frame is queued prior to when media has finished being + negotiated. This is because session supplements are called into before PJSIP's + inv_session code has told us that media has been updated. Sometimes the queued answer + frame is handled by the PBX thread before the ensuing media negotiations occur, causing + a test failure. + + As it turns out, there is another place that session supplements could be called into, which is + after media has finished getting negotiated. What this commit introduces is a means for session + supplements to indicate when they wish to be called into when handling an incoming SIP response. + By default, all session supplements will be run at the same point that they were prior to this + commit. However, session supplements may indicate that they wish to be handled earlier than + normal on redirects, or they may indicate they wish to be handled after media has been negotiated. + + In this changeset, two session supplements have been updated to indicate a preference for when + they should be run: res_pjsip_diversion executes before handling redirection in order to get + information from the Diversion header, and chan_pjsip now handles responses to INVITEs after + media negotiation to fix the race condition mentioned previously. + + ASTERISK-24212 #close + Reported by Matt Jordan + + Review: https://reviewboard.asterisk.org/r/3930 + ........ + + Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-01 10:25 +0000 [897cbf6a4f] Matt Jordan + + * main/cli: Do not attempt to show CDR data for internal channels + + Internal channels don't have CDRs. Querying the CDR engine for their variables + will make it cranky. + ........ + + Merged revisions 422506 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422507 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-09-01 09:15 +0000 [df5dbbd878] Matt Jordan + + * res_stasis: Don't play MoH to channels by default when added to holding bridges + + When ARI manipulates a bridge, it generally doesn't care what the mixing + technology is. Operations on a bridge initiated through ARI should perform + their action in generally the same way, regardless of the bridge's mixing + technology. While the mixing technology may determine how media flows to + channels, the actual operations on a bridge themselves should be the same. + + Currently, this isn't the case with holding bridges. When a channel joins + without a role, MoH is started on that channel automatically. Subsequent bridge + operations that would stop MoH would fail (as there is no Announcer channel + playing MoH to the bridge). Starting MoH on the bridge will also create two + MoH streams: one from the MoH being played on the participant channel, and one + from the announcer channel. From the perspective of ARI users, this is + counter-intuitive - I would not expect MoH to be started for me. The mixing + technology determines how media is shared between participants, not the + application experience. + + This patch does the following: + * The Stasis bridge class now inspects channels as they are going into a + bridge. If the bridge has a holding capability, and the channel has no + roles, we give it a participant role and mark the default behaviour to have + no entertainment. This allows addChannel operations to continue to set a + participant role with an entertainment option if it felt like it (or could + do it). + * The music on hold channel is now Stasis approved (tm) + + Review: https://reviewboard.asterisk.org/r/3929/ + + ASTERISK-24264 #close + Reported by: Samuel Galarneau + Tested by: Samuel Galarneau + ........ + + Merged revisions 422503 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422504 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422505 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-30 12:33 +0000 [5aefecd81e] gtjoseph + + * confbridge: Add Duration to ConfbridgeList event + + The ConfbridgeList event doesn't include how long the user has been a + member of the conference. This patch adds Duration (seconds) which + is based on user->chan->answertime. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3955/ + ........ + + Merged revisions 422444 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422445 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-30 12:24 +0000 [59d4dbd3d0] gtjoseph + + * manager: Make WaitEvent action respect eventfilters + + A WaitEvent issued via an http session isn't respecting eventfilters defined + for the user. I just added a match_filter to the predicate that controls + astman_append. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3958/ + ........ + + Merged revisions 422439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422440 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422441 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422442 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422443 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-29 14:40 +0000 [664f83a03b] Jeremy Laine (License 6561) + + * doc: Add a manpage for the smsq utility + + This patch adds a manpage for the smsq utility. Note that this is one of + the patches the Debian distro applies for the Asterisk project, as per + ASTERISK-24191. + + Review: https://reviewboard.asterisk.org/r/3895/ + + ASTERISK-24171 #close + Reported by: Jeremy Laine + patches: + smsq.8 uploaded by Jeremy Laine (License 6561) + ........ + + Merged revisions 422376 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422377 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422378 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422379 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-29 14:35 +0000 [81598fa082] Jeremy Laine (License 6561) + + * doc: Add a manpage for the aelparse utility + + This patch adds a manpage for the aelparse utility. Note that this is one of + the patches the Debian distro applies for the Asterisk project, as per + ASTERISK-24191. + + Review: https://reviewboard.asterisk.org/r/3896/ + + ASTERISK-24171 #close + Reported by: Jeremy Laine + patches: + aelparse.8 uploaded by Jeremy Laine (License 6561) + ........ + + Merged revisions 422371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422372 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422373 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422374 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-29 13:46 +0000 [2df2d785b7] Scott Griepentrog + + * The assertion that peer was not found on final event + message was being triggered on configuration reload. + This patch changes that case to just return instead. + + Review: https://reviewboard.asterisk.org/r/3953/ + + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-28 16:54 +0000 [3194892ea2] Matt Jordan + + * LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP + + The UniMRCP project distributes Asterisk modules that integrate Asterisk with + UniMRCP, and other Asterisk users use the UniMRCP library as well. + Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software + Foundation, is not a compatible license with the GPLv2. + + "Please note that this license is not compatible with GPL version 2, because it + has some requirements that are not in that GPL version. These include certain + patent termination and indemnification provisions. The patent termination + provision is a good thing, which is why we recommend the Apache 2.0 license for + substantial programs over other lax permissive licenses." + + On the other hand, UniMRCP is a great project and we'd like to let people use + it with Asterisk. + + This patch updates the LICENSE text to allow users to link Asterisk with + UniMRCP and distribute the resulting binaries. + ........ + + Merged revisions 422293 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422294 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422295 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422296 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-28 15:31 +0000 [c5916fb39f] Michael L. Young (license 5026) + + * chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure + + The reporter on the issue found some issues when upgrading from version 10 to 11 + on 55 hosts. + + Two situations that can occur with dynamic registrations. + + 1. With dnsmgr disabled, if the host is not resolvable we are not trying to + resolve the host again when it is time to attempt to register again. This + results in never registering to the host. + 2. With dnsmgr enabled, when the host is temporarily not resolvable the + address is set to 0.0.0.0:0 and then when the host is resolvable the port + is not being restored and stays set to 0. + + This patch resolves these two issues by: + + * Storing the hostname so that it can be used for resolving with DNS. + * Resolve the hostname on the next scheduled attempt to register. + * Storing the port used to reach the host so that when the hostname is + resolvable again, we can set the port again if the port is still unset after + looking up the host. + + ASTERISK-23767 #close + Reported by: David Herselman + Tested by: David Herselman, Michael L. Young + Patches: + asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff + uploaded by Michael L. Young (license 5026) + + Review: https://reviewboard.asterisk.org/r/3856/ + ........ + + Merged revisions 422274 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422275 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422276 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-28 12:29 +0000 [4e750a26fd] Richard Mudgett + + * Added ConfBridge AMI event note to UPGRADE.txt. + ........ + + Merged revisions 422255 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422256 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-28 11:06 +0000 [ef28cc0d43] Paul Belanger + + * chan_sip.c: Add 'rtpbindaddr' setting + + Users now have the ability to bind the rtpengine instance to a specific IP + address. For example, you want chan_sip (call control) on eth0 but rtp (media) + on eth1. + + ASTERISK-24280 #close + Reported by: Paul Belanger + Tested by: Paul Belanger + Review: https://reviewboard.asterisk.org/r/3952/ + Patches: + rtpengine.diff uploaded by Paul Belanger + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-28 10:50 +0000 [327d67270f] Mark Michelson + + * Fix bug that did not allow for multiple batched RLS notifications to be sent. + + A misunderstanding of how the scheduler worked caused further batched notifications + beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after + the batched notification is sent. This way, further notifications can be scheduled + when they arise. + ........ + + Merged revisions 422239 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-27 19:44 +0000 [94e1b4a8a4] Richard Mudgett + + * res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage. + + * Fix off nominal ref leak in find_or_create_contact_status(). + + * Add missing NULL check of status in update_contact_status() and + init_start_time(). + ........ + + Merged revisions 422214 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422215 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-27 19:16 +0000 [4728c05957] Richard Mudgett + + * sched: Fix typo and whitespace change. + ........ + + Merged revisions 422200 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-27 12:30 +0000 [7c1a22fba7] gtjoseph + + * confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events + + Currently there's no way to tell if a user is an admin or not when receiving + the join, leave, mute, unmute and talking events. This patch adds that + capability. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3950/ + ........ + + Merged revisions 422176 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422177 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-27 10:39 +0000 [bf85018107] Kinsey Moore + + * CallerID: Fix parsing of malformed callerid + + This allows the callerid parsing function to handle malformed input + strings and strings containing escaped and unescaped double quotes. + This also adds a unittest to cover many of the cases where the parsing + algorithm previously failed. + + Review: https://reviewboard.asterisk.org/r/3923/ + Review: https://reviewboard.asterisk.org/r/3933/ + ........ + + Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422154 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-26 18:30 +0000 [d199536a04] gtjoseph + + * confbridge: Make kick, mute and unmute handle channel targets consistently. + + Kick, mute and unmute were a little inconsistent in their handling of channel + targets. This patch cleans that up by insuring they all handle the 'all' + target consistently and adds the 'participants' target which acts on + non-admins. Documentation for kick was also cleaned up as it never + supported partial channel names. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3944/ + ........ + + Merged revisions 422090 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422091 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-26 17:14 +0000 [c5ab4adf17] Mark Michelson + + * Fix race condition in the scheduler when deleting a running entry. + + When scheduled tasks run, they are removed from the heap (or hashtab). + When a scheduled task is deleted, if the task can't be found in the + heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled, + this assertion causes a crash. + + The problem is, sometimes it just so happens that someone attempts + to delete a scheduled task at the time that it is running, leading + to a crash. This change corrects the issue by tracking which task + is currently running. If that task is attempted to be deleted, + then we mark the task, and then wait for the task to complete. + This way, we can be sure to coordinate task deletion and memory + freeing. + + ASTERISK-24212 + Reported by Matt Jordan + + Review: https://reviewboard.asterisk.org/r/3927 + ........ + + Merged revisions 422070 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 422071 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-25 11:45 +0000 [fefa6fba82] Richard Mudgett + + * res_musiconhold.c: Release any format refs before memset(). + + * Clear the channel music_state pointer before destroying the music_state + object for safety. + ........ + + Merged revisions 422037 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-25 11:16 +0000 [2b19d94a71] Richard Mudgett + + * res_musiconhold: Fix MOH restarting where it left off from the last hold. + + Restore code removed by https://reviewboard.asterisk.org/r/3536/ that + introduced a regression that prevents MOH from restarting were it left off + the last time. + + ASTERISK-24019 #close + Reported by: Jason Richards + Patches: + jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett + + Review: https://reviewboard.asterisk.org/r/3928/ + ........ + + Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421977 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421978 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421979 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-24 14:37 +0000 [497a92d079] Joshua Colp + + * res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs. + + In order to alter the Contact header on in-dialog requests and responses the + Websocket module must be attached on outgoing INVITEs. The Contact header is + modified so that the PJSIP transport layer can find and use the existing + Websocket connection based on the source IP address, port, and transport. + + ASTERISK-24143 #close + Reported by: Aleksei Kulakov + ........ + + Merged revisions 421955 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421956 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-24 14:21 +0000 [477e2e6edb] Joshua Colp + + * res_pjsip_transport_websocket: Fix a progressive memory growth. + + The packet structure used to receive messages was using the transport + pool. This meant that for each parsing the pool would grow accordingly. + Since memory can not be reclaimed without resetting it this would + cause the memory pool to grow and grow. + + This change uses a specific memory pool for the packet structure and + resets it to a fresh state after the message has been received and + handled. + ........ + + Merged revisions 421939 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421945 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-24 13:54 +0000 [2c0cbf8e64] Joshua Colp + + * res_pjsip_transport_websocket: Ensure secure Websocket clients can be called. + + This change enforces the transport in the Contact header for Websocket clients. + Previously a client may provide a transport of 'ws' when it is actually using + a transport of 'wss'. This would cause outgoing calls to fail as the existing + connection could not be found. + ........ + + Merged revisions 421931 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421932 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421933 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-24 12:22 +0000 [cee660dadf] Badalian Vyacheslav (license 5249) + + * chan_sip: Use the server reflexive ICE candidate RTCP port as provided. + + This code originally worked around an issue within res_rtp_asterisk itself. + The wrong socket was being used for the STUN check for RTCP, causing the + port to be the same as RTP. This was subsequently fixed and the RTCP port + provided for the ICE candidate is correct and does not need to be incremented. + + ASTERISK-23997 #close + Reported by: Badalian Vyacheslav + Patches: + plus1.diff submitted by Badalian Vyacheslav (license 5249) + ........ + + Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-22 11:56 +0000 [dcfffce66d] Mark Michelson + + * Fix a locking inversion in MixMonitor. + + We need to unlock the audiohook before trying to lock + the channel, since the correct locking order is channel + then audiohook. + ........ + + Merged revisions 421882 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-22 11:52 +0000 [33835e17a0] Jonathan Rose + + * ARI: Fix a crash caused by hanging during playback to a channel in a bridge + + ASTERISK-24147 #close + Reported by: Edvin Vidmar + Review: https://reviewboard.asterisk.org/r/3908/ + ........ + + Merged revisions 421879 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421880 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-22 09:09 +0000 [1498ae0830] Matt Jordan + + * main/message: Add a new-line to a DEBUG message + ........ + + Merged revisions 421859 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421860 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421861 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 17:09 +0000 [f8c4fc1121] Richard Mudgett + + * res_musiconhold.c: Remove obsolete REF_DEBUG code. + + Remove unneeded code that writes to the wrong file location in an obsolete + format. + ........ + + Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421800 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421801 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421802 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421803 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 16:43 +0000 [644e693645] Mark Michelson + + * Switch from hostname to an IP address in the SDP origin line. + + Using the hostname in the SDP origin line may not satisfy the requirement + of RFC 4566 that we use a FQDN or IP address. This change has us use the + same information from the SDP connection line if possible. If not possible, + we'll use the configured media address. And if that's not possible, we use + the result of a PJLIB call to get the IP address of ourself. + + ASTERISK-23994 #close + Reported by Private Name + + Review: https://reviewboard.asterisk.org/r/3925 + ........ + + Merged revisions 421796 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421797 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 16:37 +0000 [56a1d4930a] Mark Michelson + + * Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge. + + Because of the departable state of channels that enter Stasis bridges, Stasis has to + take responsibility for directing the channel to its intended after-bridge destination + if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures + that when such a move occurs, when the channel leaves the bridging system, any after + bridge gotos are honored. + + Review: https://reviewboard.asterisk.org/r/3920 + ........ + + Merged revisions 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421794 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 16:35 +0000 [4946981646] Jonathan Rose (license 6182) + + * res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set + + Due to a faulty function for debugging reference decrementing, it was possible + to reduce the refcount on the wrong object if two moh classes of the same name + were in the moh class container. + + (closes issue ASTERISK-22252) + Reported by: Walter Doekes + Patches: + 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) + ........ + + Merged revisions 398937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421777 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421779 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421788 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 16:28 +0000 [12d34bb12f] Mark Michelson + + * Let's try checking the name and number, instead of the name twice. + ........ + + Merged revisions 421789 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421790 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 16:19 +0000 [2150daf748] Mark Michelson + + * Improve consistency of party ID privacy usage. + + Prior to this change, the Remote-Party-ID header took the position of + "If caller name and number are not explicitly allowed, then they are private" + and P-Asserted-Identity took the position of + "Caller name and number are only private if marked explicitly so" + + Now both mechanisms of conveying party identification use the former approach. + ........ + + Merged revisions 421778 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421783 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 12:35 +0000 [77ddc5b713] Elazar Broad (License 5835) + + * chan_sip: Don't use port derived from fromdomain if it isn't set + + If a user does not provide a port in the fromdomain setting, chan_sip will set + the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will + then get used unilaterally in certain places. This causes issues with TLS, + where the default port is expected to be 5061. + + This patch modifies chan_sip such that fromdomainport is only used if it is + not the standard SIP port; otherwise, the port from the SIP pvt's recorded + self IP address is used. + + Review: https://reviewboard.asterisk.org/r/3893/ + + ASTERISK-24178 #close + Reported by: Elazar Broad + patches: + fromdomainport_fix.diff uploaded by Elazar Broad (License 5835) + ........ + + Merged revisions 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421718 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421719 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421720 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 10:25 +0000 [f3a525e9a6] Matt Jordan + + * ARI: Fix implicit answer when playback is initiated on unanswered channel + + When issuing a POST /channels/{channel_id}/play on a channel that is not + yet answered, ARI is supposed to: + * Queue up an AST_CONTROL_PROGRESS on the channel + * Start up the playback of the media + + Instead, we sneak an answer on the channel right before starting playing media. + + This is due to ARI's usage of control_streamfile. This function implicitly + answers the channel (and doesn't give ARI the option to stop it). The answering + of the channel here is probably unnecessary: + * app_voicemail, by far the biggest consumer of this function, always answers + the channels anyway + * control stream file (in res_agi) and ControlPlayback probably shouldn't be + implicitly answering the channel. Answering should not be tied directly to + playing back media. + + As it turns out, the answering of the channel here is pretty old: + 356042 twilson if (ast_channel_state(chan) != AST_STATE_UP) { + 3087 anthm res = ast_answer(chan); + 180259 tilghman } + + (As in, ancient?) + + Note that others ran into this problem and commented about it on various + mailing lists. + + Review: https://reviewboard.asterisk.org/r/3907/ + + ASTERISK-24229 #close + Reported by: Matt Jordan + ........ + + Merged revisions 421695 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421696 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 09:52 +0000 [085d5a2629] Shaun Ruffell (License 5417) + + * Clean up files that do not end with newlines + + Trivial patch to add new lines to several files missing them. This fixes + warnings when compiling with gcc 4.1.2 on CentOS 5. + + ASTERISK-24245 #close + Reported by: Shaun Ruffell + patches: + 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417) + ........ + + Merged revisions 421677 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421678 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-21 09:42 +0000 [da91946df7] Shaun Ruffell (License 5417) + + * uri: Quiet warning about type qualifiers ignored on function return type + + This patch fixes gcc warnings that occur due to the type qualifier 'const' + being ignored on a return type of int. + + ASTERISK-24246 #close + Reported by: Shaun Ruffell + patches: + 0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417) + ........ + + Merged revisions 421675 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421676 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-20 17:52 +0000 [b7f98c3da4] Richard Mudgett + + * chan_pjsip: Update media translation paths when new SDP negotiated. + + On a SIP reinvite that changes media strams, the PJSIP channel driver was + flooding the log with "Asked to transmit frame type %s, while native + formats is %s" warnings. + + * Fixes PJSIP not setting up translation paths when the formats change on + a reinvite. AFS-63 was effectively reintroduced because of the media + formats work. res_pjsip_sdp_rtp.c:set_caps() + + * Improved the unexpected frame format WARNING message to include more + information. + + * Added protective locking while altering formats on a channel. Reworked + set_format() to simplify and protect the formats under manipulation. + + * Restored some code that got lost in the media_formats work. + (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps()) + + AFS-137 #close + Reported by: Mark Michelson + + Review: https://reviewboard.asterisk.org/r/3906/ + ........ + + Merged revisions 421645 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-20 17:23 +0000 [4672c139dd] Richard Mudgett + + * cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled. + + filename_completion_function() returns memory that was not allocated by + the MALLOC_DEBUG allocation tracker so the memory must be freed by + ast_std_free(). + ........ + + Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421602 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421608 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421616 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-20 15:41 +0000 [49f8bd4ad4] Mark Michelson + + * Set the role for inbound subscriptions correctly. + + This was causing the AMI show_subscriptions test in + the testsuite to fail since all subscriptions were being + seen as subscribers instead of notifiers. + ........ + + Merged revisions 421585 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-20 15:04 +0000 [d0640ad7df] Mark Michelson + + * Move evaluation of set_var options in pjsip to the end of channel initialization. + + This allows for set_var to override certain defaults such as caller ID and codec + values. This also fixes a test suite regression. The "set_var" test suite test attempted + to use set_var to override caller ID, but a recent change caused that to no longer work. + ........ + + Merged revisions 421565 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421566 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421567 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-20 08:06 +0000 [36f4bff943] Kinsey Moore + + * Stasis: Add information to blind transfer event + + When a blind transfer occurs that is forced to create a local channel + pair to satisfy the transfer request, information about the local + channel pair is not published. This adds a field to describe that + channel to the blind transfer message struct so that this information + is conveyed properly to consumers of the blind transfer message. + + This also fixes a bug in which Stasis() was unable to properly identify + the channel that was replacing an existing Stasis-controlled channel + due to a blind transfer. + + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/3921/ + ........ + + Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421538 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-20 07:39 +0000 [01f1ff1f77] Kinsey Moore + + * AMI: Add AllVariables parameter to Status + + This adds the AllVariables parameter to the Status AMI action such that + if defined and set to "true", all channel variables will be reported in + the subsequent Status event(s). This parameter does not negate the + functionality of the "Variables" parameter so that global variables and + dialplan functions can be requested. + + Review: https://reviewboard.asterisk.org/r/3915/ + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-19 15:28 +0000 [76290adf50] Mark Michelson + + * Alter documentation for callerid_privacy to use correct values. + ........ + + Merged revisions 421485 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421488 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-19 14:55 +0000 [28a89e7685] Mark Michelson + + * Fix compilation error on certain versions of GCC. + ........ + + Merged revisions 421447 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421448 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-19 14:43 +0000 [a85a483fcd] Kinsey Moore + + * AMI Docs: Fix Status channel parameter optionality + ........ + + Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421443 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421444 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421445 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-19 11:36 +0000 [222b5cd036] Krandon Bruse (license 6631) + + * ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX + + If /channels/{channelID}/continue is called on a channel that was originated + without a PBX (such as the ARI command POST channel with a stasis application + argument), the channel will not start dialplan execution. This patch will now + run the PBX out of the stasis execution if the channel doesn't currently have + an active PBX upon continuing. + + ASTERISK-24043 #close + Reported by: Krandon Bruse + Review: https://reviewboard.asterisk.org/r/3917/ + Patches: + stasis-continue.diff submitted by Krandon Bruse (license 6631) + ........ + + Merged revisions 421416 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421423 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-19 11:16 +0000 [83a9b91da9] Richard Mudgett + + * chan_pjsip: Fix attended transfer connected line name update. + + A calls B + B answers + B SIP attended transfers to C + C answers, B and C can see each other's connected line information + B completes the transfer + A has number but no name connected line information about C + while C has the full information about A + + I examined the incoming and outgoing party id information handling of + chan_pjsip and found several issues: + + * Fixed ast_sip_session_create_outgoing() not setting up the configured + endpoint id as the new channel's caller id. This is why party A got + default connected line information. + + * Made update_initial_connected_line() use the channel's CALLERID(id) + information. The core, app_dial, or predial routine may have filled in or + changed the endpoint caller id information. + + * Fixed chan_pjsip_new() not setting the full party id information + available on the caller id and ANI party id. This includes the configured + callerid_tag string and other party id fields. + + * Fixed accessing channel party id information without the channel lock + held. + + * Fixed using the effective connected line id without doing a deep copy + outside of holding the channel lock. Shallow copy string pointers can + become stale if the channel lock is not held. + + * Made queue_connected_line_update() also update the channel's + CALLERID(id) information. Moving the channel to another bridge would need + the information there for the new bridge peer. + + * Fixed off nominal memory leak in update_incoming_connected_line(). + + * Added pjsip.conf callerid_tag string to party id information from + enabled trust_inbound endpoint in caller_id_incoming_request(). + + AFS-98 #close + Reported by: Mark Michelson + + Review: https://reviewboard.asterisk.org/r/3913/ + ........ + + Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-18 16:18 +0000 [c4c9d4ad6c] Damien Wedhorn + + * Skinny: Fixup compile warning for non dev-mode. + ........ + + Merged revisions 421376 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-18 15:20 +0000 [1de8b8035e] gtjoseph + + * func_config: Change 'Not Found' message from ERROR to DEBUG + + When you call the CONFIG dialplan function with the name of a variable that + doesn't exist in the target context you get an ERROR. This does nothing but + clutter up the logs with messages that may be perfectly acceptable. Just + because a variable wasn't in the context doesn't mean it's an error. Maybei + t's optional or just needs to be defaulted or ignored. + + This patch changes the log level from ERROR to DEBUG. If a dialplan developer + wants to debug their dialplan they still canby setting the console debug level + as needed. + + Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3919/ + ........ + + Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421328 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421329 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421337 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-17 20:14 +0000 [bb494067a5] Matt Jordan + + * Multiple revisions 421311-421312 + + ........ + r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines + + res/ari/resource_channels: Don't return allocation failure on failed function + + If a function fails to execute, it is most likely due to one of two reasons: + (1) The function doesn't exist or can't be read from + (2) The function is dangerous and is restricted based on the user's permissions + + Currently we return allocation failure, which is incorrect. This updates the + reason code to more accurately reflect why the request failed. + + ASTERISK-24215 + ........ + r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines + + res/ari/resource_channels: Fix compilation issue + + Forgot a parameter. Whoops. + ........ + + Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-17 19:57 +0000 [ba5d5da60b] Matt Jordan (License #6283) + + * Improve call forwarding reporting, especially with regards to ARI. + + This patch addresses a few issues: + + 1) The order of Dial events have been changed when performing a call forward. + The order has now been altered to + 1) Dial begins dialing channel A. + 2) When A forwards the call to B, we issue the dial end event to channel + A, indicating the dial is being canceled due to a forward to B. + 3) When the call to channel B occurs, we then issue a new dial begin to + channel B. + + 2) Call forwards are now reported on the calling channel, not the peer channel. + + 3) AMI DialEnd events have been altered to display the extension the call is + being forwarded to when relevant. + + 4) You can now get the values of channel variables for channels that are not + currently in the Stasis application. This brings the retrieval of channel + variables more in line with the rest of channel read operations since they + may be performed on channels not in Stasis. + + ASTERISK-24134 #close + Reported by Matt Jordan + + ASTERISK-24138 #close + Reported by Matt Jordan + + Patches: + forward-shenanigans.diff uploaded by Matt Jordan (License #6283) + + Review: https://reviewboard.asterisk.org/r/3899 + ........ + + Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-17 18:29 +0000 [6525f374db] Matt Jordan + + * apps/app_meetme: Fix crash when publishing MeetMe messages with no channel + + The same function, meetme_stasis_generate_msg, handles creating and publishing + Stasis message both when there are channels in the MeetMe conference and when + there are no channels in the conference. When the performance improvement was + made to use cached snapshots, this created a situation where Asterisk would + crash: obtaining a cached snapshot is not NULL tolerant. + + This patch restores the previous implementation, which used a NULL safe set + of routines to produce a blob containing the channel snapshot (if available) + and information about the MeetMe conference. + + ASTERISK-24234 #close + Reported by: Shaun Ruffell + Tested by: Shaun Ruffell + ........ + + Merged revisions 421270 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421273 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421276 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-17 18:10 +0000 [44fc6ea6ff] Richard Mudgett (License 5621) + + * apps/app_dial: Fix Dial 'z' option + + The 'z' option is supposed to disable the dial timeout in the case of a call + forward. Unfortunately, the wrong timeout timer was passed to the do_forward + function, resulting in the option not working. + + ASTERISK-24225 #close + Reported by: dimitripietro + Tested by: dimitripietro + patches: + jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621) + jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621) + ........ + + Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421235 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421236 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-17 17:35 +0000 [98ca5c0b5f] cloos (License 5956) + + * configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc + + Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is + executed with optimization. This "help" unfortunately results in re-definition + warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This + patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning. + + Review: https://reviewboard.asterisk.org/r/3912/ + + ASTERISK-24032 #close + Reported by: Kilburn + Tested by: Kilburn, wdoekes + patches: + 1.8.diff uploaded by cloos (License 5956) + 10.diff uploaded by cloos (License 5956) + 11.diff uploaded by cloos (License 5956) + 12.diff uploaded by cloos (License 5956) + 13.diff uploaded by cloos (License 5956) + ........ + + Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421228 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421229 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421230 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-17 11:11 +0000 [952da298ce] Joshua Colp + + * res_http_websocket: Include query parameters in client connection requests. + + Review: https://reviewboard.asterisk.org/r/3914/ + ........ + + Merged revisions 421210 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-15 12:26 +0000 [9b658b7c60] Jonathan Rose + + * Bridging: Fix a behavioral change when checking if a channel is leaving a bridge + + r420934 introduced some failures in the test suite. Upon investigating, it was + discovered that differences in the way we were evaluating whether a channel was in + the process of leaving a bridge were causing some reinvites not to occur (mostly + reinvites back to Asterisk when ending a call). This patch fixes that behavioral + change. + + ASTERISK-24027 #close + Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/3910/ + ........ + + Merged revisions 421186 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421187 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421195 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-15 10:50 +0000 [0d0a616e1a] Matt Jordan + + * app_voicemail/app: Remove test events that were duplicated by r421059 + + Moving the test event raised when a file is played back (which occurred in + r421059) broke the ever loving snot out of the voicemail tests. This caused + duplicate test events to get raised, as app_voicemail and main/app were raising + events prior to call ast_streamfile. The voicemail tests did not enjoy getting + multiple events. + + Since raising the playback event in ast_streamfile is far more useful to the + vast majority of tests, this patch keeps the call there and simply removes the + extraneous calls that duplicated the event. + ........ + + Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421164 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421165 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421166 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-14 16:16 +0000 [980e49614c] Matt Jordan + + * res/res_hep_rtcp: Remove dependency on PJSIP + + The res_hep_rtcp module was incorrectly including . This didn't need + to be included, as the module does not using PJPROJECT any fashion. + Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as + a dependency, this also meant that res_hep_rtcp will fail to compile on a + system without PJPROJECT. + + This patch removes the include. + + Thanks to Damien Wedhorn for pointing this out in #asterisk-dev. + + ASTERISK-24236 #close + Reported by: Damien Wedhorn, Matt Jordan + Tested by: Damien Wedhorn + ........ + + Merged revisions 421064 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421065 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-14 15:59 +0000 [513981c89d] Matt Jordan + + * main/file: Move test event to emit PLAYBACK event more consistently + + This is being done in advance of the test for ASTERISK-23953 + ........ + + Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 421060 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 421061 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421062 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-14 14:21 +0000 [0b11c48522] Matt Jordan + + * cel: Make sure channels in extra fields include their unique IDs as well + + CEL typically tracks a lot of information using the unique ID of the channel. + This is typically needed due to tying events together using the linked ID of + the various channels involved in a "call", which is derived from the channel ID + of the oldest channel involved in a bridge (or in the case of a Dial, the + parent channel). + + Previously, we had updated the extra fields to include the involved channel + names, but forgot to put in the unique ID. This patch corrects that error. + ........ + + Merged revisions 421037 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421042 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421043 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-14 11:33 +0000 [79c5c08db9] Richard Mudgett + + * ARI: Originate to app local channel subscription code optimization. + + Reduce the scope of local_peer and only get it if the ARI originate is + subscribing to the channels. + + Review: https://reviewboard.asterisk.org/r/3905/ + ........ + + Merged revisions 421009 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 421010 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-14 11:01 +0000 [e4b32731b9] Richard Mudgett + + * channel_internal_api.c: Replace some code with ao2_replace(). + + Use ao2_replace() instead of ao2_cleanup(); ao2_bump(). + + ao2_replace() has the advantange of not altering the ref count if the + replaced pointer is the same. + + Review: https://reviewboard.asterisk.org/r/3904/ + ........ + + Merged revisions 420992 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-13 12:05 +0000 [dd41d0ff01] Richard Mudgett + + * res_pjsip_send_to_voicemail.c: Fix svn file properties. + ........ + + Merged revisions 420956 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420957 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-13 11:56 +0000 [6aa510b41f] Kinsey Moore + + * PJSIP: Prevent crash no-URI contacts + + This prevents a crash from occurring when a contact with no URI is used + for the creation of an outbound out-of-dialog request with no + associated endpoint. + ........ + + Merged revisions 420949 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420950 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420953 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-13 11:24 +0000 [d4695774e7] Jonathan Rose + + * Bridges: Fix feature interruption/unintended kick caused by external actions + + If a manager or CLI user attached a mixmonitor to a call running a dynamic + bridge feature while in a bridge, the feature would be interrupted and the + channel would be forcibly kicked out of the bridge (usually ending the call + during a simple 1 to 1 call). This would also occur during any similar action + that could set the unbridge soft hangup flag, so the fix for this was to + remove unbridge from the soft hangup flags and make it a separate thing all + together. + + ASTERISK-24027 #close + Reported by: mjordan + Review: https://reviewboard.asterisk.org/r/3900/ + ........ + + Merged revisions 420934 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420940 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-13 09:31 +0000 [6a6702bb0f] Kinsey Moore + + * AMI: Improve documentation for Status action + ........ + + Merged revisions 420919 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-13 02:54 +0000 [52c94d3af4] Walter Doekes + + * logger: Don't store verbose-magic in the log files. + + In r399267, the verbose2magic stuff was edited. This time it results + in magic characters in the log files for multiline messages. + + In trunk (and 13) this was fixed by the "stripping" of those + characters from multiline messages (in r414798). + + This fix is altered to actually strip the characters and not replace + them with blanks. + + Review: https://reviewboard.asterisk.org/r/3901/ + Review: https://reviewboard.asterisk.org/r/3902/ + ........ + + Merged revisions 420897 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 420898 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420899 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420900 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-12 18:45 +0000 [969982b878] Richard Mudgett + + * chan_sip: Fix type mismatch when the format is changed. + + Symptom is most likely an invalid ao2 object bad magic number message or a + less likely crash. + ........ + + Merged revisions 420881 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-12 18:36 +0000 [8526d967c9] Richard Mudgett + + * res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup. + + * Made use ast_copy_string() instead of strcpy() for snoop uniqueid for + safety. There is no guarantee that the max channel uniqueid length will + remain the same as the snoop uniqueid space. + ........ + + Merged revisions 420879 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-12 06:18 +0000 [ca61f8ac82] Joshua Colp + + * app_voicemail: Fix the "test_voicemail_vm_info" unit test. + ........ + + Merged revisions 420856 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 16:04 +0000 [aba07a0f6e] Richard Mudgett + + * res/stasis/command.c: Fix recent commit using spaces instead of tabs. + ........ + + Merged revisions 420836 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420837 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 13:51 +0000 [ffccae8269] Matt Jordan + + * AMI/ARI: Update version to 2.5.0/1.5.0 respectively + + This is to support the backwards compatible changes made in the next version + of Asterisk. + ........ + + Merged revisions 420805 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420808 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 13:46 +0000 [7a4691b425] Kinsey Moore + + * Stasis: Use the correct return value + + Return the correct value instead of always returning 0 when setting + internal status on unreal channels. + + Reported by: Richard Mudgett + ........ + + Merged revisions 420802 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420803 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 13:38 +0000 [6f735288b0] Kinsey Moore + + * Stasis: Allow internal channels directly into bridges + + The patch to catch channels being shoehorned into Stasis() via external + mechanisms also happens to catch Announcer and Recorder channels + because they aren't known to be stasis-controlled channels in the usual + sense. This marks those channels as Stasis()-internal channels and + allows them directly into bridges. + + Review: https://reviewboard.asterisk.org/r/3903/ + ........ + + Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420796 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 12:40 +0000 [db0a97f8ce] Mark Michelson + + * Fix crashing unit tests with regards to RLS. + + The unit tests require a sorcery.conf file that has been + set up to store resource lists in memory rather than retrieving + from configuration. + + With a setup that is not conducive to running the tests, a fault + in sorcery currently causes Asterisk to crash when attempting to + run any of the tests. + + To get around the crash, this adds a function that verifies the + current environment and marks the tests as "not run" if the setup + is not correct. + ........ + + Merged revisions 420779 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420780 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 11:03 +0000 [b4e33c81e3] Mark Michelson + + * Fix crash encountered by the testsuite. + + Running testsuite tests locally produced no errors, but when + run using the continuous integration framework, crashes occurred. + + The crashes occurred due to a refcounting error that had been fixed + for a similar situation. + ........ + + Merged revisions 420758 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 08:57 +0000 [becf7c7003] Matt Jordan + + * res_hep: Remove disabling of modules + + These modules were originally specified as being disabled, as they were + introduced midstream in Asterisk 12. That makes it nicer for folks who are + upgrading to a new release in the middle of Asterisk 12. That's not the case + for Asterisk 13: it's a brand new release. There's no reason to have the + modules disabled by default in that case. + ........ + + Merged revisions 420742 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 05:41 +0000 [1e0846167b] Walter Doekes + + * general: Fix memory Corruption in __ast_string_field_ptr_build_va. + + If the space left in a stringfield is between 0 and + (alignof(ast_string_field_allocation)-1) adding new data would cause + memory corruption, because we would assume enough space (unsigned + underrun). + + Thanks Arnd Schmitter for reporting and finding out the cause! + + ASTERISK-23508 #close + Reported by: Arnd Schmitter + Tested by: Arnd Schmitter, JoshE + + Review: https://reviewboard.asterisk.org/r/3898/ + ........ + + Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 420715 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 420716 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420717 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-11 04:55 +0000 [b2afbc48e4] Walter Doekes + + * tcptls: Avoid compiler warning on non-dev-mode. + ........ + + Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ + + Merged revisions 420655 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ + + Merged revisions 420656 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420657 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-10 20:31 +0000 [6650704414] Matt Jordan + + * funcs/func_jitterbuffer: Tweak documentation + + This patch merely reformats and cleans up a bit of the jitterbuffer + documentation for the wiki. + ........ + + Merged revisions 420639 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-10 19:14 +0000 [add46fd27c] Michael K (License 6621) + + * app_queue: Add RealTime support for queue rules + + This patch gives the optional ability to keep queue rules in RealTime. It is + important to note that with this patch: + (a) Queue rules in RealTime are only examined on module load/reload + (b) Queue rules are loaded both from the queuerules.conf file as well as the + RealTime backend + To inform app_queue to examine RealTime for queue rules, a new setting has been + added to queuerules.conf's general section "realtime_rules". RealTime queue + rules will only be used when this setting is set to "yes". + + The schema for the database table supports a rule_name, time, min_penalty, and + max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or + '+' literal is provided. Otherwise, the penalties are treated as constants. + + For example: + rule_name, time, min_penalty, max_penalty + 'default', '10', '20', '30' + 'test2', '20', '30', '55' + 'test2', '25', '-11', '+1111' + 'test2', '400', '112', '333' + 'test3', '0', '4564', '46546' + 'test_rule', '40', '15', '50' + + which would result in : + + Rule: default + - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust + QUEUE_MIN_PENALTY to 20 + Rule: test2 + - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust + QUEUE_MIN_PENALTY to 30 + - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust + QUEUE_MIN_PENALTY by -11 + - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust + QUEUE_MIN_PENALTY to 112 + Rule: test3 + - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust + QUEUE_MIN_PENALTY to 4564 + Rule: test_rule + - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust + QUEUE_MIN_PENALTY to 15 + + If you use RealTime, the queue rules will be always reloaded on a module + reload, even if the underlying file did not change. With the option disabled, + the rules will only be reloaded if the file was modified. + + Review: https://reviewboard.asterisk.org/r/3607/ + + ASTERISK-23823 #close + Reported by: Michael K + patches: + app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) + ........ + + Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-10 17:02 +0000 [f7bb772804] Matt Jordan + + * Update CHANGES file + ........ + + Merged revisions 420609 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420610 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-10 16:35 +0000 [455243cdd4] Matt Jordan + + * Update UPGRADE-13.txt file + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-08 15:08 +0000 [3e452fa4d9] Jason Parker + + * Fix build in devmode. + ........ + + Merged revisions 420592 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-08 14:16 +0000 [5ce4ad8031] Jason Parker + + * app_voicemail: Add the ability to specify multiple email addresses. + + ASTERISK-24045 + Reported by: Jacob Barber + Review: https://reviewboard.asterisk.org/r/3833/ + ........ + + Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-08 12:53 +0000 [91f7b66183] Matt Jordan + + * chan_sip: Mark chan_sip and its files as extended support + ........ + + Merged revisions 420562 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-08 07:40 +0000 [86e927a714] Matt Jordan + + * make_ari_stubs: Update wiki prefix to '13' + ........ + + Merged revisions 420538 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-08 07:38 +0000 [1f35fccda1] Matt Jordan + + * res_ari_resource.c.mustache: Update template to emit module support level + ........ + + Merged revisions 420536 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-08 07:33 +0000 [008c1ad9bf] Matt Jordan + + * main/message: remove debug message + ........ + + Merged revisions 420533 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420534 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420535 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-07 22:07 +0000 [c94fef6f36] Kinsey Moore + + * CEL: Update unit tests for additional information + + This updates the CEL unit tests for the new information contained in + the attended transfer CEL extra field. + ........ + + Merged revisions 420513 from http://svn.asterisk.org/svn/asterisk/branches/12 + ........ + + Merged revisions 420514 from http://svn.asterisk.org/svn/asterisk/branches/13 + + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-07 20:37 +0000 [96be6b2228] Matt Jordan + + * Initialize svnmerge from branches/13 + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-07 20:36 +0000 [38a0df95b1] Matt Jordan + + * Remove 12 merge properties + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-08-07 20:33 +0000 [5760526f69] Matt Jordan + + * Update UPGRADE.txt for 13 branch + + git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 + +2014-10-24 Asterisk Development Team + + * Asterisk 13.0.0 Released. + +2014-10-22 21:27 +0000 [r426097] Shaun Ruffell + + * codecs/codec_dahdi.c: codec_dahdi: Cannot use struct + ast_translator.core_{src,src}_codec. This fixes a Segmentation + fault introduced in r419044 "media formats: re-architect handling + of media for performance improvements". The problem is that + codec_dahdi was using core_src_codec and core_dst_codec in the + ast_translator structure when these fields were never set. Now + instead of trying to map the new core codec descriptions to the + way DAHDI defines different codecs, we will store the DAHDI + specific formats in 'struct translator' directly so we can refer + to them without mapping. This also allows us to remove the + "global_format_map" structure, since we can now query the list of + translators directly to make sure we do not ever register a DAHDI + based translator for a specific path more than once and eliminate + the need to keep the list and the map in sync. ASTERISK-24435 + #close Reported by: Marian Koniuszko Review: + https://reviewboard.asterisk.org/r/4105/ + +2014-10-21 17:47 +0000 [r426079] Richard Mudgett + + * main/translate.c: translage.c: Fix regression when generating + translation path strings. Fix the AMI Status action read and + write translation path strings from growing for each channel in + the status event list by reseting the ast string given to + ast_translate_path_to_str() to fill in the given translation + path. + +2014-10-20 14:15 +0000 [r425991] Matthew Jordan + + * res/res_xmpp.c, main/tcptls.c, /: AST-2014-011: Fix POODLE + security issues There are two aspects to the vulnerability: (1) + res_jabber/res_xmpp use SSLv3 only. This patch updates the module + to use TLSv1+. At this time, it does not refactor + res_jabber/res_xmpp to use the TCP/TLS core, which should be done + as an improvement at a latter date. (2) The TCP/TLS core, when + tlsclientmethod/sslclientmethod is left unspecified, will default + to the OpenSSL SSLv23_method. This method allows for all + encryption methods, including SSLv2/SSLv3. A MITM can exploit + this by forcing a fallback to SSLv3, which leaves the server + vulnerable to POODLE. This patch adds WARNINGS if a user uses + SSLv2/SSLv3 in their configuration, and explicitly disables + SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk + will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly + chosen. For TLS servers, Asterisk will no longer support SSLv2 or + SSLv3. Much thanks to abelbeck for reporting the vulnerability + and providing a patch for the res_jabber/res_xmpp modules. + Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 + #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, + gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by + abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch + uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff + uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded + by mjordan (License 6283) ........ Merged revisions 425987 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-19 17:07 +0000 [r425965] George Joseph + + * Makefile, /, configure, include/asterisk/autoconfig.h.in, + configure.ac, makeopts.in: build: Force -fsigned-char on + platforms where the default for char is unsigned gcc on the ARM + platform defaults 'char' to 'unsigned char' whereas Intel and + SPARC default to 'signed char'. This is only an issue in the rare + cases where negative values are assigned to a 'char' but this + this patch insures compatibility by detecting platforms that + default to 'unsigned' and adding an '-fsigned-char' flag to + _ASTCFLAGS. If compiling for ARM (native or cross-compile) be + sure to run ./bootstrap.sh and ./configure to regenerate the + build files. You shouldn't have to do this for Intel or SPARC. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4091/ ........ Merged + revisions 425964 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-19 04:01 +0000 [r425923-425944] Matthew Jordan + + * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert 425922 + This patch for r425922 introduced a bug, wherein sending an + INVITE request with no SDP would cause Asterisk to not send an + SDP Offer in the 200 OK. The current structure of + res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as + create_outgoing_sdp has no knowledge of whether or not it is + creating an SDP as a new Offer or an Answer. This is something of + an oversight in the callback definition, as the caller of it does + have this information. + + * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Remove left over + reference to override_prefs The usage of the local override_prefs + variable in create_outgoing_sdp_stream was previously to track an + override format preference set by PJSIP_MEDIA_OFFER. Now, + however, that function simply sets the joint capabilities + structure, session->req_caps. During the media format rework, the + override_prefs was instead used to check if there were any + formats in session->req_caps. However, this usage isn't useful in + create_outgoing_sdp_stream. session->req_caps contains the + negotiated formats for *all* streams, not just the current one + being created. Thus, so long as any stream of any type has + provided a format, override_prefs will be non-zero. Hence, its + usage in checking whether or not we should look at the formats on + the endpoint or the joint capabilities is generally useless. + There's only two things useful to check: (1) Does the endpoint + have a format for the media type? (2) Did we negotiate a format + for the media type? If either of those is a 'no', then we must + kill the media stream. + +2014-10-17 22:43 +0000 [r425905] Jonathan Rose + + * configs/samples/cli_aliases.conf.sample: Sample Configurations: + make 'pjsip reload' reload all reloadable pjsip modules AST-1432 + #close Reported by: John Bigelow + +2014-10-17 13:35 +0000 [r425821-425879] Matthew Jordan + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip.c, + res/res_pjsip_session.c, /: res_pjsip_session/res_pjsip_sdp_rtp: + Be more tolerant of offers When an inbound SDP offer is received, + Asterisk currently makes a few incorrection assumptions: (1) If + the offer contains more than a single audio/video stream, + Asterisk will reject the entire stream with a 488. This is an + overly strict response; generally, Asterisk should accept the + media streams that it can accept and decline the others. (2) If + the offer contains a declined media stream, Asterisk will attempt + to process it anyway. This can result in attempting to match + format capabilities on a declined media stream, leading to a 488. + Asterisk should simply ignore declined media streams. (3) + Asterisk will currently attempt to handle offers with AVPF with + use_avpf=No/AVP with use_avpf=Yes. This mismatch results in + invalid SDP answers being sent in response. If there is a + mismatch between the media type being offered and the + configuration, Asterisk must reject the offer with a 488. This + patch does the following: * Asterisk will accept SDP offers with + at least one media stream that it can use. Some WARNING messages + have been dropped to NOTICEs as a result. * Asterisk will not + accept an offer with a media type that doesn't match its + configuration. * Asterisk will ignore declined media streams + properly. #SIPit31 Review: + https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close + Reported by: James Van Vleet ASTERISK-24381 #close Reported by: + Matt Jordan ........ Merged revisions 425868 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy + setting when sending qualify requests The outboundproxy setting + is currently ignored when sending OPTIONS requests as a result of + the qualify setting. This means that if an Asterisk server is + unable to send the packet directly to a peer, it is unable to + qualify any non-inbound registered peer (e.g. a peer SIP Trunk). + This patch grabs the outboundproxy information for a peer when a + qualify attempt is being constructed and, if it finds the + information, uses it when sending the OPTIONS request. Review: + https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close + Reported by: Damian Ivereigh patches: outboundproxy-dai.patch + uploaded by Damian Ivereigh (License 6632) ........ Merged + revisions 425818 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425819 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425820 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-17 02:41 +0000 [r425783] Richard Mudgett + + * main/core_unreal.c, main/channel.c, /: AMI: Add missing VarSet + events when a channel inherits variables. There should be AMI + VarSet events when channel variables are inherited by an outgoing + channel. Also local;2 should generate VarSet events when it gets + all of its channel variables from channel local;1. ASTERISK-24415 + #close Reported by: Richard Mudgett Patches: + jira_asterisk_24415_v12.patch (license #5621) patch uploaded by + Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/ + ........ Merged revisions 425782 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-17 01:57 +0000 [r425736-425761] Matthew Jordan + + * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio + issues when moving from remote bridge to softmix When a native + RTP bridge that is remotely bridging its participants switches to + a softmix bridge, it may not properly re-INVITE the media for one + or both participants back to Asterisk. This is due to the current + bridge_native_rtp code only re-INVITEs if it believes the channel + will survive the bridge operation. Currently, that code is + failing, as it expects the channels to have a soft hangup flag + set on it indicating that a redirect has occurred or that the + channel is going to leave the bridge. (The code did not take into + account a smart bridge operation). This patch also renames a few + things to be more reflective of the underlying types. Review: + https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close + ........ Merged revisions 425760 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_cel.c: test_cel: Update pickup test to expect + CANCEL instead of ANSWSER The CEL pickup test previously looked + for a disposition of ANSWER between the original caller/peer when + the call is picked up. This is actually incorrect: the + disposition should, at the very least, not be ANSWER as the call + was never ANSWERed. The disposition is now CANCEL; this patch + updates the test accordingly. ........ Merged revisions 425757 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: main/cdr: Use 'time' when rescheduling batched + CDRs as opposed to 'size' When refactoring CDRs to use the + configuration framework, a 'whoops' was introduced where the CDR + batch size was used when rescheduling a batch, as opposed to the + time duration. This patch corrects that obvious mistake. + ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged + revisions 425735 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 17:30 +0000 [r425714] George Joseph + + * include/asterisk/config.h, tests/test_config.c, main/config.c, /: + config: Fix inf loop using ast_category_browse and + ast_variable_retrieve Fix infinite loop when calling + ast_variable_retrieve inside an ast_category_browse loop when + there is more than 1 category with the same name. Tested-by: + George Joseph Review: https://reviewboard.asterisk.org/r/4089/ + ........ Merged revisions 425713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 14:35 +0000 [r425691] Kinsey Moore + + * res/res_pjsip_t38.c, res/res_pjsip_registrar_expire.c, + res/res_pjsip_mwi_body_generator.c, + res/res_pjsip_endpoint_identifier_user.c, + res/res_pjsip_send_to_voicemail.c, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip_endpoint_identifier_anonymous.c, + res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c, + res/res_pjsip_acl.c, res/res_pjsip_pubsub.c, + res/res_pjsip_diversion.c, res/res_pjsip_refer.c, + include/asterisk/res_pjsip.h, + res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c, + res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c, + res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c, + res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c, + res/res_pjsip_logger.c, res/res_pjsip_nat.c, + res/res_pjsip_session.c, res/res_pjsip_exten_state.c, + res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c, + res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c, + res/res_pjsip_dialog_info_body_generator.c, + res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_registrar.c, + channels/chan_pjsip.c, res/res_pjsip_transport_websocket.c, + res/res_pjsip_pidf_eyebeam_body_supplement.c, + include/asterisk/res_pjsip_session.h, /, res/res_pjsip_notify.c, + res/res_pjsip_pidf_digium_body_supplement.c, + res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip_publish_asterisk.c: PJSIP: Enforce module load + dependencies This enforces that res_pjsip, res_pjsip_session, and + res_pjsip_pubsub have loaded properly before attempting to load + any modules that depend on them since the module loader system is + not currently capable of resolving module dependencies on its + own. ASTERISK-24312 #close Reported by: Dafi Ni Review: + https://reviewboard.asterisk.org/r/4062/ ........ Merged + revisions 425690 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 06:11 +0000 [r425669] Igor Goncharovskiy + + * channels/chan_unistim.c, /: Fix loss of voice after second call + drops (on a second line) in case using multiple lines on unistim + phones. There is regression was introduced in r391379. Reported + by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........ + Merged revisions 425667 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425668 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-16 01:25 +0000 [r425646] Joshua Colp + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix a bug where ICE + state would get reset when it shouldn't. In the case where the + ICE negotiation had not yet started current state would get wiped + when it shouldn't. This also removes channel binding as in + practice this does not work well with other implementations. + ........ Merged revisions 425644 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425645 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-15 19:31 +0000 [r425627] Richard Mudgett + + * channels/chan_motif.c: chan_motif: Cleanup + jingle_tech.capabilities only once. + +2014-10-15 19:05 +0000 [r425611] Jonathan Rose + + * res/parking/parking_tests.c: parking_tests: Fix assertions and + possibly crashes in res_parking unit tests Assertions were caused + by attempting to play music on hold to a channel with no formats. + Parking unit test channels were given formats and a technology so + that they would be able to pretend to read/write frames. + ASTERISK-24413 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/4075/ + +2014-10-15 09:59 +0000 [r425590] Alexandr Anikin + + * addons/chan_ooh323.c, /: chan_ooh323: fix rtptimeout general + value checking correct condition to check rtptimeout in [general] + config section ASTERISK-24393 #close Reported by: Dmitry Melekhov + Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........ + Merged revisions 425547 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425548 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425589 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 20:46 +0000 [r425526] George Joseph + + * /, include/asterisk/config.h, tests/test_config.c, main/config.c: + config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG + the /main/config config_basic_ops test was causing a SEGV while + doing an ast_category_delete in an ast_category_browse loop. + Apparently this never worked but was also never tested. I removed + the test, added 2 notes to config.h indicating that it's not + supported and added a few lines of code to ast_category_delete to + prevent the SEGV should someone attempt it in the future. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4078/ ........ Merged + revisions 425525 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 19:00 +0000 [r425504] Jonathan Rose + + * main/sched.c, /: Scheduler: Fix a nasty scheduler caching bug + which makes new tasks not execute Tasks that were marked for + pending deletion in the scheduler would be moved to the cache for + later reuse, but after being recycled the deleted mark wouldn't + be removed resulting in fresh tasks being deleted without + reason... and immediately moved back into the cache where they + could be reused again. This could cause horrendous things to + happen in just about anything that used a scheduler. + ASTERISK-24321 #close Reported by: Steve Pitts Review: + https://reviewboard.asterisk.org/r/4071/ ........ Merged + revisions 425503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 18:12 +0000 [r425481] George Joseph + + * res/res_phoneprov.c, include/asterisk/phoneprov.h, /, + res/res_pjsip_phoneprov_provider.c: res_phoneprov: Create + accessor for ast_phoneprov_std_variable_lookup Based on feedback + from Richard, I created an accessor for + res_phoneprov/ast_phoneprov_std_variable_lookup and added load + priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by: + Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/ + ........ Merged revisions 425480 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 16:46 +0000 [r425459] Corey Farrell + + * /, res/res_fax.c: res_fax: Fix reference leak caused by gateway + sessions Fax gateway session objects can be re-used, causing the + same gateway session to be added to faxregistry.container more + than once. This change causes fax_session_new to remove the + reserved session from the container before it's id is changed, + ensuring it's possible for the session to be freed. + ASTERISK-24392 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4049/ ........ Merged + revisions 425457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 16:35 +0000 [r425455] Richard Mudgett + + * /, main/stasis_channels.c: stasis_channels.c: Resolve unfinished + Dials when doing masquerades (Part 2) Masquerades into and out of + channels that are involved in a dial operation don't create the + expected dial end event. The missing dial end event goes against + the model for things like CDRs and generating Dial end manager + actions and such. There are four cases: 1) A channel masquerades + into the caller channel. The case happens when performing a + blonde transfer using the channel driver's protocol. 2) A channel + masquerades into a callee channel. The case happens when + performing a directed call pickup. 3) The caller channel + masquerades out of dial. The case happens when using the Bridge + application on the caller channel. 4) A callee channel + masquerades out of dial. The case happens when using the Bridge + application on a peer channel. As it turned out, all four cases + need to be handled instead of just the first one. ASTERISK-24237 + Reported by: Richard Mudgett ASTERISK-24394 #close Reported by: + Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/ + ........ Merged revisions 425430 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-14 16:19 +0000 [r425415] Corey Farrell + + * /, res/res_fax.c: res_fax: Resolve module reference leak caused + by reserved sessions Remove reference to module providing + reserved session after adding a reference to the final module. + This re-reference is done to ensure that module references are + correct even if the final session selects a different module than + the reserved session. ASTERISK-18923 #close Reported by: Grigoriy + Puzankin Review: https://reviewboard.asterisk.org/r/4048/ + ........ Merged revisions 425405 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425407 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425411 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-13 16:10 +0000 [r425384] George Joseph + + * apps/app_directory.c, tests/test_sorcery.c, main/config.c, + tests/test_sorcery_realtime.c, res/res_sorcery_realtime.c, + apps/app_voicemail.c, res/res_sorcery_config.c, main/manager.c, + /, include/asterisk/config.h, pbx/pbx_realtime.c, + tests/test_config.c: manager/config: Support templates and + non-unique category names via AMI This patch provides the + capability to manipulate templates and categories with non-unique + names via AMI. Summary of changes: GetConfig and GetConfigJSON: + Added "Filter" parameter: A comma separated list of + name_regex=value_regex expressions which will cause only + categories whose variables match all expressions to be + considered. The special variable name TEMPLATES can be used to + control whether templates are included. Passing 'include' as the + value will include templates along with normal categories. + Passing 'restrict' as the value will restrict the operation to + ONLY templates. Not specifying a TEMPLATES expression results in + the current default behavior which is to not include templates. + UpdateConfig: NewCat now includes options for allowing duplicate + category names, indicating if the category should be created as a + template, and specifying templates the category should inherit + from. The rest of the actions now accept a filter string as + defined above. If there are non-unique category names, you can + now update specific ones based on variable values. To facilitate + the new capabilities in manager, corresponding changes had to be + made to config, most notably the addition of filter criteria to + many of the APIs. In some cases it was easy to change the + references to use the new prototype but others would have + required touching too many files for this patch so a wrapper with + the original prototype was created. Macros couldn't be used in + this case because it would break binary compatibility with + modules such as res_digium_phone that are linked to real symbols. + Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4033/ ........ Merged + revisions 425383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-12 21:09 +0000 [r425362] Joshua Colp + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Make the ICE + transport check case insensitive as some implementations use + 'udp'. ........ Merged revisions 425360 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425361 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-12 08:15 +0000 [r425289-425299] Walter Doekes + + * /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send + reINVITE after a BYE. After a reINVITE glare situation, Asterisk + would re-send the reINVITE even though the call had been hung up + in the mean time. This patch unschedules the reinvite when + handling the BYE. ASTERISK-22791 #close Reported by: Paolo + Compagnini Tested by: Paolo Compagnini Review: + https://reviewboard.asterisk.org/r/4056/ (testcase is in review + r4055) ........ Merged revisions 425296 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425297 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425298 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, Makefile: build: Relax badshell tilde test to allow for ~ in + middle of DESTDIR. The main Makefile has a target test called + 'badshell' that tests if DESTDIR does not happen to have an + an-expanded tilde (~). This might be the case if you run: make + install DESTDIR=~/somewhere/ That test also disallowed valid + tildes in directory names. The test is now changed to only + trigger on a tilde at the start of the path. ASTERISK-13797 + #close Reported by: Tzafrir Cohen Review: + https://reviewboard.asterisk.org/r/4064/ ........ Merged + revisions 425291 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425292 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425293 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_calendar_ews.c: res_calendar_ews: Relax neon version + check to work with 0.30 too. Allow res_calendar_ews to work not + only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close + Reported by: Tzafrir Cohen Review: + https://reviewboard.asterisk.org/r/4068/ ........ Merged + revisions 425286 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425287 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425288 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-11 21:08 +0000 [r425265] George Joseph + + * /, res/res_phoneprov.c: res_phoneprov: Cleanup module load error + handling Tested module load/reload interaction between + res_phoneprov and res_pjsip_phoneprov_provider in cases where + res_phoneprov didn't load correctly (usually misconfiguration or + missing phoneprov.conf) Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/4069/ ........ Merged + revisions 425264 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 20:48 +0000 [r425243] Joshua Colp + + * /, main/bridge.c, bridges/bridge_native_rtp.c: bridge: During a + smart bridge operation provide a more complete bridge to the old + technology. When a smart bridge operation occurs and a bridge + transitions from one technology to another the old technology is + provided the channels formerly in it and told that they are + leaving. Unfortunately the bridge provided along with them is + incomplete. The bridge, despite there being channels in it, + contains none. This forces technology implementations to have + additional logic when channels are leaving or to store their own + duplicated state. This change makes the bridge more complete so + it contains the expected channels. Now that the bridge is + complete special logic within bridge_native_rtp is no longer + needed and has been removed. Review: + https://reviewboard.asterisk.org/r/4057/ ........ Merged + revisions 425242 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 14:31 +0000 [r425221] Matthew Jordan + + * /, res/res_phoneprov.c: res/res_phoneprov: Bail on registration + if res_phoneprov didn't load If res_phoneprov failed to fully + load (due to not being configured), the providers container will + be NULL. If a module attempts to register a phone provisioning + provider, it should check for the presence of the container. If + there is no providers container, it should return an error. This + patch makes the ast_phoneprov_provider_register function do + that... otherwise this would be a silly commit message. ........ + Merged revisions 425220 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 14:23 +0000 [r425217] Joshua Colp + + * /, res/res_pjsip_phoneprov_provider.c: + res_pjsip_phoneprov_provider: Add missing dependency on + pjproject. ........ Merged revisions 425216 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 13:01 +0000 [r425155] Kinsey Moore + + * /, tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing + regression This fixes a regression in callerid parsing introduced + when another bug was fixed. This bug occurred when the name was + composed entirely of DTMF keys and quoted without a number + section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard + Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by + Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ + ........ Merged revisions 425152 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425153 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425154 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 12:10 +0000 [r425132] Joshua Colp + + * res/res_pjsip_nat.c, /: res_pjsip_nat: Place source port into + rport of responses if 'force_rport' is on. When the 'force_rport' + option is enabled the behavior should be the same as if the + remote side placed rport into the message themselves. Therefore + any responses we send should include the source port of the + request in the rport of the Via header. #SIPit31 ASTERISK-24387 + #close Reported by: Matt Jordan ........ Merged revisions 425131 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-10 07:32 +0000 [r425071] Walter Doekes + + * /, channels/chan_sip.c: chan_sip: Fix dialog leak resulting from + missing ACK to re-INVITE. If a device re-INVITEs at the same time + as the dialog is hung up, and if then the ACK to the re-INVITE + never reaches Asterisk, chan_sip would fail to destroy the dialog + after a while. This resulted in (most prominently) file handle + leaks. (Patch reindented by me.) ASTERISK-20784 #close + ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal + Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle + (License #5334) patch_asterisk_20784.txt uploaded by Nitesh + Bansal (License #6418) Reviewboard: + https://reviewboard.asterisk.org/r/4052/ (testcase can be found + at r4051) ........ Merged revisions 425068 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 425069 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 425070 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 23:35 +0000 [r425052] George Joseph + + * res/res_pjsip_phoneprov_provider.c: res_pjsip_phoneprov_provider: + fix compile breakage on AST_VECTOR endpoint->inbound_auths was + changed to a vector in 13 and I committed the 12 patch instead of + the 13 patch. Tested-by: George Joseph + +2014-10-09 21:38 +0000 [r425031] Kevin Harwell + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Crash if no + candidates received for component When starting ice if there is + not at least one remote ice candidate with an RTP component + asterisk will crash. This is due to an assertion in pjnath as it + expects at least one candidate with an RTP component. Added a + check to make sure at least one candidate contains an RTP + component and at least one candidate has an RTCP component. + ASTERISK-24383 #close Review: + https://reviewboard.asterisk.org/r/4039/ ........ Merged + revisions 425030 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 20:54 +0000 [r425008] George Joseph + + * /, res/res_pjsip_phoneprov_provider.c (added), + configs/samples/pjsip.conf.sample: res_pjsip_phoneprov_provider: + Provides pjsip integration with res_phoneprov This module allows + res_pjsip to integrate with res_phoneprov. It handles the pjsip + 'phoneprov' object type. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/3976/ ........ Merged + revisions 425007 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 18:37 +0000 [r424986] Matthew Jordan + + * /, res/res_phoneprov.c: res/res_phoneprov: Don't cancel Asterisk + load on module load failure ........ Merged revisions 424985 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 17:45 +0000 [r424964] George Joseph + + * include/asterisk/phoneprov.h (added), /, + configs/samples/phoneprov.conf.sample, + include/asterisk/chanvars.h, res/res_phoneprov.c, + res/res_phoneprov.exports.in (added), main/chanvars.c: + res_phoneprov: Refactor phoneprov to allow pluggable config + providers This patch makes res_phoneprov more modular so other + modules (like pjsip) can provide configuration information + instead of res_phoneprov relying solely on users.conf and + sip.conf. To accomplish this a new ast_phoneprov public API is + now exposed which allows config providers to register themselves, + set defaults (server profile, etc) and add user extensions. * + ast_phoneprov_provider_register registers the provider and + provides callbacks for loading default settings and loading + users. * ast_phoneprov_provider_unregister clears the defaults + and users. * ast_phoneprov_add_extension should be called once + for each user/extension by the provider's load_users callback to + add them. * ast_phoneprov_delete_extension deletes one extension. + * ast_phoneprov_delete_extensions deletes all extensions for the + provider. Tested-by: George Joseph Review: + https://reviewboard.asterisk.org/r/3970/ ........ Merged + revisions 424963 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 16:36 +0000 [r424942] Richard Mudgett + + * /, main/cdr.c: cdr.c: Make turning on CDR debug a one step + process instead of two. Now "cdr set debug on" doesn't also + require "core set verbose 1" to see CDR debug output. ........ + Merged revisions 424941 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-09 08:08 +0000 [r424880] Walter Doekes + + * /, contrib/scripts/safe_asterisk: safe_asterisk: Don't + automatically exceed MAXFILES value of 2^20. On systems with lots + of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can + exceed the per-process file limit of 2^20. This patch ensures the + value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close + Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff + uploaded by Michael Myles (License #6626) ........ Merged + revisions 424875 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 424878 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424879 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-08 18:46 +0000 [r424854] Joshua Colp + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Allow only UDP ICE + candidates. The underlying library, pjnath, that res_rtp_asterisk + uses for ICE support does not have support for ICE-TCP. As + candidates are passed through directly to it this can cause error + messages to occur when it receives something unexpected (such as + a TCP candidate). This change merely ignores all non-UDP + candidates so they never reach pjnath. ASTERISK-24326 #close + Reported by: Joshua Colp ........ Merged revisions 424852 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424853 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-08 18:24 +0000 [r424769-424850] Kinsey Moore + + * main/stasis.c: Stasis: Relegate log message to dev-mode This + error message primarily applies to development tasks and will now + only show up when dev-mode is enabled via configure. + + * main/sounds_index.c: Indexer: Format message types may not exist + In Asterisk 13+, any given message type is not guaranteed to + exist even if Asterisk comes up correctly since creation of the + message type could be declined. The indexer should not prevent + Asterisk from starting under these conditions. + + * main/stasis.c: Stasis: Only log errors for non-declined types + When message type creation is declined via stasis.conf, certain + operations log errors assuming that the declined type is being + used before initialization or after destruction. These error + messages get quite spammy for oft used message types and should + not be logged in the first place since the message type is + validly NULL. Reported by: Matt DiMeo + +2014-10-07 18:33 +0000 [r424752] Joshua Colp + + * main/data.c: data: Properly access formats in capabilities + structure when adding codecs. Formats within a capabilities + structure are addressed starting at 0, not 1. Assuming 1 causes + it to exceed an array. ASTERISK-24389 #close Reported by: Kevin + Harwell + +2014-10-07 17:41 +0000 [r424692-424731] Matthew Jordan + + * /, res/res_pjsip_outbound_registration.c: + res/res_pjsip_outbound_registration: Initialize + auth_reject_permanent parameter Prior to this patch, the + auth_reject_permanent parameter was not initialized on the + registration client state, leading to the parameter being + disabled regardless of the value specified in pjsip.conf. This + patch initialized the setting on the registration client state to + the provided configuration value. ASTERISK-24398 #close ........ + Merged revisions 424730 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Fix typo in WARNING + message + + * main/message.c, /: message: Don't close an AMI connection on + SendMessage action error If SendMessage encounters an error (such + as incorrect input provided to the action), it will currently + return -1. Actions should only return -1 if the connection to the + AMI client should be closed. In this case, SendMessage causing + the client to disconnect is inappropriate. This patch causes the + action to return 0, which simply causes the action to fail. + Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 + #close Reported by: Peter Katzmann patches: sendMessage.patch + uploaded by Peter Katzmann (License 5968) ........ Merged + revisions 424690 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424691 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-06 15:38 +0000 [r424669] Richard Mudgett + + * main/features.c, /: features.c: Fix lingering channel ref while + Bridge() application is active. Using the Bridge application to + bridge a channel that is executing an applicaiton such as Wait + results in a lingering Surrogate channel in the CLI "core show + channels" output even though it has already hungup. * Fix + bridge_exec() to not hold onto the current_dest_chan ref once it + has been put into the bridge. * Eliminated bridge_exec()'s use of + RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson + Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged + revisions 424668 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-06 12:38 +0000 [r424601-424647] Matthew Jordan + + * /, main/sdp_srtp.c: sdp_srtp: Add new lines to some WARNING + messages ........ Merged revisions 424646 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_options.c: res_pjsip/pjsip_options: Do not + 404 an OPTIONS request not sent to an endpoint An OPTIONS request + that is sent to Asterisk but not to a specific endpoint is + currently sent a 404 in response. This is because, not + surprisingly, an empty extension is never going to be found in + the dialplan. This patch makes it so that we only attempt to look + up the endpoint in the dialplan if it is specified in the OPTIONS + request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt + Jordan ........ Merged revisions 424624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions: + Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels Calling + PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your + health. It will treat the channels as a PJSIP channel, eventually + hitting an ao2 error, FRACKing on assertion error, and quite + likely crashing. This patch adds checks to the read/write + callbacks that ensure that the channel technology is of type + 'PJSIP' before attempting to operate on the channel. #SIPit31 + ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged + revisions 424621 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_hep_pjsip.c, res/res_pjsip/pjsip_distributor.c, + res/res_pjsip_logger.c: res_pjsip: Prevent crashes when PJPROJECT + presents an rdata with no message When a message that exceeds the + PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible + (although it shouldn't occur) for pjproject to pass up an rdata + object with a NULL msg in the msg_info. Needless to say, things + that attempt to dereference this are in for a rough ride. In + particular, this caused crashes in three different locations, all + of which are 'low level' enough to intercept an rdata object + early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) + res_pjsip/distributor Anything that can intercept an rdata object + before res_pjsip/distributor should be defensive when looking at + the received packet. #SIPit31 ASTERISK-24369 #close Reported by: + Matt Jordan ........ Merged revisions 424618 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Gracefully handle + errors when re-creating subscriptions A subscription that has + been persisted can - for various reasons - fail to be re-created + on startup. This patch resolves a number of crashes that occurred + when a subscription cannot be re-created on several off-nominal + paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan + +2014-10-05 00:48 +0000 [r424552-424580] Corey Farrell + + * main/manager.c, /: Release AMI connections on shutdown. + ASTERISK-24378 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4037/ ........ Merged + revisions 424578 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_motif.c: chan_motif: Correct last commit to use + ao2_cleanup to free format cap This fix applies to 13 and trunk. + ASTERISK-24384 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4043/ + + * /, channels/chan_motif.c: chan_motif: Release format capabilities + and config on module load error ASTERISK-24384 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/4043/ ........ Merged + revisions 424550 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424551 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 21:56 +0000 [r424472-424529] Richard Mudgett + + * /, CHANGES, res/res_pjsip.c: res_pjsip: Fix XML typo and update + CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, /, + main/framehook.c: audiohooks: Reevaluate the bridge technology + when an audiohook is added or removed. Adding a mixmonitor to a + channel causes the bridge to change technologies from native to + simple_bridge so the call can be recorded. However, when the + mixmonitor is stopped the bridge does not switch back to the + native technology. * Added unbridge requests to reevaluate the + bridge when a channel audiohook is removed. * Moved the unbridge + request into ast_audiohook_attach() ensure that the bridge + reevaluates whenever an audiohook is attached. This simplified + the mixmonitor and chan_spy start code as well. * Added defensive + code to stop_mixmonitor_full() in case additional arguments are + ever added to the StopMixMonitor application. * Made + ast_framehook_detach() not do an unbridge request if the + framehook does not exist. * Made ast_framehook_list_fixup() do an + unbridge request if there are any framehooks. Also simplified the + loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/4046/ ........ Merged + revisions 424506 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/core_unreal.c, main/taskprocessor.c, channels/chan_iax2.c, + res/res_pjsip_session.c, main/channel.c, channels/chan_misdn.c, + channels/chan_skinny.c, funcs/func_frame_trace.c, + channels/chan_motif.c, include/asterisk/frame.h, + main/bridge_channel.c, channels/chan_pjsip.c, + channels/chan_unistim.c, include/asterisk/res_pjsip_session.h, + addons/chan_ooh323.c, /, include/asterisk/taskprocessor.h, + channels/chan_sip.c, res/res_pjsip_session.exports.in: + chan_pjsip: Fix deadlock when masquerading PJSIP channels. + Performing a directed call pickup resulted in a deadlock when + PJSIP channels were involved. A masquerade needs to hold onto the + channel locks while it swaps channel information between the two + channels involved in the masquerade. With PJSIP channels, the + fixup routine needed to push a fixup task onto the PJSIP + channel's serializer. Unfortunately, if the serializer was also + processing a task that needed to lock the channel, you get + deadlock. * Added a new control frame that is used to notify the + channels that a masquerade is about to start and when it has + completed. * Added the ability to query taskprocessors if the + current thread is the taskprocessor thread. * Added the ability + to suspend/unsuspend the PJSIP serializer thread so a masquerade + could fixup the PJSIP channel without using the serializer. + ASTERISK-24356 #close Reported by: rmudgett Review: + https://reviewboard.asterisk.org/r/4034/ ........ Merged + revisions 424471 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 15:54 +0000 [r424448] George Joseph + + * /, main/sorcery.c: sorcery: Prevent SEGV in sorcery_wizard_create + when there's no create function When you call + ast_sorcery_create() you don't necessarily know which wizard is + going to be invoked. If it happens to be a wizard like 'config' + that doesn't have a 'create' virtual function you get a segfault + in the sorcery_wizard_create callback. This patch catches the + null function pointer, does an ast_assert, and logs an error. + Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged + revisions 424447 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 13:58 +0000 [r424424-424427] Kinsey Moore + + * configs/samples/pjsip.conf.sample, /, + res/res_pjsip/pjsip_configuration.c: PJSIP: Restore functional + default for callerid_privacy The pjsip config option default + fixups from r424263 altered the functional default from + "allowed_not_screened" to "allowed". This change restores the + functional default value when none is provided. ........ Merged + revisions 424426 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: Manager: Add missing fields and documentation + for CoreShowChannels This corrects some issues introduced in the + responses to the CoreShowChannels AMI command as well as adding + documentation for the responses. The command in Asterisk 12 was + missing the following fields: Duration, Application, + ApplicationData, and BridgedChannel and BridgedUniqueID (replaced + with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn + Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged + revisions 424423 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-03 07:54 +0000 [r424415] Joshua Colp + + * res/res_pjsip_session.c, /: res_pjsip_session: Reduce SDP size by + removing duplicate connection lines. Due to the architecture of + how media streams are handled each individual handler adds + connection details (IP address) for it. The first media stream is + then used as the top level SDP connection line. In practice each + line ends up being the same so to reduce the SDP size + stream-level connection information is also added to the SDP if + it differs from the top level SDP connection line. ........ + Merged revisions 424414 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-02 21:52 +0000 [r424394] Richard Mudgett + + * /, configs/samples/pjsip.conf.sample, res/res_pjsip.c, + res/res_pjsip/config_transport.c: res_pjsip: Make transport + cipher option accept a comma separated list of cipher names. + Improvements to the res_pjsip transport cipher option. * Made the + cipher option accept a comma separated list of OpenSSL cipher + names. Users of realtime will be glad if they have more than one + name to list. * Added the CLI command 'pjsip list ciphers' so a + user can know what OpenSSL names are available for the cipher + option. * Updated the cipher option online XML documentation to + specify what is expected for the value. * Updated + pjsip.conf.sample to not indicate that ALL is acceptable since + ALL does not imply a preference order for the ciphers and PJSIP + does not simply pass the string to OpenSSL for interpretation. + ASTERISK-24199 #close Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/4018/ ........ Merged + revisions 424393 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-02 20:15 +0000 [r424373] Jonathan Rose + + * /, + contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py + (added): Alembic: Add enumerator value to sippeers -> directmedia + - 'outgoing' The 'outgoing' value was left off of the enumerator + when first creating the column. This patch adds it, and should + gracefully upgrade keeping the existing data in tact. + ASTERISK-23781 #close Reported by: Stephen More Review: + https://reviewboard.asterisk.org/r/4013/ ........ Merged + revisions 424372 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-02 13:35 +0000 [r424338] Scott Griepentrog + + * /, configs/samples/pjsip.conf.sample: res_pjsip: document use of + rewrite_contact in sample conf Without setting rewrite_contact, + an invite to an endpoint behind NAT will not reach it - unless + the endpoint itself uses STUN or TURN to discover it's public + URI. Thus, the use of this should be in the sample documentation. + Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged + revisions 424337 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 22:52 +0000 [r424333] Jonathan Rose + + * channels/chan_pjsip.c: chan_pjsip: Fix an assertion for channels + that lack formats on creation ASTERISK-24222 #close Reported by: + Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/ + +2014-10-01 20:36 +0000 [r424313] Corey Farrell + + * res/res_hep.c, /: res_hep: Release allocation reference to + configuration. ASTERISK-24362 #close Reported by: Corey Farrell + Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged + revisions 424312 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 16:37 +0000 [r424288-424291] Joshua Colp + + * /, res/res_pjsip/pjsip_configuration.c, + configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip: + Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. + During the latest update to DTLS-SRTP support the ability to + configure the hash used for fingerprints was added. This gave us + two supported ones: SHA-1 and SHA-256. The default was + accordingly updated to SHA-256. Unfortunately this configuration + ability was not exposed within res_pjsip. This change adds a + dtls_fingerprint option that controls it. #SIPit31 ........ + Merged revisions 424290 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Accept DTLS + attributes in top level, not just media session. #SIPit31 + ........ Merged revisions 424287 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 12:27 +0000 [r424245-424266] Kinsey Moore + + * res/res_pjsip/config_transport.c, /, res/res_pjsip/location.c, + res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip/pjsip_configuration.c, + configs/samples/pjsip.conf.sample: PJSIP: Handle defaults + properly This updates the code behind PJSIP configuration options + with custom handlers to deal with the assigned default values + properly where it makes sense and adjusting the default value + where it doesn't. Before applying this patch, there were several + cases where the default value for an option would prevent that + config section from loading properly. Reported by: Thomas + Thompson Review: https://reviewboard.asterisk.org/r/4019/ + ........ Merged revisions 424263 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_nat.c: PJSIP: Force transport on contact rewrite + If contact rewriting is enabled but the contact differs in + transport from what is actually being used, messages after the + initial INVITE transaction can be sent to an incorrect + transport/port combination. In the case where this bug occurred + the remote party never received a BYE since it was sent to the + remote party's TCP port over UDP. Review: + https://reviewboard.asterisk.org/r/4032/ ........ Merged + revisions 424244 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-10-01 10:09 +0000 [r424179-424184] Walter Doekes + + * /, channels/chan_sip.c: chan_sip: Simplify some unref code by + removing unlink_peer_from_tables. ASTERISK-22945 #related + Reported by: ibercom Patches: + asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License + #6599) ........ Merged revisions 424181 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 424182 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424183 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Remove excess ref of realtime + peer before sip_poke_peer. The peer is referenced at the end of + sip_poke_peer, it should not get an extra ref before the call to + sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close + Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: + asterisk11.patch uploaded by ibercom (License #6599) Review: + https://reviewboard.asterisk.org/r/4031/ ........ Merged + revisions 424176 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 424177 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424178 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-30 11:40 +0000 [r424153-424156] Joshua Colp + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't place an + extra whitespace before 'rport' and don't put IPv6 addresses in + brackets. #SIPit31 ........ Merged revisions 424155 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the base + and mapped address for candidates is present in SDP. This change + fixes an issue where ICE candidates put into the SDP did not + contain the 'raddr' and 'rport' information for server reflexive + and relay candidates. #SIPit31 ........ Merged revisions 424151 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + Merged revisions 424152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-29 21:59 +0000 [r424129] George Joseph + + * /, res/res_pjsip/pjsip_cli.c: pjsip_cli: Suppress header print on + error or no objects If there's an error on the pjsip command line + or there are no objects, don't print the column headers. + ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George + Joseph Tested-by: Brad Latus Review: + https://reviewboard.asterisk.org/r/4025/ ........ Merged + revisions 424128 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-29 21:26 +0000 [r424126] Walter Doekes + + * /, contrib/scripts/autosupport: autosupport: Fix bashism. '==' is + bashism (bashspecific, fails when dash is /bin/sh). Anyway, a + 'case' works better there. Originally committed in r375059 and + r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported + by: Tzafrir Cohen ........ Merged revisions 424117 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 424125 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-29 21:17 +0000 [r424097-424105] Richard Mudgett + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c, + /, res/res_pjsip_authenticator_digest.c: Simplify UUID generation + in several places. Replace code using ast_uuid_generate() with + simpler and faster code using ast_uuid_generate_str(). The new + code avoids a malloc(), free(), and copy. ........ Merged + revisions 424103 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/threadpool.c: threadpool.c: Minor cleanup fixes. * Fix + threadpool_alloc() prototype. * Add missing off-nominal NULL + check of pool in threadpool_alloc(). * searializer_create() does + not need to create the object with a lock as the lock is not + used. ........ Merged revisions 424096 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-27 12:43 +0000 [r424057] Joshua Colp + + * channels/chan_pjsip.c, res/res_pjsip_session.c, /: + res_pjsip_session: Add additional checks for delaying session + refreshes. There are certain situations which no checks existed + for which need to prevent session refreshes. This includes + sending a session refresh with SDP before SDP negotiation has + completed and sending a session refresh before the dialog itself + has been established. Checks for these have been added. + Additionally COLP related UPDATEs were including SDP when it is + not needed. Review: https://reviewboard.asterisk.org/r/4008/ + ........ Merged revisions 424056 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-26 15:21 +0000 [r423992] Richard Mudgett + + * /, res/res_fax.c: res_fax: Fix out of bounds error in + update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy + Laine Patches: res_fax_bounds.patch (license #6561) patch + uploaded by Jeremy Laine Modified patch to not use magic numbers. + ........ Merged revisions 423979 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423983 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423987 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-26 08:25 +0000 [r423918] Walter Doekes + + * /, doc/asterisk.8: docs: Escape unescaped minus sign in + asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy + Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy + Lainé (License #6561) ........ Merged revisions 423915 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423916 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423917 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-25 21:01 +0000 [r423895] Richard Mudgett + + * res/res_pjsip.c, /: res_pjsip.c: Add missing off nominal cleanup + in ast_sip_push_task_synchronous(). * Made memset the std struct + in ast_sip_push_task_synchronous() because if DEBUG_THREADS is + enabled then uninitialized lock tracking data is used. ........ + Merged revisions 423894 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-24 18:32 +0000 [r423867] Richard Mudgett + + * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c: + pjsip_options.c: Fix race condition stopping periodic out of + dialog OPTIONS request. The crash on the issues is a result of an + invalid transport configuration change when asterisk is + restarted. The attempt to send the qualify request fails and we + cleaned up. However, the callback is also called which results in + a double unref of the objects involved. * Put a wrapper around + pjsip_endpt_send_request() to detect when the passed in callback + is called because of an error so callers can know to not cleanup. + * Made send_request_cb() able to handle repeated challenges (Up + to 10). * Fix periodic endpoint qualify OPTIONS sched deletion + race by avoiding it. The sched entry will no longer self stop and + must be externally stopped. * Added REF_DEBUG description tags to + struct sched_data in pjsip_options.c. * Fix some off-nominal ref + leaks in schedule_qualify(), qualify_and_schedule(). * Reordered + pjsip_options.c module start/stop code to cleanup better on + error. ASTERISK-24295 #close Reported by: Rogger Padilla Review: + https://reviewboard.asterisk.org/r/3954/ ........ Merged + revisions 423866 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-24 08:53 +0000 [r423803] Walter Doekes + + * /, channels/chan_sip.c: chan_sip: Unref outbound proxy structure + on dialog/pvt destruction. Make sure outbound proxy refs are + always unreffed on dialog destruction. Review: + https://reviewboard.asterisk.org/r/4016/ ........ Merged + revisions 423800 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423801 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423802 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-23 14:29 +0000 [r423783] Mark Michelson + + * tests/test_cel.c, tests/test_cdr.c: Make CDR and CEL unit tests + less FRACKy. Prior to this commit, CDR and CEL tests were + expected to trigger FRACKs (i.e. assertions) due to the fact that + the channels they create have no formats on them. Some code was + independently added recently that attempts to prevent FRACKs from + occurring by failing early when attempting to set up translation + paths if one or both channels support no formats. Unfortunately, + this attempt to be helpful made the CDR and CEL tests go from + simply FRACKing to outright failing and in some cases, failing so + badly as to crash Asterisk. This commit seeks to correct past + mistakes by adding the ulaw format to channels created by the CDR + and CEL unit tests. This makes setting up translation paths + succeed, eliminates previously-seen FRACKs, and ultimately causes + the unit tests to succeed again. Review: + https://reviewboard.asterisk.org/r/4014 + +2014-09-22 19:48 +0000 [r423660-423723] Walter Doekes + + * /, channels/chan_sip.c: chan_sip: On INVITE retransmission, don't + add an extra 503 response. INVITE arrives to asterisk, asterisk + responds Busy(). If the INVITE is retransmitted, asterisk would + generate a 503 in addition to the 486. Thanks Torrey Searle for + providing a working regression test. ASTERISK-24335 #close + Review: https://reviewboard.asterisk.org/r/4003/ Patches: + retrans_486_invite.patch uploaded by Torrey Searle (License + #5334) ........ Merged revisions 423720 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423721 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423722 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/editline/readline.c: cli.c: Fix tab completion "module + load" when MALLOC_DEBUG is enabled. r421600 conflicted with + r155763. ASTERISK-24348 #close ........ Merged revisions 423657 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 423658 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423659 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-21 01:15 +0000 [r423618-423641] Matthew Jordan + + * main/channel.c: main/channel: Unlock channel in off-nominal path + In r423414 (13) / r423415 (trunk), an API call that determines if + a format capability structure is empty was added. This returns + true if the format capability structure is completely empty or + "none". A check for this was added in channel.c's set_format + call. Unfortunately, when this check was true, it returned from + the function while still holding the channel lock. This caused + the CDR unit tests - which have a tendency to create channels + with no formats - to deadlock. Whoops. This patch unlocks the + channel on the off-nominal path. + + * rest-api/api-docs/events.json, /: rest-api/api-docs/events.json: + Remove non-compliant 'extends' attribute Prior to the release of + Swagger 1.2, the attribute 'extends' was being promoted as a + possible way to show that a particular object extends an existing + object. Instead, the Swagger specification went with the + 'subTypes' attribute in the base object. This patch removes the + unsupported attribute; the object that the offending objects + proposed to extend already lists them in its 'subTypes' + attribute. ASTERISK-24300 #close Reported by: Bradley Watkins + ........ Merged revisions 423620 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json: rest-api/api-docs: Correct + basePath in resources to match top resources file The + resources.json file that defines the resource JSON files used + with ARI references a basePath of 'http://localhost:8088/ari'. + This does not match what is defined in the resource files + themselves, 'http://localhost:8088/stasis'. The correct base path + is the one that includes 'ari' in the URL; this patch updates the + various resource JSON files to have the correct basePath. + ASTERISK-24339 #close Reported by: Bradley Watkins ........ + Merged revisions 423617 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-19 19:51 +0000 [r423580] Joshua Colp + + * /, res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on + unload/load and don't say the module doesn't exist on reload. + When unloading the module did not unregister the CLI commands + causing a crash upon load when they were registered again. When + reloading the module the return value from the config options + framework was not checked to determine if an error occurred or + not. This caused a message to be output saying the module did not + exist when reloading if no changes were present. AST-1433 #close + AST-1434 #close ........ Merged revisions 423579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-19 17:08 +0000 [r423561] Richard Mudgett + + * channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c: + res_pjsip_sdp_rtp.c: Fix native formats containing formats that + were not negotiated. Outgoing PJSIP calls can result in + non-negotiated formats listed in the channel's native formats if + video formats are listed in the endpoint's configuration. The + resulting call could then use a non-negotiated format resulting + in one way audio. * Simplified the update of session->req_caps in + set_caps(). Why do something in five steps when only one is + needed? AFS-162 #close Review: + https://reviewboard.asterisk.org/r/4000/ + +2014-09-19 15:18 +0000 [r423524-423530] Jonathan Rose + + * /, main/stasis_channels.c: Stasis_channels: Resolve unfinished + Dials when doing masquerades Masquerades into channels that are + in the dialing state don't end their dial and this goes against + the model for things like CDRs and generating Dial end manager + actions and such. ASTERISK-24237 #close Reported by: Richard + Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........ + Merged revisions 423525 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2 + jitterbuffer settings Caused by format changes in Asterisk 13 + ASTERISK-24265 #close Reported by: Dafi Ni Review: + https://reviewboard.asterisk.org/r/3999/ + +2014-09-19 12:45 +0000 [r423504] Kinsey Moore + + * include/asterisk/framehook.h, /, main/framehook.c, + res/res_pjsip_t38.c: PJSIP: Prevent T38 framehook being put on + wrong channel This change gives framehooks a reverse-direction + masquerade callback in addition to chan_fixup_cb similar to the + callback added to datastores to handle the same situation. The + new callback provides the same parameters as the fixup callback, + but is called on the new channel's framehooks before moving + framehooks from the old channel to the new channel. This gives + the framehooks an oppurtunity to decide whether they should + remain on the new channel or be removed. This new callback is + used to prevent the PJSIP T.38 framehook from remaining on a + masqueraded channel if the new channel is not also a PJSIP + channel. This was causing a crash when a local channel was + masqueraded into a PJSIP channel and the framehook was executed + on the local channel since the channel's tech private data was + not structured as expected. Review: + https://reviewboard.asterisk.org/r/4001/ ........ Merged + revisions 423503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 19:30 +0000 [r423482] Sean Bright + + * res/res_pjsip/config_auth.c, /: res_pjsip: Don't require a + password when doing userpass authentication. An empty password is + valid for username/password authentication so we should allow + password to be empty/not supplied. Review: + https://reviewboard.asterisk.org/r/3988 ........ Merged revisions + 423481 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 19:22 +0000 [r423478] George Joseph + + * tests/test_strings.c, /, main/utils.c, + include/asterisk/strings.h: utils: Create ast_strsep function + that ignores separators inside quotes This function acts like + strsep with three exceptions... * The separator is a single + character instead of a string. * Separators inside quotes are + treated literally instead of like separators. * You can elect to + have leading and trailing whitespace and quotes stripped from the + result and have '\' sequences unescaped. Like strsep, ast_strsep + maintains no internal state and you can call it recursively using + different separators on the same storage. Also like strsep, for + consistent results, consecutive separators are not collapsed so + you may get an empty string as a valid result. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ + Merged revisions 423476 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 18:31 +0000 [r423462] Mark Michelson + + * res/res_pjsip_pubsub.c: Add subscription state test events. These + are needed for a set of batched notification RLS tests that are + about to be committed to the testsuite. Review: + https://reviewboard.asterisk.org/r/3967 + +2014-09-18 17:11 +0000 [r423425] Jonathan Rose + + * res/res_pjsip_endpoint_identifier_ip.c, /: + res_pjsip_endpoint_identifier_ip: Fix parsing of match value with + CIDR Also fixes comma separates match lists ASTERISK-24290 #close + Reported by: Ray Crumrine Review: + https://reviewboard.asterisk.org/r/3995/ ........ Merged + revisions 423417 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 17:09 +0000 [r423418-423423] Richard Mudgett + + * bridges/bridge_softmix.c: bridge_softmix.c: Made use + ao2_replace() instead of the inline equivalent. * Clarified some + read/write format comments. * Fixed a doxygen tag typo. + + * main/astobj2.c, contrib/scripts/refcounter.py, /: + astobj2.c/refcounter.py: Fix to deal with invalid object refs. * + Make astob2 REF_DEBUG output an invalid object line when an + invalid ao2 object ref/unref is attempted. This is similar to the + constructor/destructor lines. * Fixed refcounter.py to handle + skewed objects that have constructor/destructor states. * Made + refcounter.py highlight the invalid ao2 object refs by putting + them in their own section of the processed output file. * Made + refcounter.py highlight unreffing an object by more than one that + results in a negative ref count and the object being destroyed. + The abnormally destroyed object is reported in the invalid and + finalized object sections of the output. Review: + https://reviewboard.asterisk.org/r/3971/ ........ Merged + revisions 423349 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423400 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423416 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 16:37 +0000 [r423348-423414] Mark Michelson + + * include/asterisk/format_cap.h, main/channel.c, main/format_cap.c, + main/translate.c: Add API call to determine if format capability + structure is "empty". Empty here means that there are no formats + in the format_cap structure or the only format in it is the + "none" format. I've added calls to check the emptiness of a + format_cap in a few places in order to short-circuit operations + that would otherwise be pointless as well as to prevent some + assertions from being triggered in cases where channels with no + formats are used. + + * /, res/res_fax_spandsp.c: res_fax_spandsp: Properly handle + cleanup before starting FAXes. If faxing fails at a very early + stage, then it is possible for us to pass a NULL t30 state + pointer to spandsp, which spandsp is none too pleased with. This + patch ensures that we pass the correct pointer to spandsp in the + situation where we have not yet set our local t30 state pointer. + ASTERISK-24301 #close Reported by Matt Jordan Patches: + ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License + #5049) ........ Merged revisions 423360 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423365 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c, + res/res_pjsip_dialog_info_body_generator.c, + res/res_pjsip_xpidf_body_generator.c, + res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c, + res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pidf_body_generator.c: res_pjsip_pubsub: Add some + type safety when generating NOTIFY bodies. res_pjsip_pubsub has + two separate checks that it makes when a SUBSCRIBE arrives. * It + checks that there is a subscription handler for the Event * It + checks that there are body generators for the types in the Accept + header The problem is, there's nothing that ensures that these + two things will actually mesh with each other. For instance, + Asterisk will accept a subscription to MWI that accepts pidf+xml + bodies. That doesn't make sense. With this commit, we add some + type information to the mix. Subscription handlers state they + generate data of type X, and body generators state that they + consume data of type X. This way, Asterisk doesn't end up in some + hilariously mismatched situation like the one in the previous + paragraph. ASTERISK-24136 #close Reported by Mark Michelson + Review: https://reviewboard.asterisk.org/r/3877 Review: + https://reviewboard.asterisk.org/r/3878 ........ Merged revisions + 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 15:13 +0000 [r423284] George Joseph + + * /, res/res_pjsip/location.c, + res/res_pjsip_endpoint_identifier_ip.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_auth.c: + res_pjsip: ami: Fix error in AMI output when an endpoint has no + transport When no transport is associated to an endpoint, the AMI + output for PJSIPShowEndpoint indicates an error instead of + silently ignoring the missing transport. This patch causes the + error to appear only if a transport was specified on the endpoint + and the transport doesn't exist. It also fixes an issue with + counting the objects that were actually found. ASTERISK-24161 + #close ASTERISK-24331 #close Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3998/ ........ Merged + revisions 423282 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-18 15:00 +0000 [r423281] David M. Lee + + * makeopts.in, Makefile: Only install dahdi_span_config_hook if + DAHDI is enabled This patch changes the install to only install + the hook script if DAHDI is enabled. It also adds the script to + the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so + that it's not between the _MAKEOPTS variables and their comment. + This allows installs which specify a --prefix to work normally, + as long as they don't enable DAHDI. Review: + https://reviewboard.asterisk.org/r/3972/ + +2014-09-18 14:45 +0000 [r423279] George Joseph + + * main/manager.c, /, include/asterisk/config.h, main/config.c: + config: bug: Fix SEGV in ast_category_insert when matching + category isn't found If you call ast_category_insert with a match + category that doesn't exist, the list traverse runs out of 'next' + categories and you get a SEGV. This patch adds check for the + end-of-list condition and changes the signature to return an int + for success/failure indication instead of a void. The only + consumer of this function is manager and it was also changed to + use the return value. Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3993/ ........ Merged + revisions 423276 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423277 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423278 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-17 18:05 +0000 [r423209-423255] Joshua Colp + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the + thread terminating pj stuff is registered. ........ Merged + revisions 423253 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423254 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix 100% CPU usage + due to timer heap thread spinning. Side note: I need a vacation. + ........ Merged revisions 423210 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423211 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix building when + pjproject is not used. ........ Merged revisions 423207 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423208 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-16 16:32 +0000 [r423192] Scott Griepentrog + + * apps/app_voicemail.c, include/asterisk/file.h, main/file.c: + Voicemail: get correct duration when copying file to vm Changes + made during format improvements resulted in the recording to + voicemail option 'm' of the MixMonitor app writing a zero length + duration in the msgXXXX.txt file. This change introduces a new + function ast_ratestream(), which provides the sample rate of the + format associated with the stream, and updates the app_voicemail + function for ast_app_copy_recording_to_vm to calculate the right + duration. Review: https://reviewboard.asterisk.org/r/3996/ + ASTERISK-24328 #close + +2014-09-16 12:12 +0000 [r423152-423173] Joshua Colp + + * res/res_pjsip_session.c, /: res_pjsip_session: Fix usage of wrong + memory pool when creating local SDP. ........ Merged revisions + 423172 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, /: + res_rtp_asterisk: Fix a myriad of TURN client issues. 1. The + number of file descriptors an ioqueue instance can handle is + fixed, so we now spawn the required number to handle the load. 2. + Our transport identifiers were exceeding the range supported by + pjnath. 3. The TURN client did not set up client binding causing + needless bandwidth usage. 4. The code no longer updates address + information on each packet. 5. STUN traffic was getting looped + back to Asterisk instead of going through the TURN server. 6. + Synchronization now ensures things are completely setup or + destroyed. 7. Logging now reflects the target the TURN server is + sending to/receiving from on our behalf. ASTERISK-23577 #close + Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: + Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ + ........ Merged revisions 423150 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423151 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-15 10:49 +0000 [r423069-423129] Walter Doekes + + * /, + contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py + (added): contrib: Fix verifyi typo in alembic DB script + ps_transport table. Reported by: Zogot (on IRC) Patches: tmp.diff + uploaded by Zogot, cleaned up by me. ........ Merged revisions + 423128 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/samples/sip.conf.sample, /: chan_sip: Clarify that + sipdebug=yes cannot be undone by the CLI. Document it in + sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod + Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged + revisions 423066 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 423067 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 423068 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-12 16:09 +0000 [r422985] Jonathan Rose + + * main/config.c, /: Realtime: Fix a bug that caused realtime + destroy command to crash Also has could affect with anything that + goes through ast_destroy_realtime. If a CLI user used the command + 'realtime destroy ' with only a single column/value pair, + Asterisk would crash when trying to create a variable list from a + NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson + Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged + revisions 422984 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-11 22:16 +0000 [r422965] Mark Michelson + + * /, main/app.c: Remove undocumented default behavior of + ast_play_and_record_full acceptdtmf. ast_play_and_record_full() + has a parameter called "acceptdtmf" that is a string of + acceptable DTMF digits that may be pressed by a caller to end and + accept the recording. ARI uses this function in order to perform + recording, and it provides options for what is passed as + acceptdtmf to ast_play_and_record_full(). By default, ARI passes + an empty string, with the intention that no DTMF can be used to + end the recording. The problem is that ast_play_and_record_full() + attempts to be "helpful" by setting "#" as the acceptdtmf if an + empty string or NULL pointer has been passed in. With ARI, this + results in unexpected behavior occurring if you have attempted to + intercept "#" yourself in order to perform some other + manipulation of the live recording. This change removes the + "helpful" behavior by no longer accepting "#" as a default + acceptdtmf if none is specified by the caller of + ast_play_and_record_full(). This makes the ARI scenario work as + expected. The other callers of ast_play_and_record_full() are + app_voicemail and app_minivm, and in both cases, they pass an + explicit "#" to ast_play_and_record_full() as acceptdtmf, so they + are unaffected by this change. ........ Merged revisions 422964 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-10 16:04 +0000 [r422905] George Joseph + + * /, main/config.c: config: bug: fix truncation of included config + files on permissions error ast_config_text_file_save() currently + truncates include files as they are processed. If a subsequent + include file or the main config file has a permissions error that + prevents writing, earlier include files are left truncated + resulting in a frantic search for backups. This patch causes + ast_config_text_file_save to check for write access on all files + before it truncates any of them. Will be applied 1.8 > trunk. + Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3986/ ........ Merged + revisions 422900 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422903 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422904 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-10 15:59 +0000 [r422901] Sean Bright + + * res/res_pjsip/config_auth.c, /: pjsip/config_auth.c: Add missing + whitespace to log messages. The errors generated when validating + 'auth' settings are missing a space which makes the messages a + little confusing. ........ Merged revisions 422899 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-09 20:01 +0000 [r422883] Rusty Newton + + * /, sounds/sounds.xml, sounds/Makefile: Sounds/BuildSystem: + Modifications to include new releases and Japanese language. + Modifying Makefile and sounds.xml to include new core 1.4.26 and + extra 1.4.15 sound prompt releases, plus the new Japanese core + sound prompts contributed by QLOOG. ASTERISK-23324 Reported by: + Kevin McCoy Tested by: Rusty Newton ........ Merged revisions + 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 422790 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422791 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-08 18:03 +0000 [r422851-422855] Mark Michelson + + * configs/samples/pjsip.conf.sample: Add note about configuring + list_items on a single line. + + * configs/samples/pjsip.conf.sample: Add sample configuration for + resource lists. On review /r/3977, it was recommended to note in + the sample configuration about the size limitation for resource + lists. However, since there was no section in the sample + configuration at all for resource list subscriptions, I decided + to make a separate commit where I have added the necessary sample + configuration as well as the size limitation warning. + + * res/res_pjsip_pubsub.c: Pre-allocate transmission data buffer for + RLS NOTIFY requests. PJSIP, unless a constant is modified at + compilation time, limits SIP requests to 4000 bytes. Full-state + RLS notifications can easily exceed this limit with moderately + small lists. This changeset allows for Asterisk to work around + this size limit by performing its own allocation of the + transmission data buffer. This way, Asterisk can allocate a + buffer that exceeds the built-in maximum. We still impose our own + limit of 64000 bytes, mainly because making allocations larger + than that is a bit absurd. ASTERISK-24181 #close Reported by Mark + Michelson Review: https://reviewboard.asterisk.org/r/3977 + +2014-09-08 15:41 +0000 [r422836] Jonathan Rose + + * res/res_pjsip_pubsub.c: res_pjsip_pubsub: Check supported headers + for eventlist when subscribing to resource list + https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan + According to the off-nominal plan, if evenlist support is not + specified in a SUBSCRIBE's supported header(s), that subscription + should be rejected with an error. ASTERISK-23871 Reported by: + Mark Michelson Review: + https://reviewboard.asterisk.org/r/3960/diff/#index_header + +2014-09-06 22:49 +0000 [r422767-422770] Matthew Jordan + + * /, main/cdr.c: main/cdr: Copy over location information during a + fork When a CDR is forked, a new CDR is created and appended to + the CDR chain for the Party A. The forked CDR starts life off as + a clone of the last non-finalized for the particular Party A. In + the past, merely copying over the snapshots for Party A/Party B + would be sufficient. However, as the CDRs now contain cached + information from Party A - specifically application/data, + context, and extension - we need to copy that over during a fork + as well. Huzzah for unit tests catching this when the + context/extension were derived from a cached value on the CDR + instead of on Party A. ........ Merged revisions 422769 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/rtp_engine.c, /: main/rtp_engine: Format NTP timestamps as + unsigned ints On some systems, a timeval's tv_sec/tv_usec will be + unsigned lont ints, as opposed to long ints. When the RTP engine + formats these as strings, it was previously formatting them as + signed integers, which can result in some odd negative timestamp + values (particularly on 32-bit systems). This patch formats the + values as unsigned long integers. ........ Merged revisions + 422766 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-06 19:12 +0000 [r422747] Joshua Colp + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix retrieval of + "ice-pwd" attribute if in session and not media stream. ........ + Merged revisions 422746 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-05 22:03 +0000 [r422716-422719] Matthew Jordan + + * main/cdr.c, /, apps/app_macro.c, include/asterisk/channel.h, + apps/app_stack.c: main/cdrs: Preserve context/extension when + executing a Macro or GoSub The context/extension in a CDR is + generally considered the destination of a call. When looking at a + 2-party call CDR, users will typically be presented with the + following: context exten channel dest_channel app data default + 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial + actually takes place in a Macro, the current behaviour in 12 will + result in the following CDR: context exten channel dest_channel + app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The + same is true of a GoSub: context exten channel dest_channel app + data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This + generally makes the context/exten fields less than useful. It + isn't hard to preserve these values in the CDR state machine; + however, we need to have something that informs us when a channel + is executing a subroutine. Prior to this patch, there isn't + anything that does this. This patch solves this problem by adding + a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on + a channel when it executes a Macro or a GoSub. The CDR engine + looks for this value when updating a Party A snapshot; if the + flag is present, we don't override the context/exten on the main + CDR object. In a funny quirk, executing a hangup handler must + *not* abide by this logic, as the endbeforehexten logic assumes + that the user wants to see data that occurs in hangup logic, + which includes those subroutines. Since those execute outside of + a typical Dial operation (and will typically have their own + dedicated CDR anyway), this is unlikely to cause any heartburn. + Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254 + #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis + ........ Merged revisions 422718 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: main/cdr: Fix crash/memory consumption in CDRs in + multi-party bridge scenarios This patch fixes an issue where CDRs + would get stuck generating an infinite number of CDRs, eventually + crashing Asterisk (and consuming a lot of memory along the way). + When a channel enters into a multi-party bridge, the CDR engine + creates mappings of each participant to each other participant, + picking the 'A' party as it goes. So, if we have four channels in + a multi-party bridge (Alice, Bob, Charlie, Denise), we would have + something like: Alice => Bob Alice => Charlie Alice => Denise Bob + => Charlie Bob => Denise Charlie => Denise This works fine when + participants enter the bridge a single time. When a participant + leaves a bridge, the CDRs for that channel are transitioned to a + finalized state. The bug occurs if Bob rejoins. When the CDR + engine creates mappings between the channels, it walks through + all the participants currently in the bridge, and realizes that + no one in the bridge can create a CDR with the channel (Bob). As + such it creates a new CDR for the candidate and appends it to + that candidate's chain. Unfortunately, on this particular code + path, it doesn't stop traversing the candidate's chain. Since we + just added ourselves to the chain, this causes the loop to keep + going, constantly adding new CDRs. This patch makes it so the + engine bails when it creates a CDR match in this case. Review: + https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close + Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat + ASTERISK-24208 Reported by: Frankie Chin ........ Merged + revisions 422715 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-05 20:35 +0000 [r422700] Richard Mudgett + + * funcs/func_channel.c: func_channel.c: Add missing locking to some + CHANNEL() requests. * The CHANNEL() audionativeformat, + videonativeformat, audioreadformat, and audiowriteformat now need + locking since the media format rework when accessing the + channel's format pointers. * Increased the buffer size for + CHANNEL() audionativeformat and videonativeformat output strings + since the allow=all can be a lengthy list. * Tweaked the + CHANNEL() XML documentation for secure_bridge_signaling, + secure_bridge_media, and state. * Ensured the output buffer is + initialized for secure_bridge_signaling and secure_bridge_media. + * Made use the locked_copy_string() macro instead of inlining it + for trace and checkhangup. + +2014-09-05 20:11 +0000 [r422665-422684] Jonathan Rose + + * main/dial.c, include/asterisk/dial.h: Dial API: Add a dial option + to indicate the dialed channel will replace dialer Adds an option + to the dial API that marks an outgoing dial as replacing the + dialing channel for the purpose of propagating accountcode. When + it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of + AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on + the involved channels with ast_channel_req_accountcodes. Review: + https://reviewboard.asterisk.org/r/3968/ + + * main/cli.c, /: Call IDs: Fix appearance of call ID in core show + channels when NULL NULL call IDs were meant to appear as '(none)' + but instead were showing the contents of an uninitialized + character buffer. ASTERISK-24223 Review: + https://reviewboard.asterisk.org/r/3979/ ........ Merged + revisions 422664 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-05 17:36 +0000 [r422661] Richard Mudgett + + * main/devicestate.c, channels/chan_iax2.c: devicestate.c: Minor + tweaks * In ast_state_chan2dev() use ARRAY_LEN() instead of a + sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c. + +2014-09-05 13:28 +0000 [r422646] Kinsey Moore + + * menuselect/menuselect.c: Menuselect: Fix incorrect enabling on + failed deps This corrects a situation where menuselect can + incorrectly enable a module by default that has defaultenabled + set to "no" and has failed/non-selected dependencies. The bug is + due to an inverted test when checking for whether the given + module should be set to enabled by default on load. Review: + https://reviewboard.asterisk.org/r/3975/ Reported by: John + Bigelow + +2014-09-04 21:23 +0000 [r422631] Jonathan Rose + + * main/manager.c, /: Manager: Require read permission for SYSTEM in + order to send FullyBooted Review: + https://reviewboard.asterisk.org/r/3969/ ........ Merged + revisions 422584 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422625 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422626 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-03 14:05 +0000 [r422558] Joshua Colp + + * res/res_pjsip_transport_websocket.c, /: + res_pjsip_transport_websocket: Fix crash when the Contact header + is not a URI. The code for changing the Contact header wrongly + assumed that the Contact would always contain a URI. This is + incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged + revisions 422557 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-02 20:29 +0000 [r422542] Mark Michelson + + * /, channels/chan_pjsip.c, res/res_pjsip_diversion.c, + res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h: + Resolve race condition where channels enter dialplan application + before media has been negotiated. Testsuite tests will + occasionally fail because on reception of a 200 OK SIP response, + an AST_CONTROL_ANSWER frame is queued prior to when media has + finished being negotiated. This is because session supplements + are called into before PJSIP's inv_session code has told us that + media has been updated. Sometimes the queued answer frame is + handled by the PBX thread before the ensuing media negotiations + occur, causing a test failure. As it turns out, there is another + place that session supplements could be called into, which is + after media has finished getting negotiated. What this commit + introduces is a means for session supplements to indicate when + they wish to be called into when handling an incoming SIP + response. By default, all session supplements will be run at the + same point that they were prior to this commit. However, session + supplements may indicate that they wish to be handled earlier + than normal on redirects, or they may indicate they wish to be + handled after media has been negotiated. In this changeset, two + session supplements have been updated to indicate a preference + for when they should be run: res_pjsip_diversion executes before + handling redirection in order to get information from the + Diversion header, and chan_pjsip now handles responses to INVITEs + after media negotiation to fix the race condition mentioned + previously. ASTERISK-24212 #close Reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/3930 ........ Merged revisions + 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-09-01 14:16 +0000 [r422504-422507] Matthew Jordan + + * main/cli.c, /: main/cli: Do not attempt to show CDR data for + internal channels Internal channels don't have CDRs. Querying the + CDR engine for their variables will make it cranky. ........ + Merged revisions 422506 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis.c, /, res/stasis/stasis_bridge.c: res_stasis: + Don't play MoH to channels by default when added to holding + bridges When ARI manipulates a bridge, it generally doesn't care + what the mixing technology is. Operations on a bridge initiated + through ARI should perform their action in generally the same + way, regardless of the bridge's mixing technology. While the + mixing technology may determine how media flows to channels, the + actual operations on a bridge themselves should be the same. + Currently, this isn't the case with holding bridges. When a + channel joins without a role, MoH is started on that channel + automatically. Subsequent bridge operations that would stop MoH + would fail (as there is no Announcer channel playing MoH to the + bridge). Starting MoH on the bridge will also create two MoH + streams: one from the MoH being played on the participant + channel, and one from the announcer channel. From the perspective + of ARI users, this is counter-intuitive - I would not expect MoH + to be started for me. The mixing technology determines how media + is shared between participants, not the application experience. + This patch does the following: * The Stasis bridge class now + inspects channels as they are going into a bridge. If the bridge + has a holding capability, and the channel has no roles, we give + it a participant role and mark the default behaviour to have no + entertainment. This allows addChannel operations to continue to + set a participant role with an entertainment option if it felt + like it (or could do it). * The music on hold channel is now + Stasis approved (tm) Review: + https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close + Reported by: Samuel Galarneau Tested by: Samuel Galarneau + ........ Merged revisions 422503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-30 17:32 +0000 [r422442-422445] George Joseph + + * apps/app_confbridge.c, /: confbridge: Add Duration to + ConfbridgeList event The ConfbridgeList event doesn't include how + long the user has been a member of the conference. This patch + adds Duration (seconds) which is based on user->chan->answertime. + Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3955/ ........ Merged + revisions 422444 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Make WaitEvent action respect + eventfilters A WaitEvent issued via an http session isn't + respecting eventfilters defined for the user. I just added a + match_filter to the predicate that controls astman_append. Tested + by: George Joseph Review: + https://reviewboard.asterisk.org/r/3958/ ........ Merged + revisions 422439 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422440 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422441 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-29 19:40 +0000 [r422374-422379] Matthew Jordan + + * doc/smsq.8 (added), /: doc: Add a manpage for the smsq utility + This patch adds a manpage for the smsq utility. Note that this is + one of the patches the Debian distro applies for the Asterisk + project, as per ASTERISK-24191. Review: + https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close + Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy + Laine (License 6561) ........ Merged revisions 422376 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422377 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422378 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * doc/aelparse.8 (added), /: doc: Add a manpage for the aelparse + utility This patch adds a manpage for the aelparse utility. Note + that this is one of the patches the Debian distro applies for the + Asterisk project, as per ASTERISK-24191. Review: + https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close + Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy + Laine (License 6561) ........ Merged revisions 422371 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422372 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422373 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-29 19:05 +0000 [r422359] Scott Griepentrog + + * channels/chan_sip.c: The assertion that peer was not found on + final event message was being triggered on configuration reload. + This patch changes that case to just return instead. Review: + https://reviewboard.asterisk.org/r/3953/ Commited in trunk + revision 422358 + +2014-08-28 21:54 +0000 [r422296] Matthew Jordan + + * LICENSE, /: LICENSE: Clarify language in Asterisk's LICENSE to + allow for linking to UniMRCP The UniMRCP project distributes + Asterisk modules that integrate Asterisk with UniMRCP, and other + Asterisk users use the UniMRCP library as well. Unfortunately, + the UniMRCP license is Apache 2.0, which per the Free Software + Foundation, is not a compatible license with the GPLv2. "Please + note that this license is not compatible with GPL version 2, + because it has some requirements that are not in that GPL + version. These include certain patent termination and + indemnification provisions. The patent termination provision is a + good thing, which is why we recommend the Apache 2.0 license for + substantial programs over other lax permissive licenses." On the + other hand, UniMRCP is a great project and we'd like to let + people use it with Asterisk. This patch updates the LICENSE text + to allow users to link Asterisk with UniMRCP and distribute the + resulting binaries. ........ Merged revisions 422293 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422294 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422295 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-28 20:30 +0000 [r422276] Michael L. Young + + * /, channels/chan_iax2.c: chan_iax2: Fix Dynamic IAX2 + Registrations After Temporary DNS Failure The reporter on the + issue found some issues when upgrading from version 10 to 11 on + 55 hosts. Two situations that can occur with dynamic + registrations. 1. With dnsmgr disabled, if the host is not + resolvable we are not trying to resolve the host again when it is + time to attempt to register again. This results in never + registering to the host. 2. With dnsmgr enabled, when the host is + temporarily not resolvable the address is set to 0.0.0.0:0 and + then when the host is resolvable the port is not being restored + and stays set to 0. This patch resolves these two issues by: * + Storing the hostname so that it can be used for resolving with + DNS. * Resolve the hostname on the next scheduled attempt to + register. * Storing the port used to reach the host so that when + the hostname is resolvable again, we can set the port again if + the port is still unset after looking up the host. ASTERISK-23767 + #close Reported by: David Herselman Tested by: David Herselman, + Michael L. Young Patches: + asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3856/ ........ Merged + revisions 422274 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422275 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-28 17:25 +0000 [r422256] Richard Mudgett + + * /, UPGRADE.txt: Added ConfBridge AMI event note to UPGRADE.txt. + ........ Merged revisions 422255 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-28 15:49 +0000 [r422239] Mark Michelson + + * res/res_pjsip_pubsub.c: Fix bug that did not allow for multiple + batched RLS notifications to be sent. A misunderstanding of how + the scheduler worked caused further batched notifications beyond + the first not to get scheduled. Now we reset our scheduler ID to + -1 after the batched notification is sent. This way, further + notifications can be scheduled when they arise. + +2014-08-28 00:36 +0000 [r422200-422215] Richard Mudgett + + * res/res_pjsip/pjsip_options.c, /: res/res_pjsip/pjsip_options.c: + Eliminate excessive RAII_VAR usage. * Fix off nominal ref leak in + find_or_create_contact_status(). * Add missing NULL check of + status in update_contact_status() and init_start_time(). ........ + Merged revisions 422214 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/sched.c, include/asterisk/sched.h: sched: Fix typo and + whitespace change. + +2014-08-27 17:29 +0000 [r422177] George Joseph + + * /, apps/confbridge/confbridge_manager.c, apps/app_confbridge.c: + confbridge: Add 'Admin' param to join, leave, mute, unmute and + talking events Currently there's no way to tell if a user is an + admin or not when receiving the join, leave, mute, unmute and + talking events. This patch adds that capability. Tested by: + George Joseph Review: https://reviewboard.asterisk.org/r/3950/ + ........ Merged revisions 422176 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-27 15:31 +0000 [r422154] Kinsey Moore + + * include/asterisk/utils.h, /, channels/chan_sip.c, + tests/test_callerid.c (added), tests/test_utils.c, + main/callerid.c, main/utils.c, res/res_pjsip_caller_id.c: + CallerID: Fix parsing of malformed callerid This allows the + callerid parsing function to handle malformed input strings and + strings containing escaped and unescaped double quotes. This also + adds a unittest to cover many of the cases where the parsing + algorithm previously failed. Review: + https://reviewboard.asterisk.org/r/3923/ Review: + https://reviewboard.asterisk.org/r/3933/ ........ Merged + revisions 422112 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 422113 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 422114 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-26 23:28 +0000 [r422091] George Joseph + + * apps/app_confbridge.c, /: confbridge: Make kick, mute and unmute + handle channel targets consistently. Kick, mute and unmute were a + little inconsistent in their handling of channel targets. This + patch cleans that up by insuring they all handle the 'all' target + consistently and adds the 'participants' target which acts on + non-admins. Documentation for kick was also cleaned up as it + never supported partial channel names. Tested by: George Joseph + Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged + revisions 422090 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-26 22:13 +0000 [r422071] Mark Michelson + + * main/sched.c, /: Fix race condition in the scheduler when + deleting a running entry. When scheduled tasks run, they are + removed from the heap (or hashtab). When a scheduled task is + deleted, if the task can't be found in the heap (or hashtab), an + assertion is triggered. If DO_CRASH is enabled, this assertion + causes a crash. The problem is, sometimes it just so happens that + someone attempts to delete a scheduled task at the time that it + is running, leading to a crash. This change corrects the issue by + tracking which task is currently running. If that task is + attempted to be deleted, then we mark the task, and then wait for + the task to complete. This way, we can be sure to coordinate task + deletion and memory freeing. ASTERISK-24212 Reported by Matt + Jordan Review: https://reviewboard.asterisk.org/r/3927 ........ + Merged revisions 422070 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-25 16:44 +0000 [r421979-422037] Richard Mudgett + + * res/res_musiconhold.c: res_musiconhold.c: Release any format refs + before memset(). * Clear the channel music_state pointer before + destroying the music_state object for safety. + + * res/res_musiconhold.c, /: res_musiconhold: Fix MOH restarting + where it left off from the last hold. Restore code removed by + https://reviewboard.asterisk.org/r/3536/ that introduced a + regression that prevents MOH from restarting were it left off the + last time. ASTERISK-24019 #close Reported by: Jason Richards + Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch + uploaded by rmudgett Review: + https://reviewboard.asterisk.org/r/3928/ ........ Merged + revisions 421976 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421977 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421978 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-24 19:36 +0000 [r421911-421956] Joshua Colp + + * res/res_pjsip_transport_websocket.c, /: + res_pjsip_transport_websocket: Attach the Websocket module on + outgoing INVITEs. In order to alter the Contact header on + in-dialog requests and responses the Websocket module must be + attached on outgoing INVITEs. The Contact header is modified so + that the PJSIP transport layer can find and use the existing + Websocket connection based on the source IP address, port, and + transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov + ........ Merged revisions 421955 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_transport_websocket.c: + res_pjsip_transport_websocket: Fix a progressive memory growth. + The packet structure used to receive messages was using the + transport pool. This meant that for each parsing the pool would + grow accordingly. Since memory can not be reclaimed without + resetting it this would cause the memory pool to grow and grow. + This change uses a specific memory pool for the packet structure + and resets it to a fresh state after the message has been + received and handled. ........ Merged revisions 421939 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_transport_websocket.c: + res_pjsip_transport_websocket: Ensure secure Websocket clients + can be called. This change enforces the transport in the Contact + header for Websocket clients. Previously a client may provide a + transport of 'ws' when it is actually using a transport of 'wss'. + This would cause outgoing calls to fail as the existing + connection could not be found. ........ Merged revisions 421931 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Use the server reflexive ICE + candidate RTCP port as provided. This code originally worked + around an issue within res_rtp_asterisk itself. The wrong socket + was being used for the STUN check for RTCP, causing the port to + be the same as RTP. This was subsequently fixed and the RTCP port + provided for the ICE candidate is correct and does not need to be + incremented. ASTERISK-23997 #close Reported by: Badalian + Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav + (license 5249) ........ Merged revisions 421909 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421910 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-22 16:56 +0000 [r421882] Mark Michelson + + * apps/app_mixmonitor.c: Fix a locking inversion in MixMonitor. We + need to unlock the audiohook before trying to lock the channel, + since the correct locking order is channel then audiohook. + +2014-08-22 16:44 +0000 [r421880] Jonathan Rose + + * res/res_stasis_answer.c, res/res_stasis.c, res/stasis/command.c, + res/res_stasis_playback.c, /, res/stasis/control.c, + res/stasis/stasis_bridge.c, res/stasis/command.h, + include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c: + ARI: Fix a crash caused by hanging during playback to a channel + in a bridge ASTERISK-24147 #close Reported by: Edvin Vidmar + Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged + revisions 421879 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-22 14:08 +0000 [r421860] Matthew Jordan + + * main/message.c, /: main/message: Add a new-line to a DEBUG + message ........ Merged revisions 421859 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 22:07 +0000 [r421802] Richard Mudgett + + * /, res/res_musiconhold.c: res_musiconhold.c: Remove obsolete + REF_DEBUG code. Remove unneeded code that writes to the wrong + file location in an obsolete format. ........ Merged revisions + 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 421800 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421801 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 21:42 +0000 [r421790-421797] Mark Michelson + + * res/res_pjsip_session.c, /: Switch from hostname to an IP address + in the SDP origin line. Using the hostname in the SDP origin line + may not satisfy the requirement of RFC 4566 that we use a FQDN or + IP address. This change has us use the same information from the + SDP connection line if possible. If not possible, we'll use the + configured media address. And if that's not possible, we use the + result of a PJLIB call to get the IP address of ourself. + ASTERISK-23994 #close Reported by Private Name Review: + https://reviewboard.asterisk.org/r/3925 ........ Merged revisions + 421796 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c: Ensure after-bridge behavior is correct + when moving from Stasis to a non-Stasis bridge. Because of the + departable state of channels that enter Stasis bridges, Stasis + has to take responsibility for directing the channel to its + intended after-bridge destination if the channel moves from a + Stasis bridge to a non-Stasis bridge. This change ensures that + when such a move occurs, when the channel leaves the bridging + system, any after bridge gotos are honored. Review: + https://reviewboard.asterisk.org/r/3920 ........ Merged revisions + 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_caller_id.c, /: Let's try checking the name and + number, instead of the name twice. ........ Merged revisions + 421789 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 21:25 +0000 [r421788] Jonathan Rose + + * /, res/res_musiconhold.c: res_musiconhold: Fix reference leaks + caused when reloading with REF_DEBUG set Due to a faulty function + for debugging reference decrementing, it was possible to reduce + the refcount on the wrong object if two moh classes of the same + name were in the moh class container. (closes issue + ASTERISK-22252) Reported by: Walter Doekes Patches: + 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license + 6182) ........ Merged revisions 398937 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421777 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421779 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 21:18 +0000 [r421783] Mark Michelson + + * /, res/res_pjsip_caller_id.c: Improve consistency of party ID + privacy usage. Prior to this change, the Remote-Party-ID header + took the position of "If caller name and number are not + explicitly allowed, then they are private" and + P-Asserted-Identity took the position of "Caller name and number + are only private if marked explicitly so" Now both mechanisms of + conveying party identification use the former approach. ........ + Merged revisions 421778 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-21 17:34 +0000 [r421675-421720] Matthew Jordan + + * /, channels/chan_sip.c: chan_sip: Don't use port derived from + fromdomain if it isn't set If a user does not provide a port in + the fromdomain setting, chan_sip will set the fromdomainport to + STANDARD_SIP_PORT (5060). The fromdomainport value will then get + used unilaterally in certain places. This causes issues with TLS, + where the default port is expected to be 5061. This patch + modifies chan_sip such that fromdomainport is only used if it is + not the standard SIP port; otherwise, the port from the SIP pvt's + recorded self IP address is used. Review: + https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close + Reported by: Elazar Broad patches: fromdomainport_fix.diff + uploaded by Elazar Broad (License 5835) ........ Merged revisions + 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 421718 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421719 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, UPGRADE.txt, main/app.c: ARI: Fix implicit answer when + playback is initiated on unanswered channel When issuing a POST + /channels/{channel_id}/play on a channel that is not yet + answered, ARI is supposed to: * Queue up an AST_CONTROL_PROGRESS + on the channel * Start up the playback of the media Instead, we + sneak an answer on the channel right before starting playing + media. This is due to ARI's usage of control_streamfile. This + function implicitly answers the channel (and doesn't give ARI the + option to stop it). The answering of the channel here is probably + unnecessary: * app_voicemail, by far the biggest consumer of this + function, always answers the channels anyway * control stream + file (in res_agi) and ControlPlayback probably shouldn't be + implicitly answering the channel. Answering should not be tied + directly to playing back media. As it turns out, the answering of + the channel here is pretty old: 356042 twilson if + (ast_channel_state(chan) != AST_STATE_UP) { 3087 anthm res = + ast_answer(chan); 180259 tilghman } (As in, ancient?) Note that + others ran into this problem and commented about it on various + mailing lists. Review: https://reviewboard.asterisk.org/r/3907/ + ASTERISK-24229 #close Reported by: Matt Jordan ........ Merged + revisions 421695 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/stasis/messaging.h, main/dns.c, /, main/format_cache.c: Clean + up files that do not end with newlines Trivial patch to add new + lines to several files missing them. This fixes warnings when + compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close + Reported by: Shaun Ruffell patches: + 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch + uploaded by Shaun Ruffell (License 5417) ........ Merged + revisions 421677 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/uri.h, main/uri.c: uri: Quiet warning about type + qualifiers ignored on function return type This patch fixes gcc + warnings that occur due to the type qualifier 'const' being + ignored on a return type of int. ASTERISK-24246 #close Reported + by: Shaun Ruffell patches: + 0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch + uploaded by Shaun Ruffell (License 5417) + +2014-08-20 22:49 +0000 [r421616-421645] Richard Mudgett + + * main/bridge.c, res/res_pjsip_sdp_rtp.c, main/file.c, + main/bridge_channel.c, channels/chan_pjsip.c, main/channel.c: + chan_pjsip: Update media translation paths when new SDP + negotiated. On a SIP reinvite that changes media strams, the + PJSIP channel driver was flooding the log with "Asked to transmit + frame type %s, while native formats is %s" warnings. * Fixes + PJSIP not setting up translation paths when the formats change on + a reinvite. AFS-63 was effectively reintroduced because of the + media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the + unexpected frame format WARNING message to include more + information. * Added protective locking while altering formats on + a channel. Reworked set_format() to simplify and protect the + formats under manipulation. * Restored some code that got lost in + the media_formats work. (channel.c:set_format() and + res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark + Michelson Review: https://reviewboard.asterisk.org/r/3906/ + + * /, main/cli.c: cli.c: Fix tab completion of "module load" when + MALLOC_DEBUG is enabled. filename_completion_function() returns + memory that was not allocated by the MALLOC_DEBUG allocation + tracker so the memory must be freed by ast_std_free(). ........ + Merged revisions 421600 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421602 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421608 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-20 20:40 +0000 [r421566-421585] Mark Michelson + + * res/res_pjsip_pubsub.c: Set the role for inbound subscriptions + correctly. This was causing the AMI show_subscriptions test in + the testsuite to fail since all subscriptions were being seen as + subscribers instead of notifiers. + + * /, channels/chan_pjsip.c: Move evaluation of set_var options in + pjsip to the end of channel initialization. This allows for + set_var to override certain defaults such as caller ID and codec + values. This also fixes a test suite regression. The "set_var" + test suite test attempted to use set_var to override caller ID, + but a recent change caused that to no longer work. ........ + Merged revisions 421565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-20 13:04 +0000 [r421538] Kinsey Moore + + * include/asterisk/stasis_bridges.h, tests/test_cel.c, + res/ari/ari_model_validators.c, main/stasis_bridges.c, + res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /, + res/stasis/app.c, main/bridge.c: Stasis: Add information to blind + transfer event When a blind transfer occurs that is forced to + create a local channel pair to satisfy the transfer request, + information about the local channel pair is not published. This + adds a field to describe that channel to the blind transfer + message struct so that this information is conveyed properly to + consumers of the blind transfer message. This also fixes a bug in + which Stasis() was unable to properly identify the channel that + was replacing an existing Stasis-controlled channel due to a + blind transfer. Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3921/ ........ Merged + revisions 421537 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 20:28 +0000 [r421448-421488] Mark Michelson + + * /, res/res_pjsip.c: Alter documentation for callerid_privacy to + use correct values. ........ Merged revisions 421485 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis.c, /: Fix compilation error on certain versions of + GCC. ........ Merged revisions 421447 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 19:42 +0000 [r421445] Kinsey Moore + + * main/manager.c, /: AMI Docs: Fix Status channel parameter + optionality ........ Merged revisions 421442 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421443 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421444 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 16:28 +0000 [r421423] Jonathan Rose + + * res/res_stasis.c, /: ARI: Fix a bug where + /channels/{channelID}/continue doesn't execute PBX If + /channels/{channelID}/continue is called on a channel that was + originated without a PBX (such as the ARI command POST channel + with a stasis application argument), the channel will not start + dialplan execution. This patch will now run the PBX out of the + stasis execution if the channel doesn't currently have an active + PBX upon continuing. ASTERISK-24043 #close Reported by: Krandon + Bruse Review: https://reviewboard.asterisk.org/r/3917/ Patches: + stasis-continue.diff submitted by Krandon Bruse (license 6631) + ........ Merged revisions 421416 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-19 16:11 +0000 [r421403] Richard Mudgett + + * /, res/res_pjsip_caller_id.c, channels/chan_pjsip.c, + res/res_pjsip_session.c: chan_pjsip: Fix attended transfer + connected line name update. A calls B B answers B SIP attended + transfers to C C answers, B and C can see each other's connected + line information B completes the transfer A has number but no + name connected line information about C while C has the full + information about A I examined the incoming and outgoing party id + information handling of chan_pjsip and found several issues: * + Fixed ast_sip_session_create_outgoing() not setting up the + configured endpoint id as the new channel's caller id. This is + why party A got default connected line information. * Made + update_initial_connected_line() use the channel's CALLERID(id) + information. The core, app_dial, or predial routine may have + filled in or changed the endpoint caller id information. * Fixed + chan_pjsip_new() not setting the full party id information + available on the caller id and ANI party id. This includes the + configured callerid_tag string and other party id fields. * Fixed + accessing channel party id information without the channel lock + held. * Fixed using the effective connected line id without doing + a deep copy outside of holding the channel lock. Shallow copy + string pointers can become stale if the channel lock is not held. + * Made queue_connected_line_update() also update the channel's + CALLERID(id) information. Moving the channel to another bridge + would need the information there for the new bridge peer. * Fixed + off nominal memory leak in update_incoming_connected_line(). * + Added pjsip.conf callerid_tag string to party id information from + enabled trust_inbound endpoint in caller_id_incoming_request(). + AFS-98 #close Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3913/ ........ Merged + revisions 421400 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-18 21:10 +0000 [r421376] Damien Wedhorn + + * channels/chan_skinny.c: Skinny: Fixup compile warning for non + dev-mode. + +2014-08-18 20:19 +0000 [r421337] George Joseph + + * funcs/func_config.c, /: func_config: Change 'Not Found' message + from ERROR to DEBUG When you call the CONFIG dialplan function + with the name of a variable that doesn't exist in the target + context you get an ERROR. This does nothing but clutter up the + logs with messages that may be perfectly acceptable. Just because + a variable wasn't in the context doesn't mean it's an error. + Maybei t's optional or just needs to be defaulted or ignored. + This patch changes the log level from ERROR to DEBUG. If a + dialplan developer wants to debug their dialplan they still canby + setting the console debug level as needed. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........ + Merged revisions 421327 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421328 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421329 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-18 01:13 +0000 [r421230-421312] Matthew Jordan + + * res/ari/resource_channels.c: res/ari/resource_channels: Fix + compilation issue Forgot a parameter. Whoops. + + * res/ari/resource_channels.c: res/ari/resource_channels: Don't + return allocation failure on failed function If a function fails + to execute, it is most likely due to one of two reasons: (1) The + function doesn't exist or can't be read from (2) The function is + dangerous and is restricted based on the user's permissions + Currently we return allocation failure, which is incorrect. This + updates the reason code to more accurately reflect why the + request failed. ASTERISK-24215 + + * /, apps/app_meetme.c: apps/app_meetme: Fix crash when publishing + MeetMe messages with no channel The same function, + meetme_stasis_generate_msg, handles creating and publishing + Stasis message both when there are channels in the MeetMe + conference and when there are no channels in the conference. When + the performance improvement was made to use cached snapshots, + this created a situation where Asterisk would crash: obtaining a + cached snapshot is not NULL tolerant. This patch restores the + previous implementation, which used a NULL safe set of routines + to produce a blob containing the channel snapshot (if available) + and information about the MeetMe conference. ASTERISK-24234 + #close Reported by: Shaun Ruffell Tested by: Shaun Ruffell + ........ Merged revisions 421270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_dial.c, /: apps/app_dial: Fix Dial 'z' option The 'z' + option is supposed to disable the dial timeout in the case of a + call forward. Unfortunately, the wrong timeout timer was passed + to the do_forward function, resulting in the option not working. + ASTERISK-24225 #close Reported by: dimitripietro Tested by: + dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by + rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by + rmudgett (License 5621) ........ Merged revisions 421232 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421233 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421234 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, configure.ac: configure: Undefine FORTIFY_SOURCE + prior to defining it for patched gcc Some distributions of Linux + patch gcc to define FORTIFY_SOURCE when gcc is executed with + optimization. This "help" unfortunately results in re-definition + warnings when FORTIFY_SOURCE is later defined in Asterisk's build + system. This patch undefines FORTIFY_SOURCE prior to defining it + to prevent this warning. Review: + https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close + Reported by: Kilburn Tested by: Kilburn, wdoekes patches: + 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by + cloos (License 5956) 11.diff uploaded by cloos (License 5956) + 12.diff uploaded by cloos (License 5956) 13.diff uploaded by + cloos (License 5956) ........ Merged revisions 421227 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421228 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421229 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-17 16:10 +0000 [r421210] Joshua Colp + + * res/res_http_websocket.c: res_http_websocket: Include query + parameters in client connection requests. Review: + https://reviewboard.asterisk.org/r/3914/ + +2014-08-15 17:08 +0000 [r421187] Jonathan Rose + + * main/channel.c, /: Bridging: Fix a behavioral change when + checking if a channel is leaving a bridge r420934 introduced some + failures in the test suite. Upon investigating, it was discovered + that differences in the way we were evaluating whether a channel + was in the process of leaving a bridge were causing some + reinvites not to occur (mostly reinvites back to Asterisk when + ending a call). This patch fixes that behavioral change. + ASTERISK-24027 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3910/ ........ Merged + revisions 421186 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-15 15:45 +0000 [r421042-421166] Matthew Jordan + + * apps/app_voicemail.c, /, main/app.c: app_voicemail/app: Remove + test events that were duplicated by r421059 Moving the test event + raised when a file is played back (which occurred in r421059) + broke the ever loving snot out of the voicemail tests. This + caused duplicate test events to get raised, as app_voicemail and + main/app were raising events prior to call ast_streamfile. The + voicemail tests did not enjoy getting multiple events. Since + raising the playback event in ast_streamfile is far more useful + to the vast majority of tests, this patch keeps the call there + and simply removes the extraneous calls that duplicated the + event. ........ Merged revisions 421125 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421164 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421165 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_hep_rtcp.c, /: res/res_hep_rtcp: Remove dependency on + PJSIP The res_hep_rtcp module was incorrectly including + . This didn't need to be included, as the module does + not using PJPROJECT any fashion. Unfortunately, because + res_hep_rtcp did not include pjsip in its MODULEINFO as a + dependency, this also meant that res_hep_rtcp will fail to + compile on a system without PJPROJECT. This patch removes the + include. Thanks to Damien Wedhorn for pointing this out in + #asterisk-dev. ASTERISK-24236 #close Reported by: Damien Wedhorn, + Matt Jordan Tested by: Damien Wedhorn ........ Merged revisions + 421064 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/file.c, main/app.c: main/file: Move test event to emit + PLAYBACK event more consistently This is being done in advance of + the test for ASTERISK-23953 ........ Merged revisions 421059 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 421060 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 421061 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c, main/cel.c, /: cel: Make sure channels in extra + fields include their unique IDs as well CEL typically tracks a + lot of information using the unique ID of the channel. This is + typically needed due to tying events together using the linked ID + of the various channels involved in a "call", which is derived + from the channel ID of the oldest channel involved in a bridge + (or in the case of a Dial, the parent channel). Previously, we + had updated the extra fields to include the involved channel + names, but forgot to put in the unique ID. This patch corrects + that error. ........ Merged revisions 421037 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-14 16:32 +0000 [r420957-421010] Richard Mudgett + + * /, res/ari/resource_channels.c: ARI: Originate to app local + channel subscription code optimization. Reduce the scope of + local_peer and only get it if the ARI originate is subscribing to + the channels. Review: https://reviewboard.asterisk.org/r/3905/ + ........ Merged revisions 421009 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel_internal_api.c, main/channel.c: + channel_internal_api.c: Replace some code with ao2_replace(). Use + ao2_replace() instead of ao2_cleanup(); ao2_bump(). ao2_replace() + has the advantange of not altering the ref count if the replaced + pointer is the same. Review: + https://reviewboard.asterisk.org/r/3904/ + + * /, res/res_pjsip_send_to_voicemail.c: + res_pjsip_send_to_voicemail.c: Fix svn file properties. ........ + Merged revisions 420956 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-13 16:53 +0000 [r420950] Kinsey Moore + + * res/res_pjsip.c, /: PJSIP: Prevent crash no-URI contacts This + prevents a crash from occurring when a contact with no URI is + used for the creation of an outbound out-of-dialog request with + no associated endpoint. ........ Merged revisions 420949 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-13 16:07 +0000 [r420940] Jonathan Rose + + * main/bridge_after.c, main/channel_internal_api.c, + include/asterisk/channel.h, apps/app_chanspy.c, + apps/app_mixmonitor.c, apps/app_stack.c, main/bridge_channel.c, + main/channel.c, main/pbx.c, /, main/framehook.c: Bridges: Fix + feature interruption/unintended kick caused by external actions + If a manager or CLI user attached a mixmonitor to a call running + a dynamic bridge feature while in a bridge, the feature would be + interrupted and the channel would be forcibly kicked out of the + bridge (usually ending the call during a simple 1 to 1 call). + This would also occur during any similar action that could set + the unbridge soft hangup flag, so the fix for this was to remove + unbridge from the soft hangup flags and make it a separate thing + all together. ASTERISK-24027 #close Reported by: mjordan Review: + https://reviewboard.asterisk.org/r/3900/ ........ Merged + revisions 420934 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-13 14:24 +0000 [r420919] Kinsey Moore + + * main/manager.c: AMI: Improve documentation for Status action + +2014-08-13 07:52 +0000 [r420899] Walter Doekes + + * /, main/logger.c: logger: Don't store verbose-magic in the log + files. In r399267, the verbose2magic stuff was edited. This time + it results in magic characters in the log files for multiline + messages. In trunk (and 13) this was fixed by the "stripping" of + those characters from multiline messages (in r414798). This fix + is altered to actually strip the characters and not replace them + with blanks. Review: https://reviewboard.asterisk.org/r/3901/ + Review: https://reviewboard.asterisk.org/r/3902/ ........ Merged + revisions 420897 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420898 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-12 23:43 +0000 [r420879-420881] Richard Mudgett + + * channels/chan_sip.c: chan_sip: Fix type mismatch when the format + is changed. Symptom is most likely an invalid ao2 object bad + magic number message or a less likely crash. + + * res/res_stasis_snoop.c: res_stasis_snoop.c: Fix off nominial exit + path leaving Snoop channel locked and not hungup. * Made use + ast_copy_string() instead of strcpy() for snoop uniqueid for + safety. There is no guarantee that the max channel uniqueid + length will remain the same as the snoop uniqueid space. + +2014-08-12 11:17 +0000 [r420856] Joshua Colp + + * apps/app_voicemail.c: app_voicemail: Fix the + "test_voicemail_vm_info" unit test. + +2014-08-11 20:53 +0000 [r420837] Richard Mudgett + + * res/stasis/command.c, /: res/stasis/command.c: Fix recent commit + using spaces instead of tabs. ........ Merged revisions 420836 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 18:50 +0000 [r420808] Matthew Jordan + + * rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json: AMI/ARI: Update version to + 2.5.0/1.5.0 respectively This is to support the backwards + compatible changes made in the next version of Asterisk. ........ + Merged revisions 420805 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 18:46 +0000 [r420796-420803] Kinsey Moore + + * /, res/res_stasis.c: Stasis: Use the correct return value Return + the correct value instead of always returning 0 when setting + internal status on unreal channels. Reported by: Richard Mudgett + ........ Merged revisions 420802 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis.c, res/ari/resource_bridges.c, /, + res/stasis/stasis_bridge.c, include/asterisk/stasis_app.h: + Stasis: Allow internal channels directly into bridges The patch + to catch channels being shoehorned into Stasis() via external + mechanisms also happens to catch Announcer and Recorder channels + because they aren't known to be stasis-controlled channels in the + usual sense. This marks those channels as Stasis()-internal + channels and allows them directly into bridges. Review: + https://reviewboard.asterisk.org/r/3903/ ........ Merged + revisions 420795 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 18:32 +0000 [r420758-420794] Mark Michelson + + * include/asterisk/stasis_app.h, main/stasis_channels.c, + res/ari/resource_channels.c, CHANGES, res/res_pjsip_pubsub.c, + main/manager_channels.c, apps/app_dial.c, res/stasis/app.c, + res/stasis/control.c: Improve call forwarding reporting, + especially with regards to ARI. This patch addresses a few + issues: 1) The order of Dial events have been changed when + performing a call forward. The order has now been altered to 1) + Dial begins dialing channel A. 2) When A forwards the call to B, + we issue the dial end event to channel A, indicating the dial is + being canceled due to a forward to B. 3) When the call to channel + B occurs, we then issue a new dial begin to channel B. 2) Call + forwards are now reported on the calling channel, not the peer + channel. 3) AMI DialEnd events have been altered to display the + extension the call is being forwarded to when relevant. 4) You + can now get the values of channel variables for channels that are + not currently in the Stasis application. This brings the + retrieval of channel variables more in line with the rest of + channel read operations since they may be performed on channels + not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan + ASTERISK-24138 #close Reported by Matt Jordan Patches: + forward-shenanigans.diff uploaded by Matt Jordan (License #6283) + Review: https://reviewboard.asterisk.org/r/3899 + + * res/res_pjsip_pubsub.c: Fix crashing unit tests with regards to + RLS. The unit tests require a sorcery.conf file that has been set + up to store resource lists in memory rather than retrieving from + configuration. With a setup that is not conducive to running the + tests, a fault in sorcery currently causes Asterisk to crash when + attempting to run any of the tests. To get around the crash, this + adds a function that verifies the current environment and marks + the tests as "not run" if the setup is not correct. + + * res/res_pjsip_pubsub.c: Fix crash encountered by the testsuite. + Running testsuite tests locally produced no errors, but when run + using the continuous integration framework, crashes occurred. The + crashes occurred due to a refcounting error that had been fixed + for a similar situation. + +2014-08-11 13:57 +0000 [r420742] Matthew Jordan + + * res/res_hep.c, res/res_hep_pjsip.c, res/res_hep_rtcp.c: res_hep: + Remove disabling of modules These modules were originally + specified as being disabled, as they were introduced midstream in + Asterisk 12. That makes it nicer for folks who are upgrading to a + new release in the middle of Asterisk 12. That's not the case for + Asterisk 13: it's a brand new release. There's no reason to have + the modules disabled by default in that case. + +2014-08-11 10:40 +0000 [r420657-420717] Walter Doekes + + * /, main/utils.c: general: Fix memory Corruption in + __ast_string_field_ptr_build_va. If the space left in a + stringfield is between 0 and + (alignof(ast_string_field_allocation)-1) adding new data would + cause memory corruption, because we would assume enough space + (unsigned underrun). Thanks Arnd Schmitter for reporting and + finding out the cause! ASTERISK-23508 #close Reported by: Arnd + Schmitter Tested by: Arnd Schmitter, JoshE Review: + https://reviewboard.asterisk.org/r/3898/ ........ Merged + revisions 420680 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420715 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420716 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode. + ........ Merged revisions 420654 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420655 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420656 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-11 01:31 +0000 [r420607-420639] Matthew Jordan + + * funcs/func_jitterbuffer.c: funcs/func_jitterbuffer: Tweak + documentation This patch merely reformats and cleans up a bit of + the jitterbuffer documentation for the wiki. + + * UPGRADE.txt, configs/samples/extconfig.conf.sample, CHANGES, + apps/app_queue.c, + contrib/ast-db-manage/config/versions/d39508cb8d8_create_queue_rules.py + (added), configs/samples/queuerules.conf.sample: app_queue: Add + RealTime support for queue rules This patch gives the optional + ability to keep queue rules in RealTime. It is important to note + that with this patch: (a) Queue rules in RealTime are only + examined on module load/reload (b) Queue rules are loaded both + from the queuerules.conf file as well as the RealTime backend To + inform app_queue to examine RealTime for queue rules, a new + setting has been added to queuerules.conf's general section + "realtime_rules". RealTime queue rules will only be used when + this setting is set to "yes". The schema for the database table + supports a rule_name, time, min_penalty, and max_penalty columns. + min_penalty and max_penalty can be relative, if a '-' or '+' + literal is provided. Otherwise, the penalties are treated as + constants. For example: rule_name, time, min_penalty, max_penalty + 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', + '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', + '4564', '46546' 'test_rule', '40', '15', '50' which would result + in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY + to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 + seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust + QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust + QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - + After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust + QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust + QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 + Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to + 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the + queue rules will be always reloaded on a module reload, even if + the underlying file did not change. With the option disabled, the + rules will only be reloaded if the file was modified. Review: + https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close + Reported by: Michael K patches: app_queue.c_realtime_trunk.patch + uploaded by Michael K (License 6621) + + * CHANGES: Update CHANGES file + + * UPGRADE.txt: Update UPGRADE.txt file + +2014-08-08 20:08 +0000 [r420577-420592] Jason Parker + + * apps/app_voicemail.c: Fix build in devmode. + + * CHANGES, configs/samples/voicemail.conf.sample, + apps/app_voicemail.c: app_voicemail: Add the ability to specify + multiple email addresses. ASTERISK-24045 Reported by: Jacob + Barber Review: https://reviewboard.asterisk.org/r/3833/ + +2014-08-08 17:53 +0000 [r420534-420562] Matthew Jordan + + * channels/chan_sip.c, channels/sip/security_events.c, + channels/sip/dialplan_functions.c, channels/sip/reqresp_parser.c, + channels/sip/route.c, channels/sip/utils.c, + channels/sip/config_parser.c: chan_sip: Mark chan_sip and its + files as extended support + + * rest-api-templates/make_ari_stubs.py: make_ari_stubs: Update wiki + prefix to '13' + + * rest-api-templates/res_ari_resource.c.mustache: + res_ari_resource.c.mustache: Update template to emit module + support level + + * main/message.c, /: main/message: remove debug message ........ + Merged revisions 420533 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-08 03:03 +0000 [r420514] Kinsey Moore + + * tests/test_cel.c, /: CEL: Update unit tests for additional + information This updates the CEL unit tests for the new + information contained in the attended transfer CEL extra field. + ........ Merged revisions 420513 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-08 01:31 +0000 [r420494-420496] Matthew Jordan + + * UPGRADE.txt: Update UPGRADE file for 13 branch + + * /: Remove old properties + + * / (added): ___ _ _ _ __ _____ / _ \ | | (_) | | / ||____ | / /_\ + \___| |_ ___ _ __ _ ___| | __ `| | / / | _ / __| __/ _ | '__| / + __| |/ / | | \ \ | | | \__ | || __| | | \__ | < _| |.___/ / \_| + |_|___/\__\___|_| |_|___|_|\_\ \___\____/ + +2014-08-07 21:58 +0000 [r420437] Richard Mudgett + + * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and + resolve the large SDP poll issue. Replace sip_tls_read() and + sip_tcp_read() with a single function and resolve the poll/wait + issue with large SDP payloads. ASTERISK-18345 #close Reported by: + Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) + patch uploaded by Elazar Broad Review: + https://reviewboard.asterisk.org/r/3882/ ........ Merged + revisions 420434 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420435 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420436 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-07 21:17 +0000 [r420389-420415] Kinsey Moore + + * main/stasis_bridges.c, /: Stasis: Correct blind transfer message + generation This fixes the json object creation format string and + key name for the BridgeBlindTransfer Stasis event allowing it to + be published properly. ........ Merged revisions 420414 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_bridges.c, /: Stasis: Ensure transfer messages follow + validation rules This makes Stasis() event generation for + transfer messages follow validation rules. Currently, + ast_json_null() is being used in place of omitting a key entirely + which falls afoul of these validation rules. + https://reviewboard.asterisk.org/r/3892/ ........ Merged + revisions 420408 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c: Fix build in dev mode + +2014-08-07 19:44 +0000 [r420384-420388] Mark Michelson + + * /, main/bridge.c: Ensure bridges exist when trying to determine + bridged parties when publishing transfer information. ........ + Merged revisions 420387 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/strings.c, include/asterisk/res_pjsip_presence_xml.h, + res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c, + res/res_pjsip_xpidf_body_generator.c, include/asterisk/strings.h, + res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pidf_body_generator.c: Add support for RFC 4662 + resource list subscriptions. This commit adds the ability for a + user to configure a resource list in pjsip.conf. Subscribing to + this list simultaneously subscribes the subscriber to all + resources listed. This has the potential to reduce the amount of + SIP traffic when loads of subscribers on a system attempt to + subscribe to each others' states. + +2014-08-07 18:51 +0000 [r420364] Richard Mudgett + + * include/asterisk/format_compatibility.h, + channels/iax2/format_compatibility.c, + channels/iax2/include/codec_pref.h, main/format_compatibility.c, + channels/chan_iax2.c, channels/iax2/codec_pref.c, + channels/iax2/include/format_compatibility.h: chan_iax2: Several + media format fixes. * Fixed the iax.conf bandwidth option. This + is the root cause of ASTERISK-24150. * Added checks in + iax2_request() to ensure that there are actual formats requested + for the new channel to prevent any more fracks from issues like + ASTERISK-24150. This is a consequence of the iax.conf bandwidth + option not working. * Fixed struct iax2_codec_pref.order member + size mismatch issue when converting to and from the codec + preference order list passed over the wire. In addition the + values sent over the wire are now compatible with previous + Asterisk versions. * Fixed several issues dealing with the struct + iax2_codec_pref members. Off-by-one, array limit errors, and the + order/framing members always need to be updated together. * Made + iax2_request() setup the channel's native format preference order + according to the user's wishes. The new media format strategy + needs the order specified earler. * Fixed usage of + ast_format_compatibility_bitfield2format(). The function can + return NULL if the bitfield was not associated with a function. * + Deleted dead code iax2_codec_pref_getsize() and + iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and + iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of + inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH, + IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants + again as they were in Asterisk v1.8. * Renamed prefs to + prefs_global so it won't get confused with the local pref + versions. * Fixed too small buffer in + handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in + handle_cli_iax2_show_peer() to output complete lines. * Changed + struct create_addr_info.prefs to be struct iax2_codec_pref as an + optimization so iax2_request() and iax2_call() do less work. * + Fixed a potential deadlock in ast_iax2_new() on an off-nominal + path when the pbx could not get started. * Made set_config() + setup a local prefs list along side the local capability format + bitfield. Once the config is loaded, then the local copies are + put into the global versions. * Fix unininialized codec_buf in + function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott + Griepentrog Review: https://reviewboard.asterisk.org/r/3890/ + +2014-08-07 15:30 +0000 [r420338] Kinsey Moore + + * include/asterisk/bridge_features.h, res/res_stasis.c, + res/stasis/command.c, rest-api/api-docs/events.json, /, + res/stasis/app.c, res/stasis/control.c, main/bridge.c, + main/bridge_basic.c, res/stasis/stasis_bridge.c, + include/asterisk/stasis_bridges.h, res/stasis/command.h, + include/asterisk/stasis_app.h, res/stasis/app.h, + res/stasis/control.h, apps/app_queue.c, + res/ari/ari_model_validators.c, main/cel.c, + main/stasis_bridges.c, res/ari/ari_model_validators.h, + main/channel.c, include/asterisk/datastore.h, tests/test_cel.c: + Stasis: Convey transfer information to applications This fixes a + class of issues where Stasis applications were not made aware + that their channels were being manipulated or replaced by + external entitiessuch as transfers, AMI commands, or dialplan + applications such as Bridge(). Inconsistent information such as + StasisEnd events with unknown channels as a result of masquerades + has also been corrected. To accomplish these fixes, several new + fields were added to blind and attended transfer messages as well + as StasisStart and BridgeAttendedTransfer Stasis events. + ASTERISK-23941 #close Review: + https://reviewboard.asterisk.org/r/3865/ Review: + https://reviewboard.asterisk.org/r/3857/ Review: + https://reviewboard.asterisk.org/r/3852/ Review: + https://reviewboard.asterisk.org/r/3816/ Review: + https://reviewboard.asterisk.org/r/3731/ Review: + https://reviewboard.asterisk.org/r/3729/ Review: + https://reviewboard.asterisk.org/r/3728/ ........ Merged + revisions 420325 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-07 14:37 +0000 [r420314-420315] Joshua Colp + + * include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pubsub.exports.in, res/res_pjsip_publish_asterisk.c + (added), res/res_pjsip_pubsub.c: res_pjsip_publish_asterisk: Add + support for exchanging device and mailbox state using SIP. This + module uses the inbound and outbound PUBLISH support to exchange + device and mailbox state between Asterisk instances. Each + instance is configured to publish to the other and requires no + intermediary server. The functionality provided is similar to the + XMPP and Corosync support. Review: + https://reviewboard.asterisk.org/r/3780/ + + * include/asterisk/res_pjsip_outbound_publish.h (added), + res/res_pjsip_outbound_publish.exports.in (added), + res/res_pjsip_outbound_publish.c (added): + res_pjsip_outbound_publish: Add module which provides outbound + PUBLISH support. This module implements the core parts required + for doing outbound PUBLISH. It takes care of configuration, + lifetime management, and authentication. Additional modules + implement the specific events that are published. Review: + https://reviewboard.asterisk.org/r/3780/ + +2014-08-07 14:17 +0000 [r420289-420309] Matthew Jordan + + * main/pbx.c: pbx: Filter out pattern matching hints in responses + sent to ExtensionStateList Hints that are a pattern match are + technically stored in the hint container in the same fashion as + concrete implementations of hints. The pattern matching hints, + however, are not "real" in the sense that things can subscribe to + them: rather, they are stored in the hints container so that when + a subscription is made a "real" hint can be generated for the + subscription if one does not yet exist. The extension state core + takes care of this correctly by matching against non-pattern + matching extensions prior to pattern matching extensions. Because + of this, however, the ExtensionStateList AMI action was returning + pattern matching hints when executed. These hints are meaningless + from the perspective of AMI clients: their state will never + change, they cannot be subscribed to, and events would never + normally be generated from them. As such, we now filter these out + of the response. + + * build_tools/post_process_documentation.py: build_tools: Skip + managerEvent combining for AMI action responses AMI action + responses can (and will) reference AMI events that they return. + These event references and definitions should not be combined + with AMI events raised elsewhere in the code, as they are + specifically tied to the AMI action that raised them. + ASTERISK-24156 #close Reported by: Rusty Newton + +2014-08-06 18:12 +0000 [r420212-420237] Richard Mudgett + + * contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py, + /: Fix alembic script to work properly in offline mode. When run + in offline mode, this would attempt to check the database for the + presence of a type it was going to try to create. I now check the + context to see if we're running in offline mode and change a + parameter accordingly. ........ Merged revisions 407567 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py + (added), /: Add alembic script that adds contact user_agent and + endpoint message_context. ........ Merged revisions 411514 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py + (added), /, + contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py, + contrib/ast-db-manage/config.ini.sample, + contrib/ast-db-manage/config/versions/1758e8bbf6b_increase_useragent_column_size.py + (added), + contrib/ast-db-manage/config/versions/5139253c0423_make_q_member_uniqueid_autoinc.py + (added), contrib/ast-db-manage/cdr.ini.sample, + contrib/ast-db-manage/voicemail.ini.sample: alembic: Adjust + sippeers, queue_members, and voicemail_messages tables. * + Increased the sippeers useragent max string size to 255. * + Changed the queue_members uniqueid to an auto incremented integer + instead of a string. * Increased the voicemail_messages BLOB size + to LONGBLOB on mysql. * Fixed the add_tables_for_pjsip config + change version downgrade actions to drop a table it created. * + Adjusted the sample alembic.ini files cdr.ini.sample, + config.ini.sample, and voicemail.ini.sample to give a mysql and + postgres sqlalchemy.url lines. ASTERISK-23847 #close Reported by: + Stephen More ASTERISK-23825 #close Reported by: Stephen More + ASTERISK-23909 #close Reported by: Stephen More Review: + https://reviewboard.asterisk.org/r/3870/ ........ Merged + revisions 420211 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-06 16:12 +0000 [r420149] George Joseph + + * /, pbx/pbx_lua.c, main/pbx.c: pbx_lua: fix regression with global + sym export and context clash by pbx_config. ASTERISK-23818 (lua + contexts being overwritten by contexts of the same name in + pbx_config) surfaced because pbx_lua, having the + AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before + pbx_config. Since I couldn't find any reason for pbx_lua to + export it's symbols to the rest of Asterisk, I simply changed the + flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't + realize was that the symbols need to be exported not because + Asterisk needs them but because any external Lua modules like + luasql.mysql need the base Lua language APIs exported + (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's + an issue in pbx.c where context_merge was only merging includes, + switches and ignore patterns if the context was already existing + AND has extensions, or if the context was brand new. If pbx_lua + is loaded before pbx_config, the context will exist BUT pbx_lua, + being implemented as a switch, will never place extensions in it, + just the switch statement. The result is that when pbx_config + loads, it never merges the switch statement created by pbx_lua + into the final context. This patch sets pbx_lua's modflag back to + AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge + that catches the case where an existing context has includes, + switchs or ingore patterns but no actual extensions. + ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo + Teräs Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3891/ ........ Merged + revisions 420146 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 420147 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 420148 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-06 15:32 +0000 [r420144] Walter Doekes + + * funcs/func_channel.c: Add documentation to the ability to + retrieve the source port of a SIP call. (belongs with r419970) + ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by + dtryba Review: https://reviewboard.asterisk.org/r/3781/ + +2014-08-06 12:55 +0000 [r420124] Kinsey Moore + + * configs/samples/stasis.conf.sample (added), main/named_acl.c, + apps/app_queue.c, main/stasis_bridges.c, main/loader.c, + main/stasis.c, apps/app_forkcdr.c, main/stasis_message.c, + funcs/func_cdr.c, res/res_corosync.c, res/res_stun_monitor.c, + res/res_stasis_test.c, res/res_stasis.c, apps/app_chanspy.c, + main/stasis_cache.c, main/pickup.c, main/security_events.c, + include/asterisk/stasis.h, main/devicestate.c, main/core_local.c, + res/res_stasis_snoop.c, main/endpoints.c, main/presencestate.c, + main/cdr.c, main/channel.c, main/stasis_system.c, main/manager.c, + main/test.c, main/file.c, main/app.c, pbx/pbx_realtime.c, + main/stasis_channels.c, tests/test_stasis.c, + res/parking/parking_manager.c, main/stasis_endpoints.c, + main/rtp_engine.c, main/ccss.c, main/bridge.c, + tests/test_stasis_channels.c: Stasis: Allow message types to be + blocked This introduces stasis.conf and a mechanism to prevent + certain message types from being published. Internally, this + works by preventing the chosen message types from being created + which ensures that those message types can never be published. + This patch also adjusts message publishers such that message + payloads are not created if the related message type is not + available. ASTERISK-23943 #close Review: + https://reviewboard.asterisk.org/r/3823/ + +2014-08-05 21:48 +0000 [r420098-420100] Matthew Jordan + + * res/stasis/messaging.c, /: stasis: Fix compilation issue with ao2 + tagged objects ........ Merged revisions 420099 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /, + channels/chan_sip.c, res/stasis/app.c, res/stasis/messaging.h + (added), res/ari/resource_endpoints.h, res/res_pjsip_messaging.c, + tests/test_message.c (added), res/res_xmpp.c, + include/asterisk/json.h, CHANGES, include/asterisk/manager.h, + res/ari/ari_model_validators.c, res/ari/ari_model_validators.h, + main/json.c, res/res_ari_endpoints.c, include/asterisk/message.h, + res/ari/resource_channels.c, main/message.c, res/res_stasis.c, + res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json: + Multiple revisions 420089-420090,420097 ........ r420089 | + mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines + ARI: Add channel technology agnostic out of call text messaging + This patch adds the ability to send and receive text messages + from various technology stacks in Asterisk through ARI. This + includes chan_sip (sip), res_pjsip_messaging (pjsip), and + res_xmpp (xmpp). Messages are sent using the endpoints resource, + and can be sent directly through that resource, or to a + particular endpoint. For example, the following would send the + message "Hello there" to PJSIP endpoint alice with a display URI + of sip:asterisk@mycooldomain.org: + ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There + This is equivalent to the following as well: + ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There + Both forms are available for message technologies that allow for + arbitrary destinations, such as chan_sip. Inbound messages can + now be received over ARI as well. An ARI application that + subscribes to endpoints will receive messages from those + endpoints: { "type": "TextMessageReceived", "timestamp": + "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": + "PJSIP", "resource": "alice", "state": "online", "channel_ids": + [] }, "message": { "from": "\"alice\" ", + "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", + "variables": [] }, "application": "testsuite" } The above was + made possible due to some rather major changes in the message + core. This includes (but is not limited to): - Users of the + message API can now register message handlers. A handler has two + callbacks: one to determine if the handler has a destination for + the message, and another to handle it. - All dialplan + functionality of handling a message was moved into a message + handler provided by the message API. - Messages can now have the + technology/endpoint associated with them. Various other + properties are also now more easily accessible. - A number of ao2 + containers that weren't really needed were replaced with vectors. + Iteration over ao2_containers is expensive and pointless when the + lifetime of things is well defined and the number of things is + very small. res_stasis now has a new file that makes up its + structure, messaging. The messaging functionality implements a + message handler, and passes received messages that match an + interested endpoint over to the app for processing. Note that + inadvertently while testing this, I reproduced ASTERISK-23969. + res_pjsip_messaging was incorrectly parsing out the 'to' field, + such that arbitrary SIP URIs mangled the endpoint lookup. This + patch includes the fix for that as well. Review: + https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close + Reported by: Matt Jordan ASTERISK-23969 #close Reported by: + Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 + -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties + :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, + 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing + compilation issue ........ Merged revisions 420089-420090,420097 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-05 13:59 +0000 [r420028] Jonathan Rose + + * main/format.c: chan_iax2: Fix a crash that occurs when using + allow=all for an IAX2 peer Or any combination of codecs that + includes Opus. ASTERISK-24107 #close Review: + https://reviewboard.asterisk.org/r/3885/ + +2014-08-04 21:00 +0000 [r420007] Richard Mudgett + + * main/format_cache.c, include/asterisk/format_cache.h: Remove + duplicate definitions of ast_format_vp8. + +2014-08-04 20:25 +0000 [r419970] Mark Michelson + + * channels/sip/dialplan_functions.c: Add the ability to retrieve + the source port of a SIP call. This adds the ability to call + CHANNEL(recvport) on chan_sip channels to see the port on which + an INVITE was received. ASTERISK-24040 #close Reported by dtryba + Patches: dialplan_functions.patch uploaded by dtryba (License + #6628) Review: https://reviewboard.asterisk.org/r/3781 + +2014-08-04 19:47 +0000 [r419945] Rusty Newton + + * main/manager.c, /: Manager - Improve documentation for manager + commands Getvar and Setvar. The documentation for these commands + did not make it clear that they could accept expressions and + functions. Modified to make this clear, but tried not to be + overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton + Tested by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3854 ........ Merged revisions + 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 419943 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419944 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-08-02 03:37 +0000 [r419914] Kinsey Moore + + * res/res_pjsip.c: Manager: Add PJSIPShowEndpoint[s] documentation + This adds a large swath of response documentation for + PJSIPShowEndpoint and PJSIPShowEndpoints AMI commands. It relies + heavily on the existing text in the configInfo documentation via + xi:include tags to avoid documentation duplication. Review: + https://reviewboard.asterisk.org/r/3888/ + +2014-08-01 14:48 +0000 [r419888] Mark Michelson + + * CHANGES, res/res_pjsip/pjsip_options.c: Add ContactStatusDetail + to PJSIPShowEndpoint AMI output. Now when running + PJSIPShowEndpoint, you will receive a ContactStatusDetail for + each bound contact that Asterisk is qualifying. This information + includes the URI of the contact, current reachability, and + roundtrip time. AFS-91 #close Reported by Mark Michelson Review: + https://reviewboard.asterisk.org/r/3797 + +2014-07-31 16:19 +0000 [r419851] Jonathan Rose + + * CHANGES, res/res_pjsip_notify.c: PJSIP: Send Notify AMI and CLI + commands can now send to URI instead of endpoint Review: + https://reviewboard.asterisk.org/r/3817/ + +2014-07-31 11:57 +0000 [r419822-419825] Matthew Jordan + + * main/rtp_engine.c, /, res/res_hep_rtcp.c (added), CHANGES, + channels/chan_pjsip.c, res/res_rtp_asterisk.c: res_hep_rtcp: Add + module that sends RTCP information to a Homer Server This patch + adds a new module to Asterisk, res_hep_rtcp. The module + subscribes to the RTCP topics in Stasis and receives RTCP + information back from the message bus. It encodes into HEPv3 + packets and sends the information to the res_hep module for + transmission. Using this, someone with a Homer server can get + live call quality monitoring for all RTP-based channels in their + Asterisk 12+ systems. In addition, there were a few bugs in the + RTP engine, res_rtp_asterisk, and chan_pjsip that were uncovered + by the tests written for the Asterisk Test Suite. This patch + fixes the following: 1) chan_pjsip failed to set its channel + unique ids on its RTP instance on outbound calls. It now does + this in the appropriate location, in the serialized call + callback. 2) The rtp_engine was overflowing some values when + packed into JSON. Specifically, some longs and unsigned ints + can't be be packed into integer values, for obvious reasons. + Since libjansson only supports integers, floats, strings, + booleans, and objects, we print these values into strings. 3) + res_rtp_asterisk had a few problems: (a) it would emit a source + IP address of 0.0.0.0 if bound to that IP address. We now use + ast_find_ourip to get a better IP address, and properly marshal + the result into an ast_strdupa'd string. (b) Reports can be + generated with no report bodies. In particular, this occurs when + a sender is transmitting information to a receiver (who will send + no RTP back to the sender). As such, the sender has no report + body for what it received. We now properly handle this case, and + the sender will emit SR reports with no body. Likewise, if we + receive an RTCP packet with no report body, we will still + generate the appropriate events. ASTERISK-24119 #close ........ + Merged revisions 419823 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * funcs/func_jitterbuffer.c, doc/appdocsxml.dtd, main/xmldoc.c: + xmldocs: Add support for an tag in the Asterisk XML + Documentation This patch adds support for an tag in + the XML documentation schema. For CLI help, this doesn't change + the formatting too much: - Preceeding white space is removed - + Unlike with para elements, new lines are preserved However, + having an tag in the XML schema allows for the wiki + documentation generation script to surround the documentation + with {code} or {noformat} tags, generating much better content + for the wiki - and allowing us to put dialplan examples (and + other code snippets, if desired) into the documentation for an + application/function/AMI command/etc. Review: + https://reviewboard.asterisk.org/r/3807/ + +2014-07-30 18:32 +0000 [r419806] Kinsey Moore + + * main/manager.c, res/res_manager_presencestate.c, + res/res_manager_devicestate.c, main/pbx.c: manager: Add state + list commands This patch adds three new AMI commands: * + ExtensionStateList (pbx.c) - list all known extension state hints + and their current statuses. Events emitted by the list action are + equivalent to the ExtensionStatus events. * PresenceStateList + (res_manager_presencestate) - list all known presence state + values. Events emitted are generated by the stasis message type, + and hence are PresenceStateChange events. * DeviceStateList + (res_manager_devicestate) - list all known device state values. + Events emitted are generated by the stasis message type, and + hence are DeviceStateChange events. Patch-by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3799/ + +2014-07-29 19:41 +0000 [r419789] Mark Michelson + + * main/manager.c: Do not omit the first header of a UserEvent AMI + action from the corresponding emitted UserEvent. ASTERISK-24124 + #close Reported by Matt Jordan AFS-131 #close Reported by Matt + Jordan Patches: userevent.patch uploaded by Matt Jordan (License + #6283) + +2014-07-29 10:56 +0000 [r419751-419766] Joshua Colp + + * res/res_pjsip_session.c, /: res_pjsip_session: Fix race condition + where redirecting information may not be set. Since the PJSIP + INVITE session module is invoked before any session supplements + it was possible for it to handle a redirect before the + res_pjsip_diversion module interpreted and set redirecting + information on the channel. This would cause the redirecting + information to get lost. This patch ensures that session + supplements are *always* invoked before a redirect occurs by + explicitly calling them in the redirect handler. Review: + https://reviewboard.asterisk.org/r/3850/ ........ Merged + revisions 419764 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_xpidf_body_generator.c, + res/res_pjsip_pidf_body_generator.c: + res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator: + Ensure local entity is unquoted. The local entity as provided by + PJSIP is quoted within '<' and '>'. As a result placing this + value into XML will result in malformed XML being produced. This + patch now unquotes the local entity so it can go safely into the + XML. Review: https://reviewboard.asterisk.org/r/3851/ ........ + Merged revisions 419750 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-28 18:58 +0000 [r419688] Richard Mudgett + + * apps/app_speech_utils.c, main/channel.c, /, + funcs/func_frame_trace.c, main/abstract_jb.c: datastores: Audit + ast_channel_datastore_remove usage. Audit of v1.8 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leaks in app_speech_utils and func_frame_trace. * Fixed + app_speech_utils not locking the channel when accessing the + channel datastore list. Review: + https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leak in func_jitterbuffer. (Was not in v12) Review: + https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of + ast_channel_datastore_remove() for datastore memory leaks. * + Fixed leaks in abstract_jb. * Fixed leak in + ast_channel_unsuppress(). Used by ARI mute control and + res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used + by ARI mute control and res_mutestream. Review: + https://reviewboard.asterisk.org/r/3861/ ........ Merged + revisions 419684 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 419685 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419686 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-25 18:09 +0000 [r419612] Joshua Colp + + * main/loader.c: loader: Fix an infinite loop when printing modules + using "module show". When creating the alphabetical sorted list + each module is added to a list temporarily. On the second + iteration each module already has a pointer to another module, + causing stuff to go into a loop. ASTERISK-24123 #close Reported + by: Malcolm Davenport + +2014-07-25 16:47 +0000 [r419592] Mark Michelson + + * res/res_ari_sounds.c, res/res_stasis.c, res/res_fax_spandsp.c, + res/res_timing_kqueue.c, res/res_odbc.c, + res/res_pjsip_transport_websocket.c, apps/app_voicemail.c, + res/res_calendar.c, channels/chan_unistim.c, cel/cel_radius.c, + channels/chan_multicast_rtp.c, res/res_pjsip_notify.c, + res/res_snmp.c, formats/format_sln.c, apps/app_meetme.c, + apps/app_dictate.c, codecs/codec_gsm.c, res/res_stasis_snoop.c, + res/res_musiconhold.c, res/res_format_attr_h264.c, + res/res_http_websocket.c, apps/app_followme.c, + res/res_config_sqlite.c, formats/format_siren7.c, cdr/cdr_csv.c, + formats/format_ilbc.c, channels/chan_phone.c, + apps/app_setcallerid.c, apps/app_osplookup.c, cel/cel_custom.c, + apps/app_mp3.c, res/res_agi.c, channels/chan_motif.c, + res/res_timing_timerfd.c, apps/app_confbridge.c, + res/res_format_attr_silk.c, formats/format_siren14.c, + res/res_sorcery_realtime.c, channels/chan_mgcp.c, + apps/app_jack.c, codecs/codec_lpc10.c, + res/res_pjsip_pidf_body_generator.c, res/res_config_pgsql.c, + funcs/func_dialplan.c, apps/app_nbscat.c, cdr/cdr_syslog.c, + res/res_pjsip_authenticator_digest.c, apps/app_festival.c, + res/res_fax.c, apps/app_waitforsilence.c, res/res_adsi.c, + res/res_crypto.c, res/res_ari_applications.c, + res/res_hep_pjsip.c, pbx/pbx_lua.c, res/res_pjsip_messaging.c, + res/res_pjsip_caller_id.c, channels/chan_console.c, + apps/app_getcpeid.c, res/res_stasis_answer.c, + channels/chan_oss.c, res/res_pjsip_nat.c, + res/res_pjsip_session.c, cdr/cdr_tds.c, + res/res_pjsip_header_funcs.c, res/res_parking.c, + formats/format_vox.c, res/res_pjsip_rfc3326.c, + res/res_ari_endpoints.c, res/res_stun_monitor.c, + res/res_pjsip_mwi.c, res/res_stasis_recording.c, + res/res_pjsip_xpidf_body_generator.c, apps/app_sms.c, + codecs/codec_ulaw.c, channels/chan_nbs.c, apps/app_stack.c, + channels/chan_pjsip.c, formats/format_g729.c, cel/cel_pgsql.c, + res/res_sorcery_config.c, res/res_ari.c, addons/chan_ooh323.c, + cdr/cdr_sqlite3_custom.c, codecs/codec_adpcm.c, + res/res_ari_asterisk.c, res/res_calendar_caldav.c, + apps/app_image.c, apps/app_ices.c, formats/format_wav_gsm.c, + main/cli.c, res/res_format_attr_celt.c, res/res_rtp_multicast.c, + channels/chan_dahdi.c, funcs/func_pitchshift.c, res/res_smdi.c, + res/res_pjsip_one_touch_record_info.c, pbx/pbx_ael.c, + pbx/pbx_realtime.c, apps/app_amd.c, channels/chan_alsa.c, + formats/format_h263.c, apps/app_url.c, res/res_pjsip_acl.c, + apps/app_externalivr.c, res/res_curl.c, formats/format_gsm.c, + res/res_speech.c, cdr/cdr_manager.c, res/res_calendar_exchange.c, + codecs/codec_g722.c, res/res_pjsip_multihomed.c, + res/res_ari_mailboxes.c, cel/cel_tds.c, res/res_sorcery_memory.c, + apps/app_fax.c, codecs/codec_speex.c, res/res_pjsip_sdp_rtp.c, + codecs/codec_g726.c, formats/format_ogg_vorbis.c, + apps/app_talkdetect.c, res/res_ari_channels.c, + res/res_pjsip_exten_state.c, apps/app_speech_utils.c, + apps/app_agent_pool.c, apps/app_waitforring.c, res/res_statsd.c, + addons/cdr_mysql.c, formats/format_g726.c, res/res_ari_bridges.c, + addons/app_mysql.c, res/res_stasis_playback.c, + addons/format_mp3.c, res/res_pjsip_endpoint_identifier_ip.c, + res/res_phoneprov.c, res/res_pjsip_t38.c, + res/res_pjsip_registrar_expire.c, cdr/cdr_pgsql.c, + cdr/cdr_radius.c, res/res_chan_stats.c, + res/res_format_attr_opus.c, res/res_config_odbc.c, + funcs/func_audiohookinherit.c, + res/res_pjsip_outbound_registration.c, cel/cel_manager.c, + funcs/func_odbc.c, res/res_pjsip_endpoint_identifier_anonymous.c, + funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c, + apps/app_minivm.c, res/res_pjsip_log_forwarder.c, + formats/format_h264.c, res/res_config_ldap.c, apps/app_ivrdemo.c, + addons/chan_mobile.c, apps/app_stasis.c, + res/res_pjsip_diversion.c, cdr/cdr_custom.c, apps/app_adsiprog.c, + res/res_pjsip_dtmf_info.c, res/res_rtp_asterisk.c, + res/res_calendar_icalendar.c, res/res_hep.c, channels/chan_sip.c, + channels/chan_bridge_media.c, codecs/codec_alaw.c, + apps/app_queue.c, res/res_srtp.c, funcs/func_presencestate.c, + res/res_timing_pthread.c, res/res_manager_presencestate.c, + res/res_corosync.c, apps/app_celgenuserevent.c, + cel/cel_sqlite3_custom.c, res/snmp/agent.c, pbx/pbx_dundi.c, + formats/format_g723.c, funcs/func_devstate.c, + res/res_pjsip_registrar.c, + res/res_pjsip_pidf_eyebeam_body_supplement.c, + addons/res_config_mysql.c, + res/res_pjsip_pidf_digium_body_supplement.c, apps/app_test.c, + res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c, + apps/app_alarmreceiver.c, apps/app_chanisavail.c, + res/res_format_attr_h263.c, res/res_pjsip_mwi_body_generator.c, + res/res_xmpp.c, res/res_http_post.c, channels/chan_iax2.c, + res/res_pjsip_endpoint_identifier_user.c, res/res_pjsip.c, + res/res_pktccops.c, res/res_pjsip_send_to_voicemail.c, + main/loader.c, cel/cel_odbc.c, res/res_ari_model.c, + channels/chan_skinny.c, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_mwi_external.c, apps/app_skel.c, formats/format_pcm.c, + include/asterisk/module.h, res/res_pjsip_path.c, + res/res_ari_playbacks.c, res/res_pjsip_pubsub.c, cdr/cdr_odbc.c, + funcs/func_periodic_hook.c, res/res_stasis_test.c, + formats/format_jpeg.c, res/res_pjsip_refer.c, + formats/format_g719.c, res/res_clialiases.c, + res/res_config_sqlite3.c, res/res_ari_device_states.c, + formats/format_wav.c, apps/app_saycounted.c, apps/app_dahdiras.c, + apps/app_morsecode.c, res/res_stasis_mailbox.c, + res/res_ael_share.c, res/res_mwi_external_ami.c, + res/res_pjsip_logger.c, res/res_stasis_device_state.c, + res/res_calendar_ews.c, res/res_monitor.c, apps/app_playback.c, + res/res_ari_recordings.c, res/res_manager_devicestate.c, + res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c, + res/res_ari_events.c, res/res_pjsip_dialog_info_body_generator.c, + res/res_sorcery_astdb.c, codecs/codec_dahdi.c, + apps/app_zapateller.c, pbx/pbx_config.c: Add module support level + to ast_module_info structure. Print it in CLI "module show" . + ASTERISK-23919 #close Reported by Malcolm Davenport Review: + https://reviewboard.asterisk.org/r/3802 + +2014-07-25 14:47 +0000 [r419563-419567] Matthew Jordan + + * CHANGES, res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, + res/ari/ari_model_validators.h, /, res/res_stasis_recording.c: + Multiple revisions 419565-419566 ........ r419565 | mjordan | + 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines ARI: + report duration values in LiveRecording objects This patch adds + three new fields to the LiveRecording model: - total_duration: + the total length of the live recording - talking_duration: + optional. The duration of talking energy that was detected while + the recording was made. - silence_duration: optional. The + duration of silence that was detected while the recording was + made. These values are reported in the RecordingFinished ARI + event. When a DSP is enabled on the channel during the recording + - which occurs when the recording is created with + max_silence_seconds (indicating that the user actually cares + about how much silence is in the file), we will report the + talking_duration and silence_duration in addition to the + total_duration. Review: https://reviewboard.asterisk.org/r/3770/ + ASTERISK-24037 #close Reported by: Samuel Galarneau ........ + r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) + | 1 line Update CHANGES for r419565 ........ Merged revisions + 419565-419566 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/loader.c, res/res_calendar.c: module loader: Unload modules + in reverse order of their start order When Asterisk starts a + module (calling its load_module function), it re-orders the + module list, sorting it alphabetically. Ostensibly, this was done + so that the output of 'module show' listed modules in alphabetic + order. This had the unfortunate side effect of making modules + with complex usage patterns unloadable. A module that has a large + number of modules that depend on it is typically abandoned during + the unloading process. This results in its memory not being + reclaimed during exit. Generally, this isn't harmful - when the + process is destroyed, the operating system will reclaim all + memory allocated by the process. Prior to Asterisk 12, we also + didn't have many modules with complex dependencies. However, with + the advent of ARI and PJSIP, this can make make unloading those + modules successfully nearly impossible, and thus tracking memory + leaks or ref debug leaks a real pain. While this patch is not a + complete overhaul of the module loader - such an effort would be + beyond the scope of what could be done for Asterisk 13 - this + does make some marginal improvements to the loader such that + modules like res_pjsip or res_stasis *may* be made properly + un-loadable in the future. 1. The linked list of modules has been + replaced with a doubly linked list. This allows traversal of the + module list to occur backwards. The module shutdown routine now + walks the global list backwards when it attempts to unload + modules. 2. The alphabetic reorganization of the module list on + startup has been removed. Instead, a started module is placed at + the end of the module list. 3. The ast_update_module_list + function - which is used by the CLI to display the modules - now + does the sorting alphabetically itself. It creates its own linked + list and inserts the modules into it in alphabetic order. This + allows for the intent of the previous code to be maintained. This + patch also contains a fix for res_calendar. Without + calendar.conf, the calendar modules were improperly bumping the + use count of res_calendar, then failing to load themselves. This + patch makes it so that we detect whether or not calendaring is + enabled before altering the use count. Review: + https://reviewboard.asterisk.org/r/3777/ + +2014-07-25 10:54 +0000 [r419537-419539] Joshua Colp + + * apps/app_bridgewait.c, /: app_bridgewait: Remove possibility of + race condition between channels leaving/joining. Bridges created + by app_bridgewait previously had the "dissolve when empty" flag + set. This caused the bridge core to destroy them when the last + channel had left. This introduced a race condition where we may + have a reference to the bridge but it is not actually joinable + when we try to join it. This flag has now been removed and the + bridge is guaranteed to be joinable at all times. ASTERISK-23987 + #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3836/ ........ Merged + revisions 419538 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/bridge.c: bridge: Make "bridge destroy" only available in + developer mode and add "all" to "bridge kick". The "bridge + destroy" CLI command is invasive to bridges and can leave them in + an unexpected state for the users of them. Since this command may + be useful for developers it is now only available when developer + mode is available. To take its place "all" has been added as a + valid option to the "bridge kick" CLI command. It will kick all + of the channels in the bridge out. ASTERISK-23987 Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/ + ........ Merged revisions 419536 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-24 22:48 +0000 [r419520] Richard Mudgett + + * main/bridge.c, main/bridge_basic.c, main/core_unreal.c, + UPGRADE.txt, include/asterisk/channel.h, CHANGES, + apps/app_followme.c, apps/app_queue.c, main/cel.c, + res/parking/parking_bridge_features.c, apps/app_dial.c, + main/channel.c, main/dial.c, main/pbx.c: accountcode: Slightly + change accountcode propagation. The previous behavior was to + simply set the accountcode of an outgoing channel to the + accountcode of the channel initiating the call. It was done this + way a long time ago to allow the accountcode set on the SIP/100 + channel to be propagated to a local channel so the dialplan + execution on the Local;2 channel would have the SIP/100 + accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 + Propagating the SIP/100 accountcode to the local channels is very + useful. Without any dialplan manipulation, all channels in this + call would have the same accountcode. Using dialplan, you can set + a different accountcode on the SIP/200 channel either by setting + the accountcode on the Local;2 channel or by the Dial + application's b(pre-dial), M(macro) or U(gosub) options, or by + the FollowMe application's b(pre-dial) option, or by the Queue + application's macro or gosub options. Before Asterisk v12, the + altered accountcode on SIP/200 will remain until the local + channels optimize out and the accountcode would change to the + SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount + support but ultimately had to punt on the support. The + peeraccount support was rendered useless because of how the CDR + code needed to unconditionally force the caller's accountcode + onto the peer channel's accountcode. The CEL events were thus + intentionally made to always use the channel's accountcode as the + peeraccount value. With the arrival of Asterisk v12, the + situation has improved somewhat so peeraccount support can be + made to work. Using the indicated example, the the accountcode + values become as follows when the peeraccount is set on SIP/100 + before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> + SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: + 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already + has an accountcode it can only change by the following explicit + user actions: 1) A channel originate method that can specify an + accountcode to use. 2) The calling channel propagating its + non-empty peeraccount or its non-empty accountcode if the + peeraccount was empty to the outgoing channel's accountcode + before initiating the dial. e.g., Dial and FollowMe. The + exception to this propagation method is Queue. Queue will only + propagate peeraccounts this way only if the outgoing channel does + not have an accountcode. 3) Dialplan using CHANNEL(accountcode). + 4) Dialplan using CHANNEL(peeraccount) on the other end of a + local channel pair. If a channel does not have an accountcode it + can get one from the following places: 1) The channel driver's + configuration at channel creation. 2) Explicit user action as + already indicated. 3) Entering a basic or stasis-mixing bridge + from a peer channel's peeraccount value. You can specify the + accountcode for an outgoing channel by setting the + CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue + applications. Queue adds the wrinkle that it will not overwrite + an existing accountcode on the outgoing channel with the calling + channels values. Accountcode and peeraccount values propagate to + an outgoing channel before dialing. Accountcodes also propagate + when channels enter or leave a basic or stasis-mixing bridge. The + peeraccount value only makes sense for mixing bridges with two + channels; it is meaningless otherwise. * Made peeraccount + functional by changing accountcode propagation as described + above. * Fixed CEL extracting the wrong ie value for the + peeraccount. This was done intentionally in Asterisk v1.8 when + that version had to punt on peeraccount. * Fixed a few places + dealing with accountcodes that were reading from channels without + the lock held. AFS-65 #close Review: + https://reviewboard.asterisk.org/r/3601/ + +2014-07-24 21:01 +0000 [r419504] Michael L. Young + + * main/db.c, include/asterisk/astdb.h: core/db: Revert Patch Added + In Attempt To Improve I/O Performance Reverting the patch since + it was causing a regression and after fixing the regression, + there were no performance gains. At least based on my method for + measurement. ASTERISK-24050 Review: + https://reviewboard.asterisk.org/r/3841/ + +2014-07-24 17:50 +0000 [r419438-419439] Corey Farrell + + * include/asterisk/astobj.h: Deprecate astobj.h This flags astobj.h + as deprecated, warns people to use astobj2.h instead. Only + netsock.c (also deprecated) still uses astobj.h. ASTERISK-24069 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3818/ + + * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: + complete upgrade to ao2 This change upgrades sip_registry and + sip_subscription_mwi to astobj2. ASTERISK-24067 #close Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3759/ + +2014-07-24 16:52 +0000 [r419377] Jason Parker + + * addons/chan_ooh323.c, /: Don't cause Asterisk to exit if + ooh323.conf not found. (closes issue ASTERISK-23814) ........ + Merged revisions 419374 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 419375 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419376 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-24 15:20 +0000 [r419358] Matthew Jordan + + * main/devicestate.c, channels/chan_pjsip.c: device state: Update + the core to report ONHOLD if a channel is on hold In Asterisk, it + is possible for a device to have a status of ONHOLD. This is not + typically an easy thing to determine, as a channel being on hold + is not a direct channel state. Typically, this has to be + calculated outside of the core independently in channel drivers, + notably, chan_sip and chan_pjsip. Both of these channel drivers + already have to calculate device state in a fashion more complex + than the core can handle, as they aggregate all state of all + channels associated with a peer/endpoint; they also independently + track whether or not one of those channels is currently on hold + and mark the device state appropriately. In 12+, we now have the + ability to report an AST_DEVICE_ONHOLD state for all channels + that defer their device state to the core. This is due to channel + hold state actually now being tracked on the channel itself. If a + channel driver defers its device state to the core (which many, + such as DAHDI, IAX2, and others do in most situations), the + device state core already goes out to get a channel associated + with the device. As such, it can now also factor the channel hold + state in its calculation. This patch adds this logic to the + device state core. It also uses an existing mapping between + device state and channel state to handle more channel states. + chan_pjsip has been updated slightly as well to make use of this + (as it was, for some reason, reporting a channel state of BUSY as + a device state of INUSE, which feels slightly wrong). Review: + https://reviewboard.asterisk.org/r/3771/ ASTERISK-24038 #close + +2014-07-24 13:00 +0000 [r419342] Kinsey Moore + + * include/asterisk/manager.h, doc/appdocsxml.dtd, main/xmldoc.c, + main/manager_bridges.c, main/manager.c, + include/asterisk/xmldoc.h, main/config_options.c: AMI: Allow for + command response documentation Allow for responses to AMI + actions/commands to be documented properly in XML and displayed + via the CLI. Response events are documented exactly as standard + AMI events are documented. Review: + https://reviewboard.asterisk.org/r/3812/ + +2014-07-23 16:46 +0000 [r419319] Matthew Jordan + + * main/endpoints.c, tests/test_stasis_endpoints.c, /: endpoints: + Fix failing unit tests from r419196 This patch does two things: + (1) It updates the unit tests to expect additional stasis + messages. More messages are now sent to the endpoint topic, due + to forwarding all channel messages and the forwarding + relationship set up between endpoints themselves. (2) Remove the + technology forwarding subscription during ast_endpoint_shutdown. + This prevents an improper double shutdown of an endpoint from + occurring. ........ Merged revisions 419318 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-23 14:00 +0000 [r419286] Scott Griepentrog + + * apps/app_voicemail.c, /: app_voicemail: use a consistent + generator string When updating voicemail.conf when a user changes + their pin, change the generator string to be the same as the + module name when reading so that the same config_hook will be + called. Review: https://reviewboard.asterisk.org/r/3837/ ........ + Merged revisions 419284 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419285 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-23 01:28 +0000 [r419268] Corey Farrell + + * main/manager.c, res/res_fax.c: res_fax: unregister manager + actions on unload * Unregister manager actions FAXSessions, + FAXSession and FAXStats at unload. * Update ast_manager_register2 + use ao2_t_alloc tagged with the action name. ASTERISK-24058 + #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3831/ + +2014-07-22 20:22 +0000 [r419222-419252] Michael L. Young + + * CHANGES, main/bridge_channel.c: core/bridge_channel: Substitute + Variables In Features Application Map Say you wanted to include + variables in an application map and have those variables + substituted and passed along to the application being executed; + currently this does not happen. This patch adds this ability to + pass channel variable values to an application before being + executed. ASTERISK-22608 #close Reported by: Michael L. Young + patches: features_substitute_arguments_v2.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3819/ + + * CHANGES, apps/app_mixmonitor.c: apps/app_mixmonitor: Add Options + To Play Beep At Start Or Stop We have a new periodic beep feature + but sometimes a user needs some sort of feedback, without the + need to have a periodic beep during the recording, to let them + know that MixMonitor started recording or ended the recording. + The use case where this patch is being used is when using Dynamic + Features to start and end MixMonitor. This patch adds an option + to play a beep when MixMonitor starts and an option to play a + beep when MixMonitor ends. ASTERISK-24051 #close Reported by: + Michael L. Young patches: mixmonitor-play-beep-start-stop.diff + uploaded by Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3820/ + + * main/db.c, include/asterisk/astdb.h: core/db: Improve I/O When + Updating Rows When updating a row, we are currently doing an + INSERT OR REPLACE INTO. The downside to this is that the row is + deleted if it exists and then a new row is inserted. So, we are + hitting the disk twice. One for the deletion and one for the + insertion. This patch changes this statement to an INSERT INTO + and if the insert fails because a row with that key exists, we + will IGNORE the failure. Then we will attempt to perform an + UPDATE on the existing row if that row wasn't just INSERTed. + ASTERISK-24050 #close Reported by: Michael L. Young patches: + astdb-insert-update-io-help_trunk_v2.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3815/ + +2014-07-22 17:10 +0000 [r419206] Richard Mudgett + + * codecs/codec_speex.c: codec_speex: Fix trashing normal static + frame for AST_FRAME_CNG. Made use a local static frame to + generate the AST_FRAME_CNG frame when silence starts. I don't + think the handling of the AST_FRAME_CNG has ever really worked + because there doesn't seem to be any consumers of it. Review: + https://reviewboard.asterisk.org/r/3813/ + +2014-07-22 16:20 +0000 [r419203] Matthew Jordan + + * include/asterisk/endpoints.h, + rest-api/api-docs/applications.json, include/asterisk/xmpp.h, + main/channel_internal_api.c, channels/chan_motif.c, + include/asterisk/channel.h, res/ari/resource_applications.h, + res/res_xmpp.c, channels/chan_iax2.c, main/endpoints.c, + channels/chan_pjsip.c, main/channel.c, + res/ari/resource_endpoints.c, /, channels/chan_sip.c: ARI: Fix + endpoint/channel subscription issues; allow for subscriptions to + tech This patch serves two purposes: (1) It fixes some bugs with + endpoint subscriptions not reporting all of the channel events + (2) It serves as the preliminary work needed for ASTERISK-23692, + which allows for sending/receiving arbitrary out of call text + messages through ARI in a technology agnostic fashion. The + messaging functionality described on ASTERISK-23692 requires two + things: (1) The ability to send/receive messages associated with + an endpoint. This is relatively straight forwards with the + endpoint core in Asterisk now. (2) The ability to send/receive + messages associated with a technology and an arbitrary technology + defined URI. This is less straight forward, as endpoints are + formed from a tech + resource pair. We don't have a mechanism to + note that a technology that *may* have endpoints exists. This + patch provides such a mechanism, and fixes a few bugs along the + way. The first major bug this patch fixes is the forwarding of + channel messages to their respective endpoints. Prior to this + patch, there were two problems: (1) Channel caching messages + weren't forwarded. Thus, the endpoints missed most of the + interesting bits (such as channel creation, destruction, state + changes, etc.) (2) Channels weren't associated with their + endpoint until after creation. This resulted in endpoints missing + the channel creation message, which limited the usefulness of the + subscription in the first place (a major use case being 'tell me + when this endpoint has a channel'). Unfortunately, this meant + another parameter to ast_channel_alloc. Since not all channel + technologies support an ast_endpoint, this patch makes such a + call optional and opts for a new function, + ast_channel_alloc_with_endpoint. When endpoints are created, they + will implicitly create a technology endpoint for their technology + (if one does not already exist). A technology endpoint is special + in that it has no state, cannot have channels created for it, + cannot be created explicitly, and cannot be destroyed except on + shutdown. It does, however, have all messages from other + endpoints in its technology forwarded to it. Combined with the + bug fixes, we now have Stasis messages being properly forwarded. + Consider the following scenario: two PJSIP endpoints (foo and + bar), where bar has a single channel associated with it and foo + has two channels associated with it. The messages would be + forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint + PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / + channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the + applications resource, can: - subscribe to endpoint:PJSIP/foo and + get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and + endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get + notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - + subscribe to endpoint:PJSIP and get notifications for channels + PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints + PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, + it never has events itself. It merely provides an aggregation + point for all other endpoints in its technology (which in turn + aggregate all channel messages associated with that endpoint). + This patch also adds endpoints to res_xmpp and chan_motif, + because the actual messaging work will need it (messaging without + XMPP is just sad). Review: + https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........ + Merged revisions 419196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-22 14:36 +0000 [r419180] Joshua Colp + + * channels/chan_iax2.c: chan_iax2: Restore previous behavior of + iax2_best_codec. The iax2_best_codec function was changed to + convert the formats into a format compatibilities structure and + grab the first format from it. The resulting order differs from + the previous order of iax2_best_codec which causes unexpected + formats to get chosen (such as g723). This commit brings back the + old behavior of iax2_best_codec by having a specified preference + list. Review: https://reviewboard.asterisk.org/r/3835/ + +2014-07-22 14:22 +0000 [r419110-419175] Kinsey Moore + + * addons/ooh323c/src/printHandler.c, tests/test_sorcery_realtime.c, + tests/test_json.c, addons/ooh323c/src/ooq931.c, + tests/test_astobj2_thrash.c, addons/chan_ooh323.c, /, + tests/test_optional_api.c, tests/test_abstract_jb.c, + apps/app_meetme.c, tests/test_logger.c, tests/test_event.c, + tests/test_hashtab_thrash.c, res/res_mwi_external_ami.c, + tests/test_sorcery.c, res/res_corosync.c, + tests/test_voicemail_api.c, tests/test_aoc.c, + tests/test_astobj2.c, tests/test_config.c: Fix more dev-mode + build issues ........ Merged revisions 419129 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 419162 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 419163 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/dial.c: Dial API: Prevent crash on NULL cap This prevents a + crash in the Dial API triggered by use of the Page() application + where a format capability struct was used before checking whether + it was NULL. ASTERISK-24074 #close + + * channels/chan_skinny.c, tests/test_core_format.c: Fix build in + dev-mode + +2014-07-21 16:26 +0000 [r419109] Jonathan Rose + + * channels/chan_iax2.c: chan_iax2: Restore codec choice behavior + from media formats branch After merging the media formats branch, + chan_iax2 was discarding codec preferences for the purpose of + choosing which codec a channel would use once a call started. + This patch restores the Asterisk 1.8-12 codec choice behaviors. + ASTERISK-23958 #close Review: + https://reviewboard.asterisk.org/r/3800/ + +2014-07-21 16:09 +0000 [r419093] Joshua Colp + + * channels/chan_iax2.c: chan_iax2: Only send mini frames if the + underlying format has not changed, not if it has. ASTERISK-24072 + #close Reported by: Matt Jordan + +2014-07-21 14:49 +0000 [r419077] Sean Bright + + * configure, configure.ac: Fix build when pjproject is installed in + a non-standard location. When configuring Asterisk to build + against a version of pjproject installed in a non-standard + location, the checks for "PJSIP Transaction Group Lock Support" + and "PJSIP Media Stream Replacement Support" fail. This is + because these secondary checks are not taking the CFLAGS and LIBS + returned by the pkg-config check into account. Review: + https://reviewboard.asterisk.org/r/3830 + +2014-07-21 08:41 +0000 [r419060] Corey Farrell + + * channels/sig_analog.c, res/res_smdi.c, channels/chan_motif.c, + include/asterisk/smdi.h, apps/app_voicemail.c, + channels/chan_dahdi.c: res_smdi: convert to astobj2 Remove + functions: ast_smdi_interface_unref ast_smdi_md_message_putback + ast_smdi_mwi_message_putback ast_smdi_md_message destructor + ast_smdi_mwi_message destructor Includes for astobj.h are removed + everywhere it's possible. ASTERISK-24066 #close Review: + https://reviewboard.asterisk.org/r/3758/ + +2014-07-20 22:06 +0000 [r419044] Matthew Jordan + + * apps/app_confbridge.c, res/ari/resource_channels.c, + include/asterisk/rtp_engine.h, include/asterisk/slinfactory.h, + res/res_calendar.c, codecs/codec_g722.c, + include/asterisk/res_pjsip_session.h, main/frame.c, + codecs/ex_lpc10.h, apps/app_dictate.c, res/res_fax.c, + apps/app_echo.c, include/asterisk/slin.h, codecs/codec_g726.c, + formats/format_ogg_vorbis.c, codecs/codec_gsm.c, + codecs/ex_alaw.h, formats/format_wav_gsm.c, + channels/iax2/provision.c, channels/chan_iax2.c, + res/res_format_attr_h264.c, main/data.c, main/manager.c, + include/asterisk/audiohook.h, formats/format_pcm.c, + main/config_options.c, res/res_format_attr_silk.c, + main/bridge_channel.c, res/res_speech.c, channels/chan_pjsip.c, + res/res_clioriginate.c, formats/format_g729.c, + channels/chan_unistim.c, res/res_rtp_asterisk.c, + include/asterisk/smoother.h (added), main/rtp_engine.c, + addons/format_mp3.c, formats/format_wav.c, + apps/confbridge/conf_chan_record.c, include/asterisk/speech.h, + codecs/ex_adpcm.h, channels/iax2/codec_pref.c (added), + include/asterisk/codec.h (added), formats/format_siren7.c, + include/asterisk/file.h, channels/chan_dahdi.c, + include/asterisk/image.h, funcs/func_channel.c, + main/abstract_jb.c, formats/format_h263.c, codecs/codec_dahdi.c, + main/dsp.c, apps/app_voicemail.c, apps/app_jack.c, + funcs/func_talkdetect.c, channels/chan_vpb.cc, + channels/chan_sip.c, formats/format_sln.c, + tests/test_abstract_jb.c, codecs/codec_alaw.c, UPGRADE.txt, + main/smoother.c (added), codecs/ex_speex.h, + channels/chan_console.c, apps/app_talkdetect.c, + main/format_pref.c (removed), main/indications.c, + include/asterisk/format_cap.h, main/media_index.c, + apps/app_agent_pool.c, res/res_pjsip_session.c, main/cli.c, + res/res_format_attr_celt.c, channels/chan_skinny.c, + tests/test_core_format.c (added), funcs/func_frame_trace.c, + res/res_pjsip/pjsip_configuration.c, main/file.c, + include/asterisk/frame.h, formats/format_g726.c, + apps/app_mixmonitor.c, channels/chan_mgcp.c, main/sorcery.c, + codecs/ex_ilbc.h, codecs/codec_lpc10.c, tests/test_format_cache.c + (added), apps/app_meetme.c, main/translate.c, + apps/app_originate.c, res/parking/parking_applications.c, + apps/app_ices.c, channels/iax2/parser.c, res/res_rtp_multicast.c, + pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_vox.c, + main/format_cap.c, tests/test_cel.c, include/asterisk/format.h, + formats/format_h264.c, apps/app_chanspy.c, apps/app_nbscat.c, + addons/chan_ooh323.c, bridges/bridge_holding.c, + channels/iax2/include/codec_pref.h (added), codecs/codec_adpcm.c, + apps/app_waitforsilence.c, res/res_pjsip_sdp_rtp.c, + addons/chan_ooh323.h, bridges/bridge_simple.c, + apps/app_alarmreceiver.c, bridges/bridge_softmix.c, + res/res_stasis_snoop.c, main/sounds_index.c, main/core_local.c, + main/codec_builtin.c (added), include/asterisk/format_cache.h + (added), apps/app_speech_utils.c, res/res_format_attr_opus.c, + include/asterisk/abstract_jb.h, main/channel.c, + include/asterisk/format_compatibility.h (added), apps/app_mp3.c, + tests/test_voicemail_api.c, channels/chan_alsa.c, main/app.c, + formats/format_g723.c, codecs/codec_ilbc.c, tests/test_config.c, + formats/format_gsm.c, apps/app_milliwatt.c, codecs/ex_ulaw.h, + main/asterisk.c, include/asterisk/res_pjsip.h, main/format.c, + main/ccss.c, main/bridge.c, codecs/codec_speex.c, + include/asterisk/format_pref.h (removed), apps/app_record.c, + main/slinfactory.c, res/res_adsi.c, main/core_unreal.c, + res/ari/resource_bridges.c, include/asterisk/callerid.h, + channels/pjsip/dialplan_functions.c, main/dial.c, + channels/dahdi/bridge_native_dahdi.c, main/format_cache.c + (added), include/asterisk/mod_format.h, apps/app_sms.c, + codecs/codec_resample.c, main/format_compatibility.c (added), + main/audiohook.c, formats/format_jpeg.c, res/res_stasis.c, + formats/format_g719.c, include/asterisk/translate.h, + funcs/func_speex.c, codecs/codec_a_mu.c, + channels/iax2/format_compatibility.c (added), + apps/app_festival.c, main/channel_internal_api.c, + tests/test_format_api.c (removed), codecs/ex_g722.h, + main/utils.c, res/ari/resource_sounds.c, + res/res_format_attr_h263.c, codecs/ex_g726.h, + include/asterisk/_private.h, channels/chan_oss.c, + channels/chan_misdn.c, main/codec.c (added), main/callerid.c, + addons/ooh323cDriver.c, apps/app_amd.c, codecs/codec_ulaw.c, + main/image.c, channels/chan_nbs.c, bridges/bridge_native_rtp.c, + channels/iax2/include/format_compatibility.h (added), + formats/format_siren14.c, res/res_fax_spandsp.c, + addons/chan_mobile.c, addons/ooh323cDriver.h, + channels/sip/include/sip.h, tests/test_format_cap.c (added), + channels/chan_multicast_rtp.c, include/asterisk/vector.h, + channels/chan_bridge_media.c, apps/app_fax.c, + main/bridge_basic.c, apps/app_test.c, include/asterisk/channel.h, + include/asterisk/data.h, tests/test_core_codec.c (added), + res/res_musiconhold.c, codecs/ex_gsm.h, formats/format_ilbc.c, + include/asterisk/config_options.h, channels/chan_phone.c, + include/asterisk/bridge_channel.h, apps/app_dumpchan.c, + channels/chan_motif.c, res/res_agi.c: media formats: re-architect + handling of media for performance improvements In the old times + media formats were represented using a bit field. This was fast + but had a few limitations. 1. Asterisk was limited in how many + formats it could handle. 2. Formats, being a bit field, could not + include any attribute information. A format was strictly its + type, e.g., "this is ulaw". This was changed in Asterisk 10 (see + https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal + for notes on that work) which led to the creation of the + ast_format structure. This structure allowed Asterisk to handle + attributes and bundle information with a format. Additionally, + ast_format_cap was created to act as a container for multiple + formats that, together, formed the capability of some entity. + Another mechanism was added to allow logic to be registered which + performed format attribute negotiation. Everywhere throughout the + codebase Asterisk was changed to use this strategy. + Unfortunately, in software, there is no free lunch. These new + capabilities came at a cost. Performance analysis and profiling + showed that we spend an inordinate amount of time comparing, + copying, and generally manipulating formats and their related + structures. Basic prototyping has shown that a reasonably large + performance improvement could be made in this area. This patch is + the result of that project, which overhauled the media format + architecture and its usage in Asterisk to improve performance. + Generally, the new philosophy for handling formats is as follows: + * The ast_format structure is reference counted. This removed a + large amount of the memory allocations and copying that was done + in prior versions. * In order to prevent race conditions while + keeping things performant, the ast_format structure is immutable + by convention and lock-free. Violate this tenet at your peril! * + Because formats are reference counted, codecs are also reference + counted. The Asterisk core generally provides built-in codecs and + caches the ast_format structures created to represent them. + Generally, to prevent inordinate amounts of module reference + bumping, codecs and formats can be added at run-time but cannot + be removed. * All compatibility with the bit field representation + of codecs/formats has been moved to a compatibility API. The + primary user of this representation is chan_iax2, which must + continue to maintain its bit-field usage of formats for + interoperability concerns. * When a format is negotiated with + attributes, or when a format cannot be represented by one of the + cached formats, a new format object is created or cloned from an + existing format. That format may have the same codec underlying + it, but is a different format than a version of the format with + different attributes or without attributes. * While formats are + reference counted objects, the reference count maintained on the + format should be manipulated with care. Formats are generally + cached and will persist for the lifetime of Asterisk and do not + explicitly need to have their lifetime modified. An exception to + this is when the user of a format does not know where the format + came from *and* the user may outlive the provider of the format. + This occurs, for example, when a format is read from a channel: + the channel may have a format with attributes (hence, non-cached) + and the user of the format may last longer than the channel (if + the reference to the channel is released prior to the format's + reference). For more information on this work, see the API design + notes: + https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite + Finally, this work was the culmination of a large number of + developer's efforts. Extra thanks goes to Corey Farrell, who took + on a large amount of the work in the Asterisk core, chan_sip, and + was an invaluable resource in peer reviews throughout this + project. There were a substantial number of patches contributed + during this work; the following issues/patch names simply reflect + some of the work (and will cause the release scripts to give + attribution to the individuals who work on them). Reviews: + https://reviewboard.asterisk.org/r/3814 + https://reviewboard.asterisk.org/r/3808 + https://reviewboard.asterisk.org/r/3805 + https://reviewboard.asterisk.org/r/3803 + https://reviewboard.asterisk.org/r/3801 + https://reviewboard.asterisk.org/r/3798 + https://reviewboard.asterisk.org/r/3800 + https://reviewboard.asterisk.org/r/3794 + https://reviewboard.asterisk.org/r/3793 + https://reviewboard.asterisk.org/r/3792 + https://reviewboard.asterisk.org/r/3791 + https://reviewboard.asterisk.org/r/3790 + https://reviewboard.asterisk.org/r/3789 + https://reviewboard.asterisk.org/r/3788 + https://reviewboard.asterisk.org/r/3787 + https://reviewboard.asterisk.org/r/3786 + https://reviewboard.asterisk.org/r/3784 + https://reviewboard.asterisk.org/r/3783 + https://reviewboard.asterisk.org/r/3778 + https://reviewboard.asterisk.org/r/3774 + https://reviewboard.asterisk.org/r/3775 + https://reviewboard.asterisk.org/r/3772 + https://reviewboard.asterisk.org/r/3761 + https://reviewboard.asterisk.org/r/3754 + https://reviewboard.asterisk.org/r/3753 + https://reviewboard.asterisk.org/r/3751 + https://reviewboard.asterisk.org/r/3750 + https://reviewboard.asterisk.org/r/3748 + https://reviewboard.asterisk.org/r/3747 + https://reviewboard.asterisk.org/r/3746 + https://reviewboard.asterisk.org/r/3742 + https://reviewboard.asterisk.org/r/3740 + https://reviewboard.asterisk.org/r/3739 + https://reviewboard.asterisk.org/r/3738 + https://reviewboard.asterisk.org/r/3737 + https://reviewboard.asterisk.org/r/3736 + https://reviewboard.asterisk.org/r/3734 + https://reviewboard.asterisk.org/r/3722 + https://reviewboard.asterisk.org/r/3713 + https://reviewboard.asterisk.org/r/3703 + https://reviewboard.asterisk.org/r/3689 + https://reviewboard.asterisk.org/r/3687 + https://reviewboard.asterisk.org/r/3674 + https://reviewboard.asterisk.org/r/3671 + https://reviewboard.asterisk.org/r/3667 + https://reviewboard.asterisk.org/r/3665 + https://reviewboard.asterisk.org/r/3625 + https://reviewboard.asterisk.org/r/3602 + https://reviewboard.asterisk.org/r/3519 + https://reviewboard.asterisk.org/r/3518 + https://reviewboard.asterisk.org/r/3516 + https://reviewboard.asterisk.org/r/3515 + https://reviewboard.asterisk.org/r/3512 + https://reviewboard.asterisk.org/r/3506 + https://reviewboard.asterisk.org/r/3413 + https://reviewboard.asterisk.org/r/3410 + https://reviewboard.asterisk.org/r/3387 + https://reviewboard.asterisk.org/r/3388 + https://reviewboard.asterisk.org/r/3389 + https://reviewboard.asterisk.org/r/3390 + https://reviewboard.asterisk.org/r/3321 + https://reviewboard.asterisk.org/r/3320 + https://reviewboard.asterisk.org/r/3319 + https://reviewboard.asterisk.org/r/3318 + https://reviewboard.asterisk.org/r/3266 + https://reviewboard.asterisk.org/r/3265 + https://reviewboard.asterisk.org/r/3234 + https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close + Reported by: mjordan media_formats_translation_core.diff uploaded + by kharwell (License 6464) rb3506.diff uploaded by mjordan + (License 6283) media_format_app_file.diff uploaded by kharwell + (License 6464) misc-2.diff uploaded by file (License 5000) + chan_mild-3.diff uploaded by file (License 5000) + chan_obscure.diff uploaded by file (License 5000) jingle.diff + uploaded by file (License 5000) funcs.diff uploaded by file + (License 5000) formats.diff uploaded by file (License 5000) + core.diff uploaded by file (License 5000) bridges.diff uploaded + by file (License 5000) mf-codecs-2.diff uploaded by file (License + 5000) mf-app_fax.diff uploaded by file (License 5000) + mf-apps-3.diff uploaded by file (License 5000) + media-formats-3.diff uploaded by file (License 5000) + ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License + 5909) rb3689.patch uploaded by mjordan (License 6283) + ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) + mf-attributes-3.diff uploaded by file (License 5000) + ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by + coreyfarrell (License 5909) rb3800.patch uploaded by jrose + (License 6182) chan_sip.diff uploaded by mjordan (License 6283) + rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 + #close Tested by: sgriepentrog, mjordan, coreyfarrell + sip_cleanup.diff uploaded by opticron (License 6273) + chan_sip_caps.diff uploaded by mjordan (License 6283) + rb3751.patch uploaded by coreyfarrell (License 5909) + chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 + #close Tested by: opticron direct_media.diff uploaded by opticron + (License 6273) pjsip-direct-media.diff uploaded by file (License + 5000) format_cap_remove.diff uploaded by opticron (License 6273) + media_format_fixes.diff uploaded by opticron (License 6273) + chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 + #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti + (License 5621) chan_dahdi.diff uploaded by file (License 5000) + ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, + file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by + rmudgett (License 5621) moh_cleanup.diff uploaded by opticron + (License 6273) bridge_leak.diff uploaded by opticron (License + 6273) translate.diff uploaded by file (License 5000) rb3795.patch + uploaded by rmudgett (License 5621) tls_fix.diff uploaded by + mjordan (License 6283) fax-mf-fix-2.diff uploaded by file + (License 5000) rtp_transfer_stuff uploaded by mjordan (License + 6283) rb3787.patch uploaded by rmudgett (License 5621) + media-formats-explicit-translate-format-3.diff uploaded by file + (License 5000) format_cache_case_fix.diff uploaded by opticron + (License 6273) rb3774.patch uploaded by rmudgett (License 5621) + rb3775.patch uploaded by rmudgett (License 5621) + rtp_engine_fix.diff uploaded by opticron (License 6273) + rtp_crash_fix.diff uploaded by opticron (License 6273) + rb3753.patch uploaded by mjordan (License 6283) rb3750.patch + uploaded by mjordan (License 6283) rb3748.patch uploaded by + rmudgett (License 5621) media_format_fixes.diff uploaded by + opticron (License 6273) rb3740.patch uploaded by mjordan (License + 6283) rb3739.patch uploaded by mjordan (License 6283) + rb3734.patch uploaded by mjordan (License 6283) rb3689.patch + uploaded by mjordan (License 6283) rb3674.patch uploaded by + coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell + (License 5909) rb3667.patch uploaded by coreyfarrell (License + 5909) rb3665.patch uploaded by mjordan (License 6283) + rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch + uploaded by coreyfarrell (License 5909) + format_compatibility-2.diff uploaded by file (License 5000) + core.diff uploaded by file (License 5000) + +2014-07-18 21:48 +0000 [r419022] Matthew Jordan + + * rest-api/api-docs/recordings.json, res/ari/resource_recordings.c, + res/stasis_recording/stored.c, res/res_ari_recordings.c, /, + include/asterisk/stasis_app_recording.h, + res/ari/resource_recordings.h, CHANGES: ari: Add a copy operation + for stored recordings This patch adds a new operation for stored + recordings, copy. It takes an existing stored recording and makes + a copy of it in the same directory or a relative directory under + the stored recording directory. + /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name} + This is particularly useful for voicemail-esque applications, + which may need to copy or move recordings around a directory + structure. Review: https://reviewboard.asterisk.org/r/3768/ + ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam + Galarneau ........ Merged revisions 419021 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 21:25 +0000 [r418997-419020] Corey Farrell + + * main/stasis_message_router.c, /: stasis: fix call to ao2_t_alloc + for stasis_message_router_create This fixes a build failure + introduced by r3821. struct stasis_topic is opaque, so + topic->name is unavailable. Switch to using stasis_topic_name(). + ........ Merged revisions 419019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis.c, main/stasis_cache_pattern.c, + main/stasis_message.c, main/stasis_message_router.c, /: stasis: + use ao2_t_alloc for certain object allocators Add tags to stasis + objects using the name. This makes it easier to track the source + of certain stasis ref leaks. Review: + https://reviewboard.asterisk.org/r/3821/ ........ Merged + revisions 418996 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 19:07 +0000 [r418980] Kinsey Moore + + * res/res_fax_spandsp.c: Fix build in dev-mode + +2014-07-18 17:55 +0000 [r418961-418963] Scott Griepentrog + + * res/res_pjsip_pubsub.c, main/astobj2.c, + include/asterisk/astobj2.h, main/logger.c, main/utils.c: astobj2: + assert on invalid ref and backtrace cleanup If a reference count + goes negative, instead of just logging that fact, be more helpful + with a backtrace and an assert that will DO_CRASH. This patch + also removes the duplicate ao2_bt() function and cleans up + extraneous usage of the ast_log_backtrace() call. Review: + https://reviewboard.asterisk.org/r/3765/ + + * /, channels/chan_sip.c: media formats: fix ref leak of peer for + mwi subscription Holding a reference to the peer during mwi + subscriptions resulted in a circular reference because the final + event message would not be sent until destruction of the peer. + Instead, pass the name of the peer to the event callback so that + it can fail gracefully after the peer has gone. ASTERISK-23959 + Review: https://reviewboard.asterisk.org/r/3754/ ........ Merged + revisions 418636 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/features_config.c: feature_config: insure featuregroups + and applicationmaps are initialized If the features.conf is + missing, the cfg->featurgroups and cfg->applicationmaps is not + initialized, resulting in assert on ao2_find of a null container. + This patch changes the initialization call and adds asserts for a + safeguard. Review: https://reviewboard.asterisk.org/r/3809/ + ........ Merged revisions 418886 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 16:47 +0000 [r418938] Richard Mudgett + + * funcs/func_audiohookinherit.c, /: func_audiohookinherit.c: Fixup + some XML documentation wording. ........ Merged revisions 418937 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-18 16:28 +0000 [r418911-418936] Jonathan Rose + + * main/channel.c, funcs/func_audiohookinherit.c, /, + include/asterisk/audiohook.h, main/framehook.c, res/res_fax.c, + main/bridge_basic.c, include/asterisk/res_fax.h, + bridges/bridge_native_rtp.c, main/audiohook.c, CHANGES, + include/asterisk/framehook.h, res/res_pjsip_refer.c: Channels: + Masquerades to automatically move frame/audio hooks Whenever + possible, audiohooks and framehooks will now be copied over to + the channel that the masquerading channel gets cloned into. This + should occur for all audiohooks and most framehooks. As a result, + in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now + deprecated and its behavior is essentially the new default for + all audiohooks, plus some additional audiohooks/framehooks. + Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged + revisions 418914 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_fax.c, include/asterisk/res_fax.h, CHANGES, + res/res_fax.exports.in, res/res_fax_spandsp.c: res_fax: Provide + AMI equivalents for fax CLI commands Specifically the following + equivalents were created: fax show session -> FAXSession fax show + sessions -> FAXSessions fax show stats -> FAXStats Review: + https://reviewboard.asterisk.org/r/3666/ + +2014-07-18 00:11 +0000 [r418893-418895] Sean Bright + + * config.sub, menuselect/config.guess, menuselect/config.sub, + config.guess: Update config.guess and config.sub + + * autoconf/ast_ext_tool_check.m4: Add missing file from previous + commit. + + * menuselect/aclocal.m4, menuselect/configure, + menuselect/acinclude.m4 (removed), menuselect/bootstrap.sh, + menuselect/autoconfig.h.in: Import Asterisk's autoconf magic + instead of using our own. + +2014-07-17 21:17 +0000 [r418832-418870] Matthew Jordan + + * configs/samples/acl.conf.sample (added), + configs/samples/extensions.conf.sample (added), + configs/res_parking.conf.sample (removed), + configs/samples/cel_sqlite3_custom.conf.sample (added), + configs/cdr_sqlite3_custom.conf.sample (removed), + configs/modules.conf.sample (removed), + configs/samples/cli_aliases.conf.sample (added), + configs/meetme.conf.sample (removed), + configs/cdr_pgsql.conf.sample (removed), + configs/samples/extensions.ael.sample (added), + configs/samples/cdr_adaptive_odbc.conf.sample (added), + configs/samples/motif.conf.sample (added), + configs/samples/extensions_minivm.conf.sample (added), + configs/samples/res_curl.conf.sample (added), + configs/res_config_sqlite3.conf.sample (removed), + configs/mgcp.conf.sample (removed), configs/dsp.conf.sample + (removed), configs/udptl.conf.sample (removed), + configs/sip.conf.sample (removed), configs/dbsep.conf.sample + (removed), configs/queuerules.conf.sample (removed), + configs/samples/cdr_mysql.conf.sample (added), + configs/confbridge.conf.sample (removed), + configs/samples/cdr_odbc.conf.sample (added), + configs/samples/minivm.conf.sample (added), + configs/enum.conf.sample (removed), + configs/samples/codecs.conf.sample (added), + configs/samples/chan_dahdi.conf.sample (added), + configs/samples/cdr_custom.conf.sample (added), + configs/samples/res_config_mysql.conf.sample (added), + configs/samples/dundi.conf.sample (added), + configs/samples/oss.conf.sample (added), + configs/samples/app_mysql.conf.sample (added), + configs/samples/queues.conf.sample (added), + configs/samples/cdr.conf.sample (added), + configs/samples/cdr_syslog.conf.sample (added), + configs/festival.conf.sample (removed), + configs/samples/cel_pgsql.conf.sample (added), + configs/http.conf.sample (removed), configs/phoneprov.conf.sample + (removed), configs/alarmreceiver.conf.sample (removed), + configs/samples/features.conf.sample (added), + configs/cdr_tds.conf.sample (removed), + configs/func_odbc.conf.sample (removed), + configs/samples/logger.conf.sample (added), + configs/samples/res_odbc.conf.sample (added), + configs/samples/agents.conf.sample (added), + configs/res_fax.conf.sample (removed), + configs/samples/xmpp.conf.sample (added), + configs/iaxprov.conf.sample (removed), + configs/res_pgsql.conf.sample (removed), + configs/extensions.conf.sample (removed), + configs/chan_mobile.conf.sample (removed), configs/asterisk.adsi + (removed), configs/cel_sqlite3_custom.conf.sample (removed), + configs/users.conf.sample (removed), + configs/samples/res_pktccops.conf.sample (added), + configs/samples/amd.conf.sample (added), configs/rtp.conf.sample + (removed), configs/samples/res_parking.conf.sample (added), + configs/hep.conf.sample (removed), + configs/samples/modules.conf.sample (added), + configs/cel_tds.conf.sample (removed), + configs/res_curl.conf.sample (removed), + configs/samples/skinny.conf.sample (added), + configs/samples/cdr_pgsql.conf.sample (added), + configs/samples/sip_notify.conf.sample (added), + configs/samples/test_sorcery.conf.sample (added), + configs/samples/dsp.conf.sample (added), + configs/ss7.timers.sample (removed), + configs/samples/udptl.conf.sample (added), + configs/cdr_odbc.conf.sample (removed), + configs/samples/sip.conf.sample (added), + configs/minivm.conf.sample (removed), + configs/res_config_sqlite.conf.sample (removed), + configs/codecs.conf.sample (removed), configs/osp.conf.sample + (removed), configs/samples/cel_custom.conf.sample (added), + configs/samples/dbsep.conf.sample (added), + configs/samples/app_skel.conf.sample (added), + configs/console.conf.sample (removed), + configs/cdr_manager.conf.sample (removed), + configs/cdr_custom.conf.sample (removed), + configs/chan_dahdi.conf.sample (removed), + configs/res_config_mysql.conf.sample (removed), + configs/samples/statsd.conf.sample (added), + configs/cli.conf.sample (removed), configs/queues.conf.sample + (removed), configs/cdr_syslog.conf.sample (removed), UPGRADE.txt, + configs/manager.conf.sample (removed), + configs/samples/res_corosync.conf.sample (added), + configs/features.conf.sample (removed), configs/sla.conf.sample + (removed), configs/logger.conf.sample (removed), + configs/res_odbc.conf.sample (removed), + configs/agents.conf.sample (removed), + configs/samples/ooh323.conf.sample (added), Makefile, + configs/xmpp.conf.sample (removed), + configs/samples/phoneprov.conf.sample (added), + configs/samples/alarmreceiver.conf.sample (added), + configs/samples/cdr_tds.conf.sample (added), + configs/extconfig.conf.sample (removed), + configs/samples/func_odbc.conf.sample (added), + configs/samples/res_fax.conf.sample (added), + configs/samples/iaxprov.conf.sample (added), + configs/samples/res_ldap.conf.sample (added), + configs/samples/dnsmgr.conf.sample (added), + configs/res_pktccops.conf.sample (removed), + configs/cel.conf.sample (removed), + configs/samples/res_pgsql.conf.sample (added), + configs/samples/chan_mobile.conf.sample (added), + configs/samples/asterisk.adsi (added), + configs/samples/users.conf.sample (added), + configs/samples/rtp.conf.sample (added), + configs/phone.conf.sample (removed), configs/skinny.conf.sample + (removed), configs/muted.conf.sample (removed), + configs/samples/hep.conf.sample (added), configs/iax.conf.sample + (removed), configs/samples/cel_tds.conf.sample (added), + configs/sip_notify.conf.sample (removed), + configs/samples/telcordia-1.adsi (added), + configs/samples/alsa.conf.sample (added), + configs/samples/adsi.conf.sample (added), + configs/test_sorcery.conf.sample (removed), + configs/samples/followme.conf.sample (added), + configs/samples/asterisk.conf.sample (added), + configs/extensions.lua.sample (removed), configs/say.conf.sample + (removed), configs/cel_custom.conf.sample (removed), + configs/samples/ss7.timers.sample (added), + configs/samples/cel_odbc.conf.sample (added), + configs/app_skel.conf.sample (removed), + configs/samples/ccss.conf.sample (added), + configs/cli_permissions.conf.sample (removed), + configs/statsd.conf.sample (removed), + configs/samples/res_config_sqlite.conf.sample (added), + configs/config_test.conf.sample (removed), + configs/indications.conf.sample (removed), + configs/samples/osp.conf.sample (added), + configs/samples/cdr_manager.conf.sample (added), + configs/samples/console.conf.sample (added), + configs/voicemail.conf.sample (removed), + configs/res_corosync.conf.sample (removed), + configs/misdn.conf.sample (removed), + configs/samples/cli.conf.sample (added), configs/ari.conf.sample + (removed), configs/ooh323.conf.sample (removed), + configs/samples/calendar.conf.sample (added), + configs/samples/res_stun_monitor.conf.sample (added), + configs/samples/manager.conf.sample (added), + configs/samples/pjsip_notify.conf.sample (added), + configs/samples/sla.conf.sample (added), + configs/musiconhold.conf.sample (removed), + configs/pjsip.conf.sample (removed), configs/sorcery.conf.sample + (removed), configs/vpb.conf.sample (removed), + configs/unistim.conf.sample (removed), + configs/res_ldap.conf.sample (removed), + configs/dnsmgr.conf.sample (removed), + configs/samples/extconfig.conf.sample (added), + configs/samples/res_snmp.conf.sample (added), + configs/acl.conf.sample (removed), + configs/samples/smdi.conf.sample (added), + configs/samples/cel.conf.sample (added), + configs/cli_aliases.conf.sample (removed), + configs/samples/cdr_sqlite3_custom.conf.sample (added), + configs/extensions.ael.sample (removed), + configs/cdr_adaptive_odbc.conf.sample (removed), + configs/samples/phone.conf.sample (added), + configs/extensions_minivm.conf.sample (removed), + configs/motif.conf.sample (removed), configs/telcordia-1.adsi + (removed), configs/samples/meetme.conf.sample (added), + configs/adsi.conf.sample (removed), configs/alsa.conf.sample + (removed), configs/samples/muted.conf.sample (added), + configs/followme.conf.sample (removed), + configs/asterisk.conf.sample (removed), + configs/samples/iax.conf.sample (added), + configs/samples/res_config_sqlite3.conf.sample (added), + configs/samples/mgcp.conf.sample (added), + configs/cel_odbc.conf.sample (removed), configs/ccss.conf.sample + (removed), configs/cdr_mysql.conf.sample (removed), + configs/samples/extensions.lua.sample (added), + configs/samples/say.conf.sample (added), + configs/dundi.conf.sample (removed), + configs/samples/queuerules.conf.sample (added), + configs/oss.conf.sample (removed), configs/app_mysql.conf.sample + (removed), configs/samples/confbridge.conf.sample (added), + configs/samples/cli_permissions.conf.sample (added), + configs/samples/enum.conf.sample (added), + configs/samples/config_test.conf.sample (added), + configs/cdr.conf.sample (removed), + configs/samples/indications.conf.sample (added), + configs/cel_pgsql.conf.sample (removed), + configs/res_stun_monitor.conf.sample (removed), + configs/calendar.conf.sample (removed), + configs/samples/voicemail.conf.sample (added), + configs/pjsip_notify.conf.sample (removed), + configs/samples/misdn.conf.sample (added), + configs/samples/ari.conf.sample (added), + configs/samples/festival.conf.sample (added), + configs/samples/http.conf.sample (added), + configs/res_snmp.conf.sample (removed), + configs/samples/musiconhold.conf.sample (added), + configs/samples/pjsip.conf.sample (added), + configs/samples/sorcery.conf.sample (added), + configs/samples/vpb.conf.sample (added), configs/smdi.conf.sample + (removed), configs/samples/unistim.conf.sample (added), + configs/samples (added), configs/amd.conf.sample (removed): + configs: Move sample config files into a subdirectory of configs + This moves all samples configs from configs/ to configs/samples. + This allows for additional sets of sample configuration files to + be added in the future. Review: + https://reviewboard.asterisk.org/r/3804/ + + * channels/chan_sip.c, UPGRADE.txt: chan_sip: Make + progressinband=never really mean 'never' progressinband=never in + sip.conf is easily defeated if an onward trunk sends a progress + indication of its own. This is almost certain to happen if the + onward trunk is ISDN or IAX as these technologies send a progress + indication even if early media is not required. This progress + message is passed to the caller, and causes the "never" option to + be rather badly named. This patch changes the behaviour of this + setting in the following ways: 1) In sip_write(), do not pass the + media unless we have either progressed beyond INV_EARLY_MEDIA, or + we are in INV_EARLY_MEDIA state, and early media is both set-up + and wanted. This helps resolve double-ringing on some buggy + handsets. 2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, + but SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to + avoid implicitly enabling early media. Avoid sending double ring + indications. NOTE: the meaning of the SIP_PROGRESS_SENT flag + changes slightly in this patch to also encapsulate the fact that + a channel has *sent or received* a 183 Progress indication. This + makes the updated code in sip_write() much more simple. Review: + https://reviewboard.asterisk.org/r/3700 ASTERISK-23972 #close + Reported by: Steve Davies patches: + inband_never_present_early_media2 uploaded by Steve Davies + (License 5012) + + * menuselect: Add svn:ignore property + + * UPGRADE.txt, menuselect/configure, menuselect/configure.ac, + configure, configure.ac: configure: Fix libxml2 development + library dependency checking The commit that added libxml2 support + didn't fully check for the libxml2 development script in the + Asterisk configure file. As a result, Asterisk could be + configured, then fail on menuselect. This patch fixes it so that + Asterisk should detect the libxml2 dependency failure first. + + * menuselect/makeopts.in, menuselect/autoconfig.h.in, + menuselect/menuselect.h, menuselect/example_menuselect-tree, + configure, include/asterisk/autoconfig.h.in, menuselect/Makefile, + menuselect/README, menuselect/aclocal.m4, configure.ac, + UPGRADE.txt, menuselect/configure, menuselect/configure.ac, + menuselect/menuselect.c, menuselect/acinclude.m4: menuselect: Add + libxml2 support (Patch 3) This is the final patch in adding + menuselect to Asterisk. - The first patch (r418832) added + menuselect along with mxml - The second patch (r418833) removed + mxml from menuselect This patch adds support for libxml2 to + menuselect, and makes libxml2 a required library for Asterisk. + Note that the libxml2 portion of this patch was written by Sean + Bright, and was made available on a team branch: + http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/ + Review: https://reviewboard.asterisk.org/r/3773/ ASTERISK-20703 + #close patches: some_mysterious_team_branch uploaded by + seanbright (License 5060) + + * menuselect/mxml (removed): menuselect: Remove mxml from + menuselect (Patch 2) This is the second patch that adds + menuselect to Asterisk trunk. The previous commit (r418832) added + menuselect along with mxml; this patch removes mxml completely + from Menuselect. A subsequent patch will switch menuselect over + to using libxml2, and make libxml2 a required dependency for + Asterisk. ASTERISK-20703 + + * menuselect/mxml/configure.in (added), menuselect/acinclude.m4 + (added), menuselect/mxml/mxml.list.in (added), + menuselect/mxml/README (added), menuselect/linkedlists.h (added), + menuselect/mxml (added), menuselect/mxml/config.h.in (added), + menuselect/aclocal.m4 (added), menuselect/install-sh (added), + menuselect/mxml/mxml-string.c (added), + menuselect/menuselect_stub.c (added), menuselect/make_version + (added), menuselect/mxml/mxml-entity.c (added), + menuselect/bootstrap.sh (added), menuselect/makeopts.in (added), + menuselect/autoconfig.h.in (added), menuselect/config.guess + (added), menuselect/mxml/install-sh (added), + menuselect/test/build_tools/menuselect-deps (added), /, + menuselect/contrib/menuselect-dummy (added), + menuselect/config.sub (added), menuselect/mxml/configure (added), + menuselect/mxml/Makefile.in (added), menuselect (added), + menuselect/contrib (added), menuselect/mxml/mxml.pc.in (added), + menuselect/configure.ac (added), menuselect/mxml/mxml-set.c + (added), menuselect/contrib/Makefile-dummy (added), + menuselect/mxml/ANNOUNCEMENT (added), menuselect/missing (added), + menuselect/menuselect_curses.c (added), + menuselect/example_menuselect-tree (added), menuselect/Makefile + (added), menuselect/mxml/mxml-search.c (added), menuselect/test + (added), menuselect/test/menuselect-tree (added), + menuselect/mxml/mxml.h (added), menuselect/mxml/mxml-index.c + (added), menuselect/configure (added), + menuselect/menuselect_newt.c (added), menuselect/mxml/mxml-attr.c + (added), menuselect/mxml/mxml-private.c (added), + menuselect/menuselect.c (added), menuselect/mxml/CHANGES (added), + menuselect/mxml/COPYING (added), menuselect/mxml/mxml-file.c + (added), menuselect/menuselect.h (added), + menuselect/menuselect_gtk.c (added), menuselect/README (added), + menuselect/strcompat.c (added), menuselect/mxml/mxml-node.c + (added), menuselect/test/build_tools (added): menuselect: Add + menuselect to Asterisk trunk (Patch 1) This is the first patch + that adds menuselect to Asterisk trunk, and removes the + svn:externals property. This is being done for two reasons: (1) + The removal of external repositories eases a future migration to + git (2) Asterisk is now the only thing that uses menuselect; as a + result, there's little need to keep it in an external repository + Subsequent patches will remove the mxml dependency from + menuselect and tidy up the build system. ASTERISK-20703 + +2014-07-17 14:28 +0000 [r418811] Kinsey Moore + + * /, main/bridge_channel.c: TEST_FRAMEWORK: Fix threewaytransfer + reporting Ensure that three-way transfers can be reported even if + featuremap is non-NULL. ........ Merged revisions 418810 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-16 23:08 +0000 [r418788] Corey Farrell + + * /, channels/dahdi/bridge_native_dahdi.c: Remove include of + astobj.h from channels/dahdi/bridge_native_dahdi.c. The include + was unneeded, this is split off from r3758 as it applies to 12. + ........ Merged revisions 418787 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-16 14:03 +0000 [r418717-418757] Matthew Jordan + + * res/res_pjsip/pjsip_configuration.c, CHANGES, res/res_pjsip.c, + channels/chan_pjsip.c, include/asterisk/res_pjsip.h, + contrib/ast-db-manage/config/versions/1d50859ed02e_create_accountcode.py + (added), /, configs/pjsip.conf.sample: res_pjsip: Support setting + a default accountcode on endpoints Most channel drivers let you + specify a default accountcode to be set on channels associated + with a particular peer/endpoint/object. Prior to this patch, + chan_pjsip/res_pjsip did not support such a setting. This patch + adds a new setting to the res_pjsip endpoint object, + 'accountcode'. When a channel is created that is associated with + an endpoint with this value set, the channel will automatically + have its accountcode property set to the value configured for the + endpoint. Review: https://reviewboard.asterisk.org/r/3724/ + ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged + revisions 418756 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * cdr/cdr_pgsql.c, CHANGES, configs/cdr_pgsql.conf.sample, + configs/res_pgsql.conf.sample, cel/cel_pgsql.c, + res/res_config_pgsql.c, configs/cel_pgsql.conf.sample: cel_pgsql, + cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name + support This patch adds support for the PostgreSQL + application_name connection setting. When the appropriate + PostgreSQL module's configuration is set with an application + name, the name will be passed to PostgreSQL on connection and + displayed in the database's pg_stat_activity view, as well as in + CSV logs. This aids in managing which applications/servers are + connected to a PostgreSQL database, as well as tracing the + activity of those connections. Review: + https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close + Reported by: Gergely Domodi patches: pgsql_application_name.patch + uploaded by Gergely Domodi (License 6610) + + * codecs/codec_adpcm.c, main/format.c: codec_adpcm: Change + description of codec "ADPCM" to "Dialogic ADPCM" Technically, + ADPCM is a method that can be applied to several codecs. + Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec. See + http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information + about said codec. Review: https://reviewboard.asterisk.org/r/3744 + patches: rb3744.patch uploaded by dennis.guse (License 6513) + + * UPGRADE.txt, main/manager.c, /: manager: Return ActionID on + nominal responses to PresenceState action When the PresenceState + action is executed, the nominal path fails to include the + ActionID in the successful response. This patch adds a call to + astman_start_ack, which guarantees that an ActionID (if provided) + will be sent back to the AMI client. Unlike the Asterisk 11 and + 12 patches, this patch also deprecates the duplicate Message key + in the response to the action, replacing it with the key + 'PresenceMessage'. Review: + https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close + ........ Merged revisions 418713 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418714 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-15 23:03 +0000 [r418716] Kinsey Moore + + * /, main/bridge_channel.c: TEST_FRAMEWORK: Fix ref leak in feature + activation This fixes two reference leaks that would occur when + TEST_FRAMEWORK was enabled and features were successfully + executed. ........ Merged revisions 418715 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-15 17:57 +0000 [r418654] Jonathan Rose + + * funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty + strings as argument Previously these two dialplan functions would + issue warnings and return failure when an empty string is used as + the argument. Now they will not issue a warning and will + successfully return an empty string. ASTERISK-23911 #close + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3745/ ........ Merged + revisions 418641 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 418649 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-15 12:11 +0000 [r418616] Sean Bright + + * main/asterisk.c: Update Asterisk copyright year in + main/asterisk.c It's been 2014 for like... 6 months. + +2014-07-14 14:55 +0000 [r418566-418587] Richard Mudgett + + * include/asterisk/logger.h, /: logger.h: Extract DEBUG_ATLEAST() + to complement VERBOSITY_ATLEAST(). ........ Merged revisions + 418586 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/jabber.h (removed), include/asterisk/jingle.h + (removed), include/asterisk/frame_defs.h (removed), + configs/h323.conf.sample (removed): Actually delete the removed + files. + +2014-07-13 21:57 +0000 [r418507] Corey Farrell + + * /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work + around REF_DEBUG race which causes out of order log entries * + Update refcounter.py to use delta's to track the current + reference count. * Use result from internal_ao2_ref to write + old_refcount to refs_log. Review: + https://reviewboard.asterisk.org/r/3756/ ........ Merged + revisions 418504 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 418505 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418506 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-13 20:08 +0000 [r418488] Scott Griepentrog + + * include/asterisk/astobj2.h: astobj2: correct define for + ao2_t_cleanup This change maps the ao2_t_cleanup() function to + the correct debug function so that it can be used. Review: + https://reviewboard.asterisk.org/r/3764/ + +2014-07-13 16:48 +0000 [r418448-418467] Corey Farrell + + * main/manager.c, /, apps/app_skel.c: Fix minor reference leaks in + app_skel and TEST_FRAMEWORK * Cleanup games object in app_skel. * + Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+). + Review: https://reviewboard.asterisk.org/r/3757/ ........ Merged + revisions 418465 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418466 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/jabber.h, include/asterisk/jingle.h, + configs/h323.conf.sample: Remove files left behind on removal of + h323, jingle and jabber. This change removes h323.conf.sample, + jingle.h, jabber.h left behind by r3698. Review: + https://reviewboard.asterisk.org/r/3755/ + +2014-07-11 23:00 +0000 [r418419] Matthew Jordan + + * main/astobj2.c, include/asterisk/astobj2.h: astobj2: Add tag + variants for ao2_bump, ao2_cleanup, and ao2_replace Tags are + useful in hunting down ref imbalances; this patch adds tag + variants for these commonly used macros/functions. Review: + https://reviewboard.asterisk.org/r/3750/ + +2014-07-11 21:10 +0000 [r418397] Corey Farrell + + * /, include/asterisk/astobj2.h: astobj2: tweak ao2_replace to do + nothing when it would be a NoOp This change causes ao2_replace to + do nothing when src == dst. This avoids REF_DEBUG logging when + we're not actually doing anything. Review: + https://reviewboard.asterisk.org/r/3743/ ........ Merged + revisions 418396 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-11 16:42 +0000 [r418370] Scott Griepentrog + + * /, main/config.c: config: inform config hook of change when + writing file When updated configuration is written back to the + conf file - for example when a user changes their voicemail pin, + make sure that any config hook that wants to know of changes is + informed. Review: https://reviewboard.asterisk.org/r/3708/ + ........ Merged revisions 418366 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418369 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-10 15:36 +0000 [r418325] Matthew Jordan + + * /, include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert + indentation to tabs This is a whitespace only change. ........ + Merged revisions 418323 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-10 01:59 +0000 [r418226-418264] Richard Mudgett + + * channels/sig_pri.c, /: chan_dahdi/sig_pri: Fix type mismatch in + the idledial feature's channel creation. Square pegs in round + holes don't work very well. ........ Merged revisions 418261 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 418262 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 418263 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/stasis/stasis_bridge.h (added), main/bridge_channel.c, + res/res_stasis.c, /, res/stasis/stasis_bridge.c (added), + include/asterisk/bridge_channel.h, main/bridge_basic.c: ARI: Make + mixing bridges propagate linkedids and accountcodes. * Create a + Stasis bridge sub-class to propagate linkedids and accountcodes. + * Fixed the basic bridge sub-class to update peeraccount codes + when the number of channels in the bridge drops back down to two + parties. * Refactored ast_bridge_channel_update_accountcodes() to + handle channels joining/leaving the bridge. * Fixed the basic + bridge sub-class to not call the base bridge class pull method + twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard + Mudgett Review: https://reviewboard.asterisk.org/r/3720/ ........ + Merged revisions 418225 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-08 14:48 +0000 [r418174-418183] Matthew Jordan + + * rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json: manager/ARI: Update version to + 2.4.0/1.4.0; Update UPGRADE.txt ........ Merged revisions 418182 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix undefined + function when PJPROJECT is not installed The + dtls_perform_handshake function was mistakenly placed under the + guards for USE_PJPROJECT. If PJPROJECT was not installed, the + function would not be defined, while other functions would + attempt to still use it. This prevented res_rtp_asterisk from + being loaded. ASTERISK-24001 #close Reported by: Don Fanning + ........ Merged revisions 418172 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-07 16:08 +0000 [r418117] Joshua Colp + + * include/asterisk/res_pjsip_body_generator_types.h, + res/res_pjsip_dialog_info_body_generator.c (added), + res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c, /, + include/asterisk/res_pjsip_presence_xml.h: + res_pjsip_dialog_info_body_generator: Add dialog-info+xml support + for presence. This module implements dialog-info+xml for the + purposes of presence. This means that phones such as Grandstreams + can now subscribe to receive presence information for an + extension. ASTERISK-21443 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3705/ ........ Merged + revisions 418116 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-07 02:15 +0000 [r418090] Matthew Jordan + + * include/asterisk/stasis_app.h, res/ari/resource_channels.c, + res/res_stasis.c, /, res/stasis/app.c: ARI/res_stasis: Subscribe + to both Local channel halves when originating to app This patch + fixes two bugs: 1. When originating a channel into a Stasis + application, we already create a subscription for the channel + that is going into our Stasis app. Unfortunately, when you create + a Local channel and pass it off to a Stasis app, you really + aren't creating just one channel: you're creating two. This patch + snags the second half of the Local channel pair (assuming it is a + Local channel pair, but luckily core_local is kind about such + assumptions) and subscribes to it as well. 2. Subscriptions are a + bit sticky right now. If a subscription is made, the 'interest' + count gets bumped on the Stasis subscription - but unless + something explicitly unsubscribes the channel, said subscription + sticks around. This is not much of a problem is a user is + creating the subscription - if they made it, they must want it. + However, when we are creating implicit subscriptions, we need to + make sure something clears them out. This patch takes a + pessimistic approach: it watches the cache updates coming from + Stasis and, if we notice that the cache just cleared out an + object, we delete our subscription object. This keeps our ao2 + container of Stasis forwards in an application from growing out + of hand; it also is a bit more forgiving for end users who may + not realize they were supposed to unsubscribe from that channel + that just hung up. Review: + https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close + ........ Merged revisions 418089 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-07 01:22 +0000 [r418067-418084] Kinsey Moore + + * tests/test_cel.c, main/cel.c, channels/chan_pjsip.c, + res/res_pjsip_session.c, /: CEL: Fix incorrect/missing extra + field information This corrects two issues with the extra field + information in Asterisk 12+ in channel event logs. It is possible + to inject custom values into the dialstatus provided by + ast_channel_dial_type() Stasis messages that fall outside the + enumeration allowed for the DIALSTATUS channel variable. CEL now + filters for the allowed values and ignores other values. The + "hangupsource" extra field key is always blank if the far end + channel is a chan_pjsip channel. This is because the hangupsource + is never set for the pjsip channel driver. This change sets the + hangupsource whenever a hangup is queued for chan_pjsip channels. + This corrects an issue with the pjsip channel driver where the + hangupcause information was not being set properly. Review: + https://reviewboard.asterisk.org/r/3690/ ........ Merged + revisions 418071 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/http.c: HTTP: Fix build for gcc 4.10 ........ Merged + revisions 418066 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-04 15:26 +0000 [r418019-418050] Matthew Jordan + + * main/Makefile: main/Makefile: fix compilation error of buildinfo + occurring on 'make install' Egads. Another bad deletion of too + much when attempting to remove h323 stuff. + + * configure.ac, build_tools/menuselect-deps.in, configure, + main/Makefile: configure: Remove last vestiges of h323; DO create + menuselect-deps The previous patch (r418034) fixed the 'glitch' + that the channels/h323 Makefile no longer existed. Unfortunately, + removing the entire line was a bit of a blunder, as it meant that + build_tools/menuselect-deps was never generated. Hilarity ensued + when actually trying to compile. But hey! At least configure + worked. This patch fixes *that* glitch, and removes some more of + the vestiges of h323. (It had tendrils in the main Makefile? + Crazy.) + + * configure.ac, configure: configure: Update script to pass if + channels/h323/Makefile.in does not exist This simply removes that + check from the configure script, as r418019 removed chan_h323. + + * apps/app_dahdibarge.c (removed), configs/gtalk.conf.sample + (removed), main/pbx.c, apps/app_readfile.c (removed), + channels/chan_sip.c, configs/jingle.conf.sample (removed), + UPGRADE.txt, res/res_musiconhold.c, channels/chan_gtalk.c + (removed), channels/Makefile, CHANGES, res/res_jabber.c + (removed), channels/h323 (removed), utils/conf2ael.c, + channels/chan_jingle.c (removed), res/ael/pval.c, + configs/jabber.conf.sample (removed), + configs/asterisk.conf.sample, res/res_agi.c, channels/chan_h323.c + (removed), addons/Makefile, pbx/pbx_realtime.c, utils/ael_main.c, + include/asterisk/options.h, main/asterisk.c, + addons/app_saycountpl.c (removed): Remove many deprecated modules + Billing records are fair, To get paid is quite bright, You should + really use ODBC; Good-bye cdr_sqlite. Microsoft did once push + H.323, Hell, we all remember NetMeeting. But try to compile + chan_h323 now And you will take quite a beating. The XMPP and SIP + war was fierce, And in the distant fray Was birthed + res_jabber/chan_jingle; But neither to stay. For everyone did + care and chase what Google professed. "Free Internet Calling" was + what devotees cried, But Google did change the specs so often + That the developers were happy the day chan_gtalk died. And then + there was that odd application Dedicated to the Polish tongue. + app_saycountpl was subsumed by Say; One could say its bell was + rung. To read and parse a file from the dialplan You could (I + guess) use an application. app_readfile did fill that purpose, + but I think A function is perhaps better in its creation. Barging + is rude, I'm not sure why we do it. Inwardly, the caller will + probably sigh. But if you really must do it, Don't use + app_dahdibarge, use ChanSpy. We all despise the sound of tinny + robots It makes our queues so cold. To control such an + abomination It's better to not use Wait/SetMusicOnHold. It's + often nice to know properties of a channel It makes our calls + right We have a nice function called CHANNEL And so SIPCHANINFO + is sent off into the night. And now things get odd; Apparently + one could delimit with a colon Properties from the SIPPEER + function! Commas are in; all others are done. Finally, a word on + pipes and commas. We're sorry. We can't say it enough. But those + compatibility options in asterisk.conf; To maintain them forever + was just too tough. This patch removes: * cdr_sqlite * chan_gtalk + * chan_jingle * chan_h323 * res_jabber * app_saycountpl * + app_readfile * app_dahdibarge It removes the following + applications/functions: * WaitMusicOnHold * SetMusicOnHold * + SIPCHANINFO It removes the colon delimiter from the SIPPEER + function. Finally, it also removes all compatibility options that + were configurable from asterisk.conf, as these all applied to + compatibility with Asterisk 1.4 systems. Review: + https://reviewboard.asterisk.org/r/3698/ + +2014-07-03 22:22 +0000 [r417933-417976] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, /, UPGRADE.txt, + channels/sig_pri.c: chan_dahdi: Add inband_on_setup_ack + compatibility option. The new inband_on_setup_ack option causes + Asterisk to assume inband audio may be present when a + SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says + that in scenarios with overlap dialing, when a dialtone is sent + from the network side, progress indicator 8 "Inband info now + available" MAY be sent to the CPE if no digits were received with + the SETUP. It is thus implied that the ie is mandatory if digits + came with the SETUP and dialtone is needed. This option should be + enabled, when the network sends dialtone and you want to hear it, + but the network doesn't send the progress indicator when needed. + NOTE: For Q.SIG setups this option should be enabled when + outgoing overlap dialing is also enabled because Q.SIG does not + send the progress indicator with the SETUP ACK. The commit + -r413714 (AST-1338) which causes this issue was dealing with a + SIP-to-ISDN interoperability issue. This commit is a merge of the + two patches indicated below. ASTERISK-23897 #close Reported by: + Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded + by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) + patch uploaded by rmudgett Review: + https://reviewboard.asterisk.org/r/3633/ ........ Merged + revisions 417956 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417957 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417958 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_channels.c, res/res_ari.c, main/manager.c, /: + res_ari: Fix some off-nominal paths just dropping the HTTP + connection. * Removed some incorrect newlines on ast_http_error() + messages in manager.c. * Removed an incorrect newline in + res_ari_channels.c. Addendum to ASTERISK-23552 ........ Merged + revisions 417932 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-03 17:34 +0000 [r417910-417916] Jonathan Rose + + * CHANGES, channels/chan_dahdi.c: chan_dahdi: Add AMI commands for + controlling PRI debugging output Adds the following AMI commands: + PRIDebugSet - Set PRI debug levels for a specific span + PRIDebugFileSet - Set the file used for PRI debug message output + PRIDebugFileUnset - Disables file output for PRI debug messages + Review: https://reviewboard.asterisk.org/r/3681/ + + * CHANGES, pbx/pbx_config.c, main/pbx.c: pbx_config: Add manager + actions to add/remove extensions Adds two new manager commands to + pbx_config - DialplanExtensionAdd and DialplanExtensionRemove + which allow manager users to create and delete extensions + respectively. Review: https://reviewboard.asterisk.org/r/3650/ + +2014-07-03 17:16 +0000 [r417901] Richard Mudgett + + * res/res_phoneprov.c, main/http.c, UPGRADE.txt, + include/asterisk/tcptls.h, res/res_http_post.c, + res/res_http_websocket.c, configs/http.conf.sample, + include/asterisk/http.h, main/tcptls.c, res/res_ari.c, + main/manager.c, /: HTTP: Add persistent connection support. + Persistent HTTP connection support is needed due to the increased + usage of the Asterisk core HTTP transport and the frequency at + which REST API calls are going to be issued. * Add http.conf + session_keep_alive option to enable persistent connections. * + Parse and discard optional chunked body extension information and + trailing request headers. * Increased the maximum + application/json and application/x-www-form-urlencoded body size + allowed to 4k. The previous 1k was kind of small. * Removed a + couple inlined versions of ast_http_manid_from_vars() by calling + the function. manager.c:generic_http_callback() and + res_http_post.c:http_post_callback() * Add missing va_end() in + ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use + in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott + Griepentrog Review: https://reviewboard.asterisk.org/r/3691/ + ........ Merged revisions 417880 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-03 16:55 +0000 [r417900] Matthew Jordan + + * main/tcptls.c, configure, include/asterisk/autoconfig.h.in, + configure.ac: main/tcptls: Add checks for OpenSSL Elliptic Curve + support The patch for ASTERISK-23905 that added PFS support in + Asterisk depends on the elliptic curve library support being + present in OpenSSL. As it turns out, some versions of OpenSSL + don't have this library - notably the version running on our + build agents. This patch fixes the build by providing a configure + check for the specific library calls that the PFS patch relies + on. Review: https://reviewboard.asterisk.org/r/3709/ + +2014-07-03 16:14 +0000 [r417877-417879] sgalarneau : + + * res/ari/resource_events.h, rest-api/api-docs/channels.json, + res/ari/resource_channels.h, rest-api/api-docs/events.json, /: + ARI: Improvements to body parameters documentation The variables + body parameter under the originate and originate with id + operations of the channel resource showed invalid JSON in its + description. The variables body parameter under the userEvent + operation of the event resource made no mention that the custom + key/value pairs should be wrapped in a variables key in order to + be added to the custom user event. ASTERISK-23975 #close Review: + https://reviewboard.asterisk.org/r/3692/ ........ Merged + revisions 417878 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api-templates/api.wiki.mustache, + rest-api-templates/swagger_model.py, /: api.wiki.mustache: Update + wiki template to support body parameters This patch updates the + api.wiki.mustache template and the swagger_model python script to + understand if an operation has a body parameter. If an operation + does have a body parameter, it will now be displayed in the + corresponding wiki entry. ........ Merged revisions 407389 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-03 14:08 +0000 [r417863] Tzafrir Cohen + + * Makefile, contrib/scripts/dahdi_span_config_hook (added): + dahdi_span_config_hook: automatically register new dahdi channels + Install a hook script for DAHDI to register new spans with + Asterisk automatically by running: asterisk -rx 'dahdi create + channel FIRST LAST' Review: + https://reviewboard.asterisk.org/r/3157/ + +2014-07-03 12:10 +0000 [r417800-417803] Matthew Jordan + + * main/tcptls.c, CHANGES: main/tcptls: Add support for Perfect + Forward Secrecy This patch enables Perfect Forward Secrecy (PFS) + in Asterisk's core TLS API. Modules that wish to enable PFS + should consider the following: - Ephemeral ECDH (ECDHE) is + enabled by default. To disable it, do not specify a ECDHE cipher + suite in a module's configuration, for example: + tlscipher=AES128-SHA:DES-CBC3-SHA - Ephemeral DH (DHE) is + disabled by default. To enable it, add DH parameters into the + private key file, i.e., tlsprivatekey. For an example, see the + default dh2048.pem at + http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt + - Because clients expect the server to prefer PFS, and because + OpenSSL sorts its cipher suites by bit strength, (see "openssl + ciphers -v DEFAULT") consider re-ordering your cipher suites in + the conf file. For example: + tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH + will use PFS when offered by the client. Clients which do not + offer PFS fall-back to AES-128 (or even 3DES as recommend by RFC + 3261). Review: https://reviewboard.asterisk.org/r/3647/ + ASTERISK-23905 #close Reported by: Alexander Traud patches: + tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520) + tlsPFS.patch uploaded by Alexander Traud (License 6520) + + * /, main/utils.c: main/untils: Prevent potential infinite loop in + ast_careful_fwrite A loop in ast_careful_fwrite exists that will + continually attempt to write to a file stream, even in the + presence of EAGAIN/EINTR errors. However, if a connection that + uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's + call to fflush may return EAGAIN/EINTER along with EOF. A + subsequent call to fflush will return EOF but not clear errno, + resulting in an infinite loop. This patch clears errno after it + is detected and handled the loop, such that any subsequent call + to fflush will not get erroneously stuck. Review: + https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close + Reported by: Steve Davies patches: fflush_loop_fix uploaded by + one47 (License 5012) ........ Merged revisions 417797 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417798 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417799 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-02 21:13 +0000 [r417770] Jonathan Rose + + * res/ari/resource_events.h, res/ari/resource_asterisk.h, + res/ari/resource_applications.h, res/ari/resource_playbacks.h, + res/ari/resource_channels.h, res/ari/resource_sounds.h, /, + res/ari/resource_bridges.h, res/ari/resource_recordings.h, + rest-api-templates/ari_resource.h.mustache, + res/ari/resource_device_states.h, res/ari/resource_endpoints.h, + res/ari/resource_mailboxes.h: ARI: Remove unnecessary \briefs + from automatically generated documentation Review: + https://reviewboard.asterisk.org/r/3440/ ........ Merged + revisions 412653 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-07-01 14:42 +0000 [r417679-417706] Joshua Colp + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Don't leak memory or + reset state if DTLS configuration is set multiple times. ........ + Merged revisions 417705 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, + contrib/ast-db-manage/config/versions/51f8cb66540e_add_further_dtls_options.py + (added), include/asterisk/res_pjsip_session.h, main/rtp_engine.c, + /, channels/chan_sip.c, main/sdp_srtp.c, res/res_pjsip_sdp_rtp.c, + res/res_pjsip/pjsip_configuration.c, configs/sip.conf.sample, + include/asterisk/rtp_engine.h, res/res_pjsip.c, + channels/sip/include/sip.h, include/asterisk/res_pjsip.h, + include/asterisk/sdp_srtp.h: Recorded merge of revisions 417677 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS + negotiation on RTCP. This change fixes up DTLS support in + res_rtp_asterisk so it can accept and provide a SHA-256 + fingerprint, so it occurs on RTCP, and so it occurs after ICE + negotiation completes. Configuration options to chan_sip and + chan_pjsip have also been added to allow behavior to be tweaked + (such as forcing the AVP type media transports in SDP). + ASTERISK-22961 #close Reported by: Jay Jideliov Review: + https://reviewboard.asterisk.org/r/3679/ Review: + https://reviewboard.asterisk.org/r/3686/ ........ Merged + revisions 417678 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-30 18:39 +0000 [r417663] Mark Michelson + + * res/res_pjsip_pubsub.c: Reverse logic during subscription + persistence recreation. In the abstraction effort, this bit of + logic got messed up. We want to recreate the persistence if + things go well, not if things fail. + +2014-06-30 13:02 +0000 [r417590-417649] Matthew Jordan + + * apps/app_voicemail.c: apps/app_voicemail: Fix compilation error + introduced in r417591 Not sure why that change to + ast_channel_alloc was made but ... okay. + + * apps/app_voicemail.c, main/say.c, CHANGES: app_voicemail, say: + Add support for Japanese Language This patch adds support for the + Japanese language to both the say family of applications, as well + as for VoiceMail and VoiceMailMain. A new pack of language sounds + will be released at the same time as the next major version of + Asterisk to support the new language features. The language + features can be enabled using a language code of 'ja'. Review: + https://reviewboard.asterisk.org/r/3477 ASTERISK-23324 #close + Reported by: Kevin McCoy patches: + app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy + (License 6586) say.c.20140226.jb.patch uploaded by Kevin McCoy + (License 6586) + + * /, channels/chan_sip.c: chan_sip: be more tolerant of whitespace + between attributes in SDP fmtp line This patch is essentially a + backport of a small portion of r397526 from ASTERISK-21981. In + that patch, pass through support and format attribute negotiation + was added for Opus. Part of that included being more tolerant to + whitespace in the fmtp line of an SDP; that part of the patch is + being applied here. As the author of the backport pointed out, in + SDP, the fmtp line is allowed to include whitespace between + attributes. RFC 3267 chapter 8.3 (from 2001) includes an example + for this. This was not removed in the updated RFC 4867 in 2007. + Review: https://reviewboard.asterisk.org/r/3658 #ASTERISK-23916 + #close Reported by: Alexander Traud patches: + sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud + (License 6520) ........ Merged revisions 417587 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417588 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417589 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 23:21 +0000 [r417571] Richard Mudgett + + * /, main/event.c: event.c: Fix type mismatch errors in ie_maps[]. + In v12+ the type values from the table are only used by the CEL + unit tests. Since the unit tests were only comparing a generated + expected event with a real event to see if the ie contents + matched and using the same table IE_PLTYPE values to read the + event contents, the type mismatches were not detected. ........ + Merged revisions 417565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 19:27 +0000 [r417485-417511] Corey Farrell + + * /, main/astobj2.c: Ensure REF_DEBUG records entrys for attempts + to ao2_ref an invalid object This change ensures that + __ao2_ref_debug writes to ref_log when given a non-NULL pointer + to an invalid ao2 object. This is to ensure that we record any + attempt manipulate references of already freed objects. + ASTERISK-23948 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3677/ ........ Merged + revisions 417500 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417505 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417509 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, contrib/scripts/refcounter.py: refcounter.py: prevent use of + excessive RAM with large refs logs When processing a 212MB refs + file, refcounter.py used over 3GB of RAM. This change greatly + reduces memory usage in two ways: * Saving object history in + whole lines instead of separated values. * Not saving + normal/skewed/leaked object lists unless they are requested. + ASTERISK-23921 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3668/ ........ Merged + revisions 417480 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417481 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417483 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 13:50 +0000 [r417461] Matthew Jordan + + * res/res_pjsip/pjsip_configuration.c, res/res_pjsip_pubsub.c, + res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h, /, + res/res_pjsip_outbound_registration.c: res_pjsip: Add ActionID to + events created as a result of PJSIP AMI actions A number of + various PJSIP AMI actions were failing to parse out and place the + ActionID into their responses. This patch updates the various + PJSIP actions such that the passed in ActionID is emitted on any + event list complete events, as well as any intermediate events + created as a result of the action. #ASTERISK-23947 #close + Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3675/ ........ Merged + revisions 417460 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-27 02:04 +0000 [r417423-417447] Kinsey Moore + + * tests/test_cel.c: CEL: Update unit tests for bridge tech field + Update the CEL unit tests that handle BRIDGE_ENTER and + BRIDGE_EXIT events to expect the "bridge_technology" extra field + key. + + * CHANGES: CHANGES: Add missing changes Add missing CHANGES changes + from r417361 and r417383. + +2014-06-26 18:27 +0000 [r417400-417421] Matthew Jordan + + * res/res_http_websocket.exports.in, /: res_http_websocket: Export + symbol for ast_websocket_set_timeout Thanks to Sean Bright for + pointing out that this was missed in #asterisk-dev. ........ + Merged revisions 417419 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417420 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_pjsip.c, /: chan_pjsip: Add a test event for fast + picture updates This will drive the test on review r3419. Note + that the patch for this was done by Ben Ford, although it was + slightly modified for this commit. ASTERISK-23562 Reported by: + Matt Jordan ........ Merged revisions 417399 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-26 14:48 +0000 [r417361-417383] Kinsey Moore + + * main/cel.c: CEL: Add bridge tech to relevant CEL records Add the + "bridge_technology" extra field key to BRIDGE_ENTER and + BRIDGE_EXIT CEL events to convey the bridge technology in use at + the time the record was generated. + + * main/bridge.c, include/asterisk/channel.h, + include/asterisk/bridge_features.h, + tests/test_channel_feature_hooks.c (added), + main/bridge_channel.c, main/channel.c: Bridging: Allow channels + to define bridging hooks This patch allows the current owner of a + channel to define various feature hooks to be made available once + the channel has entered a bridge. This includes any hooks that + are setup on the ast_bridge_features struct such as DTMF hooks, + bridge event hooks (join, leave, etc.), and interval hooks. + Review: https://reviewboard.asterisk.org/r/3649/ + +2014-06-26 12:43 +0000 [r417317-417360] Matthew Jordan + + * CHANGES, apps/app_jack.c: app_jack: Support audio with a sampling + rate higher than 8kHz This patch enables the jack-audiohook to + cope with dynamic sampling rates from and to Asterisk. + Information from the channel is taken to derive the channel's + sampling rate, suiting SLINxx format and frame->datalen. There + are stil a few limitations after this patch: * Required + information is taken from the channel during initialization as + the audiohook does not provide this information. + Audiohook.internal_sampl_rate(...) is set later, but no callback + is available to inform app_jack. * Frame.datalen is computed + using "rate / 50" assuming a ptime of 20ms. There is no internal + API available to determine datalen for a SLINxx. * Ringbuffer + size is now dynamic depending on the value of frame.datalen (see + above) and the number of frames, which are in + RINGBUFFER_FRAME_CAPACITY, that need to fit. Review: + https://reviewboard.asterisk.org/r/3618 Note that the patch being + committed here is based on the patch posted on ASTERISK-23836. + However, Matthis Schmieder also provided a patch to enable this + functionality, and that patch is noted below. ASTERISK-20696 + #close Reported by: Matthis Schmieder patches: app_jack.patch + uploaded by Matthis Schmieder (License 6445) ASTERISK-23836 + #close Reported by: Dennis Guse patches: patch-app_jack.c + uploaded by Dennis Guse (License 6513) + + * main/udptl.c, /: udptl: Correct FEC to not consider negative + sequence numbers as missing When using FEC, with span=3 and + entries=4 Asterisk will attempt to repair the packet with + sequence number 5, as it will see that packet -4 is missing. The + result is Asterisk sending garbage packets that can kill a fax. + This patch adds a check to see if the sequence number is valid + before checking if the packet is missing. Review: + https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close + Reported by: Torrey Searle patches: udptl_fec.patch uploaded by + Torrey Searle (License 5334) ........ Merged revisions 417318 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 417320 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/internal.h, configs/ari.conf.sample, + res/res_http_websocket.c, res/res_pjsip.c, + configs/pjsip.conf.sample, include/asterisk/http_websocket.h, + configs/sip.conf.sample, res/res_pjsip/config_transport.c, + res/ari/ari_websockets.c, res/res_pjsip_transport_websocket.c, + res/ari/config.c, channels/sip/include/sip.h, + include/asterisk/res_pjsip.h, res/res_ari.c, /, + channels/chan_sip.c, UPGRADE.txt: res_http_websocket: Close + websocket correctly and use careful fwrite When a client takes a + long time to process information received from Asterisk, a write + operation using fwrite may fail to write all information. This + causes the underlying file stream to be in an unknown state, such + that the socket must be disconnected. Unfortunately, there are + two problems with this in Asterisk's existing websocket code: 1. + Periodically, during the read loop, Asterisk must write to the + connected websocket to respond to pings. As such, Asterisk + maintains a reference to the session during the loop. When + ast_http_websocket_write fails, it may cause the session to + decrement its ref count, but this in and of itself does not break + the read loop. The read loop's write, on the other hand, does not + break the loop if it fails. This causes the socket to get in a + 'stuck' state, preventing the client from reconnecting to the + server. 2. More importantly, however, is that the fwrite in + ast_http_websocket_write fails with a large volume of data when + the client takes awhile to process the information. When it does + fail, it fails writing only a portion of the bytes. With some + debugging, it was shown that this was failing in a similar + fashion to ASTERISK-12767. Switching this over to + ast_careful_fwrite with a long enough timeout solved the problem. + Note that this version of the patch, unlike r417310 in Asterisk + 11, exposes configuration options beyond just chan_sip's + sip.conf. Configuration options to configure the write timeout + have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 + #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3624/ ........ Merged + revisions 417310 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-26 10:06 +0000 [r417251] Corey Farrell + + * /, channels/chan_sip.c: chan_sip: Fix handling of "From" headers + longer than 256 characters From headers were processed using a + 256 character buffer on the stack. This change replaces that with + a heap allocation by ast_strdup. ASTERISK-23790 #close Reported + by: uniken1 Tested by: uniken1 Review: + https://reviewboard.asterisk.org/r/3669/ Patches: + chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes + (license 5674) ........ Merged revisions 417248 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 417249 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 417250 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-25 20:57 +0000 [r417233] Mark Michelson + + * res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pidf_body_generator.c, + res/res_pjsip_pubsub.exports.in, res/res_pjsip_mwi.c, + res/res_pjsip_xpidf_body_generator.c: Abstract PJSIP-specific + elements from the pubsub API. This helps to pave the way for RLS + work that is to come. Since this is a self-contained change and + subscription tests still pass, this work is being committed + directly to trunk instead of a working branch. ASTERISK-23865 + #close Review: https://reviewboard.asterisk.org/r/3628 + +2014-06-25 18:57 +0000 [r417213] Corey Farrell + + * main/astobj2_container.c, /: ao2_container node object ignores + REF_DEBUG in all places except one Almost every reference + operation against container node's uses __ao2_alloc or __ao2_ref, + thereby preventing ref logging for the nodes. One node reference + is released with ao2_t_ref, causing refcounter.py to falsely + report skews and leaks for many nodes. ASTERISK-23922 #close + Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3670/ ........ Merged + revisions 417212 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-25 00:45 +0000 [r417193] Damien Wedhorn + + * channels/chan_skinny.c: Skinny: cleanup some log messages around + sessions. + +2014-06-24 02:50 +0000 [r417167] Corey Farrell + + * include/asterisk/netsock.h, main/utils.c, main/netsock.c, + include/asterisk/res_pjsip_session.h: Move eid functions to + utils.c, mark netsock.h deprecated Move eid functions from + netsock.c to utils.c. These functions were already published by + utils.h. Flag netsock.h as deprecated and switch + res_pjsip_session.h to use netsock2.h. The only code that still + uses netsock.h is chan_iax2. ASTERISK-23920 #close Reported by: + Corey Farrell Review: https://reviewboard.asterisk.org/r/3661/ + +2014-06-23 18:50 +0000 [r417143] Joshua Colp + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Return the length of + data written when sending via ICE instead of 0. ASTERISK-23834 + #close Reported by: Richard Kenner ........ Merged revisions + 417141 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 417142 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-23 16:04 +0000 [r417120] Richard Mudgett + + * /, main/core_unreal.c: core_unreal: Fix off by one buffer + overwrite error. Appending the ;2 to the user supplied ;1 + uniqueid to create the ;2 version if the user did not also supply + an extra uniqueid for the ;2 channel resulted in allocating a + buffer that was one byte too small. * Fix off by one error in + ast_unreal_new_channels() when generating the ;2 uniqueid from + the user suppled ;1 version. * Pulled some long assignment lines + from if tests to improve line break readability in + ast_unreal_new_channels(). ........ Merged revisions 417119 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-23 07:44 +0000 [r417059] Tzafrir Cohen + + * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c: + suspended destructions of pri spans on events If a DAHDI span + disappears, we wish for its representation in Asterisk to be + destroyed as well. The information about the span's removal may + come from several paths: 1. DAHDI sends DAHDI_EVENT_REMOVE on + every channel. 2. An extra DAHDI_EVENT_REMOVED is sent on every + subsequent call to DAHDI_GET_EVENT. 3. Every read (including the + internal one by libpri on the D-channel) returns -ENODEV. + Asterisk responsds to DAHDI_EVENT_REMOVE on a channel by + destroying it. Destroying a channel requires holding the channel + list lock (iflock). Destroying a channel that is part of a span + requires holding the span's lock. Destroying a channel from a + context that holds the span lock, while at the same time another + channel is destroyed directly, leads to a deadlock. Solution: + don't destroy span while holding the channels list lock. Thus + changes in this patch: * Deferring removal of PRI spans in + response to events: doomed spans are collected on a list. * + Doomed spans are removed periodically by the monitor thread. * + ENODEV reads from the D-channel will warant the same deferred + removal. Review: https://reviewboard.asterisk.org/r/3548/ + +2014-06-22 18:53 +0000 [r416996] George Joseph + + * include/asterisk/astobj2.h, Makefile.rules, Makefile, /: astobj2: + Add an ao2_replace macro to astobj2.h This macro replaces one + object reference with another cleaning up the original. param dst + Pointer to the object that will be cleaned up. param src Pointer + to the object replacing it. src's ref count is bumped if it's + non-NULL. dst's ref count is decremented if it's non-NULL. src is + assigned to dst, This patch was reviewed on IRC by coreyfarrell + and mjordan. Tested by: George Joseph ........ Merged revisions + 416995 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-20 23:18 +0000 [r416872-416935] George Joseph + + * /, configure, include/asterisk/autoconfig.h.in: build: Allow + autoconf/ast_ext_tool_check to handle cross-compiling better. + ast_ext_tool_check.m4 isn't handling cases where a path to a + package is provided (E.G. --with-mysqlclient=/some/sysroot) and + the package has a config tool (E.G. mysql_config) and the package + has its own subdirectories in include or lib. For example, + mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but + ast_ext_tool_check sets MYSQLCLIENT_LIB to + ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its + includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not + directly in ${LIBXML2_DIR}/usr/include. Both cause configure to + fail and there are others in the same boat. The problem is caused + by logic in ast_ext_tool_check that overrides the result of the + config tool's --cflags and --libs options if package_DIR is set. + This patch prepends package_DIR (if specified) to the -L and -I + results from the package's config tool instead of overriding + them. A regenerated ./configure and + include/asterisk/autoconfig.h.in are included but can be + regenerated by running ./bootstrap.sh at any time. Tested by: + George Joseph Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3550/ ........ Merged + revisions 416929 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416930 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416931 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * autoconf/ast_ext_tool_check.m4, /: build: Allow + autoconf/ast_ext_tool_check to handle cross-compiling better. + ast_ext_tool_check.m4 isn't handling cases where a path to a + package is provided (E.G. --with-mysqlclient=/some/sysroot) and + the package has a config tool (E.G. mysql_config) and the package + has its own subdirectories in include or lib. For example, + mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but + ast_ext_tool_check sets MYSQLCLIENT_LIB to + ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its + includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not + directly in ${LIBXML2_DIR}/usr/include. Both cause configure to + fail and there are others in the same boat. The problem is caused + by logic in ast_ext_tool_check that overrides the result of the + config tool's --cflags and --libs options if package_DIR is set. + This patch prepends package_DIR (if specified) to the -L and -I + results from the package's config tool instead of overriding + them. Tested by: George Joseph Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3550/ ........ Merged + revisions 416870 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416871 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-20 20:57 +0000 [r416848-416850] Jonathan Rose + + * res/parking/parking_manager.c, /: res_parking: Make manager + commands register with module information Previously module + information was not included due to an oversight. Review: + https://reviewboard.asterisk.org/r/3626/ ........ Merged + revisions 416849 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/logger.c, CHANGES, include/asterisk/logger.h, + main/manager.c: Logger: Add manager command 'LoggerRotate' to + rotate logger Part of a series of AMI command equivalents to + existing CLI commands Review: + https://reviewboard.asterisk.org/r/3651/ + +2014-06-20 17:06 +0000 [r416830] Richard Mudgett + + * apps/app_voicemail.c, include/asterisk/app.h, main/app.c, + apps/app_directory.c, apps/app_chanspy.c: voicemail API + callbacks: Extract the sayname API call to its own registerd + callback. * Extract the sayname API call to its own registerd + callback. This allows the app_directory and app_chanspy + applications to say a mailbox owner's name using an alternate + provider when app_voicemail is not available because you are + using res_mwi_external. app_directory still uses the + voicemail.conf file. AFS-64 #close Reported by: Mark Michelson + +2014-06-20 15:27 +0000 [r416738-416807] George Joseph + + * main/astobj2_private.h, main/astobj2_container_private.h, + main/astobj2_container.c, main/astobj2_hash.c, + main/astobj2_rbtree.c, build_tools/cflags.xml, /, + tests/test_astobj2.c: astobj2: Additional refactoring to push + impl specific code down into the impls. Move some implementation + specific code from astobj2_container.c into astobj2_hash.c and + astobj2_rbtree.c. This completely removes the need for + astobj2_container to switch on RTTI and it no longer has any + knowledge of the implementation details. Also adds AO2_DEBUG as a + new compile option in menuselect which controls astobj2 debugging + independently of AST_DEVMODE and REF_DEBUG. Tested by: George + Joseph Review: https://reviewboard.asterisk.org/r/3593/ ........ + Merged revisions 416806 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, + include/asterisk/netsock2.h, include/asterisk/acl.h, + main/netsock2.c: pjsip cli: Change Identify to show CIDR notation + instead of netmasks. * Added ast_sockaddr_cidr_bits() to count + the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which + uses ast_sockaddr_cidr_bits() for the netmask instead of + ast_sockaddr_stringify_addr. * Changed + res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() + instead of ast_ha_join() for the CLI output. This is a CLI change + only. AMI was not affected. Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3652/ ........ Merged + revisions 416737 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-19 19:40 +0000 [r416736] Kinsey Moore + + * /, main/bridge.c, res/parking/parking_tests.c, + channels/sip/reqresp_parser.c, main/logger.c, main/test.c: Fix + build warnings with TEST_FRAMEWORK enabled ........ Merged + revisions 416732 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416733 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416734 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-19 16:04 +0000 [r416589-416670] George Joseph + + * pbx/pbx_lua.c, /: Remove the problematic and unneeded + AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c + AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be + incorrectly loaded before pbx_config. pbx_config was therefore + blowing away contexts that were created by pbx_lua. With + AST_MODFLAG_DEFAULT the load order is now correct and contexs are + being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed + anyway since no other modules needed its global symbols that + early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by: + Dennis Guse Tested by: George Joseph Review: + https://reviewboard.asterisk.org/r/3629/ ........ Merged + revisions 416668 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416669 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/extensions.lua.sample, /: Update extensions.lua.sample + with naming conflict guidance. The sample extensions.lua was + causing pbx_lua to fail to load when parsing 'app.goto("default", + "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This + patch adds guidance to extensions.lua.sample and changed + 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", + 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by: + gtjoseph Review: https://reviewboard.asterisk.org/r/3627/ + ........ Merged revisions 416581 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-18 04:22 +0000 [r416561] Matthew Jordan + + * /, main/stasis_channels.c: stasis_channels: Update the stasis + cache if manager variables are needed In r416211, the publishing + of variable changes was modified such that a cached channel + snapshot was used if manager variables were not requested with + each AMI event. This was done to reduce the amount of channel + snapshots created. However, an assumption was made that + generating a channel snapshot and publishing the snapshot to the + channel topic was sufficient to ensure that the cache would be + updated; this is not the case. The channel snapshot type must be + used to force a snapshot update. This patch updates the + publication of channel variables such that the cache is updated + prior to publication of the channel variable message if manager + variables are in use. This ensures that all AMI events receive + the variable update when they are supposed to. Note that this + issue was caught by the Asterisk Test Suite (go go testing) + ........ Merged revisions 416557 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-17 18:45 +0000 [r416444-416503] Mark Michelson + + * /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to + set inheritable channel variables. ........ Merged revisions + 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 416501 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416502 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pidf_body_generator.c, /, + res/res_pjsip_xpidf_body_generator.c: Fix string growth algorithm + for XML presence bodies. pjpidf_print() does not return < 0 if + there is not enough room for the document to be printed. Rather, + it returns 39, the length of the XML prolog. The algorithm also + had a bug in that it would return if it attempted to grow the + string larger. ........ Merged revisions 416442 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-17 16:33 +0000 [r416443] Kinsey Moore + + * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated + start calls Currently, music on hold will stop and then start + again from the beginning if ast_moh_start() is called multiple + times. This can happen if a call is put on hold repeatedly (the + channel receives multiple HOLD control frames) and can be + triggered from ARI by starting MoH on a channel multiple times. + This is fairly jarring/annoying to users. This change prevents + MoH from being restarted if the requested music class is the same + as the one currently playing. This includes an extra check to + prevent the errors previously experienced in the testsuite and + has 100+ test runs behind it. Review: + https://reviewboard.asterisk.org/r/3615/ ........ Merged + revisions 416439 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416440 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416441 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-16 18:27 +0000 [r416416] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/sig_ss7.h, configure, channels/chan_dahdi.h, + configure.ac, UPGRADE.txt, configs/ss7.timers.sample (added), + CHANGES, channels/sig_ss7.c: chan_dahdi: Adds support for major + update to libss7. * SS7 support now requires libss7 v2.0 or + later. The new libss7 is not backwards compatible. * Added SS7 + support for connected line and redirecting. * Most SS7 CLI + commands are reworked as well as new SS7 commands added. See + online CLI help. * Added several SS7 config option parameters + described in chan_dahdi.conf.sample. * ISUP timer support + reworked and now requires explicit configuration. See + ss7.timers.sample. Special thanks to Kaloyan Kovachev for his + support and persistence in getting the original patch by adomjan + updated and ready for release. SS7-27 #close Reported by: adomjan + +2014-06-16 16:22 +0000 [r416394] Kevin Harwell + + * include/asterisk/http_websocket.h, tests/test_websocket_client.c, + res/res_http_websocket.c: res_http_websocket: read/write string + fixup There was a problem when reading a string from the + websocket. It assumed the received data had a null terminator and + tried to write the data to an ast_str. This of course could/would + read past the end of the given buffer while writing the data to + the internal buffer of ast_str. Modified the the code to + correctly place a null terminator on the result string. + +2014-06-16 09:04 +0000 [r416339] Igor Goncharovskiy + + * cel/cel_sqlite3_custom.c, main/db.c, res/res_config_sqlite3.c, + cdr/cdr_sqlite3_custom.c, /: We have faced situation when using + CDR and CEL by sqlite3 modules. With system having high load + (~100 concurrent calls created by sipp) we found many cdr and cel + records missed. There is special finction in sqlite3, that make + able to fix this situation - sqlite3_wait_timeout, that also can + replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this + function can be used for aastdb and res_config_sqlite3 to avoid + missed writes to sqlite db. #ASTERISK-23766 #close Reported by: + Igor Goncharovsky Review: + https://reviewboard.asterisk.org/r/3559/ ........ Merged + revisions 416336 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416337 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416338 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-16 02:40 +0000 [r416267-416319] Matthew Jordan + + * /, channels/chan_sip.c: channels/chan_sip: Forbid remote bridging + if T.38 is negotiated When a framehook is removed - such as the + fax gateway framehook - the bridge framework will re-evaluate the + bridge mixing technologies to see if it can improve the bridging. + When this occurs, get_rtp_info will be called to determine if + local or remote bridging can be used. Using remote bridging will + cause a fax to fail, as direct media negotiation will cause some + small number of packets to not arrive at the remote endpoint. + This patch forces local native bridging if T.38 negotiation is in + progress or has been established. ........ Merged revisions + 416318 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/channel_internal_api.c: channel_internal_api: Publish a + snapshot change when linkedids change Snapshots are now not + published *quite* as much as they used to. One instance where + they are not published any longer is during bridge enter and exit + - the state of the channel doesn't change, the bridge does. + However, channels are changed when a linkedid is propagated; + previously, the channel's state would be updated and published + during the bridge enter event. Now this must be explicitly done. + ........ Merged revisions 416300 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_stasis_endpoints.c: test_stasis_endpoints: Remove + expected channel snapshot We no longer publish a channel snapshot + when it is associated with an endpoint; after all, the channel + itself hasn't changed - the endpoint state has changed. This + updates the channel_messages unit test accordingly. ........ + Merged revisions 416298 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This + patch reverts r416150. When the comparison between mohclass->name + and state->class->name is made, you are not guaranteed that (a) + state->class is non-NULL or that state or state->class are in a + safe state. Crashes caught by the bridges/transfer_capabilities + test. ........ Merged revisions 416251 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416252 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416255 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-14 19:26 +0000 [r416237] Corey Farrell + + * res/res_manager_devicestate.c, res/res_manager_presencestate.c: + res_manager_devicestate and res_manager_presencestate missing + support level Add MODULEINFO comment block to define support + level core for these new modules. Review: + https://reviewboard.asterisk.org/r/3620/ + +2014-06-13 18:24 +0000 [r416216] Matthew Jordan + + * res/res_agi.c, res/res_pjsip/pjsip_configuration.c, + main/stasis_channels.c, res/ari/resource_channels.c, + main/bridge_channel.c, main/pbx.c, main/stasis_cache.c, /, + apps/app_meetme.c, main/pickup.c, main/channel_internal_api.c, + include/asterisk/channel.h, main/core_local.c, main/aoc.c, + main/endpoints.c, main/cel.c, apps/app_queue.c, + main/stasis_bridges.c, apps/app_agent_pool.c, main/cli.c, + main/channel.c, main/dial.c, main/manager.c, + include/asterisk/stasis_channels.h: stasis: Reduce creation of + channel snapshots to improve performance During some performance + testing of Asterisk with AGI, ARI, and lots of Local channels, we + noticed that there's quite a hit in performance during channel + creation and releasing to the dialplan (ARI continue). After + investigating the performance spike that occurs during channel + creation, we discovered that we create a lot of channel snapshots + that are technically unnecessary. This includes creating + snapshots during: * AGI execution * Returning objects for ARI + commands * During some Local channel operations * During some + dialling operations * During variable setting * During some + bridging operations And more. This patch does the following: - It + removes a number of fields from channel snapshots. These fields + were rarely used, were expensive to have on the snapshot, and + hurt performance. This included formats, translation paths, Log + Call ID, callgroup, pickup group, and all channel variables. As a + result, AMI Status, "core show channel", "core show channelvar", + and "pjsip show channel" were modified to either hit the live + channel or not show certain pieces of data. While this is + unfortunate, the performance gain from this patch is worth the + loss in behaviour. - It adds a mechanism to publish a cached + snapshot + blob. A large number of publications were changed to + use this, including: - During Dial begin - During Variable + assignment (if no AMI variables are emitted - if AMI variables + are set, we have to make snapshots when a variable is changed) - + During channel pickup - When a channel is put on hold/unhold - + When a DTMF digit is begun/ended - When creating a bridge + snapshot - When an AOC event is raised - During Local channel + optimization/Local bridging - When endpoint snapshots are + generated - All AGI events - All ARI responses that return a + channel - Events in the AgentPool, MeetMe, and some in Queue - + Additionally, some extraneous channel snapshots were being made + that were unnecessary. These were removed. - The result of + ast_hashtab_hash_string is now cached in stasis_cache. This + reduces a large number of calls to ast_hashtab_hash_string, which + reduced the amount of time spent in this function in gprof by + around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged + revisions 416211 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-13 13:11 +0000 [r416149-416153] Kinsey Moore + + * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated + start calls Currently, music on hold will stop and then start + again from the beginning if ast_moh_start() is called multiple + times. This can happen if a call is put on hold repeatedly (the + channel receives multiple HOLD control frames) and can be + triggered from ARI by starting MoH on a channel multiple times. + This is fairly jarring/annoying to users. This change prevents + MoH from being restarted if the requested music class is the same + as the one currently playing. Review: + https://reviewboard.asterisk.org/r/3615/ ........ Merged + revisions 416150 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416151 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cel.c, /: CEL: Expose parking retreiver in extra field This + exposes the retreiver of a parked call under the "retreiver" key + of the extra field when this information is available. Review: + https://reviewboard.asterisk.org/r/3608/ ........ Merged + revisions 416148 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-13 05:16 +0000 [r416071] Richard Mudgett + + * main/http.c, include/asterisk/tcptls.h, main/tcptls.c, + main/manager.c, /, channels/chan_sip.c: AST-2014-007: Fix of fix + to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close + Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/3617/ ........ Merged + revisions 416066 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 416067 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 416070 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 21:27 +0000 [r416024] Rusty Newton + + * main/pbx.c: main/pbx - documentation - enhance 'core show hints' + and 'core show hint' help text Adds descriptive help text to + 'core show hints' and 'core show hint'. The text describes the + various columns for the sake of clarity. It takes into account + recent changes to the content displayed by the commands + https://reviewboard.asterisk.org/r/3604/ and + https://reviewboard.asterisk.org/r/3611/. ASTERISK-23764 Review: + https://reviewboard.asterisk.org/r/3610/ + +2014-06-12 20:17 +0000 [r415982] Kinsey Moore + + * res/res_pjsip_pubsub.c, /: Fix build in devmode for GCC 4.10 + ........ Merged revisions 415980 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 17:00 +0000 [r415907] Richard Mudgett + + * include/asterisk/utils.h, main/tcptls.c, main/manager.c, /, + channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c, + include/asterisk/tcptls.h, res/res_http_websocket.c, + configs/http.conf.sample: AST-2014-007: Fix DOS by consuming the + number of allowed HTTP connections. Simply establishing a TCP + connection and never sending anything to the configured HTTP port + in http.conf will tie up a HTTP connection. Since there is a + maximum number of open HTTP sessions allowed at a time you can + block legitimate connections. A similar problem exists if a HTTP + request is started but never finished. * Added http.conf + session_inactivity timer option to close HTTP connections that + aren't doing anything. Defaults to 30000 ms. * Removed the + undocumented manager.conf block-sockets option. It interferes + with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections + now have better authentication timeout protection. Though I + didn't remove the bizzare TLS timeout polling code from chan_sip. + * chan_sip can now handle SSL certificate renegotiations in the + middle of a session. It couldn't do that before because the + socket was non-blocking and the SSL calls were not restarted as + documented by the OpenSSL documentation. * Fixed an off nominal + leak of the ssl struct in handle_tcptls_connection() if the FILE + stream failed to open and the SSL certificate negotiations + failed. The patch creates a custom FILE stream handler to give + the created FILE streams inactivity timeout and timeout after a + specific moment in time capability. This approach eliminates the + need for code using the FILE stream to be redesigned to deal with + the timeouts. This patch indirectly fixes most of ASTERISK-18345 + by fixing the usage of the SSL_read/SSL_write operations. + ASTERISK-23673 #close Reported by: Richard Mudgett ........ + Merged revisions 415841 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415854 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415896 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 15:50 +0000 [r415839] Scott Griepentrog + + * /, apps/app_queue.c: app_queue: delayed state can cause early + leavewhenempty ringing In app_queue, device state changes arrive + in event messages and update the queue member status value. That + value is checked in get_member_status() to decide that the caller + should leave when there are no available members. Although event + messages can be delayed by other activity, there is no adverse + affect by lagged status except in one specific case: there is + only one available member, it was just rung, and leavewhenempty + is enabled set for ringing members. This change adds a direct + check of the device state only under this condition where the + caller may be dropped incorrectly, resolving this issue without + affecting performance of app_queue normally. AST-1248 #close + Review: https://reviewboard.asterisk.org/r/3595/ Reported by: + Thomas Arimont ........ Merged revisions 415833 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415835 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415836 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 15:39 +0000 [r415834] Jonathan Rose + + * apps/app_mixmonitor.c, /, UPGRADE.txt: MixMontior: Add class + authorization requirements to MixMonitor AMI commands MixMonitor + AMI commands StartMixMonitor and StopMixMonitor lacked class + authorization. StopMixMonitor now requires that the manager user + either have the call or system class authorization. + StartMixMonitor is a slightly larger issue since it can execute + shell commands if the right arguments are passed into it, and we + consider this a permission escalation. A security release will be + issued for problem this shortly. ASTERISK-23609 #close Reported + by: Corey Farrell ........ Merged revisions 415825 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415832 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 14:39 +0000 [r415813] Kevin Harwell + + * res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: unauthenticated + remote crash in PJSIP pub/sub framework A remotely exploitable + crash vulnerability exists in the PJSIP channel driver's pub/sub + framework. If an attempt is made to unsubscribe when not + currently subscribed and the endpoint's "sub_min_expiry" is set + to zero, Asterisk tries to create an expiration timer with zero + seconds, which is not allowed, so an assertion raised. The fix + was to reject a subscription that is attempting to unsubscribe + when not being already subscribed. Asterisk now checks for this + situation appropriately and responds with a 400 instead of + crashing. AST-2014-005 ASTERISK-23489 #close ........ Merged + revisions 415812 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 14:15 +0000 [r415795] Mark Michelson + + * res/res_pjsip.c, /: Fix potential deadlock situation in + res_pjsip. SIP transaction timeouts are handled in the PJSIP + monitor thread. When this happens on a subscription, and the + subscription is destroyed, the subscription destruction is + dispatched synchronously to the threadpool. The issue is that the + PJSIP dialog is locked by the monitor thread, and then the + dispatched task attempts to lock the dialog. This leads to a + deadlock that causes SIP traffic to no longer be accepted on the + Asterisk server. The fix here is to treat the monitor thread as + if it were a threadpool thread when it attempts to dispatch + synchronous tasks. This way, the dispatched task turns into a + simple function call within the same thread, and the locking + issue is averted. AST-2014-008 ASTERISK-23802 #close ........ + Merged revisions 415794 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 11:34 +0000 [r415767] Joshua Colp + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, + res/res_pjsip_exten_state.c, include/asterisk/res_pjsip.h, + include/asterisk/res_pjsip_pubsub.h, + res/res_pjsip_pubsub.exports.in, /, + contrib/ast-db-manage/config/versions/c6d929b23a8_create_pjsip_subscription_persistence_.py + (added), res/res_pjsip_mwi.c: res_pjsip_pubsub: Persist + subscriptions in sorcery so they are recreated on startup. This + change makes res_pjsip_pubsub persist inbound subscriptions in + sorcery. By default this uses the local astdb but it can also be + configured to store within an outside database. When Asterisk is + started these subscriptions are recreated if they have not + expired. Notifications are sent to the devices which have + subscribed and they are none the wiser that the system has + restarted. Review: https://reviewboard.asterisk.org/r/3598/ + ........ Merged revisions 415766 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-12 07:52 +0000 [r415749] Walter Doekes + + * UPGRADE.txt, contrib/scripts/safe_asterisk, Makefile, /: + safe_asterisk: Overwrite old safe_asterisk on make install. From + now on, make install will overwrite safe_asterisk with the latest + version. You need to move any local modifications to files inside + /etc/asterisk/startup.d, if you have any. See also commits + r394939 and r397938. ASTERISK-21965 #close Patches: + safe_asterisk.patch uploaded by jkister (License 6232, modified + by me) ........ Merged revisions 415748 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-11 23:01 +0000 [r415730] Richard Mudgett + + * main/format.c, /: format.c: Fix misuse of hash container + function. The supplied hash function to a container must be + idempotent given the object's key value to figure out which + container bucket the object belongs in. Returning a random number + or the current container count is not idempotent. The "computed + hash" value doesn't help find the object later in those cases. * + Fixed the format_list container to actually be a list since that + is how the container is used. Conceptually, if more than 283 + formats were added to the format_list then odd things may have + happened before the fix. ........ Merged revisions 415728 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415729 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-11 20:22 +0000 [r415698-415715] Scott Griepentrog + + * main/pbx.c: CLI: correct presence information on core show hints + Adds presence to core show hint and changes presence string + conversion to use the correct function. ASTERISK-23858 #close + Review: https://reviewboard.asterisk.org/r/3611/ + + * main/pbx.c: CLI: add presence information to core show hints Adds + presence state value to output of core show hints. Also reformats + the output slightly so it doesn't use as much space as it would + otherwise. Was: 1000@demo : SIP/1000 State:Unavailable Watchers 0 + Now: 1000@demo : SIP/1000 State:Unavailable Presence:Idle + Watchers 0 AFS-53 #close Review: + https://reviewboard.asterisk.org/r/3604/ + +2014-06-10 18:32 +0000 [r415679] Kinsey Moore + + * main/channel.c, /: Fix build in dev mode due to signed/unsigned + mismatch ........ Merged revisions 415678 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-10 16:06 +0000 [r415659] Jonathan Rose + + * main/message.c, /, res/res_pjsip_notify.c: PJSIP: PJSIPNotify - + Strip content-length headers and add documentation Documentation + for how to add custom headers/content to notifies created with + the PJSIPNotify manager action was a little sparse and it also + wasn't vetting application of Content-length headers like its + chan_sip equivalent was (so two Content-length headers could be + applied... and PJSIP determines the content length anyway, so it + just opens people up for error). This patch also flips the + variable order so that the variables are interpreted in the same + order as they are put in the AMI action. Review: + https://reviewboard.asterisk.org/r/3587/ ........ Merged + revisions 415658 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-10 09:28 +0000 [r415630] Alexandr Anikin + + * addons/chan_ooh323.c, /: chan_ooh323: fix loading module failure + if there no accessible h323_log or ooh323 config file change + return 1 to return AST_MODULE_LOAD_FAILURE on module load routine + few cosmetic changes ASTERISK-23814 #close (closes issue + ASTERISK-23814) Reported by: Igor Goncharovsky Patches: + ASTERISK-23814-ast11.patch ........ Merged revisions 415599 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415602 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-09 20:21 +0000 [r415580] Mark Michelson + + * res/res_pjsip_header_funcs.c, /: chan_pjsip: Fix bug where custom + SIP headers could be duplicated on outgoing INVITEs. When using + PJSIP_HEADER() to add custom headers to outgoing INVITE requests, + certain situations could result in the headers being duplicated. + For instance, if the request were retransmitted, or if the INVITE + were re-sent with authentication credentials, the custom headers + would be re-added to the request. The fix here is to, after + adding the custom headers to the outbound INVITE, remove the + datastore that holds the custom headers to add. This way, there + is no risk in accidentally adding them if the session supplement + is called into a second or third time. ........ Merged revisions + 415579 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-09 12:12 +0000 [r415524] Walter Doekes + + * /, UPGRADE.txt, contrib/scripts/safe_asterisk: safe_asterisk: + Cleanup additions to r415132. * Replaced a stray echo that + should've been a message call in safe_asterisk. This replaces a + conditional log message by a slightly different message. Please + update your log parsing scripts. * Made the $NOTIFY mail Subject + more verbose by adding the machine name and exitstatus. (Note + that a 'make install' still won't overwrite your old + safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492 + #close ........ Merged revisions 415521 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415522 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415523 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-09 03:50 +0000 [r415466] Corey Farrell + + * /, main/autoservice.c: autoservice: stop thread on graceful + shutdown This change adds thread shutdown to autoservice for + graceful shutdowns only. ast_register_cleanup is backported to + 1.8 to allow this. The logger callid is also released on shutdown + in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3594/ ........ Merged + revisions 415463 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415464 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-08 18:12 +0000 [r415444] Matthew Jordan + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + main/bridge_channel.c, main/channel.c, main/pbx.c, /, + main/framehook.c, main/bridge_after.c: bridges/bridge_native_rtp: + Reconfigure bridge on removal of framehook This patch is a re-do + of r414122. When r414122 was merged, a major problem with it was + uncovered. UNBRIDGE soft hangup flags have a catastrophic effect + on the pbx core if they leak out from the bridge layer: the + channel gets hung up. With the number of threads involved in a + blind transfer, and with the initial patch, it was likely that + this would occur. This caused a large number of test failures + This patch is nearly identical with the one proposed in r414122, + save for the following changes: - We explicitly clear the + UNBRIDGE flag when setting an after goto on a channel in a bridge + - Defensively, if we encounter an UNBRIDGE flag in the pbx core, + we handle it https://reviewboard.asterisk.org/r/3585/ ........ + Merged revisions 415443 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-07 00:42 +0000 [r415428] Richard Mudgett + + * include/asterisk/bridge.h, /: bridge.h: Remove redundant struct + ast_bridge_channel forward declaration. ........ Merged revisions + 415427 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 21:44 +0000 [r415411] Jonathan Rose + + * include/asterisk/manager.h, main/config.c, main/manager.c, /, + channels/chan_sip.c, include/asterisk/config.h: chan_sip: Fix + order of variables specified in SIPNotify action Prior to this + patch, sequential variables would be ordered in reverse from the + order specified in the manager action. Review: + https://reviewboard.asterisk.org/r/3588/ ........ Merged + revisions 415359 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415390 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415410 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 20:45 +0000 [r415358] Kevin Harwell + + * main/uri.c, tests/test_websocket_client.c: core uri: Custom uri + parsing error when no query parameters If using the custom URI + parsing code (not external uriparser lib) and there was no query + parameters the resulting pointer would be NULL and then an + attempt was made to subtract from it. The pointer is now set to a + valid value if there is no query parameter(s). Also, in the + 'ast_uri_make_host_with_port' function when setting the + terminator on the resulting string it was writing it one past the + end of allocated memory. It now writes the string terminator + appropriately. + +2014-06-06 19:13 +0000 [r415343] Kinsey Moore + + * /, res/res_pjsip_sdp_rtp.c: PJSIP: Remove premature write of raw + formats Currently, there are situations that can occur when using + chan_pjsip and certain dialplan applications (notably ChanSpy()) + that can cause the channel to get no audio with scrolling + warnings about format mismatches. This is caused by a failure to + update translation paths on a mid-call native format update since + the raw formats have already been updated by res_pjsip_sdp_rtp.c + in set_caps(). Removing the premature raw format updates allows + the translation paths to be setup correctly and the raw read and + write formats with them. AFS-63 #close ........ Merged revisions + 415342 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 14:12 +0000 [r415319] George Joseph + + * tests/test_astobj2.c, main/astobj2_private.h (added), + main/astobj2.c, main/astobj2_container_private.h (added), + main/astobj2_container.c (added), main/astobj2_hash.c (added), + main/astobj2_rbtree.c (added), /, include/asterisk/astobj2.h: + Split astobj2.c into more maintainable components. Split + astobj2.c into the following files to improve maintainability. + astobj2.c - object primitives, object primitive misc and + initialization code. astobj2_private.h - internal object + declarations needed by the containers. astobj2_container.c - + generic conainer and container misc code. + astobj2_container_hash.c - hash container specific code. + astobj2_container_rbtree.c - rbtree container specific code. + astobj2_container_private.h - generic container definitions and + rtti prototypes. https://reviewboard.asterisk.org/r/3576/ + ........ Merged revisions 415317 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-06 12:49 +0000 [r415302] Rusty Newton + + * /, configs/cli_aliases.conf.sample: configs/cli_aliases.conf: Two + new aliases, plus enhancements for context names. Changed naming + of included alias templates to avoid confusion between version + names. For example, asterisk12 was for asterisk 1.2, so I changed + it to asterisk_1dot2, so that later we can use asterisk_12 for + Asterisk 12. Added alias for "features reload" to the template + for Asterisk 11 style syntax template, as features reload was + removed in 12, but you can still do "module reload features" + Added alias for "pjsip reload" to the friendly template. It is + shorter than "module reload res_pjsip.so" and if some are like + me; I constantly forget that reloading chan_pjsip doesn't parse + config. Remembering "pjsip reload" is just easier. ASTERISK-23654 + #close ASTERISK-23654 #comment Fixed by adding two new aliases + and enhancements for context names. Review: + https://reviewboard.asterisk.org/r/3572/ ........ Merged + revisions 415301 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-05 19:04 +0000 [r415231-415288] Richard Mudgett + + * main/config.c: config: Fix indentation and missing curlies in + config_text_file_load(). + + * main/config.c, /: config: Fix config files not reloading when + only an included file changes. The twisted logic determining if a + config file should be reloaded was mostly broken and disabled. + The incorrect test that ASTERISK-23383 fixed actually reenabled + the broken logic. The incorrect test was causing the timestamp to + always be cleared which caused config files with includes to + always be reloaded. * Made wildcard includes always cause a + reload. Determining if a file was deleted cannot be determined + without restructuring the cache to determine if any files are + missing from the last files actually loaded. Also without + refactoring config_text_file_load(), the glob loop couldn't check + more than one file for changes anyway. * Made remove the cache + entry if the file no longer exists when trying to get its + timestamp or it is no longer a regular file. This fixes the + corner case where the file was loaded, then deleted, then the + config reloaded, then the file restored with the same timestamp, + and then the config reloaded again. * Made remove the cache entry + include list when actually loading the file. This gets rid of any + stale includes the file had from the last time the file was + loaded. ASTERISK-23683 #close Reported by: tootai Review: + https://reviewboard.asterisk.org/r/3575/ ........ Merged + revisions 415225 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415229 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415230 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-05 17:22 +0000 [r415223] Kevin Harwell + + * tests/test_uri.c (added), include/asterisk/http_websocket.h, + main/http.c, main/uri.c (added), tests/test_websocket_client.c + (added), res/res_http_websocket.c, include/asterisk/http.h, + include/asterisk/uri.h (added), + res/res_http_websocket.exports.in: res_http_websocket: Create a + websocket client Added a websocket server client in Asterisk. + Asterisk has a websocket server, but not a client. The ability to + have Asterisk be able to connect to a websocket server can + potentially be useful for future work (for instance this could + allow ARI to connect back to some external system, although more + work would be needed in order to incorporate that). Also a couple + of things to note - proxy connection support has not been + implemented and there is limited http response code handling + (basically, it is connect or not). Also added an initial new URI + handling mechanism to core. Internet type URI's are parsed into a + data structure that contains pointers to the various parts of the + URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell + Review: https://reviewboard.asterisk.org/r/3541/ + +2014-06-05 14:49 +0000 [r415208] Matthew Jordan + + * /, apps/app_confbridge.c: app_confbridge: Allow muting of users + waiting to enter a ConfBridge Prior to this patch, users waiting + to enter a ConfBridge were not considered when muted via the CLI + or via AMI. Instead, a confusing message would be emitted stating + that the channel did not exist. This patch allows a user to be + muted when waiting to enter a ConfBridge conference. This is + equivalent to start when muted, only toggled via the CLI or AMI. + Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824 + #close patches: rb3582.patch uploaded by tm1000 (License 6524) + ........ Merged revisions 415206 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415207 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-05 11:59 +0000 [r415192] Kinsey Moore + + * /, channels/chan_pjsip.c: PJSIP: Send initial connected line + information This makes chan_pjsip send connected line information + when it is called so that connected line information is available + on the connected channel. (closes issue DPMA-442) Reported by: + John Bigelow Review: https://reviewboard.asterisk.org/r/3584/ + ........ Merged revisions 415191 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-04 20:16 +0000 [r415173] Walter Doekes + + * /, contrib/scripts/safe_asterisk: safe_asterisk: Cleanup and + debian compatibility. Cleans up the safe_asterisk script and adds + the ASTSAFE_FOREGROUND option that allows the debian asterisk + init script to capture the right pid. * Drop the vim #modeline + which wasn't used. Use test consistently without the odd + configure xno syntax. Double quote all paths. General cleanup. * + Don't output message()s to the console but only to TTY if set. * + Allow TTY to be "no" as well as empty (debian compatibility with + debian/patches/safe_asterisk-config). * Add option to export + ASTSAFE_FOREGROUND=1 from the init script that calls this to + disable backgrounding. Debian uses a similar method in + debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review: + https://reviewboard.asterisk.org/r/3574/ ........ Merged + revisions 415132 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415171 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415172 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-04 14:13 +0000 [r415116-415118] Matthew Jordan + + * /, channels/chan_pjsip.c: chan_pjsip: Add debug in RTP Engine + glue callback This patch adds some debug statements that aid with + determining why a direct media request may or may not be + initiated. ........ Merged revisions 415117 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: res_pjsip_session: Add debug + statement for session refreshes This small patch adds a debug + level 3 statement indicating how a session refresh is being sent + - either as a re-INVITE or as an UPDATE - and where the session + refresh is going. ........ Merged revisions 415115 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-04 07:27 +0000 [r415080] Corey Farrell + + * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c: + app_confbridge: Correct verification of conference name length + Conference names were not checked for maximum length, allowing + unexpected behaviour. This change adds checking to ensure the + maximum length is not exceeded. The maximum length is also + changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close + Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: + confbridge-enforce_max-1.8.patch uploaded by coreyfarrell + (license 5909) confbridge-enforce_max-11up.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 415060 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 415066 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 415078 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-03 07:36 +0000 [r415000] Walter Doekes + + * /, funcs/func_odbc.c: func_odbc: Fix fixed size buffers fix + (r414968). The change that removed the fixed size buffers in + odbc-related code -- removing arbitrary column width limits -- + was incomplete. This change adds: no segfault on writesql without + insertsql and return value checks after strdup. While I was in + the vicinity I cleaned up the linefeeds in the odbc function + descriptions, moved some code for clarity, removed some blobs and + noted (but didn't fix) that the 'odbc write ... exec' CLI command + doesn't behave as the dialplan equivalent when insertsql= is + used. ASTERISK-23582 #close Review: + https://reviewboard.asterisk.org/r/3579/ ........ Merged + revisions 414997 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414998 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-06-01 15:32 +0000 [r414976] Joshua Colp + + * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Take the + bridge type choice of both channels into account. The + bridge_native_rtp module currently uses the bridge result of the + first channel that joins a bridge as the ultimate result. This + means that if the first channel has direct media enabled but the + second does not a direct media bridge will still occur. This + change makes it so that both sides are taken into account. If + either side forbids the bridge or responds with a local bridge + result then either a generic or local bridge occurs. + ASTERISK-23541 #close Reported by: Justin E Review: + https://reviewboard.asterisk.org/r/3577/ ........ Merged + revisions 414975 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-30 14:53 +0000 [r414949] Kinsey Moore + + * res/res_pjsip_refer.c, /: PJSIP: Prevent crash on blind transfer + Blind transfers don't go too well with NULL channels which can + occur if the channel has already been transferred away. (closes + issue ASTERISK-23718) Reported by: Jonathan Rose ........ Merged + revisions 414948 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-30 12:42 +0000 [r414883-414935] Matthew Jordan + + * main/audiohook.c, CHANGES, res/ari/ari_model_validators.c, + res/ari/ari_model_validators.h, funcs/func_talkdetect.c (added), + include/asterisk/stasis_channels.h, + rest-api/api-docs/events.json, /, main/stasis_channels.c: + TALK_DETECT: A channel function that raises events when talking + is detected This patch adds a new channel function TALK_DETECT + that, when set on a channel, causes events indicating the + start/stop of talking on a channel to be emitted to both AMI and + ARI clients. The function allows setting both the silence + threshold (the length of silence after which we decide no one is + talking) as well as the talking threshold (the amount of energy + that counts as talking). Parameters can be updated on a channel + after talk detection has been enabled, and talk detection can be + removed at any time. The events raised by the function use a + nomenclature similar to existing AMI/ARI events. For AMI: + ChannelTalkingStart/ChannelTalkingStop For ARI: + ChannelTalkingStarted/ChannelTalkingFinished Review: + https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close + Reported by: Matt Jordan ........ Merged revisions 414934 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, /: main/config.c: AMI action UpdateConfig EmptyCat + clears all categories When invoking UpdateConfig AMI action with + Action set to EmptyCat, Asterisk will make all categories empty + in the config but the one requested with a Cat variable. This is + due to a bug in ast_category_empty (main/config.c) that makes an + incorrect comparison for a category name. This patch corrects the + comparison such that only the requested category is cleared. + Review: https://reviewboard.asterisk.org/r/3573/ #ASTERISK-23803 + #close Reported by: zvision patches: manager.c.diff uploaded by + zvision (License 5755) ........ Merged revisions 414880 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414881 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-29 18:51 +0000 [r414861] Kinsey Moore + + * main/pbx.c, /: PBX: Prevent incorrect hint parsing Dynamic and + pattern matching hints should not be checked for their last known + state until they are instantiated by subscribers. (closes issue + AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted + by Matt Jordan (license 6283) ........ Merged revisions 414813 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 414859 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414860 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 22:54 +0000 [r414798] Matthew Jordan + + * main/loader.c, include/asterisk/logger.h, res/res_config_curl.c, + cel/cel_odbc.c, res/res_config_odbc.c, + bridges/bridge_builtin_features.c, main/optional_api.c, + main/logger.c, main/config_options.c, cdr/cdr_odbc.c, + apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c, + main/xmldoc.c, apps/app_voicemail.c, cel/cel_pgsql.c, + channels/chan_unistim.c, res/res_config_pgsql.c, main/pbx.c, + cdr/cdr_sqlite3_custom.c, res/res_fax.c, main/bridge.c, + apps/app_waitforsilence.c, cdr/cdr_adaptive_odbc.c, + res/parking/parking_applications.c, cdr/cdr_pgsql.c, + res/res_jabber.c: Logger/CLI/etc.: Fix some aesthetic issues; + reduce chatty verbose messages This patch addresses some + aesthetic issues in Asterisk. These are all just minor tweaks to + improve the look of the CLI when used in a variety of settings. + Specifically: * A number of chatty verbose messages were removed + or demoted to DEBUG messages. Verbose messages with a verbosity + level of 5 or higher were - if kept as verbose messages - demoted + to level 4. Several messages that were emitted at verbose level 3 + were demoted to 4, as announcement of dialplan applications being + executed occur at level 3 (and so the effects of those + applications should generally be less). * Some verbose messages + that only appear when their respective 'debug' options are + enabled were bumped up to always be displayed. * + Prefix/timestamping of verbose messages were moved to the + verboser handlers. This was done to prevent duplication of + prefixes when the timestamp option (-T) is used with the CLI. * + Verbose magic is removed from messages before being emitted to + non-verboser handlers. This prevents the magic in multi-line + verbose messages (such as SIP debug traces or the output of + DumpChan) from being written to files. * _Slightly_ better + support for the "light background" option (-W) was added. This + includes using ast_term_quit in the output of XML documentation + help, as well as changing the "Asterisk Ready" prompt to bright + green on the default background (which stands a better chance of + being displayed properly than bright white). Review: + https://reviewboard.asterisk.org/r/3547/ + +2014-05-28 20:53 +0000 [r414781] Rusty Newton + + * /, configs/pjsip.conf.sample: pjsip.conf: privkey_file should be + priv_key_file, mediaencryption=yes should be mediaencryption=sdes + privkey_file was missed in the snake case update. An example + included an invalid value for the mediaencryption option. + ........ Merged revisions 414780 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 17:46 +0000 [r414764-414766] Matthew Jordan + + * rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, + rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, include/asterisk/manager.h, + rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json: AMI/ARI: Update version + numbers Update the semantic versioning of ARI to 1.3.0 and AMI to + 2.3.0 to account for backwards compatible changes going from + 12.2.0 to 12.3.0. ........ Merged revisions 414765 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/ast-db-manage/cdr/env.py, /: ast-db-manage/cdr/env.py: + Don't fail if a config file can't be loaded When generating SQL + files via the repotools alembic_creator.py script, a + configuration object is used programatically with SQLAlechemy, as + opposed to a configuration file. This patch ignores failures to + interpret a config file, as ... there isn't one in this case. + ........ Merged revisions 414763 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 16:56 +0000 [r414748-414750] Richard Mudgett + + * res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h, /, + res/res_pjsip_t38.c: res_pjsip_session: Fix leaked video RTP + ports. Simply enabling PJSIP to negotiage a video codec (e.g., + h264) would leak video RTP ports if the codec were not negotiated + by an incoming call. * Made add_sdp_streams() associate the + handler with the media stream if the handler handled the media + stream. Otherwise, when the ast_sip_session_media object was + destroyed it didn't know how to clean up the RTP resources. * + Fixed sdp_requires_deferral() associating the handler with the + media stream when deciding if the SDP processing needs to be + deferred for T.38. Like the leaked video RTP ports, the T.38 + handler needs to clean up allocated resources from deciding if + SDP processing needs to be deffered. * Cleaned up some dead code + in handle_incoming_sdp() and sdp_requires_deferral(). + ASTERISK-23721 #close Reported by: cervajs Review: + https://reviewboard.asterisk.org/r/3571/ ........ Merged + revisions 414749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, CHANGES, apps/app_agent_pool.c: app_agent_pool: Return to + dialplan if the agent fails to ack the call. Improvements to the + agent pool functionality. * AgentRequest no longer hangs up the + caller if the agent fails to connect with the caller. It now + continues in the dialplan. * AgentRequest returns AGENT_STATUS + set to NOT_CONNECTED if the agent failed to connect with the + call. Most likely because the agent did not acknowledge the call + in time or got disconnected. * The agent alerting play file + configured by the agent.conf custom_beep option can now be + disabled by setting the option to an empty string. The agent is + effectively alerted to a call presence when MOH stops. * Fixed + bridge reference leak when the agent connects with a caller. + ASTERISK-23499 #close Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3551/ ........ Merged + revisions 414747 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 11:37 +0000 [r414696] Joshua Colp + + * res/res_config_odbc.c, /, funcs/func_odbc.c: res_config_odbc: Use + dynamically sized buffers to store row data so values do not get + truncated. ASTERISK-23582 #close ASTERISk-23582 #comment Reported + by: Walter Doekes Review: + https://reviewboard.asterisk.org/r/3557/ ........ Merged + revisions 414693 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414694 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414695 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-28 09:43 +0000 [r414567-414679] Walter Doekes + + * /, channels/chan_unistim.c: chan_unistim: Unlock mutex in rare + OOM condition. #ASTERISK-23792 #close Reported by: Peter Whisker + Review: https://reviewboard.asterisk.org/r/3567/ ........ Merged + revisions 414677 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414678 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Start session timer at 200, not + at INVITE. Asterisk started counting the session timer at INVITE + while the other end correctly started at 200. This meant that for + short session-expiries (90 seconds) combined with long ringing + times (e.g. 30 seconds), asterisk would wrongly assume that the + timer was hit before the other end thought it was time to send a + session refresh. This resulted in prematurely ended calls. This + changes the session timer to start counting first at 200 like RFC + says it should. (Also removed a few excess NULL checks that would + never hit, because if they did, asterisk would have crashed + already.) ASTERISK-22551 #close Reported by: i2045 Review: + https://reviewboard.asterisk.org/r/3562/ ........ Merged + revisions 414620 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414628 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414636 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_config_odbc.c, /: res_config_odbc: Fix old and new + ast_string_field memory leaks. The ODBC realtime driver uses ^NN + parameter encoding to cope with the special meaning of the + semi-colon. A semi-colon in a field is interpreted as if the key + was supplied twice, something which isn't otherwise possible with + fixed database columns. E.g. allow=alaw;ulaw is parsed as + allow=alaw and allow=ulaw. A literal semi-colon is rewritten to + ^3B when stored in the database. The module uses a stringfield to + efficiently store the encoded parameters. However, this + stringfield wasn't always freed in some off-nominal cases. Commit + r413241 fixed initialization so the encoding for INSERT and + DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked + apparently.) But that commit forgot the frees. This change cleans + that up. Review: https://reviewboard.asterisk.org/r/3555/ + ........ Merged revisions 414564 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414565 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414566 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-25 02:37 +0000 [r414543] Matthew Jordan + + * /, main/core_unreal.c: core_unreal: Prevent double free of + core_unreal pvt When a channel is destroyed (such as via + ast_channel_release in off nominal paths in core_unreal), it will + attempt to free (via ast_free) the channel tech pvt. This is + problematic for a few reasons: 1. The channel tech pvt is an ao2 + object in core_unreal. Free'ing the pvt directly is no good. 2. + The channel tech pvt's reference count is dropped just prior to + calling ast_channel_release, resulting in the pvt's destruction. + Hence, the channel destructor is free'ing an invalid pointer. + This patch keeps the dropping of the reference count, but sets + the pvt to NULL on the channel prior to releasing it. This models + what would occur if the channel was hung up directly. ........ + Merged revisions 414542 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-23 17:36 +0000 [r414529] Matthew Jordan + + * tests/test_cel.c, /: test_cel: Fix unit tests broken due to event + def changes from res_corosync This patch instructs test_cel to + skip any IE types it doesn't care about. The addition of the raw + and bitfield types caused the tests to fail. ........ Merged + revisions 414528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-23 14:36 +0000 [r414475] Kinsey Moore + + * main/event.c, /: Fix signed/unsigned build warnings ........ + Merged revisions 414474 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 16:19 +0000 [r414417] Richard Mudgett + + * /, apps/app_meetme.c: app_meetme: Don't interrupt MOH for + waitmarked users. Occasionally, when the last marked user leaves + the conference, waitmarked users don't get MOH if MOH is supposed + to be played while a waitmarked user is waiting for another + marked user. * Made not interrupt MOH when the user is a + waitmarked user. The waitmarked user doesn't need to hear any + leave announcements from the conference as the user would have + already heard different leave announcements if they were enabled. + Apparently DAHDI occasionally sends unending non-silent streams + to these users or a normal user still in the conference has + continuous high background noise. These non-silent streams cause + MOH to be suspended while the never ending "announcement" is + played. Issue caused by ASTERISK-13680. AST-1349 #close Reported + by: Tyler Stewart Review: + https://reviewboard.asterisk.org/r/3543/ ........ Merged + revisions 414401 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414402 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414404 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 16:09 +0000 [r414406] Scott Griepentrog + + * rest-api/api-docs/events.json, /, res/stasis/app.c, + res/ari/resource_events.c, include/asterisk/stasis_app.h, + include/asterisk/stasis.h, apps/app_userevent.c, + res/ari/resource_events.h, res/ari/ari_model_validators.c, + CHANGES, main/stasis.c, res/ari/ari_model_validators.h, + include/asterisk/stasis_channels.h, res/res_ari_events.c, + main/stasis_channels.c, res/res_stasis.c, + main/manager_channels.c, main/stasis_endpoints.c: ARI: Add + ability to raise arbitrary User Events User events can now be + generated from ARI. Events can be signalled with arbitrary json + variables, and include one or more of channel, bridge, or + endpoint snapshots. An application must be specified which will + receive the event message (other applications can subscribe to + it). The message will also be delivered via AMI provided a + channel is attached. Dialplan generated user event messages are + still transmitted via the channel, and will only be received by a + stasis application they are attached to or if the channel is + subscribed to. This change also introduces the multi object blob + mechanism used to send multiple snapshot types in a single + message. The dialplan app UserEvent was also changed to use multi + object blob, and a new stasis message type created to handle + them. ASTERISK-22697 #close Review: + https://reviewboard.asterisk.org/r/3494/ ........ Merged + revisions 414405 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 15:52 +0000 [r414403] Jonathan Rose + + * include/asterisk/bridge.h, res/parking/parking_bridge_features.c, + channels/chan_mgcp.c, res/res_pjsip_refer.c, + channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, main/parking.c, main/bridge.c, + main/bridge_basic.c, res/parking/parking_applications.c, + include/asterisk/parking.h: res_pjsip_refer: Fix bugs involving + Parking/PJSIP/transfers PJSIP would never send the final 200 + Notify for a blind transfer when transferring to parking. This + patch fixes that. In addition, it fixes a reference leak when + performing blind transfers to non-bridging extensions. Review: + https://reviewboard.asterisk.org/r/3485/ ........ Merged + revisions 414400 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-22 14:02 +0000 [r414331-414348] Matthew Jordan + + * /, UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag ........ + Merged revisions 414345 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414346 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414347 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_corosync.c, include/asterisk/stasis.h, main/app.c, + main/devicestate.c, main/event.c, main/stasis.c, + include/asterisk/devicestate.h, include/asterisk/event.h, + main/stasis_message.c, /, include/asterisk/event_defs.h: + res_corosync: Update module to work with Stasis (and compile) + This patch fixes res_corosync such that it works with Asterisk + 12. This restores the functionality that was present in previous + versions of Asterisk, and ensures compatibility with those + versions by restoring the binary message format needed to pass + information from/to them. The following changes were made in the + core to support this: * The event system has been partially + restored. All event definition and event types in this patch were + pulled from Asterisk 11. Previously, we had hoped that this + information would live in res_corosync; however, the approach in + this patch seems to be better for a few reasons: (1) + Theoretically, ast_events can be used by any module as a binary + representation of a Stasis message. Given the structure of an + ast_event object, that information has to live in the core to be + used universally. For example, defining the payload of a device + state ast_event in res_corosync could result in an incompatible + device state representation in another module. (2) Much of this + representation already lived in the core, and was not easily + extensible. (3) The code already existed. :-) * Stasis message + types now have a message formatter that converts their payload to + an ast_event object. * Stasis message forwarders now handle + forwarding to themselves. Previously this would result in an + infinite recursive call. Now, this simply creates a new + forwarding object with no forwards set up (as it is the thing it + is forwarding to). This is advantageous for res_corosync, as + returning NULL would also imply an unrecoverable error. Returning + a subscription in this case allows for easier handling of message + types that are published directly to an aggregate topic that has + forwarders. Review: https://reviewboard.asterisk.org/r/3486/ + ASTERISK-22912 #close ASTERISK-22372 #close ........ Merged + revisions 414330 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-21 22:24 +0000 [r414297] Richard Mudgett + + * /, main/core_unreal.c: core_unreal: Only block media frames when + a generator is on both ends of an unreal channel. The fix for + ASTERISK-12292 was a bit too aggressive. You could have + generators pointed at each other on local channels but need to + get other kinds of frames such as DTMF or CONNECTED_LINE frames + accross. ........ Merged revisions 414269 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414270 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414272 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-21 19:08 +0000 [r414217] Scott Griepentrog + + * /, funcs/func_strings.c: pbx.c: prevent potential crash from + recursive replace() Recurisve usage of replace() resulted in + corruption of the temporary string storage and potential crash. + By changing the string to be allocated separtely per instance, + this is eliminated. ASTERISK-23650 #comment Reported by: Roel van + Meer ASTERISK-23650 #close Review: + https://reviewboard.asterisk.org/r/3539/ ........ Merged + revisions 414214 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414215 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414216 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-19 19:52 +0000 [r414196] Paul Belanger + + * res/res_stasis_answer.c, /: Replace __ast_answer with + ast_raw_answer in app_control_answer While load testing an ARI + application, I noticed asterisk was returning HTTP 500 internal + server errors on channels/:id/answer. After talking to + #asterisk-dev, the issue appeared to be a lack of media flowing + after __ast_answer() was called. So now, we call ast_raw_answer + instead and no longer wait for media. ASTERISK-23758 #close + Review: https://reviewboard.asterisk.org/r/3549/ ........ Merged + revisions 414195 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-19 01:10 +0000 [r414123-414138] Matthew Jordan + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + main/bridge_channel.c, res/res_pjsip_refer.c, + res/res_pjsip_session.c, main/channel.c, /, main/framehook.c: + Undo r414123 The Test Suite caught a few problems, undoing until + those are resolved + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + main/bridge_channel.c, res/res_pjsip_session.c, main/channel.c, + /, main/framehook.c: bridge_native_rtp/bridge_channel: Fix direct + media issues due to frame hook This patch fixes issues with + direct media bridges that occur after a blind transfer. These + issues were caught by the (currently failing) + pjsip/transfers/blind_transfer/caller_direct_media test. The test + currently fails primarily for two reasons: (1) When Bob and + Charlie (the transfer target and the transfer destination) enter + a bridge together, the framehook remains on the transfer target + channel until both channels are in the bridge. As it consumes + voice frames, the initial bridge type is a simple bridge. The + framehook is removed when both channels are in the bridge; + however, this does not currently cause the bridging framework to + re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE + poke to the transfer target channel when a framehook is removed + so the bridge can re-evaluate itself. (2) When a channel leaves a + native RTP bridge, it may be leaving due to being hung up. + Sending a re-INVITE to a channel that is about to be hung up is + not nice - in fact, there's a good chance we'll send the BYE + request before the channel has had a chance to send back a 200 + OK. To be somewhat nicer, this patch adds a function to channel.h + that allows the bridging framework to query for exactly why a + channel is leaving a bridge via the channel's soft hangup flags. + This allows it to only send the re-INVITE if there's a chance the + channel will survive the native bridging experience. Review: + https://reviewboard.asterisk.org/r/3535/ ........ Merged + revisions 414122 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-16 20:06 +0000 [r413994-414070] Richard Mudgett + + * /, channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone + detection. * Check if waitingfordt (waitfordialtone) is enabled + in dahdi_read() to allow the DSP to operate early enough to + detect dialtone. * Made use the correct variable in + my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve + Davies Patches: dialtone_detect_fix (license #5012) patch + uploaded by Steve Davies Review: + https://reviewboard.asterisk.org/r/3534/ ........ Merged + revisions 414067 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 414068 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414069 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sig_pri.c, /: sig_pri.c: Pull the pri_dchannel() + PRI_EVENT_RING case into its own function. * Populate the + CALLERID(ani2) value (and the special CALLINGANI2 channel + variable) with the ANI2 value in addition to the PRI specific + ANI2 channel variable. * Made complete snapshot staging with the + channel lock held. All channel snapshots need to be done while + the channel lock is held. ........ Merged revisions 414050 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 414051 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI + conference data structure. Starting a conference recording using + the admin menu overwrites the DAHDI conference data structure + used to modify the admin user's conference mute mode. * Made no + longer pass the user's DAHDI conference data structure into the + menu functions. The menu now uses its own DAHDI conference data + structure to start the recording channel. * Moved the unlock + conf->playlock to before playing the conf-full message. No sense + keeping the lock while that prompt is playing. The user is never + going to get into the conference at that point. ........ Merged + revisions 413991 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413992 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413993 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-14 15:41 +0000 [r413897] Walter Doekes + + * /, res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a + few free()'s that should be ast_free()'s. Reverted an old + workaround that isn't necessary. Reorder a tiny bit of code. + Remove a bit of commented-out code. Review: + https://reviewboard.asterisk.org/r/3536/ ........ Merged + revisions 413894 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413895 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413896 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-13 18:09 +0000 [r413878] Jonathan Rose + + * main/netsock2.c, /, channels/chan_sip.c, + include/asterisk/netsock2.h: chan_sip: Add TLS and SRTP status to + CLI command 'sip show channel' ASTERISK-23564 #close Reported by: + Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ + ........ Merged revisions 413876 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413877 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-13 13:53 +0000 [r413790-413793] Walter Doekes + + * res/res_format_attr_h264.c, /: h264: Fix H264 SDP payload format. + https://tools.ietf.org/html/rfc3984#section-8.1 says + profile-level-id takes 3 bytes in base16 (6 hex digits). This + fixes video setup in certain cases. ASTERISK-23664 #close + ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume + Maudoux. Review: https://reviewboard.asterisk.org/r/3530/ + ........ Merged revisions 413791 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413792 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/rtp_engine.c: rtp: Fix case typo in H263+ mime. + http://tools.ietf.org/html/rfc3555#section-4.2.6 says the + canonical mime subtype is "H263-1998", not "h263-1998". Original + code was added in r183101 on 2009-03-19 02:26:50 +0100. This + fixes issues with Polycom phones. ASTERISK-23665 #close + ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume + Maudoux, backported by me. Review: + https://reviewboard.asterisk.org/r/3529/ ........ Merged + revisions 413787 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413788 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413789 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-13 00:35 +0000 [r413770-413772] Richard Mudgett + + * configure.ac, channels/sig_pri.c, /, configure, + include/asterisk/autoconfig.h.in: chan_dahdi/sig_pri: Prevent + unnecessary PROGRESS events when overlap dialing is enabled. When + overlap dialing is enabled, the lack of inband audio available + information in the SETUP_ACKNOWLEDGE events causes an + interoperability problem with SIP. sig_pri doesn't know if there + is dialtone present when a SETUP_ACKNOWLEDGE is received so it + assumes it is there and posts an AST_CONTROL_PROGRESS frame. The + SIP channel driver then sends out a 183 Session Progress and + blocks the desired 180 Ringing message when the ALERTING message + comes in. * Made the configure script detect if the installed + version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * + Using the new API, made generate an AST_CONTROL_PROGRESS frame on + an incoming SETUP_ACKNOWLEDGE message when the message indicates + inband audio is present instead of assuming that dialtone is + present. * Using the new API, made SETUP_ACKNOWLEDGE send out an + inband audio available indication only if dialtone is expected. + The change also makes the fallback behaviour of sending the + PROGRESS message better by sending it only if dialtone is + expected. * Changed receiving a PROCEEDING message to not + generate an AST_CONTROL_PROGRESS frame if the progress indication + ie indicates non-end-to-end-ISDN. This helps interoperability + with SIP. * Changed sending a PROCEEDING message in response to + an AST_CONTROL_PROCEEDING frame to not indicate inband audio + available. It was silly to do so anyway because the channel + driver doesn't know if inband audio is even available. This helps + interoperability with SIP. This patch and a corresponding change + in libpri work together to allow Asterisk to control the inband + audio available progress indication ie on the SETUP_ACKNOWLEDGE + message when dialtone is present. AST-1338 #close Reported by: + Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ + ........ Merged revisions 413714 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413765 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413771 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/sig_pri.c: Fix compiler warning from GCC 4.10 fixup. + ........ Merged revisions 413766 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-12 22:33 +0000 [r413713] Jonathan Rose + + * apps/app_chanspy.c, /: app_chanspy: Fix a test that was failing + on account of r413551 ASTERISK-23381 #close ASTERISK-23381 + #comment Reported by: Robert Moss Review: + https://reviewboard.asterisk.org/r/3505/ ........ Merged + revisions 413710 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413712 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-11 02:09 +0000 [r413651-413682] Joshua Colp + + * main/bridge_basic.c, include/asterisk/channel.h, + bridges/bridge_native_rtp.c, include/asterisk/framehook.h, + main/channel.c, /, main/framehook.c: framehooks: Add callback for + determining if a hook is consuming frames of a specific type. In + the past framehooks have had no capability to determine what + frame types a hook is actually interested in consuming. This has + meant that code has had to assume they want all frames, thus + preventing native bridging. This change adds a callback which + allows a framehook to be queried for whether it is consuming a + frame of a specific type. The native RTP bridging module has also + been updated to take advantange of this, allowing native bridging + to occur when previously it would not. ASTERISK-23497 #comment + Reported by: Etienne Lessard ASTERISK-23497 #close Review: + https://reviewboard.asterisk.org/r/3522/ ........ Merged + revisions 413681 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/channel.h, bridges/bridge_native_rtp.c, + include/asterisk/framehook.h, main/channel.c, /, + main/framehook.c, main/bridge_basic.c: Undoing framehook support. + Issues were uncovered by Bamboo. + + * /, main/framehook.c, main/bridge_basic.c, + include/asterisk/channel.h, bridges/bridge_native_rtp.c, + include/asterisk/framehook.h, main/channel.c: framehooks: Add + callback for determining if a hook is consuming frames of a + specific type. In the past framehooks have had no capability to + determine what frame types a hook is actually interested in + consuming. This has meant that code has had to assume they want + all frames, thus preventing native bridging. This change adds a + callback which allows a framehook to be queried for whether it is + consuming a frame of a specific type. The native RTP bridging + module has also been updated to take advantange of this, allowing + native bridging to occur when previously it would not. + ASTERISK-23497 #comment Reported by: Etienne Lessard + ASTERISK-23497 #close Review: + https://reviewboard.asterisk.org/r/3522/ ........ Merged + revisions 413650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-09 23:18 +0000 [r413589-413599] Kinsey Moore + + * /, funcs/func_env.c: Fix 32bit build for func_env ........ Merged + revisions 413592 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413595 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413597 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_festival.c, pbx/dundi-parser.c, apps/app_getcpeid.c, + main/netsock.c, funcs/func_channel.c, main/audiohook.c, + pbx/pbx_config.c, res/res_pjsip_registrar.c, main/xmldoc.c, + channels/iax2/firmware.c, apps/app_voicemail.c, main/format.c, + cel/cel_pgsql.c, main/rtp_engine.c, main/parking.c, + main/bridge.c, res/res_jabber.c, res/res_http_websocket.c, + main/config.c, res/res_format_attr_opus.c, main/loader.c, + res/parking/parking_bridge.c, main/cdr.c, main/manager.c, + include/asterisk/astobj.h, main/bucket.c, apps/app_dumpchan.c, + main/app.c, res/res_pjsip/config_transport.c, + res/res_pjsip_refer.c, channels/chan_mgcp.c, + res/res_rtp_asterisk.c, main/slinfactory.c, main/core_unreal.c, + res/res_pjsip_sdp_rtp.c, res/res_crypto.c, main/acl.c, + channels/sig_pri.c, res/res_monitor.c, res/res_srtp.c, + main/data.c, res/res_corosync.c, channels/sip/config_parser.c, + res/res_fax_spandsp.c, apps/app_stack.c, main/asterisk.c, + main/udptl.c, res/res_sorcery_config.c, main/security_events.c, + res/res_timing_dahdi.c, res/res_pjsip_t38.c, + res/res_musiconhold.c, main/taskprocessor.c, + res/res_format_attr_h263.c, res/res_xmpp.c, res/res_pktccops.c, + funcs/func_hangupcause.c, channels/chan_phone.c, + main/manager_bridges.c, cel/cel_odbc.c, channels/chan_skinny.c, + channels/chan_motif.c, res/res_agi.c, main/logger.c, + funcs/func_srv.c, channels/chan_alsa.c, apps/app_confbridge.c, + res/res_pjsip_pubsub.c, channels/sip/include/sip.h, main/sched.c, + apps/app_adsiprog.c, main/pbx.c, channels/chan_sip.c, + res/res_fax.c, main/aoc.c, res/res_calendar_ews.c, + res/parking/parking_bridge_features.c, channels/iax2/parser.c, + main/callerid.c, main/file.c, + res/res_pjsip/pjsip_configuration.c, main/adsi.c, + main/config_options.c, pbx/pbx_dundi.c, funcs/func_iconv.c, + main/bridge_channel.c, res/res_odbc.c, channels/chan_pjsip.c, + res/parking/parking_manager.c, res/res_calendar.c, /, + funcs/func_sysinfo.c, main/utils.c, cdr/cdr_adaptive_odbc.c, + res/res_calendar_caldav.c, res/res_stasis_snoop.c, + res/res_format_attr_h264.c, main/channel.c, res/ael/pval.c, + res/res_ari_model.c, channels/chan_dahdi.c, + channels/sig_analog.c, funcs/func_frame_trace.c, + res/res_format_attr_silk.c, main/manager_channels.c, + apps/app_dial.c, res/res_calendar_icalendar.c, main/translate.c, + apps/app_queue.c, channels/chan_jingle.c, res/res_stun_monitor.c, + main/abstract_jb.c, res/res_stasis_recording.c, apps/app_sms.c, + main/event.c, apps/app_verbose.c, main/dsp.c, + channels/chan_unistim.c, main/frame.c, res/res_stasis_playback.c, + main/ccss.c, funcs/func_env.c, main/devicestate.c, + bridges/bridge_softmix.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/enum.c, main/cli.c, + res/res_format_attr_celt.c, apps/confbridge/conf_config_parser.c, + main/io.c, channels/pjsip/dialplan_functions.c, + res/res_config_odbc.c, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, formats/format_pcm.c, + apps/app_minivm.c, main/stdtime/localtime.c, main/stun.c: Allow + Asterisk to compile under GCC 4.10 This resolves a large number + of compiler warnings from GCC 4.10 which cause the build to fail + under dev mode. The vast majority are signed/unsigned mismatches + in printf-style format strings. ........ Merged revisions 413586 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 413587 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413588 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-09 18:15 +0000 [r413572] Richard Mudgett + + * main/http.c: http.c: Remove dead code. + +2014-05-09 17:03 +0000 [r413557] Jonathan Rose + + * apps/app_chanspy.c, /: app_chanspy: Fix a bug where Barge mode + could fail If the barge audiohook was attached prior to the spyee + and its peer actually being bridged, the audiohook would not be + applied and the connected peer would not be able to hear audio + from the spy when the spy is in barge mode. (closes issue + ASTERISK-23381) Reported by: Robert Moss Review: + https://reviewboard.asterisk.org/r/3505/ ........ Merged + revisions 413551 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413556 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-08 00:36 +0000 [r413488] Joshua Colp + + * apps/app_queue.c, main/manager.c, /: app_queue: Extend + documentation for various Manager actions and events. ........ + Merged revisions 413485 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413486 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413487 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-07 21:58 +0000 [r413469] Mark Michelson + + * funcs/func_presencestate.c: Ensure that presence state is decoded + properly on Asterisk startup. The CustomPresence provider + callback will automatically base64 decode stored data if the 'e' + option was present when the state was set. However, since the + provider callback was bypassed on Asterisk startup, encoded + presence subtypes and messages were being sent instead. This fix + makes it so the provider callback is always used when providing + presence state updates. + +2014-05-07 20:59 +0000 [r413453-413455] Richard Mudgett + + * apps/app_confbridge.c, /: app_confbridge: Fixed "CBAnn" channels + not going away. Fixed a ref leak in conf_handle_talker_cb() + everytime the conference bridge was found to report a channel's + talker status change. The resulting leak caused the "CBAnn" + channels and the conference bridge to never be destroyed. Thanks + to Richard Kenner on the asterisk-user's list for locating the + problem. Reported by: Richard Kenner ........ Merged revisions + 413454 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_confbridge.c, /: app_confbridge: Fix ref leak in CLI + "confbridge kick" command. Fixed ref leak in the CLI "confbridge + kick" command when the channel to be kicked was not in the + conference. ........ Merged revisions 413451 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413452 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-07 17:56 +0000 [r413307-413399] Mark Michelson + + * res/res_config_odbc.c, /: Fix encoding of custom prepare extra + data. Patches: res_config_odbc-take2.patch by John Hardin + (License #6512) ........ Merged revisions 413396 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413397 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/presence_xml.c, /, + res/res_pjsip_pidf_digium_body_supplement.c: Improve XML + sanitization in NOTIFYs, especially for presence subtypes and + messages. Embedded carriage return line feed combinations may + appear in presence subtypes and messages since they may be + derived from user input in an instant messenger client. As such, + they need to be properly escaped so that XML parsers do not vomit + when the messages are received. ........ Merged revisions 413372 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_registrar.c, /: Check for an act on failures to + update contacts during registration. There was an underlying + issue in a realtime backend where database updates would fail. + Since we were not checking for failure, we would end up in a + strange state where the old database entry was still present but + Asterisk thought that it had been updated. Now when an entry + fails to update, we print a warning and delete the old contact + from sorcery so there is no mismatch between foreground and + backend state. Patches: res_pjsip_registrar.patch by John Hardin + (License #6512) ........ Merged revisions 413358 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_config_odbc.c, /: Ensure that all parts of SQL UPDATEs + and DELETEs are encoded. Patches: res_config_odbc.patch by John + Hardin (License #6512) ........ Merged revisions 413304 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413305 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-02 20:28 +0000 [r413227-413263] Mark Michelson + + * /, res/res_config_odbc.c: Prevent crashes in res_config_odbc due + to uninitialized string fields. Patches: odbc-crash.patch by John + Hardin (License #6512) ........ Merged revisions 413241 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413251 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413258 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_config_pgsql.c, /: Return the number of rows affected by + a SQL insert, rather than an object ID. The realtime API + specifies that the store callback is supposed to return the + number of rows affected. res_config_pgsql was instead returning + an Oid cast as an int, which during any nominal execution would + be cast to 0. Returning 0 when more than 0 rows were inserted + causes problems to the function's callers. To give an idea of how + strange code can be, this is the necessary code change to fix a + device state issue reported against chan_pjsip in Asterisk 12+. + The issue was that the registrar would attempt to insert contacts + into the database. Because of the 0 return from res_config_pgsql, + the registrar would think that the contact was not successfully + inserted, even though it actually was. As such, even though the + contact was query-able and it was possible to call the endpoint, + Asterisk would "think" the endpoint was unregistered, meaning it + would report the device state as UNAVAILABLE instead of + NOT_INUSE. The necessary fix applies to all versions of Asterisk, + so even though the bug reported only applies to Asterisk 12+, the + code correction is being inserted into 1.8+. Closes issue + ASTERISK-23707 Reported by Mark Michelson ........ Merged + revisions 413224 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 413225 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-02 16:39 +0000 [r413211] Richard Mudgett + + * UPGRADE.txt, res/res_pjsip_refer.c, /, channels/chan_sip.c: + res_pjsip_refer: Add Referred-By header on INVITE for blind + transfers. Per rfc3892, the Referred-By header in a REFER must be + copied into the referenced request (IE. The outgoing INVITE to + the transfer target). * Automatically put the Referred-By header + in the outgoing INVITE message if the SIPREFERREDBYHDR channel + variable is defined with a value. * Made + chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance + so chan_pjsip has a better chance to interoperate. * Fixed + refer_blind_callback() and refer_incoming_refer_request() to not + modify the data in the pointer returned by + pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data + since the calling routine doesn't own the buffer. ASTERISK-23501 + #close Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/3514/ ........ Merged + revisions 413210 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-02 16:06 +0000 [r413197] Jonathan Rose + + * res/parking/res_parking.h, /, CHANGES, + res/parking/parking_bridge_features.c, + res/parking/parking_manager.c: Parking: Add 'AnnounceChannel' + argument to manager action 'Park' (closes ASTERISK-23397) + Reported by: Denis Review: + https://reviewboard.asterisk.org/r/3446/ ........ Merged + revisions 413196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-01 16:21 +0000 [r413174-413183] Mark Michelson + + * funcs/func_presencestate.c: Make behavior of the PRESENCE_STATE + 'e' option more consistent. When writing presence state, if 'e' + is specified, then the presence state will be stored in the astdb + encoded. However, consumers of presence state events or those + that query for the presence state will be given decoded + information. If base64 encoding is desired for consumers, then + the information can be base64-encoded manually and the 'e' option + can be omitted. closes issue ASTERISK-23671 Reported by Mark + Michelson Review: https://reviewboard.asterisk.org/r/3482 + + * res/res_pjsip_exten_state.c, /: Remove unnecessary repetition + checks from res_pjsip_exten_state The PBX core already takes care + of ensuring that repeated state changes are not communicated to + exten state consumers. Because the check in res_pjsip_exten_state + was incomplete, it was causing valid presence state changes not + to be sent out. For instance, if the presence state did not + change but the message or subtype did, then no presence-related + NOTIFY request would be sent out. closes issue ASTERISK-23672 + Reported by Mark Michelson ........ Merged revisions 413173 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-05-01 12:31 +0000 [r413160] Joshua Colp + + * res/res_pjsip/config_transport.c, /: res_pjsip: Add the ability + to configure ciphers based on name. Previously this code would + only accept the OpenSSL identifier instead of the documented + name. ASTERISK-23498 #close ASTERISK-23498 #comment Reported by: + Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/ + ........ Merged revisions 413159 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 21:03 +0000 [r413144] Richard Mudgett + + * main/message.c, /, channels/chan_sip.c, + include/asterisk/message.h, res/res_pjsip_messaging.c: + chan_sip.c: Fixed off-nominal message iterator ref count and + alloc fail issues. * Fixed early exit in sip_msg_send() not + destroying the message iterator. * Made + ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() + tolerant of a NULL iter parameter in case + ast_msg_var_iterator_init() fails. * Made + ast_msg_var_iterator_destroy() clean up any current message data + ref. * Made struct ast_msg_var_iterator, + ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), + ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() + use iter instead of i. * Eliminated RAII_VAR usage in + res_pjsip_messaging.c:vars_to_headers(). ........ Merged + revisions 413139 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413142 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 20:39 +0000 [r413141] Joshua Colp + + * /, channels/chan_pjsip.c: chan_pjsip: Fix deadlock when + retrieving call-id of channel. If a task was in-flight which + required the channel or bridge lock it was possible for the + synchronous task retrieving the call-id to deadlock as it holds + those locks. After discussing with Mark Michelson the synchronous + task was removed and the call-id accessed directly. This should + be safe as each object involved is guaranteed to exist and the + call-id will never change. ........ Merged revisions 413140 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 13:08 +0000 [r413125] Kinsey Moore + + * res/res_http_websocket.c, /: Websocket: Add session locking and + delay close This resolves a race condition where data could be + written to a NULL FILE pointer causing a crash as a websocket + connection was in the process of shutting down by adding locking + to websocket session writes and by deferring session teardown + until session destruction. (closes issue ASTERISK-23605) Review: + https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan + ........ Merged revisions 413123 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413124 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-30 12:42 +0000 [r413118-413122] Joshua Colp + + * /, res/stasis/control.c: res_stasis: Add progress indications to + operations which perform media. This change fixes operations + which did not account for the fact that they may be executed on + channels which have not been answered. These operations will now + indicate progress when invoked. ASTERISK-23560 #close + ASTERISk-23560 #comment Reported by: Jan Svoboda Review: + https://reviewboard.asterisk.org/r/3495/ ........ Merged + revisions 413121 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix issue where + sending a hold SDP twice could cause an unhold. This change fixes + a bug where if an SDP with media address and sendonly was + received twice the underlying call would go off hold, instead of + remaining on hold. This occured because the code did not properly + take into account that the SDP may contain both a valid media + address and the sendonly attribute. The code now examines the + sendonly attribute and media address first, so if the SDP is + received again no change will occur. ASTERISK-23558 #comment + Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/3472/ ........ Merged + revisions 413119 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip: + Add support for picking up calls in the configured pickup group. + AST-1363 Review: https://reviewboard.asterisk.org/r/3478/ + ........ Merged revisions 413117 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-29 15:10 +0000 [r413103] George Joseph + + * /, include/asterisk/spinlock.h: Add "destroy" implementation for + spinlock. The original commit for spinlock was missing "destroy" + implementations. Most of them are no-ops but phtread_spin and + pthread_mutex do need their locks destroyed. ........ Merged + revisions 413102 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-29 11:27 +0000 [r413089] Joshua Colp + + * channels/chan_pjsip.c, /: chan_pjsip: Implement core ability to + get Call-ID of a channel. This changes implement the + "get_pvt_uniqueid" which is used to return the technology + specific unique identifier. In the case of SIP this is the + Call-ID of the dialog. Review: + https://reviewboard.asterisk.org/r/3480/ ........ Merged + revisions 413088 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-28 20:07 +0000 [r413074] Kinsey Moore + + * /, main/bridge.c, main/bridge_basic.c: Bridging: Don't lock NULL + bridges When bridge locking was added for bridge snapshot + creation, some locations where bridge locking was added were not + guaranteed to actually have a bridge and locking NULL AO2 objects + tends to cause segfaults. This ensures that NULL bridges aren't + locked. ........ Merged revisions 413073 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-28 14:40 +0000 [r413060] Mark Michelson + + * res/res_manager_presencestate.c (added), main/devicestate.c, + CHANGES, main/presencestate.c, res/res_manager_devicestate.c + (added): Add DeviceStateChanged and PresenceStateChanged AMI + events. These events are controlled by two new modules, + res_manager_devicestate and res_manager_presencestate. Review: + https://reviewboard.asterisk.org/r/3417 + +2014-04-28 07:43 +0000 [r413048] Igor Goncharovskiy + + * UPGRADE.txt, CHANGES, channels/chan_unistim.c, + configs/unistim.conf.sample: Introducing changes proposed to + chan_unistim driver: 1) Added the unistim.conf variable + dtmf_duration which can select the DTMF playback duration from + 0ms to 150ms (0 is off and is the new default) 2) Enabled the + transmission of month names, which are sent with the date and + changed the dateformat variable to accept the values 0-3 as per + the UNISTIM standard (2 & 3 match the previous 1 & 2 formats). 3) + Enabled the "Mute" packet so muting microphone works as expected + and microphone muted for all calls while LED light on 4) Changed + Duree to Timer on i2004 display (closes issue ASTERISK-23592) + +2014-04-27 19:29 +0000 [r413036] Olle Johansson + + * main/tcptls.c: tcptls.c : Log errors as ERROR, not warning or + something else. + +2014-04-25 19:26 +0000 [r413012] Matthew Jordan + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Add support for DTLS + handshake retransmissions On congested networks, it is possible + for the DTLS handshake messages to get lost. This patch adds a + timer to res_rtp_asterisk that will periodically check to see if + the handshake has succeeded. If not, it will retransmit the DTLS + handshake. Review: https://reviewboard.asterisk.org/r/3337 + ASTERISK-23649 #close Reported by: Nitesh Bansal patches: + dtls_retransmission.patch uploaded by Nitesh Bansal (License + 6418) ........ Merged revisions 413008 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 413009 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-24 14:37 +0000 [r412993] Kevin Harwell + + * /, + contrib/ast-db-manage/config/versions/e96a0b8071c_increase_pjsip_column_size.py + (added): pjsip realtime: increase the size of some columns The + string lengths on certain columns created through alembic for + PJSIP were too short. For instance, columns containing URIs are + currently set to 40 characters, but this can be too small and + result in truncated values. Added an alembic migration script + that increases the size of these columns and a few others to 255. + ASTERISK-23639 #close Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3475/ ........ Merged + revisions 412992 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-23 20:13 +0000 [r412977] George Joseph + + * include/asterisk/spinlock.h (added), /, configure, + include/asterisk/autoconfig.h.in, configure.ac: This patch adds + support for spinlocks in Asterisk. There are cases in Asterisk + where it might be desirable to lock a short critical code section + but not incur the context switch and yield penalty of a mutex or + rwlock. The primary spinlock implementations execute exclusively + in userspace and therefore don't incur those penalties. Spinlocks + are NOT meant to be a general replacement for mutexes. They + should be used only for protecting short blocks of critical code + such as simple compares and assignments. Operations that may + block, hold a lock, or cause the thread to give up it's timeslice + should NEVER be attempted in a spinlock. The first use case for + spinlocks is in astobj2 - internal_ao2_ref. Currently the + manipulation of the reference counter is done with an + ast_atomic_fetchadd_int which works fine. When weak reference + containers are introduced however, there's an additional + comparison and assignment that'll need to be done while the lock + is held. A mutex would be way too expensive here, hence the + spinlock. Given that lock contention in this situation would be + infrequent, the overhead of the spinlock is only a few more + machine instructions than the current ast_atomic_fetchadd_int + call. ASTERISK-23553 #close Review: + https://reviewboard.asterisk.org/r/3405/ ........ Merged + revisions 412976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-23 18:03 +0000 [r412925] Richard Mudgett + + * /, main/http.c: http: Fix spurious ERROR message in responses + with no content. Backport -r411687 and fix the fix because + content_length is the length of out plus the length of the file + controlled by fd. When a response has an out content length of 0, + fwrite would be called to write a buffer with no data in it. This + resulted in the following classic error message: [Apr 3 11:49:17] + ERROR[26421] http.c: fwrite() failed: Success This patch makes it + so that we only attempt to write the content of out if the out + string is non-zero. ........ Merged revisions 412922 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412923 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-23 15:02 +0000 [r412910] Russell Bryant + + * res/res_monitor.c, funcs/func_periodic_hook.exports.in (added), + main/asterisk.dynamics, funcs/func_periodic_hook.c: Fix error + loading res_monitor. For some odd reason, loading app_mixmonitor + was fine, but res_monitor was not. This patch fixes a set of + issues related to func_periodic_hook exporting the beep functions + that gets res_monitor working again. + +2014-04-22 10:09 +0000 [r412883] Joshua Colp + + * /, res/stasis/app.c: res_stasis: Fix crash when handling a failed + blind transfer message. This changes fixes a crash that occurs + when stasis determines if it should send a message out to an + application or not. The code incorrectly assumed that a bridge + snapshot would always be present when in reality for failure + cases it may not be. ASTERISK-23573 #close ........ Merged + revisions 412882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-21 17:56 +0000 [r412759-412824] Jonathan Rose + + * CHANGES, /: chan_sip: trust_id_outbound CHANGES message + improvement (closes issue AST-1301) (closes issue ASTERISK-19465) + Reported by: Krzysztof Chmielewski ........ Merged revisions + 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 412822 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412823 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: chan_sip: Add sendrpid trust options + In r411189, some behavior was changed which made sendrpid + behavior act in a more trusting manner by sending full user data + for peers set with private caller presence in P-Asserted-Identity + headers. Since this changed long time expected behaviors, we + decided to pull that patch when that was pointed out by the + community. Instead, this patch provides a trust_id_outbound + setting which will expose the data per RFC-3325 if set to 'yes' + and simply not send the PAI/RPID headers at all if set to 'no'. + By default trust_id_outbound will be set to 'legacy' which will + preserve the behavior prior to these patches. Extra special + thanks to Walter Doekes for providing advice and feedback. + (closes issue AST-1301) (closes issue ASTERISK-19465) Reported + by: Krzysztof Chmielewski Review: + https://reviewboard.asterisk.org/r/3447/ ........ Merged + revisions 412744 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412746 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412747 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-21 16:16 +0000 [r412729-412750] Kinsey Moore + + * main/http.c, main/manager.c, /: HTTP: Add TCP_NODELAY to accepted + connections This adds the TCP_NODELAY option to accepted + connections on the HTTP server built into Asterisk. This option + disables the Nagle algorithm which controls queueing of outbound + data and in some cases can cause delays on receipt of response by + the client due to how the Nagle algorithm interacts with TCP + delayed ACK. This option is already set on all non-HTTP AMI + connections and this change would cover standard HTTP requests, + manager HTTP connections, and ARI HTTP requests and websockets in + Asterisk 12+ along with any future use of the HTTP server. + Review: https://reviewboard.asterisk.org/r/3466/ ........ Merged + revisions 412745 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412748 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_confbridge.c, /: Confbridge: Fix ConfbridgeKick AMI + documentation This adds documentation for the "all" channel + option for the ConfbridgeKick AMI action and adjusts AMI + responses accordingly. (issue ASTERISK-23282) Reported by: Dorian + Logan ........ Merged revisions 412730 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_confbridge.c: Confbridge: Add references for kick all + option After the ability to kick all attendees from a conference + was added, a rework removed the comment about that feature from + the CLI documentation. This adds that documentation and adds + "all" to the participant tab completion list for the confbridge + kick command. (closes issue ASTERISK-23282) Reported by: Dorian + Logan ........ Merged revisions 412728 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-21 08:36 +0000 [r412714] Igor Goncharovskiy + + * /, channels/chan_unistim.c: Fix wrong dialtone. The "modulation" + should not be referenced for tone+tone as it refers to the on-off + characteristic - this often resulted in a single tone rather than + the multitone as in the UK. ........ Merged revisions 412712 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-19 02:14 +0000 [r412697-412699] Matthew Jordan + + * /, main/asterisk.c: main/asterisk: Fix startup sequence for + realtime features When ASTERISK-23265/ASTERISK-23320 was fixed, + it inadvertently led to realtime features breaking. This was due + to features loading prior to realtime. This patch fixes this by + loading features after loading dynamic modules. ASTERISK-23487 + #close Reported by: Denis Tested by: Denis ........ Merged + revisions 412698 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_sms.c: app_sms: Fix uninitialized values; hangup + channel when REL is sent successfully This patch fixes two issues + in app_sms: (1) Firstly, the 'flags' field on the stack in + sms_exec() is uninitialised, causing it to use the wrong protocol + in some cases. This patch correctly initializes the flags fields. + (2) Secondly, when disconnect supervision is not working or + inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was + failing to terminate the call after it sent the REL(ease) message + and the peer stopped talking to it. This patch fixes the code to + handle the 'bad stop bit' message more gracefully in that case, + and hang up the call. Review: + https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close + Reported by: David Woodhouse patches: asterisk-fix-sms.patch + uploaded by David Woodhouse (License 5754) ........ Merged + revisions 412655 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412656 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412657 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 20:09 +0000 [r412641] Jonathan Rose + + * /, res/ari/resource_bridges.h, res/stasis/control.c, + include/asterisk/stasis_app.h, res/stasis/control.h, + res/ari/resource_channels.c, CHANGES, res/res_stasis.c, + rest-api/api-docs/bridges.json, res/ari/resource_bridges.c, + res/res_ari_bridges.c, res/res_stasis_playback.c: ARI: Make + bridges/{bridgeID}/play queue sound files Previously multiple + play actions against a bridge at one time would cause the sounds + to play simultaneously on the bridge. Now if a sound is already + playing, the play action will queue playback to occur after the + completion of other sounds currently on the queue. (closes issue + ASTERISK-22677) Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/3379/ ........ Merged + revisions 412639 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 17:17 +0000 [r412589] Rusty Newton + + * sounds/sounds.xml, sounds/Makefile, /: sounds: Fix Sounds + Makefile and XML that didn't support new sound prompt sets In + sounds/Makefile 1 Adds and moves some lines necessary for the + en_GB core set. I'm just following how the other sets are defined + here. 2 removes the ES extra sounds related lines as we don't + have ES extra sound sets. In sounds/sounds.xml 3 Adds member + definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in + extra sound sets ASTERISK-23550 #close Review: + https://reviewboard.asterisk.org/r/3464/ ........ Merged + revisions 412586 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412587 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 17:02 +0000 [r412584] Mark Michelson + + * /, res/res_pjsip/location.c: Allow for multiple contacts to be + configured in a single contact= line. This is useful for + configuring multiple permanent contacts for an AOR when using + realtime AORs. Review: https://reviewboard.asterisk.org/r/3462 + ........ Merged revisions 412582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 16:44 +0000 [r412580-412583] Richard Mudgett + + * main/dial.c, main/pbx.c, /, apps/app_originate.c, + include/asterisk/pbx.h: Originated calls: Fix several originate + call problems. * Restore the reason value set by + pbx_outgoing_attempt() to use AST_CONTROL_xxx values as all the + consumers were expecting rather than cause codes. * Fixed the + dial routines to set cause codes for more than just ast_request() + so pbx_outgoing_attempt() reason codes will function. * Fix + inconsistent locked_channel return status in + pbx_outgoing_attempt(). The chanel may not have been locked or + the channel may have been a stale pointer. * Fixed the + OutgoingSpoolFailed channel to run dialplan whenever the dialing + fails for an originate exten and 1 < synchronous. * Fix incorrect + ast_cond_wait() usage in pbx_outgoing_attempt(). Indroduced by + issue ASTERISK-22212 patch. * Made struct pbx_outgoing use the + ao2 lock instead of its own lock for the cond wait mutex. No + sense in having two locks associated with the same struct when + only one is needed. Review: + https://reviewboard.asterisk.org/r/3421/ ........ Merged + revisions 412581 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, apps/app_queue.c, apps/app_dial.c, /: + app_dial and app_queue: Make lock the forwarding channel while + taking the channel snapshot. * Fixed + ast_channel_publish_dial_forward() not locking the forwarded + channel when taking the channel snapshot. * Fixed + app_dial.c:do_forward() using the wrong channel to get the + original call forwarding string. * Removed unnecessary locking + when calling ast_channel_publish_dial() and + ast_channel_publish_dial_forward() in app_dial and app_queue. + Holding channel locks when calling + ast_channel_publish_dial_forward() with a forwarded channel could + result in pausing the system while the stasis bus completes + processsing a forwarded channel subscription. Review: + https://reviewboard.asterisk.org/r/3451/ ........ Merged + revisions 412579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-18 14:25 +0000 [r412566] Kinsey Moore + + * res/ari/ari_websockets.c, res/res_ari.c, main/manager.c, /: ARI: + Add debug logging for events and responses This adds DEBUG level + logging for ARI websocket events and HTTP responses similar to + what is available for AMI. Logging for ARI HTTP requests is + already adequate for debugging purposes. ........ Merged + revisions 412565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-17 22:50 +0000 [r412552] Joshua Colp + + * /, res/res_pjsip/location.c, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + res/res_pjsip_registrar.c: res_pjsip: Handle reloading when + permanent contacts exist and qualify is configured. This change + fixes a problem where permanent contacts being qualified were not + being updated. This was caused by the permanent contacts getting + a uuid and not a known identifier, causing an inability to look + them up when updating in the qualify code. A bug also existed + where the new configuration may not be available immediately when + updating qualifies. (closes issue ASTERISK-23514) Reported by: + Richard Mudgett Review: https://reviewboard.asterisk.org/r/3448/ + ........ Merged revisions 412551 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-17 22:42 +0000 [r412536-412550] Jonathan Rose + + * /, main/app.c: Fix a silly shadowed variable mistake that was + missed from play tones patch ........ Merged revisions 412549 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/ari/resource_bridges.h, main/app.c, + rest-api/api-docs/channels.json, CHANGES, + rest-api/api-docs/bridges.json, res/ari/resource_channels.h, + include/asterisk/app.h, res/res_stasis_playback.c: ARI: Add tones + playback resource Adds a tones URI type to the playback resource. + The tone can be specified by name (from indications.conf) or by a + tone pattern. In addition, tonezone can be specified in the URI + (by appending ;tonezone=). Tones must be stopped manually + in order for a stasis control to move on from playback of the + tone. Tones may be paused, resumed, restarted, and stopped. They + may not be rewound or fast forwarded (tones can't be controlled + in a way that lets you skip around from note to note and pausing + and resuming will also restart the tone from the beginning). + Tests are currently in development for this feature + (https://reviewboard.asterisk.org/r/3428/). (closes issue + ASTERISK-23433) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3427/ ........ Merged + revisions 412535 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-17 20:25 +0000 [r412467-412484] Matthew Jordan + + * channels/chan_oss.c, /, main/Makefile: main/Makefile: Fix build + failure on SmartOS/Illumos/SunOS This patch fixes two issues when + building on SmartOS: - channels/chan_oss.c: it makes sure + soundcard.h is found - main/Makefile: only use + "-Wl,--version-script" when GNU LD is used as the Sun Linker + doesn't support that. Similar checks are already used elswhere in + the Makefile Review: https://reviewboard.asterisk.org/r/3426 + ASTERISK-23576 #close Reported by: Sebastian Wiedenroth patches: + fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597) + ........ Merged revisions 412468 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412483 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sip/include/sip.h, channels/chan_sip.c, CHANGES: + chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL + URIs This patch is a continuation of + https://reviewboard.asterisk.org/r/3349/, committed in r412303. + It resolves a finding oej had that the phone-context be available + in a channel variable separate from SIPDOMAIN. This patch adds + that variable as SIPURIPHONECONTEXT. It also allows a local + number (or global number specified in the TEL URI) to be used to + look up as a peer. (issue ASTERISK-17179) Review: + https://reviewboard.asterisk.org/r/3349/ + +2014-04-17 15:17 +0000 [r412454] Kevin Harwell + + * res/res_pjsip_refer.c, /: res_pjsip_refer: Channel variable + SIPREFERTOHDR not being set during blind transfer The + SIPREFERTOHDR channel variable is not being set on any channel + when performing a blind transfer using PJSIP. The + 'refer->refer_to' was not being set during a blind transfer. + Updated so the 'refer_to' is set to the target uri on a blind + transfer. (closes issue ASTERISK-23502) Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/3445/ ........ Merged + revisions 412453 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-16 19:14 +0000 [r412440] Kinsey Moore + + * /, include/asterisk/stasis_app.h: Stasis: Add a usage note on + stasis_app_get_bridge This function returns an ast_bridge without + a refcount bump and the caller must increment the count if it + intends to hold the pointer. (closes issue ASTERISK-23588) + Review: https://reviewboard.asterisk.org/r/3450/ Reported by: + Matt Jordan ........ Merged revisions 412439 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-15 23:21 +0000 [r412427] Russell Bryant + + * bridges/bridge_builtin_features.c, include/asterisk/monitor.h, + CHANGES, apps/app_queue.c, funcs/func_periodic_hook.c, + apps/app_mixmonitor.c, include/asterisk/beep.h (added), + res/res_monitor.c: (mix)monitor: Add options to enable a periodic + beep Add an option to enable a periodic beep to be played into a + call if it is being recorded. If enabled, it uses the + PERIODIC_HOOK() function internally to play the 'beep' prompt + into the call at a specified interval. This option is provided + for both Monitor() and MixMonitor(). Review: + https://reviewboard.asterisk.org/r/3424/ + +2014-04-15 18:30 +0000 [r412384-412414] Richard Mudgett + + * main/stasis_channels.c, main/features_config.c, + res/res_parking.c, main/rtp_engine.c, /: Eliminate some more + unnecessary RAII_VAR() uses. RAII_VAR() is not a hammer + appropriate to pound all nails. ........ Merged revisions 412413 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /, res/stasis/app.c, res/res_fax.c, + res/res_pjsip/security_events.c, + res/parking/parking_applications.c, channels/chan_oss.c, + main/stasis_bridges.c, res/res_pjsip_session.c, + res/stasis_recording/stored.c, main/cdr.c, res/res_parking.c, + channels/chan_skinny.c, res/res_pjsip/location.c, + res/res_stasis_recording.c, main/stasis_channels.c, + res/ari/resource_channels.c, res/parking/parking_manager.c, + res/ari/resource_recordings.c, res/res_pjsip_refer.c, + res/res_ari.c, main/pbx.c: Remove unused RAII_VAR() declarations. + * Remove unused RAII_VAR() declarations. The compiler cannot + catch these because the cleanup function "references" the unused + variable. Some actually allocated and released resources that + were never used. * Fixed some whitespace issues in + stasis_bridges.c. ........ Merged revisions 412399 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/rtp_engine.h, main/rtp_engine.c, /, + channels/chan_sip.c: chan_sip.c: Fix channel staging assertion + failure. The failing assertion ensures that the final snapshot + gets generated so CDR records can get finalized. The only place + where a channel staging snapshot flag could be left set is in + chan_sip.c:handle_request_bye(). The function could return before + clearing the flag because the channel could dissappear while the + function had to have the channel unlocked. * Fixed + handle_request_bye() channel snapshot staging coverage area to + not have a return in the middle of it and be unable to clear the + staging flag. * Pushed the channel snapshot staging coverage area + into ast_rtp_instance_set_stats_vars() to ensure that the staging + is not interrutped. * Made callers of + ast_rtp_instance_set_stats_vars() not call it with any channels + or channel driver private locks held to eliminate the deadlock + potential. The callers must hold references to the passed in + channel and rtp objects. * Eliminated sip_hangup() trying to get + the bridge peer. It is futile at this point because the channel + could never be in a bridge. Review: + https://reviewboard.asterisk.org/r/3431/ ........ Merged + revisions 412385 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip.c: Moved some sip_pvt unrefs + after their last use. * Moved sip_pvt unref in ast_hangup() and + handle_request_do() to the end of the function. The unref needs + to happen after the last use of the pointer. ........ Merged + revisions 412348 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-15 16:13 +0000 [r412331] Jonathan Rose + + * configs/sip.conf.sample, /, channels/chan_sip.c: Reverting + r411189 so that it can be put up for public review --- r411189 | + jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines + chan_sip: Send real CallerID information with + P-Assserted-Identity (RFC-3325) Prior to this patch, the + P-Asserted-Identity header would include anonymous caller id + information which seems to go against the point of the + P-Asserted-Identity header. Now the real caller ID information + will be included in this header. Also, no privacy header would be + included. This patch adds 'Privacy: id' to outgoing SIP messages + that include the P-Asserted-Identity header. (closes issue + AST-1301) --- ........ Merged revisions 412328 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412329 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412330 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-14 15:54 +0000 [r412307] Corey Farrell + + * main/autoservice.c, /: autoservice: fix reference leak of logger + callid. autoservice acquires a local reference to the logger + callid of each channel in a loop. This local reference was not + released, causing the callid of every channel in autoservice to + leak. This change moves the callid unref inside the loop. + ASTERISK-23616 #close Reported by: ibercom ........ Merged + revisions 412305 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-12 02:27 +0000 [r412292] Matthew Jordan + + * channels/sip/reqresp_parser.c, CHANGES, channels/chan_sip.c: + chan_sip: Support RFC-3966 TEL URIs in inbound INVITE requests + This patch adds support for handling TEL URIs in inbound INVITE + requests. This includes the Request URI and the From URI. The + number specified in the Request URI will be the destination of + the inbound channel in the dialplan. The phone-context specified + in the Request URI will be stored in the TELPHONECONTEXT channel + variable. Review: https://reviewboard.asterisk.org/r/3349 + ASTERISK-17179 #close Reported by: Geert Van Pamel Tested by: + Geert Van Pamel patches: + asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van + Pamel (License 6140) + asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by + Geert Van Pamel (License 6140) + +2014-04-12 01:35 +0000 [r412279-412280] Russell Bryant + + * funcs/func_periodic_hook.c: func_periodic_hook: move module ref + The previous code left one error path where the module would be + unref'd twice instead of once. It was done once in the error + handling block, and again inside of datastore destruction. Now + the module ref is only released in the datastore destructor and + only acquired when the datastore has been successfully allocated. + + * funcs/func_periodic_hook.c: func_periodic_hook: add module ref + counting This module lacked necessary module ref count + incrementing and decrementing when used. This patch adds it. + There's already a datastore used, so doing the ref counting along + with the lifetime of the datastore provides a convenient place to + do it. + +2014-04-11 21:43 +0000 [r412213-412228] Richard Mudgett + + * apps/app_stack.c, /: app_stack: Add missing unlock in off-nominal + path of STACK_PEEK function. ASTERISK-23620 #close Reported by: + Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch + (license #5021) patch uploaded by Bradley Watkins ........ Merged + revisions 412225 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412226 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412227 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * utils/Makefile, utils: utils dir: Remove no longer needed traces + of refcounter except in the clean make target. * Removed no + longer needed files from the svn:ignore property to make them + visible. + +2014-04-11 12:43 +0000 [r412194] Kinsey Moore + + * /, main/bridge.c, main/bridge_basic.c, + include/asterisk/stasis_bridges.h, tests/test_cel.c, + apps/app_confbridge.c, res/ari/resource_bridges.c: bridging: + Ensure locking during snapshot creation While the vast majority + of bridge snapshot creation is locked properly, there are + currently some instances that are not. This adds the missing + locking to ensure bridge state is not malleable during snapshot + creation. (closes issue ASTERISK-22904) Review: + https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan + ........ Merged revisions 412193 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-11 08:28 +0000 [r412168-412180] Olle Johansson + + * main/audiohook.c: Formatting: Remove invisible characters + + * main/audiohook.c: Formatting only. + +2014-04-11 02:59 +0000 [r412154] Matthew Jordan + + * main/astobj2.c, contrib/scripts/refcounter.py (added), + main/asterisk.c, utils/refcounter.c (removed), + build_tools/cflags.xml, utils/utils.xml, /, channels/chan_sip.c, + channels/sip/security_events.c, include/asterisk/astobj2.h, + UPGRADE.txt: main/astobj2: Make REF_DEBUG a menuselect item; + improve REF_DEBUG output This patch does the following: (1) It + makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables + REF_DEBUG globally throughout Asterisk. (2) The ref debug log + file is now created in the AST_LOG_DIR directory. Every run will + now blow away the previous run (as large ref files sometimes + caused issues). We now also no longer open/close the file on each + write, instead relying on fflush to make sure data gets written + to the file (in case the ao2 call being performed is about to + cause a crash) (3) It goes with a comma delineated format for the + ref debug file. This makes parsing much easier. This also now + includes the thread ID of the thread that caused ref change. (4) + A new python script instead for refcounting has been added in the + contrib/scripts folder. (5) The old refcounter implementation in + utils/ has been removed. Review: + https://reviewboard.asterisk.org/r/3377/ ........ Merged + revisions 412114 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 412115 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 412153 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-11 01:12 +0000 [r412102] Russell Bryant + + * res/res_monitor.c: monitor: use app options parsing helper code + This app is pretty ancient, so it was never converted to use the + option parsing helper code. I'd like to add an option to this app + that takes an argument, and that's a pain to do when not using + this helper, so start by doing this conversion. Review: + https://reviewboard.asterisk.org/r/3429/ + +2014-04-10 21:28 +0000 [r412089] Matthew Jordan + + * /, res/res_hep_pjsip.c: res_hep_pjsip: Use the channel name + instead of the call ID when it is available During discussions + with Alexandr Dubovikov at Kamailio World, it became apparent + that while the SIP call ID is a useful identifier prior to an + Asterisk channel being created, it is far more preferable to use + the channel name (or some channel based identifier) when the + channel is available. Homer is smart enough to tie the various + messages together. This patch opts to use the channel name when + it is available, falling back to the call ID otherwise. ........ + Merged revisions 412088 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-10 21:10 +0000 [r412075] Kevin Harwell + + * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Set the body + generation result to 0 for a valid path The result of the + "ast_sip_pubsub_generate_body_content" was not set/initialized. + Consequently, the nominal path potentially returned an invalid + value, thus not sending mwi notifications. ........ Merged + revisions 412074 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-09 21:43 +0000 [r412050] Mark Michelson + + * /, CHANGES, apps/app_mixmonitor.c: Add a Command header to the + AMI Mixmonitor action. This fixes a parsing error that occurred + during the processing of the AMI action. The error did not result + in MixMonitor itself misbehaving, but it could result in the AMI + response not giving correct information back. The new header + allows for one to specify a post-process command to run when + recording finishes. Previously, in order to do this, the + post-process command would have to be placed at the end of the + Options: header. Patches: mixmonitor_command_2.patch by jhardin + (License #6512) ........ Merged revisions 412048 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-09 18:17 +0000 [r412035] Kinsey Moore + + * /, res/res_stasis_answer.c: res_stasis_answer: Add missing + newlines ........ Merged revisions 412034 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-08 21:25 +0000 [r411946-411990] Richard Mudgett + + * /, main/asterisk.c: Internal timing: Add notice that the -I and + internal_timing option are no longer needed. Add notice messages + during execution that the -I command line option and the + astersik.conf internal_timing option are no longer needed. The + internal timing functionality is now always enabled if there is a + timing module loaded. NOTE: Since the command line options and + the asterisk.conf config file are processed before the logging + system is initialized, the messages are output to stderr. Change + requested as a result of asterisk-dev list comments about the + commit for ASTERISK-22846 that removed the -I and internal_timing + options. Review: https://reviewboard.asterisk.org/r/3423/ + ........ Merged revisions 411964 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411974 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411985 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, /: config: Fix CB_ADD_LEN() to work as originally + intended. Fix a long standing bug in CB_ADD_LEN() behaving like + CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes + ........ Merged revisions 411960 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411961 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411962 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix + confbridge.conf dsp_talking_threshold option setting wrong + parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported + by: John Knott ........ Merged revisions 411944 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411945 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-08 14:49 +0000 [r411928] Joshua Colp + + * /, res/res_pjsip.c: res_pjsip: Ignore explicit transport + configuration if a WebSocket transport is specified. This change + makes it so if a transport is configured on an endpoint that is a + WebSocket type the option will be ignored. In practice this is + fine because the WebSocket transport can not create outgoing + connections, it can only reuse existing ones. By ignoring the + option the existing PJSIP logic for using the existing connection + will be invoked and stuff will proceed. (closes issue + ASTERISK-23584) Reported by: Rusty Newton ........ Merged + revisions 411927 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-08 00:26 +0000 [r411897] Russell Bryant + + * funcs/func_periodic_hook.c: func_periodic_hook: List more modules + as dependencies This module makes use of some existing Asterisk + components. app_chanspy was already listed as a dependency. There + are a few function modules used, as well, so list them. + +2014-04-07 20:41 +0000 [r411884] Kinsey Moore + + * /, res/res_pjsip_pubsub.c: PJSIP: Ensure test event has new state + The change that fixed the pubsub test event's use of a dangling + pointer also changed when it was processed relative to the pjsip + subscription state change processing. This change corrects the + order of events while holding a reference to the pointer that was + previously dangling. ........ Merged revisions 411883 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-07 16:15 +0000 [r411870] Jonathan Rose + + * main/manager_channels.c, /: AGI/Manager: Prevent multiple + NewExten events during AGI application changes AGI applications + would trigger NewExten events every time the state of the AGI + application changed. This has historically not been the behavior + and this behavior was introduced with a CDR patch. This patch + corrects that. (closes issue ASTERISK-23390) Reported by: + Benjamin Keith Ford Review: + https://reviewboard.asterisk.org/r/3406/ ........ Merged + revisions 411868 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-07 14:57 +0000 [r411812] Walter Doekes + + * apps/app_queue.c, /: app_queue: Re-add HoldTime to + QueueCallerAbandon event (simple typo during ast12 refactor). + Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........ + Merged revisions 411811 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-07 14:29 +0000 [r411791-411806] Kinsey Moore + + * /, res/res_stasis.c: Stasis: Fix Stasis() bridge refcount issue + The Stasis() dialplan application monitors what bridge a channel + is in and so necessarily holds on to a bridge pointer. This + change ensures that it also holds on to a reference for that + bridge to prevent the bridge pointer from becoming a dangling + pointer. ........ Merged revisions 411804 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pubsub.c, /: PJSIP: Fix crash introduced in r411671 + The test event introduced in revision 411671 uses a dangling + pointer to access information about pubsub state changes. This + moves the event to within the lifetime of the pointer. ........ + Merged revisions 411790 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-05 13:06 +0000 [r411768] Russell Bryant + + * CHANGES, funcs/func_periodic_hook.c (added): func_periodic_hook: + New function for periodic hooks. This commit introduces a new + dialplan function, PERIODIC_HOOK(). It allows you run to a + dialplan hook on a channel periodically. The original use case + that inspired this was the ability to play a beep periodically + into a call being recorded. The implementation is much more + generic though and could be used for many other things. The + implementation makes heavy use of existing Asterisk components. + It uses a combination of Local channels and ChanSpy() to run some + custom dialplan and inject any audio it generates into an active + call. The other important bit of the implementation is how it + figures out when to trigger the beep playback. This + implementation uses the audiohook API, even though it's not + actually touching the audio in any way. It's a convenient way to + get a callback and check if it's time to kick off another beep. + It would be nice if this was timer event based instead of polling + based, but unfortunately I don't see a way to do it that won't + interfere with other things. Review: + https://reviewboard.asterisk.org/r/3362/ + +2014-04-04 19:19 +0000 [r411702-411724] Richard Mudgett + + * include/asterisk/options.h, main/asterisk.c, main/channel.c, /, + channels/chan_sip.c, configs/asterisk.conf.sample, UPGRADE.txt, + include/asterisk/channel.h, utils/extconf.c: internal_timing: + Remove the option and always make it enabled if a timing module + is loaded. The masquerade supertest frequently fails because + either the local channel chain doesn't completely optimize out or + the DTMF handshake doesn't completely get accross. Local channel + optimization requires frames flowing to trigger when optimization + can happen. When optimization happens the media frame that + triggered the optimization is dropped. Sending DTMF requires + frames to flow in the other direction for timing purposes while + sending nothing. If internal timing is not enabled when MOH is + playing, Asterisk switches to received timing when an audio frame + is received. With optimization dropping media frames and MOH not + sending frames unless it receives frames, occasionaly there are + no more frames being passed and the test fails. * The asterisk + command line -I option and the asterisk.conf internal_timing + option are removed. Asterisk now always uses internal timing when + needed if any timing module is loaded. The issue ASTERISK-14861 + did this quite awhile ago in v1.4 but effectively is broken if + other internal timing modules besides DAHDI are used. The + ast_read_generator_actions() now only does received timing if it + has no choice for frame generators like MOH, silence, and + playback streaming. * Cleaned up some code dealing with frame + generators in ast_deactivate_generator(), + generator_write_format_change(), ast_activate_generator(), and + ast_channel_stop_silence_generator(). * Removed + ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and + ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ + Merged revisions 411715 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411716 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411717 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/utils.c, res/res_musiconhold.c, main/channel.c, + main/stasis_cache.c, /: Add some asserts that were handy when + looking for a stasis cache problem. * Assert if a channel is + destroyed but has the snapshot staging flag set. In this case the + final channel destruction snapshot would never get taken. * + Assert if what we just got out of the stasis cache is not what we + were looking for. This assert would have saved several days + searching for a bug and a lot of my hair. * Assert if the music + on hold message posts could not find the associated channel. A + crash will happen later when manager tries to send the MOH AMI + message. This assert catches the problem when the stasis message + is posted instead of by the thread processing the defective + message. * Always generate a backtrace when an ast_assert() + fails. Review: https://reviewboard.asterisk.org/r/3411/ ........ + Merged revisions 411701 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-04 15:13 +0000 [r411688] Matthew Jordan + + * /, main/http.c: http: Fix spurious ERROR message in responses + with no content When a response has a content length of 0, fwrite + would be called to write a buffer with no data in it. This + resulted in the following classic error message: [Apr 3 11:49:17] + ERROR[26421] http.c: fwrite() failed: Success This patch makes it + so that we only attempt to write out the content if the + calculated content_length is non-zero. ........ Merged revisions + 411687 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-03 12:06 +0000 [r411671] Kinsey Moore + + * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Add test event for + state change This adds a test event when subscription state + changes so that integration tests may trigger new actions at the + appropriate times. Review: + https://reviewboard.asterisk.org/r/3383/ ........ Merged + revisions 411670 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-03 11:47 +0000 [r411669] Matthew Jordan + + * res/res_hep.c, /: res_hep: Fix crash when hep.conf not available + Parts of res_hep properly checked for a valid configuration + object before attempting to access the configuration. A check, + however, was missed when a packet is sent. This patch fixes the + crash caused by not checking if the configuration object is + valid. ........ Merged revisions 411668 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-02 18:57 +0000 [r411656] Mark Michelson + + * main/sorcery.c, /, res/res_mwi_external.c, + res/res_pjsip/config_system.c, configs/sorcery.conf.sample, + main/bucket.c, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c, + tests/test_sorcery.c, tests/test_sorcery_realtime.c: Prevent + duplicate sorcery wizards from being applied to sorcery object + types. This commit contains several changes to sorcery: 1) + Application of sorcery configuration based on module name is + automatically performed when sorcery is opened for a module. 2) + Sorcery will not attempt to apply the same wizard to an object + type more than once. 3) Sorcery gives more exact results when + attempting to apply a wizard, whether as the default or based on + configuration. Sorcery unit tests still pass for me after making + these changes. Review: https://reviewboard.asterisk.org/r/3326 + ........ Merged revisions 411159 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-01 22:42 +0000 [r411637-411639] Richard Mudgett + + * res/parking/parking_bridge.c, /: res_parking: Minor tweaks. * Use + ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of + ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables. + * Use ast_copy_string() instead of inlining it. * Remove an + already done TODO comment. * Some whitespace tweaks. ........ + Merged revisions 411638 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, /: stasis_channels.c: Eliminate another + overuse of RAII_VAR(). ........ Merged revisions 411636 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-04-01 16:52 +0000 [r411587] Joshua Colp + + * /, apps/app_queue.c: app_queue: Fix a bug where realtime members + would be deleted during reload causing waiting callers to get + ejected. This patch causes realtime queue members to remain in + queues during the reload process. Previously these members would + be removed causing any waiting callers to be ejected from the + queue with a reason of "EXITEMPTY". ASTERISK-23547 #close + ASTERISK-23547 #comment Patch + app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo + Rossi (license 6409) Review: + https://reviewboard.asterisk.org/r/3404/ ........ Merged + revisions 411584 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411585 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411586 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 18:32 +0000 [r411556] Matthew Jordan + + * include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added), + res/res_hep.exports.in (added), configs/hep.conf.sample (added), + CHANGES, res/res_hep.c (added), /: res_hep/res_hep_pjsip: Add a + HEPv3 capture agent module and a logger for PJSIP This patch adds + the following: (1) A new module, res_hep, which implements a + generic packet capture agent for the Homer Encapsulation Protocol + (HEP) version 3. Note that this code is based on a patch provided + by Alexandr Dubovikov; I basically just wrapped it up, added + configuration via the configuration framework, and threw in a + taskprocessor. (2) A new module, res_hep_pjsip, which forwards + all SIP message traffic that passes through the res_pjsip stack + over to res_hep for encapsulation and transmission to a HEPv3 + capture server. Much thanks to Alexandr for his Asterisk patch + for this code and for a *lot* of patience waiting for me to port + it to 12/trunk. Due to some dithering on my part, this has taken + the better part of a year to port forward (I still blame CDRs for + the delay). ASTERISK-23557 #close Review: + https://reviewboard.asterisk.org/r/3207/ ........ Merged + revisions 411534 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 18:00 +0000 [r411533] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, + addons/chan_ooh323.c, /, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c: + process stack command even if gatekeeper client isn't register + don't destroy gatekeeper client if it is not started don't + destroy gatekeeper client in some sort of gatekeeper errors + signal rtp create condition when call cleared before rtp + structure created (closes issue ASTERISK-23460) Reported by: + Dmitry Melekhov Patches: ASTERISK-23460-2.patch Tested by: Dmitry + Melekhov ........ Merged revisions 411531 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411532 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 17:41 +0000 [r411515-411530] Matthew Jordan + + * rest-api/api-docs/channels.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/playbacks.json, UPGRADE.txt, + rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES, + include/asterisk/manager.h, rest-api/api-docs/bridges.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/mailboxes.json, + rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json: Update API versions and + UPGRADE/CHANGES for 12.2.0 This patch does the following: * It + updates the AMI version to 2.2.0 to indicate backwards compatible + changes have been made since the last release * It updates the + ARI version to 1.2.0 to indicate backwards compatible changes + have been made since the last release * It updates the + UPGRADE/CHANGES files with changes that were not mentioned + ........ Merged revisions 411529 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * UPGRADE.txt, res/res_config_odbc.c: res_config_odbc: Fix for + nullable integer columns and keyfield existence check in + update_odbc. This patch fixes setting nullable integer columns to + NULL instead of an empty string, which fails for PostgreSQL, for + example. The current code is supposed to do so, but the check is + broken. The patch also allows the first column in the list to be + a nullable integer. Also, the check for existence of a mandatory + column checked for the first column in the list instead of the + key field lookup column. This patch fixes that issue as well. + Finally, the compatibility option allow_empty_string_in_nontext, + which was added to previous revisions to allow for some database + backends with certain schemas to function, has been removed. + Review: https://reviewboard.asterisk.org/r/3335 ASTERISK-23459 + #close ASTERISK-23351 #close (closes issue ASTERISK-23459) + Reported by: zvision patches: res_config_odbc.diff uploaded by + zvision (License 5755) + +2014-03-28 16:18 +0000 [r411469] Scott Griepentrog + + * main/tcptls.c, main/manager.c, /, main/http.c: http: response + body often missing after specific request This patch works around + a problem with the HTTP body being dropped from the response to a + specific client and under specific circumstances: a) Client + request comes from node.js user agent "Shred" via use of + swagger-client library. b) Asterisk and Client are *not* on the + same host or TCP/IP stack In testing this problem, it has been + determined that the write of the HTTP body is lost, even if the + data is written using low level write function. The only solution + found is to instruct the TCP stack with the shutdown function to + flush the last write and finish the transmission. See review for + more details. ASTERISK-23548 #close (closes issue ASTERISK-23548) + Reported by: Sam Galarneau Review: + https://reviewboard.asterisk.org/r/3402/ ........ Merged + revisions 411462 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411463 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-28 15:48 +0000 [r411375-411460] Matthew Jordan + + * UPGRADE.txt, /: UPGRADE: Note IAX2 compatibility issue between + 1.4 and 1.8+ systems. ........ Merged revisions 411457 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411458 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411459 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * contrib/realtime/mysql/voicemail_messages.sql (removed), + contrib/realtime/postgresql/realtime.sql (removed), + contrib/realtime/mysql/voicemail_data.sql (removed), + contrib/realtime/mysql/musiconhold.sql (removed), + contrib/realtime/mysql/queue_log.sql (removed), + contrib/realtime/mysql/voicemail.sql (removed), + contrib/realtime/mysql/sippeers.sql (removed), /, + contrib/realtime/mysql/iaxfriends.sql (removed), + contrib/realtime/mysql/meetme.sql (removed): contrib/realtime: + Remove empty SQL script files Since the relatime scripts are now + managed by Alembic, the previous realtime scripts were previously + removed. However, the removal process messed up, as the files + were still in the repository. The contents were just empty. This + removes the files from the tree. ........ Merged revisions 411442 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/sip/include/sip.h: chan_sip: Add MESSAGE request to + allowed methods The allowed methods advertised by chan_sip did + not previously note the MESSAGE request. Even in Asterisk 1.8, we + do accept in-dialog MESSAGE requests; we should advertise that we + support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504 + #comment Reported by: Martin Kontsek ASTERISK-23504 #comment + Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587) + Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged + revisions 411372 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411373 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411374 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-27 19:21 +0000 [r411312-411328] Corey Farrell + + * funcs/func_global.c, apps/app_speech_utils.c, + apps/confbridge/conf_config_parser.c, + funcs/func_callcompletion.c, funcs/func_frame_trace.c, + funcs/func_callerid.c, main/message.c, /, res/res_mutestream.c, + channels/pjsip/dialplan_functions.c, + res/res_pjsip_header_funcs.c, funcs/func_pitchshift.c, + funcs/func_groupcount.c, funcs/func_volume.c, funcs/func_odbc.c, + funcs/func_channel.c, funcs/func_cdr.c, funcs/func_blacklist.c, + apps/app_stack.c, apps/app_voicemail.c, res/res_calendar.c, + apps/app_jack.c, funcs/func_dialplan.c, funcs/func_speex.c, + channels/chan_sip.c, funcs/func_math.c, funcs/func_strings.c, + funcs/func_jitterbuffer.c, res/res_xmpp.c, channels/chan_iax2.c, + main/features_config.c, res/res_jabber.c: Fix dialplan function + NULL channel safety issues (closes issue ASTERISK-23391) Reported + by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3386/ ........ Merged + revisions 411313 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411314 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411315 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/format.c, include/asterisk.h, /: main/formats: Fix crash in + ast_format_cmp during non-clean shutdown. * Update asterisk.h to + reflect availability of ast_register_cleanup in 11.9. * Use + ast_register_cleanup for format_attr_shutdown. (closes issue + ASTERISK-23103) Reported by: JoshE ........ Merged revisions + 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 411311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-27 14:21 +0000 [r411296] Mark Michelson + + * main/sorcery.c, /: Give sorcery instances a reference to their + wizards. On graceful shutdown, sorcery wizards are all killed + off, but it is possible for sorcery instances to still have + dangling pointers after this, possibly causing a crash. Giving + the sorcery instances a reference to their wizards ensures that + the wizard reference will remain valid for the lifetime of the + sorcery instance. Review: https://reviewboard.asterisk.org/r/3401 + ........ Merged revisions 411295 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-26 22:45 +0000 [r411246] Joshua Colp + + * /, main/say.c: say: Fix a bug where SayNumber in Polish tries to + play incorrect sound. This change fixes a bug where calling + SayNumber with a number divisible by 100 using the Polish + language would cause the code to attempt to play a sound file + with an empty name. (closes issue ASTERISK-23509) Reported by: + zvision Review: https://reviewboard.asterisk.org/r/3378/ ........ + Merged revisions 411243 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411244 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411245 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-26 16:15 +0000 [r411194] Jonathan Rose + + * /, channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send + real CallerID information with P-Assserted-Identity (RFC-3325) + Prior too this patch, the P-Asserted-Identity header would + include anonymous caller id information which seems to go against + the point of the P-Asserted-Identity header. Now the real caller + ID information will be included in this header. Also, no privacy + header would be included. This patch adds 'Privacy: id' to + outgoing SIP messages that include the P-Asserted-Identity + header. (closes issue AST-1301) ........ Merged revisions 411189 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 411190 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411193 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-26 16:05 +0000 [r411192] Richard Mudgett + + * /, + contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py: + Fix 'alembic branches' merge conflict as described by the web + page. ........ Merged revisions 411191 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 18:44 +0000 [r411174] Sean Bright + + * /, res/ari/config.c: ARI: Don't complain about missing ARI users + when we aren't enabled Currently, if ARI is not enabled it will + still complain that there are no configured users. This patch + checks to see if ARI is enabled before logging and error or + iterating the container to validate the users. Review: + https://reviewboard.asterisk.org/r/3391/ ........ Merged + revisions 411173 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 17:40 +0000 [r411158] Mark Michelson + + * /, res/res_pjsip/pjsip_configuration.c, UPGRADE.txt, + res/res_pjsip_messaging.c, res/res_pjsip.c, + include/asterisk/res_pjsip.h: Add a "message_context" option for + PJSIP endpoints. ........ Merged revisions 411157 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 16:57 +0000 [r411142] Richard Mudgett + + * res/res_pjsip/pjsip_options.c, res/res_pjsip.c, + include/asterisk/res_pjsip.h, /: res_pjsip: Fix contact + authenticate_qualify endpoint lookup when qualifing a contact. * + Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of + find_endpoints() with find_an_endpoint() since only the first + found endpoint is ever needed. * Fixed qualify_contact_cb() to + update the contact with the aor authenticate_qualify setting. + Otherwise, permanent contacts in the aor type sections would have + a config line order dependancy. * Fixed off nominal path contact + ref leak in qualify_contact(). The comment saying the unref is + not needed was wrong. * Fixed off nominal path use of the + endpoint parameter if it is NULL in send_out_of_dialog_request(). + * Added missing off nominal path unref of pjsip tdata in + send_out_of_dialog_request(). * Fixed off nominal path failing to + call the callback in send_request_cb() when the request is + challenged for authentication. * Eliminated silly RAII_VAR() use + in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen + to better reflect reality. (closes issue ASTERISK-23254) Reported + by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/ + ........ Merged revisions 411141 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 16:06 +0000 [r411092] Kinsey Moore + + * /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If + update_provisional_keepalive() is called while + send_provisional_keepalive_full() is waiting on the PVT lock, + then pvt->provisional_keepalive_sched_id will be changed to a new + sched_id value by update_provisional_keepalive(), but that new + sched_id then may be overwritten with -1 by + send_provisional_keepalive_full(), killing the pvt's reference to + a schedule and "leaking" the reference. (closes issue + ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ + Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies + Patches: provisional_keepalive_fix.diff uploaded by Steve Davies + (license 5012) ........ Merged revisions 411088 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411089 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411091 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 15:56 +0000 [r411090] Jonathan Rose + + * /, res/res_stasis.c: ARI: Resolve a subscription leak against + implicit bridge subscriptions When a channel in a stasis + application is joined to a bridge, a subscription for that bridge + is created implicitly for the stasis application serving the + channel. Prior to this patch, subsequent removals of the channel + from the bridge would leave the subscription open. Review: + https://reviewboard.asterisk.org/r/3380/ ........ Merged + revisions 411086 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-25 15:47 +0000 [r411073-411087] Richard Mudgett + + * utils/conf2ael.c, main/lock.c, utils/ael_main.c: Revert -r411073. + It didn't help and blew up the system. + + * utils/ael_main.c, utils/conf2ael.c, main/lock.c: locking: Add + temporary sanity checks. Add some temporary sanity checks to hunt + for locking problems with the masquerade supertest. + +2014-03-24 21:39 +0000 [r411024] Joshua Colp + + * /, channels/chan_sip.c: chan_sip: Always use fromdomain if set + for domain, even if callerid is set to restricted. (closes issue + ASTERISK-20841) Reported by: Kelly Goedert ........ Merged + revisions 411021 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 411022 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 411023 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-21 16:04 +0000 [r410996] Richard Mudgett + + * /, res/res_pjsip_registrar.c: res_pjsip_registrar.c: + Miscellaneous cleanup in rx_task(). * Fix variable shadowing of + 'updated' by renaming it to 'contact_update'. * Checked + 'contact_update' for ast_sorcery_copy() failure. * Removed silly + use of RAII_VAR() for 'contact_update'. ........ Merged revisions + 410995 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-21 15:50 +0000 [r410981-410994] Sean Bright + + * res/ael/ael.flex, utils/Makefile, pbx/pbx_ael.c, + res/ael/ael_lex.c: Make the AEL load process less chatty. + Switched a bunch of LOG_NOTICEs to ast_debug. This time without + breaking the build. + + * pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Revert + r410981. aelparse blew up. + + * main/config.c: Remove a LOG_NOTICE from + ast_config_engine_register. There is enough indication from the + CLI that we are loading a realtime engine as it is. + + * pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Make the AEL + load process less chatty. Switched a bunch of LOG_NOTICEs to + ast_debug. + +2014-03-20 23:02 +0000 [r410967] Jonathan Rose + + * apps/app_confbridge.c, /: app_confbridge: Fix bug - users with + startmuted set don't start muted (closes issue ASTERISK-23461) + Reported by: Chico Manobela Review: + https://reviewboard.asterisk.org/r/3373/ ........ Merged + revisions 410965 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410966 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-20 16:35 +0000 [r410950] Richard Mudgett + + * include/asterisk/rtp_engine.h, main/dial.c, main/manager.c, /, + main/channel_internal_api.c, main/core_unreal.c, + include/asterisk/channel.h, res/ari/resource_channels.c, + res/res_stasis_snoop.c: assigned-uniqueids: Miscellaneous cleanup + and fixes. * Fix memory leak in ast_unreal_new_channels(). Made + it generate the ;2 uniqueid on a stack variable instead of + mallocing it. * Made send error response to ARI and AMI requests + instead of just logging excessive uniqueid length and allowing + truncation. action_originate() and + ari_channels_handle_originate_with_id(). * Fixed minor truncating + uniqueid hole when generating the ;2 uniqueid string length. + Created public and internal lengths of uniqueid. The internal + length can handle a max public uniqueid plus an appended ;2. * + free() and ast_free() are NULL tolerant so they don't need a NULL + test before calling. * Made use better struct initialization + format instead of the position dependent initialization format. + Also anything not explicitly initialized in the struct is + initialized to zero by the compiler. * Made + ast_channel_internal_set_fake_ids() use the safer + ast_copy_string() instead of strncpy(). Review: + https://reviewboard.asterisk.org/r/3371/ ........ Merged + revisions 410949 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-19 17:27 +0000 [r410934] Mark Michelson + + * /, res/res_pjsip_endpoint_identifier_ip.c: PJSIP: Allow for + identify sections to be specified in sorcery.conf. "identify" is + a special type of configuration object in PJSIP because unlike + the other objects, it is not provided by the base res_pjsip + module. Instead, it is provided by the + res_pjsip_endpoint_identifier_ip module. If using the default + sorcery wizard (config,criteria=type=identify) then things work + because the module that applies the default wizard is the correct + module. However, if attempting to use sorcery.conf to apply an + alternate wizard, it was not possible. If you attempted to + specify the identify object type in the res_pjsip section, then + the object could not be registered since the object was + undocumented for the res_pjsip module. There was no alternate + configuration section defined for it, so you were out of luck if + you wanted to override the default wizard. With this change, the + identify section will properly have a sorcery.conf-based wizard + applied when the identify definition is within the + res_pjsip_endpoint_identifier_ip section. ........ Merged + revisions 410933 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-19 14:25 +0000 [r410905-410919] Joshua Colp + + * res/res_stasis.c, /: res_stasis: Fix a bug where the default + bridge type was not set. ........ Merged revisions 410918 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * CHANGES, res/res_stasis.c, rest-api/api-docs/bridges.json, /, + res/ari/resource_bridges.h: res_stasis: Extend bridge type to be + a comma separated list of bridge attributes. This change turns + the bridge type field into a comma separated list of attributes. + These attributes include: mixing, holding, dtmf_events, and + proxy_media. By setting the various attributes a user can control + the type of bridge created with the behavior they need for their + application. (closes issue ASTERISK-23437) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........ + Merged revisions 410904 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-19 02:33 +0000 [r410891] Matthew Jordan + + * res/res_ari.c, /: res_ari: Fix documentation schema error + ........ Merged revisions 410890 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 23:32 +0000 [r410877] Rusty Newton + + * res/res_ari.c, /: res_ari: Add notes about Asterisk HTTP server + to the "enabled" config option for the res_ari general section + Added note and see-also reminding user to enable the HTTP server. + (closes issue ASTERISK-22499) Reported by: Rusty Newton ........ + Merged revisions 410876 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 15:45 +0000 [r410863] Scott Griepentrog + + * /, main/http.c: ARI: allow json content type with zero length + body When a request was received with a Content-type of json, the + body was sent for json parsing - even if it was zero length. This + resulted in ARI requests failing that were valid, such as a + channel DELETE with no parameters. The code has now been changed + to skip json parsing with zero content length. (closes issue + SWP-6748) Reported by: Samuel Galarneau Review: + https://reviewboard.asterisk.org/r/3360/ ........ Merged + revisions 410858 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 15:28 +0000 [r410862] Matthew Jordan + + * main/cdr.c, /: cdr: Add asserts for when we don't know about a + CDR for a channel In the CDR core, every channel should either be + filtered out (due to being an 'internal' channel used as an + implementation detail, such as playing media back into a bridge) + or it should get a CDR. Even if that CDR ends up being discarded, + we still give the channel a CDR in case we end up needing it. If + we hit a situation where a channel does not have a CDR, we should + blow up in -dev-mode. Asserts are appropriate for that. This + patch adds those asserts, as they would have quickly caught the + error fixed by r410814. ........ Merged revisions 410861 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 12:45 +0000 [r410845] Joshua Colp + + * /, res/res_pjsip/config_system.c: res_pjsip: Fix memory leak of + nameservers in off-nominal resolver creation failure. Thanks + Walter Doekes! ........ Merged revisions 410844 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 11:52 +0000 [r410831] Sean Bright + + * res/res_fax_spandsp.c, /: res_fax_spandsp: Use g711_free() when + available. Per Johann Steinwendtner on the asterisk-dev mailing + list: + http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html + g711_free() was introduced in spandsp 0.0.6pre4 and + g711_release() became a noop. I opted not to remove the call to + g711_release() since it is harmless and to call g711_free() if we + have a sufficiently recent version of spandsp. (issue + ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged + revisions 410829 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410830 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-18 02:09 +0000 [r410814] Richard Mudgett + + * main/stasis_cache.c, /: stasis_cache: Use the right variable in + the cache entry ao2 cmp function. ........ Merged revisions + 410813 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-17 22:54 +0000 [r410794-410796] Joshua Colp + + * include/asterisk/dns.h, CHANGES, + res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c, + main/dns.c, /, res/res_pjsip/config_system.c: res_pjsip: Enable + PJSIP DNS client support. This change enables DNS client support + within PJSIP. System nameservers are automatically discovered + using res_init or res_ninit. If this fails then PJSIP will resort + to using gethostbyname for resolution. By enabling this support + we gain SRV support, failover, and weight support. (closes issue + ASTERISK-23435) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3343/ ........ Merged + revisions 410795 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Make address + replacement less aggressive. This change makes the + res_pjsip_multihomed module less aggressive when changing the + address in messages. It will now only occur if the transport in + use is bound to the any address OR if the system determined + source address matches the bound address of the transport in use. + Review: https://reviewboard.asterisk.org/r/3369/ ........ Merged + revisions 410793 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-17 22:24 +0000 [r410775] Russ Meyerriecks + + * /, main/callerid.c: callerid: Logic error in checksum processing + Callerid checksum-ing was being handled incorrectly here. When + the checksum is calculated to be 0x00, it will perform 0x100-0x00 + which results in 0x100. This value will then fail the otherwise + correct callerid message. This patch changes the logic to simply + add the calculated checksum to the transmitted 2's compliment + checksum. Review: https://reviewboard.asterisk.org/r/3356/ + (closes issue ASTERISK-23488) ........ This is a merge of merged + revisions 410750 410747 from + http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a + broken patch to be comitted to trunk so I pre-merge merged them. + +2014-03-17 19:35 +0000 [r410684-410699] Mark Michelson + + * res/res_mwi_external.c, res/res_pjsip/config_system.c, + configs/sorcery.conf.sample, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c, + tests/test_sorcery.c, tests/test_sorcery_realtime.c, + main/sorcery.c, /: Revert changes to sorcery that accidentally + got committed. These changes were still up for review and have + not been approved yet. I must have had the changes in my working + copy when making a different change. ........ Merged revisions + 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * bridges/bridge_softmix.c, tests/test_sorcery.c, main/channel.c, + res/res_pjsip/config_system.c, res/res_mwi_external.c, + include/asterisk/bridge_channel.h, funcs/func_frame_trace.c, + configs/sorcery.conf.sample, res/res_pjsip/pjsip_configuration.c, + include/asterisk/sorcery.h, tests/test_sorcery_astdb.c, + include/asterisk/frame.h, main/bridge_channel.c, + tests/test_sorcery_realtime.c, main/sorcery.c, + res/res_stasis_playback.c, main/frame.c, /: Fix stuck channel in + ARI through the introduction of synchronous bridge actions. + Playing back a file to a channel in an ARI bridge would attempt + to wait until the playback concluded before returning. The method + used involved signaling the waiting thread in the ARI custom + playback function. The problem with this is that there were some + corner cases that were not accounted for: * If a bridge channel + could not be found, then we never would attempt the playback but + would still attempt to wait for the playback to complete. * If + the bridge playfile action failed to queue, we would still + attempt to wait for the playback to complete. * If the bridge + playfile action were queued but some circumstance caused the + playback not to occur (the bridge dies, the channel is removed + from the bridge), then we would never be notified. The solution + to this is to move the waiting logic into the bridge code. A new + bridge API function is added to queue a synchronous action on a + bridge. The waiting thread is notified when the queued frame has + been freed, either due to an error occurring or due to successful + playback. As a failsafe, the waiting thread has a 10 minute + timeout just in case there is a frame leak somewhere. Review: + https://reviewboard.asterisk.org/r/3338 ........ Merged revisions + 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-17 16:48 +0000 [r410672] Richard Mudgett + + * /, apps/confbridge/conf_chan_announce.c: app_confbridge: Add + missing destructor call to announcer channel destructor. ........ + Merged revisions 410671 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-16 20:27 +0000 [r410651] Matthew Jordan + + * /, res/stasis/app.c: stasis/app.c: Add some extra debugging for + subscription counts Events are sent to a connected ARI + application based on the things that ARI application cares about. + These subscriptions can be set up implicitly - such as when that + ARI application creates a new object - or explicitly, via the + application resource's subscription operations. Debugging *why* + something was being sent to an application - or why something was + not being sent to an application - was a bit tricky, as there was + no debug information for the subscriptions. This patch adds some + debug level 3 statements that show the subscription counts for + applications. (Level 3 was chosen as it matches the verbose level + 3 statements elsewhere) ........ Merged revisions 410650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-15 15:24 +0000 [r410639] Russell Bryant + + * include/asterisk/framehook.h: framehook.h: Fix some doc typos. + There were a number of instances in this header file where + "function all" was intended to be "function call". This patch + fixes that up. + +2014-03-14 21:56 +0000 [r410626] Mark Michelson + + * /, tests/test_sorcery_realtime.c: Fix failing realtime sorcery + tests. The store realtime callback needs to return a positive + value for sorcery to treat the store as a success. ........ + Merged revisions 410625 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 21:36 +0000 [r410624] Jonathan Rose + + * main/manager.c, /: manager: fix memory leak in manager_add_filter + function (closes issue ASTERISK-23420) Reported by: Etienne + Lessard Patches: manager_eventfilter_leak uploaded by Etienne + Lessard (license 6394) ........ Merged revisions 410609 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410623 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 20:55 +0000 [r410591-410608] Mark Michelson + + * /, main/db.c: Remove an extra ast_cond_wait() that slipped + through the patch. ........ Merged revisions 410606 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410607 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/config.c, res/res_sorcery_realtime.c: Handle the return + values of realtime updates and stores more accurately. Realtime + backends' update and store callbacks return the number of rows + affected, or -1 if there was a failure. There were a couple of + issues: * The config API was treating 0 as a successful return, + and positive values as a failure. Now the config API treats + anything >= 0 as a success. * res_sorcery_realtime was treating 0 + as a successful return from the store procedure, and any positive + values as a failure. Now sorcery treats anything > 0 as a + success. It still considers 0 a "failure" since there is no + change to report to observers. Review: + https://reviewboard.asterisk.org/r/3341 ........ Merged revisions + 410592 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c: Prevent conflicts regarding unsolicited + and solicited MWI to an endpoint. If an endpoint is receiving + unsolicited MWI for a mailbox and then attempts to subscribe to + an AOR that provides MWI for the same mailbox, then the SUBSCRIBE + is rejected with a 500 response. Review: + https://reviewboard.asterisk.org/r/3345 ........ Merged revisions + 410590 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 17:56 +0000 [r410589] Scott Griepentrog + + * /, CHANGES: uniqueid: Update CHANGES to reflect new features Note + the new features provided by uniqueid in the CHANGES file. (issue + ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/ + ........ Merged revisions 410588 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:42 +0000 [r410575] Jonathan Rose + + * /, main/acl.c, res/res_pjsip/pjsip_configuration.c, + contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py, + CHANGES, res/res_pjsip/config_transport.c, + include/asterisk/acl.h: PJSIP: TOS values should be represented + as decimals in sorcery objects (closes issue ASTERISK-23235) + Reported by: George Joseph Review: + https://reviewboard.asterisk.org/r/3324/ ........ Merged + revisions 410574 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:19 +0000 [r410567] Mark Michelson + + * /, main/db.c: Prevent delayed astdb syncs. The syncing thread + sleeps for a second before waiting to be told to attempt to sync + again. If a signal were sent during this sleeping period, we + would end up having to wait until the next sync signal occurred + in order to sync up the astdb. This code rearrangement also + ensures that any pending transactions will be synced prior to + Asterisk shutting down. Patches: db_sync.patch by John Hardin + (License #6512) ........ Merged revisions 410556 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410559 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:17 +0000 [r410560] Jonathan Rose + + * res/ari/resource_bridges.c, /: ARI/bridges: Forward + Playback/Recording Started/Finished to bridge topic (closes issue + ASTERISK-23444) Reported by: Ben Merrills Review: + https://reviewboard.asterisk.org/r/3340/ ........ Merged + revisions 410558 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-14 16:01 +0000 [r410542-410557] Richard Mudgett + + * include/asterisk/app.h, /, res/res_mwi_external.c, main/app.c: + res_mwi_external: Clear the stasis cache entry when the external + MWI is deleted. One of the things missing when external MWI + support was added was the ability to clear the stasis cache entry + of deleted external MWI mailboxes. Review: + https://reviewboard.asterisk.org/r/3325/ ........ Merged + revisions 410555 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/cdr.c: cdr.c: Add missing aow_unlock(cdr) in off nominal + path of handle_dial_message(). * Trivial common code hoisting in + handle_bridge_leave_message(). * Some whitespace fixing. ........ + Merged revisions 410541 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-13 19:33 +0000 [r410528] Kinsey Moore + + * res/stasis/control.h, res/res_stasis.c, /, res/stasis/control.c: + ARI: Ensure managing application receives ChannelEnteredBridge + messages This fixes an issue where a Stasis application running + over ARI and subscribed to ari/events could miss the + ChannelEnteredBridge event because it did not subscribe to the + new bridge fast enough. To accomplish this, it subscribes the + application controlling the channel to the new bridge before + adding it to that bridge which required the stasis_app_control + structure to maintain a reference to the stasis_app. (closes + issue ASTERISK-23295) Review: + https://reviewboard.asterisk.org/r/3336/ ........ Merged + revisions 410527 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-13 13:25 +0000 [r410511] Joshua Colp + + * res/res_pjsip_multihomed.c, /: Multiple revisions 410509-410510 + ........ r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar + 2014) | 2 lines res_pjsip_multihomed: Fix a bug where the 200 OK + for a REGISTER would contain the wrong contact. ........ r410510 + | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines + res_pjsip_multihomed: Remove change for testing fix. ........ + Merged revisions 410509-410510 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-12 19:06 +0000 [r410492-410494] Richard Mudgett + + * res/res_musiconhold.c, main/channel.c, /: res_musiconhold.c: + Generate MOH start/stop events whenever the MOH stream is + started/stopped. * Made res_musiconhold.c always post the + MusicOnHoldStart/MusicOnHoldStop events when it actually + starts/stops the music streams. This allows the events to always + happen when MOH starts/stops. The event posting code was moved to + the MOH alloc/release routines. * Made channel_do_masquerade() + stop any MOH on the original channel before masquerading so the + original channel will get a stop event with correct information. + * Cleaned up a couple odd codings in moh_files_alloc() and + moh_alloc() dealing with the music state variable. (issue + ASTERISK-23311) Reported by: Benjamin Keith Ford Review: + https://reviewboard.asterisk.org/r/3306/ ........ Merged + revisions 410493 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_state.c, + apps/confbridge/conf_state_single.c, + apps/confbridge/conf_state_inactive.c, + apps/confbridge/conf_state_single_marked.c, /: app_confbridge: + Make explicitly stop MOH if a user is kicked or hangs up while + MOH is playing. When MOH is playing to a user in a conference and + the user is kicked or hangs up from the conference then the AMI + MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event: + MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported + by: Benjamin Keith Ford Review: + https://reviewboard.asterisk.org/r/3306/ ........ Merged + revisions 410490 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410491 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-12 12:51 +0000 [r410452-410472] Joshua Colp + + * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Fix a bug + where outgoing messages for TCP would go out using UDP. This + change fixes a bug where the code which changes the transport did + not check whether the message is going out over UDP or not before + changing it. For TCP and TLS transports we don't need to change + the transport as the correct one is already chosen. ........ + Merged revisions 410471 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_multihomed.c (added), /: res_pjsip_multihomed: Add + module which places the correct address within messages. Due to + how messages are handled within PJSIP it is not until a message + is actually sent that the destination is reliably known. This + means that the addresses placed within the message may not be of + the interface the message is being sent out on. This module + determines what interface a message is being sent on and updates + the message to contain the correct address if applicable. This + module was tested by myself in a virtualized environment with + multiple interfaces and also by Kinsey Moore in the following + configuration: Networks: * 10.24.16.0/21 ** hard phone ** default + gateway * 10.24.64.0/21 ** softphone with pjsip-based stack + Transport details: bind address: 0.0.0.0 protocol: UDP All + endpoints were tested with explicitly configured transports and + unconfigured transports. This was tested with inbound and + outbound calls, both of which were experiencing detrimental + effects from incorrect IP addresses in SIP messages. These + effects were only experienced by the soft phone on the 10.24.64.0 + network since the messages to the hard phone on the 10.24.16.0 + network had the correct IP address. (closes issue ASTERISK-23020) + Reported by: xrobau Review: + https://reviewboard.asterisk.org/r/3102/ ........ Merged + revisions 410451 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 17:21 +0000 [r410395] Richard Mudgett + + * /, main/http.c: AST-2014-001: Stack overflow in HTTP processing + of Cookie headers. Sending a HTTP request that is handled by + Asterisk with a large number of Cookie headers could overflow the + stack. Another vulnerability along similar lines is any HTTP + request with a ridiculous number of headers in the request could + exhaust system memory. (closes issue ASTERISK-23340) Reported by: + Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. + Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions + 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 410381 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 16:33 +0000 [r410369] Scott Griepentrog + + * res/ari/resource_channels.c, main/manager.c, /: unqiueid: correct + max uniqueid length test This patch adds null string test prior + to checking for a max uniqueid value that was added in r410157. + ........ Merged revisions 410368 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 13:30 +0000 [r410346] Kinsey Moore + + * /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad + session timers request This change allows chan_sip to avoid + creation of the channel and consumption of associated file + descriptors altogether if the inbound request is going to be + rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey + Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey + Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by + Corey Farrell (license 5909) ........ Merged revisions 410308 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 410311 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410329 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-10 12:53 +0000 [r410307] Joshua Colp + + * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c: AST-2014-003: + res_pjsip: When handling 401/407 responses don't assume a request + will have an endpoint. This change removes the assumption that an + outgoing request will always have an endpoint and makes the + authenticate_qualify option work once again. (closes issue + ASTERISK-23210) Reported by: Joshua Colp ........ Merged + revisions 410306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-08 16:50 +0000 [r410288] George Joseph + + * res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip_endpoint_identifier_ip.c, + include/asterisk/res_pjsip_cli.h, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_cli.c, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/config_transport.c, main/sorcery.c, + include/asterisk/res_pjsip.h: pjsip_cli: Create pjsip show + channel and contact, and general cli code cleanup. Created the + 'pjsip show channel' and 'pjsip show contact' commands. + Refactored out the hated ast_hashtab. Replaced with + ao2_container. Cleaned up function naming. Internal only, no + public name changes. Cleaned up whitespace and brace formatting + in cli code. Changed some NULL checking from "if"s to + ast_asserts. Fixed some register/unregister ordering to reduce + deadlock potential. Fixed ast_sip_location_add_contact where the + 'name' buffer was too short. Fixed some self-assignment issues in + res_pjsip_outbound_registration. (closes issue ASTERISK-23276) + Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged + revisions 410287 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-08 15:45 +0000 [r410275] Matthew Jordan + + * /, res/ari/resource_channels.c: resource_channels: Check if a + passed in ID is NULL before checking its length Calling strlen on + a NULL string is explosive. This patch checks whether or not the + passed in string is NULL or zero length before checking to see if + the string is too long. ........ Merged revisions 410274 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 22:56 +0000 [r410227] Corey Farrell + + * /, channels/chan_sip.c: chan_sip: Fix deadlock of monlock between + unload_module and do_monitor Release monlock before calling + pthread_join. This ensures do_monitor cannot freeze by locking + monlock during module unload. (closes issue ASTERISK-21406) + Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3284/ ........ Merged + revisions 410224 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 410225 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 22:08 +0000 [r410212] Scott Griepentrog + + * /, include/asterisk/sorcery.h: sorcery: correct field register + argument list This fixes mistakes I previously made in merging + gtjoseph's changes with mine. ........ Merged revisions 410211 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 21:54 +0000 [r410208-410210] Matthew Jordan + + * /, main/config_options.c: config_options: Display the see-also + information for CLI config option help The config option help + information has always parsed the tags in the XML + documentation. Unfortunately, it just never bothered displaying + them on the CLI. With this patch, when you execute 'config show + help [module] [obj] [option]', it will display what other options + are useful to you. (closes issue ASTERISK-22008) Reported by: + Richard Mudgett ........ Merged revisions 410209 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip.c, /: res_pjsip: Fix documentation for one touch + recording see-also links The one touch recording options have + several see-also links between the various configuration options. + These were 'broken' by the snake casing of those options. This + patch corrects the see-also links such that they reference the + correct option names. ........ Merged revisions 410194 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 21:23 +0000 [r410207] Mark Michelson + + * main/sorcery.c, res/res_sorcery_realtime.c, /, + include/asterisk/sorcery.h, tests/test_sorcery_realtime.c: Make + res_sorcery_realtime filter unknown retrieved results. When + retrieving data from a database or other realtime backend, it's + quite possible to retrieve variables that Asterisk does not care + about but that are legitimate to exist. Asterisk does not need to + throw a hissy fit when these variables are encountered but rather + just filter them out. Review: + https://reviewboard.asterisk.org/r/3305 ........ Merged revisions + 410187 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 21:11 +0000 [r410191] Scott Griepentrog + + * main/sorcery.c, /, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c: pjsip: allow and disallow + show same codecs In order to prevent confusion over the allow and + disallow list of codecs being the same an option for registering + a field as an alias is added. The alias field will be read from + the configuration file, but afterwards is not listed as a known + field. With disallow set as an alias, the CLI command pjsip show + endpoint # will list the allow= field, but not the disallow + field. (closes issue ASTERISK-23092) Review: + https://reviewboard.asterisk.org/r/3193/ ........ Merged + revisions 410190 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 20:41 +0000 [r410174-410185] Richard Mudgett + + * include/asterisk/devicestate.h, main/stasis_cache.c, + main/stasis_message.c, /, tests/test_devicestate.c, + include/asterisk/stasis.h, main/app.c, main/devicestate.c, + tests/test_stasis.c: stasis cache: Enhance to keep track of an + item from different entities. A stasis cache entry now contains + more than a single message/snapshot. It contains + messages/snapshots for the local entity as well as any remote + entities that post to the cached item. In addition callbacks can + be supplied when the cache is created to compute and post the + aggregate message/snapshot representing all entities stored in + the cache entry. * All stasis messages now have an eid to + indicate what entity posted it. * The stasis cache enhancements + allow device state to cache and aggregate the device states from + local and remote entities in a single operation. The cached + aggregate device state is available immediately after it is + posted to the stasis bus. This improves performance by + eliminating a cache dump and associated ao2 container traversals + to calculate the aggregate state. (closes issue ASTERISK-23204) + Reported by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/3281/ ........ Merged + revisions 410184 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_cel.c, channels/sig_pri.c, channels/sig_ss7.c, + include/asterisk/bridge.h, tests/test_cdr.c, channels/sig_pri.h, + channels/chan_dahdi.c, channels/sig_ss7.h, /: uniqueid: Fix + chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler + errors. (issue ASTERISK-23120) ........ Merged revisions 410171 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 15:47 +0000 [r410158] Scott Griepentrog + + * tests/test_cdr.c, res/res_clioriginate.c, res/res_ari_bridges.c, + tests/test_substitution.c, res/res_stasis_playback.c, + channels/chan_multicast_rtp.c, apps/app_meetme.c, /, + main/bridge_basic.c, include/asterisk/channel_internal.h, + tests/test_app.c, apps/confbridge/conf_chan_record.c, + main/core_unreal.c, channels/chan_gtalk.c, + include/asterisk/stasis_app_playback.h, + res/ari/resource_bridges.c, channels/chan_jingle.c, + channels/chan_phone.c, pbx/pbx_spool.c, + res/ari/resource_bridges.h, res/parking/parking_tests.c, + channels/chan_motif.c, apps/app_confbridge.c, + res/ari/resource_channels.c, include/asterisk/pbx.h, + res/res_stasis.c, include/asterisk/bridge.h, + apps/app_voicemail.c, res/ari/resource_channels.h, + apps/app_dial.c, res/res_calendar_exchange.c, + channels/chan_vpb.cc, apps/app_page.c, apps/app_chanisavail.c, + include/asterisk/dial.h, main/core_local.c, + res/parking/parking_bridge_features.c, + tests/test_stasis_endpoints.c, res/parking/parking_bridge.c, + channels/chan_skinny.c, include/asterisk/stasis_app_snoop.h, + addons/chan_mobile.c, main/bridge_channel.c, + channels/chan_pjsip.c, channels/chan_mgcp.c, + channels/chan_unistim.c, main/pbx.c, + res/res_calendar_icalendar.c, main/ccss.c, + channels/chan_bridge_media.c, main/bridge.c, + tests/test_stasis_channels.c, apps/app_bridgewait.c, + apps/app_originate.c, res/res_calendar_caldav.c, + include/asterisk/channel.h, res/parking/parking_applications.c, + apps/app_followme.c, main/cel.c, apps/app_queue.c, + res/res_ari_channels.c, res/res_calendar_ews.c, + rest-api/api-docs/bridges.json, main/dial.c, + channels/chan_dahdi.c, channels/chan_h323.c, tests/test_cel.c, + rest-api/api-docs/channels.json, + include/asterisk/bridge_internal.h, + apps/confbridge/conf_chan_announce.c, res/res_calendar.c, + include/asterisk/core_unreal.h, addons/chan_ooh323.c, + res/stasis/control.c, channels/chan_sip.c, + main/channel_internal_api.c, include/asterisk/stasis_app.h, + res/res_stasis_snoop.c, channels/chan_console.c, + channels/chan_iax2.c, channels/chan_oss.c, apps/app_agent_pool.c, + main/channel.c, main/manager.c, channels/chan_misdn.c, + tests/test_voicemail_api.c, channels/chan_alsa.c, + channels/chan_nbs.c, main/message.c: uniqueid: channel linkedid, + ami, ari object creation with id's Much needed was a way to + assign id to objects on creation, and much change was necessary + to accomplish it. Channel uniqueids and linkedids are split into + separate string and creation time components without breaking + linkedid propgation. This allowed the uniqueid to be specified by + the user interface - and those values are now carried through to + channel creation, adding the assignedids value to every function + in the chain including the channel drivers. For local channels, + the second channel can be specified or left to default to a ;2 + suffix of first. In ARI, bridge, playback, and snoop objects can + also be created with a specified uniqueid. Along the way, the + args order to allocating channels was fixed in chan_mgcp and + chan_gtalk, and linkedid is no longer lost as masquerade occurs. + (closes issue ASTERISK-23120) Review: + https://reviewboard.asterisk.org/r/3191/ ........ Merged + revisions 410157 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-07 05:04 +0000 [r410108] Matthew Jordan + + * /, channels/chan_sip.c: chan_sip: Allow static realtime members + to be qualified during module load. When a static realtime peer + with qualify=yes is loaded, Asterisk will fail to send an OPTIONS + request due to the lastms being equal to 0. This results in the + peer being unable to receive calls from Asterisk because the + status is permanently UNKNOWN. This patch allows an OPTIONS + request to be sent during module load by ignoring the lastms + value on startup only. Review: + https://reviewboard.asterisk.org/r/3294/ (closes issue + ASTERISK-17523) Reported by: Maciej Krajewski Tested by: + wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor + Peirce (license 6112) ........ Merged revisions 410105 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 410106 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410107 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 23:47 +0000 [r410092] Richard Mudgett + + * main/sorcery.c, /: sorcery.c: Fix off-nominal path ref and memory + leak in ast_sorcery_objectset_json_create(). * Made exit a loop + early on error in ast_sorcery_objectset_json_create(). * Removed + some dead code in ast_sorcery_objectset_create2(). ........ + Merged revisions 410089 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 23:43 +0000 [r410091] Russell Bryant + + * /, res/res_musiconhold.c: moh: fix a refcount error with realtime + MOH I observed a crash in res_musiconhold on an Asterisk 11 + system using realtime MOH. Investigation of the backtrace showed + a corrupt mohclass, implying that it got destroyed before the + code expected it to. I went looking for reference counting errors + that could have caused this crash and this patch this result. It + contains 2 changes. 1) Remove a usless block of code that was + impossible to reach. There was even a comment indicating that it + was impossible to reach. The conditional includes + "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's + inside of an if block with the opposite check + "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no + good reason to keep it around. 2) A similar block to #1 contained + a reference counting error. It stores state->class in the local + variable mohclass without increasing its reference count. The + reference count on mohclass is decremented at the end of the + function. This block of code probably very rarely runs, which + would help explain why this system was working fine for many + months before experiencing a crash. Review: + https://reviewboard.asterisk.org/r/3282/ ........ Merged + revisions 410043 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 410044 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 410090 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 22:39 +0000 [r410042] George Joseph + + * res/res_pjsip/config_auth.c, funcs/func_sorcery.c (added), + res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c, + main/bucket.c, res/res_pjsip_endpoint_identifier_ip.c, + include/asterisk/config.h, include/asterisk/sorcery.h, + res/res_pjsip/pjsip_configuration.c, res/res_pjsip_acl.c, + CHANGES, tests/test_sorcery.c, res/res_pjsip/config_transport.c, + main/config.c, main/sorcery.c: sorcery: Create AST_SORCERY + dialplan function. This patch creates the AST_SORCERY dialplan + function which allows someone to retrieve any value from a + sorcery-based config file. It's similar to AST_CONFIG. The + creation of the function itself was fairly straightforward but it + required changes to the underlying sorcery infrastructure that + rippled into individual sorcery objects. The changes stemmed from + inconsistencies in how sorcery created ast_variable objectsets + from sorcery objects and the inconsistency in how individual + objects used that feature especially when it came to parameters + that can be specified multiple times like contact in aor and + match in identify. You can read more here... + http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html + So, what this patch does, besides actually creating the + AST_SORCERY function, is the following... * Creates + ast_variable_list_append which is a helper to append one + ast_variable list to another. * Modifies the + ast_sorcery_object_field_register functions to accept the + already-defined sorcery_fields_handler callback. * Modifies + ast_sorcery_objectset_create to accept a parameter indicating + return type preference...a single ast_variable with all values + concatenated or an ast_variable list with multiple entries. Also + fixed a few bugs. * Modifies individual sorcery object + implementations to use the new function definition of the + ast_sorcery_object_field_register functions. * Modifies + location.c and res_pjsip_endpoint_identifier_ip.c to implement + sorcery_fields_handler handlers so they return multiple + occurrences as an ast_variable_list. * Added a whole bunch of + tests to test_sorcery. (closes issue ASTERISK-22537) Review: + http://reviewboard.asterisk.org/r/3254/ + +2014-03-06 19:04 +0000 [r410029] Jonathan Rose + + * include/asterisk/acl.h, /, main/acl.c, + res/res_pjsip/pjsip_configuration.c, UPGRADE.txt, + contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py + (added), res/res_pjsip/config_transport.c: pjsip configuration: + Make transport TOS values consistent with endpoints Transport TOS + values were interpreted as DSCP values without being documented + as such. Endpoint TOS values (tos_audio/tos_video) behaved + normally as TOS values have historically. This patch makes the + transport TOS values behave as TOS values and makes all TOS + values readable as string values (e.g. AF11). In addition, + alembic scripts have been updated to use the proper field types + for all TOS/COS values. (issue ASTERISK-23235) Reported by: + George Joseph Review: https://reviewboard.asterisk.org/r/3304/ + ........ Merged revisions 410028 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 18:20 +0000 [r410027] Joshua Colp + + * res/ari/resource_channels.c, CHANGES, + res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, res/ari/resource_bridges.c, + res/ari/ari_model_validators.h, /, + include/asterisk/stasis_app_recording.h, + res/res_stasis_recording.c: res_stasis_recording: Add a + "target_uri" field to recording events. This change adds a + target_uri field to the live recording object. It contains the + URI of what is being recorded. (closes issue ASTERISK-23258) + Reported by: Ben Merrills Review: + https://reviewboard.asterisk.org/r/3299/ ........ Merged + revisions 410025 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 15:58 +0000 [r410012] Mark Michelson + + * res/res_pjsip_mwi.c, /: Don't attempt to link in an aggregate MWI + subscription if an endpoint does not aggregate MWI. Attempting to + link a NULL object into an ao2 container had been benign + previously, but since enabling DO_CRASH in the testsuite, this is + now causing a crash. It's better to be right here anyway. + ........ Merged revisions 410011 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 02:22 +0000 [r409996] Matthew Jordan + + * res/res_fax_spandsp.c, /: res_fax_spandsp: Fix crash when passing + ulaw/alaw data to spandsp When acting as a T.38 fax gateway, + res_fax_spandsp would at times cause a crash in libspandsp. This + would occur when, during fax tone detection, a ulaw/alaw frame + would be passed to modem_connect_tones_rx. That particular + routine expects the data to be in slin format. This patch looks + at the frame type and, if the data is ulaw/alaw, converts the + format to slin before passing it to modem_connect_tones_rx. + Review: https://reviewboard.asterisk.org/r/3296 (closes issue + ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal + Rybarik patches: spandsp_g711decode.diff uploaded by Michal + Rybarik (license 6578) ........ Merged revisions 409990 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409991 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-06 00:33 +0000 [r409970-409977] Richard Mudgett + + * apps/confbridge/conf_state_multi.c, + apps/confbridge/conf_state_inactive.c, /: app_confbridge: Remove + some noop code. ........ Merged revisions 409976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_musiconhold.c: res_musiconhold.c: Remove some + unnecessary RAII_VAR() usage. * Made the moh_register() define + use useful parameter names. ........ Merged revisions 409967 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 20:41 +0000 [r409904-409919] Kinsey Moore + + * main/config.c, /: config: Fix inverted test The test of the + result of the stat() call was inverted such that its output was + only used if the call failed. This inverts the test so that the + output of stat() is used correctly. This was causing full reloads + on unchanged files. (closes issue ASTERISK-23383) Reported by: + David Woolley ........ Merged revisions 409916 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409917 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409918 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * bridges/bridge_native_rtp.c, /: bridge_native_rtp: Fix crash + involving masquerade It is possible for a channel to be + masqueraded out of a bridge which means it may no longer have RTP + glue to check upon leaving said bridge. If this situation + occurred (it's possible at least during dial and call pickup) + then Asterisk would crash. This change makes sure the glue is + checked before use. (closes issue AST-1290) Reported by: John + Bigelow ........ Merged revisions 409900 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 18:51 +0000 [r409889] Richard Mudgett + + * contrib/ast-db-manage/cdr/versions, + contrib/ast-db-manage/cdr/versions/210693f3123d_create_cdr_table.py, + /, + contrib/ast-db-manage/config/versions/28887f25a46f_create_queue_tables.py + (added), contrib/ast-db-manage/cdr.ini.sample (added), + contrib/ast-db-manage/cdr/env.py, contrib/ast-db-manage/cdr + (added), contrib/ast-db-manage/cdr/script.py.mako: alembic: Add + missing queue and CDR table creation scripts. * Added the queues + and queue_members tables to the config alembic scripts. * Added + the CDR table alembic creation script. The CDR table is more of + an example for new setups since the actual table can be fully + customized in cdr_adaptive_odbc.conf. (closes issue + ASTERISK-23233) Reported by: jmls Review: + https://reviewboard.asterisk.org/r/3227/ ........ Merged + revisions 409885 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 18:47 +0000 [r409888] Mark Michelson + + * funcs/func_presencestate.c, /: Fix documentation for + PRESENCE_STATE to properly illustrate how to create a presence + hint. There was a missing comma. This was discovered by Dan + Kaplan. ........ Merged revisions 409886 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409887 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 16:58 +0000 [r409836] David M. Lee + + * main/config.c, /, configure, include/asterisk/autoconfig.h.in, + configure.ac: Corrected cross-platform stat nanosecond code When + nanosecond time resolution was added for identifying config file + changes, it didn't cover all of the myriad of ways that one might + obtain nanosecond time resolution off of struct stat. Rather than + complicate the #if even further figuring out one system from the + next, this patch directly tests for the three struct members I + know about today, and #ifdef's accordingly. Review: + https://reviewboard.asterisk.org/r/3273/ ........ Merged + revisions 409833 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409834 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409835 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 16:26 +0000 [r409831-409832] Moises Silva + + * res/res_http_websocket.c: Fix res/res_http_websocket.c build + failure in 32bit due to incorrect print format for uint64_t + + * res/res_http_websocket.c, /: Fix WebRTC over WSS not working + Several fixes for the WebSockets implementation in + res/res_http_websocket.c * Flush the websocket session FILE* as + fwrite() may not actually guarantee sending the data to the + network. If we do not flush, it seems that buffering on the SSL + socket for outbound messages causes issues * Refactored + ast_websocket_read to take into account that SSL file descriptors + may be ready to read via fread() but poll() will not actually say + so because the data was already read from the network buffers and + is now in the libc buffers (closes issue ASTERISK-23099) (closes + issue ASTERISK-21930) Review: + https://reviewboard.asterisk.org/r/3248/ ........ Merged + revisions 409681 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409697 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 12:06 +0000 [r409780] Sean Bright + + * contrib/scripts/astgenkey, contrib/scripts/astgenkey.8, /: Fix + references to 'keys' CLI commands in astgenkey ........ Merged + revisions 409777 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409778 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409779 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-05 06:17 +0000 [r409747] Igor Goncharovskiy + + * channels/chan_unistim.c: Add update_peer function to + unistim_rtp_glue, improve other unistim_rtp_glue functions + conforming to other channel drivers. Do not forget auto-detected + and user-selected phone settings on 'unistim reload' ........ + Merged revisions 409705 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409745 from + http://svn.asterisk.org/svn/asterisk/branches/11 + +2014-03-05 01:05 +0000 [r409683] Richard Mudgett + + * /, include/asterisk/stasis_internal.h: stasis: Made + internal_stasis_subscribe() prototype and definition match + exactly. ........ Merged revisions 409682 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 19:34 +0000 [r409627] Michael L. Young + + * funcs/func_audiohookinherit.c, /: func_audiohookinheritance: + Check If A Channel Was Specified This patch prevents a crash when + using the function audiohookinheritance without setting the + channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal + Tested by: Joel Vandal Patches: + asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/3272/ ........ Merged + revisions 409623 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409625 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409626 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 17:22 +0000 [r409587] Jonathan Rose + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix one way audio + problems with hold/unhold when using ICE ICE sessions will now be + restarted if sessions are changed to use new sets of remote + candidates. (closes issue ASTERISK-22911) Reported by: Vytis + Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/ + ........ Merged revisions 409565 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409570 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 16:55 +0000 [r409569] Kinsey Moore + + * /, main/astobj2.c: AO2: Add an assert for bad objects This adds + an assert that will only be active if Asterisk is compiled with + DO_CRASH and allows the testsuite to fail tests that would + otherwise require log file parsing. ........ Merged revisions + 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 409567 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409568 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-04 14:55 +0000 [r409475] Sean Bright + + * /, channels/chan_sip.c: Minor whitespace change to 'sip show + peers' output. (closes issue ASTERISK-23406) Reported by: ibercom + Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom + ........ Merged revisions 409472 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409473 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409474 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-03 19:44 +0000 [r409423] Joshua Colp + + * /, res/res_stasis_recording.c: res_stasis_recording: Fix memory + leak of the absolute name. ........ Merged revisions 409422 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-03 02:08 +0000 [r409364] Matthew Jordan + + * main/asterisk.c, /: doxygen: Tweak the link back to ye olde + Digium website ........ Merged revisions 409361 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409362 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409363 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-02 17:03 +0000 [r409350] Tzafrir Cohen + + * /, Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a + legal option of gcc. Unofficially gcc considers it to be + equivalent of -O3. clang chalks on it, though. This commit sets + the default optimization flag to be -O3, like gcc actually + considered it. Review: https://reviewboard.asterisk.org/r/3280/ + ........ Merged revisions 409308 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409344 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409346 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-01 20:28 +0000 [r409288] Joshua Colp + + * res/res_pjsip_session.c, /: res_pjsip_session: Set options + (100rel, timers) on incoming sessions. This change passes options + to the UAS creation function. This in turn sets up 100rel and + session timer properties on the incoming session. Reported by + Julian Russell on asterisk-users mailing list. ........ Merged + revisions 409287 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-03-01 00:05 +0000 [r409257-409275] Richard Mudgett + + * /, main/devicestate.c: devicestate.c: Simplified some logic in + _ast_device_state(). ........ Merged revisions 409274 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_cache.c, /: stasis_cache.c: Remove some unnecessary + RAII_VAR() usage. ........ Merged revisions 409272 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis.c, /: stasis.c: Misc code cleanups. * Remove some + unnecessary RAII_VAR() usage. * Made the struct + stasis_subscription ao2 object use the ao2 lock instead of a + redundant join_lock in the struct for ast_cond_wait(). * Removed + locks on some ao2 objects that don't need the lock. * Made the + topic pool entries container use the ao2 template functions. * + Add some missing allocation failure checks. * Add missing cleanup + in off nominal path of dispatch_message(). ........ Merged + revisions 409270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Add precautionary p->owner + checks. * Add precautionary p->owner checks in sip_hangup(), + get_refer_info(), get_also_info(), and + interpret_t38_parameters(). * Simplify some tangled logic in + get_refer_info(), get_also_info(), and add_rpid(). * Removed some + dead code in handle_request_invite(). (closes issue + ASTERISK-23323) Reported by: Walter Doekes Patches: + issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) + uploaded by wdoekes (modified) + issueA23323-more_p_owner_checks-11.x.patch (license #5674) + uploaded by wdoekes (modified) + issueA23323-more_p_owner_checks-12.x.patch (license #5674) + uploaded by wdoekes (modified) + issueA23323-more_p_owner_checks-trunk.patch (license #5674) + uploaded by wdoekes (modified) ........ Merged revisions 409207 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 409255 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409256 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-28 21:24 +0000 [r409237] Kinsey Moore + + * apps/app_queue.c, /: app_queue: Fix documented AMI event name + During the rewrite of AMI events to use the Stasis bus, the name + of the QueueMemberPaused event was changed to QueueMemberPause. + This corrects documentation to reflect that. ........ Merged + revisions 409234 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-28 18:03 +0000 [r409159] Richard Mudgett + + * /, channels/chan_sip.c: chan_sip: Fix crash in + ast_channel_hangupcause_set(). * Fix crash in + ast_channel_hangupcause_set() because p->owner not checked before + calling. Regression introduced by the fix for ASTERISK-22621. + (closes issue ASTERISK-23135) Reported by: OK (issue + ASTERISK-23323) Reported by: Walter Doekes ........ Merged + revisions 409156 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409157 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409158 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 19:54 +0000 [r409132] Jonathan Rose + + * res/res_rtp_asterisk.c, /: Multiple revisions 409129-409130 + ........ r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb + 2014) | 15 lines res_rtp_asterisk: Fix checklist creating + problems in ICE sessions Prior to this patch, local candidate + lists including SRFLX would fail to start properly when building + ICE candidate check lists. This patch fixes that problem by + making sure that each SRFLX candidate is associated with the + proper base address so that the check list can create matches + properly. This patch was written by jcolp. The issue will be left + open to await testing by the issue participants. (issue + ASTERISK-23213) Reported by: Andrea Suisani Review: + https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose + | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines + res_rtp_asterisk: correct build error from r409129 Accidentally + placed a declaration below functional code (issue ASTERISK-23213) + Reported by: Andrea Suisani Review: + https://reviewboard.asterisk.org/r/3256/ ........ Merged + revisions 409129-409130 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409131 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 16:26 +0000 [r409091] David M. Lee + + * utils/astman.c, /: Fix memory stomping bug in astman. This memset + complained in dev mod on my Ubuntu box. The memset is both + unnecessary and dangerous. At this point, m hasn't been + initialized yet, so the memset will write off to whatever address + happens to be on the stack at the time. ........ Merged revisions + 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 409083 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409087 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 16:08 +0000 [r409055] Corey Farrell + + * /, configs/res_fax.conf.sample: res_fax: Comment out default + settings from res_fax.conf. Comment out many settings in + res_fax.conf.sample. The defaults are set in res_fax.c, so + setting the same value in sample config does nothing but make the + sample config more fragile. (closes issue ASTERISK-23231) + Reported by: David Brillert Review: + https://reviewboard.asterisk.org/r/3261/ ........ Merged + revisions 409052 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 409053 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 409054 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-27 12:29 +0000 [r409000] Matthew Jordan + + * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Apply + packetization rules on inbound SDP handling The setting + 'use_ptime' is supposed to tell Asterisk to honour the ptime + attribute in an offer, preferring it to whatever packetization + preferences have been set internally. Currently, however, + something rather quirky will happen: (1) The SDP answer will be + constructed in create_outgoing_sdp_stream. This will use the + preferences from the endpoint, such that the 200 OK response will + add the packetization preferences from the endpoint, and not what + was offered. (2) When the 200 response is issued, + apply_negotiated_sdp_stream is called. This will call + apply_packetization, which will use the ptime attribute from the + offer internally. We end up telling the offerer to use the + internal ptime attribute, but we end up using the offered ptime + attribute. Hilarity ensues. This patch modifies the behaviour by + calling apply_packetization from negotiate_incoming_sdp_stream, + which is called prior to create_outgoing_sdp_stream. This causes + the format preferences on the session's media object to be set to + the inbound ptime value (if 'use_ptime' is enabled), such that + the construction of the answer gets the right value immediately. + Review: https://reviewboard.asterisk.org/r/3244/ ........ Merged + revisions 408999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 23:35 +0000 [r408984] Richard Mudgett + + * /, tests/test_stasis.c: test_stasis.c: Misc cleanups. * Make the + consumer ao2 object use the ao2 lock instead of a redundant lock + in the struct for ast_cond_wait(). * Fixed some curly brace + placements. * Fixed use of malloc(0). malloc(0) has variant + behavior. It is up to the implementation to determine if it + returns NULL or a valid pointer that can be later passed to + free(). ........ Merged revisions 408983 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 19:00 +0000 [r408971] Scott Griepentrog + + * channels/chan_pjsip.c, /: pjsip: avoid edge case potential crash + in answer() When accidentally compiling against a wrong version + of pjsip headers with a different pjsip_inv_session size, the + invite_tsx structure could be null in the answer() function. This + led to a crash because it attempted to send the session response + with an uninitialized packet pointer. This patch presets packet + to null and adds a diagnostic log message to explain why the call + fails. Review: https://reviewboard.asterisk.org/r/3267/ ........ + Merged revisions 408970 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 17:04 +0000 [r408958] Joshua Colp + + * res/res_ari.c, /: res_ari: Make some additional error responses + consistent with the rest of the system. This change makes some + error cases use ast_ari_response_error to construct their error + responses instead of manually doing it. This ensures they are + consistent with the other error responses. Based on the original + patch as done by Paul Belanger on the associated review. Review: + https://reviewboard.asterisk.org/r/2904/ ........ Merged + revisions 408957 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 13:47 +0000 [r408942-408944] Kinsey Moore + + * include/asterisk/res_pjsip_session.h, /: PJSIP: Fix some bad + spacing ........ Merged revisions 408943 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_refer.c: PJSIP: Prevent crash if channel has + gone away It is currently possible for an ast_sip_session to + exist without an associated channel as is the case when a new + invite is coming in or just after a hangup is issued on a + chan_pjsip channel. Part of the attended transfer code assumed + the channel would be non-NULL and used it as such causing a + crash. This bug was exposed thanks to the attended transfer ARI + test in the test suite. (closes issue ASTERISK-23287) Reported + by: Matt Jordan ........ Merged revisions 408941 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-26 08:57 +0000 [r408932] Igor Goncharovskiy + + * channels/chan_unistim.c: Implement functions handling keypress, + display icons and text for i2004 KEM support. + +2014-02-25 17:51 +0000 [r408881-408883] Kevin Harwell + + * res/res_pjsip_exten_state.c, /, + res/res_pjsip_pidf_digium_body_supplement.c (added), + include/asterisk/res_pjsip_body_generator_types.h: + res_pjsip_exten_state: Presence for digium phones Added presence + support for digium phones. Review: + https://reviewboard.asterisk.org/r/3239/ ........ Merged + revisions 408882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_send_to_voicemail.c (added), + res/res_pjsip_header_funcs.c: res_pjsip_send_to_voicemail: + transferring to voicemail for digium phones Added the ability for + transferring directly to voicemail on digium phones. Added a new + module that checks for the presence of a custom header and/or + diversion header within a sip REFER. If either is found and they + specify a sending to voicemail action then variables are added to + the channel allowing the user access to them in the dialplan. + Dialplan can then be written that branches based upon these + values allowing, for instace, for a single number to be used for + dialing and/or accessing voicemail directly. Also fixed a problem + where the PJSIP_HEADER function was allowing non pjsip channels + through (checked to make sure it has the correct channel type + before proceeding). Review: + https://reviewboard.asterisk.org/r/3245/ ........ Merged + revisions 408880 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-25 17:44 +0000 [r408879] Rusty Newton + + * configs/voicemail.conf.sample, /: configs/voicemail.conf.sample - + Make mailcmd sample text more explicit Made the wording a bit + more explicit. Didn't really change the meaning. ........ Merged + revisions 408876 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408877 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408878 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-22 23:31 +0000 [r408859] Matthew Jordan + + * /, main/asterisk.c: main: Initialize dialplan providing core + components prior to module pre-load It is possible to pre-load + pbx_config. As a result, pbx_config - which will load and parse + the dialplan - will attempt to use various dialplan components, + such as device state providers and presence state providers, + prior to them being initialized by the core. This would lead to a + crash, as the components had not created their Stasis cache + entries. This patch moves a number of core component + initializations before the module pre-load. This guarantees that + if someone does pre-load pbx_config - or other pbx modules - that + the Stasis caches for the various core components are created. + (closes issue ASTERISK-23320) Reported by: xrobau (closes issue + ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy, + Rusty Newton ........ Merged revisions 408855 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-22 18:01 +0000 [r408840] Alexandr Anikin + + * addons/chan_ooh323.c, /: ignore AST_CONTROL_PVT_CAUSE_CODE + without any messages (closes issue ASTERISK-23336) Reported by: + Alexander Semych ........ Merged revisions 408838 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408839 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-22 02:31 +0000 [r408788] Corey Farrell + + * /, utils/extconf.c, utils/conf2ael.c, res/ael/pval.c, main/pbx.c: + Remove extra defines of AST_PBX_MAX_STACK. * Ensure + AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix + incorrect function parameters in utils/extconf.c. (closes issue + ASTERISK-23141) Reported by: Maxim Review: + https://reviewboard.asterisk.org/r/3241/ ........ Merged + revisions 408785 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408786 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408787 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 18:37 +0000 [r408731] Kevin Harwell + + * main/rtp_engine.c, /: rtp_engine: Dynamic payload change in rtp + mapping not supported Asterisk didn't support the dynamic payload + change in rtp mapping in the 200 OK response. Scenario: Asterisk + sends the INVITE proposing alaw and telephone-event, it proposes + rtpmap:101 for telephone-event. Peer responds with 2xx, it + answers with alaw and telephone-event also, but it proposes a + different rtpmap number (rtpmap:103) for telephone-event. + Expected Behaviour: Asterisk should honour the rtpmapping in the + response and send DTMF packets using 103 as payload type for + DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload + type 101. With this patch asterisk now supports changes that can + occur in the rtp mapping in the response. (closes issue + ASTERISK-23279) Reported by: NITESH BANSAL Review: + https://reviewboard.asterisk.org/r/3225/ Patches: + dynamic_payload_change.patch uploaded by nbansal (license 6418) + ........ Merged revisions 408729 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408730 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 18:19 +0000 [r408712-408723] Richard Mudgett + + * main/manager.c, /: manager: Fix AMI Status action of a single + channel. Fixed use of uninitialized ao2 container iterator in an + off-nominal condition. Either a memory allocation error or the + requested channel is an internal channel not exposed to the + outside. ........ Merged revisions 408715 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/sorcery.c, res/ari/resource_endpoints.c, /, + apps/app_meetme.c, res/res_fax.c, res/res_stasis_recording.c, + main/stasis_channels.c, res/res_sorcery_astdb.c, + include/asterisk/json.h: json: Fix off-nominal json ref counting + issues. * Fixed off-nominal json ref counting issue with using + the following API calls: ast_json_object_set() and + ast_json_array_append(). * Fixed off-nominal error reporting in + ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal + json ref counting issues in report_receive_fax_status() and + dial_to_json(). ........ Merged revisions 408713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/json.c, /: json: Fix json API wrapper code for json library + versions earlier than 2.3.0. * Fixed json ref counting issue with + json API wrapper code for ast_json_object_update_existing() and + ast_json_object_update_missing() when the json library is earlier + than version 2.3.0. ........ Merged revisions 408711 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 16:49 +0000 [r408699] Corey Farrell + + * channels/chan_sip.c: chan_sip: prevent add_route from adding + empty header. Fix regression caused by ASTERISK-22582. Empty + Route headers were added when the route had a single strict hop. + (closes issue ASTERISK-23306) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3236/ + +2014-02-21 16:27 +0000 [r408645-408652] Kevin Harwell + + * main/rtp_engine.c, /: rtp_engine: Output mixup in + ${CHANNEL(rtpqos,audio,all)} Fixed the output of + CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter. + (closes issue ASTERISK-23261) Reported by: rsw686 Patches: + rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged + revisions 408646 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408647 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408649 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel.c, /: channel.c: MOH is not working for transferee + after attended transfer Updated the code to check to see if MOH + is playing on the transferor and if so then start it on the + channel that replaces it during a masquerade. Example scenario of + the problem: Alice calls Bob and then Bob begins the attended + transfer process into a queue. Upon going on hold Alice hears + music and so does Bob once he is in the queue. Bob then transfers + Alice into the queue and then music for Alice stops even though + she should be hearing it since has now replaced Bob in the queue. + The problem that was occurring is that once the channel was + masqueraded the app (queues, confbridge, etc...) had no way of + knowing that the channel had just been swapped out thus it did + not start music for the present channel. Credit to Olle Johansson + for pointing me in the right direction on this issue. (closes + issue ASTERISK-19499) Reported by: Timo Teräs Review: + https://reviewboard.asterisk.org/r/3226/ ........ Merged + revisions 408642 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408643 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408644 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 10:45 +0000 [r408592] Alexandr Anikin + + * /, addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay + variables ........ Merged revisions 408589 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408590 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408591 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-21 00:50 +0000 [r408539] Michael L. Young + + * /, apps/app_chanspy.c: app_chanspy: Documentation Update To + Clarify "x" Option When using the "x" option (specify a DTMF + digit to exit the application), it is not obvious in the + documentation that this only works when spying on a channel. If a + channel being used to spy on other channels is waiting to connect + to a channel or is no longer attached to a channel, the DTMF is + ignored. As noted on the issue tracker, since there are + workarounds available and this is a rarely used option we are + opting for a documentation change here. (closes issue + ASTERISK-22661) Reported by: Chris Hillman Patches: + asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2990/ ........ Merged + revisions 408536 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408537 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408538 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-20 21:12 +0000 [r408519-408523] George Joseph + + * /, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c: pjsip_cli: Add pjsip + commands 'show registrations' and 'show contacts'. Added 'show + registrations' and 'show contacts' to pjsip cli to make things a + little more consistent. The output is exactly the same as the + list command. Just needed to add entries to their respective + ast_cli_entry structures. (closes issue ASTERISK-23275) Review: + http://reviewboard.asterisk.org/r/3210/ ........ Merged revisions + 408522 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_cli.c, main/config.c: pjsip_cli: Fix + memory leak in ast_sip_cli_print_sorcery_objectset. Fixed memory + leaks in ast_sip_cli_print_sorcery_objectset and + ast_variable_list_sort. (closes issue ASTERISK-23266) Review: + http://reviewboard.asterisk.org/r/3200/ ........ Merged revisions + 408520 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/sorcery.h, + res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c, + tests/test_sorcery.c, main/sorcery.c, /, + res/res_pjsip/config_system.c: sorcery: Create sorcery instance + registry. In order to retrieve an arbitrary sorcery instance from + a dialplan function (or any place else) there needs to be a + registry of sorcery instances. ast_sorcery_init now creates a + hashtab as a registry. ast_sorcery_open now checks the hashtab + for an existing sorcery instance matching the caller's module + name. If it finds one, it bumps the refcount and returns it. If + not, it creates a new sorcery instance, adds it to the hashtab, + then returns it. ast_sorcery_retrieve_by_module_name is a new + function that does a hashtab lookup by module name. It can be + called by the future dialplan function. res_pjsip/config_system + needed a small change to share the main res_pjsip sorcery + instance. tests/test_sorcery was updated to include a test for + the registry. (closes issue ASTERISK-22537) Review: + http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions + 408518 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-20 19:02 +0000 [r408503] Matthew Jordan + + * res/res_pjsip.c, /: res_pjsip: Update documentation for + 'use_avpf' option When 'use_avpf' is set to True, inbound offers + must use the AVPF/SAVPF RTP profile. However, when 'use_avpf' is + set to False, Asterisk will accept both AVP/SAVP or AVPF/SAVPF + RTP profiles in inbound offers. The documentation previously + implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was + set to False and a UA offered said profile in an INVITE request. + ........ Merged revisions 408502 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-20 02:44 +0000 [r408450] Rusty Newton + + * /, apps/app_queue.c: apps/app_queue - Fix incorrect Macro + parameter documentation Macro is executed on the called channel, + not the calling channel. (closes issue ASTERISK-23069) Reported + By: Bryan Anderson ........ Merged revisions 408447 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408448 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408449 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-19 19:09 +0000 [r408386-408390] Richard Mudgett + + * /, main/config.c: config: Add file size and nanosecond resolution + fields to the cached modified config file information. Repeatedly + modifying config files and reloading too fast sometimes fails to + reload the configuration because the cached modification + timestamp has one second resolution. * Added file size and + nanosecond resolution fields to the cached config file + modification timestamp information. Now if the file size changes + or the file system supports nanosecond resolution the modified + file has a better chance of being detected for reload. * Added a + missing unlock in an off-nominal code path. (closes issue + AST-1303) Review: https://reviewboard.asterisk.org/r/3235/ + ........ Merged revisions 408387 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408388 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408389 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix regex + handling and keep simple prefix matching performance. The sorcery + astDB wizzard does not handle regex correctly if the pattern + begins with an anchor character. This patch attempts to convert + the anchored regex pattern to a prefix pattern supported by astDB + for performance reasons. If it is not able to convert the pattern + it falls back to getting all astDB members of the family and + doing a normal regex pattern matching on the retrieved records. + Review: https://reviewboard.asterisk.org/r/3161/ ........ Merged + revisions 408385 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-19 12:04 +0000 [r408315-408332] Alexandr Anikin + + * addons/ooh323c/src/ooCapability.c, /, + addons/ooh323c/src/ooh245.c: process receiveAndTransmit user + input remote caps instead of receive only send receiveAndTransmit + user input our caps instead of receive only ........ Merged + revisions 408328 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408330 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408331 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * addons/ooh323c/src/ooh323.c, /: Allow different socket and + signalling ip on h.323 connection if gk mode is active Reported + by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by: + Gabriele Odone (closes issue ASTERISK-22738) ........ Merged + revisions 408312 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408314 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-18 19:19 +0000 [r408299] Richard Mudgett + + * contrib/ast-db-manage/config/env.py, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py, + contrib/ast-db-manage/config, + contrib/ast-db-manage/voicemail/env.py, + contrib/ast-db-manage/voicemail, + contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py, + contrib/ast-db-manage/config/versions, + contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py, + contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py, + contrib/ast-db-manage/voicemail/versions, contrib/ast-db-manage, + /: alembic: Add svn:ignore *.pyc to directories and + svn:executable to *.py files. ........ Merged revisions 408297 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-17 15:36 +0000 [r408272] Mark Michelson + + * /, res/res_pjsip/location.c, UPGRADE.txt, res/res_pjsip.c, + res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h: Store + SIP User-Agent information in contacts. When an endpoint sends a + REGISTER request to Asterisk, we now will associate the + User-Agent header with all contacts that were bound in that + REGISTER request. ........ Merged revisions 408270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-16 03:25 +0000 [r408199-408227] Matthew Jordan + + * /, main/pbx.c: pbx: Handle a completely empty dialplan during a + context merge It is highly unlikely, but - at least in Asterisk + 12 - theoretically possible to load Asterisk with no dialplan + whatsoever. If that occurs, and some other module (that is not a + pbx module) attempts to merge its contexts into the dialplan, the + existing merge routine will crash. This is because it is not + insane, and rightly believes that you provided some sort of + dialplan, somewhere. This patch will gracefully merge the + contexts in such a case. Note that this is highly unlikely to + occur in 1.8/11, as features will most likely provide some + dialplan via parking. However, in Asterisk 12, parking is now + provided by res_parking, and hence may create its dialplan later. + (closes issue ASTERISK-23297) Reported by: CJ Oster Review: + https://reviewboard.asterisk.org/r/3222 ........ Merged revisions + 408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 408201 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408220 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, Makefile: buildsystem: Unbreak the build (infloop) on Asterisk + 11+ Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/ + ) broke the build. This patch fixes it by ignoring the .lastclean + dependencies if the MENUSELECT_EMBED variable is not defined. + patches: tmp.diff uploaded by wdoekes (License 5674) Review: + https://reviewboard.asterisk.org/r/3228/ ........ Merged + revisions 408193 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408194 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 21:44 +0000 [r408139-408141] Scott Griepentrog + + * main/stasis_endpoints.c, /: ARI: correct upper/lower case URI + discrepancies URI's are supposed to be case sensitive and all + lower case. In practice some portions of URI's in ARI are case + insensitive and others are not, such as TECH, which in one + instance would match a lower case name and in another would not. + In this patch, the ast_endpoint_lastest_snapshot() function is + modified to change the TECH portion to full upper case before + lookup. This resolves the discrepancy noted by the reporter. + However I chose to avoid forcing the /ari prefix of the URI's to + be lower case for now. Except for the two cases here, all URI's + should be lower case, unless they are part of a resource name or + id. Review: https://reviewboard.asterisk.org/r/3211/ Reported by: + Zane Conkle (closes issue ASTERISK-23125) ........ Merged + revisions 408140 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/format.c, /: format.c: correct possible null pointer + dereference In ast_format_sdp_parse and ast_format_sdp_generate + the check checks for a valid interface and function were + potentially confusing, and hid an error in the test of the + presence of the function that is called later. This patch clears + up and corrects the test. Review: + https://reviewboard.asterisk.org/r/3208/ (closes issue + ASTERISK-23098) Reported by: marcelloceschia Patches: + main_format.patch uploaded by marcelloceschia (license 6036) + ASTERISK-23098.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 408137 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408138 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 13:31 +0000 [r408086] Walter Doekes + + * Makefile, /: buildsystem: Don't force main to depend on + everything else. Directory 'main' only needs to depend on + embedded modules. If no module embedding is selected, the + dependency is dropped. Review: + https://reviewboard.asterisk.org/r/3212/ ........ Merged + revisions 408083 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 408084 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 408085 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 12:41 +0000 [r408070] Matthew Jordan + + * /, channels/chan_sip.c: chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER + prior to calling bridge blind transfer This patch moves setting + SIP_DEFER_BY_ON_TRANSFER prior to calling + ast_bridge_transfer_blind. This prevents a BYE from being sent + prior to the NOTIFY request that informs the transferor if the + transfer succeeded or failed. This patch also clears said flag + from the off nominal NOTIFY paths in the local_attended_transfer + code, as once we've sent the NOTIFY request it is safe to send by + the BYE request. This was caught by the + blind-transfer-accountcode test in the Asterisk Test Suite. + (closes issue ASTERISK-23290) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3214/ ........ Merged + revisions 408069 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-14 08:52 +0000 [r408059] Tzafrir Cohen + + * Makefile, build_tools/install_subst (added): install_subst: + helper script for installing with path substitution A helper + script to copy a source file substituting any + __ASTERISK__DIR__ with the content of $ASTDIR. Review: + https://reviewboard.asterisk.org/r/3202/ + +2014-02-13 18:52 +0000 [r407990-408006] Mark Michelson + + * res/res_pjsip_pubsub.c, /, res/res_pjsip_mwi.c: Remove all PJSIP + MWI-specific use from our MWI code. PJSIP has built-in MWI code + that could be useful to some degree, but our utilization of the + API actually made our code a bit more cluttered since we had to + have special cases peppered throughout. With this change, we move + to using the pjsip_evsub API instead, which streamlines the code + by removing special cases. Review: + https://reviewboard.asterisk.org/r/3205 ........ Merged revisions + 408005 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/location.c: Fix crash in AMI PJSIPShowEndpoint + action. If an AOR has no permanent contacts, then the + permanent_contacts container is never allocated. This makes the + code safe in the face of NULLs. I also changed the variable that + counts contacts from "num" to "total_contacts" since there are + now two variables that are indicate numbers of things. ........ + Merged revisions 407988 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-13 15:51 +0000 [r407989] Kinsey Moore + + * main/logger.c, CHANGES: Logger: Add dynamic logger channels This + adds the ability to dynamically add and remove logger channels + from Asterisk via the CLI. (closes issue AST-1150) Review: + https://reviewboard.asterisk.org/r/3185/ + +2014-02-12 08:25 +0000 [r407970] Walter Doekes + + * /, main/config.c: realtime: Fix ast_update2_realtime() on + raspberry pi. The old code depended on undefined va_arg + behaviour: calling a function twice with the same va_list + parameter and expecting it to continue where it left off. The + changed code behaves like the manpage says it should. Also added + a bunch of early returns to trap errors (e.g. OOM) instead of + crashing. The problem was found by Julian Lyndon-Smith. The + deviant behaviour on the raspberry PI also uncovered another bug + (fixed in r407875) in the res_config_pgsql.so driver. Reported + by: jmls Tested by: jmls Review: + https://reviewboard.asterisk.org/r/3201/ ........ Merged + revisions 407968 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-11 20:17 +0000 [r407958] Joshua Colp + + * main/sched.c: scheduler: Remove hashtab usage. This is a first + stab at tweaking the performance profile of the scheduler. + Removing the hashtab usage removes an extra memory allocation + when scheduling something and makes it so rescheduling does not + incur any memory allocation at all. Review: + https://reviewboard.asterisk.org/r/3199/ + +2014-02-11 03:18 +0000 [r407940] Matthew Jordan + + * res/ari/resource_channels.c, /: ari/resource_channels: Add + channel variables earlier in the creation process This patch + tweaks the behaviour of POST /channels with channel variables + such that the variables are passed into the pbx.c routines that + perform the origination. This allows the variables to be assigned + to the newly created channels immediately upon their + construction, as opposed to be assigned after the originate has + completed. The upshot of this is that the variables are available + on the channels if they execute in the dialplan, as opposed to + only being available once the channels are answered. Review: + https://reviewboard.asterisk.org/r/3183/ ........ Merged + revisions 407937 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-10 18:28 +0000 [r407926] Corey Farrell + + * channels/sip/include/reqresp_parser.h, + channels/sip/include/route.h (added), channels/chan_sip.c, + channels/sip/route.c (added), channels/sip/include/sip.h: + chan_sip: Isolate code that manages struct sip_route. * Move + route code to sip/route.c + sip/include/route.h * Rename + functions to sip_route_* * Replace ad-hoc list code with macro's + from linkedlists.h * Create sip_route_process_header() to + processes Path and Record-Route headers (previously done with + different code in build_route and build_path) * Add use of const + where possible * Move struct uriparams, struct contact and + contactliststruct from sip.h to reqresp_parser.h. sip/route.c + uses reqresp_parser.h but not sip.h, this was a problem. These + moved declares are not used outside of reqresp_parser. * While + modifying reqprep() the lack of {} caused me trouble. I added + them. * Code outside route.c treats sip_route as an opaque + structure, using macro's or procedures for all access. (closes + issue ASTERISK-22582) Reported by: Corey Farrell Review: + https://reviewboard.asterisk.org/r/3173/ + +2014-02-10 16:49 +0000 [r407876] Walter Doekes + + * res/res_config_pgsql.c, /: res_config_pgsql: Fix + ast_update2_realtime calls. Fix so multiple updates from a single + call works (add missing ','). Remove bogus ast_free's that + weren't supposed to be there. Moved a few spaces for readability. + Review: https://reviewboard.asterisk.org/r/3194/ ........ Merged + revisions 407873 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407874 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407875 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-10 16:01 +0000 [r407859] Kinsey Moore + + * apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c, + apps/confbridge/conf_state_empty.c, + apps/confbridge/conf_config_parser.c, + configs/confbridge.conf.sample, /, + apps/confbridge/include/confbridge.h, UPGRADE.txt: ConfBridge: + Correct prompt playback target Currently, when the first marked + user enters the conference that contains waitmarked users, a + prompt is played indicating that the user is being placed into + the conference. Unfortunately, this prompt is played to the + marked user and not the waitmarked users which is not very + helpful. This patch changes that behavior to play a prompt + stating "The conference will now begin" to the entire conference + after adding and unmuting the waitmarked users since the design + of confbridge is not conducive to playing a prompt to a subset of + users in a conference in an asynchronous manner. (closes issue + PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/ + Reported by: Steve Pitts ........ Merged revisions 407857 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407858 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 20:52 +0000 [r407767] Richard Mudgett + + * /, channels/chan_iax2.c: chan_iax2: Add some more iaxs[] NULL + checks to a routine already full of them. ........ Merged + revisions 407764 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407765 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407766 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 20:17 +0000 [r407752] Matthew Jordan + + * /, main/security_events.c: security_events: Fix assertion failure + in dev-mode on optional IE parsing When formatting an optional + IE, the value is, of course, optional. As such, it is entirely + appropriate for ast_json_object_get to return NULL. If that + occurs, we now simply skip the IE that was requested, as it was + not provided by the entity that raised the event. Thanks to + George Joseph (gtjoseph) for catching this and reporting it in + #asterisk-dev ........ Merged revisions 407750 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 20:01 +0000 [r407749] Joshua Colp + + * main/timing.c, res/res_timing_pthread.c, res/res_timing_dahdi.c, + res/res_timing_timerfd.c, include/asterisk/timing.h, + res/res_timing_kqueue.c: timing: Improve performance for most + timing implementations. This change allows timing implementation + data to be stored directly on the timer itself thus removing the + requirement for many implementations to do a container lookup for + the same information. This means that API calls into timing + implementations can directly access the information they need + instead of having to find it. Review: + https://reviewboard.asterisk.org/r/3175/ + +2014-02-07 19:40 +0000 [r407748] Matthew Jordan + + * /, funcs/func_cdr.c: funcs/func_cdr: Handle empty time values + when extracting parsed values When extracting timestamps that are + parsed, time stamp values that are not set (time values of + 0.000000) should not actually result in a parsed string. The + value should be skipped, and the result of the CDR function + should be an empty string. Prior to this patch, the result was + fed to the time formatting, which would result in an output of a + date/time in 1969. ........ Merged revisions 407747 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 18:29 +0000 [r407731] Richard Mudgett + + * channels/chan_iax2.c, include/asterisk/frame.h, + configs/iax.conf.sample, /: chan_iax2: Block unnecessary control + frames to/from the wire. Establishing an IAX2 call between + Asterisk v1.4 and v1.8 (or later) results in an unexpected call + disconnect. The problem happens because newer values in the enum + ast_control_frame_type are not consistent between the branch + versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) + using IAX2 2) v1.8 answers and sends a connected line update + control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 + receives the control frame as an end-of-q (on v1.4 + AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the + receive queue becomes empty. Several things are done by this + patch to fix the problem and attempt to prevent it from happening + again in the future: * Added a warning at the definition of enum + ast_control_frame_type about how to add new control frame values. + * Made block sending and receiving control frames that have no + reason to go over the wire. * Extended the connectedline iax.conf + parameter to also include the redirecting information updates. * + Updated the connectedline iax.conf parameter documentation to + include a notice that the parameter must be "no" when the peer is + an Asterisk v1.4 instance. (closes issue AST-1302) Review: + https://reviewboard.asterisk.org/r/3174/ ........ Merged + revisions 407678 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407727 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407729 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 16:47 +0000 [r407677] Matthew Jordan + + * /, main/security_events.c: security_events: Fix error caused by + DTD validation error The appdocsxml.dtd specifies that a + "required" attribute in a parameter may have a value of yes, no, + true, or false. On some systems, specifying "False" instead of + "false" would cause a validation error. This patch fixes the + casing to explicitly match the DTD. ........ Merged revisions + 407676 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-07 13:15 +0000 [r407625] Tzafrir Cohen + + * /, configs/indications.conf.sample: indications.conf: add stutter + tone; end properly * If the "stutter" (voicemail indication) tone + is indeed a stutter tone, and it ends with a constant tone, make + sure that it is the dial tone. This was done for India (in), + Mexico (mx) and the Philippines (ph). * If no "stutter" tone + exists for a country, provide one. This was done for Spain (es), + Malaysia (my) and Venezuela (ve). Review: + https://reviewboard.asterisk.org/r/3158/ ........ Merged + revisions 407622 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407623 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 21:24 +0000 [r407602] Matthew Jordan + + * /, main/security_events.c, UPGRADE.txt, CHANGES: security_events: + Add AMI documentation; output optional fields This patch adds + documentation for the Security Events that are emited over AMI. + It also notes these events in the UPGRADE/CHANGES file. ........ + Merged revisions 407589 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 19:58 +0000 [r407588] Rusty Newton + + * /, configs/pjsip.conf.sample: configs/pjsip.conf.sample: + Configuration section naming in pjsip.conf.sample needs a little + clarification There is a bit of nuance to how you name things in + pjsip.conf. This is a documentation patch to at least clear it up + a little for users. Review: + https://reviewboard.asterisk.org/r/3180/ ........ Merged + revisions 407587 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 18:11 +0000 [r407574] Kevin Harwell + + * /, + contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py: + pjsip realtime: already created enum failure for postgresql If an + enum had been previously created the alembic script would attempt + to re-create it and an error would be generated while running + migrations for a postgresql server. The work around for this is + to use the ENUM object type for postgres as opposed to the + generic enum type used by sqlalchemy. Using this type in the + script seems to work properly for both postgres and mysql. + ........ Merged revisions 407572 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-06 17:55 +0000 [r407573] Richard Mudgett + + * res/res_pjsip_logger.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip_endpoint_identifier_ip.c, + include/asterisk/res_pjsip_cli.h, res/res_pjsip/pjsip_cli.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/config_domain_aliases.c: res_pjsip: Updates and + adds more PJSIP CLI commands. * Adds identify, transport, and + registration support to the PJSIP CLI. * Creates three additional + callbacks, one for an iterator, one for a comparator, and one for + a container. This eliminates the link dependency from higher + level modules to lower level ones. * Eliminates duplicate sorting + in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. * + Pushes CLI command registration down to the implementing source + file. * Adds several ast_sip_destroy_sorcery functions to + complement existing ast_sip_sorcery_initialize functions. The + destroy functions unregister PJSIP CLI commands and PJSIP CLI + formatters. Reported by: George Joseph Review: + https://reviewboard.asterisk.org/r/3104/ ........ Merged + revisions 407568 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 23:04 +0000 [r407514] Rusty Newton + + * /, formats/format_wav.c: formats/format_wav: enhancing log + message "Not a wav file" to be clear on what is supported + Modifying the log message to be more specific as to what is + supported. Specifically it seems format_wav supports only PCM + encoded versions with a lower-case '.wav' extension. (closes + issues ASTERISK-22310) Reported by: Jim Credland Review: + https://reviewboard.asterisk.org/r/3188/ ........ Merged + revisions 407511 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407512 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407513 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 20:56 +0000 [r407462] Jonathan Rose + + * CHANGES, /: CHANGES: Improved description of Name/Creator changes + to bridge ARI, adds AMI The changes log was written with language + that was a little too internal Asterisk specific, so it's been + changed to be more in the frame of reference of an ARI user. + Also, previously the AMI event changes were omitted from the + change log as well as the ability to include a bridge name in the + ARI post bridges command. ........ Merged revisions 407461 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 20:43 +0000 [r407459] Kinsey Moore + + * main/logger.c, /: Logger: Fix handling of absolute paths This + fixes path handling for log files so that an extra / is not + appended to the file path when the path is absolute (begins with + /). This would previously result in different but functionally + equivalent paths in the output of 'logger show channels'. + ........ Merged revisions 407455 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407456 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 19:42 +0000 [r407443] Kevin Harwell + + * res/res_pjsip/config_global.c, /: res_pjsip: When no global type + the debug option defaults to "yes" If the global section was not + specified in pjsip.conf then the configuration object does not + exist in sorcery so when retrieving "debug" option it would + return NULL. Then the NULL result was passed to ast_false utils + function which would return false because it wasn't set to some + representation of false, thus enabling sip debug logging. Made it + so if the global config object does not exist then it will return + a default of "no" for sip debugging. (issue ASTERISK-23038) + Reported by: Rusty Newton ........ Merged revisions 407442 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 17:42 +0000 [r407422-407425] Jonathan Rose + + * CHANGES: CHANGES: Update changes log to include r403414 entry + Adds note of additional 0 for operator option on app_record + + * CHANGES, /: CHANGES: Update changes log to include new bridge + fields added in r404042 ........ Merged revisions 407419 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-05 15:29 +0000 [r407407] Matthew Jordan + + * rest-api/api-docs/playbacks.json, UPGRADE.txt, + rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES, + include/asterisk/manager.h, rest-api/api-docs/bridges.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/mailboxes.json, + rest-api/api-docs/asterisk.json, + rest-api/api-docs/applications.json, + rest-api/api-docs/channels.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + /: ARI/AMI: Update versions; update UPGRADE/CHANGES notes for + 12.1.0 changes Due to backwards compatible changes made to + AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0, + respectively. ........ Merged revisions 407402 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 20:15 +0000 [r407275-407340] Richard Mudgett + + * include/asterisk/devicestate.h, /, main/devicestate.c: + devicestate: Make ast_devstate_changed_literal() return value and + doxygen consistent. Nothing actually cares about the value + anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose + ........ Merged revisions 407337 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407338 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407339 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix assertion + for pjsip.conf authorization list options. (closes issue + ASTERISK-23168) Reported by: George Joseph Review: + https://reviewboard.asterisk.org/r/3143/ ........ Merged + revisions 407324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/sip.conf.sample, main/tcptls.c, /: tcptls.c: Made TLS + handle a certificate chain file. Thanks to Guillaume Martres for + doing the necessary research to validate the change. (closes + issue ASTERISK-17727) Reported by: LN Patches: + use_certificate_chain.patch (license #5864) patch uploaded by st + documente_certificate_chain.patch (license #6576) patch uploaded + by Guillaume Martres ........ Merged revisions 407272 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407273 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407274 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 16:55 +0000 [r407260] Matthew Jordan + + * /, funcs/func_cdr.c: funcs/func_cdr: Fix non-epoch timestamps + broken by improper char array deref Thanks to snuffy for pointing + this issue out and fixing it. (closes issue ASTERISK-23250) + Reported by: snuffy patches: func_cdr-fix.diff uploaded by snuffy + (License 5024) ........ Merged revisions 407259 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 02:22 +0000 [r407217] Joshua Colp + + * res/res_clialiases.c, /: res_clialiases: Fix crash when reloading + and re-aliasing an alias that is in use. The code assumed that + unregistering the alias would always succeed while in practice + this is not actually true. A common case is the "reload" command + itself. If the cli_aliases.conf configuration file was changed + and reload executed the command would fail to unregister and + ultimately point to freed memory. The reload process now checks + whether unregistering succeeded or not and if not the old CLI + alias is retained. (closes issue ASTERISK-19773) Reported by: + Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth + Blades ........ Merged revisions 407205 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407210 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407213 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-04 02:07 +0000 [r407198] Damien Wedhorn + + * /, channels/chan_skinny.c: Skinny - Fix deadlock when pickup of + no call. Locking issues in skinny when picking up a call that + doesn't exist. Cleaned up sub locking by fully removing and using + the chan lock instead. Also changed ast_call_pickup to check + whether chan was masq'd. (closes issue ASTERISK-23249) Reported + by: wedhorn Tested by: snuffy, myself Patches: + skinny-locking01.diff uploaded by wedhorn (license 5019) ........ + Merged revisions 407197 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-03 01:31 +0000 [r407169] Matthew Jordan + + * main/cdr.c, /: cdrs: Check for applications to lock onto during + dial begin handling This patch brings CDR processing further in + line with r407085. During some dial operations, the application + would not be locked to the Dial application and would instead + continue to show the previously known application. In particular, + this would occur when a Parked call would time out. This was due + to a previous snapshot already locking the application to Park - + processing this in a Dial Begin allows the Dial application to + reassert its rightful place. (CDRs. Ugh.) But hooray for the + Parked Call tests for catching this in the Asterisk Test Suite. + ........ Merged revisions 407166 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-01 16:26 +0000 [r407154] Joshua Colp + + * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /, + res/stasis/app.c, res/ari/ari_model_validators.c, + res/res_stasis.c, main/stasis_bridges.c: res_stasis: Enable + transfers and provide events when they occur. This change enables + transfers within ARI created bridges and adds events for when + they occur. Unlike other events these will be received if *any* + subscribed object is involved in the transfer. (closes issue + ASTERISK-22984) Reported by: David M. Lee Review: + https://reviewboard.asterisk.org/r/3120/ ........ Merged + revisions 407153 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-02-01 00:25 +0000 [r407105] Corey Farrell + + * apps/app_stack.c, /: app_stack: protect against missing + parameters to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2 + parameters and LOCAL_PEEK requires 1 parameter. This protects + against situations where those parameters are blank or missing by + logging an error and returning. (closes issue ASTERISK-23220) + Reported by: James Sharp ........ Merged revisions 407100 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 407103 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407104 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 23:40 +0000 [r407083-407085] Matthew Jordan + + * apps/app_dial.c, main/cdr.c, main/pbx.c, /, main/bridge_after.c, + UPGRADE.txt, main/manager_channels.c: CDRs: fix a variety of dial + status problems, h/hangup handler creating CDRs This patch fixes + a number of small-ish problems that were noticed when witnessing + the records that the FreePBX dialplan produces: (1) Mid-call + events (as well as privacy options) have the ability to change + the overall state of the Dial operation after the called party + answers. This means that publishing the DialEnd event when the + called party is premature; we have to wait for the execution of + these subroutines to complete before we can signal the overall + status of the DialEnd. This patch moves that publication and adds + handlers for the mid-call events. (2) The AST_FLAG_OUTGOING + channel flag is cleared if an after bridge goto datastore is + detected. This flag was preventing CDRs from being recorded for + all outbound channels that had a 'continue' option enabled on + them by the Dial application. (3) The CDR engine now locks the + 'Dial' application as being the CDR application if it detects + that the current CDR has entered that app. This is similar to the + logic that is done for Parking. In general, if we entered into + Dial, then we want that CDR to record the application as such - + this prevents pre-dial handlers, mid-call handlers, and other + shenaniganry from changing the application value. (4) The CDR + engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more + places to determine if the channel is in hangup logic or dead. In + either case, we don't want to record changes in the channel. (5) + The default option for "endbeforehexten" has been changed to + "yes". In general, you don't want to see CDRs in the 'h' exten or + in hangup logic. Since the semantics of that option changed in + 12, it made sense to update the default value as well. (6) + Finally, because we now have the ability to synchronize on the + messages published to the CDR topic, on shutdown the CDR engine + will now synchronize to the messages currently in flight. This + helps to ensure that all in-flight CDRs are written before + shutting down. (closes issue ASTERISK-23164) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3154 ........ + Merged revisions 407084 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_dial.c, /: app_dial: Allow macro/gosub pre-bridge + execution to occur on priorities The parsing for the destination + of the macro/gosub uses the '^' character to separate out + context, extension, and priority. However, the logic for the + macro/gosub execution was written such that it would only do the + actual macro/gosub jump if a '^' character existed. This doesn't + apply when the macro/gosub jump occurs in a priority/priority + label. This patch changes the logic so that the parsing still + occurs, but the jump will occur even for priorities/priority + labels. (issue ASTERISK-23164) Review: + https://reviewboard.asterisk.org/r/3154 ........ Merged revisions + 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 407074 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 407082 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 23:15 +0000 [r407035-407037] Kevin Harwell + + * res/res_pjsip_logger.c, CHANGES, res/res_pjsip.c, + include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, + contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py + (added), /, configs/pjsip.conf.sample, UPGRADE.txt: res_pjsip: + Config option to enable PJSIP logger at load time. Added a + "debug" configuration option for res_pjsip that when set to "yes" + enables SIP messages to be logged. It is specified under the + "system" type. Also added an alembic script to add the option to + realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton + Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged + revisions 407036 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_exten_state.c, /: res_pjsip_exten_state: Exporting + global symbols caused load order issues Removed the exportation + of global symbols from the module as it is no longer needed and + it could potentially cause load problems as on some systems it + would try to load before res_pjsip_pubsub ........ Merged + revisions 407034 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 23:04 +0000 [r407033] Richard Mudgett + + * CHANGES, apps/app_chanspy.c: ChanSpy: Add ability to specify + channel uniqueids as well as channel names. * Made ChanSpy accept + a channel uniqueid or a fully specified channel name as the + chanprefix parameter if the 'u' option is specified. (closes + issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/ + +2014-01-31 22:39 +0000 [r407030-407032] Mark Michelson + + * include/asterisk/res_pjsip_presence_xml.h (added), /: Add file + that apparently got missed in the merge. ........ Merged + revisions 407031 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pidf_body_generator.c (added), + include/asterisk/res_pjsip_exten_state.h (removed), + res/res_pjsip_pubsub.exports.in, /, + include/asterisk/res_pjsip_body_generator_types.h (added), + res/res_pjsip_mwi.c, res/res_pjsip_xpidf_body_generator.c + (added), res/res_pjsip_mwi_body_generator.c (added), + res/res_pjsip_pubsub.c, res/res_pjsip_pidf.c (removed), + res/res_pjsip_pidf_eyebeam_body_supplement.c (added), + res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c + (added), include/asterisk/res_pjsip_pubsub.h: Decouple + subscription handling from NOTIFY/PUBLISH body generation. When + the PJSIP pubsub framework was created, subscription handlers + were required to state what event they handled along with what + body types they knew how to generate. While this serves well when + implementing a base RFC, it has problems when trying to extend + the body to support non-standard or proprietary body elements. + The code also was NOTIFY-specific, meaning that when the time + comes that we start writing code to send out PUBLISH requests + with MWI or presence bodies, we would likely find ourselves + duplicating code that had previously been written. This changeset + introduces the concept of body generators and body supplements. A + body generator is responsible for allocating a native structure + for a given body type, providing the primary body content, + converting the native structure to a string, and deallocating + resources. A body supplement takes the primary body content (the + native structure, not a string) generated by the body generator + and adds nonstandard elements to the body. With these elements + living in their own module, it becomes easy to extend our support + for body types and to re-use resources when sending a PUBLISH + request. Body generators and body supplements register themselves + with the pubsub core, similar to how subscription and publish + handlers had done. Now, subscription handlers do not need to know + what type of body content they generate, but they still need to + inform the pubsub core about what the default body type for a + given event package is. The pubsub core keeps track of what body + generators and body supplements have been registered. When a + SUBSCRIBE arrives, the pubsub core will check that there is a + subscription handler for the event in the SUBSCRIBE, then it will + check that there is a body generator that can provide the content + specified in the Accept header(s). Because of the nature of body + generators and supplements, it means res_pjsip_exten_state and + res_pjsip_mwi have been completely gutted. They no longer worry + about body types, instead calling + ast_sip_pubsub_generate_body_content() when they need to generate + a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150 + ........ Merged revisions 407016 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 22:23 +0000 [r407015-407029] Kevin Harwell + + * contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py, + contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py, + /, UPGRADE.txt: alembic: script modifications due to errors A + couple of the scripts had errors that would not allow a full + migration to take place. The extensions table needed to make its + 'id' column a primary key in order to work with mysql. The other + script ...add_endpoints... was missing tables that it was trying + to add columns to. Added the primary key on id for extensions and + added the tables in for the missing pjsip configuration options. + While it is not ideal to modify already released scripts this was + a case where it had to be done due to errors in the script and + lacking a better alternative. Review: + https://reviewboard.asterisk.org/r/3167/ ........ Merged + revisions 407019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c: res_pjsip_mwi: Subscribe fails when + missing aor name When subscribing to MWI (res_pjsip_mwi) and the + sip uri did not contain a name (ex: sip:) then the + subscription would fail since it would be unable to locate an + associated aor. This patch makes it so that when a subscribe + comes with no aor name then it will subscribe to all aors on the + located endpoint. (closes issue ASTERISK-23072) Reported by: Bob + M Review: https://reviewboard.asterisk.org/r/3164/ ........ + Merged revisions 407014 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 15:08 +0000 [r407001] Kinsey Moore + + * res/res_pjsip_nat.c, /: PJSIP: Fix address for ACK in NAT + situations In NAT scenarios where a call is placed to a + Grandstream phone, res_pjsip will sometimes send the ACK to a 200 + OK to the private address of the device behind the NAT instead of + the address of the NAT device. This corrects that behavior by + rewriting the address in the Contact header in the incoming 200 + OK and the dialog's target address if necessary (since it has + already been rewritten to the incorrect private address). (closes + issue ASTERISK-23106) Review: + https://reviewboard.asterisk.org/r/3168/ Reported by: Matt Jordan + ........ Merged revisions 407000 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-31 05:31 +0000 [r406988] Damien Wedhorn + + * /, channels/chan_skinny.c: Skinny: fix up possible double unlock + of chan. Return before chan is possibly unlocked a second time + when hanging up a channel in SUBSTATE_OFFHOOK. ........ Merged + revisions 406987 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-30 20:36 +0000 [r406936] Corey Farrell + + * main/udptl.c, res/res_rtp_asterisk.c, /: res_rtp_asterisk & + udptl: fix port selection to work with SELinux restrictions + ast_bind to a port reserved for another program by SELinux causes + errno == EACCES. This caused random failures when binding rtp or + udptl sockets. Treat EACCES as a non-fatal error, try next port. + (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ + Merged revisions 406933 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406934 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406935 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-30 17:35 +0000 [r406920] Sean Bright + + * main/manager.c, /: Make a NOTICE about an invalid channel name + more useful. ........ Merged revisions 406918 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406919 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-29 00:44 +0000 [r406863] Russell Bryant + + * /, configs/queues.conf.sample: queues.conf.sample Fix documented + default for persistentmembers Closes issue ASTERISK-22662 + ........ Merged revisions 406860 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406861 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406862 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 23:40 +0000 [r406789-406848] Kevin Harwell + + * res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: potential crash on + timeout What seems to be happening is if a subscription has been + terminated and the subscription timeout/expires is less than the + time it takes for all pending transactions (currently on the + subscription) to end then the subscription timer will not have + been canceled yet and sub will be null. Since the subscription + has already been canceled nothing needs to be done so a null + check in the asterisk code is sufficient in working around this + problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins + ........ Merged revisions 406847 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * cdr/cdr_radius.c, cel/cel_radius.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: cdr_radius, + cel_radius: build agains libfreeradius-client Asterisk's RADIUS + module currently build against libradiusclient-ng, but this + project has been superseeded by libfreeradius-client. The API is + 99% compatible except that the header name has changed, the + library name has changed, and the configuration file location has + changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé + Patches: freeradius-client.patch uploaded by sharky (license + 6561) ........ Merged revisions 406801 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406802 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406803 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/include/res_pjsip_private.h, /, + include/asterisk/compat.h: res_pjsip,compat: INFINITY and NAN + undefined On some systems the values for INFINITY and NAN are not + defined thus causing a build error on those systems. Added + definitions for those if they had not previously been defined. + (closes issue ASTERISK-23056) Reported by: capouch Patches: + inf-nan-patch.txt uploaded by capouch (license 6564) ........ + Merged revisions 406788 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 19:19 +0000 [r406778] Kinsey Moore + + * /, res/res_stasis_device_state.c: ARI: Make double subscribe + respond with success Currently, attempting to subscribe an + application to a device state that it has already subscribed to + will generate a 500 error response. This will now be treated as a + subscription refresh even though ARI subscriptions don't + currently support lifetimes and will respond with the normal + response for a successful subscription (200 OK). (closes issue + ASTERISK-23143) Reported by: Matt Jordan ........ Merged + revisions 406775 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 16:43 +0000 [r406724] Scott Griepentrog + + * main/rtp_engine.c, /: rtp_engine: improved handling of + get_rtp_info failure In ast_rtp_instance_make_compatible(), after + a failure of channel tech call get_rtp_info() to return + peer_instance, the null pointer would be passed to ao2_ref, + producing an error that looked like a refernce counting problem + but is not. This patch corrects that and adds helpful LOG_ERROR + messages to indicate which failure path occurred. (issue + AST-1276) Review: https://reviewboard.asterisk.org/r/3156/ + ........ Merged revisions 406721 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406722 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406723 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-28 00:20 +0000 [r406710] Richard Mudgett + + * /, tests/test_cel.c, tests/test_cdr.c: test_cdr.c, test_cel.c: + Correctly destroy created bridges. * Fixed the + test_cel_attended_transfer_bridges_link unit test to also account + for the local channel link being destroyed now that the bridges + are actually destroyed. * Made CDR unit test use its own version + of do_sleep() from the CEL unit tests. ........ Merged revisions + 406707 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-27 22:54 +0000 [r406647-406696] Kevin Harwell + + * CHANGES: manager: ExtensionStatus event status human readable + Added a note in the changes file about the new 'StatusText' field + that was added to the 'ExtensionStatus' event. (issue + ASTERISK-23154) Reported by: Jonathan Rose + + * main/manager.c: manager: ExtensionStatus event status human + readable When an 'ExtensionStatus' event was raised it included + the status as a numerical value, but did not include a text + description of the status. Added a 'StatusText' field to the + event which is a string representation of the extension status. + Also added this to the 'Extension State' command response. + (closes issue ASTERISK-23154) Reported by: Jonathan Rose + +2014-01-27 20:38 +0000 [r406646] Russell Bryant + + * main/config.c, /: Allow nested #includes in extconfig.conf + extconfig.conf was hard-coded to not allow nested includes for + some reason. The code has been this way since a patch was merged + for ASTERISK-3333 (revision 4889), which was a significant update + to this code ("Merge config updates"). I can't figure out any + good reason why this should be limited. This patch just removes + the limit and uses the default nesting depth limit. Closes issue + ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/ + ........ Merged revisions 406643 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406644 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406645 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-27 08:17 +0000 [r406618] Walter Doekes + + * main/manager.c, UPGRADE.txt, configs/manager.conf.sample: + manager: The eventfilter= option now takes an extended regex. In + pre-trunk versions (...12) it accepts a basic regex, which is + confusing because all other regexes in asterisk are of the + extended kind. Review: https://reviewboard.asterisk.org/r/3147/ + +2014-01-27 01:25 +0000 [r406595] Russell Bryant + + * main/file.c, include/asterisk/channel.h, main/channel.c, /: + Protect ast_filestream object when on a channel The + ast_filestream object gets tacked on to a channel via + chan->timingdata. It's a reference counted object, but the + reference count isn't used when putting it on a channel. It's + theoretically possible for another thread to interfere with the + channel while it's unlocked and cause the filestream to get + destroyed. Use the astobj2 reference count to make sure that as + long as this code path is holding on the ast_filestream and + passing it into the file.c playback code, that it knows it's + valid. Bug reported by Leif Madsen. Review: + https://reviewboard.asterisk.org/r/3135/ ........ Merged + revisions 406566 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406567 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406574 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-26 23:04 +0000 [r406517] Richard Mudgett + + * /, main/tcptls.c: tcptls.c: Add missing cleanup on off nominal + path. ........ Merged revisions 406514 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406515 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406516 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-26 14:19 +0000 [r406503] Tzafrir Cohen + + * contrib/scripts/live_ast: live_ast: run wrapped programs with + exec live_ast can be used as a wrapper script to run asterisk, + gdb or valgrind. In those cases it runs them and returns the + result. It is more useful to use 'exec' to avoid having another + odd process in the chain. Review: + https://reviewboard.asterisk.org/r/3110/ + +2014-01-26 02:11 +0000 [r406490] Joshua Colp + + * res/res_pjsip_session.c, /: res_pjsip_session: Be less strict + with core requested outgoing capabilities. The core may + (depending on circumstances) request a single codec on outgoing + calls. Many channel drivers ignore or treat this as a suggestion + while still including configured codecs. The res_pjsip_session + logic treated this as an explicit request, leaving out other + configured codecs. This change makes res_pjsip_session behave + like other channel driver and simply adds the requested codec to + the list. (closes issue ASTERISK-23082) Reported by: xrobau + Review: https://reviewboard.asterisk.org/r/3140/ ........ Merged + revisions 406489 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 23:33 +0000 [r406466] Richard Mudgett + + * /, main/cel.c: CEL: Protect data structures during reload and + shutdown. The CEL data structures need to be protected during a + configuration reload and shutdown. Asterisk crashed during a + shutdown because CEL events were still in flight and the CEL data + structures were already destroyed. * Protected the cel_backends, + cel_dialstatus_store, and cel_linkedids ao2 containers with a + global ao2 object wrapper. * Added NULL checks before use of the + cel_backends, cel_dialstatus_store, and cel_linkedids ao2 + containers in case the CEL module is already shutdown. * Fixed + overloading of the cel_linkedids held objects reference count. + During shutdown any held objects would be leaked. * Fixed memory + leak of cel_linkedids held objects if the LINKEDID_END is not + being tracked. The objects in the cel_linkedids container were + not removed if the LINKEDID_END event is not used. * Added access + protection to the cel_backends container during the CLI "cel show + status" command. * Made cel_backends, cel_dialstatus_store, and + cel_linkedids use the standard ao2 callback templates for the + hash and cmp functions. * Eliminated unnecessary uses of + RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated + resources on failure. (closes issue AST-1253) Reported by: + Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3128/ ........ Merged + revisions 406417 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406418 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406465 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 22:34 +0000 [r406416] Jonathan Rose + + * main/utils.c, CHANGES: Thread Debugging: Add LWP to core show + locks output This patch adds the LWP to core show locks output if + it is available. Review: https://reviewboard.asterisk.org/r/3142/ + +2014-01-24 22:18 +0000 [r406407] Richard Mudgett + + * main/manager.c, /: manager: Register atexit shutdown routine only + once. * Made register atexit shutdown routine only once in + __init_manager(). * Fixed some initial load failure conditions in + __init_manager(). * Made reset options to defaults on reload when + the reload will actually happen. * Removed unnecessary container + traversals of the white/black filters during manager_free_user(). + * ast_free() does not need a NULL check before calling. ........ + Merged revisions 406359 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406400 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406401 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 21:46 +0000 [r406399] Jonathan Rose + + * res/res_config_pgsql.c, /: res_config_pgsql: Fix a memory leak + and use RAII_VAR for cleanup when practical Review: + https://reviewboard.asterisk.org/r/3141/ ........ Merged + revisions 406360 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406361 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406389 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-24 18:13 +0000 [r406343] Richard Mudgett + + * main/manager.c, /: manager: Protect data structures during + shutdown. Occasionally, the manager module would get an + "INTERNAL_OBJ: bad magic number" error on a "core restart + gracefully" command if an AMI connection is established. * Added + ao2_global_obj protection to the sessions global container. * + Fixed the order of unreferencing a session object in + session_destroy(). * Removed unnecessary container traversals of + the white/black filters during session_destructor(). (closes + issue AST-1242) Reported by: Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3144/ ........ Merged + revisions 406341 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406342 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-23 23:43 +0000 [r406328] Mark Michelson + + * /: Today is not my day for writing code that compiles. ........ + Merged revisions 406327 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-23 22:56 +0000 [r406312] Michael L. Young + + * /, addons/res_config_mysql.c: res_config_mysql: Fix Setting The + Column Name Incorrectly When support for a realtime sorcery + module was added in revision 386731, the wrong property was + accidentally used for setting the column name to be updated in + the database table. This patch fixes the typo. (closes issue + ASTERISK-23177) Reported by: Denis Tested by: Denis Patches: + asterisk-23177-use-field-name.diff by Michael L. Young (license + 5026) ........ Merged revisions 406311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-23 21:18 +0000 [r406298] Mark Michelson + + * res/res_pjsip_pidf.c, /: Multiple revisions 406294-406295 + ........ r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu, + 23 Jan 2014) | 11 lines Fix presence body errors found during + testing: * PIDF bodies were reporting an "open" state in many + cases where it should have been reporting "closed" * XPIDF bodies + had XML nodes placed incorrectly within the hierarchy. * SIP URIs + in XPIDF bodies did not go through XML sanitization * XML + sanitization had some errors: * Right angle bracket was being + replaced with "&rt;" instead of ">" * Double quote, + apostrophe, and ampersand were not being escaped. ........ + r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan + 2014) | 11 lines Fix presence body errors found during testing: * + PIDF bodies were reporting an "open" state in many cases where it + should have been reporting "closed" * XPIDF bodies had XML nodes + placed incorrectly within the hierarchy. * SIP URIs in XPIDF + bodies did not go through XML sanitization * XML sanitization had + some errors: * Right angle bracket was being replaced with "&rt;" + instead of ">" * Double quote, apostrophe, and ampersand were + not being escaped. ........ Merged revisions 406294-406295 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-22 22:24 +0000 [r406269] Scott Griepentrog + + * main/pbx.c, /, utils/extconf.c: pbx.c: Pre-initialize timezone to + avoid crash on destroy In ast_build_timing, initialize the + timezone value to NULL in order to avoid deferencing an + uninitialized value later when calling ast_destroy_timing. The + timezone value could be uninitialized if ast_build_timing were to + fail due to a zero length time string. (closes issue + ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review: + https://reviewboard.asterisk.org/r/3134/ Patches: + ast_build_timing-initialize-timezone.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 406241 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406245 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406264 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-22 19:36 +0000 [r406153-406224] Kinsey Moore + + * /, apps/app_confbridge.c: ConfBridge: Fix channel parameter + documentation Confbridge AMI and CLI commands for mute, unmute, + and setting the single video source can accept channel prefixes + in lieu of a full channel name, but documentation states only + that it is required and is a channel name. This corrects the + documentation. (closes issue PQ-1397) Reported by: Steve Pitts + ........ Merged revisions 406217 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406223 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: chan_sip: Decline image streams on + unsupported transports This change allows chan_sip to decline + individual image streams over unsupported transports in the SDP + of the 200 response. Previously, an image stream offer with + RTP/AVP as the transport would cause chan_sip to respond with a + 488. (closes issue ASTERISK-22988) Reported by: adomjan Original + patch by: adomjan ........ Merged revisions 406170 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406171 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406172 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /: res_stasis_playback: Correct error + argument order Several of the playback error messages for invalid + media input in res_stasis_playback.c had the media name and + channel name reversed. They now correctly identify the channel + name and media name. Reported by: skrusty ........ Merged + revisions 406152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-21 21:48 +0000 [r406134] Rusty Newton + + * /, res/res_pjsip.c: res_pjsip: Documentation improvement for + Endpoint and AOR mailbox options. Making the help text for both + more explicit regarding the format of mailbox identifiers. i.e. + clarifying the format for app_voicemail mailboxes vs mailboxes + from external MWI sources through modules such as + res_external_mwi. ........ Merged revisions 406133 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-21 21:08 +0000 [r406082] Walter Doekes + + * main/manager.c, /, configs/manager.conf.sample: manager: Clarify + eventfilter documentation. Textual changes only. Review: + https://reviewboard.asterisk.org/r/3133/ ........ Merged + revisions 406079 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406080 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406081 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-21 20:28 +0000 [r406006-406078] Kinsey Moore + + * channels/chan_mgcp.c, /: chan_mgcp: Enforce locking for oseq This + restricts direct usage of global oseq so that all accesses are + locked and threads are not racing to get oseq values that they + did not claim. This also fixes a build error in res_pktccops + under dev mode. (closes issue ASTERISK-23100) Reported by: + adomjan Patch by: adomjan ........ Merged revisions 406037 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 406038 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 406049 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c, res/res_pjsip.c: PJSIP: + Handle headers in a list appropriately The PJSIP header parsing + function (pjsip_parse_hdr) can generate more than one header + instance from a single header field. These header instances exist + as a list attached to the returned header and must be handled + appropriately when they are added to a message or else only the + first header instance will be used. This changes the linked list + functions used in outbound proxy code to merge the lists + properly. ........ Merged revisions 406020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_sounds.h, res/ari/resource_bridges.h, + res/ari/resource_device_states.h, res/ari/resource_mailboxes.h, + res/ari/resource_asterisk.h, rest-api/api-docs/channels.json, + res/ari/resource_applications.h, res/ari/resource_channels.c, + res/res_ari_playbacks.c, res/res_ari_sounds.c, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, res/res_ari_bridges.c, /, + res/res_ari_device_states.c, + rest-api-templates/ari_resource.h.mustache, + res/res_ari_mailboxes.c, res/res_ari_asterisk.c, + res/res_ari_applications.c, + rest-api-templates/res_ari_resource.c.mustache, + rest-api-templates/body_parsing.mustache (added), + res/res_ari_channels.c, res/ari/resource_playbacks.h, + rest-api-templates/param_parsing.mustache: ARI: Support channel + variables in originate This adds back in support for specifying + channel variables during an originate without compromising the + ability to specify query parameters in the JSON body. This was + accomplished by generating the body-parsing code in a separate + function instead of being integrated with the URI query parameter + parsing code such that it could be called by paths with body + parameters. This is transparent to the user of the API and + prevents manual duplication of code or data structures. (closes + issue ASTERISK-23051) Review: + https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan + ........ Merged revisions 406003 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 23:25 +0000 [r405985] Damien Wedhorn + + * /, channels/chan_skinny.c: Skinny: fix up handling of fragmented + packets. Bad offset in reading second or more fragment of skinny + packets. Fixed to offset by char (single byte) rather than size + of req. ........ Merged revisions 405982 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 22:23 +0000 [r405947] Richard Mudgett + + * channels/sig_pri.c, /: chan_dahdi/PRI: Suppress CONNECTED_LINE + updates when nothing in the udpate is valid. * Also simplified + some subddress handling code. (closes issue ASTERISK-23008) + Reported by: Michael Cargile ........ Merged revisions 405926 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 405927 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405928 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 21:56 +0000 [r405925] Damien Wedhorn + + * /, channels/chan_skinny.c: Skinny: fix up session logging. + Logging from the skinny session loop was providing some incorrect + reasons for exiting the loop. Cleaned up messages and handling so + correct reason displayed. ........ Merged revisions 405924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-20 18:18 +0000 [r405910] Jonathan Rose + + * channels/chan_pjsip.c, /: chan_pjsip: Provide a means for + tracking device state when holding/unholding Previously PJSIP did + not track hold/unhold and it would always simply be 'inuse'. This + patch fixes that. review: + https://reviewboard.asterisk.org/r/3129/ ........ Merged + revisions 405908 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-19 00:01 +0000 [r405894] Damien Wedhorn + + * /, channels/chan_skinny.c: Skinny: fix reversed device reset from + CLI. Existing code would do a full device restart when "skinny + reset device" was entered at the CLI and do a reset when "skinny + reset device restart" entered. ........ Merged revisions 405893 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 22:09 +0000 [r405878] Sean Bright + + * /, channels/chan_sip.c: Make sure the maxptime attribute is added + to the correct offers. ........ Merged revisions 405877 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 21:33 +0000 [r405862-405876] Scott Griepentrog + + * main/format_pref.c, main/sorcery.c, main/frame.c, /, + include/asterisk/format_pref.h, res/res_pjsip_sdp_rtp.c: pjsip: + fix support for allow=all This change adds improvements to + support for allow=all in pjsip.conf so that it functions as + intended. Previously, the allow/disallow socery configuration + would set & clear codecs from the media.codecs and media.prefs + list, but if all was specified the prefs list was not updated. + Then a call would fail when create_outgoing_sdp_stream() created + an SDP with no audio codecs. A new function + ast_codec_pref_append_all() is provided to add all codecs to the + prefs list - only those not already on the list. This enables the + configuration to specify a codec preference, but still add all + codecs, and even then remove some codecs, as shown in this + example: allow = ulaw, alaw, all, !g729, !g723 Also, the display + order of allow in cli output is updated to match the + configuration by using prefs instead of caps when generating a + human readable string. Finally, a change to + create_outgoing_sdp_stream() skips a codec when it does not have + a payload code instead of the call failing. (closes issue + ASTERISK-23018) Reported by: xrobau Review: + https://reviewboard.asterisk.org/r/3131/ ........ Merged + revisions 405875 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/http.c: http: supported chunked Transfer-Encoding This + change implements support for HTTP Transfer-Encoding chunked in + both JSON and Form (post vars) body content. A new function + ast_http_get_contents() handles both regular and chunked mode + body, returning after the entire body is received. (closes issue + ASTERISK-23068) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3125/ ........ Merged + revisions 405861 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 18:55 +0000 [r405778-405844] Rusty Newton + + * res/res_pjsip.c, /: Fixing some XML syntax issues with my + previous commit at r405777 for ASTERISK-23071 ........ Merged + revisions 405843 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c, doc/asterisk.8, main/features.c, + configs/sip.conf.sample, apps/app_queue.c, apps/app_transfer.c, + channels/chan_iax2.c: Documentation: doc fixes across various + parts of the code for ASTERISK issues 23061,23028,23046,23027 + Fixes typos of "transfered" instead of "transferred" in various + code. Fixes incorrect gosub param help text for app_queue. Fixes + Asterisk man pages containing unquoted minus signs. Adds note + about the "textsupport" option in sip.conf.sample. (issue + ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) + (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes + issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue + ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis + Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine + (license 6561) hyphen.patch uploaded by Jeremy Laine (license + 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) + ........ Merged revisions 405791 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405792 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405829 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip.c, /: res_pjsip: enhance documentation for + mailboxes options, for both endpoints and aors Made documentation + more explicit as to the use of the both options. (issue + ASTERISK-23071) (closes issue ASTERISK-23071) Reported by: Matt + Jordan ........ Merged revisions 405777 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-17 14:17 +0000 [r405766] Walter Doekes + + * res/res_musiconhold.c, CHANGES: Enable wide band audio in + musiconhold streams. Review: + https://reviewboard.asterisk.org/r/3112/ + +2014-01-16 20:06 +0000 [r405747-405749] Kevin Harwell + + * res/res_pjsip/pjsip_options.c, /: res_pjsip: AOR option + qualify_frequency not respected on startup If an endpoint had + previously dynamically registered a contact and the contact + information was successfully stored in astdb then upon restart + the qualify notifications would not be sent out if the + qualify_frequency was set. This was due to the fact that only + permanent contacts were being checked and scheduled for qualifies + on startup. Modified the code to check and schedule all + registered contacts at startup. (closes issue ASTERISK-23062) + Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3124/ ........ Merged + revisions 405748 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Originate doesn't abort on failed + format_cap allocation action_originate responds to the remote + system with an error when cap==NULL, but doesn't return (abort + the originate). Patched to return. (closes issue ASTERISK-23034) + Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded + by coreyfarrell (license 5909) ........ Merged revisions 405745 + from http://svn.asterisk.org/svn/asterisk/branches/11 ........ + Merged revisions 405746 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-16 19:33 +0000 [r405744] Kinsey Moore + + * /, res/res_pjsip.c: PJSIP: Fix outbound OPTIONS support When path + support was added and contacts were made available during request + creation and transmission, the code path used by outbound qualify + support was not modified correctly and was causing request + creation to fail. This ensures that outbound request creation + with only a contact and no dialog, endpoint, or uri can succeed + which restores qualify support. Reported by: gtjoseph Reported + by: kharwell ........ Merged revisions 405743 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-16 19:13 +0000 [r405644-405695] Kevin Harwell + + * /, res/res_fax.c, configs/res_fax.conf.sample: res_fax: + check_modem_rate() returned incorrect rate for V.27 According to + the new standard for V.27 and V.32 they are able to transmit at a + bit rate of 4,800 or 9,600. The check_mode_rate function needed + to be updated to reflect this. Also, because of this change the + default 'minrate' value was updated to be 4800. (closes issue + ASTERISK-22790) Reported by: Paolo Compagnini Patches: + res_fax.txt uploaded by looserouting (license 6548) ........ + Merged revisions 405656 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405693 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405694 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_pjsip.c: chan_pjsip: initial device state on + endpoints is INVALID When endpoints get loaded their device state + gets set to 'INVALID' because the channel driver has not been + loaded yet. Fixed by updating the device state for every endpoint + upon load of the channel driver. (closes issue ASTERISK-23065) + Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/3123/ ........ Merged + revisions 405643 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-15 16:51 +0000 [r405586-405589] Jonathan Rose + + * CHANGES: Make 12 - 12.1 CHANGES log the same as in 12 + + * CHANGES, /: Include CHANGES info for r405553 ........ Merged + revisions 405585 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-15 16:36 +0000 [r405584] Joshua Colp + + * /, cel/cel_manager.c: cel_manager: Don't crash if configuration + file is invalid. The cel_manager module did not properly handle + the case where the configuration file was invalid. The module + will now output a warning message and disable itself if this + occurs. Reported by: Bryan Walters ........ Merged revisions + 405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 405582 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405583 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-15 13:16 +0000 [r405566] Kinsey Moore + + * res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c, + res/res_pjsip_path.c (added), res/res_pjsip_mwi.c, + res/res_pjsip/pjsip_distributor.c, res/res_pjsip_diversion.c, + channels/chan_pjsip.c, res/res_pjsip_registrar.c, + res/res_pjsip_refer.c, include/asterisk/res_pjsip.h, + include/asterisk/res_pjsip_session.h, res/res_pjsip_notify.c, /, + res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c, + res/res_pjsip_t38.c, res/res_pjsip.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip_nat.c, + res/res_pjsip_session.c, + contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py + (added), res/res_pjsip_header_funcs.c: PJSIP: Add Path header + support This adds Path support to chan_pjsip in res_pjsip_path.c + with minimal additions in res_pjsip_registrar.c to store the path + and additions in res_pjsip_outbound_registration.c to enable + advertisement of path support to registrars and intervening + proxies. Path information is stored on contacts and is enabled + via Address of Record (AoRs) and Registration configuration + sections. While adding path support, it became necessary to be + able to add SIP supplements that handled messages outside of + sessions, so a framework for handling these types of hooks was + added in parallel to the already-existing session supplements and + several senders of out-of-dialog requests were refactored as a + result. (closes issue ASTERISK-21084) Review: + https://reviewboard.asterisk.org/r/3050/ ........ Merged + revisions 405565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 23:44 +0000 [r405554] Jonathan Rose + + * res/res_stasis_mailbox.exports.in (added), + res/ari/ari_model_validators.h, rest-api/api-docs/mailboxes.json + (added), include/asterisk/stasis_app_mailbox.h (added), + res/ari/resource_mailboxes.c (added), /, res/ari.make, + res/res_ari_mailboxes.c (added), res/ari/resource_mailboxes.h + (added), res/res_stasis_mailbox.c (added), + rest-api/resources.json, res/ari/ari_model_validators.c: ARI: Add + mailboxes resource for controlling and polling external MWI Adds + the following AMI commands: PUT mailboxes/mailboxName modifies + mailbox state and implicitly creates new mailboxes GET + mailboxes/mailboxName retrieves a JSON representation of a single + mailbox if it exists GET mailboxes retrieves a JSON array of all + mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that + res_mwi_external must be loaded for these functions to actually + do anything. Review: https://reviewboard.asterisk.org/r/3117/ + ........ Merged revisions 405553 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 21:46 +0000 [r405542] Richard Mudgett + + * main/strings.c, /: string container: Remove unnecessary RAII_VAR + usage and string object lock. ........ Merged revisions 405541 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 18:15 +0000 [r405437] Scott Griepentrog + + * /, channels/chan_sip.c: chan_sip: fix Local From tag on outbound + register regression In ASTERISK-12117, an improvement to insure + consistant local from tags on outbound registrations resulted in + an undesirable behavior - caused by leftover unexpired sip_pvt + dialogs (with the previous cseq number), resulting in many + uncessary REGISTER requests. Instead of significant rework of + transmit_register(), this change deletes the dialogs after a 200 + OK response indiciating a successful registration, keeping the + old dialogs from interfering with normal operation. (closes issue + ASTERISK-22946) Reported by: Stephan Eisvogel Review: + https://reviewboard.asterisk.org/r/3109/ ........ Merged + revisions 405433 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405434 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405435 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 18:14 +0000 [r405436] Richard Mudgett + + * apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample, + main/cli.c, include/asterisk/logger.h, main/pbx.c, + main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c, + main/logger.c, UPGRADE.txt: verbosity: Fix performance of console + verbose messages. The per console verbose level feature as + previously implemented caused a large performance penalty. The + fix required some minor incompatibilities if the new rasterisk is + used to connect to an earlier version. If the new rasterisk + connects to an older Asterisk version then the root console + verbose level is always affected by the "core set verbose" + command of the remote console even though it may appear to only + affect the current console. If an older version of rasterisk + connects to the new version then the "core set verbose" command + will have no effect. * Fixed the verbose performance by not + generating a verbose message if nothing is going to use it and + then filtered any generated verbose messages before actually + sending them to the remote consoles. * Split the "core set debug" + and "core set verbose" CLI commands to remove the per module + verbose support that cannot work with the per console verbose + level. * Added a silent option to the "core set verbose" command. + * Fixed "core set debug off" tab completion. * Made "core show + settings" list the current console verbosity in addition to the + root console verbosity. * Changed the default verbose level of + the 'verbose' setting in the logger.conf [logfiles] section. The + default is now to once again follow the current root console + level. As a result, using the AMI Command action with "core set + verbose" could again set the root console verbose level and + affect the verbose level logged. (closes issue AST-1252) Reported + by: Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/3114/ ........ Merged + revisions 405431 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405432 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-14 16:43 +0000 [r405420] Mark Michelson + + * res/res_pjsip/pjsip_distributor.c: Fix erroneous behavior when + sending auth rejection to artificial endpoint. We were not + including an authentication challenge when sending a 401 response + to unmatched endpoints. This was due to the conversion to use a + vector for authentication section names on an endpoint. The + vector for artificial endpoints was empty, resulting in the + challenge being sent back containing no challenges. This is + worked around by placing a bogus value in the artificial + endpoint's auth vector. This value is never looked up by + anything, since they instead will directly call + ast_sip_get_artificial_auth(). + +2014-01-14 03:27 +0000 [r405369] Damien Wedhorn + + * /, channels/chan_skinny.c: Skinny: do not add call to missed + calls list if answered elsewhere. Patch updates skinny devices + with a SKINNY_CONNECTED callstate if an inbound ringing or + callwaiting call is answered elsewhere. ........ Merged revisions + 405367 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-13 13:34 +0000 [r405339] Kinsey Moore + + * /, res/res_pjsip/pjsip_cli.c: res_pjsip: Fix CLI tab completion + issues This fixes several issues with the new res_pjsip CLI tab + completion such as output of headers during tab completion and + being able to tab-complete more items than the code actually + handled (further items would simply be ignored). (closes issue + ASTERISK-23081) Review: https://reviewboard.asterisk.org/r/3115/ + Reported by: xrobau ........ Merged revisions 405338 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-12 22:24 +0000 [r405326] Joshua Colp + + * res/ari/resource_playbacks.c, res/ari/resource_channels.c, + include/asterisk/ari.h, res/ari/resource_bridges.c, + res/ari/resource_recordings.c, res/ari/resource_device_states.c, + res/res_ari.c, res/ari/resource_endpoints.c, /, + res/ari/resource_applications.c: res_ari: Fix various memory + leaks. This change fixes a few memory leaks that were found based + on a mailing list post. 1. Some JSON response messages were never + freed. This was caused by the documentation stating that message + references were stolen when in reality they were not. The code + now follows the documentation and usage has been updated. 2. HTTP + response headers were never freed. 3. The variable list for + wildcards paths was never freed. (closes issue ASTERISK-23128) + Reported by: Kenneth Watson (on list) Review: + https://reviewboard.asterisk.org/r/3119/ ........ Merged + revisions 405325 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-12 22:13 +0000 [r405313-405314] Matthew Jordan + + * apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h, + apps/app_cdr.c, main/cdr.c: CDRs: Synchronize dialplan + applications that manipulate CDRs with the engine In + https://reviewboard.asterisk.org/r/3057/, applications and + functions that manipulate CDRs were made to interact over Stasis. + This was done to synchronize manipulations of CDRs from the + dialplan with the updates the engine itself receives over the + message bus. This change rested on a faulty premise: that + messages published to the CDR topic or to a topic that forwards + to the CDR topic are synchronized with the messages handled by + the CDR topic subscription in the CDR engine. This is not the + case. There is no ordering guaranteed for two messages published + to the same topic; ordering is only guaranteed if a message is + published to the same subscriber. Stasis was modified in r405311 + to allow a publisher to synchronize on the subscriber. This patch + uses that API to synchronize the CDR publishers with the CDR + engine message router, which maintains the overall topic + subscription. (closes issue ASTERISK-22884) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ + Merged revisions 405312 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis.c, main/stasis_message_router.c, /, + include/asterisk/stasis.h, + include/asterisk/stasis_message_router.h, tests/test_stasis.c: + stasis: Add methods to allow for synchronous publishing to + subscriber This patch adds an API call to Stasis that allows a + publisher to publish a stasis message that will not return until + a specific subscriber handles the message. Since a subscriber can + have their own forwarding topic which orders messages from many + topics, this allows a publisher who knows of that subscriber to + synchronize to that subscriber regardless of the forwarding + relationships between topics. This is of particular use for + dialplan applications that need to synchronize on a particular + subscriber's handling of a message. (issue ASTERISK-22884) + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3099/ ........ Merged + revisions 405311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-10 20:00 +0000 [r405299] Mark Michelson + + * /, res/res_pjsip/security_events.c: Print "" for + artificial endpoint in PJSIP security events. Previously, this + printed a UUID, which was not very clear when dealing with an + artificial endpoint. Review: + https://reviewboard.asterisk.org/r/3113 ........ Merged revisions + 405298 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-10 18:17 +0000 [r405284] Richard Mudgett + + * /, main/logger.c: Logging callid: Fix some sizeof() references + per coding guidelines. ........ Merged revisions 405281 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405282 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 23:52 +0000 [r405270] Jonathan Rose + + * res/res_pjsip_session.c: PJSIP: Add unhold on reinvite without + SDP behavior Review: https://reviewboard.asterisk.org/r/3106/ + +2014-01-09 23:50 +0000 [r405269] Damien Wedhorn + + * channels/chan_dahdi.c, /: Fix chan_dahdi copile issue in + dev-mode. Error "unused variable i in dahdi_create_channel_range" + when compiling in dev-mode. Small restructure to + dahdi_create_channel_range to move the for(x) loop and int i,x to + a block within the IFDEF. ........ Merged revisions 405268 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 23:39 +0000 [r405267] Kevin Harwell + + * res/res_pjsip.c, /, res/res_pjsip_messaging.c: + res_pjsip_messaging: potential for field values in from/to + headers to be missing Added in ability to specify display name + format ("name" ) for a given URI and made + sure it was fully propagated to the outgoing message. Also made + it so outoing messages in res_pjsip always send as "sip:". + (closes issue ASTERISK-22924) Reported by: Anthony Messina + Review: https://reviewboard.asterisk.org/r/3094/ ........ Merged + revisions 405266 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 20:34 +0000 [r405254] Kinsey Moore + + * main/astobj2.c, res/res_pjsip_session.c, /, + include/asterisk/astobj2.h: astobj2: Correct ao2_iterator opacity + violations This corrects the ao2_iterator opacity violations in + res_pjsip_session.c by adding a global function to get the number + of elements inside the container hidden behind the iterator. + (closes issue ASTERISK-23053) Review: + https://reviewboard.asterisk.org/r/3111/ Reported by: Richard + Mudgett ........ Merged revisions 405253 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 16:52 +0000 [r405236] Kevin Harwell + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fails to resume + WebRTC call from hold In ast_rtp_ice_start if the ice session + create check list failed, start check was never initiated and + ice_started was never set to true. Upon re-entering the function + (for instance, [un]hold) it would try to create the check list + again with duplicate remote candidates. Fixed so that if the + create check list fails the necessary data structures are + properly re-initialized for any subsequent retries. Note, it was + decided to not stop ice support (by calling ast_rtp_ice_stop) on + a check list failure because it possible things might still work. + However, a debug message was added to help with any future + troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis + Valentinavičius Patches: works_on_my_machine.patch uploaded by + xytis (license 6558) ........ Merged revisions 405234 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405235 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 15:50 +0000 [r405217] Matthew Jordan + + * /, apps/app_confbridge.c, + apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix + crash caused when waitmarked/marked users leave together When + waitmarked users join a ConfBridge, the conference state is + transitioned from EMPTY -> INACTIVE. In this state, the users are + maintined in a waiting users list. When a marked user joins, the + ConfBridge conference transitions from INACTIVE -> MULTI_MARKED, + and all users are put onto the active list of users. This process + works correctly. When the marked user leaves, if they are the + last marked user, the MULTI_MARKED state does the following: (1) + It plays back a message to the bridge stating that the leader has + left the conference. This requires an unlocking of the bridge. + (2) It moves waitmarked users back to the waiting list (3) It + transitions to the appropriate state: in this case, INACTIVE + However, because it plays the prompt back to the bridge before + moving the users and before finishing the state transition, this + creates a race condition: with the bridge unlocked, waitmarked + users who leave the conference (or are kicked from it) can cause + a state transition of the bridge to another state before the + conference is transitioned to the INACTIVE state. This causes the + state machine to get a bit wonky, often leading to a crash when + the MULTI_MARKED state attempts to conclude its processing. This + patch fixes this problem: (1) It prevents kicked users from being + kicked again. That's just a nicety. (2) More importantly, it + fixes the race condition by only playing the prompt once the + state has transitioned correctly to INACTIVE. If waitmarked users + sneak out during the prompt being played, no harm no foul. + Review: https://reviewboard.asterisk.org/r/3108/ Note that the + patch committed here is essentially the same as uploaded by Simon + Moxon on ASTERISK-22740, with the addition of the double kick + prevention. (closes issue AST-1258) Reported by: Steve Pitts + (closes issue ASTERISK-22740) Reported by: Simon Moxon patches: + ASTERISK-22740.diff uploaded by Simon Moxon (license 6546) + ........ Merged revisions 405215 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405216 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-09 14:15 +0000 [r405163] Walter Doekes + + * /, apps/app_dumpchan.c: "Minimun" typo. ........ Merged revisions + 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 405161 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405162 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-08 17:23 +0000 [r405144] Mark Michelson + + * /, res/res_pjsip/security_events.c: Use proper case for checking + if digest authentication is used. ........ Merged revisions + 405131 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-08 16:34 +0000 [r405129-405130] Kinsey Moore + + * /, configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support + for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is + available on newer operating systems. (closes issue + ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/ + Reported by: George Joseph Patch by: George Joseph ........ + Merged revisions 405090 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 405091 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405124 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: Add the missing part of r400140 When the + patch to add retry-on-forbidden-response was committed, part of + the patch for chan_sip was not committed which caused the feature + to be entirely nonfunctional. This corrects the code in question. + (closes issue ASTERISK-17138) Review: + https://reviewboard.asterisk.org/r/2874 ........ Merged revisions + 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 405081 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 405083 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-07 19:56 +0000 [r405020-405035] Joshua Colp + + * /, res/res_pjsip_acl.c: res_pjsip_acl: Fix another case of + assuming a contact will always contain a URI. ........ Merged + revisions 405034 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_nat.c: res_pjsip_nat: Don't assume a Contact + header will always contain a URI. If the 'rewrite_contact' option + was enabled and a Contact header was received which contained a + '*' a crash would occur. This change makes the res_pjsip_nat + module ignore the Contact header if it contains only a '*'. + (closes issue ASTERISK-23101) Reported by: Matt Jordan ........ + Merged revisions 405019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-06 21:55 +0000 [r404953-405007] Richard Mudgett + + * apps/app_voicemail.c, /: app_voicemail: Explicitly set + defaultenabled=yes ........ Merged revisions 405006 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_mwi_external_ami.c (added): External MWI AMI support. + The external MWI AMI interface provides a thin wrapper around the + core external MWI resource. The resource adds the following AMI + actions: MWIGet, MWIDelete, and MWIUpdate. (closes issue AFS-46) + Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged + revisions 404954 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_mwi_external.c (added), configs/sorcery.conf.sample, + include/asterisk/res_mwi_external.h (added), + res/res_mwi_external.exports.in (added), apps/app_voicemail.c: + External MWI core support. * The core external MWI resource + provides for MWI message counts persistence using sorcery. With + sorcery, the user is able to configure which sorcery wizzard + backend to use if the default astdb is not desired. * The core + external MWI resoruce provides some debugging CLI commands + enabled by defining MWI_DEBUG_CLI. The debugging CLI commands + are: "mwi delete all", "mwi delete like ", "mwi delete + mailbox ", "mwi list all", "mwi list like ", "mwi + show mailbox ", and "mwi update mailbox [ + []]". (closes issue AFS-43) Review: + https://reviewboard.asterisk.org/r/3061/ ........ Merged + revisions 404952 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-05 16:01 +0000 [r404924-404936] Joshua Colp + + * /, res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Don't assume that a registration + client will always exist. ........ Merged revisions 404935 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Create registration client in pj + thread. Depending on which threading was loading the outbound + registration it was possible for the registration client to be + allocated outside of a pj thread. This change moves the creation + inside the synchronous task where it is guaranteed it will occur + in a pj thread. Reported by: Rob Thomas ........ Merged revisions + 404923 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-04 10:52 +0000 [r404912] Tzafrir Cohen + + * main/asterisk.c, /: asterisk.c: suppress live_dangerously warning + on rasterisk Even since the fixes of AST-2013-007, Asterisk + prints the following warning on startup if the user decided to + live dangerously: Privilege escalation protection disabled! See + https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This + message is intended for the logs and interactive startup. No need + for it to appear on a remote console. This commit removes it from + there. (closes issue ASTERISK-23084) Review: + https://reviewboard.asterisk.org/r/3101/ ........ Merged + revisions 404861 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404888 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404911 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 22:00 +0000 [r404860] Kevin Harwell + + * cel/cel_pgsql.c, /: cel_pgsql: module not correctly reloading + Upon reload the module unconditionally "unloaded" the module + (freeing memory and setting pointers to NULL) and then when + attempting a "load" if the config file had not changed then + nothing would be reinitialized. By moving the "unload" to occur + conditionally (reload only) after an attempted configuration + load, but before module "loading" alleviates the issue. The + module now loads/unloads/reloads correctly. (closes issue + ASTERISK-22871) Reported by: Matteo ........ Merged revisions + 404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 404858 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404859 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 21:45 +0000 [r404844-404856] Matthew Jordan + + * /, res/res_pjsip_logger.c: res_pjsip_logger: Add the + ASTERISK_FILE_VERSION macro Registering yourself with the + Asterisk core is the nice thing to do, even when you're a logging + module. ........ Merged revisions 404855 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_authenticator_digest.c, tests/test_utils.c: + res_pjsip_authenticator_digest: Fix md5 hash buffer An md5 hash + is 32 bytes long. The char buffer must be at least 33 bytes to + avoid clobbering of the stack. This patch also fixes a potential + clobbering in test_utils.c. Thanks to Andrew Nagy for reporting + and testing this out in #asterisk-dev Reported by: Andrew Nagy + Tested by: Andrew Nagy ........ Merged revisions 404843 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 20:02 +0000 [r404787-404832] Kevin Harwell + + * main/manager.c: manager: UserEvent including action on output AMI + action UserEvent event response would include the action header + in its keyvalue pairs list. Adjusted the start of the header loop + to skip over the action part. (closes issue ASTERISK-22899) + Reported by: outtolunc Patches: + svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license + 5198) + + * channels/chan_dahdi.c, /: chan_dahdi: dahdi show channels slices + PRI channel dnid on output dahdi show channels output slices the + callerid (which is dnid copied over on PRI channels). If the + channel naming structures look like: 'DAHDI/i1/1408409XXXX-6' + then the output slices 1408409XXXX down to 1408409XXX. This patch + just opens it up to 15 chars so you can see the whole thing. + (closes issue ASTERISK-22918) Reported by: outtolunc Patches: + svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc + (license 5198) ........ Merged revisions 404784 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404785 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404786 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 18:33 +0000 [r404783] Richard Mudgett + + * tests/test_stasis.c, /: test_stasis.c: Fix ref leak in normal + execution path. ........ Merged revisions 404764 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 18:31 +0000 [r404782] Kevin Harwell + + * /, apps/app_meetme.c: app_meetme: compiler warning Fixed a + compiler warning (errors in 'dev-mode') given by gcc version + 4.8.1. The one in app_meetme involved the + 'sizeof-pointer-memaccess' (see: + http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it + would no longer issue a warning and can compile again in + 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/ + ........ Merged revisions 404742 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404773 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404781 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-03 17:27 +0000 [r404726-404738] Joshua Colp + + * res/res_pjsip/pjsip_configuration.c, /, res/res_pjsip/location.c: + res_pjsip: Ensure more URI validation happens in pj threads. + ........ Merged revisions 404737 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_outbound_registration.c: + res_pjsip_outbound_registration: Ensure URI validation happens in + a pjlib thread. This change moves outbound registration URI + validation into the task executed within a pjlib thread. Reported + by: Andrew Nagy ........ Merged revisions 404725 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-02 19:38 +0000 [r404677] Scott Griepentrog + + * /, funcs/func_strings.c: func_strings: use memmove to prevent + overlapping memory on strcpy When calling REPLACE() with an empty + replace-char argument, strcpy is used to overwrite the the + matching . However as the src and dest arguments to + strcpy must not overlap, it causes other parts of the string to + be overwritten with adjacent characters and the result is + mangled. Patch replaces call to strcpy with memmove and adds a + test suite case for REPLACE. (closes issue ASTERISK-22910) + Reported by: Gareth Palmer Review: + https://reviewboard.asterisk.org/r/3083/ Patches: + func_strings.patch uploaded by Gareth Palmer (license 5169) + ........ Merged revisions 404674 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404675 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404676 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2014-01-02 19:08 +0000 [r404664] Kevin Harwell + + * channels/chan_pjsip.c, include/asterisk/res_pjsip.h, /, + configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c, + CHANGES, res/res_pjsip.c: res_pjsip: add 'set_var' support on + endpoints Added a new 'set_var' option for ast_sip_endpoint(s). + For each variable specified that variable gets set upon creation + of a pjsip channel involving the endpoint. (closes issue + ASTERISK-22868) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/3095/ ........ Merged + revisions 404663 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-31 22:51 +0000 [r404620-404653] Joshua Colp + + * channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip: + Handle hanging up before calling. Channel creation in Asterisk is + broken up into two steps: requesting and calling. In some cases a + channel may be requested but never called. This happens in the + ChanIsAvail dialplan application for determining if something is + reachable or not. The PJSIP channel driver did not take this + situation into account and attempted to end a session that was + never called out on. The code now checks the session state to + determine if the session has been called out on and if not + terminates it instead of ending it. (closes issue ASTERISK-23074) + Reported by: Kilburn ........ Merged revisions 404652 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_endpoint_identifier_ip.c: + res_pjsip_endpoint_identifier_ip: Accept hostnames in the 'match' + field. Hostnames specified in the 'match' field will be resolved + and all addresses returned. Each address will be added to the + endpoint identifier for the matching process. Reported by: Rob + Thomas ........ Merged revisions 404613 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-31 21:39 +0000 [r404606] Kevin Harwell + + * cel/cel_pgsql.c, /: cel_pgsql: deadlock on unload and + core_event_dispatcher A deadlock can happen between a thread + unloading or reloading the cel_pgsql module and the + core_event_dispatcher taskprocessor thread. Description of what + is happening: Thread 1 (for example, a netconsole thread): a + "module reload cel_pgsql" is launched the thread enter the + "my_unload_module" function (cel_pgsql.c) the thread acquire the + write lock on psql_columns the thread enter the + "ast_event_unsubscribe" function (event.c) the thread try to + acquire the write lock on ast_event_subs[sub->type] Thread 2 + (core_event_dispatcher taskprocessor thread): the taskprocessor + pop a CEL event the thread enter the "handle_event" function + (event.c) the thread acquire the read lock on + ast_event_subs[sub->type] the thread callback the "pgsql_log" + function (cel_pgsql.c), since it's a subscriber of CEL events the + thread try to acquire a read lock on psql_columns (closes issue + ASTERISK-22854) Reported by: Etienne Lessard Patches: + cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license + 6394) ........ Merged revisions 404603 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404604 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404605 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-31 20:27 +0000 [r404593] Joshua Colp + + * res/res_pjsip_outbound_registration.c, /: + res_pjsip_outbound_registration: Add validation for 'server_uri' + and 'client_uri'. When applying configuration for outbound + registrations the 'server_uri' and 'client_uri' fields were not + validated. The code will now confirm that they exist and that + they contain parseable SIP URIs. Reported by: Andrew Nagy + ........ Merged revisions 404592 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-30 23:25 +0000 [r404582] Kevin Harwell + + * main/channel.c, /: channels.c: core show channeltypes slicing + 'core show channeltypes' type column is being sliced, resulting + in incomplete type names. (closes issue ASTERISK-22919) Reported + by: outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded + by outtolunc (license 5198) ........ Merged revisions 404579 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404581 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-24 17:12 +0000 [r404567-404569] David M. Lee + + * UPGRADE-12.txt, /: Added note to UPGRADE.txt about the default + value of live_dangerously changing ........ Merged revisions + 404568 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/http.c: http: Properly reject requests with + Transfer-Encoding set Asterisk does not support any of the + transfer encodings specified in HTTP/1.1, other than the default + "identity" encoding. According to RFC 2616: A server which + receives an entity-body with a transfer-coding it does not + understand SHOULD return 501 (Unimplemented), and close the + connection. A server MUST NOT send transfer-codings to an + HTTP/1.0 client. This patch adds the 501 Unimplemented response, + instead of the hard work of actually implementing other + recordings. This behavior is especially problematic for Node.js + clients, which use chunked encoding by default. (closes issue + ASTERISK-22486) Review: https://reviewboard.asterisk.org/r/3092/ + ........ Merged revisions 404565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-24 02:20 +0000 [r404554] Joshua Colp + + * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Ensure dialog + manipulation happens on proper thread. When destroying a + subscription we remove the serializer from its dialog and + decrease its reference count. Depending on which thread dropped + the subscription reference count to 0 it was possible for this to + occur in a thread where it is not possible. (closes issue + ASTERISK-22952) Reported by: Matt Jordan ........ Merged + revisions 404553 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-23 16:38 +0000 [r404542] Tzafrir Cohen + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + UPGRADE-12.txt: chan_dahdi: enable ignore_failed_channels by + default If ignore_failed_channels is set to "true" for a channel, + the channel will continue to be configured even if configuring it + has failed. This allows Asterisk to start before all the DAHDI + initialization is done and thus not force the starting order + dahdi -> asterisk. Review: + https://reviewboard.asterisk.org/r/3063/ + +2013-12-21 03:35 +0000 [r404532] Matthew Jordan + + * /, res/res_pjsip/pjsip_cli.c: res_pjsip/pjsip_cli: fix + compilation error caused by passing ast_free When wanting to pass + *free as a function pointer, ast_free_ptr has to be used instead + of ast_free. This allows it to be compiled with MALLOC_DEBUG + enabled. ........ Merged revisions 404531 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 22:04 +0000 [r404511-404512] David M. Lee + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /, + rest-api/api-docs/applications.json: ari: Remove support for + specifying channel vars during origination. When we added support + for specifying channel variables for an origination, we didn't + consider how that would interact with another feature, namely + specifying request parameters in a JSON request body. The method + of specifying channel variables (as a flat JSON object passed in + the JSON body) interferes with parsing parameters out of the + request body. Unfortunately, fixing this would be a backward + incompatible change. In the interest of keeping the API sane and + keeping our release schedule, we're dropping the feature for + specifying channel variables in the origination request. We will + bring the feature back soon, as a backward compatible addition to + the API. (closes issue ASTERISK-23051) Review: + https://reviewboard.asterisk.org/r/3088 ........ Merged revisions + 404509 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Remove automerge properties ........ Merged revisions 404488 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 21:32 +0000 [r404507] Matthew Jordan + + * include/asterisk/config.h, main/config.c, main/channel.c, + res/res_pjsip/location.c, include/asterisk/res_pjsip_cli.h + (added), res/res_pjsip/pjsip_cli.c (added), + include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip_registrar.c, main/sorcery.c, + include/asterisk/res_pjsip.h, CREDITS, + res/res_pjsip/config_auth.c, /, + res/res_pjsip_endpoint_identifier_ip.c: res_pjsip: Add PJSIP CLI + commands Implements the following cli commands: pjsip list aors + pjsip list auths pjsip list channels pjsip list contacts pjsip + list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show + channels pjsip show endpoint(s) Also... Minor modifications made + to the AMI command implementations to facilitate reuse. New + function ast_variable_list_sort added to config.c and config.h to + implement variable list sorting. (issue ASTERISK-22610) patches: + pjsip_cli_v2.patch uploaded by george.joseph (License 6322) + ........ Merged revisions 404480 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 21:18 +0000 [r404461] Scott Griepentrog + + * /, main/say.c: say.c: correct time for polish In + ast_say_date_with_format_pl(), change ast_say_number() to use + tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported + by: Robert Mordec Review: + https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch + uploaded by veilen (license 6555) ........ Merged revisions + 404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 404457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 20:28 +0000 [r404452] Mark Michelson + + * /, res/res_pjsip_refer.c: Fix issue where PJSIP blind transferer + dialog may not complete as planned. When transferring to a + dialplan extension that will not place any outbound calls, the + only control frames that the PJSIP REFER framehook will receive + are inconsequential (such as unhold or srcchange). As such, we + shouldn't allow for the reception of those types of frames + prevent us from signaling to the transferring party that the + transfer has completed successfully once voice frames are read. + Thanks to Jonathan Rose for pointing this out. ........ Merged + revisions 404439 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 20:05 +0000 [r404438] Matthew Jordan + + * /, res/ari/resource_applications.h, + res/res_stasis_device_state.c: res_stasis_device_state: Set + resource type for subscriptions to deviceState The documentation + for ARI already specifies that the device state resource when + used for subscribing for events is "deviceState", not + "device_state". The code, however, used "device_state"; although + this was inconsistent as well in doxygen comments in + resource_applications. Because the actual resource being + subscribed to is /deviceStates/{device}/, it makes sense for the + resource type specifier to be deviceState. Note that the key + value in the events is still "device_state". ........ Merged + revisions 404437 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 20:00 +0000 [r404436] Richard Mudgett + + * res/ari/resource_channels.c, tests/test_scoped_lock.c, + tests/test_stasis.c, res/parking/parking_manager.c, + res/ari/resource_bridges.c, res/ari/resource_endpoints.c, /, + res/res_pjsip/location.c, tests/test_cel.c: ao2_iterator: + Mini-audit of the ao2_iterator loops in the new code files. * + Fixed several places where ao2_iterator_destroy() was not called. + * Fixed several iterator loop object variable reference problems. + * Fixed res_parking AMI actions returning non-zero. Only the AMI + logoff action can return non-zero. Review: + https://reviewboard.asterisk.org/r/3087/ ........ Merged + revisions 404434 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 19:25 +0000 [r404433] Matthew Jordan + + * include/asterisk/manager.h, /: manager: bump version to 2.0.0 AMI + has received substantial updates over the past year. Not only has + the syntax been vastly improved and made consistent (which + entails many event changes), but the underlying things that those + events convey have changed substantially as well. After some + conversation in #asterisk-dev, it was agreed that this is a good + time to jump to 2. At the same time, since ARI will most likely + use semantic versioning, we might as well use that for AMI as + well. That also affords us greater meaning for the AMI version. + ........ Merged revisions 404421 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 19:06 +0000 [r404420] Richard Mudgett + + * /, main/sounds_index.c: Whitespace fixes. ........ Merged + revisions 404419 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-20 17:22 +0000 [r404406] Rusty Newton + + * /, configs/pjsip.conf.sample: Documentation: Updates for info + about NAT-related settings and fixes for pjsip.conf.sample Added + another NAT example to pjsip.conf.sample. We had a few mentions + of NAT configuration throughout the sample, but I added another + for a little bit more clarity. Additionally many pjsip options + were affected by the change to snake case, so I fixed any + instances of those options in pjsip.conf. I regenerated the + config option list (at the bottom of the file) from a new xml + config doc dump, so all the snake case changes should be + reflected there, as well as any other changes to those options. + (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/ + ........ Merged revisions 404405 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 20:48 +0000 [r404387] Scott Griepentrog + + * main/security_events.c: security_events: log events with + descriptive names This patch updates the log messages to include + descriptive names for event types. This is an improvement over + having only cryptic type numbers. (closes issue ASTERISK-22909) + Reported by: outtolunc Review: + https://reviewboard.asterisk.org/r/3081/ Patches: + svn_security_events.c.names.diff.txt uploaded by outtolunc + (license 5198) + +2013-12-19 18:16 +0000 [r404376] Richard Mudgett + + * /, CHANGES: Put notice in CHANGES as well as UPGRADE.txt. + ........ Merged revisions 404375 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 18:00 +0000 [r404370-404372] Joshua Colp + + * res/res_pjsip/pjsip_outbound_auth.c, /: res_pjsip: Ignore 401/407 + responses for transactions and dialogs we don't know about. Under + normal conditions it is unlikely we will ever receive a response + for a transaction or dialog we don't know about but if any are + received ignore them. ........ Merged revisions 404371 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_session.c: res_pjsip_session: Fix SDP + negotiation when resending an INVITE with authentication. The + process for resending an INVITE with authentication involves + restarting the UAC session. We were incorrectly passing in that a + new offer is being sent, causing the SDP negotiation to get into + a (technically speaking) funky state. ........ Merged revisions + 404369 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 17:45 +0000 [r404368] Mark Michelson + + * include/asterisk/channel.h, res/res_pjsip.c, main/channel.c, /, + include/asterisk/autochan.h: Fix a deadlock that occurred due to + a conflict of masquerades. For the explanation, here is a + copy-paste of the review board explanation: Initially, it was + discovered that performing an attended transfer of a multiparty + bridge with a PJSIP channel would cause a deadlock. A PBX thread + started a masquerade and reached the point where it was calling + the fixup() callback on the "original" channel. For chan_pjsip, + this involves pushing a synchronous task to the session's + serializer. The problem was that a task ahead of the fixup task + was also attempting to perform a channel masquerade. However, + since masquerades are designed in a way to only allow for one to + occur at a time, the task ahead of the fixup could not continue + until the masquerade already in progress had completed. And of + course, the masquerade in progress could not complete until the + task ahead of the fixup task had completed. Deadlock. The initial + fix was to change the fixup task to be asynchronous. While this + prevented the deadlock from occurring, it had the frightful side + effect of potentially allowing for tasks in the session's + serializer to operate on a zombie channel. Taking a step back + from this particular deadlock, it became clear that the problem + was not really this one particular issue but that masquerades + themselves needed to be addressed. A PJSIP attended transfer + operation calls ast_channel_move(), which attempts to both set up + and execute a masquerade. The problem was that after it had set + up the masquerade, the PBX thread had swooped in and tried to + actually perform the masquerade. Looking at changes that had been + made to Asterisk 12, it became clear that there never is any time + now that anyone ever wants to set up a masquerade and allow for + the channel thread to actually perform the masquerade. Everyone + always is calling ast_channel_move(), performs the masquerade + itself before returning. In this patch, I have removed all blocks + of code from channel.c that will attempt to perform a masquerade + if ast_channel_masq() returns true. Now, there is no distinction + between setting up a masquerade and performing the masquerade. It + is one operation. The only remaining checks for + ast_channel_masq() and ast_channel_masqr() are in ast_hangup() + since we do not want to interrupt a masquerade by hanging up the + channel. Instead, now ast_hangup() will wait for a masquerade to + complete before moving forward with its operation. The + ast_channel_move() function has been modified to basically + in-line the logic that used to be in ast_channel_masquerade(). + ast_channel_masquerade() has been killed off for real. + ast_channel_move() now has a lock associated with it that is used + to prevent any simultaneous moves from occurring at once. This + means there is no need to make sure that ast_channel_masq() or + ast_channel_masqr() are already set on a channel when + ast_channel_move() is called. It also means the channel container + lock is not pulling double duty by both keeping the container + locked and preventing multiple masquerades from occurring + simultaneously. The ast_do_masquerade() function has been renamed + to do_channel_masquerade() and is now internal to channel.c. The + function now takes explicit arguments of which channels are + involved in the masquerade instead of a single channel. While it + probably is possible to do some further refactoring of this + method, I feel that I would be treading dangerously. Instead, all + I did was change some comments that no longer are true after this + changeset. The other more minor change introduced in this patch + is to res_pjsip.c to make ast_sip_push_task_synchronous() run the + task in-place if we are already a SIP servant thread. This is + related to this patch because even when we isolate the channel + masquerade to only running in the SIP servant thread, we would + still deadlock when the fixup() callback is reached since we + would essentially be waiting forever for ourselves to finish + before actually running the fixup. This makes it so the fixup is + run without having to push a task into a serializer at all. + (closes issue ASTERISK-22936) Reported by Jonathan Rose Review: + https://reviewboard.asterisk.org/r/3069 ........ Merged revisions + 404356 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 17:13 +0000 [r404355] Richard Mudgett + + * main/udptl.c, addons/chan_ooh323.c, /, channels/chan_sip.c, + include/asterisk/udptl.h: udptl: Dead code elimination. + ast_udptl_bridge was not used. Removing dead code starting with + ast_udptl_bridge() eliminated the code in this change. Note: This + code has actually been dead since Asterisk v1.4 when it was first + put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ + Merged revisions 404354 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 17:03 +0000 [r404353] Scott Griepentrog + + * /, res/res_fax.c: res_fax.c: crash on framehook with no dsp in + fax detect In fax_detect_framehook() a null pointer reference can + occur where a voice frame is processed but no dsp is attached to + the fax detection structure. The code block that rejects frames + that detection cannot be processed on is checking for dsp but + falls through when it should instead return, as this change + implements. (closes issue ASTERISK-22942) Reported by: adomjan + Review: https://reviewboard.asterisk.org/r/3076/ ........ Merged + revisions 404351 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404352 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 16:52 +0000 [r404350] Richard Mudgett + + * configs/skinny.conf.sample, res/res_xmpp.c, res/res_jabber.c, + CHANGES, channels/chan_iax2.c, channels/sig_pri.c, + channels/h323/chan_h323.h, configs/iax.conf.sample, + channels/sig_pri.h, channels/chan_dahdi.c, + include/asterisk/app.h, channels/chan_skinny.c, + channels/chan_dahdi.h, channels/chan_h323.c, main/app.c, + UPGRADE-12.txt, configs/sip.conf.sample, + channels/sip/include/sip.h, channels/chan_mgcp.c, + apps/app_voicemail.c, channels/chan_unistim.c, + configs/chan_dahdi.conf.sample, /, channels/chan_sip.c, + configs/voicemail.conf.sample, funcs/func_vmcount.c: Voicemail: + Remove mailbox identifier format (box@context) assumptions in the + system. This change is in preparation for external MWI support. + Removed code from the system for normal mailbox handling that + appends @default to the mailbox identifier if it does not have a + context. The only exception is the legacy hasvoicemail users.conf + option. The legacy option will only work for app_voicemail + mailboxes. The system cannot make any assumptions about the + format of the mailbox identifer used by app_voicemail. chan_sip + and chan_dahdi/sig_pri had the most changes because they both + tried to interpret the mailbox identifier. chan_sip just stored + and compared the two components. chan_dahdi actually used the box + information. The ISDN MWI support configuration options had to be + reworked because chan_dahdi was parsing the box@context format to + get the box number. As a result the mwi_vm_boxes chan_dahdi.conf + option was added and is documented in the chan_dahdi.conf.sample + file. Review: https://reviewboard.asterisk.org/r/3072/ ........ + Merged revisions 404348 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 16:33 +0000 [r404346] Scott Griepentrog + + * main/db.c, /: astdb: crash in sqlite3 during shutdown When + Asterisk is shut down, the astdb_atexit() function releases + (finalize) the previously initiated (prepared) SQL statements in + sqlite3. Another thread making a subsequent request can cause a + crash in sqlite3. This patch eliminates that issue by resetting + the statement pointer after it is released/cleared. The sqlite3 + code detects the null pointer, and aborts the operation cleanly. + (closes issue AST-1265) Reported by: Alexander Hömig (closes + issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter + Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged + revisions 404344 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404345 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 12:18 +0000 [r404333] Joshua Colp + + * main/channel.c, /: channel: Add a missing ast_channel_unlock when + allocating a Surrogate channel. ........ Merged revisions 404332 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 08:35 +0000 [r404321] Alexandr Anikin + + * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooGkClient.c, + addons/chan_ooh323.c, /, addons/ooh323c/src/ooGkClient.h: Handle + temporary failures on gk registration Introduce new 'stopped' + state for gk client and restart gk client on failures Remove + ooh323 stack command lock as it is not need now. (closes issue + ASTERISK-21960) Reported by: Dmitry Melekhov Patches: + ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested + by: Dmitry Melekhov ........ Merged revisions 404318 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404320 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 02:59 +0000 [r404307] Damien Wedhorn + + * /, channels/chan_skinny.c: Fixup some skinny bugs causing Fracks + and ao2 cleanup issues. Moved channel locking into setsubstate so + that a process can complete working on a sub before another + starts changing it. The existing code was causing some Fracks + with schedule deletion. Removed multiple rtp cleanup. Now only + cleansup up once, fixing ao2 object cleanup issues. ........ + Merged revisions 404306 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 00:50 +0000 [r404295] Matthew Jordan + + * include/asterisk/cdr.h, CHANGES, apps/app_cdr.c, main/cdr.c, + apps/app_forkcdr.c, main/pbx.c, /, funcs/func_cdr.c, + apps/app_disa.c, UPGRADE-12.txt: app_cdr,app_forkcdr,func_cdr: + Synchronize with engine when manipulating state When doing the + rework of the CDR engine that pushed all of the logic into cdr.c + and made it respond to changes in channel state over Stasis, we + knew that accessing the CDR engine from the dialplan would be + "slightly" non-deterministic. Dialplan threads would be accessing + CDRs while Stasis threads would be updating the state of said + CDRs - whereas in the past, everything happened on the dialplan + threads. Tests have shown that "slightly" is in reality "very". + This patch synchronizes things by making the dialplan + applications/functions that manipulate CDRs do so over Stasis. + ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to + send their requests over to the CDR engine, and synchronize on + the channel Stasis topic via a subscription so that they return + their values/control to the dialplan at the appropriate time. + While going through this, the following changes were also made: * + DISA, which can reset the CDR when a user successfully + authenticates, now just uses the ResetCDR app to do this. This + prevents having to duplicate the same Stasis synchronization + logic in that application. * Answer no longer disables CDRs. It + actually didn't work anyway - calling DISABLE on the channel's + CDR doesn't stop the CDR from getting the Answer time - it just + kills all CDRs on that channel, which isn't what the caller would + intend. (closes issue ASTERISK-22884) (closes issue + ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/ + ........ Merged revisions 404294 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-19 00:32 +0000 [r404293] Damien Wedhorn + + * /, channels/chan_skinny.c: Fixup skinny registration following + network issues. On session registration, if device is already + reporting that it is connected to a device, an innocuous packet + (update time) is sent to the already connected device. If the tcp + connection is down, the device will be unregistered and the new + connection allowed. Without this patch, network issues can see a + situation where a device can not reregister until after + 3*timeout. ........ Merged revisions 404292 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 23:00 +0000 [r404280] Jason Parker + + * main/manager.c, /: Add AMI event for presence state. Review: + https://reviewboard.asterisk.org/r/3039/ ........ Merged + revisions 404275 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404279 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 21:12 +0000 [r404264] Richard Mudgett + + * addons/ooh323c/src/ooTimer.c, /: ooh323c: Fix gcc 4.6.3 compiler + warnings. ........ Merged revisions 404212 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404219 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404263 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 20:48 +0000 [r404260-404262] Kevin Harwell + + * channels/chan_oss.c, /: chan_oss.c: channel being locked twice + and unlocked once Removed channel lock as it is now being down in + ast_channel_alloc ........ Merged revisions 404261 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c, + addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c, + channels/chan_pjsip.c, res/parking/parking_manager.c, + channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c, + funcs/func_timeout.c, /, apps/app_meetme.c, main/bridge.c, + tests/test_stasis_channels.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/sig_pri.c, apps/app_queue.c, + main/cel.c, main/stasis_bridges.c, channels/chan_jingle.c, + channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c, + channels/sig_analog.c, include/asterisk/stasis_channels.h, + res/res_agi.c, channels/chan_motif.c, tests/test_cel.c, + apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c, + apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc, + addons/chan_ooh323.c, main/pickup.c, include/asterisk/aoc.h, + include/asterisk/stasis_bridges.h, apps/app_userevent.c, + apps/app_disa.c, channels/chan_console.c, + include/asterisk/channelstate.h, main/core_local.c, + channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c, + res/parking/parking_bridge_features.c, apps/app_agent_pool.c, + main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c: + channel locking: Add locking for channel snapshot creation + Original commit message by mmichelson (asterisk 12 r403311): + "This adds channel locks around calls to create channel snapshots + as well as other functions which operate on a channel and then + end up creating a channel snapshot. Functions that expect the + channel to be locked prior to being called have had their + documentation updated to indicate such." The above was initially + committed and then reverted at r403398. The problem was found to + be in core_local.c in the publish_local_bridge_message function. + The ast_unreal_lock_all function locks and adds a reference to + the returned channels and while they were being unlocked they + were not being unreffed when no longer needed. Fixed by unreffing + the channels. Also in bridge.c a lock was obtained on + "other->chan", but then an attempt was made to unlock "other" and + not the previously locked channel. Fixed by unlocking + "other->chan" (closes issue ASTERISK-22709) Reported by: John + Bigelow ........ Merged revisions 404237 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 19:36 +0000 [r404211] Alexandr Anikin + + * addons/chan_ooh323.c, configs/ooh323.conf.sample: Introduce new + config option 'aniasdni'. If yes then asterisk set dialed number + as own id back to the caller on incoming h.323 calls. Option can + be set globally or per user section. (closes issue + ASTERISK-22020) Reported by: Ross Beer + +2013-12-18 19:28 +0000 [r404210] Joshua Colp + + * channels/chan_mgcp.c, main/pbx.c, channels/chan_sip.c, + apps/confbridge/conf_chan_record.c, tests/test_app.c, + tests/test_stasis_channels.c, main/core_unreal.c, + include/asterisk/channel.h, channels/chan_console.c, + channels/chan_oss.c, channels/chan_jingle.c, + channels/chan_misdn.c, channels/chan_h323.c, tests/test_cel.c, + channels/chan_nbs.c, channels/chan_pjsip.c, res/res_calendar.c, + apps/app_voicemail.c, channels/chan_unistim.c, + tests/test_substitution.c, channels/chan_vpb.cc, + addons/chan_ooh323.c, channels/chan_multicast_rtp.c, /, + apps/app_meetme.c, res/res_stasis_snoop.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c, + channels/chan_phone.c, channels/chan_skinny.c, + res/parking/parking_tests.c, channels/chan_motif.c, + tests/test_voicemail_api.c, channels/chan_alsa.c, main/message.c, + addons/chan_mobile.c, tests/test_cdr.c: channels: Return + allocated channels locked. This change makes ast_channel_alloc + return allocated channels locked. By doing so no other thread can + acquire, lock, and manipulate the channel before it is completely + set up. (closes issue AST-1256) Review: + https://reviewboard.asterisk.org/r/3067/ ........ Merged + revisions 404204 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 19:10 +0000 [r404198] Alexandr Anikin + + * addons/chan_ooh323.c: Implement module reload command for + chan_ooh323 (close issue ASTERISK-22817) Patches: + ooh323_module_reload.patch + +2013-12-18 12:46 +0000 [r404185] Matthew Jordan + + * rest-api/api-docs/applications.json, + rest-api/api-docs/playbacks.json, + rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json, + rest-api/resources.json, rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, + rest-api/api-docs/deviceStates.json, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + /, rest-api/api-docs/asterisk.json: ari: Bump the version of ARI + to 1.0.0 (closes issue ASTERISK-23007) ........ Merged revisions + 404184 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 12:01 +0000 [r404138] Joshua Colp + + * res/res_calendar.c, /: res_calendar: Protect channel when adding + datastore. This change adds a missing channel lock when adding a + datastore to a channel. ........ Merged revisions 404135 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404136 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404137 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 00:36 +0000 [r404100] Rusty Newton + + * /, funcs/func_strings.c: func_strings: Documentation fix for + QUOTE() Example output was inaccurate. (issue ASTERISK-22970) + (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches: + func_strings.patch uploaded by Gareth Palmer (license 5169) + ........ Merged revisions 404081 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 404087 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 404099 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-18 00:17 +0000 [r404051] Matthew Jordan + + * /, LICENSE: LICENSE: Update language to include ARI ........ + Merged revisions 404050 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 23:57 +0000 [r404049] Jonathan Rose + + * /, tests/test_cel.c, tests/test_cdr.c: tests: fix + ast_bridge_base_new calls not using the additional arguments + r404042 gave ast_bridge_base_new two new arguments for setting a + bridge creator and name. Unfortunately since a couple test + modules aren't compiled by default, I missed the fact that this + change impacted those tests and caused compilation failures + against them. ........ Merged revisions 404048 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 23:38 +0000 [r404047] Rusty Newton + + * include/asterisk/test.h, main/channel.c, main/rtp_engine.c, /, + channels/chan_iax2.c, apps/app_chanspy.c, apps/app_mixmonitor.c: + Several components: fixing Typos in comments and code, + "avaliable" instead of "available" (issue ASTERISK-23021) (closes + issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty + Newton Patches: available.patch uploaded by Jeremy Lainé (license + 6561) ........ Merged revisions 404046 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 23:25 +0000 [r404043] Jonathan Rose + + * apps/app_bridgewait.c, res/ari/ari_model_validators.c, + doc/appdocsxml.xslt, main/stasis_bridges.c, + rest-api/api-docs/bridges.json, res/ari/resource_bridges.c, + apps/app_agent_pool.c, res/parking/parking_bridge.c, + res/ari/ari_model_validators.h, main/manager_bridges.c, + res/ari/resource_bridges.h, include/asterisk/bridge_internal.h, + apps/app_confbridge.c, res/res_stasis.c, + include/asterisk/bridge.h, res/res_ari_bridges.c, /, + main/bridge.c, main/bridge_basic.c, + include/asterisk/stasis_bridges.h, include/asterisk/stasis_app.h: + bridging: Give bridges a name and a known creator Bridges have + two new optional properties, a creator and a name. Certain + consumers of bridges will automatically provide bridges that they + create with these properties. Examples include app_bridgewait, + res_parking, app_confbridge, and app_agent_pool. In addition, a + name may now be provided as an argument to the POST function for + creating new bridges via ARI. (closes issue AFS-47) Review: + https://reviewboard.asterisk.org/r/3070/ ........ Merged + revisions 404042 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 18:35 +0000 [r404028-404030] Joshua Colp + + * res/res_sorcery_config.c, /: res_sorcery_config: Output an error + message when an object can't be created. If object creation fails + an error message will now be output with the id, type, and + configuration file. ........ Merged revisions 404029 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/framehook.c: framehooks: Re-iterate if framehook provides + different frame. Framehooks can be used in a reactive manner to + execute specific logic when a frame is received with a certain + type and payload. Since it is possible for framehooks to provide + frames it was possible for this reactive framehook to be unaware + of frames it is looking for. This change makes it so that when + framehooks return a modified frame the code will now re-iterate + (from the beginning) and call any previous framehooks that have + not provided a modified frame themselves. Review: + https://reviewboard.asterisk.org/r/3046/ ........ Merged + revisions 404027 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 14:41 +0000 [r404008-404009] David M. Lee + + * /, configs/asterisk.conf.sample, main/asterisk.c: Changed the + default for live_dangerously to no ........ Merged revisions + 404006 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/pjsip, /: Setting svn:ignore ........ Merged revisions + 403748 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-17 12:59 +0000 [r403994] Matthew Jordan + + * /, res/ari/resource_channels.c: ari/resource_channels: When + creating a channel, specify a default format (SLIN) When creating + channels via ARI, the current code fails to provide any default + format capabilities. For non-virtual channels this isn't really a + problem - the channels typically receive their capabilities as a + result of the underlying channel driver configuration. For + virtual channels (such as Local channels), the lack of any format + capabilities causes the Asterisk core to make some 'odd' choices + with respect to the translation paths. The issue reporter had + some paths that had 3 hops on each channel leg, causing multiple + transcodings and some really crappy audio/performance. By + specifying a baseline of SLIN, we prevent that from occurring. + Note that this is what AMI does when it performs an Originate, as + does res_clioriginate. Review: + https://reviewboard.asterisk.org/r/3068/ (issue ASTERISK-22962) + Reported by: Matt DiMeo ........ Merged revisions 403993 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-16 19:11 +0000 [r403960] David M. Lee + + * include/asterisk/pbx.h, main/asterisk.c, funcs/func_realtime.c, + main/pbx.c, main/tcptls.c, funcs/func_db.c, /, + README-SERIOUSLY.bestpractices.txt, configs/asterisk.conf.sample, + funcs/func_shell.c, funcs/func_env.c, funcs/func_lock.c, + UPGRADE-12.txt: security: Inhibit execution of privilege + escalating functions This patch allows individual dialplan + functions to be marked as 'dangerous', to inhibit their execution + from external sources. A 'dangerous' function is one which + results in a privilege escalation. For example, if one were to + read the channel variable SHELL(rm -rf /) Bad Things(TM) could + happen; even if the external source has only read permissions. + Execution from external sources may be enabled by setting + 'live_dangerously' to 'yes' in the [options] section of + asterisk.conf. Although doing so is not recommended. Also, the + ABI was changed to something more reasonable, since Asterisk 12 + does not yet have a public release. (closes issue ASTERISK-22905) + Review: http://reviewboard.digium.internal/r/432/ ........ Merged + revisions 403913 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403917 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403959 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-16 18:31 +0000 [r403958] Jonathan Rose + + * /, main/bridge.c: transfers: Fix bug setting both BLINDTRANSFER + and ATTENDEDTRANSFER The ast_bridge_set_transfer_variables + function is supposed to wipe whichever variable isn't being set. + Instead it was setting both to the new value. Oops. (issue + AFS-24) ........ Merged revisions 403957 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-16 16:12 +0000 [r403857-403865] Scott Griepentrog + + * main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to + prevent memory corruption During dialplan execution in + pbx_extension_helper(), the contexts global read lock prevents + link list corruption, but was released with a pointer to the + ast_exten and data later used in variable substitution. Instead, + this patch removes pbx_substitute_variables() and locates a copy + of the ast_exten data on the stack before releasing the lock, + where ast_exten could get free'd by another thread performing a + module reload. (issue AST-1179) Reported by: Thomas Arimont + (issue AST-1246) Reported by: Alexander Hömig Review: + https://reviewboard.asterisk.org/r/3055/ ........ Merged + revisions 403862 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403863 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403864 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_sms.c: app_sms: BufferOverflow when receiving odd + length 16 bit message This patch prevents an infinite loop + overwriting memory when a message is received into the + unpacksms16() function, where the length of the message is an odd + number of bytes. (closes issue ASTERISK-22590) Reported by: Jan + Juergens Tested by: Jan Juergens ........ Merged revisions 403856 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-15 01:39 +0000 [r403824] Matthew Jordan + + * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions: + Use the right buffer length when printing URIs While + entertaining, sizeof(buflen) is not the same as buflen. Doh. + ........ Merged revisions 403823 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-14 17:28 +0000 [r403810-403812] Joshua Colp + + * include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c, + res/res_pjsip/pjsip_options.c, res/res_pjsip.c: res_pjsip: Apply + outbound proxy to all SIP requests. Objects which are involved in + SIP request creation and sending now allow an outbound proxy to + be specified. For cases where an endpoint is used the outbound + proxy specified there will be applied. (closes issue + ASTERISK-22673) Reported by: Antti Yrjola Review: + https://reviewboard.asterisk.org/r/3022/ ........ Merged + revisions 403811 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, apps/app_queue.c, + res/ari/ari_model_validators.c, apps/app_dial.c, + res/ari/ari_model_validators.h, main/dial.c, + include/asterisk/stasis_channels.h, + rest-api/api-docs/events.json, /, res/stasis/app.c: res_stasis: + Expose event for call forwarding and follow forwarded channel. + This change adds an event for when an originated call is + redirected to another target. This event contains the original + channel and the newly created channel. If a stasis subscription + exists on the original originated channel for a stasis + application then a new subscription will also be created on the + stasis application to the redirected channel. This allows the + application to follow the call path completely. (closes issue + ASTERISK-22719) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/3054/ ........ Merged + revisions 403808 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 21:35 +0000 [r403797] Jonathan Rose + + * /, res/res_pjsip_messaging.c, main/message.c: documentation: Add + PJSIP technology to messaging documentation ........ Merged + revisions 403796 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 20:17 +0000 [r403784] Richard Mudgett + + * /, main/test.c: test.c: Fix too sticky unit test failed status. + Rerunning a failed unit test after loading any required modules + should allow the test to report a pass status if it now passes. + ........ Merged revisions 403782 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 20:13 +0000 [r403783] Jonathan Rose + + * /, main/bridge.c, main/bridge_basic.c, include/asterisk/bridge.h, + res/parking/parking_bridge_features.c, + res/parking/parking_manager.c: Transfers: Make Asterisk set + ATTENDEDTRANSFER/BLINDTRANSFER more reliably There were still a + few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be + set on channels involved with blind and attended transfers. This + would happen with features that were initialized by channel + driver specific mechanisms in multiparty calls. This patch + resolves those cases while attempted to keep the behavior for + setting those variables as consistent as possible. (closes issue + AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........ + Merged revisions 403781 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 18:33 +0000 [r403750-403768] Kevin Harwell + + * main/channel.c, /, channels/chan_sip.c, + include/asterisk/channel.h, bridges/bridge_native_rtp.c, + channels/chan_pjsip.c: bridge_native_rtp: Deadlock during 4-way + conference creation The change contains a slightly adjusted patch + that was on the issue (submitted by kmoore). A fix was made by + adding in a bridge lock while calling bridge_start/stop from the + framehook callback. Since the framehook callback is not called + from the bridging core the bridge is not locked, but needs to be + before calling bridge_start. (closes issue ASTERISK-22749) + Reported by: Kinsey Moore Review: + https://reviewboard.asterisk.org/r/3066/ Patches: + lock_inversion.diff uploaded by kmoore (license 6273) ........ + Merged revisions 403767 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /, + main/http.c: ARI: Allow specifying channel variables during a + POST /channels Added the ability to specify channel variables + when creating/originating a channel in ARI. The variables are + sent in the body of the request and should be formatted as a + single level JSON object. No nested objects allowed. For example: + {"variable1": "foo", "variable2": "bar"}. (closes issue + ASTERISK-22872) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3052/ ........ Merged + revisions 403752 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_answer.c, rest-api/api-docs/bridges.json, + res/ari/resource_bridges.c, res/res_ari_bridges.c, + res/stasis/command.c, res/res_stasis_playback.c, /, + res/stasis/control.c, res/stasis/command.h, + include/asterisk/stasis_app.h, + include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c: + ARI: Adding a channel to a bridge while a live recording is + active blocks Added the ability to have rules that are checked + when adding and/or removing channels to/from a bridge. In this + case, if a channel is currently recording and someone attempts to + add it to a bridge an "is recording" rule is checked, fails, and + a 409 conflict is returned. Also command functions now return an + integer value that can be descriptive of what kind of problems, + if any, occurred before or during execution. (closes issue + ASTERISK-22624) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2947/ ........ Merged + revisions 403749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 05:00 +0000 [r403737] Matthew Jordan + + * /, channels/Makefile: channels/Makefile: clean pjsip directory + ........ Merged revisions 403736 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-13 00:40 +0000 [r403726] Richard Mudgett + + * include/asterisk/app.h, tests/test_voicemail_api.c, main/app.c: + test_voicemail_api: Add check for a registered voicemail provider + before tests. It is much nicer diagnosing a test failure if + app_voicemail is actually loaded. + +2013-12-12 19:46 +0000 [r403714] Scott Griepentrog + + * contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py + (added), /: realtime: Create extensions in alembic ast-db-manage + contribution When the alembic scripts were written for creating + Asterisk realtime databases the extensions table for dialplan + wasn't included. This update creates the extensions table. + (closes issue ASTERISK-22815) Reported by: Zone Conkle Review: + https://reviewboard.asterisk.org/r/3064/ ........ Merged + revisions 403713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-12 19:18 +0000 [r403707] Jonathan Rose + + * /, channels/chan_pjsip.c: chan_pjsip: Revert r403587 This patch + was intended to eliminate a deadlock that occurs when masquerades + occur in pjsip channels, but has some potential side effects. + Mark Michelson is currently working on addressing this problem + from another angle. (issue ASTERISK-22936) Reported by: Jonathan + Rose ........ Merged revisions 403705 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-11 20:24 +0000 [r403687] Kevin Harwell + + * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, /, + configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip_messaging.c, + res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c: + res_pjsip_messaging: send message to a default outbound endpoint + In some cases messages need to be sent to a direct URI (sip:). This patch adds in that support by using a default + outbound endpoint. When sending messages, if no endpoint can be + found then the default one is used. To facilitate this a new + default_outbound_endpoint option was added to the globals section + for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ + ........ Merged revisions 403680 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-11 19:22 +0000 [r403652] Russell Bryant + + * /, channels/chan_sip.c: Reset peer outboundproxy on sip.conf + reload If you set a peer's outboundproxy and then removed it from + the config, this would not get picked up in a config reload. This + patch fixes that by resetting it in set_peer_defaults(). Closes + ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/ + ........ Merged revisions 403634 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403635 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403639 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-11 19:19 +0000 [r403643] Richard Mudgett + + * apps/app_voicemail.c, include/asterisk/app.h, + include/asterisk/doxyref.h, main/app.c: app_voicemail: Voicemail + callback registration/unregistration function improvements. * The + voicemail registration/unregistration functions now take a struct + of callbacks instead of a lengthy parameter list of callbacks. * + The voicemail registration/unregistration functions now prevent a + competing module from interfering with an already registered + callback supplying module. + +2013-12-11 13:06 +0000 [r403617-403619] Matthew Jordan + + * channels/pjsip/dialplan_functions.c, + include/asterisk/res_pjsip_session.h, channels/pjsip (added), /, + funcs/func_channel.c, channels/pjsip/include, + channels/pjsip/include/dialplan_functions.h, res/res_pjsip_t38.c, + channels/pjsip/include/chan_pjsip.h, channels/Makefile, + channels/chan_pjsip.c, main/xmldoc.c: func_channel, chan_pjsip: + Add CHANNEL read function support for chan_pjsip This patch adds + CHANNEL read support for chan_pjsip. This allows the dialplan to + use the CHANNEL function on a chan_pjsip channel to obtain + run-time information about the channel from the PJSIP channel + driver and the PJSIP stack. This includes: * RTP information, + including source/destination media addresses, whether or not the + media is secure, held, and other properties. * RTCP information. + This includes sets of parseable information, as well as + individual statistic attriutes. * PJSIP information. This + includes URIs, local/remote signalling addresses, whether or not + the signalling is secure, and other properties. * The endpoint + name. This can be used in conjunction with the PJSIP_ENDPOINT + function to obtain more detailed endpoint information. Review: + https://reviewboard.asterisk.org/r/3038/ ........ Merged + revisions 403618 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * Makefile, funcs/func_pjsip_endpoint.c (added), doc/snapshots.xslt + (removed), /, doc/appdocsxml.xslt (added), doc/appdocsxml.dtd, + main/sorcery.c: func_pjsip_endpoint: Add PJSIP_ENDPOINT function + for querying endpoint details This patch adds a new function, + PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint, + any property configured on an endpoint. This function is a + companion to the CHANNEL function, which can be used to extract + the endpoint name for a channel. Review: + https://reviewboard.asterisk.org/r/3035 ........ Merged revisions + 403616 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-10 15:15 +0000 [r403605] Mark Michelson + + * res/res_pjsip_authenticator_digest.c: Fix correct authentication + behavior for artificial endpoint. When switching to using a + vector for authentication, I initialized the vector for the + artificial endpoint to be of size 1. However, this does not + result in AST_VECTOR_SIZE() returning 1 since there isn't + actually anything in the vector. Rather than trifle with the + vector by putting unnecessary elements in, I simply changed the + callback in res_pjsip_authenticator_digest.c to explicitly report + that the artificial endpoint requires authentication. Thanks to + Joshua Colp for pointing this out. + +2013-12-09 22:59 +0000 [r403576-403588] Jonathan Rose + + * /, channels/chan_pjsip.c: chan_pjsip: Fix a sticking channel lock + caused by channel masquerades (closes issue ASTERISK-22936) + Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/3042/ ........ Merged + revisions 403587 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * CHANGES, main/dial.c, apps/app_page.c, include/asterisk/dial.h: + app_page: Add predial handlers for app_page. (closes issue + AFS-14) Review: https://reviewboard.asterisk.org/r/3045/ + +2013-12-09 19:24 +0000 [r403544-403560] Richard Mudgett + + * /, res/res_sorcery_astdb.c: Reverting regex part of -r403545 at + request of file. res_sorcery_astdb.c: Fix get multiple records by + regex. * Fix sorcery_astdb_retrieve_regex() pattern matching. Let + the regexec() function match the stored key values instead of + having astdb prefilter them. Previoiusly you could only use a + simple regex pattern when the pattern began with '^'. ........ + Merged revisions 403559 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix get multiple + records by regex. * Fix sorcery_astdb_retrieve_regex() pattern + matching. Let the regexec() function match the stored key values + instead of having astdb prefilter them. Previoiusly you could + only use a simple regex pattern when the pattern began with '^'. + * Fix off nominal memory leak in sorcery_astdb_retrieve_regex(). + ........ Merged revisions 403545 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/sorcery.c, /: sorcery: Eliminate shadowing a varaible that + caused confusion. * Eliminated shadowing of the + __ast_sorcery_apply_config() name parameter causing confusion. * + Fix potential crash from sorcery.conf user input in + __ast_sorcery_apply_config() if the user supplied a malformed + config line that is missing the sorcery object type name. * + Remove redundant test in __ast_sorcery_apply_config(). !config + and config == CONFIGS_STATUS_FILEMISSING are identical. ........ + Merged revisions 403541 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 18:32 +0000 [r403543] Joshua Colp + + * /, main/endpoints.c: endpoints: Keep a reference to channel ids + when creating snapshot. The snapshot process for endpoints uses + the channel ids present on the endpoint itself. Without keeping a + reference it was possible for the strings to be freed underneath + any consumer of an endpoint snapshot. A reference is now held by + the snapshot to the channel ids and released when the snapshot is + destroyed. (issue ASTERISK-22801) Reported by: Matt Jordan + ........ Merged revisions 403542 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 18:14 +0000 [r403528] Richard Mudgett + + * main/sorcery.c, /: sorcery: Whitespace You would think that a new + file would start off without any whitespace oddities. ........ + Merged revisions 403527 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 17:29 +0000 [r403512-403526] Mark Michelson + + * apps/app_confbridge.c, CHANGES, + apps/confbridge/conf_state_multi_marked.c: Add a + CONFBRIDGE_RESULT channel variable to discern why a channel left + a ConfBridge. Review: https://reviewboard.asterisk.org/r/3009 + + * CHANGES, apps/app_mixmonitor.c: Create function for retrieving + Mixmonitor instance data. For the time, this is only useful for + retrieving the filename. The purpose of this function is to + better facilitate multiple mixmonitors per channel. Setting a + MIXMONITOR_FILENAME channel variable is not conducive to such + behavior, so allowing finer grained access to individual + mixmonitor properties improves the situation. The + MIXMONITOR_FILENAME channel variable is still set, though, so + there is no worry about backwards compatibility. Review: + https://reviewboard.asterisk.org/r/3023 + +2013-12-09 16:41 +0000 [r403511] Joshua Colp + + * res/res_pjsip_nat.c, /: res_pjsip_nat: Add NAT module to session + dialogs. Due to the way pjproject internally works it was + possible for the NAT module to not be invoked on messages with-in + a session dialog. This means that the various parts of the + message would not get rewritten with the source IP address and + port. This change uses a session supplement to add the NAT module + to the dialog on the first incoming or outgoing INVITE. (closes + issue ASTERISK-22941) Reported by: Leif Madsen ........ Merged + revisions 403510 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-09 16:10 +0000 [r403499] Mark Michelson + + * res/res_pjsip/config_auth.c, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_pjsip_authenticator_digest.c, + res/res_pjsip_outbound_registration.c, + res/res_pjsip/pjsip_configuration.c, + res/res_pjsip/pjsip_distributor.c, res/res_pjsip.c, + include/asterisk/res_pjsip.h: Switch PJSIP auth to use a vector. + Since Asterisk has a vector API now, places where arrays are + manually resized don't really make sense any more. Since the auth + work in PJSIP was freshly-written, it was easy to reform it to + use a vector. Review: https://reviewboard.asterisk.org/r/3044 + +2013-12-09 03:21 +0000 [r403436-403466] Matthew Jordan + + * /, res/res_fax_spandsp.c: res_fax_spandsp: Always init T.38 + session to avoid crashes during state change Prior to this patch, + res_fax_spandsp was conservative with how it initialized the + spandsp T.38 context. It would only initialize it if the driver + thought the current state was a T.38 fax. While this works fine + in nominal situations, in certain off nominal situations, + res_fax_spandsp can believe that a T.38 fax will not occur when + in fact one has started. In particular, this was discovered when + res_fax would fall back to audio after timing out on a T.38 + upgrade. The SIP channel driver would continue to retry the + re-INVITE and - if the remote end responded after res_fax timed + out with a 200 OK - a T.38 frame would be delivered to the + res_fax stack when it no longer expected it. As it turns out, + there does not appear to be any downside to always initializing + the T.38 context, other than the actual memory allocation. Since + that avoids this off nominal situation (and others which are + equally likely hard to predict), this is the safest way to avoid + this problem. Much thanks to Torrey as well for providing a + scenario that reproduces this issue. (closes issue + ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey + Searle patches: always-init-t38.patch uploaded by awinters + (License 6477) A_PARTY.xml uploaded by tsearle (License 5334) + ........ Merged revisions 403449 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403450 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_config_sqlite.c: res_config_sqlite: Check for CDR + unregistration failures If the CDR unregistration fails due to an + inflight CDR, the res_config_sqlite module needs to bail on + unloading itself. Otherwise, the config could be unloaded + (including the CDR table name) while the CDR engine posts a CDR + to the still registered backend, resulting in a crash. ........ + Merged revisions 403435 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-05 23:40 +0000 [r403414] Jonathan Rose + + * apps/app_record.c: app_record: Add an option that allows DTMF '0' + to act as an additional terminator Using this terminator when + active results in ${RECORD_STATUS} being set to 'OPERATOR' + instead of 'DTMF' (closes issue AFS-7) Review: + https://reviewboard.asterisk.org/r/3041/ + +2013-12-05 22:10 +0000 [r403402-403404] David M. Lee + + * addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c, + channels/chan_pjsip.c, res/parking/parking_manager.c, + channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c, /, + apps/app_meetme.c, funcs/func_timeout.c, main/bridge.c, + tests/test_stasis_channels.c, main/core_unreal.c, + include/asterisk/channel.h, channels/chan_gtalk.c, main/cel.c, + apps/app_queue.c, channels/sig_pri.c, main/stasis_bridges.c, + channels/chan_jingle.c, channels/chan_phone.c, + channels/chan_dahdi.c, main/dial.c, channels/sig_analog.c, + include/asterisk/stasis_channels.h, res/res_agi.c, + channels/chan_motif.c, channels/chan_h323.c, tests/test_cel.c, + apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c, + apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc, + addons/chan_ooh323.c, channels/chan_sip.c, main/pickup.c, + include/asterisk/aoc.h, include/asterisk/stasis_bridges.h, + apps/app_userevent.c, apps/app_disa.c, main/core_local.c, + include/asterisk/channelstate.h, channels/chan_console.c, + channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c, + res/parking/parking_bridge_features.c, apps/app_agent_pool.c, + main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, + pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c, + channels/chan_nbs.c: Reverting r403311. It's causing ARI tests to + hang. ........ Merged revisions 403398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c: ari: Fix deadlock problem with functions + that use autoservice. The code for getting channel variables from + ARI assumed that you needed to lock the channel in order to + properly execute functions and read channel variables. + Apparently, this is not the case, since any dialplan function + that puts the channel into autoservice deadlocks when attempting + to remove the channel from autoservice. ........ Merged revisions + 403342 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Multiple revisions 403304,403310 ........ r403304 | dlee | + 2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line Fixed the + filename for the ari.conf docs ........ r403310 | file | + 2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines Revert + revision 403304: Fixed the filename for the ari.conf docs The + changed value refers to the name of the module. The name of the + configuration file is specified in the configFile section. + ........ Merged revisions 403304,403310 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-04 21:42 +0000 [r403378] Kevin Harwell + + * /, res/res_pjsip_registrar.c: res_pjsip_registrar: undefined + function pointer symbol Used a static wrapper around the + offending function to alleviate the issue. Reported by: rmudgett + ........ Merged revisions 403377 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-04 20:54 +0000 [r403365] Joshua Colp + + * res/res_pjsip_t38.c, /: res_pjsip_t38: Don't pass T.38 control + frames through to other hooks. This crept up during gateway + testing where the gateway would receive the request to negotiate + and assume it came from the remote side, causing the gateway + state machine to go a little, to a use a technical term, "wonky". + ........ Merged revisions 403364 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-04 18:41 +0000 [r403350] Mark Michelson + + * /, res/res_pjsip.c: Initialize the hash value argument to + pj_hash_get() to 0. Passing a non-zero value causes PJLIB to use + the given input as the hash value. Passing zero causes the + parameter to become an output parameter that receives the hash + value that was computed based on the given key. This change + essentially makes ast_sip_dict_get() properly retrieve the + desired value. ........ Merged revisions 403349 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 18:01 +0000 [r403330] Joshua Colp + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_pjsip_session.c: res_pjsip_session: Add support for + PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. Newer versions of PJSIP + have changed to using a flag for the + PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds + a configure check to detect the presence of the flag and use it + if found. ........ Merged revisions 403329 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 17:35 +0000 [r403327] Richard Mudgett + + * include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip_registrar_expire.c, res/res_pjsip/pjsip_options.c, + tests/test_sorcery.c, include/asterisk/bucket.h, main/sorcery.c, + /, main/bucket.c: sorcery, bucket: Change observer remove calls + to take const callbacks struct. * Make + ast_sorcery_observer_remove() accept a const callbacks struct. * + Make ast_sorcery_observer_remove() tolerant of the sorcery + parameter being NULL. Now it can be called within a module unload + routine if the sorcery initialization fails. * Fix + ast_sorcery_observer_add() to fail if the container link fails. + ........ Merged revisions 403324 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 17:07 +0000 [r403314] Mark Michelson + + * channels/chan_nbs.c, main/bridge_channel.c, res/res_stasis.c, + channels/chan_pjsip.c, res/parking/parking_manager.c, + apps/app_voicemail.c, channels/chan_unistim.c, + channels/chan_vpb.cc, addons/chan_ooh323.c, /, + include/asterisk/aoc.h, apps/app_meetme.c, main/bridge.c, + apps/app_userevent.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/endpoints.c, main/stasis_bridges.c, + main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, + main/dial.c, channels/sig_analog.c, channels/chan_skinny.c, + res/res_agi.c, channels/chan_motif.c, pbx/pbx_realtime.c, + channels/chan_alsa.c, main/stasis_channels.c, + apps/app_confbridge.c, addons/chan_mobile.c, tests/test_cdr.c, + res/res_pjsip_refer.c, channels/chan_mgcp.c, apps/app_dial.c, + main/pbx.c, channels/chan_sip.c, main/pickup.c, + funcs/func_timeout.c, tests/test_stasis_channels.c, + main/core_unreal.c, include/asterisk/stasis_bridges.h, + apps/app_disa.c, include/asterisk/channel.h, main/core_local.c, + include/asterisk/channelstate.h, channels/chan_console.c, + main/cel.c, apps/app_queue.c, channels/sig_pri.c, + channels/chan_oss.c, res/parking/parking_bridge_features.c, + apps/app_agent_pool.c, channels/chan_jingle.c, + channels/chan_misdn.c, include/asterisk/stasis_channels.h, + channels/chan_h323.c, tests/test_cel.c: Add channel locking for + channel snapshot creation. This adds channel locks around calls + to create channel snapshots as well as other functions which + operate on a channel and then end up creating a channel snapshot. + Functions that expect the channel to be locked prior to being + called have had their documentation updated to indicate such. + ........ Merged revisions 403311 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-03 16:39 +0000 [r403313] Joshua Colp + + * main/media_index.c, /: media_index: Make media indexing tolerable + of bad symlinks. Media indexing will now skip over files and + directories that stat will not return information about. This can + occur under normal conditions when a symbolic link points to a + location that no longer exists. ........ Merged revisions 403312 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-02 18:12 +0000 [r403292] Alexandr Anikin + + * addons/chan_ooh323.c, /: Check and reject non-digits e164 values + on peers and general sections in ooh323.conf Regenerate e164 + endpoint list on reload ooh323 (issue ASTERISK-22901) Reported + by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch ........ + Merged revisions 403288 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403290 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-12-01 21:13 +0000 [r403257-403272] Joshua Colp + + * /, res/res_pjsip_session.c: res_pjsip_session: Apply fromuser and + fromdomain to all requests as documented. ........ Merged + revisions 403271 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_t38.c, /: res_pjsip_t38: Add the framehook to the + channel only on first INVITE. The check for determining whether + the T.38 framehook should be added to the channel or not has now + been changed to guarantee adding only occurs on the first + incoming or outgoing INVITE. ........ Merged revisions 403258 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/security_events.c, res/res_pjsip/pjsip_options.c, + res/res_pjsip.c, res/res_pjsip_transport_websocket.c, + include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c: + res_pjsip_transport_websocket: Fix security events and simplify + implementation. Transport type determination for security events + has been simplified to use the type present on the message itself + instead of searching through configured transports to find the + transport used. The actual WebSocket transport has also been + simplified. It now leverages the existing PJSIP transport manager + for finding the active WebSocket transport for outgoing messages. + This removes the need for res_pjsip_transport_websocket to store + a mapping itself. (closes issue ASTERISK-22897) Reported by: Max + E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/ + ........ Merged revisions 403256 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-30 14:12 +0000 [r403241] Joshua Colp + + * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /, + res/ari/ari_model_validators.c: res_ari: Add Recording events to + the validator. ........ Merged revisions 403240 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-28 02:12 +0000 [r403208-403224] Joshua Colp + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't produce an + invalid media stream with no formats. Depending on configuration + it was possible for a media stream to be created without any + media formats. The produced SDP would fail internal validation + and cause a crash. The code will now no longer add media streams + with no formats to the SDP, allowing it to pass validation and + work. (closes issue ASTERISK-22858) Reported by: Anthony Messina + ........ Merged revisions 403223 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_header_funcs.c, /: res_pjsip_header_funcs: Don't + add headers to re-INVITEs. When sending a re-INVITE to an + endpoint it was possible for received headers to be added as well + (since they are stored for retrieval using the PJSIP_HEADER + dialplan function). This caused a broken (and potentially large) + SIP INVITE to be produced and sent. This changes the module so it + will no longer add headers to re-INVITEs. (closes issue + ASTERISK-22882) Reported by: David M. Lee ........ Merged + revisions 403221 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /: res_stasis_playback: Add 'number', + 'digits', and 'characters' URI scheme implementations. This + change adds new URI scheme implementations for playing numbers, + digits, and characters. This is done as part of the normal + playback mechanism and can be used with queueing to create a + combined sentence. Review: + https://reviewboard.asterisk.org/r/3028/ ........ Merged + revisions 403209 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c, + res/res_pjsip_session.c, include/asterisk/res_pjsip.h: + res_pjsip_session: Add configurable behavior for redirects. The + action taken when a redirect occurs is now configurable on a + per-endpoint basis. The redirect can either be treated as a + redirect to a local extension, to a URI that is dialed through + the Asterisk core, or to a URI that is dialed within PJSIP + itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged + revisions 403207 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-27 17:32 +0000 [r403192] Richard Mudgett + + * include/asterisk/astdb.h: astdb: Tweak some doxygen comments. + +2013-11-27 16:12 +0000 [r403180] Joshua Colp + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix crash when + reloading certain configurations. Certain options available that + specify a SIP URI perform validation on the provided URI using + the PJSIP URI parser. This operation requires that the thread + executing it be registered with the PJLIB library. During reloads + this was done on a thread which was NOT registered with it. This + fixes the problem by creating a task which reloads the + configuration on a PJSIP thread. (closes issue ASTERISK-22923) + Reported by: Anthony Messina ........ Merged revisions 403179 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-27 15:48 +0000 [r403177] David M. Lee + + * res/res_ari_channels.c, include/asterisk/ari.h, + rest-api-templates/param_parsing.mustache, + include/asterisk/http.h, res/res_ari_recordings.c, + res/res_ari_endpoints.c, main/http.c, + rest-api-templates/swagger_model.py, res/res_ari_playbacks.c, + res/res_ari_sounds.c, rest-api-templates/asterisk_processor.py, + res/res_ari_bridges.c, tests/test_ari.c, res/res_ari.c, /, + res/res_ari_device_states.c, res/res_ari_asterisk.c, + rest-api-templates/res_ari_resource.c.mustache, + res/res_ari_applications.c: ari:Add application/json parameter + support The patch allows ARI to parse request parameters from an + incoming JSON request body, instead of requiring the request to + come in as query parameters (which is just weird for POST and + DELETE) or form parameters (which is okay, but a bit asymmetric + given that all of our responses are JSON). For any operation that + does _not_ have a parameter defined of type body (i.e. + "paramType": "body" in the API declaration), if a request + provides a request body with a Content type of + "application/json", the provided JSON document is parsed and + searched for parameters. The expected fields in the provided JSON + document should match the query parameters defined for the + operation. If the parameter has 'allowMultiple' set, then the + field in the JSON document may optionally be an array of values. + (closes issue ASTERISK-22685) Review: + https://reviewboard.asterisk.org/r/2994/ + +2013-11-27 15:31 +0000 [r403161-403174] Joshua Colp + + * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Update + handling of some options to work with new option names. Some + options (such as call_group and pickup_group) share the same + configuration handler and decide what logic to use based on the + name of the option. These handlers were not updated to check for + the new option names and were treating the options as invalid. + This change simply updates the handlers with the proper names of + the options. (closes issue ASTERISK-22922) Reported by: Anthony + Messina ........ Merged revisions 403173 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Fix + a configure issue with PJSIP transaction group lock detection. + The configure check did not use the provided paths for pjproject + if provided when looking for transaction group lock support. + ........ Merged revisions 403160 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-23 17:48 +0000 [r403133-403135] Kevin Harwell + + * res/ari.make, rest-api/api-docs/applications.json, + res/ari/resource_device_states.h (added), + include/asterisk/stasis_app_device_state.h (added), + res/ari/resource_applications.h, res/res_stasis.c, + include/asterisk/devicestate.h, rest-api/api-docs/events.json, + res/res_stasis_device_state.exports.in (added), res/stasis/app.c, + res/res_ari_device_states.c (added), /, + include/asterisk/stasis_app.h, main/devicestate.c, + res/stasis/app.h, rest-api/resources.json, + res/res_stasis_device_state.c (added), + res/ari/ari_model_validators.c, res/ari/ari_model_validators.h, + res/ari/resource_device_states.c (added), + rest-api/api-docs/deviceStates.json (added), + rest-api-templates/ari.make.mustache: ARI: Implement device state + API Created a data model and implemented functionality for an ARI + device state resource. The following operations have been added + that allow a user to manipulate an ARI controlled device: + Create/Change the state of an ARI controlled device PUT + /deviceStates/{deviceName}&{deviceState} Retrieve all ARI + controlled devices GET /deviceStates Retrieve the current state + of a device GET /deviceStates/{deviceName} Destroy a device-state + controlled by ARI DELETE /deviceStates/{deviceName} The ARI + controlled device must begin with 'Stasis:'. An example + controlled device name would be Stasis:Example. A + 'DeviceStateChanged' event has also been added so that an + application can subscribe and receive device change events. Any + device state, ARI controlled or not, can be subscribed to. While + adding the event, the underlying subscription control mechanism + was refactored so that all current and future resource + subscriptions would be the same. Each event resource must now + register itself in order to be able to properly handle + [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/3025/ ........ Merged + revisions 403134 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_registrar.c, main/sorcery.c, + include/asterisk/res_pjsip.h, include/asterisk/acl.h, + res/res_pjsip/config_auth.c, include/asterisk/utils.h, + res/res_pjsip.exports.in, /, + res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, main/utils.c, + res/res_pjsip.c, res/res_pjsip_exten_state.c, + include/asterisk/res_pjsip_pubsub.h, res/res_pjsip/location.c, + res/res_pjsip_outbound_registration.c, res/res_pjsip_mwi.c, + res/res_pjsip/pjsip_configuration.c, include/asterisk/sorcery.h, + include/asterisk/strings.h, + res/res_pjsip/include/res_pjsip_private.h, + res/res_pjsip_pubsub.c, res/res_pjsip/config_transport.c: + res_pjsip: AMI commands and events. Created the following AMI + commands and corresponding events for res_pjsip: + PJSIPShowEndpoints - Provides a listing of all pjsip endpoints + and a few select attributes on each. Events: EndpointList - for + each endpoint a few attributes. EndpointlistComplete - after all + endpoints have been listed. PJSIPShowEndpoint - Provides a detail + list of attributes for a specified endpoint. Events: + EndpointDetail - attributes on an endpoint. AorDetail - raised + for each AOR on an endpoint. AuthDetail - raised for each + associated inbound and outbound auth TransportDetail - transport + attributes. IdentifyDetail - attributes for the identify object + associated with the endpoint. EndpointDetailComplete - last event + raised after all detail events. PJSIPShowRegistrationsInbound - + Provides a detail listing of all inbound registrations. Events: + InboundRegistrationDetail - inbound registration attributes for + each registration. InboundRegistrationDetailComplete - raised + after all detail records have been listed. + PJSIPShowRegistrationsOutbound - Provides a detail listing of all + outbound registrations. Events: OutboundRegistrationDetail - + outbound registration attributes for each registration. + OutboundRegistrationDetailComplete - raised after all detail + records have been listed. PJSIPShowSubscriptionsInbound - A + detail listing of all inbound subscriptions and their attributes. + Events: SubscriptionDetail - on each subscription detailed + attributes SubscriptionDetailComplete - raised after all detail + records have been listed. PJSIPShowSubscriptionsOutbound - A + detail listing of all outboundbound subscriptions and their + attributes. Events: SubscriptionDetail - on each subscription + detailed attributes SubscriptionDetailComplete - raised after all + detail records have been listed. (issue ASTERISK-22609) Reported + by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ + ........ Merged revisions 403131 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-23 12:52 +0000 [r403118-403120] Joshua Colp + + * res/res_stasis_playback.c, rest-api/api-docs/events.json, /, + res/res_stasis_recording.c, res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, + res/ari/ari_model_validators.h: ari: Add events for playback and + recording. While there were events defined for playback and + recording these were not actually sent. This change implements + the to_json handlers which produces them. (closes issue + ASTERISK-22710) Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/3026/ ........ Merged + revisions 403119 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_snoop.exports.in (added), /, + include/asterisk/stasis_app_snoop.h (added), + rest-api/api-docs/channels.json, res/res_stasis_snoop.c (added), + main/audiohook.c, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h: ari: Add + Snoop operation for spying/whispering on channels. The Snoop + operation can be invoked on a channel to spy or whisper on it. It + returns a channel that any channel operations can then be invoked + on (such as record to do monitoring). (closes issue + ASTERISK-22780) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/3003/ ........ Merged + revisions 403117 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-23 00:22 +0000 [r403106] Rusty Newton + + * apps/app_voicemail.c: app_voicemail: when forwarding a message, + play vm-msgforwarded instead of vm-msgsaved In the last release + of sounds, 1.4.25 we added a vm-msgforwarded prompt for various + core languages. Now we use that prompt. (issue ASTERISK-21413) + (closes issue ASTERISK-21413) Reported by: netwrkr Tested by: + newtonr + +2013-11-22 23:57 +0000 [r403095] Kinsey Moore + + * tests/test_stasis.c, /, tests/test_stasis_channels.c: Make sure + unit tests compile This fixes the unit tests that were broken by + r403069 and several functions requiring a new parameter for + sanitization of JSON messages generated from object snapshots. + ........ Merged revisions 403094 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 22:37 +0000 [r403083] Kevin Harwell + + * /, + contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py, + res/res_pjsip/pjsip_configuration.c: res_pjsip: convert + configuration settings names to snake case some more Updated the + alembic script for pjsip. Also, the dtls config parsing stuff was + expecting strings with no underscores, so removed the underscores + from the option name before passing it to the parser. ........ + Merged revisions 403082 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 20:10 +0000 [r403070] Kinsey Moore + + * res/res_stasis.c, main/stasis_endpoints.c, + res/ari/resource_endpoints.c, main/rtp_engine.c, /, + res/stasis/app.c, include/asterisk/stasis_bridges.h, + include/asterisk/stasis.h, include/asterisk/stasis_app.h, + main/stasis_bridges.c, res/ari/resource_bridges.c, main/json.c, + main/stasis_message.c, include/asterisk/stasis_channels.h, + main/stasis_channels.c, res/ari/resource_channels.c, + include/asterisk/stasis_endpoints.h: ARI: Don't leak + implementation details This change prevents channels used as + implementation details from leaking out to ARI. It does this by + preventing creation of JSON blobs of channel snapshots created + from those channels and sanitizing JSON blobs of bridge snapshots + as they are created. This introduces a framework for excluding + information from output targeted at Stasis applications on a + consumer-by-consumer basis using channel sanitization callbacks + which could be extended to bridges or endpoints if necessary. + This prevents unhelpful error messages from being generated by + ast_json_pack. This also corrects a bug where BridgeCreated + events would not be created. (closes issue ASTERISK-22744) + Review: https://reviewboard.asterisk.org/r/2987/ Reported by: + David M. Lee ........ Merged revisions 403069 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 17:27 +0000 [r403051] Kevin Harwell + + * res/res_pjsip_acl.c, res/res_pjsip.c, + res/res_pjsip/config_transport.c, res/res_pjsip/config_global.c, + /, configs/pjsip.conf.sample, res/res_pjsip/config_system.c, + contrib/scripts/sip_to_pjsip/sip_to_pjsip.py, + res/res_pjsip/pjsip_configuration.c: res_pjsip: convert + configuration settings names to snake case Renamed, where + appropriate, the configuration options for chan/res_pjsip to use + snake case (compound words separated by an underscore). For + example, faxdetect will become fax_detect, recordofffeature will + become record_off_feature, etc... Review: + https://reviewboard.asterisk.org/r/3002/ ........ Merged + revisions 403022 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 17:12 +0000 [r403017] Joshua Colp + + * /, main/translate.c: translate: Move freeing of frame to after it + is used. When translating from one format to another it is + possible to inform the translation function that the source frame + should be freed. This was previously done immediately but shortly + afterwards the frame that was freed was accessed and used again. + This change moves code around a bit so that the frame is now + freed after it has been completely used. (closes issue + ASTERISK-22788) Reported by: Corey Farrell Patches: + translate-access-after-free-11up.patch uploaded by coreyfarrell + (license 5909) translate-access-after-free-1.8.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 403014 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 403015 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 403016 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-22 16:43 +0000 [r403013] Richard Mudgett + + * apps/app_directed_pickup.c, CHANGES: PickupChan: Add ability to + specify channel uniqueids as well as channel names. * Made + PickupChan() search by channel uniqueids if the search could not + find a channel by name. * Ensured PickupChan() never considers + the picking channel for pickup. * Made PickupChan() option p use + a common search by name routine. The original search was + erroneously case sensitive. (issue AFS-42) Review: + https://reviewboard.asterisk.org/r/3017/ + +2013-11-21 22:38 +0000 [r402995] Jonathan Rose + + * CHANGES, apps/app_directory.c: app_directory: Set variable + indicating reason directory exited By the time the directory + application exits, a channel variable DIRECTORY_RESULT will be + set for the channel that invoked it which can be used to + determine the reason for exit. The changes log and the + app_directory documentation contain specific details about each + of the possible values for DIRECTORY_RESULT. Review: + https://reviewboard.asterisk.org/r/3016/ + +2013-11-21 22:36 +0000 [r402982-402994] David M. Lee + + * rest-api-templates/ari_resource.c.mustache, /, + rest-api-templates/res_ari_resource.c.mustache: ari: Fix #include + to match generated headers for snakeCase resource files ........ + Merged revisions 402993 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api-templates/make_ari_stubs.py, /: ari: Fix generators for + resources with camelCase names. For the new deviceState resource, + we need to properly generate device_state.[ch] files. ........ + Merged revisions 402981 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 19:22 +0000 [r402969] Matthew Jordan + + * res/res_pjsip_session.c, /: res_pjsip_session: Fix memory leak of + direct media format capabilities The direct media format + capabilities are always allocated in ast_sip_session_alloc and + were not freed in the session destructor. Whoops. (This being the + third whoops caught by Scott and Nitesh's valgrind work for the + Asterisk Test Suite. Nifty!) ........ Merged revisions 402968 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 19:09 +0000 [r402945-402957] Richard Mudgett + + * include/asterisk/app.h, /: voicemail: Fixup some doxygen + comments. ........ Merged revisions 402956 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/bucket.c: bucket: Fix scheme ref leak in + __ast_bucket_scheme_register(). ........ Merged revisions 402944 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 17:53 +0000 [r402942-402943] Matthew Jordan + + * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix use of + uninitialized value in PJSIP In PJMEDIA, + pjmedia_sdp_rtpmap_to_attr will attempt to use the string + rtpmap.param regardless of its length value. Simply setting the + length to 0 does not prevent the garbage on the stack in + rtpmap.param.ptr from being formatted in a sprintf call. This + patch initializes the string to NULL so that at the very least, + something is provided to the function that is predictable. + ........ Merged revisions 402941 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_mwi.c: res_pjsip_mwi: Fix memory leak of MWI + subscriptions container This patch fixes a reference counting + memory leak on the ao2_container created as part of + create_mwi_subscriptions. When we create the container in this + routine, the intent is to hand lifetime ownership over to the + global container unsolicited_mwi. When + ao2_global_obj_replace_unref is called, the reference count on + mwi_subscriptions (the container) will be bumped by 1; however, + the function does not decrement the reference count on + mwi_subscriptions when this occurs. This will prevent the + container from being fully disposed of when Asterisk exits (or on + any subsequent call to this operation, such as during a reload). + ........ Merged revisions 402940 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-21 15:57 +0000 [r402928-402929] David M. Lee + + * res/res_stasis.c, /: stasis: Fixed scoping problem with bridge + tracking. ........ Merged revisions 402817 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_channels.c, res/res_ari_channels.c, + res/ari/resource_channels.h, /, res/stasis/control.c, + include/asterisk/stasis_app.h, rest-api/api-docs/channels.json: + ari: Add silence generator controls This patch adds the ability + to start a silence generator on a channel via ARI. This generator + will play silence on the channel (avoiding audio timeouts on the + peer) until it is stopped, or some other media operation is + started (like playing media, starting music on hold, etc.). + (closes issue ASTERISK-22514) Review: + https://reviewboard.asterisk.org/r/3019/ ........ Merged + revisions 402926 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-19 23:17 +0000 [r402892] Joshua Colp + + * /, res/res_pjsip_caller_id.c: res_pjsip_caller_id: Don't + overwrite user portion of the From header when fromuser is set. + The fromuser option is used to explicitly set the user within the + From header. The res_pjsip_caller_id module did not take this + setting into account when determining if the From header could be + modified or not. (closes issue ASTERISK-22866) Reported by: + Anthony Messina ........ Merged revisions 402891 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-16 13:51 +0000 [r402865] Joshua Colp + + * res/res_pjsip/pjsip_distributor.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac: res_pjsip: Add + support for building against pjproject with SIP transaction group + lock support. SIP transaction group lock support has been + backported into our pjproject. Since the code now internally uses + a group lock the code is now changed to unlock it if present. + Note that the act of finding the transaction is what actually + returns it locked. For further information about group locks + check out the wiki page at: + http://trac.pjsip.org/repos/wiki/Group_Lock (issue + ASTERISK-22818) Reported by: Matt Jordan ........ Merged + revisions 402864 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-15 22:38 +0000 [r402854] Jonathan Rose + + * apps/app_confbridge.c, CHANGES, + apps/confbridge/conf_config_parser.c, + configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h: Confbridge: Add option to + review the recording similar to announce_join_leave Review: + https://reviewboard.asterisk.org/r/3008/ + +2013-11-15 14:37 +0000 [r402839] Kinsey Moore + + * /, main/cel.c: CEL: Fix crash when using CELGenUserEvent This + fixes a crash when CELGenUserEvent is called from the dialplan + while CEL is disabled. Currently, CEL does not create its topics + and forwards if it is not enabled and external entities may + depend on these topics blindly since they should always be + available. This patch breaks up route creation and topic/forward + creation such that the CEL topics and forwards will always exist + while the router and its associated routes will be torn down and + recreated as necessary. (closes issue ASTERISK-22799) Review: + https://reviewboard.asterisk.org/r/3010/ Reported by: Matt Jordan + ........ Merged revisions 402838 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-14 21:36 +0000 [r402820-402829] Richard Mudgett + + * apps/app_directed_pickup.c: Pickup: Pickup() and PickupChan() + parameter parsing improvements. * Made Pickup() and PickupChan() + tollerate empty pickup values. i.e., You can now have + Pickup(&&exten@context). * Made PickupChan() use the standard + option flag parsing code. + + * apps/app_directed_pickup.c: Pickup: Ensure using PICKUPMARK never + considers the picking channel. + +2013-11-14 20:32 +0000 [r402819] Jonathan Rose + + * CHANGES, main/pbx.c, apps/app_sayunixtime.c: Say: If + SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF + Similar to how background works, if a say application is called + with this variable set to 'true', 'yes', 'on', etc. then using + DTMF while the say action is in progress will result in the + channel jumping to that extension in the dialplan. Review: + https://reviewboard.asterisk.org/r/3011/ + +2013-11-13 23:11 +0000 [r402805] Joshua Colp + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /, + res/stasis/control.c, include/asterisk/stasis_app.h: + res_ari_channels: Add the ability to stop locally generated + ringing on a channel. Using the 'ring' operation it is possible + to start locally generated ringback if the channel is answered. + This change adds the ability to stop it by using DELETE. ........ + Merged revisions 402804 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 23:17 +0000 [r402788-402795] Kevin Harwell + + * res/ari/resource_endpoints.c, /: ari endpoints: GET + /ari/endpoints/{invalid-tech} should return a 404 Was returning a + 404 on a valid technology with an empty list of endpoints. Now + checking against the channel tech to make sure the tech itself is + valid and not just an empty list of endpoints. (issue + ASTERISK-22803) Reported by: David M. Lee ........ Merged + revisions 402793 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c, + /, res/res_ari_endpoints.c: ari endpoints: GET + /ari/endpoints/{invalid-tech} should return a 404 Implementation + listing endpoints by technology returned an empty array if no + matching endpoints were found. Fixed so a "404 Not Found" will be + returned instead. (closes issue ASTERISK-22803) Reported by: + David M. Lee ........ Merged revisions 402787 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 19:38 +0000 [r402768-402778] Mark Michelson + + * /, main/channel.c: Switch to a scoped lock to avoid missing + unlocks in failure returns. ........ Merged revisions 402769 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/channel.c, /: Move a NULL check to a place that makes more + sense. Two variables were being checked for NULLity immediately + after being declared NULL. I moved the NULL check until after the + variables are allocated. This allows for the "channelvars" option + in manager.conf to work as intended again. ........ Merged + revisions 402767 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 16:49 +0000 [r402758] Kevin Harwell + + * res/res_pjsip_messaging.c, res/res_pjsip_header_funcs.c, /: + pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer + dereferences Both res_pjsip_messaging and res_pjsip_header_funcs + were causing asterisk to crash because they were trying to + dereference a NULL pointer. In the case of res_pjsip_messaging it + was attempting to "print" a contact header that did not exist. In + fact contact headers should not be part of a SIP MESSAGE, so the + offending code was simply removed. In the case of + res_pjsip_header_funcs a null private channel tech was being + passed to the function and then later dereferenced. Added null + checks (and error logging) to the read/write function handlers to + guard against crashing. (closes issue ASTERISK-22821) Reported + by: Anthony Messina ........ Merged revisions 402757 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 16:34 +0000 [r402756] Kinsey Moore + + * /, apps/app_celgenuserevent.c: CELGenUserEvent: Fix error message + from ast_json_pack This prevents NULL from being passed into an + ast_json_pack call when no extra information is passed to the + application which prevents an error message about NULL arguments + from being generated. ........ Merged revisions 402755 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 15:27 +0000 [r402741] David M. Lee + + * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /: + Fixed a typ. ........ Merged revisions 402738 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-12 15:03 +0000 [r402711] Kinsey Moore + + * channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID + read Asterisk will sometimes core dump during caller id read on + analog channels due to a negative return value from the read() in + my_get_callerid that slips through as a negative length argument + to callerid_feed() if the errno returned by DAHDI is ELAST. This + change ensures that the negative return is treated properly even + when it is ELAST. (closes issue ASTERISK-22746) Reported by: + Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch + uploaded by Michael Walton (License 6502) ........ Merged + revisions 402708 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402709 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402710 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-11 20:28 +0000 [r402698] Jonathan Rose + + * apps/app_confbridge.c: Confbridge: add test events for dynamic + menus test Adds a couple of test events for conference menu + actions so that it's easy to discern when those menu actions have + been triggered. (issue ASTERISK-22760) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2999/ + +2013-11-11 19:31 +0000 [r402688] Mark Michelson + + * apps/app_confbridge.c, /: Get rid of some inaccurate comments. + I'm doing some unrelated work in app_confbridge and finding these + "invalid pin" comments to be annoying. Get out! ........ Merged + revisions 402686 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402687 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-11 15:37 +0000 [r402648] Kinsey Moore + + * /, apps/app_queue.c: app_queue: Honor penalty limits of 0 In the + current app_queue code from 1.8 up to trunk the upper and lower + penalties can be set to 0 but the value is interpreted to be + disabled instead of actually setting limits. This is especially + evident if min and max limits are set to 0 and members with + penalties of 0 and 1 are in the queue since the member with + penalty 1 will still receive calls. This patch adjusts the + special disabled value to be INT_MAX instead of 0. (closes issue + ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/ + Reported by: Schmooze Com ........ Merged revisions 402645 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402646 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402647 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 23:07 +0000 [r402607] Scott Griepentrog + + * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: + keep same local (from) tag for outgoing register requests For + outbound register requests the tag on the From line was updated + every 20 seconds prior to a successful registration and also once + for each registration renewal. That behavior can possibly cause + the registration to be denied because of the different tag, and + is not aligned with the intention of RFC 3261 8.1.3.5 "... + request constitutes a new transaction and SHOULD have the same + value of the Call-ID, To, and From of the previous request...". + This updates chan_sip to have a field to keep the local tag in + the registration structure and use that tag for registration + requests where the callid is also unchanged. (closes issue + ASTERISK-12117) Reported by: Pawel Pierscionek Review: + https://reviewboard.asterisk.org/r/2988/ ........ Merged + revisions 402604 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402605 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402606 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 20:37 +0000 [r402595] Richard Mudgett + + * /, res/res_stasis.c: res_stasis.c: Fix locking issues with the + app_bridge_moh container. * Fix unlinking from the + app_bridges_moh container in remove_bridge_moh() without a lock + under normal circumstances. * Made check + ast_bridge_set_after_callback() return value in + bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK() + locking over too much scope in stasis_app_bridge_moh_channel() + and stasis_app_bridge_moh_stop(). * Fixed unusual usage of + ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge + from off nominal path in stasis_app_bridge_create(). * Fixed + strange construct in stasis_app_unsubscribe(). From a bad merge? + * Made load_module() cleanup on failure. Review: + https://reviewboard.asterisk.org/r/2962/ ........ Merged + revisions 402593 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 19:33 +0000 [r402585] Jonathan Rose + + * /, main/security_events.c, configs/manager.conf.sample, CHANGES, + include/asterisk/manager.h, main/manager.c: security_events: Push + out security events over AMI events Security Events will now be + written to any listener of the new 'security' class Review: + https://reviewboard.asterisk.org/r/2998/ ........ Merged + revisions 402584 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 19:22 +0000 [r402583] Mark Michelson + + * res/res_pjsip.c, /: Clarify an ambiguous error message. ........ + Merged revisions 402582 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 18:53 +0000 [r402571-402572] David M. Lee + + * /, res/res_pjsip/config_system.c: res_pjsip: Print a helpful + error message if sorcery registration fails ........ Merged + revisions 402570 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_playbacks.h, /: Changes from make ari-stubs + after r402560 ........ Merged revisions 402561 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 17:59 +0000 [r402562] Kevin Harwell + + * rest-api/resources.json, res/ari/resource_playback.h (removed), + res/res_ari_playbacks.c (added), res/ari/resource_playbacks.h + (added), /, res/ari.make, rest-api/api-docs/playback.json + (removed), res/ari/resource_playback.c (removed), + res/res_ari_playback.c (removed), + rest-api/api-docs/playbacks.json (added), + res/ari/resource_playbacks.c (added): ARI playback: Rename ARI + Playback to Playbacks Before playback was the only non plural + resource. It has been renamed to playbacks for consistency. + (closes issue ASTERISK-22737) Reported by: Paul Belanger ........ + Merged revisions 402560 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 17:29 +0000 [r402557] David M. Lee + + * res/res_ari.c, main/manager.c, /, main/http.c: ari: Add + application/x-www-form-urlencoded parameter support ARI POST + calls only accept parameters via the URL's query string. While + this works, it's atypical for HTTP API's in general, and + specifically frowned upon with RESTful API's. This patch adds + parsing for application/x-www-form-urlencoded request bodies if + they are sent in with the request. Any variables parsed this way + are prepended to the variable list supplied by the query string. + (closes issue ASTERISK-22743) Review: + https://reviewboard.asterisk.org/r/2986/ ........ Merged + revisions 402555 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-08 14:58 +0000 [r402546] Kevin Harwell + + * apps/app_dahdiras.c, utils/extconf.c, main/asterisk.c: + app_dahdiras: Use waitpid instead of wait4. Several places in the + code were using wait4 while other places were using waitpid. This + change makes all places use waitpid in order to make things more + consistent and since the 'rusage' object passed in/out of wait4 + was never used. (closes issue ASTERISK-22557) Reported by: + YvesGael Patches: asterisk-11.5.1-wait4.patch uploaded by hurdman + (license 6537) + +2013-11-07 23:42 +0000 [r402538] Jonathan Rose + + * res/res_pjsip_authenticator_digest.c, /: PJSIP: Improve error + handling in digest authenticator Previously, regardless of + whether failure to authenticate was due to lacking any + authentication or actually failing authentication, the Digest + Authenticator would simply return that a challenge was still + needed. It will continue to do that when no authentication + information is in the received SIP digest, but when + authentication information is present and does not pass + authentication, that will be treated as an authentication error. + This is to ensure that PJSIP will issue security events indicated + failed auths. ........ Merged revisions 402537 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-07 21:10 +0000 [r402529] David M. Lee + + * res/ari/resource_applications.c, res/ari/resource_playback.c, + rest-api/api-docs/channels.json, res/ari/resource_applications.h, + res/ari/resource_channels.c, res/ari/resource_playback.h, + rest-api/api-docs/recordings.json, res/ari/resource_recordings.c, + rest-api-templates/ari_resource.c.mustache, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, rest-api/api-docs/endpoints.json, + res/ari/resource_endpoints.c, res/ari/resource_recordings.h, + res/ari/resource_events.c, res/res_ari_playback.c, + res/res_ari_applications.c, res/ari/resource_endpoints.h, + res/ari/resource_events.h, rest-api/api-docs/sounds.json, + res/ari/resource_sounds.c, res/res_ari_channels.c, + rest-api/api-docs/bridges.json, res/ari/resource_bridges.c, + res/ari/resource_sounds.h, res/res_ari_recordings.c, + res/ari/resource_bridges.h, rest-api/api-docs/asterisk.json, + res/ari/resource_asterisk.c, res/res_ari_endpoints.c, + rest-api/api-docs/applications.json, + rest-api/api-docs/playback.json, res/res_ari_events.c, + res/ari/resource_asterisk.h, rest-api-templates/swagger_model.py, + res/res_ari_sounds.c, res/res_ari_bridges.c, /, + rest-api-templates/ari_resource.h.mustache, + rest-api-templates/rest_handler.mustache, res/res_ari_asterisk.c, + rest-api-templates/res_ari_resource.c.mustache: ari: User better + nicknames for ARI operations While working on building client + libraries from the Swagger API, I noticed a problem with the + nicknames. channel.deleteChannel() channel.answerChannel() + channel.muteChannel() Etc. We put the object name in the nickname + (since we were generating C code), but it makes OO generators + redundant. This patch makes the nicknames more OO friendly. This + resulted in a lot of name changing within the res_ari_*.so + modules, but not much else. There were a couple of other fixed I + made in the process. * When reversible operations (POST /hold, + POST /unhold) were made more RESTful (POST /hold, DELETE + /unhold), the path for the second operation was left in the API + declaration. This worked, but really the two operations should + have been on the same API. * The POST /unmute operation had still + not been REST-ified. Review: + https://reviewboard.asterisk.org/r/2940/ ........ Merged + revisions 402528 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-06 21:58 +0000 [r402518] Kevin Harwell + + * /, apps/app_queue.c: app_queue: crash if first agent is "busy" If + the first agent/member (via CLI "queue show") in a queue is + "busy" (dnd, circuit busy, etc...) and no agents answered then + app_queue would crash. This occurred because while the calling of + agent(s) remained valid the channel on "busy" agent would be set + to NULL and then later dereferenced upon a second "rna" function + call. The original intention of the code is to have only valid + "call attempt" objects (channels != NULL) checked while + attempting to call agent(s). It does this by building a + "call_next" list of valid "call attempt" objects. In the case of + the "busy" agent subsequent builds of the valid "call attempt" + list would sometimes include (the case mentioned above) an + invalid "call attempt" object. The fix was to make sure the "call + attempt" list was appropriately built on every iteration. A NULL + sanity check was also added at the original offending spot of the + crash just in case another one slipped by somehow. (closes issue + ASTERISK-22644) Reported by: Marco Signorini Review: + https://reviewboard.asterisk.org/r/2983/ ........ Merged + revisions 402517 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-05 21:17 +0000 [r402502-402508] Matthew Jordan + + * /, channels/chan_sip.c: chan_sip: Use AST_AF* defined constant + when calling ast_get_ip While the structure passed to ast_get_ip + should be set memset to 0, thus initializing the ss_family member + to 0, explicitly setting it to AST_AF_UNSPEC is more portable. + ........ Merged revisions 402507 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix incorrect usage of + ast_get_ip involving uninitialized struct This started off as a + fix for the failing IAX2 acl_call test in the Asterisk Test + Suite. When inspecting why that test was failing, it became clear + that all attempts to bind to any local loopback address was + failing: [Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding + IAX2 to address 127.0.0.1:4569 [Nov 2 15:56:28] DEBUG[15787] + netsock2.c: Splitting '127.0.0.1' into... [Nov 2 15:56:28] + DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''. [Nov 2 + 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", + "(null)", ...): ai_family not supported [Nov 2 15:56:28] + WARNING[15787] acl.c: Unable to lookup '127.0.0.1' While there's + conceivably other ways for getaddrino to return EAI_FAMILY, the + most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not + provided as the desired family. The culprit was the call to + ast_get_ip, defined in acl.h. This function uses the family from + the passed in addr object (which it will also populate when it + returns!) when it eventually calls getaddrinfo. This patch fixes + the use of ast_get_ip that were not specifying the family in + chan_iax2. This prevents uninitialized use of the structure, so + that the addresses resolve correctly. Review: + https://reviewboard.asterisk.org/r/2991 ........ Merged revisions + 402505 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/acl.h, /, include/asterisk/netsock2.h: netsock2: + Define AST_AF_* enum constants to their AF_* equivalents This + patch explicitly defines AST_AF_* enum constants to their + sys/socket.h defined equivalents. It is certainly unclear why + these constants actually have to exist, given that netsock2.h + includes sys/socket.h; however, since the code base is already + liberally sprinkled with the usage of AST_AF_* (as well as with + direct calls to AF_*), this will at least keep the semantics + consistent between their usage across systems. ........ Merged + revisions 402503 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_channels.c, /: stasis_channels: Don't give preference + to ANI info in channel snapshots When publishing channel + snapshots, we currently compute the caller ID name and number by + giving preference first to ani.{name|number}, then to + id.{name|number}. However, when a channel driver (such as + chan_sip) updates the caller ID, it typically only updates the + caller ID stored in id.{name|number}. This means that we are + currently giving preference to stale information. When looking at + the rest of the code base, the only other place where we appear + to use this same logic is in app_amd. Everywhere else, we treat + the party information in ani as being separate to the party + information in id. This patch publishes only the caller ID name + and number in the snapshot field for caller_name and caller_num. + Note that the information in ANI is still available in + caller_ani. Review: https://reviewboard.asterisk.org/r/2992/ + ........ Merged revisions 402501 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-04 21:02 +0000 [r402453] Kevin Harwell + + * /, channels/chan_sip.c: chan_sip: notify dialog info ignores + presentation indicator in callerid The presentation indicator in + a callerid (e.g. set by dialplan function + Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog + Info Notifies are generated during extension monitoring. Added a + check to make sure the name and/or number presentations on the + callee (remote identity) are set to allow. If they are restricted + then "anonymous" is used instead. (closes issue AST-1175) + Reported by: Thomas Arimont Review: + https://reviewboard.asterisk.org/r/2976/ ........ Merged + revisions 402450 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402452 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-02 04:30 +0000 [r402406-402439] Richard Mudgett + + * main/stasis.c, main/stasis_message_router.c, /, + include/asterisk/vector.h: vector: Uppercase API to follow C + convention. C does not support templates like C++. ........ + Merged revisions 402438 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/lock.h, main/stasis.c, + main/stasis_message_router.c, /, include/asterisk/vector.h: + vector: Update API to be more flexible. Made the vector macro API + be more like linked lists. 1) Added a name parameter to + ast_vector() to name the vector struct. 2) Made the API take a + pointer to the vector struct instead of the struct itself. 3) + Added an element cleanup macro/function parameter when removing + an element from the vector for ast_vector_remove_cmp_unordered() + and ast_vector_remove_elem_unordered(). 4) Added + ast_vector_get_addr() in case the vector element is not a simple + pointer. * Converted an inline vector usage in + stasis_message_router to use the vector API. It needed the API + improvements so it could be converted. * Fixed topic reference + leak in router_dtor() when the stasis_message_router is + destroyed. * Fixed deadlock potential in stasis_forward_all() and + stasis_forward_cancel(). Locking two topics at the same time + requires deadlock avoidance. * Made internal_stasis_subscribe() + tolerant of a NULL topic. * Made stasis_message_router_add(), + stasis_message_router_add_cache_update(), + stasis_message_router_remove(), and + stasis_message_router_remove_cache_update() tolerant of a NULL + message_type. * Promoted a LOG_DEBUG message to LOG_ERROR as + intended in dispatch_message(). Review: + https://reviewboard.asterisk.org/r/2903/ ........ Merged + revisions 402429 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_state_single.c, + apps/confbridge/conf_state_inactive.c, + apps/confbridge/conf_state_single_marked.c, /, + apps/confbridge/include/confbridge.h, + apps/confbridge/conf_state_multi.c, apps/app_confbridge.c, + apps/confbridge/conf_state_multi_marked.c, + apps/confbridge/conf_state.c: confbridge: Separate user muting + from system muting overrides. The system overrides the user + muting requests when MOH is playing or a waitmarked user is + waiting for a marked user to join. System muting overrides + interfere with what the user may wish the muting to be when the + system override ends. * User muting requests are now independent + of the system muting overrides. The effective muting is now the + logical or of the user request and system override. * Added a + Muted flag to the CLI "confbridge list " command. * + Added a Muted header to the AMI ConfbridgeList action + ConfbridgeList event. (closes issue AST-1102) Reported by: John + Bigelow Review: https://reviewboard.asterisk.org/r/2960/ ........ + Merged revisions 402425 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402427 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, apps/confbridge/conf_config_parser.c, + configs/confbridge.conf.sample, /: config: Allow ConfBridge DTMF + menus to have '#' as the first digit. ConfBridge allows custom + DTMF menus to be created in the confbridge.conf file by assigning + a DTMF key sequence to a sequence of actions as follows: + DTMF-sequence = action,action... Unfortunately, the normal config + file processing code interprets an initial '#' character as + starting a directive such as #include. * Add the ability to + escape the first non-blank character in a config line so the '#' + character can be used without triggering the directive processing + code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported + by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch + (license #5621) patch uploaded by rmudgett (modified) Review: + https://reviewboard.asterisk.org/r/2969/ ........ Merged + revisions 402407 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402416 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/app.h, /, main/app.c: voicemail: Simplify + callback pointer declarations and add doxygen. * Typedefed and + added doxegen for the voicemail callback functions. * Simplified + the prototypes for ast_install_vm_functions() and + ast_install_vm_test_functions() to use the new function typedefs. + * Simplified the voicemail callback function pointer variable + declarations to use the new function typedefs. ........ Merged + revisions 402398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 22:48 +0000 [r402397] Jonathan Rose + + * apps/confbridge/conf_config_parser.c, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + CHANGES: app_confbridge: Make the CONFBRIDGE function be able to + create dynamic menus Also adds the ability to clear all profile + items and makes behavior more consistent with documentation as + when choosing whether to use CONFBRIDGE datastore profiles or the + application arguments to the confbridge application. (closes + issue ASTERISK-22760) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2971/ + +2013-11-01 21:51 +0000 [r402388] Scott Griepentrog + + * main/manager_bridges.c, /, main/bridge.c, + include/asterisk/bridge.h: Manager: Add equivalent AMI actions + for the bridge CLI commands. Adds the following AMI events, + closely following their CLI counterparts: BridgeDestroy + BridgeKick BridgeTechnologyList BridgeTechnologySuspend + BridgeTechnologyUnsuspend BridgeDestroy kicks an entire bridge, + where BridgeKick kicks just one channel off the bridge. When + kicking a channel, specifying the bridge also (optional) insures + it is not removed from the wrong bridge. The BridgeTechnology + events allow viewing and changing suspension status, which + affects only subsequent not active bridging. (closes + ASTERISK-22356) Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/2973/ ........ Merged + revisions 402387 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 16:31 +0000 [r402368] David M. Lee + + * /, rest-api-templates/api.wiki.mustache: ari wiki docs: add notes + about allowMultiple parameters. This patch adds a note to any + parameter that has 'allowMultiple' set in the Swagger + documentation. (closes issue ASTERISK-22704) ........ Merged + revisions 402367 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 14:38 +0000 [r402359] Joshua Colp + + * include/asterisk/stasis_app.h, rest-api/api-docs/channels.json, + res/ari/resource_channels.c, res/res_ari_channels.c, + res/ari/resource_channels.h, res/res_stasis_playback.c, /, + res/stasis/control.c: res_ari_channels: Add ring operation, dtmf + operation, hangup reasons, and tweak early media. The ring + operation sends ringing to the specified channel it is invoked + on. The dtmf operation can be used to send DTMF digits to the + specified channel of a specific length with a wait time in + between. Finally hangup reasons allow you to specify why a + channel is being hung up (busy, congestion). Early media behavior + has also been tweaked slightly. When playing media to a channel + it will no longer automatically answer. If it has not been + answered a progress indication is sent instead. (closes issue + ASTERISK-22701) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2916/ ........ Merged + revisions 402358 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 12:40 +0000 [r402349] Kinsey Moore + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c, + include/asterisk/rtp_engine.h: chan_sip: Fix RTCP port for SRFLX + ICE candidates This corrects one-way audio between Asterisk and + Chrome/jssip as a result of Asterisk inserting the incorrect RTCP + port into RTCP SRFLX ICE candidates. This also exposes an ICE + component enumeration to extract further details from candidates. + (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: + https://reviewboard.asterisk.org/r/2967/ ........ Merged + revisions 402345 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402348 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-11-01 12:33 +0000 [r402337-402347] Joshua Colp + + * /, include/asterisk/stasis_app.h, res/ari/resource_channels.c: + res_ari_channels: Fix a deadlock when originating multiple + channels close to eachother. If a Stasis application is specified + an implicit subscription is done on the originated channel. This + was previously done with the channel lock held which is dangerous + as the underlying code locks the container and iterates items. + This change releases the lock on the originated channel before + subscribing occurs. (closes issue ASTERISK-22768) Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/2979/ + ........ Merged revisions 402346 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c: res_stasis: Ensure the channel is always + departed from the bridge when it leaves. This change adds a + command to the command queue to explicitly depart the channel + from the bridge when it is told it has left. If the channel has + already been departed or has entered a different bridge this + command will become a no-op. (closes issue ASTERISK-22703) + Reported by: John Bigelow (closes issue ASTERISK-22634) Reported + by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2965/ ........ Merged + revisions 402336 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-31 22:09 +0000 [r402328] Mark Michelson + + * /, contrib/scripts/sip_to_pjsip/sip_to_pjsip.py, + contrib/scripts/sip_to_res_sip (removed), + contrib/scripts/sip_to_pjsip (added), + contrib/scripts/sip_to_pjsip/astconfigparser.py, + contrib/scripts/sip_to_pjsip/astdicts.py: Update the conversion + script from sip.conf to pjsip.conf (closes issue ASTERISK-22374) + Reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/2846 ........ Merged revisions + 402327 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-31 16:06 +0000 [r402286-402290] Matthew Jordan + + * main/loader.c, /: core/loader: Don't call dlclose in a while loop + For awhile now, we've noticed continuous integration builds + hanging on CentOS 6 64-bit build agents. After resolving a number + of problems with symbols, strange locks, and other shenanigans, + the problem has persisted. In all cases, gdb shows the Asterisk + process stuck in loader.c on one of the infinite while loops that + calls dlclose repeatedly until success. The documentation of + dlclose states that it returns 0 on success; any other value on + error. It does not state that repeatedly calling it will + eventually clear those errors. Most likely, the repeated calls to + dlclose was to force a close by exhausting the references on the + library; however, that will never succeed if: (a) There is some + fundamental error at work in the loaded library that precludes + unloading it (b) Some other loaded module is referencing a symbol + in the currently loaded module This results in Asterisk sitting + forever. Since we have matching pairs of dlopen/dlclose, this + patch opts to only call dlclose once, and log out as an ERROR if + dlclose fails to return success. If nothing else, this might help + to determine why on the CentOS 6 64-bit build agent things are + not closing successfully. Review: + https://reviewboard.asterisk.org/r/2970 ........ Merged revisions + 402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 402288 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402289 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/media_index.c, /: medix_index: Display errors when library + calls fail Based on feedback from ipengineer in #asterisk, when + the media indexer cannot access a sound file on the system (or + otherwise fails) Asterisk displays a "Cannot frob file" error but + fails to tell you why. This is especially problematic as the + media_indexer failing will rpevent Asterisk from starting, as it + is in the core. We now display the errno error messages so folks + can figure out what they've done wrong. ........ Merged revisions + 402285 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-31 14:45 +0000 [r402277] David M. Lee + + * /, res/stasis/app.c: stasis: add functions embarrassingly missing + from r400522 I neglected to implement two of the endpoint + subscription functions when I did the work. Normally, you'll only + hit that when you unsubscribe from a specific endpoint. ........ + Merged revisions 402276 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-30 17:54 +0000 [r402266] Kevin Harwell + + * channels/chan_pjsip.c, /, res/res_pjsip_messaging.c: + pjsip_messaging: Added debug for in dialog messaging (issue + ASTERISK-22777) Reported by: Matt Jordan ........ Merged + revisions 402265 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 23:43 +0000 [r402227] Rusty Newton + + * /, sounds/Makefile: Updates for 1.4.25 core sounds and 1.4.14 + extra sounds, plus new en_GB language set The new sound packages + relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, + ASTERISK-20782 Modified sounds/Makefile for the new sound + versions and to account for the new en_GB language set. (issue + ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue + ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged + revisions 402224 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402225 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 12:57 +0000 [r402155] Matthew Jordan + + * main/xmldoc.c, main/channel.c, main/pbx.c, /, main/translate.c: + Remove some spammy debug messages; improve clarity of others + Debug messages aren't free. Even when the debug level is + sufficiently low such that the messages are never evaluated, + there is a cost to having to parse Asterisk logs that contain + debug messages that (a) fail to convey sufficient information or + (b) occur so frequently as to be next to meaningless. Based on + having to stare at lots of DEBUG messages, this patch makes the + following changes: * channel.c: When copying variables from a + parent channel to a child channel, specify the channels involved. + Do not log anything for a variable that is not inherited; the + fact that it doesn't have an _ or __ already signifies that it + won't be inherited. * pbx.c: Specify what function evaluation has + occurred that created the result. * translate.c: Bump up the + translator path messages to 10. I've never once had to use these + debug messages, and for each format that is registered (on + startup) and unregistered (on shutdown) the entire f^2 matrix is + logged out. For short tests in the Asterisk Test Suite, this + should make finding the actual test much easier. * xmldoc.c: The + debug message that 'blah' is not found in the tree is expected. + Often, description elements - which are not required - are not + provided. This debug message adds no additional value, as it is + not indicative of an error or helpful in debugging which element + did not contain a 'blah' element as a child. If an element is + supposed to contain a child element, then that XML tree should + have failed validation in the first place. Review: + https://reviewboard.asterisk.org/r/2966/ ........ Merged + revisions 402150 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 402151 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402154 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 12:51 +0000 [r402149-402153] Kinsey Moore + + * rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, res/ari/resource_channels.h, /: ARI: + Remove channels/{channelId}/dial This removes the + /ari/channels/{channelId}/dial URI since it is redundant, overly + complex, is likely to become more externally complex over time, + and is too high-level compared with other ARI operations. See the + following for further information: + http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html + (closes issue ASTERISK-22784) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2968/ ........ Merged + revisions 402152 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * bridges/bridge_native_rtp.c, /: bridge_native_rtp: Ensure bridge + is torn down When a bridge transitions away from one tech to + another, the tech going away is provided a dummy bridge with no + channels in it to tear down. Currently this means that the + teardown code exits prematurely and does not tear anything down. + This change tears down RTP bridging for the channel provided in + the leave bridge tech callback. This also reverts the majority of + r400403 since it is now redundant. (closes issue ASTERISK-22628) + (closes issue ASTERISK-22676) Reported by: John Bigelow Reported + by: Kevin Harwell Tested by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2905/ Patches: + native_rtp_fix.diff uploaded by Kinsey Moore (License 6273) + ........ Merged revisions 402148 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-29 11:15 +0000 [r402140] Joshua Colp + + * /, rest-api/api-docs/playback.json, res/res_ari_playback.c: + res_ari_playback: Add missing 404 error response for GET and + DELETE. (closes issue ASTERISK-22722) Reported by: Richard + Mudgett ........ Merged revisions 402139 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-28 22:10 +0000 [r402128-402130] David M. Lee + + * /, doc: Ignore full docs ........ Merged revisions 402127 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Put back several merge revisions that were lost in r402054 + + * /: Put back several merge revisions that were lost in r401962 + +2013-10-28 15:08 +0000 [r402113-402117] Michael L. Young + + * /, UPGRADE-11.txt, UPGRADE-12.txt: Fix UPGRADE.txt Due To Merging + From Branch 11 When merging in the patch for ASTERISK-22728, the + UPGRADE.txt file was changed incorrectly. That change should have + gone into ASTERISK-11.txt. This commit is to fix that. Also, + another comment in the UPGRADE-11.txt was missing and this commit + adds that as well. ........ Merged revisions 402115 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c, UPGRADE-12.txt: chan_sip: Clarify + 'Forcerport' Setting Displayed When Running "sip show peers" + While looking at ASTERISK-22236, Walter Doekes pointed out that + when running "sip show peers", the setting being displayed can be + confusing. The display of "N" used to mean NAT (i.e. yes). The + NAT setting has gone through many different changes resulting in + the display of different characters to try and convey what the + current setting is for 'Forcerport' (A for Auto and Forcerport is + currently on, a for Auto but Forcerport is off, Y for yes, and N + for no). During the initial code review to try and clarify these + settings (especially since "N" no longer meant what it used to + mean in prior versions of Asterisk), Mark Michelson suggested + using the full space available to display the settings which + helped to make the settings very clear. That was a great + suggestion. Therefore, this patch does the following: * The + column for 'Forcerport' now will show: Auto (Yes), Auto (No), + Yes, or No. * A column for the 'Comedia' setting has been added. + It too will display the setting in a non-cryptic way: Auto (Yes), + Auto (No), Yes, or No. * UPGRADE.txt has been updated to document + this change. (closes issue ASTERISK-22728) Reported by: Walter + Doekes Tested by: Michael L. Young Patches: + asterisk-forcerport-display-clarification_v3.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2941 ........ Merged revisions + 402111 from http://svn.asterisk.org/svn/asterisk/branches/11 + ........ Merged revisions 402112 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-27 23:22 +0000 [r402073-402091] Matthew Jordan + + * main/cdr.c, /: Filter out internal channels from dial message + handling Surrogate channels would pop up from time to time in + dial message handling. This would cause a WARNING message to + appear, indicating that the Surrogate channel had no CDR. This + patch filters out those channels that have the internal + implementation flag set, such that the WARNING message isn't + displayed. ........ Merged revisions 402090 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * cdr/cdr_sqlite3_custom.c, /, cdr/cdr_syslog.c, cdr/cdr_sqlite.c, + cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, + include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, + cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, + cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c: Prevent CDR backends + from unregistering while billing data is in flight This patch + makes it so that CDR backends cannot be unregistered while active + CDR records exist. This helps to prevent billing data from being + lost during restarts and shutdowns. Review: + https://reviewboard.asterisk.org/r/2880/ ........ Merged + revisions 402081 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, contrib/ast-db-manage/config/env.py, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py, + contrib/ast-db-manage/voicemail/env.py: Update Alembic database + scripts for external scripting and PostgreSQL, Oracle This patch + does the following: 1) The env scripts have been updated to be + tolerant of a NULL configuration file. This occurs when + configuration is provided by an external script, such that the + actual config.ini file is not used. 2) Enum types have all been + given names. This is needed for PostgreSQL script generation. 3) + The identifier meetme_confno_starttime_endtime is greater than 30 + characters, and hence invalid for Oracle databases. This has been + truncated down to meetme_confno_start_end. ........ Merged + revisions 400383 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-26 12:56 +0000 [r402065] Joshua Colp + + * channels/chan_pjsip.c, include/asterisk/res_pjsip_session.h, /: + chan_pjsip: Fix a crash when direct media is enabled and an ACK + is received after the channel is hung up. (closes issue + ASTERISK-22731) Reported by: Kinsey Moore ........ Merged + revisions 402064 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-26 00:36 +0000 [r402056] Richard Mudgett + + * res/res_stasis.c, /: res_stasis.c: Made use the ao2_container + callback templates. * Made res_stasis.c use the OBJ_SEARCH_XXX + defines. ........ Merged revisions 402055 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-26 00:27 +0000 [r402054] Scott Griepentrog + + * main/rtp_engine.c, /, include/asterisk/rtp_engine.h: rtp_engine: + fix rtp payloads copy and improve argument names In function + ast_rtp_instance_early _bridge_make_compatible the use of + instance 0/1 as arguments doesn't clearly communicate a direction + that the copying of payloads from the source channel to the + destination channel will occur, making it more probable to have + the arguments to ast_rtp_codecs_payloads_copy() put in the + reverse order. This patch renames the arguments with _dst and + _src suffixes and corrects the copy direction. (closes issue + ASTERISK-21464) Reported by: Kevin Stewart Review: + https://reviewboard.asterisk.org/r/2894/ ........ Merged + revisions 402000 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows + rtpmap:119 being copied per this change, but is not in sip invite + ........ Merged revisions 402042 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 402043 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 23:58 +0000 [r402004-402045] Richard Mudgett + + * /, main/taskprocessor.c: taskprocessor: Made use pthread_equal() + to compare thread ids. * Removed another silly use of RAII_VAR(). + RAII_VAR() and SCOPED_LOCK() are not silver bullets that allow + you to turn off your brain. ........ Merged revisions 402044 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/app.c: You'd think that new files would be free of + whitespace issues. But you would be wrong. ........ Merged + revisions 402003 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 22:01 +0000 [r401999-402002] Jonathan Rose + + * res/ari/resource_bridges.c, res/res_ari_bridges.c, /, + rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c, rest-api/api-docs/bridges.json: ARI: + channel/bridge recording errors when invalid format specified + Asterisk will now issue 422 if recording is requested against + channels or bridges with an unknown format (closes issue + ASTERISK-22626) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2939/ ........ Merged + revisions 402001 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_recording.c, rest-api/api-docs/channels.json, + res/ari/resource_channels.c, res/ari/ari_model_validators.c, + res/res_ari_channels.c, rest-api/api-docs/bridges.json, + rest-api/api-docs/recordings.json, res/ari/resource_bridges.c, + res/ari/ari_model_validators.h, res/res_ari_bridges.c, + rest-api/api-docs/events.json, /: ARI recordings: Issue HTTP + failures for recording requests with file conflicts If a file + already exists in the recordings directory with the same name as + what we would record, issue a 422 instead of relying on the + internal failure and issuing success. (closes issue + ASTERISK-22623) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2922/ ........ Merged + revisions 401973 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 20:51 +0000 [r401962] Scott Griepentrog + + * include/asterisk/pbx.h, main/pbx.c, /: pbx.c: fix confused match + caller id that deleted exten still in hash This fixes a bug where + a zero length callerid match adjacent to a no match callerid + extension entry would be deleted together, which then resulted in + hashtable references to free'd memory. A third state of the + matchcid value has been added to indicate match to any extension + which allows enforcing comparison of matchcid on/off without + errors. (closes issue AST-1235) Reported by: Guenther Kelleter + Review: https://reviewboard.asterisk.org/r/2930/ ........ Merged + revisions 401959 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401960 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401961 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 17:41 +0000 [r401898-401939] Jonathan Rose + + * /, res/res_pjsip/pjsip_distributor.c, + res/res_pjsip_endpoint_identifier_user.c: PJSIP: Add log messages + when requests are received for non-existent endpoints (closes + issue ASTERISK-22552) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2934/ ........ Merged + revisions 401938 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * utils/clicompat.c, utils/refcounter.c, /: Put clicompat-r2.patch + back in We've figured out how to resolve the problems this was + causing in 12/trunk, so this can go back in now. (issue + ASTERISK-22467) Reported by: Corey Farrell Patches: + clicompat-r2.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 401914 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401935 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401936 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, utils/clicompat.c: revert clicompat-r2.patch from r401704 + Patch caused the following build errors against testsuite + https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244 + (issue ASTERISK-22467) Reported by: Corey Farrell ........ Merged + revisions 401895 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401896 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401897 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 16:09 +0000 [r401886] Kevin Harwell + + * /, channels/chan_sip.c: chan_sip: Allow a sip peer to accept both + AVP and AVPF calls Adapts the behaviour of avpf to only impact + the format of outgoing calls. For inbound calls, both AVP and + AVPF calls will be accepted regardless of the value of avpf in + the configuration. (closes issue ASTERISK-22005) Reported by: + Torrey Searle Patches: optional_avpf_trunk.patch uploaded by + tsearle (license 5334) ........ Merged revisions 401884 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401885 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-25 13:49 +0000 [r401873] David M. Lee + + * tests/test_json.c, /: test_json: Fix deprecation warnings After a + series of upgrades over recent weeks, I've discovered that + test_json.c won't compile in dev mode any more for me. One of + gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate + tempnam. Which, in general, is a good thing. But for test code + that just needs a temporary file, it's just annoying. This patch + replaces usage of tempname with mkstemp, avoiding the deprecation + warning. It also removes the temporary files when the test is + complete, which apparently we weren't doing before (oops). + Review: https://reviewboard.asterisk.org/r/2957/ ........ Merged + revisions 401872 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-24 21:06 +0000 [r401836] Kevin Harwell + + * /, main/logger.c: Logging: Logging types ignored after specifying + a verbose level If one specified a verbose level within a logging + facility in logger.conf then any component after it was ignored. + Fixed so all values are correctly read. (closes issue + ASTERISK-22456) Reported by: Kevin Harwell ........ Merged + revisions 401833 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401835 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-24 20:48 +0000 [r401834] David M. Lee + + * rest-api-templates/models.wiki.mustache, + rest-api/api-docs/events.json, /, + rest-api-templates/swagger_model.py, + rest-api-templates/ari_model_validators.c.mustache: The Swagger + 1.2 specification for type extension ended up being slightly + different than my proposal. Instead of putting an 'extends' field + on the subtype, the base type has a 'subTypes' field, which is a + list of the subTypes. Given that its a messaging model and not an + object model, kinda makes sense. This patch changes the + events.json api-doc, and the python translators to take the new + format into account. Other changes that are in Swagger 1.2 were + not adopted, since the spec is still in flux, and could change + before it's finalized. A summary of changes to the Swagger-1.2 + spec can be found at + https://github.com/wordnik/swagger-core/wiki/1.2-transition. + (closes issue ASTERISK-22440) Review: + https://reviewboard.asterisk.org/r/2909/ ........ Merged + revisions 401701 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-24 20:34 +0000 [r401622-401832] Jonathan Rose + + * /, main/utils.c: utils: Fix memory leaks and missed + unregistration of CLI commands on shutdown Final set of patches + in a series of memory leak/cleanup patches by Corey Farrell + (closes issue ASTERISK-22467) Reported by: Corey Farrell Patches: + main-utils-1.8.patch uploaded by coreyfarrell (license 5909) + main-utils-11.patch uploaded by coreyfarrell (license 5909) + main-utils-12up.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 401829 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401830 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401831 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_linkedlists.c: test_linkedlists: Fix memory leak + (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + test_linkedlists-1.8.patch uploaded by coreyfarrell (license + 5909) test_linkedlists-11up.patch uploaded by coreyfarrell + (license 5909) ........ Merged revisions 401790 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401791 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401792 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer + reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + jitterbuf-jb_reset-leak-1.8.patch + jitterbuf-jb_reset-leak-11up.patch ........ Merged revisions + 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 401787 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401788 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/astobj2.c, /: astobj2: Unregister debug CLI commands at exit + (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell + (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 401781 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401783 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401784 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_voicemail.c, /: app_voicemail: Memory Leaks against + tests (issue ASTERISK-22467) Reported by: Corey Farrell Patches: + app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909) + app_voicemail-11up.patch uploaded by coreyfarrell (license 5909) + ........ Merged revisions 401743 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401744 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401745 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/app.c, main/asterisk.c, utils/clicompat.c, + channels/chan_dahdi.c, codecs/ilbc/doCPLC.c, main/data.c, /: + memory leaks: Memory leak cleanup patch by Corey Farrell (second + set) Also covers ast_app_parse_timelen-fail-zero-length.patch, + but the patch was replaced with one of my own. (issue + ASTERISK-22467) Reported by: Corey Farrell Patches: + chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license + 5909) clicompat-r2.patch uploaded by coreyfarrell (license 5909) + codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909) + data-cleanup-test-registration.patch uploaded by coreyfarrell + (license 5909) main-asterisk-kill-listener.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 401704 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401705 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401706 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, tests/test_dlinklists.c, funcs/func_math.c, + channels/sip/reqresp_parser.c, main/test.c, + main/editline/readline.c: memory leaks: Memory leak cleanup patch + by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by: + Corey Farrell Patches: + chan_sip-parse_contact_header_test-free-contacts.patch uploaded + by coreyfarrell (license 5909) cli-filename-completion-leak.patch + uploaded by coreyfarrell (license 5909) func_math.patch uploaded + by corefarrell (license 5909) main-test-cleanup.patch uploaded by + coreyfarrell (license 5909) test_dlinklists.patch uploaded by + coreyfarrell (license 5909) ........ Merged revisions 401660 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401661 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401662 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk: + Address jittery DTMF events in RTP streams (closes issue + ASTERISK-21170) Reported by: NITESH BANSAL Patches: + dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418) + Review: https://reviewboard.asterisk.org/r/2938/ ........ Merged + revisions 401619 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401620 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401621 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 16:52 +0000 [r401582] Richard Mudgett + + * /, cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a + filter when the CDR value is empty. Extra CDR records are written + if a filtered CDR value is empty because the filter is not + checked. (closes issue ASTERISK-22272) Reported by: Jordi Llull + Chavarria ........ Merged revisions 401577 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401579 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401581 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 16:48 +0000 [r401580] John Bigelow + + * /, main/bridge_channel.c: Add a test suite event to indicate when + the atxfer 3-way feature is detected This adds a test suite event + that indicates to tests when the attended transfer three-way call + feature is detected. Review: + https://reviewboard.asterisk.org/r/2912/ ........ Merged + revisions 401578 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 15:23 +0000 [r401540] Kinsey Moore + + * channels/chan_mgcp.c, /: chan_mgcp: Properly handle malformed + media lines This corrects a situation in which a media line was + not parsed properly and resulted in a crash. (closes issue + ASTERISK-21190) Reported by: adomjan Patches: + chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448) + ........ Merged revisions 401537 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401538 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401539 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 11:16 +0000 [r401500] Joshua Colp + + * /, channels/chan_sip.c: chan_sip: Fix an issue where an + incompatible audio format may be added to SDP. If preferred + codecs included any non-audio format the code would mistakenly + add the audio format, even if it was not a joint capability with + the remote side. (closes issue ASTERISK-21131) Reported by: + nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by + nbougues (license 6470) ........ Merged revisions 401497 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401498 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401499 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-23 02:36 +0000 [r401489] Michael L. Young + + * channels/chan_iax2.c, configs/iax.conf.sample, /: chan_iax2: Fix + Binding To Multiple Addresses Again When reworking chan_iax2 for + IPv6, the ability to bind to multiple addresses was removed by + mistake. This patch restores this functionality and adds notes + about IPv6 addresses in the sample config. (closes issue + ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L. + Young Patches: asterisk-22741-fix-binding-multiple-addr.diff + uploaded by Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2945/ ........ Merged + revisions 401488 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-22 23:10 +0000 [r401450] Matthew Jordan + + * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP + is not available during SSRC change In r400089, a patch was put + in to correct erroneous RTCP statistic resets. Unfortunately, + ast_rtp_read can be called on an RTP instance that does not have + RTCP information. This patch prevents that crash by only + resetting the statistics if we do actually have an RTCP instance. + (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John + Bigelow ........ Merged revisions 401445 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401446 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401447 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-22 19:04 +0000 [r401421-401435] Richard Mudgett + + * apps/app_queue.c, /: app_queue: Fix CLI "queue remove member" + queue_log entry. The queue_log entry resulting from CLI "queue + remove member" when log_membername_as_agent is enabled is wrong. + It always uses the interface name instead of the member name in + the queue_log entry. * Get the queue member before removing it + from the queue so the member name is available for the queue_log + entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve + Patches: fix_membername.diff (license #6505) patch uploaded by + Oscar Esteve (modified to fix potential ref leak) ........ Merged + revisions 401433 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401434 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/bridge_channel.c, + include/asterisk/bridge_channel_internal.h, /, main/bridge.c: + Bridging: Fix orphaned bridge if neither of the joining channels + can join. The original issue noted that the bridge is orphaned + when res_parking.so is not loaded and a call uses the dial kK + flags. A similar issue happens when only one of the park flags is + used. In this case you have the bridge with one or the other + channel left in it. The channel and bridge will stay around until + the channel hangs up. * Fixed the initial bridge channel push + failure to act as if the channel were kicked out of the bridge. + The bridge then decides if it needs to be dissolved. (closes + issue ASTERISK-22629) Reported by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2928/ ........ Merged + revisions 401424 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/parking/parking_bridge_features.c, + res/parking/parking_bridge.c, /: res_parking: Give parking + timeout comebacktoorigin channel DTMF features. Parking timeouts + did not set any DTMF features for the channel calling the parker + back. * Added code to set the parkedcalltransfers, + parkedcallreparking, parkedcallhangup, and parkedcallrecording + options appropriately for the channels when a parking timeout + occurs. The recall channel DTMF options are set using the + BRIDGE_FEATURES channel variable to allow the other timeout + options to have the DTMF features available. (closes issue + ASTERISK-22630) Reported by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2942/ ........ Merged + revisions 401422 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_parking.c: res_parking: Update XML documention for + DTMF features after parking timeout. * Updated the XML + documentation to indicate that the parkedcalltransfers, + parkedcallreparking, parkedcallhangup, and parkedcallrecording + configuration options also apply to parking timeouts. (issue + ASTERISK-22630) Reported by: Kevin Harwell Review: + https://reviewboard.asterisk.org/r/2942/ ........ Merged + revisions 401420 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-22 15:17 +0000 [r401411] Joshua Colp + + * apps/app_dial.c: Add an 'R' option to Dial which sends ringing + until early media has been received. (closes issue + ASTERISK-10487) Reported by: Gaspar Zoltan Patches: 10487.patch + uploaded by n8ideas (license 6075) + +2013-10-21 21:06 +0000 [r401365] Mark Michelson + + * /, main/bridge_channel.c: Remove a noisy debug message from + bridging code. This particular debug message, during a stress + test, was logged so often that it appeared that there may be a + memory leak in the logger code. In actuality, there was no memory + leak, but the logger thread was having a hard time keeping up + with the demands of the rest of the system. Since this debug + message has no value at all, the best way to fix the problem was + to just remove the message. (closes issue AST-1225) reported by + John Bigelow Patches: spammy_log.diff uploaded by Mark Michelson + (License #5049) ........ Merged revisions 401364 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-21 19:50 +0000 [r401328] Kevin Harwell + + * /, main/editline/term.c: Segfault in LIBEDIT_INTERNAL after + tgetstr(), when libncurses5-dev isn't installed Include the + appropriate declarations when not using termcap, but term+curses + and [n]curses do not exist. (closes issue ASTERISK-22351) + Reported by: A. Iglesias Patches: + issueA22351_libedit_internal_without_ncurses_dev.patch uploaded + by wdoekes (license 5674) ........ Merged revisions 401325 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401326 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401327 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-21 18:59 +0000 [r401316-401317] David M. Lee + + * rest-api/api-docs/channels.json, /: Fixing r401281; the model + name is Channel, with a capital C ........ Merged revisions + 401315 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_ari.c, /: Fixed malformed Access-Control-Allow-Methods + header. Was causing Safari to barf on POST and DELETE. ........ + Merged revisions 401106 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-19 21:57 +0000 [r401292] Kinsey Moore + + * /, channels/chan_iax2.c: Fix IAX2 incoming call address lookups + This fixes address lookup for incoming calls without a peer + definition. The address family was unset instead of being set to + AST_AF_UNSPEC which was causing lookup failures on "127.0.0.1". + This is one of the causes of the current failure of the app_page + integration test. Review: + https://reviewboard.asterisk.org/r/2933/ ........ Merged + revisions 401291 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-19 14:45 +0000 [r401282] Joshua Colp + + * res/ari/resource_channels.h, main/pbx.c, /, + rest-api/api-docs/channels.json, res/ari/resource_channels.c, + res/res_ari_channels.c: Return a channel snapshot when + originating using ARI, and subscribe the Stasis application to + it. This change allows a user of ARI to know what channel it has + originated and also follow any progress. If a Stasis application + is provided it will be automatically subscribed to the originated + channel immediately. (closes issue ASTERISK-22485) Reported by: + David Lee Review: https://reviewboard.asterisk.org/r/2910/ + ........ Merged revisions 401281 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 22:52 +0000 [r401272] Richard Mudgett + + * /, res/parking/parking_controller.c: res_parking: Remove setting + useless flag. ........ Merged revisions 401271 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 21:51 +0000 [r401263] David M. Lee + + * contrib/scripts/get_swagger_ui.sh (added), Makefile, /, + static-http: This is just a quick script for dumping swagger-ui + into static-http, so that it can be served by the Asterisk web + server. I had to change the Makefile in order to recursively + install content from the static-http directory, hence the code + review instead of just putting it in. Review: + https://reviewboard.asterisk.org/r/2924/ ........ Merged + revisions 401261 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 18:44 +0000 [r401249] Mark Michelson + + * main/sorcery.c, main/cli.c, main/manager.c, /, main/bridge.c, + main/bucket.c: Resolve some memory leaks due to incorrect for + loop / ao2 ref usage. A common idiom in Asterisk is to due + something like: for (ao2_obj = list_beginning; ao2_obj = + next_item; ao2_ref(ao2_obj, -1)) { ...do stuff... } This is nice + because it automatically takes care of the object references for + you. However, there is a pitfall here. If a break statement is in + the for loop, then the current reference is not cleaned up. In + some cases, this is on purpose, but in others there is a leak. + This commit fixes the leak cases. ........ Merged revisions + 401248 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 16:59 +0000 [r401233-401240] Richard Mudgett + + * /, res/res_fax.c, include/asterisk/channel.h, apps/app_dial.c, + main/channel.c: Add channel lock protection around translation + path setup. Most callers of ast_channel_make_compatible() happen + before the channels enter a two party bridge. With the new + bridging framework, two party bridging technologies may also call + ast_channel_make_compatible() when there is more than one thread + involved with the two channels. * Added channel lock protection + in set_format() and ast_channel_make_compatible_helper() when + dealing with the channel's native formats while setting up a + translation path. * Fixed best_src_fmt and best_dst_fmt usage + consistency in ast_channel_make_compatible_helper(). The call to + ast_translator_best_choice() got them backwards. * Updated some + callers of ast_channel_make_compatible() and the function + documentation. There is actually a difference between the two + channels passed in. * Fixed the deadlock potential in res_fax.c + dealing with ast_channel_make_compatible(). The deadlock + potential was already there anyway because res_fax called + ast_channel_make_compatible() with chan locked. (closes issue + ASTERISK-22542) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2915/ ........ Merged + revisions 401239 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, include/asterisk/bridge.h: Tweak ast_bridge_depart() doxygen. + ........ Merged revisions 401232 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 16:06 +0000 [r401216-401224] Mark Michelson + + * include/asterisk/bridge.h, /: Remove the bit about requiring + ast_bridge_depart() to be called before ast_bridge_destroy(). + ........ Merged revisions 401223 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/bridge.h, /: Clarify in ast_bridge_destroy() + about how departable channels must be handled. ........ Merged + revisions 401212 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 15:14 +0000 [r401184] Michael L. Young + + * /, channels/chan_sip.c: Remove Port Restriction When Checking For + NAT When trying to determine if a peer is behind NAT, we should + not be using the ports when comparing addresses. This patch + removes the port from being checked and just useds the addresses + now. (closes issue ASTERISK-22729) Reported by: Michael L. Young + Tested by: Michael L. Young Patches: + asterisk-remove-using-port-for-nat-check.diff uploaded by Michael + L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2927/ ........ Merged + revisions 401182 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401183 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-18 14:50 +0000 [r401181] Walter Doekes + + * main/channel.c, /: Properly copy/remove the device state cache + flag over a masquerade. In r378303 the + AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the + devstate system to not cache states for non-real devices. + However, when optimizing away channels (ast_do_masquerade), that + flag wasn't copied. In my case, using Local devices as queue + members created a situation where the endpoint was considered in + use, but the state change of the device being available again was + ignored (not cached). The endpoint channel was optimized into the + (previously) Local channel, but kept the do-not-cache flag. The + end result being that the queue member apparently stayed in use + forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes + Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged + revisions 401178 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401179 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401180 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-17 20:39 +0000 [r401169] Michael L. Young + + * /, channels/chan_sip.c: Fix Setting A chan_sip Dialog's + SIP_NAT_FORCE_RPORT Flag A condition was added in a commit to fix + ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was + set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the + dialog. This condition should not have been there since it + assumed that if Asterisk is in an environment where NAT is + involved, that the auto_* nat settings or force_rport setting + would be on in the global settings. If the nat setting in the + global setting is set to 'nat=no' and then turned on for peers + (which is not quite the recommended way, although it is allowed) + this flag is never copied to the dialog resulting in problems + like, REGISTER replies going to the wrong port. This patch + removes this conditional check and will now always use the peer's + flag which by this point in the code the checks on whether the + peer is behind NAT or not (if using auto_force_rport) have + already been run. (closes issue ASTERISK-22236) Reported by: + Filip Frank Tested by: Michael L. Young Patches: + asterisk-2236-always-set-rport.diff uploaded by Michael L. Young + (license 5026) Review: https://reviewboard.asterisk.org/r/2919/ + ........ Merged revisions 401167 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401168 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-17 18:25 +0000 [r401159] Jonathan Rose + + * res/res_parking.c, /: res_parking: Fix bug where reloading + immediately wipes new parkpos extensions (closes issue + ASTERISK-22631) Reported by: Kevin Harwell ........ Merged + revisions 401158 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-17 15:41 +0000 [r401122] Kinsey Moore + + * /, res/res_xmpp.c, res/res_jabber.c: Reduce log level of a + non-pubsub error message Drop an error log message to debug level + 1 since distributed device state functions correctly when + receiving this message and it spams the logs. (closes issue + ASTERISK-22410) Reported by: abelbeck Patches: + asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch + uploaded by abelbeck (License 5903) + asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded + by abelbeck (License 5903) ........ Merged revisions 401119 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401120 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401121 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 21:22 +0000 [r401108] Richard Mudgett + + * /, res/ari/resource_playback.c: ARI: Fix crash when POST + /playback/{id}/control does not have an operation parameter. + (closes issue ASTERISK-22680) Reported by: John Bigelow ........ + Merged revisions 401107 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 17:01 +0000 [r401097] David M. Lee + + * rest-api/resources.json, /: Oops. Leftover /stasis reference + ........ Merged revisions 401096 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 14:02 +0000 [r401088] Kinsey Moore + + * rest-api/api-docs/bridges.json, res/ari/resource_channels.h, /, + res/ari/resource_bridges.h, rest-api/api-docs/channels.json: + Clarify documentation for channel and bridge list This makes it + clear that the ARI API calls for listing channels and bridges + will list all channels or bridges in the system and not just + those that are in or are controlled by a Stasis application. + (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........ + Merged revisions 401087 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 12:19 +0000 [r401079] Walter Doekes + + * /, apps/app_queue.c: Don't check all realtime queues when doing + "queue show some_queue". When using realtime queues, queues have + to be fetched from the database every now and then to see if any + info has been changed or to see if the queue has been removed. + When fetching info for an individual queue, the pruning of other + queues is unnecessarily costly. Review: + https://reviewboard.asterisk.org/r/2907/ ........ Merged + revisions 401049 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 401076 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401077 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-16 00:12 +0000 [r401041] Paul Belanger + + * /, rest-api/api-docs/bridges.json, res/res_ari_bridges.c: Use + POST / DELETE to toggle ARI bridge moh Review: + https://reviewboard.asterisk.org/r/2911/ ........ Merged + revisions 401040 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 23:44 +0000 [r401020-401039] Richard Mudgett + + * main/translate.c: translate.c: Some minor code tweaks. * + Consistently compare format2index() return value so matrix_get() + cannot get passed negative values. * Optimize + ast_translator_best_choice() to defer initializing things until + needed. Also cached the matrix_get() return value rather than + repeatedly calling it. + + * /, channels/dahdi/bridge_native_dahdi.c: bridge_native_dahdi: + Return channel join failure if could not make the channels + compatible. ........ Merged revisions 401030 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_iax2.c: chan_iax2: Fix channel left locked in + off nominal code path. ........ Merged revisions 401016 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 401017 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 20:03 +0000 [r401019] Kinsey Moore + + * rest-api/api-docs/bridges.json, res/res_ari_bridges.c, /: Ensure + bridge record error responses validate This adds the list of + expected errors to the /bridges/{bridgeId}/record ARI + documentation so that outbound 4xx errors validate properly. + Previously, this would result in a response validation failure. + (closes issue ASTERISK-22627) Reported by: Joshua Colp ........ + Merged revisions 401018 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 15:30 +0000 [r401007] Paul Belanger + + * rest-api/api-docs/channels.json, res/res_ari_channels.c, /: Use + POST / DELETE to toggle hold / moh for ARI channels This change + updates how we handle toggle events, rather then create two + different function names, we'll just use POST / DELETE from HTTP + to handle it. Review: https://reviewboard.asterisk.org/r/2906/ + ........ Merged revisions 400999 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 15:26 +0000 [r400998] Mark Michelson + + * /, channels/chan_sip.c: Prevent chan_sip from sending duplicate + BYEs. When a 200 OK for an initial INVITE is received, we were + doing the right thing by ACKing and sending an immediate BYE. + However, we also were doing the wrong thing and queuing an answer + frame, thus causing the call to be answered. This would cause the + call to be hung up by the channel thread, thus resulting in a + second BYE being sent out. In this fix, I also have set the + hangupcause to be correct since the initial BYE being sent by + Asterisk had an unknown hangup cause. I have changed to using + "Bearer capabilty not available" since the call was hung up due + to an SDP offer/answer error. (closes issue ASTERISK-22621) + reported by Kinsey Moore ........ Merged revisions 400970 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400971 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400984 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-15 13:44 +0000 [r400959] David M. Lee + + * /, rest-api-templates/asterisk_processor.py: My doc correction in + r400842 had a silly bug. Because I added a wiki_description to + models and not their properties, the rendered wiki page had the + model description instead of the property descriptions, which + looks very silly indeed. (closes issue ASTERISK-22705) ........ + Merged revisions 400958 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 22:52 +0000 [r400913-400950] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/chan_dahdi.h: chan_dahdi: Add config support for hwgain + settings. * Add hwtxgain and hwrxgain config options to + chan_dahdi.conf with documentation in chan_dahdi.conf.sample. + (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches: + jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch + uploaded by rmudgett + + * channels/chan_dahdi.c, /, channels/chan_dahdi.h: chan_dahdi: + Reflect the set software gain in the CLI "dahdi show channel" + output. * Remember the swgain setting from CLI "dahdi set swgain" + command so the CLI "dahdi show channel" output will reflect the + current setting. * Updated CLI "dahdi set hwgain" and "dahdi set + swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco + Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621) + patch uploaded by rmudgett ........ Merged revisions 400907 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400909 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400911 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 22:03 +0000 [r400912] Mark Michelson + + * /, channels/chan_sip.c: chan_sip: Do not increment the SDP + version between 183 and 200 responses. Bumping the SDP version + number can cause interoperability problems since receivers of the + responses will expect that a 200 SDP will be identical to a + previous 183 SDP. (closes issue ASTERISK-21204) reported by + NITESH BANSAL Patches: + dont-increment-session-version-in-2xx-after-183.patch uploaded by + NITESH BANSAL (License #6418) ........ Merged revisions 400906 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 400908 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400910 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 15:54 +0000 [r400891] Kevin Harwell + + * /, res/res_pjsip_outbound_registration.c: pjsip outbound + registration: Log message says received a 408 when we didn't If + the server didn't exist that we are trying to register to the log + message would say that a 408 was received from that server when + in reality one wasn't. Added log messages stating no response was + received if the response does not exist. (closes issue + ASTERISK-22554) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2893/ ........ Merged + revisions 400890 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-14 15:01 +0000 [r400882] Matthew Jordan + + * res/res_pjsip_mwi.c, /: Remove duplicate module info block The + module info block was repeated twice. Once is sufficient. + ........ Merged revisions 400881 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-13 15:42 +0000 [r400873] Joshua Colp + + * res/res_pjsip_session.c, /: Fix a race condition in + res_pjsip_session with rapidly terminating the session. The + INVITE session state callback wrongly assumes that a session will + always exist, but when rapidly terminating the session this + assumption goes out the window. As all handler code for the + INVITE session state callback requires the session it will now + just exit immediately if no session exists. (closes issue + ASTERISK-22668) Reported by: John Bigelow ........ Merged + revisions 400872 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-12 16:53 +0000 [r400864] Kinsey Moore + + * /, res/res_pjsip_outbound_authenticator_digest.c: Fix realm + comparison for outbound auth When generating the list of + authentication credentials to pass to PJSIP, Asterisk was using + the raw pointer of a pj_str_t which is not always + NULL-terminated. This sometimes resulted in incorrect text for + the realm and a failure to match the realm for authentication + purposes which was causing the outbound nominal auth pjsip basic + call test to bounce. This now uses the pj_str_t that contains the + realm instead of generating a new one. Thanks to John Bigelow for + helping to narrow this down. ........ Merged revisions 400863 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-11 17:05 +0000 [r400855] Richard Mudgett + + * include/asterisk/channel.h, /: channel.h: whitespace changes. + ........ Merged revisions 400854 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-11 16:36 +0000 [r400851-400852] David M. Lee + + * /, res/ari/resource_bridges.h, rest-api/api-docs/playback.json, + rest-api-templates/api.wiki.mustache, res/res_ari_playback.c, + rest-api/api-docs/channels.json, res/ari/resource_playback.h, + rest-api/api-docs/bridges.json, + rest-api-templates/asterisk_processor.py, + res/ari/resource_channels.h, + rest-api-templates/models.wiki.mustache: Multiple revisions + 400508,400842-400843,400848 ........ r400508 | dlee | 2013-10-03 + 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line Corrected response + class for stopPlayback ........ r400842 | dlee | 2013-10-10 + 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line Correct some ARI wiki + rendering errors ........ r400843 | dlee | 2013-10-10 14:26:19 + -0500 (Thu, 10 Oct 2013) | 1 line Updated /play resource docs. + The playback of http: resources isn't implemented... yet ........ + r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 + lines Fix a stupid copy/paste error in ARI docs. Patches: + ari-doc-patch.txt uploaded by jbigelow (license 5091) ........ + Merged revisions 400508,400842-400843,400848 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Fixed merge tracking for r400360, which was somehow lost + +2013-10-11 16:28 +0000 [r400850] Richard Mudgett + + * bridges/bridge_softmix.c, /: Softmix: Fix crash when switching + from softmix to another bridge technology. The crash is caused by + a race condition when switching between native RTP and softmix + bridging technologies. In this situation, the bridging technology + is switched from native RTP to softmix, and then back to native + RTP fast enough that the softmix private data gets destroyed + before the softmix mixing thread gets started. Thanks to Kinsey + Moore for the crash analysis. * Fix race condition when starting + the softmix mixing thread and switching to another bridge + technology. (closes issue ASTERISK-22678) Reported by: John + Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621) + patch uploaded by rmudgett Tested by: John Bigelow ........ + Merged revisions 400849 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-10 18:21 +0000 [r400825-400834] Joshua Colp + + * /, res/res_pjsip/location.c: Perform validation of permanent + contacts on AORs in res_pjsip. ........ Merged revisions 400833 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c: Fix an + assertion in res_pjsip when specifying an invalid outbound proxy. + This change fixes two issues when setting an outbound proxy: 1. + The outbound proxy URI was not parsed and validated during + configuration. 2. If an outgoing dialog was created and the + outbound proxy could not be set an assertion would occur because + the usage count on the dialog was not decremented. The + documentation has also been updated to specify that a full URI + must be specified for the outbound proxy. (closes issue + ASTERISK-22672) Reported by: Antti Yrjola ........ Merged + revisions 400824 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-09 11:02 +0000 [r400772-400813] Matthew Jordan + + * res/res_pjsip_header_funcs.c, /: Use 'z' as the format specifier + for size_t Using 'lu' will produce a compiler warning for some + versions of gcc and on some architectures. 'z' should be portable + as a format specifier for size_t. ........ Merged revisions + 400812 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_header_funcs.c (added), /: Add PJSIP_HEADER + function for manipulation of SIP headers in the PJSIP stack This + patch adds support to the PJSIP stack in Asterisk for SIP header + manipulation. Note that this is analagous to + SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming + supplemental session callback is registered that takes the + pjsip_hdrs from the incoming session and stores them in a linked + list in the session datastore. Calls to PJSIP_HEADER traverse + over the list and return the nth matching header where 'n' is the + 'number' argument to the function. When adding a header, the + first call creates a datastore and linked list and adds the + datastore to the session. The header is then created as a + pjsip_hdr and added to the list. An outgoing supplemental session + callback then traverses the list and adds the headers to the + outgoing pjsip_msg. When removing a header, the list created with + PJSIP_HEADER(add,...) is traversed and all matching entries are + removed. (closes issue ASTERISK-22498) Reported by: George Joseph + patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph + (License 6322) ........ Merged revisions 400771 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 22:33 +0000 [r400770] Kinsey Moore + + * /, configure, configure.ac: Add warning when compiling with iODBC + support When running configure, libiodbc2 development headers + will fulfill the requirement for ODBC development headers, but + will not function properly. This adds a warning when libiodbc2 + development headers are detected instead of unixodbc development + headers. (closes issue ASTERISK-22459) Reported by: Patrick + Maille Tested by: Walter Doekes Patches: + issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes + (License 5674) ........ Merged revisions 400767 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400768 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400769 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 21:20 +0000 [r400759] Richard Mudgett + + * apps/app_agent_pool.c, /: app_agent_pool: Fix AMI/CLI AgentLogoff + soft preventing agents from logging back in. * Clear the + deferred_logoff flag when an agent logs in. (closes issue + ASTERISK-22669) Reported by: John Bigelow ........ Merged + revisions 400754 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 20:52 +0000 [r400750] Mark Michelson + + * /, res/res_pjsip.c, res/res_pjsip/config_transport.c: Switch from + using pjsip_strerror to pj_strerror. pjsip_strerror is only aware + of PJSIP-specific error codes. pj_strerror() is aware of all + PJProject error codes and OS-specific error codes. This + specifically fixes an oft-seen error in transport configuration + code where EADDRINUSE would result in "Unknown PJSIP error + 120098" instead of a useful message. ........ Merged revisions + 400749 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 20:18 +0000 [r400728-400744] Richard Mudgett + + * configs/confbridge.conf.sample, /, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + CHANGES, apps/confbridge/conf_config_parser.c: app_confbridge: + Can now set the language used for announcements to the + conference. ConfBridge now has the ability to set the language of + announcements to the conference. The language can be set on a + bridge profile in confbridge.conf or by the dialplan function + CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) + Reported by: Jonathan White Patches: M19983_rev2.diff (license + #5138) patch uploaded by junky (modified) Tested by: rmudgett + ........ Merged revisions 400741 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400742 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix + duplicate default_user profile. * Fixed looking in the wrong + profiles container to see if the default_user profile is already + created in verify_default_profiles(). The bridge profile + container is never going to hold user profiles. :) ........ + Merged revisions 400723 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400724 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 18:19 +0000 [r400684-400704] Kinsey Moore + + * funcs/func_config.c, /: Fix func_config list entry allocation The + AST_CONFIG dialplan function defined in func_config.c allocates + its config file list entries using ast_malloc. List entry + allocations destined for use with Asterisk's linked list API must + be ast_calloc()d or otherwise initialized so that list pointers + are set to NULL. These uses of ast_malloc have been replaced by + ast_calloc to prevent dereferencing of uninitialized pointer + values when traversing the list. (closes issue ASTERISK-22483) + Reported by: Brian Scott ........ Merged revisions 400694 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400697 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400701 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: Fix STUN crash when using IPv6 any + address Ensure that when chan_sip binds to the IPv6 any address + ([::]), IPv4 candidates are also added. (closes issue + ASTERISK-21917) Reported by: Torrey Searle Patches: + 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License + 5334) ........ Merged revisions 400681 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400682 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 15:44 +0000 [r400683] Mark Michelson + + * res/res_pjsip/pjsip_options.c, /: Push CLI qualify into the + threadpool. If you run Asterisk in the background and then + connect to it through a separate console, the thread that runs + CLI commands is not registered with PJLIB. Thus PJLIB does not + like it when you attempt to send OPTIONS requests from that + thread. So now we push the task into the threadpool, which we + know to be registered with PJLIB. Thanks to Antti Yrjola for + reporting this. ........ Merged revisions 400680 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-08 15:12 +0000 [r400662-400672] Richard Mudgett + + * /, res/res_agi.c, apps/app_queue.c: Make app_queue and res_agi + independent of AMI being enabled. The + https://reviewboard.asterisk.org/r/2888/ review changes manager + to not subscribe to stasis when it is disabled for performance + reasons. When manager is disabled app_queue and res_agi decline + to load and fail to clean up what they have already allocated. * + Made app_queue and res_agi clean up allocated resources when they + decline to load. * Made app_queue and res_agi use their own + subscriptions to the stasis topics instead of borrowing manager's + message router structure inappropriately. (closes issue + ASTERISK-22604) Reported by: rmudgett Review: + https://reviewboard.asterisk.org/r/2902/ ........ Merged + revisions 400671 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, include/asterisk/stasis.h, apps/app_queue.c, + include/asterisk/manager.h: Miscellaneous stand alone comment + cleanups. ........ Merged revisions 400661 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-06 17:13 +0000 [r400625] Michael L. Young + + * /, apps/app_queue.c: app_queue: Fix Queuelog EXITWITHKEY only + logging two of four fields Commit r62462 added two extra fields + for logging "the original position the caller entered the queue + at, and the amount of time the caller was waiting in the queue." + But when r75969 was merged from 1.4 into trunk (r75977), these + two fields disappeared. Those two extra fields were not logged in + 1.4 and when the patch was merged, those fields went away. + Therefore, this is a regression and was caught by the reporter + because he was reading the awesome "Asterisk: The Definitive + Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M. + Tested by: Dalius M. Patches: + asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2901/ ........ Merged + revisions 400622 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400623 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-05 00:59 +0000 [r400593] Richard Mudgett + + * /, channels/iax2/include/parser.h: chan_iax2: Fix compile error. + ........ Merged revisions 400588 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 21:41 +0000 [r400568] Michael L. Young + + * main/acl.c, include/asterisk/netsock2.h, CHANGES, + channels/chan_iax2.c, channels/iax2/parser.c, main/netsock.c, + main/netsock2.c, /, channels/iax2/include/parser.h: Add IPv6 + Support To chan_iax2 This patch adds IPv6 support to chan_iax2. + Yay! (closes issue ASTERISK-22025) Patches: + iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) + Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged + revisions 400567 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 19:32 +0000 [r400553] David M. Lee + + * rest-api/api-docs/applications.json (added), /: Added missing + file from r400522 ........ Merged revisions 400552 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 19:11 +0000 [r400533-400543] Jonathan Rose + + * res/res_pjsip_logger.c, /: chan_pjsip: Make logger togglable + without loading/unloading This patch makes the res_pjsip_logger + do a few things... First, it will be built and installed by + default now, so end users won't need to enable it in menuselect. + Second, while it is loaded, it no longer will immediately issue + log messages. Upon loading, it is in the disabled state and must + be turned on with the new CLI command. The CLI command 'pjsip set + logger has been added and can be used to do the + following: pjsip set logger on: Enables logger for all PJSIP + traffic pjsip set logger off: Disables logger for all PJSIP + traffic pjsip set logger host : Enables logger for the + specific host Review: https://reviewboard.asterisk.org/r/2900/ + ........ Merged revisions 400542 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, + contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py + (added), configs/extconfig.conf.sample, + configs/sorcery.conf.sample, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py: + chan_pjsip: Add alembic scripts for generating db tables for + PJSIP Also updates sample configurations for sorcery and + extconfig to demonstrate how to use databases created by that + alembic script. (closes issue ASTERISK-22133) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........ + Merged revisions 400532 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 16:01 +0000 [r400523] Matthew Jordan + + * res/res_stasis.c, main/asterisk.c, + rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json, + res/stasis/app.c, /, + rest-api-templates/ari_model_validators.h.mustache, + include/asterisk/endpoints.h, res/res_ari_applications.c (added), + res/ari/resource_endpoints.h, include/asterisk/stasis_app.h, + res/stasis/app.h, rest-api/resources.json, + include/asterisk/_private.h, res/ari/ari_model_validators.c, + main/endpoints.c, res/ari/ari_model_validators.h, main/json.c, + res/res_ari_model.c, res/ari.make, + res/ari/resource_applications.c (added), + res/ari/resource_applications.h (added): ARI: Add subscription + support This patch adds an /applications API to ARI, allowing + explicit management of Stasis applications. * GET /applications - + list current applications * GET /applications/{applicationName} - + get details of a specific application * POST + /applications/{applicationName}/subscription - explicitly + subscribe to a channel, bridge or endpoint * DELETE + /applications/{applicationName}/subscription - explicitly + unsubscribe from a channel, bridge or endpoint Subscriptions work + by a reference counting mechanism: if you subscript to an event + source X number of times, you must unsubscribe X number of times + to stop receiveing events for that event source. Review: + https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) + Reported by: Matt Jordan ........ Merged revisions 400522 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-04 15:49 +0000 [r400511-400521] Joshua Colp + + * /, res/res_pjsip.c: Enclose the To URI and update its user + portion if a request user has been specified. ........ Merged + revisions 400520 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: Replace the connection address at the + SDP level if altering the SDP with the external media address. + ........ Merged revisions 400510 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 23:20 +0000 [r400482] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Don't ignore expires value in + contact header if it lacks semicolon (closes issue + ASTERISK-22574) Reported by: Filip Jenicek Patches: + chan_sip_expires.patch uploaded by Filip Jenicek (license 6277) + ........ Merged revisions 400469 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400470 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400471 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 21:46 +0000 [r400461] Matthew Jordan + + * /, main/channel_internal_api.c: Remove publication of a channel + snapshot when the technology is set This patch removes said + publication for a few reasons: (1) It is unnecessary. Association + of the channel technology with a specific channel is an + implementation detail that should be assumed to "just happen", + and consumers of Stasis don't need to be informed about it. (2) + Publication of said message can now cause crashes, as the actual + creation of a channel in normal locations now stages its + messages. As a result, things that create dummy channels (such as + the SIP RTP QOS unit test) and associate them with a channel + technology were now crashing, as the channel itself was not known + by Stasis. ........ Merged revisions 400460 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 20:22 +0000 [r400452] Mark Michelson + + * bridges/bridge_native_rtp.c, /, + include/asterisk/bridge_technology.h: Fix assumption in + bridge_native_rtp.c regarding number of participants in a bridge. + When a party leaves a bridge, there may be more participants in + the bridge than expected. As such, it is important not to make + assumptions regarding the list of channels in a bridge. This + change makes it so that when a party leaves a native RTP bridge, + we unbridge it and the party it was bridged with. Previously, the + first and last channels in the list were unbridged since it was + assumed that these were the two channels that had been bridged. + As previously stated, a new party had been inserted into the + bridge, so this logic did not work properly. (closes issue + ASTERISK-22615) reported by Matt Jordan Review: + https://reviewboard.asterisk.org/r/2899 ........ Merged revisions + 400403 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 19:32 +0000 [r400443] Joshua Colp + + * /, main/cdr.c: When serializing CDR variables (like for "core + show channels") don't output an error if CDRs aren't enabled. + ........ Merged revisions 400442 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 19:30 +0000 [r400441] Kinsey Moore + + * /, main/security_events.c: Fix security events for AMI invalid + password In r337595, additional security events were added for + chan_sip authentication failures. The new IEs added to the + existing invalid password event were defined as required IEs, but + existing users of the event did not set the new IEs and could not + since they didn't apply to existing uses. They are now marked as + optional IEs. (closes issue ASTERISK-22578) Reported by: Matt + Jordan ........ Merged revisions 400421 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400440 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 19:06 +0000 [r400402] Joshua Colp + + * res/ari/resource_channels.c, /: Fix a crash caused by muting and + unmuting a channel in ARI without specifying a direction. (closes + issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by + Matt Jordan, whose office I have taken over in the name of + Canada. ........ Merged revisions 400401 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 18:51 +0000 [r400399] Richard Mudgett + + * /, main/cel.c: cel: Some whitespace cleanups ........ Merged + revisions 400398 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 18:32 +0000 [r400385-400397] Kinsey Moore + + * res/res_rtp_multicast.c, /: res_rtp_multicast: Ensure SSRC is set + properly This fixes a bug where the SSRC field on multicast RTP + can be stuck at 0 which can cause problems for endpoints trying + to make sense of incoming streams. (closes issue ASTERISK-22567) + Reported by: Simone Camporeale Patches: + 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale + (License 6536) ........ Merged revisions 400393 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400394 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400395 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/xml.c: Detect and use xsltCleanupGlobals when available This + introduces usage of an additional libxslt cleanup function, + xsltCleanupGlobals, when the configure script detects that it is + available. Early versions of the library did not include this + function. (closes issue ASTERISK-22570) Reported by: Corey + Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey + Farrell (License 5909) ........ Merged revisions 400384 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 16:28 +0000 [r400374] Richard Mudgett + + * channels/chan_vpb.cc, /: chan_vpb: Make compile again. ........ + Merged revisions 400373 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 14:59 +0000 [r400363-400364] Mark Michelson + + * tests/test_cel.c, /: Get rid of uses of stasis_topic_wait() + ........ Merged revisions 400362 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * pbx/pbx_spool.c, main/manager.c, main/format_cap.c, + channels/chan_skinny.c, res/res_agi.c, channels/chan_motif.c, + channels/chan_alsa.c, apps/app_confbridge.c, + addons/chan_mobile.c, channels/chan_mgcp.c, + res/res_clioriginate.c, channels/chan_bridge_media.c, + channels/chan_sip.c, tests/test_format_api.c, + res/res_pjsip_sdp_rtp.c, bridges/bridge_simple.c, + apps/app_originate.c, res/parking/parking_applications.c, + main/core_local.c, channels/chan_console.c, channels/chan_oss.c, + include/asterisk/format_cap.h, res/res_pjsip_session.c, + res/ari/resource_bridges.c, channels/chan_jingle.c, + channels/chan_misdn.c, channels/dahdi/bridge_native_dahdi.c, + res/res_pjsip/pjsip_configuration.c, main/file.c, + channels/chan_h323.c, channels/chan_nbs.c, + bridges/bridge_native_rtp.c, tests/test_config.c, + res/res_stasis.c, channels/chan_pjsip.c, channels/chan_unistim.c, + channels/chan_multicast_rtp.c, addons/chan_ooh323.c, + main/rtp_engine.c, /, main/ccss.c, apps/app_meetme.c, + bridges/bridge_holding.c, main/bridge_basic.c, + bridges/bridge_softmix.c, channels/chan_gtalk.c, + channels/chan_iax2.c, main/media_index.c, main/channel.c, + channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c: Cache + string values of formats on ast_format_cap() to save processing. + Channel snapshots have string representations of the channel's + native formats. Prior to this change, the format strings were + re-created on ever channel snapshot creation. Since channel + native formats rarely change, this was very wasteful. Now, string + representations of formats may optionally be stored on the + ast_format_cap for cases where string representations may be + requested frequently. When formats are altered, the string cache + is marked as invalid. When strings are requested, the cache + validity is checked. If the cache is valid, then the cached + strings are copied. If the cache is invalid, then the string + cache is rebuilt and copied, and the cache is marked as being + valid again. Review: https://reviewboard.asterisk.org/r/2879 + ........ Merged revisions 400356 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-03 14:52 +0000 [r400361] Joshua Colp + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c, /: Fix crashes in + res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and + external_media_address is set. The callback function for changing + the media address in streams wrongly assumes that a connection + line will always be present. This is false as no line is present + if a stream has been rejected. (closes issue ASTERISK-22645) + Reported by: Rusty Newton ........ Merged revisions 400360 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 22:22 +0000 [r400335] Mark Michelson + + * main/stasis_wait.c (removed), res/ari/resource_endpoints.c, /, + include/asterisk/stasis.h, tests/test_cel.c, + include/asterisk/stasis_endpoints.h, channels/chan_pjsip.c, + main/stasis.c, main/stasis_endpoints.c: Multiple revisions + 400318-400319 ........ r400318 | mmichelson | 2013-10-02 17:08:49 + -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from + stasis. Since caches are updated on publisher threads, there is + no need to wait for the cache updates to occur after a stasis + message is published. In the case of chan_pjsip device state + changes, this set of changes caused an improvement to + performance. Review: https://reviewboard.asterisk.org/r/2890 + ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, + 02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........ + Merged revisions 400318-400319 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 21:33 +0000 [r400317] Michael L. Young + + * channels/chan_iax2.c, /: Cast Integer Argument To Unsigned Char + The member reg in the peercnt structure is an unsigned char and + peercnt_modify() is expecting an unsigned char argument which + gets assigned to peercnt->reg. This patch fixes that by casting + the integer argument being passed to peercnt_modify to unsigned + char. ........ Merged revisions 400314 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400315 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400316 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 21:26 +0000 [r400313] Matthew Jordan + + * main/cdr.c, main/manager.c, /, main/cel.c: Only create Stasis + subscriptions when enabled Subscribing to Stasis isn't free. As + such, this patch makes AMI, CDR, and CEL - the "big 3" - only + subscribe when enabled. Toggling their availability via a .conf + file will unsubscribe/subscribe as appropriate. Review: + https://reviewboard.asterisk.org/r/2888/ ........ Merged + revisions 400312 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 20:31 +0000 [r400304] Richard Mudgett + + * main/pbx.c, /: Originate: Make setting caller id on outgoing call + use either name or number. Previous code was requiring both name + and number to be available. Also restored a comment block on why + caller id is also set on an outgoing call leg in addition to + connected line from earlier versions of Asterisk. ........ Merged + revisions 400303 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 19:20 +0000 [r400295] Kinsey Moore + + * /, rest-api/api-docs/asterisk.json: Correct allowable values for + ARI general information filter ........ Merged revisions 400291 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 19:17 +0000 [r400287] Matthew Jordan + + * main/cdr.c, /: Fix the CDR CLI command 'cdr show active + {channel}' When the switch from channel names to channel unique + IDs happened, the poor CLI command got left in the dust. This + fixes the command so that users can once again see how Asterisk + is messing up your billing information. ........ Merged revisions + 400286 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 18:44 +0000 [r400285] Joshua Colp + + * /, res/res_pjsip_t38.c: Fix a crash in res_pjsip_t38 caused by + the wrong assumption that a session will always have a channel. + When starting up or shutting down this assumption is false. + ........ Merged revisions 400284 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 18:28 +0000 [r400282] Tzafrir Cohen + + * Makefile, doc/astdb2sqlite3.8 (added), /, doc/astdb2bdb.8 + (added): man pages for astdb2bdb and astdb2sqlite3 Review: + https://reviewboard.asterisk.org/r/2898/ ........ Merged + revisions 400279 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400281 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 17:12 +0000 [r400269-400271] Richard Mudgett + + * apps/app_stack.c, res/stasis_recording/stored.c, main/json.c, + main/stasis_cache.c, res/res_ari.c, /, main/utils.c: + MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is + enabled. * There were several places in ARI where an external + library was mallocing memory that must always be released with + free(). When MALLOC_DEBUG is enabled, free() is redirected to the + MALLOC_DEBUG version. Since the external library call still uses + the normal malloc(), MALLOC_DEBUG complains that the freed memory + block is not registered and will not free it. These cases must + use ast_std_free(). * Changed calls to asprintf() and vasprintf() + to the equivalent ast_asprintf() and ast_vasprintf() versions + respectively. ........ Merged revisions 400270 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sig_ss7.c, /: sig_ss7: Fix compiler warnings. ........ + Merged revisions 400268 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-02 16:23 +0000 [r400246-400266] Joshua Colp + + * channels/chan_alsa.c, main/stasis_channels.c, channels/sig_ss7.c, + channels/chan_pjsip.c, channels/chan_mgcp.c, + channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, /, + channels/chan_sip.c, main/bridge.c, include/asterisk/channel.h, + channels/chan_gtalk.c, channels/chan_console.c, + channels/sig_pri.c, channels/chan_iax2.c, channels/chan_jingle.c, + main/channel.c, channels/chan_dahdi.c, main/dial.c, + include/asterisk/stasis_channels.h, channels/chan_skinny.c, + channels/chan_motif.c: Reduce channel snapshot creation and + publishing by up to 50%. This change introduces the ability to + stage channel snapshot creation and publishing by suppressing the + implicit creation and publishing that some functions have. Once + all operations are executed the staging is marked as done and a + single snapshot is created and published. Review: + https://reviewboard.asterisk.org/r/2889/ ........ Merged + revisions 400265 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: Fix a random one way audio issue in + PJSIP. Due to the asynchronous design of the PJMEDIA SDP + negotiator it was possible for the SDP to be negotiated *after* a + channel was created and after it was being wait on by an + application. It is only after negotiation occurs that the file + descriptors for RTP are placed on the channel. Since the channel + was already being waited on these file descriptors were not + monitored, causing incoming media to never be read. This change + wakes up any application waiting on the channel so that added + file descriptors end up being monitored. (closes issue AST-1227) + Reported by: John Bigelow ........ Merged revisions 400256 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/stasis/control.c, include/asterisk/stasis_app.h, + res/ari/resource_channels.c: Allow specifying a channel to dial + an extension and context in an ARI dial operation. (issue + ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged + revisions 400254 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip_session.c: Retrieve and store the hostname only + once so multiple threads do not potentially initialize it at the + same time. ........ Merged revisions 400245 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-01 21:19 +0000 [r400228-400237] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: chan_dahdi: Fix + analog parking using flash-hook. Transferring an analog call + using a flash-hook to parking would fail to park the call and + result in an invalid ao2 object unref. * Park the correct bridged + channel. ........ Merged revisions 400236 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/features_config.c, /: Features: Rearm the parking config + options have moved warning for each reload. ........ Merged + revisions 400227 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-10-01 15:54 +0000 [r400218] Matthew Jordan + + * main/cdr.c, /: Filter out internal channels for bridge leave + messages and parked call messages Granted, if you manage to park + a Conference announcer channel, something has gone horrifically + wrong. ........ Merged revisions 400217 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 21:40 +0000 [r400206] Jonathan Rose + + * configs/features.conf.sample, /, configs/res_parking.conf.sample: + configuration samples: Pull all parking related stuff out of + features.conf This patch also adds documentation for parking from + features.conf to res_parking.conf ........ Merged revisions + 400205 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 19:58 +0000 [r400195-400197] Matthew Jordan + + * /, funcs/func_cdr.c: Parse arguments passed to the CDR_PROP + function correctly I can only blame this on a bad merge, because + this in no way worked properly the way it was written. Mea culpa. + The function should now parse its arguments correctly and + function properly. (Note that the API used by the CDR_PROP + function has working unit tests... this was merely bad coding of + the actual registered function) (closes issue ASTERISK-22613) + Reported by: Private Name ........ Merged revisions 400196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Remove spurious event raised when CDRs are + reloaded The Reload event is now raised by the module loading + core. As such, the Reload event in the CDR engine was a duplicate + and not needed. ........ Merged revisions 400194 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 18:55 +0000 [r400186] David M. Lee + + * tests/test_devicestate.c, include/asterisk/sem.h (added), + tests/test_taskprocessor.c, res/res_pjsip_mwi.c, + res/res_pjsip/include/res_pjsip_private.h, tests/test_stasis.c, + res/parking/parking_manager.c, res/res_security_log.c, + channels/chan_mgcp.c, main/stasis_cache_pattern.c, main/pbx.c, + include/asterisk/vector.h (added), /, main/ccss.c, + apps/app_meetme.c, include/asterisk/taskprocessor.h, + configs/stasis.conf.sample (removed), configure.ac, + res/parking/parking_applications.c, channels/sig_pri.c, + apps/app_queue.c, main/cel.c, main/stasis.c, + channels/chan_dahdi.c, funcs/func_presencestate.c, + main/stasis_message_router.c, configure, + apps/confbridge/confbridge_manager.c, res/res_agi.c, + main/manager_system.c, res/res_stasis_test.c, main/sem.c (added), + main/manager_channels.c, res/res_pjsip_refer.c, + main/manager_mwi.c, apps/app_voicemail.c, main/stasis_cache.c, + main/stasis_wait.c, main/stasis_config.c (removed), + include/asterisk/stasis_internal.h, res/stasis/app.c, + channels/chan_sip.c, include/asterisk/autoconfig.h.in, + main/manager_endpoints.c, main/channel_internal_api.c, + include/asterisk/stasis.h, main/devicestate.c, + main/taskprocessor.c, res/res_xmpp.c, main/sounds_index.c, + include/asterisk/stasis_message_router.h, channels/chan_iax2.c, + res/res_jabber.c, main/endpoints.c, main/astobj2.c, + res/res_chan_stats.c, res/parking/parking_bridge_features.c, + tests/test_stasis_endpoints.c, main/cdr.c, main/channel.c, + main/manager_bridges.c, main/manager.c, channels/chan_skinny.c: + Multiple revisions 399887,400138,400178,400180-400181 ........ + r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 + line Minor performance bump by not allocate manager variable + struct if we don't need it ........ r400138 | dlee | 2013-09-30 + 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance + improvements This patch addresses several performance problems + that were found in the initial performance testing of Asterisk + 12. The Stasis dispatch object was allocated as an AO2 object, + even though it has a very confined lifecycle. This was replaced + with a straight ast_malloc(). The Stasis message router was + spending an inordinate amount of time searching hash tables. In + this case, most of our routers had 6 or fewer routes in them to + begin with. This was replaced with an array that's searched + linearly for the route. We more heavily rely on AO2 objects in + Asterisk 12, and the memset() in ao2_ref() actually became + noticeable on the profile. This was #ifdef'ed to only run when + AO2_DEBUG was enabled. After being misled by an erroneous comment + in taskprocessor.c during profiling, the wrong comment was + removed. Review: https://reviewboard.asterisk.org/r/2873/ + ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep + 2013) | 24 lines Taskprocessor optimization; switch Stasis to use + taskprocessors This patch optimizes taskprocessor to use a + semaphore for signaling, which the OS can do a better job at + managing contention and waiting that we can with a mutex and + condition. The taskprocessor execution was also slightly + optimized to reduce the number of locks taken. The only + observable difference in the taskprocessor implementation is that + when the final reference to the taskprocessor goes away, it will + execute all tasks to completion instead of discarding the + unexecuted tasks. For systems where unnamed semaphores are not + supported, a really simple semaphore implementation is provided. + (Which gives identical performance as the original taskprocessor + implementation). The way we ended up implementing Stasis caused + the threadpool to be a burden instead of a boost to performance. + This was switched to just use taskprocessors directly for + subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ + ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep + 2013) | 28 lines Optimize how Stasis forwards are dispatched This + patch optimizes how forwards are dispatched in Stasis. + Originally, forwards were dispatched as subscriptions that are + invoked on the publishing thread. This did not account for the + vast number of forwards we would end up having in the system, and + the amount of work it would take to walk though the forward + subscriptions. This patch modifies Stasis so that rather than + walking the tree of forwards on every dispatch, when forwards and + subscriptions are changed, the subscriber list for every topic in + the tree is changed. This has a couple of benefits. First, this + reduces the workload of dispatching messages. It also reduces + contention when dispatching to different topics that happen to + forward to the same aggregation topic (as happens with all of the + channel, bridge and endpoint topics). Since forwards are no + longer subscriptions, the bulk of this patch is simply changing + stasis_subscription objects to stasis_forward objects (which, + admittedly, I should have done in the first place.) Since this + required me to yet again put in a growing array, I finally + abstracted that out into a set of ast_vector macros in + asterisk/vector.h. Review: + https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee + | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove + dispatch object allocation from Stasis publishing While looking + for areas for performance improvement, I realized that an unused + feature in Stasis was negatively impacting performance. When a + message is sent to a subscriber, a dispatch object is allocated + for the dispatch, containing the topic the message was published + to, the subscriber the message is being sent to, and the message + itself. The topic is actually unused by any subscriber in + Asterisk today. And the subscriber is associated with the + taskprocessor the message is being dispatched to. First, this + patch removes the unused topic parameter from Stasis subscription + callbacks. Second, this patch introduces the concept of + taskprocessor local data, data that may be set on a taskprocessor + and provided along with the data pointer when a task is pushed + using the ast_taskprocessor_push_local() call. This allows the + task to have both data specific to that taskprocessor, in + addition to data specific to that invocation. With those two + changes, the dispatch object can be removed completely, and the + message is simply refcounted and sent directly to the + taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ + ........ Merged revisions 399887,400138,400178,400180-400181 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-30 15:57 +0000 [r400142] Kinsey Moore + + * /, channels/chan_sip.c, configs/pjsip.conf.sample, + res/res_pjsip_outbound_registration.c, configs/sip.conf.sample, + CHANGES: chan_sip: Allow Asterisk to retry after 403 on register + This adds a global option in chan_sip to allow it to continue + attempting registration if a 403 is received, clearing the cached + nonce and treating it as a non-fatal response. Normally, this + would cause registration attempts to that endpoint to stop. This + also adds a similar per-outbound-registration option to + chan_pjsip which allows the retry interval to be altered for 403 + responses to REGISTER requests. (closes issue ASTERISK-17138) + Review: https://reviewboard.asterisk.org/r/2874/ Reported by: + Rudi ........ Merged revisions 400137 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400140 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400141 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-28 22:57 +0000 [r400059-400122] Matthew Jordan + + * /, res/res_pjsip_notify.c, configs/pjsip_notify.conf.sample + (added): res_pjsip_notify: Add documentation We forgot to add + documentation for res_pjsip_notify, which would prevent it from + being loaded. Whoops. This patch also updates res_pjsip_notify to + use pjsip_notify.conf, which now has its own sample file in the + configs directory as well. Review: + https://reviewboard.asterisk.org/r/2835/ ........ Merged + revisions 400121 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Correct erroneous + lost packet information in RTCP reports RTCP's calculation of the + number of lost packets in an RTP stream is based on that stream's + sequence number count, the number of received packets, and how + many packets we expect to receive. When the SSRC for an RTP + stream changes, there can - and almost always will be - a large + jump in the next packet's timestamp and sequence number. If we + don't reset the number of received packets, sequence number + count, and other metrics used by RTCP, the next RR/SR report will + use the previous SSRC's values to calculate the lost packet count + for the new SSRC - resulting in a very large number of lost + packets. This patch modifies res_rtp_asterisk such that, if it + detects a SSRC change, it will reset the various values used by + the RTCP calculations. From the perspective of RTCP, this appears + as a new media stream - which is what it is. Review: + https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) + Reported by: Thomas Arimont ........ Merged revisions 400089 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400093 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400108 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, configure, configure.ac: Add check for openSUSE when detecting + bfd library In ASTERISK-17842, some additional library checks + were added to the configure script so that the bfd library could + be found on CentOS and Fedora systems. As it turns out, openSUSE + requires an additional library. This patch adds another check to + the configure script for openSUSE that will add that library. + Review: https://reviewboard.asterisk.org/r/2885/ (closes issue + AST-1169) Reported by: Guenther Kelleter ........ Merged + revisions 400073 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400075 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400077 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: CDR: Improve handling of parking; resolve + assertion when originating into park This patch covers two + problems: 1) Currently, when a call is transferred into a parking + lot from a bridge (using either the blind transfer or one touch + parking mechanisms), the application fails to be set to "Park" in + the resulting CDR record for the parked channel. This is due to + the ParkedCall message arriving before the BridgeEnter for the + channel entering the parking bridge. The ParkedCall message isn't + handled as the CDR for the channel has already been finalized + (due to the channel having left its two party bridge), and the + BridgeEnter - which creates the new CDR - doesn't have the + parking information. This patch modifies the behavior so that + reception of a ParkedCall message will - if not handled by a CDR + chain - cause a new CDR to be created and put into the Parking + state. 2) It fixes a FRACK that occurred when a channel is + originated into a parking space. The DialedPending state - which + occurs for both Dialed and Originated channels - assumed that it + couldn't handle the parking transitions due to it having a Party + B; however, Originated channels don't have a Party B. As such, + the existing CDR needs to transition into the parking state - + this patch does that. Review: + https://reviewboard.asterisk.org/r/2877/ (closes issue + ASTERISK-22482) Reported by: Richard Mudgett ........ Merged + revisions 400062 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_queue.c: app_queue: Make manager events tolerant of + Local channel shenanigans app_queue currently attempts to handle + Local channel optimizations in an effort to provide accurate + information in Stasis messages (and their corresponding AMI + events) as well as the Queue log. Sometimes, however, things + don't go as planned. Consider the following scenario: SIP/foo <-> + L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local + channel optimization. app_queue will normally do the following: * + Listen for the Local optimization events and update our agent + accordingly to SIP/agent in the queue log and messages * When we + get a hangup, publish the AgentComplete event based on our + information (SIP/foo and SIP/agent) However, as with all things + that depend on sanity from something as capricious as Local + channels, things can go wrong: (1) SIP/agent immediately hangs up + upon answering. This triggers a race condition between + termination messages coming from SIP/agent and the ongoing Local + channel optimization messages. (Note that this can also occur + with SIP/foo) (2) In a race condition, Asterisk can (rarely) + deliver the hangup messages prior to the Local channel + optimization. In that case, the messages *may* arrive to + app_queue in the following order: * Hangup SIP/Agent * Hangup + SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When + app_queue receives the hangup of the agent or the caller, it will + attempt to publish the AgentComplete event. However, it now has a + problem - it thinks its agent is the ;1 side of the Local + channel, as it never received the optimization event. At the same + time, that channel is already gone. This results in getting NULL + from the Stasis cache. What's more, we can't really wait for the + optimization message, as we are currently handling the hangup of + the channel that the optimization event would tell us to use. + This patch modifies the behavior in app_queue such that, since we + still have a lot of pertinent queue information (interface, queue + name, etc.), we now raise the event with what information we + know. The channels involved now may or may not be present. Users + will still at least get the "AgentComplete" event, which + "completes" the known Agent information. Review: + https://reviewboard.asterisk.org/r/2878/ (closes issue + ASTERISK-22507) Reported by: Richard Mudgett ........ Merged + revisions 400060 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/manager.c, /: manager: Fix crash when appending a manager + channel variable In r399887, a minor performance improvement was + introduced by not allocating the manager variable struct if it + wasn't used. Unfortunately, when directly accessing an + ast_channel struct, manager assumed that the struct was always + allocated. Since this was no longer the case, things got a bit + crashy. This fixes that problem by simply bypassing appending + variables if the manager channel variable struct isn't there. + ........ Merged revisions 400058 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 21:58 +0000 [r400016-400021] Richard Mudgett + + * apps/app_cdr.c, res/res_parking.c, /: app_cdr and res_parking: + Fix some resource leaks. * app_cdr left the ResetCDR application + registered. * res_parking leaked a ref to config global. (closes + issue ASTERISK-22566) Reported by: Corey Farrell Patches: + ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey + Farrell ........ Merged revisions 400020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/sip/reqresp_parser.c, /, channels/chan_sip.c: chan_sip: + Increase some scratch buffer sizes dealing with caller id. * + Eliminated an unnecessary initialization in check_user_full(). + (closes issue ASTERISK-22477) Reported by: Michael Shepelev + ........ Merged revisions 400013 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 400014 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 400015 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 19:18 +0000 [r400000] Sean Bright + + * configs/sip.conf.sample: Remove some trailing whitespace and + steal revision 400000. + +2013-09-27 18:28 +0000 [r399991] Kevin Harwell + + * /, res/res_pjsip.c, res/res_pjsip_session.c, + include/asterisk/res_pjsip.h, res/res_pjsip.exports.in: + res_pjsip: crash when using localnet and + external_signaling_address options There was a collision of + mod_data use on the transaction between using a nat hook and an + session response callback. During state change it was assumed + what was in the mod_data was nothing or the response callback. + However, it was possible for it to also contain a nat hook thus + resulting in a bad cast and a crash. Added the ability to store + multiple data elements in mod_data via a hash table. In this + instance, mod_data now stores a hash table of the two values that + can be retrieved using an associated string key. (closes issue + ASTERISK-22394) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2843/ ........ Merged + revisions 399990 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 17:46 +0000 [r399978] Jonathan Rose + + * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip: + Reject calls on 200 OKs if no SDP has been received When Asterisk + receives a 200 OK in response to an invite, that peer should have + sent an SDP at some point by then. If the channel has never + received an SDP, media won't have been set and the remote address + won't be known. Endpoints in general should not be doing this. + This patch makes it so that Asterisk will simply hang up a call + if it sends a 200 OK at this point. So far this odd behavior for + endpoints has only been observed in tests which involved manually + created SIP transactions in SIPp. (closes issue ASTERISK-22424) + Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/2827/ ........ Merged + revisions 399939 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399962 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 17:11 +0000 [r399938] Richard Mudgett + + * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c, + /: astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a + strange feature that came into the world under suspicious + circumstances to support an abuse of the ao2_container by + chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is + safe to remove it. The simplified code should help performance + slightly and make understanding the code easier. Review: + https://reviewboard.asterisk.org/r/2887/ ........ Merged + revisions 399937 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 14:35 +0000 [r399925] Mark Michelson + + * /, bridges/bridge_native_rtp.c: Fix refleaks of ast_rtp_instance + structures. These refleaks were causing bridged calls not to + close their RTP ports. Thus a call would leave open 4 ports (RTP + for party A, RTCP for party A, RTP for party B, and RTCP for + party B). This led to an eventual depletion of available RTP + ports. ........ Merged revisions 399924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-27 14:08 +0000 [r399913] Kinsey Moore + + * tests/test_cel.c, main/cel.c, /, include/asterisk/cel.h: Restore + usefulness of the CEL Peer field This change makes the CEL peer + field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and + fills the field with a comma-separated list of all channels in + the bridge other than the channel that is entering or exiting the + bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes + issue ASTERISK-22393) ........ Merged revisions 399912 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-26 18:51 +0000 [r399898] Kevin Harwell + + * res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h, + res/res_pjsip.exports.in, /, res/res_pjsip/security_events.c: + pjsip: race condition in registrar While handling a registration + request a race condition could occur if/when two+ clients + registered at the same time. This happened when one request + obtained a copy of the current contacts for an AOR and another + request did the same before the first request updated. Thus the + second would update and overwrite the first (or vice-versa + depending on which actually updated first). In the case of it + being the same contact two "add" events would be raised. pjsip + registration handling is now serialized to alleviate this issue. + (closes issue AST-1213) Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2860/ ........ Merged + revisions 399897 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-26 14:13 +0000 [r399875] Rusty Newton + + * /, apps/app_dial.c: Adding a few words to the Dial option 'r' + help text to clarify its tone argument description ........ + Merged revisions 399874 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 20:38 +0000 [r399844] Richard Mudgett + + * channels/sig_ss7.c, channels/chan_dahdi.c, /: chan_dahdi: CLI + "core stop gracefully" has needless delay for PRI and SS7. The + PRI and SS7 link control threads are not stopped correctly when + the chan_dahdi.so module is unloaded. The link control threads + pri_dchannel() and ss7_linkset() are not awakened from a poll() + to cancel the thread. * Added a SIGURG signal after requesting + the thread cancel to break the link control thread poll() + immediately. For SS7 it was slightly worse, the link poll() + timeout would always be whatever was the last libss7 scheduled + event time used. If no libss7 scheduled event was pending, the + thread could run more often than necessary. * Set nextms to 60 + seconds for the ss7_linkset() poll() if there is no other libss7 + scheduled event. ........ Merged revisions 399818 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399834 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399842 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 19:43 +0000 [r399799] Rusty Newton + + * /, res/res_pjsip.c: Broke the build - Fixing XML DTD violation + added in r399782, missing tags inside a ........ + Merged revisions 399798 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 19:29 +0000 [r399797] Michael L. Young + + * /, channels/chan_sip.c: chan_sip: Fix Realtime Peer Update + Problem When Un-registering And Expires Header In 200ok 1st Issue + When a realtime peer sends an un-REGISTER request, Asterisk + un-registers the peer but the database table record still has + regseconds and fullcontact for the peer. This results in calls + attempting to be routed to the peer which is no longer + registered. The expected behavior is to get busy/congested when + attempting to call an un-registered peer through the dialplan. + What was discovered is that we are clearing out the peer's + registration in the database in parse_register_contact() when + calling expire_register() but then upon returning from + parse_register_contact(), update_peer() is run which stores back + in the database table regseconds and fullcontact. 2nd Issue The + reporter pointed out that the 200 ok being returned by Asterisk + after un-registering a peer contains a Contact header with + ;expires= and the Expires header is not set to 0. This is + actually a regression. Tests were created for this second issue + (ASTERISK-22548). The tests have been reviewed and a Ship It! was + received on those tests. This patch does the following: * Do not + ignore the Expires header value even when it is set to 0. The + patch sets the pvt->expiry earlier on in the function so that it + is set properly and used. * If pvt->expiry is 0, do not call + update_peer since that means the peer has already been + un-registered and there is no need to update the database record + again since nothing has changed. (closes issue ASTERISK-22428) + Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L. + Young Patches: + asterisk-22428-rt-peer-update-and-expires-header.diff by Michael + L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2869/ ........ Merged + revisions 399794 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399795 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399796 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 18:38 +0000 [r399782] Rusty Newton + + * /, res/res_pjsip.c: Fixing documentation for the configOption + "external_media_address" of both Endpoints and Transports + Re-using some of Mark Michelson's text from an E-mail discussion + for: * Modifying synopsis for both options * Adding description + to both options * Changing name of "external_media_address" for + Endpoint configuration to "media_address" in anticipation of the + option name being changed. (As it is not really specific to + external destinations) (issue ASTERISK-22405) (closes issue + ASTERISK-22405) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2850/ ........ Merged + revisions 399781 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-24 22:55 +0000 [r399737-399750] Richard Mudgett + + * /, main/astobj2.c: astobj2: Made use OBJ_SEARCH_xxx identifiers + as field enum values internally. * Made ao2_unlink to protect + itself from stray OBJ_SEARCH_xxx values passed in. ........ + Merged revisions 399749 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Prevent some needless + breaking of the native IAX2 bridge. * Clean up some twisted code + in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and + AST_CONTROL_SRCCHANGE to a list of frames to prevent the native + bridge loop from breaking. * Passing the + AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a + native IAX2 bridge. (issue ABE-2912) Review: + https://reviewboard.asterisk.org/r/2870/ ........ Merged + revisions 399697 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399708 from + http://svn.asterisk.org/svn/asterisk/branches/11 For v12 and + above this is really just documentation until IAX2 native + bridging is restored. ........ Merged revisions 399736 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-24 19:22 +0000 [r399667-399696] Matthew Jordan + + * apps/app_queue.c, /: app_queue: Don't be quite so aggressive in + initializing the array We only need the first character. ........ + Merged revisions 399695 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_queue.c, /: app_queue: Initialize array holding + MixMonitor exec options If the channel variable MONITOR_EXEC is + set, app_queue will pass the specified execution parameters to + the MixMonitor application when a queue is recorded. If that + channel variable is not set, the buffer that holds the escaped + value was not being initialized to NULL, and so would be passed + to the MixMonitor application with garbage. Hilarity ensued as + app_mixmonitor attempted to execute gobeldy-gook. ........ Merged + revisions 399681 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/stasis_bridges.c, tests/test_cdr.c, main/cdr.c, /: Fix a + performance problem CDRs There is a large performance price + currently in the CDR engine. We currently perform two + ao2_callback calls on a container that has an entry for every + channel in the system. This is done to create matching pairs + between channels in a bridge. As such, the portion of the CDR + logic that this patch deals with is how we make pairings when a + channel enters a mixing bridge. In general, when a channel enters + such a bridge, we need to do two things: (1) Figure out if anyone + in the bridge can be this channel's Party B. (2) Make pairings + with every other channel in the bridge that is not already our + Party B. This is a two step process. In the first step, we look + through everyone in the bridge and see if they can be our Party B + (single_state_process_bridge_enter). If they can - yay! We mark + our CDR as having gotten a Party B. If not, we keep searching. If + we don't find one, we wait until someone joins who can be our + Party B. Step 2 is where we changed the logic + (handle_bridge_pairings and bridge_candidate_process). + Previously, we would first find candidates - those channels in + the bridge with us - from the active_cdrs_by_channel container. + Because a channel could be a candidate if it was Party B to an + item in the container, the code implemented multiple + ao2_container callbacks to get all the candidates. We also had to + store them in another container with some other meta information. + This was rather complex and costly, particularly if you have 300 + Local channels (600 channels!) going at once. Luckily, none of it + is needed: when a channel enters a bridge (which is when we're + figuring all this stuff out), the bridge snapshot tells us the + unique IDs of everyone already in the bridge. All we need to do + is: For all channels in the bridge: If the channel is us or our + Party B that we got in step 1, skip it Compare us and the + candidate to figure out who is Party A (based on some specific + rules) If we are Party A: Make a new CDR for us, append it to our + chain, and set the candidate as Party B If they are Party A: If + they don't have a Party B: Make a new CDR for them, append us to + their chain, and us as Party B Otherwise: Copy us over as Party B + on their existing CDR. This patch does that. Because we now use + channel unique IDs to find the candidates during bridging, + active_cdrs_by_channel now looks up things using uniqueid instead + of channel name. This makes the more complex code simpler; it + does, however, have the drawback that dialplan applications and + functions will be slightly slower as they have to iterate through + the container looking for the CDR by name. That's a small price + to pay however as the bridging code will be called a lot more + often. This patch also does two other minor changes: (1) It + reduces the container size of the channels in a bridge snapshot + to 1. In order to be predictable for multi-party bridges, the + order of the channels in the container must be stable; that is, + it must always devolve to a linked list. (2) CDRs and the + multi-party test was updated to show the relationship between two + dialed channels. You still want to know if they talked - + previously, dialed channels were always ignored, which is wrong + when they have managed to get a Party B. (closes issue + ASTERISK-22488) Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/2861/ ........ Merged + revisions 399666 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-23 12:03 +0000 [r399625] Joshua Colp + + * res/res_pjsip.c, res/res_pjsip_session.c, /: Fix crash in + res_pjsip on load if error occurs, and prevent unloading of + res_pjsip and res_pjsip_session. During load time in res_pjsip if + an error occurred the operation would attempt to rollback all + operations done during load. This is not permitted by PJSIP as it + will assert if the operation has not been done. This fix changes + the code so it will only rollback what has been initialized + already. Further changes also prevent res_pjsip and + res_pjsip_session from being unloaded. This is due to limitations + within PJSIP itself. The library environment can only be changed + to a certain extent and does not provide the ability, currently, + to deinitialize certain required functionality. (closes issue + ASTERISK-22474) Reported by: Corey Farrell ........ Merged + revisions 399624 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-21 04:49 +0000 [r399578-399608] Richard Mudgett + + * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix ref leaks in + ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the + loop so it is unref'ed after every loop. Moved message_blob to + loop and switched it to a regular variable. The regular variable + was used since message_blob is used in a very contained way. + (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches: + rtcp_report-leak.patch (license #5909) patch uploaded by Corey + Farrell Tested by: Corey Farrell ........ Merged revisions 399607 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/media_index.c: media_index: Fix + process_description_file() memory leak of file_id_persist. + ........ Merged revisions 399596 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/features_config.c: features_config: Fix config ref leak + of parkinglots. This leak happend for just about every channel + created. ........ Merged revisions 399585 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, apps/app_queue.c: app_queue: Fix json blob ref leak. The json + ref from queue_member_blob_create() was never released. ........ + Merged revisions 399583 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/json.c, /: json: Make it obvious that ast_json_unref() is + NULL safe. It looked like the safety check was done after the + NULL pointer was used. ........ Merged revisions 399576 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 22:44 +0000 [r399566] Kinsey Moore + + * main/config_options.c, /: Ensure global types in the config + framework are initialized If a config object was allocated but + one of its global objects was never encountered, then the global + object's defaults were never applied. Ensure that global objects + are initialized properly upon allocation instead of on + configuration. Review: https://reviewboard.asterisk.org/r/2866/ + ........ Merged revisions 399564 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399565 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 22:06 +0000 [r399554] Jonathan Rose + + * main/dial.c, /: originate/call forwarding: Fix a crash when + forwarding a call from originate (closes issue ASTERISK-22487) + Reported by: David M. Lee Review: + https://reviewboard.asterisk.org/r/2868/ ........ Merged + revisions 399553 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 16:18 +0000 [r399533] Joshua Colp + + * /, channels/chan_pjsip.c: Add a missing session supplement + unregistration in chan_pjsip for ACKs. (closes issue + ASTERISK-22453) Reported by: Corey Farrell Patches: + chan_pjsip_session_unregister_supplement.patch uploaded by Corey + Farrell (license 5909) ........ Merged revisions 399531 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-20 14:26 +0000 [r399515] Kevin Harwell + + * /, main/logger.c: Fix memory leak in logger. Fixed a memory leak + discovered in the logger where a temporary string buffer was not + being freed. (closes issue ASTERISK-22540) Reported by: John + Hardin ........ Merged revisions 399513 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399514 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-19 23:20 +0000 [r399503] Richard Mudgett + + * /, main/optional_api.c: optional_api: Make always use the + standard malloc functions even with MALLOC_DEBUG. ........ Merged + revisions 399501 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-19 17:01 +0000 [r399459] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Make direct media reinvites for + T38 put Asterisk in the media path Prior to this patch, Asterisk + would incorrectly use the previous endpoint addresses in SDP in + spite of providing its own port. T38 is never meant to be done + through directmedia and Asterisk should always be in the media + path for these streams. (closes issue ASTERISK-17273) Reported + by: Kevin Stewart (closes issue ASTERISK-18706) Reported by: + Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/ + ........ Merged revisions 399456 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 20:04 +0000 [r399405] Kinsey Moore + + * /, main/abstract_jb.c: Fix jitter buffer log file creation This + adjusts '/'-to-'#' replacement to replace all instances of '/' + instead of just the first to ensure that the jitter buffer log + file gets the correct name as per Richard Kenner's suggestion. + (closes issue ASTERISK-21036) Reported by: Richard Kenner + ........ Merged revisions 399402 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399403 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399404 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 17:23 +0000 [r399368-399378] Matthew Jordan + + * /, build_tools/prep_tarball: Update prep_tarball with new + documentation files on the Asterisk wiki This will now pull both + a command reference for the version being prepared, as well as an + Admin Guide that applies to all versions of Asterisk. (issue + ASTERISK-22439) Reported by: Olle Johansson ........ Merged + revisions 399351 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399373 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399376 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when + a timing module isn't loaded If bridge_softmix fails to be + created because no timing source is present in Asterisk, this + will currently fail gracefully but with (most likely) a generic + error message by whatever module tried to create the softmix + bridge. This patch adds a more explicit warning so you can + actually diagnose and fix the problem. Review: + https://reviewboard.asterisk.org/r/2857/ ........ Merged + revisions 399353 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399365 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 17:15 +0000 [r399352] Richard Mudgett + + * main/config_options.c: Make config framework able to reload + module configs with multiple config files. The config framework + is supposed to be able to load configs that come from multiple + config files. The principle example is chan_sip's sip.conf and + users.conf. Unfortunately, it only does this correctly on initial + load. This patch causes the module's config to be reloaded + entirely if any of the config files change. (closes issue + ASTERISK-22009) Reported by: Richard Mudgett Review: + https://reviewboard.asterisk.org/r/2859/ + +2013-09-18 14:56 +0000 [r399340] Kevin Harwell + + * res/res_pjsip_messaging.c, /: res_pjsip_messaging: Register + message technology as pjsip pjsip's message technology was being + registered as 'sip', which was causing it to not load due it + conflicting with chan_sip's registered 'sip' technology for + messaging. It now registers as 'pjsip'. However, due to this + change the "to" field for outgoing pjsip messages need to be + prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to + res_pjsip_messaging will automatically have their "to" fields + altered in order to accommodate the change. Outgoing messages + also handle changing it back to 'sip' before being sent so the + pjsip library will properly handle it. (closes issue + ASTERISK-22445) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2833/ ........ Merged + revisions 399339 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-18 00:13 +0000 [r399295] Michael L. Young + + * /, main/features_config.c: Fix Segfault In features-config.c When + Application Has No Arguments Some applications do not require + arguments. Therefore, when parsing application maps in + features.conf, it is possible that app_data will be set to NULL. + * This patch sets app_data to "" if it is NULL. Review: + https://reviewboard.asterisk.org/r/2804 ........ Merged revisions + 399294 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 23:10 +0000 [r399284] Mark Michelson + + * res/res_pjsip_sdp_rtp.c, res/res_pjsip/pjsip_configuration.c, + res/res_pjsip_t38.c, include/asterisk/res_pjsip.h, /: Change the + "external_media_address" PJSIP endpoint option to + "media_address". The endpoint option does not apply to + communication with external entities. Rather, the option is + applied to all communications with the endpoint. The + external_media_address transport configuration option may + override the endpoint option if it turns out that we are going to + be communicating with an external entity. Two things of note: 1) + I have not updated the XML documentation. This is being taken + care of by Rusty as part of his work on issue ASTERISK-22405 2) + This commit is likely to cause testsuite failures since there are + tests that use the external_media_address endpoint option, and + they will need to be changed over. Well, I'm planning to get that + updated ASAP after this commit. (closes issue ASTERISK-22528) + reported by Rusty Newton ........ Merged revisions 399283 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 18:44 +0000 [r399269] Kevin Harwell + + * main/logger.c, main/asterisk.c, /: Remote console: more output + discrepancies The remote console continued to have issues with + its output. In this case CLI command output would either not show + up (if verbose level = 0) or would contain verbose prefixes (if + verbose level > 0) once log messages were sent to the remote + console. The fix now now adds verbose prefix data to all new + lines contained in a verbose log string. (closes issue + ASTERISK-22450) Reported by: David Brillert (closes issue + AST-1193) Reported by: Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/2825/ ........ Merged + revisions 399267 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399268 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 17:55 +0000 [r399258] Richard Mudgett + + * /, include/asterisk/features_config.h: Fix doxygen to use correct + units of features.conf options. ........ Merged revisions 399257 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 17:10 +0000 [r399238-399248] Mark Michelson + + * main/bridge_basic.c, main/features_config.c, /: Fix other + timeouts (atxferloopdelay and atxfernoanswertimeout) to use + seconds instead of milliseconds. Thanks to Richard Mudgett for + pointing this out. ........ Merged revisions 399247 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/features_config.c, /, include/asterisk/features_config.h, + main/bridge_basic.c: Switch transferdigittimeout to be configured + as seconds instead of milliseconds. This was an unintentional + consequence of the update of features.conf to use the config + framework in Asterisk 12. Thanks to Marco Signorini on the + Asterisk developers list for pointing out the problem. ........ + Merged revisions 399237 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-17 14:58 +0000 [r399226] Kevin Harwell + + * apps/confbridge/conf_state_multi_marked.c, /: Confbridge: empty + conference not being torn down Confbridge would not properly tear + down an empty conference bridge when all users were kicked via + end_marked=yes and at least one user was also set to wait_marked. + This occurred because while end_marked users were being kicked + and at least one was also set to wait_marked then the leave + wait_marked handler would be called on that user, but there would + be no waiting user (still considered active). The waiting users + would decrement and now be negative. The conference would remain, + but be put into an inactive state. The solution was to move from + the active list to the wait list, those users with wait_marked + set right before kicking. This allows both the active and wait + users to decrement correctly and the confbridge to tear down + properly. A crashed also occurred when trying to list the + specific conference from the CLI. This happened because the + conference specified was invalid. Since the conference properly + tears down now there is no way to reference it thus alleviating + the crash as well. (closes issue ASTERISK-21859) Reported by: + Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/ + ........ Merged revisions 399222 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399225 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-16 18:36 +0000 [r399161-399208] Richard Mudgett + + * tests/test_ari_model.c, /: Fix module load errors for + test_ari_model.so. You cannot use a function pointer variable + with an external function from another dynamically loaded module + because data variables are always resolved even with RTLD_LAZY. * + Added wrapper functions for ast_ari_validate_int() and + ast_ari_validate_string() to use instead for the function pointer + variable. (closes issue ASTERISK-22457) Reported by: David M. Lee + ........ Merged revisions 399207 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_speech_utils.c, /, res/res_speech.exports.in: + app_speech_utils: Fix unresolved symbol ast_speech_get_setting(). + Fixes regression introduced by -r374096. * Made + res_speech.export.in export ast_* symbols instead of specific + functions. * Made app_speech_utils.c declare that it is dependent + upon res_speech. (issue ASTERISK-17136) Reported by: Richard + Kenner ........ Merged revisions 399197 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix saving the wrong expiry + time in astdb. When a new IAX2 client registers, the astdb + database is updated with the value of minregexpire defined in + iax.conf instead of using the expiry time that is provided by the + client. The provided expiry time of the client is updated after + inserting the astdb entry. As a consequence, restarting or + reloading asterisk creates clients whose registration may expire + before they reregister. The clients are therefore unavailable + after minregexpire seconds until they reregister. * Move updating + of the expiry time to before inserting into the astdb. (closes + issue ASTERISK-22504) Reported by: Stefan Wachtler Patches: + chan_iax2.c.patch (license #6533) patch uploaded by Stefan + Wachtler ........ Merged revisions 399158 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399159 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399160 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-16 02:37 +0000 [r399147] Matthew Jordan + + * main/cdr.c, /: Filter internal channels out of bridge enter/leave + message handling Some channels exist merely as an implementation + detail in Asterisk, such as ConfBridge's announcer/recorder + channels. These channels should never be exposed to the outside + world, or to interfaces that report on Asterisk. We already + filter out such channels in snapshot processing; however, we + failed to filter out bridge related messages that involved these + channels. This patch filters out bridge related messages that are + for such channels. This prevents a spurious WARNING message from + being displayed when those channels move in and out of bridges. + ........ Merged revisions 399146 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 22:19 +0000 [r399138] Richard Mudgett + + * res/parking/parking_bridge_features.c, apps/app_agent_pool.c, + include/asterisk/features.h, main/channel.c, + res/parking/parking_tests.c, include/asterisk/bridge_channel.h, + main/features.c, tests/test_cel.c, main/bridge_channel.c, + tests/test_cdr.c, apps/confbridge/conf_chan_announce.c, + include/asterisk/bridge.h, res/res_pjsip_refer.c, /, + channels/chan_sip.c, res/stasis/control.c, main/bridge.c, + main/bridge_basic.c, main/core_unreal.c, + res/parking/parking_applications.c, main/core_local.c: Restore + Dial, Queue, and FollowMe 'I' option support. The Dial, Queue, + and FollowMe applications need to inhibit the bridging initial + connected line exchange in order to support the 'I' option. * + Replaced the pass_reference flag on ast_bridge_join() with a + flags parameter to pass other flags defined by enum + ast_bridge_join_flags. * Replaced the independent flag on + ast_bridge_impart() with a flags parameter to pass other flags + defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, + and FollowMe applications are now the only callers of + ast_bridge_call() and ast_bridge_call_with_flags(), changed the + calling contract to require the initial COLP exchange to already + have been done by the caller. * Made all callers of + ast_bridge_impart() check the return value. It is important. As a + precaution, I also made the compiler complain now if it is not + checked. * Did some cleanup in parking_tests.c as a result of + checking the ast_bridge_impart() return value. An independent, + but associated change is: * Reduce stack usage in + ast_indicate_data() and add a dropping redundant connected line + verbose message. (closes issue ASTERISK-22072) Reported by: + Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ + ........ Merged revisions 399136 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 20:55 +0000 [r399101] David M. Lee + + * /, main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not + defined. If MALLOC_DEBUG is enabled, then the debug destructor + for the container is used, which would erroneously write to + /tmp/refs. This patch only uses the debug destructor if ref_debug + is used. (closes issue ASTERISK-22536) ........ Merged revisions + 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 399099 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399100 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:50 +0000 [r399082-399084] Mark Michelson + + * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c, + include/asterisk/res_pjsip.h, res/res_pjsip.exports.in, /: Create + more accurate Contact headers for dialogs when we are the UAS. + (closes issue AST-1207) reported by John Bigelow Review: + https://reviewboard.asterisk.org/r/2842 ........ Merged revisions + 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip/config_auth.c, /, + res/res_pjsip_outbound_authenticator_digest.c, + res/res_pjsip_authenticator_digest.c: Change how realms are + handled for outbound authentication. With this change, if no + realm is specified in an outbound auth section, then we will + simply match the realm that was present in the 401/407 challenge. + (closes issue ASTERISK-22471) Reported by George Joseph (closes + issue ASTERISK-22386) Reported by Rusty Newton Patches: + outbound_auth_realm_v4.patch uploaded by George Joseph (License + #6322) ........ Merged revisions 399059 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:43 +0000 [r399080-399081] David M. Lee + + * /: Recorded merge of revisions 399035,399049 from + http://svn.asterisk.org/svn/asterisk/branches/12 These were lost + in r399071 + + * /: Put merge tracking for r399039 back. + +2013-09-13 14:27 +0000 [r399071] Rusty Newton + + * /, res/res_pjsip_endpoint_identifier_ip.c: Broke the build! + Forgot para tags within my description. + https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304 + ........ Merged revisions 399064 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:22 +0000 [r399042-399051] David M. Lee + + * res/res_pjsip_log_forwarder.c (added), res/res_pjsip_logger.c, + res/res_rtp_asterisk.c, /: res_pjsip: Forward PJSIP logging to + Asterisk logging This patch uses PJSIP's pj_log_set_log_func() to + forward PJSIP's log messages to Asterisk's logger. This is done + in a new module: res_pjsip_log_forwarder.so. This patch sets + defaultenabled on the existing res_pjsip_logger.so to no, since + logging every SIP packet seems a bit odd to do by default, and is + (hopefully) less necessary with regular PJSIP logging. It also + removes res_rtp_asterisk's disabling of PJSIP logging. (closes + issue ASTERISK-22360) Reported by: Joshua Colp Review: + https://reviewboard.asterisk.org/r/2830/ ........ Merged + revisions 399049 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_http_websocket.c: ARI: Fix WebSocket response when + subprotocol isn't specified When I moved the ARI WebSocket from + /ws to /ari/events, I added code to allow a WebSocket to connect + without specifying the subprotocol if there's only one + subprotocol handler registered for the WebSocket. Naively, I + coded it to always respond with the subprotocol in use. + Unfortunately, according to RFC 6455, if the server's response + includes a subprotocol header field that "indicates the use of a + subprotocol that was not present in the client's handshake [...], + the client MUST _Fail the WebSocket Connection_.", emphasis + theirs. This patch correctly omits the Sec-WebSocket-Protocol if + one is not specified by the client. (closes issue ASTERISK-22441) + Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged + revisions 399039 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 14:17 +0000 [r399036] Kinsey Moore + + * /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This + change ensures that MeetMeAdmin commands requiring a user + actually get a user and fixes another issue where an extra + dereference could occur for a last-entered user being ejected if + a user identifier was also provided. (closes issue + ASTERISK-21907) Reported by: Alex Epshteyn Review: + https://reviewboard.asterisk.org/r/2844/ ........ Merged + revisions 399033 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 399034 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 399035 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-13 13:28 +0000 [r399032] Rusty Newton + + * /, res/res_pjsip_endpoint_identifier_ip.c: 'identify' + configObject doesn't have a synopsis Add a straightforward + synopsis and description to the identify config object in XML + documentation. (issue ASTERISK-22311) (closes issue + ASTERISK-22311) Reported By: Rusty Newton ........ Merged + revisions 399031 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 23:42 +0000 [r399020-399022] Richard Mudgett + + * /, main/bridge.c: CLI bridge: Fix "bridge destroy " and + "bridge kick " tab completion. These two commands must + deal with the live bridges container for tab completion and not + the stasis cache. ........ Merged revisions 399021 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/bridge.c, /: astobj2: Register the bridges container for + debug inspection. ........ Merged revisions 399019 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 23:23 +0000 [r399018] Rusty Newton + + * /, res/res_pjsip_acl.c: Documentation fix and improvements to XML + configuration help res_pjsip_acl * One bug fix. Made the synopsis + for "type" to accurate. * changing the usage of "IP-domains" to + "IP addresses" * clarifying the usage for the options, by adding + a relevant description for each * modified other areas of the XML + help for clarity, such as the module description and a few + synopsis changes here and there. See the patch. (issue + ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty + Newton Review: https://reviewboard.asterisk.org/r/2823/ ........ + Merged revisions 399017 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 20:27 +0000 [r399006] Jonathan Rose + + * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip: + Revert r398835 due to failing tests involving originate (issue + ASTERISK-22424) Reported by: Jonathan Rose ........ Merged + revisions 398977 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398986 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398991 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 16:44 +0000 [r398939] Richard Mudgett + + * main/core_unreal.c, /: core_local: Fix memory corruption race + condition. The masquerade super test is failing on v12 with high + fence violations and crashing. The fence violations are showing + that party id allocated memory strings are somehow getting + corrupted in the bridge_reconfigured_connected_line_update() + function. The invalid string values happen to be the freed memory + fill pattern. After much puzzling, I deduced that the + bridge_reconfigured_connected_line_update() is copying a string + out of the source channel's caller party id struct just as + another thread is updating it with a new value. The copying + thread is using the old string pointer being freed by the + updating thread. A search of the code found the + unreal_colp_redirect_indicate() routine updating the caller party + id's without holding the channel lock. A latent bug in v1.8 and + v11 hatched in v12 because of the bridging and connected line + changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2839/ ........ Merged + revisions 398938 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 15:23 +0000 [r398928] David M. Lee + + * res/res_pjsip.c, /: Fix symbol collision with pjsua. We shouldn't + be exporting any symbols that start with pjsip_. ........ Merged + revisions 398927 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-12 00:04 +0000 [r398883-398887] Rusty Newton + + * /, apps/app_queue.c: 'queue add member' help text correction You + are adding dial strings to the queue, not channels. An aribitrary + string could be used, but you are typically referencing a + channel. Correcting the command help text. (issue ASTERISK-22263) + (closes issue ASTERISK-22263) Reported By: Rusty Newton ........ + Merged revisions 398884 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398885 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398886 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * configs/chan_dahdi.conf.sample, /: Documentation fix - + waitfordialtone is not boolean, it's time in milliseconds + Changing text in chan_dahdi.conf sample to be accurate. (issue + ASTERISK-22308) (closes issue ASTERISK-22308) Reported By: + Malcolm Davenport ........ Merged revisions 398880 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398881 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398882 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-11 20:03 +0000 [r398838] Jonathan Rose + + * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip: + Reject calls without prior SDP on 200 OK If we receive a 200 OK + without SDP, we will now check to see if the remote address has + been established for that channel's RTP session and if the to tag + for that channel has changed from the most recent to tag in a + response less than 200. If either a change has been made since + the last to-tag was received or the remote address is unset, then + we will drop the call. (closes issue ASTERISK-22424) Reported by: + Jonathan Rose Review: + https://reviewboard.asterisk.org/r/2827/diff/#index_header + ........ Merged revisions 398835 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398836 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398837 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-11 18:03 +0000 [r398822] Russell Bryant + + * configs/confbridge.conf.sample, /: Fix typo in + confbridge.conf.sample The denoise filter requires func_speex, + not codec_speex. Fix this in the description of the denoise=yes + option in confbridge.conf. ........ Merged revisions 398820 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398821 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-11 14:23 +0000 [r398808] Kevin Harwell + + * res/res_pjsip_caller_id.c, channels/chan_pjsip.c, /: pjsip: + reinvite for connected line updates occurs when it should not + Connected line updates are now only sent out if an actual update + needs to occur. This happens under the following conditions: 1. + The endpoint we are sending to is trusted. 2. Either a + P-Asserted-Identity or Remote Party-ID header needs to be + added/sent. 3. The connected id's number and name are valid. Also + added an SDP when an update is sent out. (closes issue AST-1212) + Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2831/ ........ Merged + revisions 398806 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-10 18:05 +0000 [r398760] Richard Mudgett + + * main/event.c, res/res_musiconhold.c, main/indications.c, + main/asterisk.c, main/xmldoc.c, main/cli.c, /, + funcs/func_dialgroup.c, main/heap.c, + res/res_pjsip/pjsip_configuration.c: Fix incorrect usages of + ast_realloc(). There are several locations in the code base where + this is done: buf = ast_realloc(buf, new_size); This is going to + leak the original buf contents if the realloc fails. Review: + https://reviewboard.asterisk.org/r/2832/ ........ Merged + revisions 398757 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398758 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398759 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-10 17:50 +0000 [r398751-398755] David M. Lee + + * utils/check_expr.c, /: Fixed utils directory breakage from + r398748, this time with extra hate. ........ Merged revisions + 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398753 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398754 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * utils/check_expr.c, /, utils/ael_main.c, utils/conf2ael.c: Fixed + utils directory breakage from r398648 ........ Merged revisions + 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398749 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398750 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-09 23:29 +0000 [r398732] Richard Mudgett + + * main/astmm.c, /: MALLOC_DEBUG: Change fence magic number to be + completely different from the freed magic number. Race conditions + between freeing a nul terminated string and ast_strdup()'ing it + are more likely to be detected if the fence and freed magic + numbers are completely different. ........ Merged revisions + 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398721 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398726 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-09 22:00 +0000 [r398695] Mark Michelson + + * res/res_pjsip_endpoint_identifier_ip.c, /: Add extra debugging to + res_pjsip_endpoint_identifier_ip ........ Merged revisions 398694 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-09 20:13 +0000 [r398641-398652] David M. Lee + + * /, main/utils.c, include/asterisk/lock.h, main/lock.c: Fix + DEBUG_THREADS when lock is acquired in __constructor__ This patch + fixes some long-standing bugs in debug threads that were + exacerbated with recent Optional API work in Asterisk 12. With + debug threads enabled, on some systems, there's a lock ordering + problem between our mutex and glibc's mutex protecting its module + list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one + thread, the module list will be locked before acquiring our + mutex. In another thread, our mutex will be locked before locking + the module list (which happens in the depths of calling + backtrace()). This patch fixes this issue by moving backtrace() + calls outside of critical sections that have the mutex acquired. + The bigger change was to reentrancy tracking for + ast_cond_{timed,}wait, which wrongly assumed that waiting on the + mutex was equivalent to a single unlock (it actually suspends all + recursive locks on the mutex). (closes issue ASTERISK-22455) + Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged + revisions 398648 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398649 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398651 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/ari/resource_channels.h, /, rest-api/api-docs/channels.json: + Multiple revisions 398638-398639 ........ r398638 | dlee | + 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note + about expected behavior of originate ........ r398639 | dlee | + 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note + about expected behavior of originate (the rest of the commit) + ........ Merged revisions 398638-398639 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-08 23:30 +0000 [r398629] Matthew Jordan + + * tests/test_cdr.c, /: Update CDR Unit tests to reflect container + changes in r398579 When a channel joins a multi-party bridge, the + ordering of the CDRs that is created is determined by the + ordering of the channels who happen to be in that bridge. When + r398579 changed the number of buckets in the container to + something sensible, it changed the ordering that the CDRs was + created in, causing one of the multiparty tests to fail. This + fixes the test with the now expected ordering. ........ Merged + revisions 398628 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-07 01:03 +0000 [r398603-398620] Kinsey Moore + + * /, res/res_xmpp.c: Prevent XMPP timeout on blank responses + Sometimes the Google Voice servers have a bad habit of sending + out 1 byte replies to the xmpp resource. When a blank 1 byte + reply is received from the socket the buffer attempts to wait + (endlessly) for the rest of the reply from google which + effectively blocks the socket and google voice calls will no + longer come into the server. This patch allows the xmpp module to + correctly detect empty packets and send out ping replies to + google. It also sets a socket timeout on the default socket which + prevents the xmpp socket from closing and preventing future + google voice calls from coming into the server. Furthermore + instead of sending an empty reply back to google we send a proper + xmpp ping reply back. This also adds several more socket + messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy + Review: https://reviewboard.asterisk.org/r/2771 Patches: + xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........ + Merged revisions 398618 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398619 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_xmpp.c, res/res_jabber.c: Multiple revisions + 398558,398577 ........ r398558 | kmoore | 2013-09-06 14:28:16 + -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed + MWI The mailbox and context are swapped on the receiving end for + all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and + all more recent versions. This swaps those values to be correct + when publishing to the internal event system from Jabber/XMPP + distributed MWI state. (closes issue ASTERISK-22435) Reported by: + abelbeck Tested by: Michael Keuter Patches: + asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by + abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch + uploaded by abelbeck ........ Merged revisions 398523 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | + 10 lines Commit the remainder of r398523 This is a missing part + of the commit in revision 398523 that corrects the name of a + variable. (issue ASTERISK-22435) ........ Merged revisions 398576 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 398558,398577 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398580 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-06 21:17 +0000 [r398557-398583] Richard Mudgett + + * main/cdr.c, /: cdr: Change the number of container buckets to be + similar to the channels container. * Fix the temporary cdr + candidate containers to use a prime number of buckets. ........ + Merged revisions 398579 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/core_local.c, /: core_local: Fix LocalOptimizationBegin AMI + event missing Source channel snapshot. * Fix the + LocalOptimizationBegin AMI event by eliminating an artificial + buffer size limitation that is too small anyway. ........ Merged + revisions 398572 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: cdr: Fix some ref leaks. * Added missing + unregister of the cdr container in cdr_engine_shutdown(). * Fixed + ref leak in off nominal path of cdr_object_alloc(). * Removed + some unnecessary NULL checks in cdr_object_dtor(). ........ + Merged revisions 398562 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/astobj2.h, main/cel.c, main/features_config.c, + apps/app_agent_pool.c, main/cdr.c, main/udptl.c, /, + main/parking.c, main/stasis_config.c: astobj2: Add warn unused + attribute to some functions. * Fixed resulting warnings with + improper use of ao2_global_obj_replace(). * Made a couple uses of + ao2_global_obj_replace_unref(x, NULL) into the equivalent and + more appropriate ao2_global_obj_release() call. ........ Merged + revisions 398533 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-06 18:53 +0000 [r398512-398522] Kinsey Moore + + * main/http.c, /, res/stasis/app.c: Fix build warnings When + AST_DEVMODE is not defined, ast_asserts are not compiled into the + binary. In some cases, this means variables are not referenced or + are set but unused which causes warnings to show up. (closes + issue ASTERISK-22446) Reported by: Jason Parker (qwell) ........ + Merged revisions 398521 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_h323.c: Fix chan_h323 compilation This fixes the + things in chan_h323 that were missed or ignored in the great + channel opaquification and gets chan_h323 back into a compiling + state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov + Patches: chan_h323.patch uploaded by Dmitry Melekhov ........ + Merged revisions 398510 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398511 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-05 21:48 +0000 [r398384-398499] Richard Mudgett + + * /, main/astobj2.c: astobj2: Only define ao2_bt() once. * Make + ao2_bt() not use single char variable names. * Fix ao2_bt() + formatting. ........ Merged revisions 398498 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Reduce indentation in + __attempt_transmit(). * Reduce indentation in + __attempt_transmit(). * Don't update the static last error time + variable every time in __schedule_action() and socket_read(). + ........ Merged revisions 398456 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398457 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398458 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker + thread idle_list. * Fix stray reference to idle_list in + cleanup_thread_list(). This may be the reason for the note in + iax2_process_thread() about threads not being removed from the + task lists. * Move cleanup_thread_list(&idle_list) to after the + other lists are cleaned up. ........ Merged revisions 398416 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398417 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398418 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock + avoidance. * Fix bridgecallno deadlock avoidance. When doing + deadlock avoidance, you need to retest the status of values for + each loop to see if you still need the lock for bridgecallno. * + As a safety check, after acquiring the bridgecallno lock you + should check if iaxs[bridgecallno] is NULL just like the current + callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE + to after processing any deferred frames to ensure that the + iostate is IDLE when it is placed back into the idle list. + defer_full_frame() tries to ensure iax2_process_thread() wakes up + to process the frame. ........ Merged revisions 398379 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398380 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398381 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-05 14:10 +0000 [r398369] Mark Michelson + + * /, res/res_pjsip_outbound_registration.c: Clarify server_uri and + client_uri registration settings. Used some of Rusty's suggested + language plus also included more SIPesque descriptions of where + the URIs are actually used in an outgoing REGISTER. (closes issue + ASTERISK-22390) reported by Rusty Newton ........ Merged + revisions 398368 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 23:07 +0000 [r398304] Richard Mudgett + + * channels/iax2/parser.c, /: chan_iax2: Add missing control frame + names to debug frame decode output. ........ Merged revisions + 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 398302 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398303 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 22:49 +0000 [r398300] Mark Michelson + + * /, res/res_pjsip_outbound_authenticator_digest.c: Give more + detail regarding failures to create request with auth + credentials. (issue ASTERISK-22386) ........ Merged revisions + 398299 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 21:37 +0000 [r398284-398287] Jonathan Rose + + * /, tests/test_voicemail_api.c: unit tests: test_voicemail_api + leaks stringfields from snapshots (closes issue ASTERISK-22414) + Reported by: Corey Farrell Patches: + test_voicemail_api-leaks-11.patch uploaded by coreyfarrell + (license 5909) ........ Merged revisions 398285 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398286 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * apps/app_voicemail.c, /: app_voicemail: Fix leaking config + objects when msg_id doesn't match (issues ASTERISK-22414) + Reported by: Corey Farrell Patch: + test_voicemail_api-leaks-11.patch uploaded by coreyfarrell + (license 5909) ........ Merged revisions 398281 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398283 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 16:03 +0000 [r398238] Richard Mudgett + + * channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output + printed with arbitrary verbose levels. Fix the misdn debug output + to remote consoles. chan_misdn uses ast_console_puts() which + doesn't know about verbose levels. Better to use ast_verbose() + instead. Without this patch the misdn debug messages are appended + to the verbose level which ever was set by the message sent to + the console before, i.e. any undefined level. (closes issue + AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch + (license #6372) patch uploaded by Guenther Kelleter ........ + Merged revisions 398235 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398236 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398237 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-04 14:32 +0000 [r398227] Kevin Harwell + + * /, res/res_pjsip_outbound_registration.c: Debug messages for + pjsip outbound registration Added debug messages indicating that + an outbound registration attempt was made and it was successful + in pjsip. (closes issue ASTERISK-22388) Reported by: Rusty Newton + ........ Merged revisions 398226 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 20:28 +0000 [r398217] Alexandr Anikin + + * /, addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling + on empty tcs received ........ Merged revisions 398214 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398215 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 18:09 +0000 [r398207] Kinsey Moore + + * res/res_pjsip_dtmf_info.c, /: Prevent a crash in + res_pjsip_dtmf_info.c This change makes sure that a content type + header exists before checking the contents of the header against + known SIP INFO DTMF content types. ........ Merged revisions + 398206 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 17:19 +0000 [r398205] David M. Lee + + * Makefile, /: Fixed 'make clean' for wiki docs ........ Merged + revisions 398198 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-03 14:29 +0000 [r398197] Walter Doekes + + * /, cel/cel_custom.c: Be a little more verbose when loading + cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/ + ........ Merged revisions 398167 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398168 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398196 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 20:58 +0000 [r398150] David M. Lee + + * main/asterisk.c, include/asterisk/optional_api.h, /, + main/optional_api.c: Fix graceful shutdown crash. The cleanup + code for optional_api needs to happen after all of the optional + API users and providers have unused/unprovided. Unfortunately, + regsitering the atexit() handler at the beginning of main() isn't + soon enough, since module destructors run after that. ........ + Merged revisions 398149 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 20:37 +0000 [r398148] Rusty Newton + + * /, configs/pjsip.conf.sample: New pjsip.conf.sample (issue + ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt + Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........ + Merged revisions 398147 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 19:55 +0000 [r398124-398140] Kevin Harwell + + * /, res/res_pjsip_outbound_registration.c, + include/asterisk/sorcery.h, res/res_pjsip.c, + res/res_pjsip/config_transport.c, main/sorcery.c: Add a + reloadable option for sorcery type objects Some configuration + objects currently won't place nice if reloaded. Specifically, in + this case the pjsip transport objects. Now when registering an + object in sorcery one may specify that the object is allowed to + be reloaded or not. If the object is set to not reload then upon + reloading of the configuration the objects of that type will not + be reloaded. The initially loaded objects of that type however + will remain. While the transport objects will not longer be + reloaded it is still possible for a user to configure an endpoint + to an invalid transport. A couple of log messages were added to + help diagnose this problem if it occurs. (closes issue + ASTERISK-22382) Reported by: Rusty Newton (closes issue + ASTERISK-22384) Reported by: Rusty Newton Review: + https://reviewboard.asterisk.org/r/2807/ ........ Merged + revisions 398139 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/config.c, res/res_security_log.c, /, channels/chan_sip.c, + main/translate.c, main/named_acl.c, main/indications.c: Fix + various memory leaks main/config.c - cleanup cache fie includes + res/res_security_log.c - unregister logger level + channesl/chan_sip.c - cleanup io context and notify_types + main/translator.c - cleanup at shutdown main/named_acl.c - + cleanup cli commands main/indications.c - + ast_get_indication_tone() unref default_tone_zone if used (closes + issues ASTERISK-22378) Reported by: Corey Farrell Patches: + config_shutdown.patch uploaded by coreyfarrell (license 5909) + res_security_log.patch uploaded by coreyfarrell (license 5909) + chan_sip-11.patch uploaded by coreyfarrell (license 5909) + indications_refleak.patch uploaded by coreyfarrell (license 5909) + named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license + 5909) translate_shutdown.patch uploaded by coreyfarrell (license + 5909) ........ Merged revisions 398102 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398103 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398116 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 18:38 +0000 [r398101] Matthew Jordan + + * /, UPGRADE-12.txt (added), UPGRADE.txt: Update UPGRADE.txt file + for Asterisk 12 This simply pulls in the changes that were + breaking from the CHANGES file and updates a few other areas + accordingly. It also removes the 10 => 11 notes, which are + traditionally removed from each major version and stored in the + appropriate UPGRADE-X.txt file. ........ Merged revisions 398100 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 18:30 +0000 [r398064-398099] Jonathan Rose + + * main/features_config.c, /, main/config_options.c: + features_config: Ignore parkinglots in features.conf instead of + failing to load Parkinglots are defined in res_features.conf now, + but this patch fixes features_config so that features don't fail + to load when parkinglots are present in features.conf Review: + https://reviewboard.asterisk.org/r/2801/ ........ Merged + revisions 398068 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/features_config.c, main/udptl.c, /: features_config: Don't + require features.conf to be present for Asterisk to load (closes + issue ASTERISK-22426) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2806/ ........ Merged + revisions 398020 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 17:59 +0000 [r398063] Kevin Harwell + + * main/manager.c, /, res/res_agi.c: Memory leak fix + ast_xmldoc_printable returns an allocated block that must be + freed by the caller. Fixed manager.c and res_agi.c to stop + leaking these results. (closes issue ASTERISK-22395) Reported by: + Corey Farrell Patches: manager-leaks-12.patch uploaded by + coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded + by coreyfarrell (license 5909) ........ Merged revisions 398060 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 398061 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398062 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 17:11 +0000 [r398024-398026] Richard Mudgett + + * tests/test_substitution.c, /: test_substitution: Fix failing + test. Revert the -r392190 change. The original test was correct. + The CDR code was actually returning an unititialized buffer. + ........ Merged revisions 398025 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * tests/test_substitution.c, /: test_substituition: Fix failed test + reporting to actually report failure. You cannot put the "Testing + pass/fail" on a single line before actually performing the + test. Now any additional failure information is logged before the + test pass/fail announcement. * Added an additional CDR(answer,u) + test. ........ Merged revisions 398018 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 398019 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398023 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 16:27 +0000 [r398003-398017] Kevin Harwell + + * /, apps/app_mixmonitor.c: Fix memory leaks (closes issue + ASTERISK-22368) Reported by: Corey Farrell Patches: + issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes + (license 5674) ........ Merged revisions 398004 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged + revisions 398011 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398016 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/asterisk.c, /: Check return value on fwrite ........ Merged + revisions 398000 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 398002 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 13:40 +0000 [r397987-397990] David M. Lee + + * rest-api-templates/swagger_model.py, res/ari/ari_websockets.c, + channels/sip/include/sip.h, main/asterisk.c, res/res_ari.c, + tests/test_optional_api.c (added), /, channels/chan_sip.c, + include/asterisk/autoconfig.h.in, configure.ac, + rest-api-templates/res_ari_resource.c.mustache, + res/ari/internal.h, res/res_http_websocket.c, CHANGES, + include/asterisk/compiler.h, include/asterisk/ari.h, + main/loader.c, include/asterisk/optional_api.h, + build_tools/cflags.xml, configure, res/res_ari_events.c, + include/asterisk/http_websocket.h, main/optional_api.c (added): + optional_api: Fix linking problems between modules that export + global symbols With the new work in Asterisk 12, there are some + uses of the optional_api that are prone to failure. The details + are rather involved, and captured on [the wiki][1]. This patch + addresses the issue by removing almost all of the magic from the + optional API implementation. Instead of relying on weak symbol + resolution, a new optional_api.c module was added to Asterisk + core. For modules providing an optional API, the pointer to the + implementation function is registered with the core. For modules + that use an optional API, a pointer to a stub function, along + with a optional_ref function pointer are registered with the + core. The optional_ref function pointers is set to the + implementation function when it's provided, or the stub function + when it's now. Since the implementation no longer relies on + magic, it is now supported on all platforms. In the spirit of + choice, an OPTIONAL_API flag was added, so we can disable the + optional_api if needed (maybe it's buggy on some bizarre platform + I haven't tested on) The AST_OPTIONAL_API*() macros themselves + remained unchanged, so existing code could remain unchanged. But + to help with debugging the optional_api, the patch limits the + #include of optional API's to just the modules using the API. + This also reduces resource waste maintaining optional_ref + pointers that aren't used. Other changes made as a part of this + patch: * The stubs for http_websocket that wrap system calls set + errno to ENOSYS. * res_http_websocket now properly increments + module use count. * In loader.c, the while() wrappers around + dlclose() were removed. The while(!dlclose()) is actually an + anti-pattern, which can lead to infinite loops if the module + you're attempting to unload exports a symbol that was directly + linked to. * The special handling of nonoptreq on systems without + weak symbol support was removed, since we no longer rely on weak + symbols for optional_api. [1]: + https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue + ASTERISK-22296) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2797/ ........ Merged + revisions 397989 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_stasis_playback.c, /, + include/asterisk/stasis_app_recording.h, + res/ari/resource_recordings.h, res/res_stasis_recording.c, + res/Makefile, res/ari/ari_model_validators.c, + rest-api/api-docs/recordings.json, res/stasis_recording (added), + res/ari/resource_recordings.c, res/ari/ari_model_validators.h, + res/res_ari_recordings.c: ARI: Implement /recordings/stored API's + his patch implements the ARI API's for stored recordings. While + the original task only specified deleting a recording, it was + simple enough to implement the GET for all recordings, and for an + individual recording. The recording playback operation was + modified to use the same code for accessing the recording as the + REST API, so that they will behave consistently. There were + several problems with the api-docs that were also fixed, bringing + the ARI spec in line with the implementation. There were some + 'wishful thinking' fields on the stored recording model (duration + and timestamp) that were removed, because I ended up not + implementing a metadata file to go along with the recording to + store such information. The GET /recordings/live operation was + removed, since it's not really that useful to get a list of all + recordings that are currently going on in the system. (At least, + if we did that, we'd probably want to also list all of the + current playbacks. Which seems weird.) (closes issue + ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/ + ........ Merged revisions 397985 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /: Multiple revisions 397975-397976 ........ r397975 | rmudgett | + 2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line pbx.c: Make + ast_str_substitute_variables_full() not mask variables. ........ + r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013) + | 1 line Revert last commit. ........ Merged revisions + 397975-397976 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 01:20 +0000 [r397978] Richard Mudgett + + * main/pbx.c, /: pbx.c: Make pbx_substitute_variables_helper_full() + not mask variables. ........ Merged revisions 397977 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-30 00:11 +0000 [r397962-397969] Mark Michelson + + * res/res_pjsip_pidf.c, /: Sanitize XML output for PIDF bodies. + PJSIP's PIDF API does not replace angle brackets with their + appropriate counterparts for XML. So we have to do it ourself. In + this particular case, the problem had to do with attempting to + place an unsanitized SIP URI into an XML node. Now we don't get a + 488 from recipients of our PIDF NOTIFYs. ........ Merged + revisions 397968 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_pidf.c, /: Fix method for creating activities + string in PIDF bodies. The previous method did not allocate + enough space to create the entire string, but adjusted the + string's slen value to be larger than the actual allocation. This + resulted in garbled text in NOTIFY requests from Asterisk. This + method allocates the proper amount of space first and then writes + the content into the buffer. ........ Merged revisions 397960 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 22:49 +0000 [r397959] Kevin Harwell + + * apps/app_dumpchan.c, main/logger.c, apps/app_verbose.c, + main/asterisk.c, channels/chan_misdn.c, /: Verbose logging + discrepancies Refactored cases where a combination of + ast_verbose/options_verbose were present. Also in general tried + to eliminate, in as many places as possible, where the + options_verbose global variable was being used. Refactored the + way local and remote consoles handle verbose message logging in + an attempt to solve the various discrepancies that sometimes + would show between the two. (closes issue AST-1193) Reported by: + Guenther Kelleter Review: + https://reviewboard.asterisk.org/r/2798/ ........ Merged + revisions 397948 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397958 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 22:26 +0000 [r397956-397957] Mark Michelson + + * /, res/res_pjsip_pubsub.c: Fix when the subscription_terminated + callback is called for subscription handlers. The previous + placement would result in the resubscribe() callback called + instead of the subscription_terminated() callback being called + when a subscription was ended via a SUBSCRIBE request. This would + result in confusing PJSIP and having it throw an assertion. + ........ Merged revisions 397955 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * res/res_pjsip_session.c, /: Fix a race condition where a canceled + call was answered. RFC 5407 section 3.1.2 details a scenario + where a UAC sends a CANCEL at the same time that a UAS sends a + 200 OK for the INVITE that the UAC is canceling. When this + occurs, it is the role of the UAC to immediately send a BYE to + terminate the call. This scenario was reproducible by have a + Digium phone with two lines place a call to a second phone that + forwarded the call to the second line on the original phone. The + Digium phone, upon realizing that it was connecting to itself, + would attempt to cancel the call. The timing of this happened to + trigger the aforementioned race condition about 80% of the time. + Asterisk was not doing its job of sending a BYE when receiving a + 200 OK on a cancelled INVITE. The result was that the ast_channel + structure was destroyed but the underlying SIP session, as well + as the PJSIP inv_session and dialog, were still alive. Attempting + to perform an action such as a transfer, once in this state, + would result in Asterisk crashing. The circumstances are now + detected properly and the session is ended as recommended in RFC + 5407. (closes issue AST-1209) reported by John Bigelow ........ + Merged revisions 397945 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 21:37 +0000 [r397947] Kevin Harwell + + * main/file.c, main/app.c, main/config_options.c, main/cel.c, + main/asterisk.c, main/cdr.c, main/manager.c, /, + main/stasis_config.c: Memory leaks fix (closes ASTERISK-22376) + Reported by: John Hardin Patches: memleak.patch uploaded by + jhardin (license 6512) memleak2.patch uploaded by jhardin + (license 6512) ........ Merged revisions 397946 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 20:22 +0000 [r397939] Matthew Jordan + + * configs/safe_asterisk.conf.sample (removed), /, CHANGES, + contrib/scripts/safe_asterisk, Makefile: Revert r394939 due to + (numerous) objections The patch from ASTERISK-21965 was committed + perhaps a bit too hastily. Walter and Tzafrir have pointed out + numerous issues with the approach and have propsed an alternative + in r/2757. Since it's not a time critical issue and is not worth + holding up the release of 12 for it, I've gone ahead and reverted + r394939 from 12/trunk and re-opened ASTERISK-21965. ........ + Merged revisions 397938 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 16:21 +0000 [r397932] David M. Lee + + * rest-api-templates/make_ari_stubs.py, /, + rest-api-templates/api.wiki.mustache, + rest-api-templates/asterisk_processor.py: Account for {} in + Swagger notes ........ Merged revisions 397927 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 16:05 +0000 [r397925] Matthew Jordan + + * Makefile, /: Recursively search for '.c' files when making + documentation with 'make full' Without this, documentation + defined in sub-folders is ignored. Since having properly + generated documentation is especially important in Asterisk 12 - + not having it can cause a module to not load - 'make full' needs + to look in all .c files. ........ Merged revisions 397924 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 15:43 +0000 [r397923] Mark Michelson + + * /, apps/app_queue.c, main/cel.c, main/stasis_bridges.c: Multiple + revisions 397921-397922 ........ r397921 | mmichelson | + 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines Resolve + assumptions that bridge snapshots would be non-NULL for transfer + stasis events. Attempting to transfer an unbridged call would + result in crashes in either CEL code or in the conversion to AMI + messages. ........ r397922 | mmichelson | 2013-08-29 10:42:29 + -0500 (Thu, 29 Aug 2013) | 3 lines Remove extra debug message. + ........ Merged revisions 397921-397922 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-29 12:30 +0000 [r397912] Matthew Jordan + + * contrib/ast-db-manage/config, + contrib/ast-db-manage/config/script.py.mako, + contrib/ast-db-manage/voicemail.ini.sample, + contrib/ast-db-manage/voicemail/env.py, + contrib/ast-db-manage/voicemail, + contrib/ast-db-manage/voicemail/script.py.mako, + contrib/ast-db-manage/README.md, + contrib/ast-db-manage/config/versions, + contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py, + contrib/ast-db-manage (added), + contrib/ast-db-manage/voicemail/versions, /, + contrib/ast-db-manage/config.ini.sample, + contrib/ast-db-manage/config/env.py, + contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py: + Actually *add* the database schema management utilities In + r397874, the scripts were removed... but not replaced. Thanks to + Michael Young for noticing this! ........ Merged revisions 397911 + from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 23:15 +0000 [r397886-397903] Richard Mudgett + + * main/cdr.c, /, funcs/func_cdr.c, main/stdtime/localtime.c: Fix + some uninitialized buffers for CDR handling valgrind found. * + Made ast_strftime_locale() ensure that the output buffer is + initialized. The std library strftime() returns 0 and does not + touch the buffer if it has an error. However, the function can + also return 0 without an error. (closes issue ASTERISK-22412) + Reported by: rmudgett ........ Merged revisions 397902 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Fixed problems with ast_cdr_serialize_variables(). + * Fixed return value of ast_cdr_serialize_variables() on error. + It needs to return 0 indicating no CDR variables found. * Made + ast_cdr_serialize_variables() check the return value of + cdr_object_format_property() and assert if nonzero. A member of + the cdr_readonly_vars[] was not handled. * Removed unused + elements from cdr_readonly_vars[]: total_duration, total_billsec, + first_start, and first_answer. ........ Merged revisions 397900 + from http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Made the on/off in CLI "cdr set debug [on|off]" + case insensitive. ........ Merged revisions 397898 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/cdr.c, /: Make CDR variable name chandling consistently case + insensitive. ........ Merged revisions 397896 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, main/cdr.c: Make CDR code deal with channel names case + insensitively. ........ Merged revisions 397894 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, funcs/func_cdr.c, main/cdr.c: Some CDR code optimization. + ........ Merged revisions 397892 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, funcs/func_cdr.c: Whitespace and curly braces. ........ Merged + revisions 397885 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 21:09 +0000 [r397877] Mark Michelson + + * /, res/res_pjsip_refer.c: Improve detection of answer on SIP + blind transfer. A problem encountered during testing was that + res_pjsip_refer would not ever send a NOTIFY with a 200 OK + sipfrag. This is because the framehook that was supposed to send + the NOTIFY would never be told that an answer had occurred. This + happened for two reasons: 1) The transferee channel on which the + framehook was on was already up. 2) Answers are rarely if ever + written to channels. Rather, the ast_answer() or ast_raw_answer() + function is used to answer channels. Thanks to a suggestion by + Matt Jordan, the best way to detect that the call had been + answered was to find out when the transferee channel joined a + bridge. With stasis this is an easy task. So now, in addition to + the framehook logic, there is a stasis subscription used to + determine when the transferee has entered a bridge. Once it has + entered, an appropriate NOTIFY is sent. ........ Merged revisions + 397876 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 20:55 +0000 [r397872-397875] Matthew Jordan + + * contrib/realtime/mysql/queue_log.sql, + contrib/realtime/mysql/voicemail.sql, + contrib/realtime/mysql/sippeers.sql, /, + contrib/realtime/mysql/iaxfriends.sql, + contrib/realtime/mysql/meetme.sql, + contrib/realtime/mysql/voicemail_messages.sql, + contrib/realtime/postgresql/realtime.sql, + contrib/realtime/mysql/voicemail_data.sql, CHANGES, + contrib/realtime/mysql/musiconhold.sql: Add database schema + management using Alembic This patch replaces contrib/realtime/ + with a new setup for managing the database schema required for + database integration with Asterisk. In addition to initializing a + database with the proper schema, alembic can do a database + migration to assist with upgrading Asterisk in the future. + Hopefully this helps make setting up and operating Asterisk with + a database easier. With this the schema only needs to be + maintained in one place instead of once per database. The schemas + I have added here have a bit of improvement over the examples + that were there before (some added consistency and added some + missing indexes). Managing the schema in one place here also + applies to all databases supported by SQLAlchemy. See + contrib/ast-db-manage/README.md for more details. Review: + https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant + (license 6300) ........ Merged revisions 397874 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * CHANGES, /: Update CHANGES file for Asterisk 12 This updates the + Asterisk 12 CHANGES file with the things that were missed during + the development cycle. Review: + https://reviewboard.asterisk.org/r/2795/ ........ Merged + revisions 397870 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 16:13 +0000 [r397857-397860] Richard Mudgett + + * /, main/pbx.c: pbx.c: Make ast_str_substitute_variables_full() + not mask variables. ........ Merged revisions 397859 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * main/chanvars.c: ast_free() is null tollerant. + + * include/asterisk/threadstorage.h, /: Match use of ast_free() with + ast_calloc() and add some curly braces. ........ Merged revisions + 397856 from http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-28 15:43 +0000 [r397855] Mark Michelson + + * res/res_pjsip/pjsip_distributor.c, /: Fix dialog matching in the + SIP distributor. Dialog matching is performed in the distributor + for the sole purpose of retrieving an associated serializer so + the request may be serialized. This patch fixes two problems. + First, incoming CANCEL requests that had no to-tag (which really + should be *all* CANCEL requests) would not match with a dialog. + An earlier bug fix to deal with early CANCEL requests would + result in the CANCEL being replied to with a 481. The fix for + this is to find the matching INVITE transaction and get the + dialog from that transaction. Second, no SIP responses were + matching dialogs. This is because we were inverting the tags that + we were passing into PJSIP's dialog finding function. This logic + has been corrected by setting local and remote tag variables + based on whether the incoming message is a request or response. + ........ Merged revisions 397854 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 19:19 +0000 [r397820] David M. Lee + + * rest-api-templates/param_parsing.mustache, res/res_ari_bridges.c, + /, res/stasis/app.c, res/res_ari_events.c, + res/res_ari_asterisk.c, + rest-api-templates/res_ari_resource.c.mustache, res/stasis/app.h, + res/res_stasis.c, main/stasis_bridges.c: ARI: WebSocket event + cleanup Stasis events (which get distributed over the ARI + WebSocket) are created by subscribing to the channel_all_cached + and bridge_all_cached topics, filtering out events for + channels/bridges currently subscribed to. There are two issues + with that. First was a race condition, where messages in-flight + to the master subscribe-to-all-things topic would get sent out, + even though the events happened before the channel was put into + Stasis. Secondly, as the number of channels and bridges grow in + the system, the work spent filtering messages becomes excessive. + Since r395954, individual channels and bridges have caching + topics, and can be subscribed to individually. This patch takes + advantage, so that channels and bridges are subscribed to on + demand, instead of filtering the global topics. The one case + where filtering is still required is handling BridgeMerge + messages, which are published directly to the bridge_all topic. + Other than the change to how subscriptions work, this patch + mostly just moves code around. Most of the work generating JSON + objects from messages was moved to .to_json handlers on the + message types. The callback functions handling app subscriptions + were moved from res_stasis (b/c they were global to the model) to + stasis/app.c (b/c they are local to the app now). (closes issue + ASTERISK-21969) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/2754/ ........ Merged + revisions 397816 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 18:52 +0000 [r397811] Richard Mudgett + + * /, main/astmm.c: Made MALLOC_DEBUG less CPU intensive by default. + Storing a backtrace for each allocation in anticipation of a + memory management problem is very CPU intensive. * Added the CLI + "memory backtrace {on|off}" command to request that the backtrace + be gathered only on request. The backtrace is off by default. + (issue ASTERISK-22221) Reported by: Matt Jordan ........ Merged + revisions 397809 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 18:10 +0000 [r397753-397760] Matthew Jordan + + * /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid + SDP If the SIP channel driver processes an invalid SDP that + defines media descriptions before connection information, it may + attempt to reference the socket address information even though + that information has not yet been set. This will cause a crash. + This patch adds checks when handling the various media + descriptions that ensures the media descriptions are handled only + if we have connection information suitable for that media. Thanks + to Walter Doekes, OSSO B.V., for reporting, testing, and + providing the solution to this problem. (closes issue + ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches: + issueA22007_sdp_without_c_death.patch uploaded by wdoekes + (License 5674) ........ Merged revisions 397756 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 397757 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged + revisions 397758 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397759 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK + on dialog that has no channel A remote exploitable crash + vulnerability exists in the SIP channel driver if an ACK with SDP + is received after the channel has been terminated. The handling + code incorrectly assumed that the channel would always be + present. This patch adds a check such that the SDP will only be + parsed and applied if Asterisk has a channel present that is + associated with the dialog. Note that the patch being applied was + modified only slightly from the patch provided by Walter Doekes + of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin + Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches: + issueA21064_fix.patch uploaded by wdoekes (License 5674) ........ + Merged revisions 397710 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 397711 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged + revisions 397712 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397713 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-27 16:51 +0000 [r397746] Richard Mudgett + + * channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c, + channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, channels/chan_motif.c: Fix uninitialized + value in struct ast_control_pvt_cause_code usage. ........ Merged + revisions 397744 from + http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged + revisions 397745 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-26 23:48 +0000 [r397691] Matthew Jordan + + * /, main/bridge_channel.c: Better handle clearing the OUTGOING + flag when a channel leaves a bridge When a channel with the + OUTGOING flag leaves a bridge, and it will survive being pulled + from the bridge (either because it will execute dialplan, go into + another bridge, or live in a friendly autoloop), we have to clear + the OUTGOING flag. This is the signal to the CDR engine that this + channel is no longer a second class citizen, i.e., it is not + "dialed". The soft hangup flags are only half the picture. If a + channel is being moved from one bridge to another, the soft + hangup flags aren't set; however, the state of the bridge_channel + will not be hung up. Since the channel does not have one of the + two hang up states, that implies that the channel is still + technically alive. This patch modifies the check so that it + checks both the soft hangup flags as well as the bridge_channel + state. If either suggests that the channel is going to persist, + we clear the OUTGOING flag. ........ Merged revisions 397690 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-26 21:32 +0000 [r397674] David M. Lee + + * /, main/bucket.c: Fixed bucket.c for systems where tv_usec is not + an unsigned long. ........ Merged revisions 397673 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-26 16:25 +0000 [r397644-397651] Richard Mudgett + + * /, include/asterisk/bridge_channel.h, main/bridge_channel.c: + bridging: Fix a livelock with local channel optimization. Use a + better means of waking up the bridge channel thread. ........ + Merged revisions 397650 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * channels/Makefile, /: chan_dahdi: Add some missing build cleanup. + ........ Merged revisions 397643 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-25 18:12 +0000 [r397622-397631] Matthew Jordan + + * tests/test_bucket.c, /: Fix bucket unit tests After the review + for buckets was completed (r2715), the handling of names in the + bucket core was deferred to the wizards. As such, the bucket unit + tests cannot expect that passing a URI with a scheme specified + but no actual resource name will automatically fail. The tests + have been updated to not make this check. ........ Merged + revisions 397630 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * include/asterisk/config_options.h, /, main/config_options.c, + tests/test_config.c: Fix the config_options_test The config + options test requires the entire configuration item to be + transparent from the documentation system. So we let it do that + too. As an aside, please do not use this power for evil. + Documentation is your friend, and you really should document your + configurations. Hiding your module's configuration information + from the system attempting to enforce some sanity in the universe + is something only a Bond villain would contemplate. ........ + Merged revisions 397628 from + http://svn.asterisk.org/svn/asterisk/branches/12 + + * /, res/res_pjsip/pjsip_configuration.c: Add rtpengine + configuration parameter The rtpengine configuration parameter was + documented in the XML documentation, but it was not actually + registered with the sorcery object. This adds the parameter with + a default of "asterisk", such that res_rtp_asterisk is chosen as + the default RTP implementation. (closes issue ASTERISK-22380) + Reported by: Rusty Newton Tested by: Rusty Newton ........ Merged + revisions 397621 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-08-23 22:40 +0000 [r397615] Matthew Jordan + + * /: Set new merge properties on 12 + +2013-08-23 22:20 +0000 [r397613] Joshua Colp + + * main/bucket.c: Fix building of trunk. Note: This is why I commit + on the weekend. + diff --git a/asterisk-20.2.0-rc1-summary.html b/asterisk-20.2.0-rc1-summary.html new file mode 100644 index 0000000000..904683d6cc --- /dev/null +++ b/asterisk-20.2.0-rc1-summary.html @@ -0,0 +1,191 @@ +Release Summary - asterisk-20.2.0-rc1

Release Summary

asterisk-20.2.0-rc1

Date: 2023-03-02

<asteriskteam@digium.com>

Table of Contents

    +
  1. Summary
    2. +
  2. Contributors
    3. +
  3. Closed Issues
    4. +
  4. Other Changes
    5. +
  5. Diffstat
    6. +

Summary

[Back to Top]

This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.

The data in this summary reflects changes that have been made since the previous release, asterisk-20.1.0.

Contributors

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This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.

+ + +
CodersTestersReporters
14 Naveen Albert
7 Sean Bright
6 George Joseph
6 Mike Bradeen
2 Igor Goncharovsky
1 Peter Fern
1 Holger Hans Peter Freyther
1 Boris P. Korzun
1 Ben Ford
1 Alexei Gradinari
1 cmaj
1 Asterisk Development Team
1 Nick French
1 sungtae kim
11 N A
5 Michael Bradeen
3 Sean Bright
3 George Joseph
1 Sebastian Gutierrez
1 Jaco Kroon
1 Yury Kirsanov
1 Benjamin Keith Ford
1 Nick French
1 AvayaXAsterisk
1 Ross Beer
1 Igor Goncharovsky
1 Boris P. Korzun
1 Joshua C. Colp
1 Stanislav Abramenkov
1 Julien Alie
1 Oleg
1 cmaj
1 Danila Evgrafov
1 Sebastian Gutierrez
1 Julien Alie
1 Yury Kirsanov

Closed Issues

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This is a list of all issues from the issue tracker that were closed by changes that went into this release.

New Feature

Category: Applications/NewFeature

ASTERISK-29810: app_signal: Add channel signaling applications
Reported by: N A
    +
  • [88b2c741ca] Naveen Albert -- app_signal: Add signaling applications
    • +
ASTERISK-30180: app_broadcast: Add a channel audio multicasting application
Reported by: N A
    +
  • [e06fe8e344] Naveen Albert -- app_broadcast: Add Broadcast application
    • +

Category: Resources/res_pjsip_rfc3326

ASTERISK-30319: Add BYE Reason support for SIP
Reported by: Igor Goncharovsky
    +
  • [3526441e41] Igor Goncharovsky -- res_pjsip_rfc3326: Add SIP causes support for RFC3326
    • +

Category: Resources/res_pjsip_session

ASTERISK-30262: res_pjsip_session: Allow a context to be specified for overlap dialing
Reported by: N A
    +
  • [a1da8042d1] Naveen Albert -- res_pjsip_session: Add overlap_context option.
    • +

Bug

Category: Applications/app_mixmonitor

ASTERISK-30198: Error `Too many open files` occurs after about ~8000 calls when using mixmonitor
Reported by: Julien Alie
    +
  • [58404b5c22] Peter Fern -- streams: Ensure that stream is closed in ast_stream_and_wait on error
    • +

Category: Applications/app_queue

ASTERISK-30417: Copy/Paste error in UnpauseQueueMember
Reported by: Sean Bright
    +
  • [aeb16aa7d8] Sean Bright -- app_queue: Minor docs and logging fixes for UnpauseQueueMember.
    • +

Category: Applications/app_stasis

ASTERISK-29604: ari: Segfault with lots of calls
Reported by: Danila Evgrafov
    +
  • [f99849f8d5] sungtae kim -- res_stasis_snoop: Fix snoop crash
    • +

Category: Applications/app_voicemail/ODBC

ASTERISK-30240: app voicemail odbc build error with gcc 11.1
Reported by: Michael Bradeen
    +
  • [20d4775d0a] Naveen Albert -- app_voicemail_odbc: Fix string overflow warning.
    • +

Category: Channels/chan_iax2

ASTERISK-30354: chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall
Reported by: N A
    +
  • [ede67a99be] Naveen Albert -- chan_iax2: Fix jitterbuffer regression prior to receiving audio.
    • +
ASTERISK-30162: when chan_iax is used to relay calls, no ringing indication is played
Reported by: Jaco Kroon
    +
  • [ede67a99be] Naveen Albert -- chan_iax2: Fix jitterbuffer regression prior to receiving audio.
    • +

Category: Channels/chan_pjsip

ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late
Reported by: Oleg
    +
  • [c7598ee947] Naveen Albert -- res_pjsip_session: Use Caller ID for extension matching.
    • +

Category: Channels/chan_sip/General

ASTERISK-29604: ari: Segfault with lots of calls
Reported by: Danila Evgrafov
    +
  • [f99849f8d5] sungtae kim -- res_stasis_snoop: Fix snoop crash
    • +

Category: Core/BuildSystem

ASTERISK-27830: Asterisk crashes on Invalid UTF-8 string
Reported by: AvayaXAsterisk
    +
  • [ceda5a9859] George Joseph -- res_pjsip: Replace invalid UTF-8 sequences in callerid name
    • +

Category: Core/General

ASTERISK-30345: loader.c: Modules that decline to load cannot be reloaded
Reported by: N A
    +
  • [d33bd6d67a] Naveen Albert -- loader: Allow declined modules to be unloaded.
    • +

Category: Core/HTTP

ASTERISK-30379: http: fix NULL pointer dereference while enable_status on TLS-only
Reported by: Boris P. Korzun
    +
  • [edc90c96ac] Boris P. Korzun -- http.c: Fix NULL pointer dereference bug
    • +

Category: Core/ManagerInterface

ASTERISK-30351: manager: Originate variables are not added when setvar used in manager.conf
Reported by: Sebastian Gutierrez
    +
  • [7b8f7428da] Naveen Albert -- manager: Fix appending variables.
    • +

Category: Core/PBX

ASTERISK-30367: pbx: Fix outdated channel snapshots with pbx_exec
Reported by: N A
    +
  • [cc8d9b947b] Naveen Albert -- pbx_app: Update outdated pbx_exec channel snapshots.
    • +

Category: Documentation

ASTERISK-30347: xmldocs: Remove references to removed applications
Reported by: N A
    +
  • [36bea9ad33] Naveen Albert -- app_sendtext: Remove references to removed applications.
    • +

Category: PBX/pbx_ael

ASTERISK-30406: pbx_ael: Global variables are not expanded.
Reported by: Sean Bright
    +
  • [56051d1ac5] Sean Bright -- pbx_ael: Global variables are not expanded.
    • +

Category: Resources/res_http_media_cache

ASTERISK-30375: res_http_media_cache: Crash when URL has no path component.
Reported by: Sean Bright
    +
  • [3d9b9a2b16] Holger Hans Peter Freyther -- res_http_media_cache: Do not crash when there is no extension
    • +

Category: Resources/res_phoneprov

ASTERISK-30388: res_phoneprov: Stale SERVER variable when multi-homed
Reported by: cmaj
    +
  • [5b0e3444c3] cmaj -- res_phoneprov.c: Multihomed SERVER cache prevention
    • +

Category: Resources/res_pjsip

ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they shouldn't be
Reported by: Joshua C. Colp
    +
  • [24102ba236] George Joseph -- res_pjsip_transport_websocket: Add remote port to transport
    • +
ASTERISK-30100: res_pjsip: Path is ignored on INVITE to endpoint
Reported by: Yury Kirsanov
    +
  • [115a1b4f0a] Igor Goncharovsky -- res_pjsip: Fix path usage in case dialing with '@'
    • +

Category: Resources/res_pjsip_caller_id

ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late
Reported by: Oleg
    +
  • [c7598ee947] Naveen Albert -- res_pjsip_session: Use Caller ID for extension matching.
    • +

Category: Resources/res_pjsip_pubsub

ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13
Reported by: Ross Beer
    +
  • [37e558f6ef] Mike Bradeen -- res_pjsip: Prevent SEGV in pjsip_evsub_send_request
    • +

Category: Resources/res_pjsip_sdp_rtp

ASTERISK-30350: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold
Reported by: Benjamin Keith Ford
    +
  • [881faf544f] Ben Ford -- res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
    • +

Category: Resources/res_rtp_asterisk

ASTERISK-30391: res_rtp_asterisk: Issue with transcoding g722 after MES changes
Reported by: George Joseph
    +
  • [2f5aece0c9] George Joseph -- res_rtp_asterisk: Don't use double math to generate timestamps
    • +
  • [4710f37ef6] George Joseph -- res_rtp_asterisk: Asterisk Media Experience Score (MES)
    • +

Category: pjproject/pjsip

ASTERISK-30424: pjproject_bundled: cross-compilation broken when ssl autodetected
Reported by: Nick French
    +
  • [200dc7d0e8] Nick French -- pjproject_bundled: Fix cross-compilation with SSL libs.
    • +
ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13
Reported by: Ross Beer
    +
  • [37e558f6ef] Mike Bradeen -- res_pjsip: Prevent SEGV in pjsip_evsub_send_request
    • +

Improvement

Category: Applications/app_directory

ASTERISK-30405: app_directory: Add 's' option to skip channel call
Reported by: Michael Bradeen
    +
  • [2308afed8e] Mike Bradeen -- app_directory: Add a 'skip call' option.
    • +
ASTERISK-30404: app_directory: Add reading directory configuration from custom file
Reported by: Michael Bradeen
    +
  • [70856e865f] Mike Bradeen -- app_directory: add ability to specify configuration file
    • +

Category: Applications/app_read

ASTERISK-30411: app_read: add option to include terminating digit on empty, terminated strings
Reported by: Michael Bradeen
    +
  • [5c11d7adea] Mike Bradeen -- app_read: Add an option to return terminator on empty digits.
    • +

Category: Applications/app_senddtmf

ASTERISK-30422: app_senddtmf: add the option for senddtmf to answer
Reported by: Michael Bradeen
    +
  • [98742388b6] Mike Bradeen -- app_senddtmf: Add option to answer target channel.
    • +

Category: Core/General

ASTERISK-30361: json.h: Add missing ast_json_object_real_get
Reported by: N A
    +
  • [3b3fef2347] Naveen Albert -- json.h: Add ast_json_object_real_get.
    • +

Category: Functions/General

ASTERISK-29913: func_json: Adds multi-level and array parsing to JSON_DECODE
Reported by: N A
    +
  • [8a45cd7af4] Naveen Albert -- func_json: Enhance parsing capabilities of JSON_DECODE
    • +
ASTERISK-30353: func_frame_trace: Print text for text frames
Reported by: N A
    +
  • [68e345286b] Naveen Albert -- func_frame_trace: Print text for text frames.
    • +

Category: Functions/func_callerid

ASTERISK-30332: func_callerid: Warn if invalid redirecting reason provided
Reported by: N A
    +
  • [cbb1fd2cb9] Naveen Albert -- func_callerid: Warn about invalid redirecting reason.
    • +

Category: Resources/res_pjsip

ASTERISK-30325: Upgrade Asterisk to bundled pjproject 2.13
Reported by: Stanislav Abramenkov
    +
  • [58636a6ea6] Mike Bradeen -- res_pjsip: Upgraded bundled pjsip to 2.13
    • +

Category: Resources/res_rtp_asterisk

ASTERISK-30280: Create capability to assign a Media Experience Score to RTP streams
Reported by: George Joseph
    +
  • [d454801c2d] George Joseph -- res_rtp_asterisk: Asterisk Media Experience Score (MES)
    • +

Commits Not Associated with an Issue

[Back to Top]

This is a list of all changes that went into this release that did not reference a JIRA issue.

+ + + + + + + + + +
RevisionAuthorSummary
93813c9dcaAsterisk Development TeamUpdate CHANGES and UPGRADE.txt for 20.2.0
e5c5cd6e25Sean Brighttest.c: Avoid passing -1 to FD_* family of functions.
827222d607Sean Brighttest_crypto.c: Fix getcwd(…) build error.
aef0c0ce0eSean Brightapp_queue: Reset all queue defaults before reload.
96d9ad51acSean Brightdoxygen: Fix doxygen errors.
ef16eaee36Sean Brightapp_playback.c: Fix PLAYBACKSTATUS regression.
e86d5d7fdaAlexei Gradinariformat_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)
62ca063fcaGeorge JosephRevert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"

Diffstat Results

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This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.

.lastclean                                                                                                                           |    1
+.version                                                                                                                             |    1
+ChangeLog                                                                                                                            |105422 ----------
+asterisk-20.1.0-summary.html                                                                                                         |  279
+asterisk-20.1.0-summary.txt                                                                                                          |  768
+b/CHANGES                                                                                                                            |   66
+b/UPGRADE.txt                                                                                                                        |   13
+b/apps/app_broadcast.c                                                                                                               |  619
+b/apps/app_directory.c                                                                                                               |   36
+b/apps/app_mf.c                                                                                                                      |    1
+b/apps/app_playback.c                                                                                                                |    3
+b/apps/app_queue.c                                                                                                                   |   14
+b/apps/app_read.c                                                                                                                    |   23
+b/apps/app_senddtmf.c                                                                                                                |   31
+b/apps/app_sendtext.c                                                                                                                |    4
+b/apps/app_signal.c                                                                                                                  |  471
+b/apps/app_voicemail.c                                                                                                               |    7
+b/channels/chan_iax2.c                                                                                                               |   17
+b/channels/chan_pjsip.c                                                                                                              |  114
+b/channels/pjsip/dialplan_functions.c                                                                                                |   67
+b/configs/samples/pjsip.conf.sample                                                                                                  |    2
+b/configs/samples/queues.conf.sample                                                                                                 |   10
+b/configure                                                                                                                          |18025 -
+b/configure.ac                                                                                                                       |   13
+b/contrib/ast-db-manage/config/versions/f261363a857f_add_overlap_context.py                                                          |   21
+b/formats/format_wav.c                                                                                                               |    2
+b/funcs/func_callerid.c                                                                                                              |    1
+b/funcs/func_frame_trace.c                                                                                                           |    1
+b/funcs/func_json.c                                                                                                                  |  232
+b/include/asterisk/autoconfig.h.in                                                                                                   |  124
+b/include/asterisk/channel.h                                                                                                         |    4
+b/include/asterisk/crypto.h                                                                                                          |   12
+b/include/asterisk/file.h                                                                                                            |    1
+b/include/asterisk/json.h                                                                                                            |    9
+b/include/asterisk/pbx.h                                                                                                             |    6
+b/include/asterisk/res_aeap.h                                                                                                        |    8
+b/include/asterisk/res_aeap_message.h                                                                                                |    3
+b/include/asterisk/res_geolocation.h                                                                                                 |    4
+b/include/asterisk/res_pjsip.h                                                                                                       |   66
+b/include/asterisk/res_stir_shaken.h                                                                                                 |    2
+b/include/asterisk/rtp_engine.h                                                                                                      |   54
+b/include/asterisk/time.h                                                                                                            |   88
+b/include/asterisk/utf8.h                                                                                                            |   53
+b/include/asterisk/xml.h                                                                                                             |   18
+b/main/bridge_basic.c                                                                                                                |    2
+b/main/file.c                                                                                                                        |    4
+b/main/http.c                                                                                                                        |   10
+b/main/loader.c                                                                                                                      |   25
+b/main/manager.c                                                                                                                     |    6
+b/main/pbx_app.c                                                                                                                     |    2
+b/main/rtp_engine.c                                                                                                                  |   74
+b/main/stasis_channels.c                                                                                                             |   33
+b/main/test.c                                                                                                                        |   13
+b/main/utf8.c                                                                                                                        |  544
+b/menuselect/autoconfig.h.in                                                                                                         |   22
+b/menuselect/configure                                                                                                               | 3476
+b/res/ael/pval.c                                                                                                                     |   14
+b/res/res_aeap/transaction.h                                                                                                         |    4
+b/res/res_aeap/transport.h                                                                                                           |    2
+b/res/res_geolocation/geoloc_eprofile.c                                                                                              |   14
+b/res/res_http_media_cache.c                                                                                                         |    9
+b/res/res_phoneprov.c                                                                                                                |   20
+b/res/res_pjsip.c                                                                                                                    |  337
+b/res/res_pjsip/pjsip_config.xml                                                                                                     |   10
+b/res/res_pjsip/pjsip_configuration.c                                                                                                |    5
+b/res/res_pjsip/pjsip_manager.xml                                                                                                    |    3
+b/res/res_pjsip/pjsip_transport_events.c                                                                                             |    2
+b/res/res_pjsip_caller_id.c                                                                                                          |  227
+b/res/res_pjsip_path.c                                                                                                               |   73
+b/res/res_pjsip_pubsub.c                                                                                                             |  101
+b/res/res_pjsip_rfc3326.c                                                                                                            |   31
+b/res/res_pjsip_sdp_rtp.c                                                                                                            |    4
+b/res/res_pjsip_session.c                                                                                                            |   24
+b/res/res_rtp_asterisk.c                                                                                                             |  547
+b/res/res_speech_aeap.c                                                                                                              |   51
+b/res/res_stasis_snoop.c                                                                                                             |   10
+b/res/res_stir_shaken.c                                                                                                              |    2
+b/tests/test_crypto.c                                                                                                                |   32
+b/tests/test_res_rtp.c                                                                                                               |  189
+b/third-party/pjproject/configure.m4                                                                                                 |    7
+b/third-party/pjproject/patches/0000-remove-third-party.patch                                                                        |    6
+b/third-party/pjproject/patches/0010-Make-sure-that-NOTIFY-tdata-is-set-before-sending-it_new-129fb323a66dd1fd16880fe5ba5e6a57.patch |   46
+contrib/realtime/mysql/mysql_cdr.sql                                                                                                 |   41
+contrib/realtime/mysql/mysql_config.sql                                                                                              | 1402
+contrib/realtime/mysql/mysql_voicemail.sql                                                                                           |   35
+contrib/realtime/postgresql/postgresql_cdr.sql                                                                                       |   45
+contrib/realtime/postgresql/postgresql_config.sql                                                                                    | 1524
+contrib/realtime/postgresql/postgresql_voicemail.sql                                                                                 |   39
+third-party/pjproject/patches/0100-allow_multiple_auth_headers.patch                                                                 |  413
+third-party/pjproject/patches/0200-potential-buffer-overflow-in-pjlib-scanner-and-pjmedia.patch                                      |  306
+third-party/pjproject/patches/0201-potential-stack-buffer-overflow-when-parsing-message-as-a-STUN-client.patch                       |   44
+third-party/pjproject/pjproject-2.12.1.tar.bz2.md5                                                                                   |    1
+92 files changed, 13653 insertions(+), 122894 deletions(-)

\ No newline at end of file diff --git a/asterisk-20.2.0-rc1-summary.txt b/asterisk-20.2.0-rc1-summary.txt new file mode 100644 index 0000000000..4480bccc94 --- /dev/null +++ b/asterisk-20.2.0-rc1-summary.txt @@ -0,0 +1,496 @@ + Release Summary + + asterisk-20.2.0-rc1 + + Date: 2023-03-02 + + + + ---------------------------------------------------------------------- + + Table of Contents + + 1. Summary + 2. Contributors + 3. Closed Issues + 4. Other Changes + 5. Diffstat + + ---------------------------------------------------------------------- + + Summary + + [Back to Top] + + This release is a point release of an existing major version. The changes + included were made to address problems that have been identified in this + release series, or are minor, backwards compatible new features or + improvements. Users should be able to safely upgrade to this version if + this release series is already in use. Users considering upgrading from a + previous version are strongly encouraged to review the UPGRADE.txt + document as well as the CHANGES document for information about upgrading + to this release series. + + The data in this summary reflects changes that have been made since the + previous release, asterisk-20.1.0. + + ---------------------------------------------------------------------- + + Contributors + + [Back to Top] + + This table lists the people who have submitted code, those that have + tested patches, as well as those that reported issues on the issue tracker + that were resolved in this release. For coders, the number is how many of + their patches (of any size) were committed into this release. For testers, + the number is the number of times their name was listed as assisting with + testing a patch. Finally, for reporters, the number is the number of + issues that they reported that were affected by commits that went into + this release. + + Coders Testers Reporters + 14 Naveen Albert 11 N A + 7 Sean Bright 5 Michael Bradeen + 6 George Joseph 3 Sean Bright + 6 Mike Bradeen 3 George Joseph + 2 Igor Goncharovsky 1 Sebastian Gutierrez + 1 Peter Fern 1 Jaco Kroon + 1 Holger Hans Peter Freyther 1 Yury Kirsanov + 1 Boris P. Korzun 1 Benjamin Keith Ford + 1 Ben Ford 1 Nick French + 1 Alexei Gradinari 1 AvayaXAsterisk + 1 cmaj 1 Ross Beer + 1 Asterisk Development Team 1 Igor Goncharovsky + 1 Nick French 1 Boris P. Korzun + 1 sungtae kim 1 Joshua C. Colp + 1 Stanislav Abramenkov + 1 Julien Alie + 1 Oleg + 1 cmaj + 1 Danila Evgrafov + 1 Sebastian Gutierrez + 1 Julien Alie + 1 Yury Kirsanov + + ---------------------------------------------------------------------- + + Closed Issues + + [Back to Top] + + This is a list of all issues from the issue tracker that were closed by + changes that went into this release. + + New Feature + + Category: Applications/NewFeature + + ASTERISK-29810: app_signal: Add channel signaling applications + Reported by: N A + * [88b2c741ca] Naveen Albert -- app_signal: Add signaling applications + ASTERISK-30180: app_broadcast: Add a channel audio multicasting + application + Reported by: N A + * [e06fe8e344] Naveen Albert -- app_broadcast: Add Broadcast application + + Category: Resources/res_pjsip_rfc3326 + + ASTERISK-30319: Add BYE Reason support for SIP + Reported by: Igor Goncharovsky + * [3526441e41] Igor Goncharovsky -- res_pjsip_rfc3326: Add SIP causes + support for RFC3326 + + Category: Resources/res_pjsip_session + + ASTERISK-30262: res_pjsip_session: Allow a context to be specified for + overlap dialing + Reported by: N A + * [a1da8042d1] Naveen Albert -- res_pjsip_session: Add overlap_context + option. + + Bug + + Category: Applications/app_mixmonitor + + ASTERISK-30198: Error `Too many open files` occurs after about ~8000 calls + when using mixmonitor + Reported by: Julien Alie + * [58404b5c22] Peter Fern -- streams: Ensure that stream is closed in + ast_stream_and_wait on error + + Category: Applications/app_queue + + ASTERISK-30417: Copy/Paste error in UnpauseQueueMember + Reported by: Sean Bright + * [aeb16aa7d8] Sean Bright -- app_queue: Minor docs and logging fixes + for UnpauseQueueMember. + + Category: Applications/app_stasis + + ASTERISK-29604: ari: Segfault with lots of calls + Reported by: Danila Evgrafov + * [f99849f8d5] sungtae kim -- res_stasis_snoop: Fix snoop crash + + Category: Applications/app_voicemail/ODBC + + ASTERISK-30240: app voicemail odbc build error with gcc 11.1 + Reported by: Michael Bradeen + * [20d4775d0a] Naveen Albert -- app_voicemail_odbc: Fix string overflow + warning. + + Category: Channels/chan_iax2 + + ASTERISK-30354: chan_iax2: Lack of formats prior to receiving voice frames + causes jitterbuffer to stall + Reported by: N A + * [ede67a99be] Naveen Albert -- chan_iax2: Fix jitterbuffer regression + prior to receiving audio. + ASTERISK-30162: when chan_iax is used to relay calls, no ringing + indication is played + Reported by: Jaco Kroon + * [ede67a99be] Naveen Albert -- chan_iax2: Fix jitterbuffer regression + prior to receiving audio. + + Category: Channels/chan_pjsip + + ASTERISK-28767: chan_pjsip: Caller ID not used when checking for + extension, callerid supplement executed too late + Reported by: Oleg + * [c7598ee947] Naveen Albert -- res_pjsip_session: Use Caller ID for + extension matching. + + Category: Channels/chan_sip/General + + ASTERISK-29604: ari: Segfault with lots of calls + Reported by: Danila Evgrafov + * [f99849f8d5] sungtae kim -- res_stasis_snoop: Fix snoop crash + + Category: Core/BuildSystem + + ASTERISK-27830: Asterisk crashes on Invalid UTF-8 string + Reported by: AvayaXAsterisk + * [ceda5a9859] George Joseph -- res_pjsip: Replace invalid UTF-8 + sequences in callerid name + + Category: Core/General + + ASTERISK-30345: loader.c: Modules that decline to load cannot be reloaded + Reported by: N A + * [d33bd6d67a] Naveen Albert -- loader: Allow declined modules to be + unloaded. + + Category: Core/HTTP + + ASTERISK-30379: http: fix NULL pointer dereference while enable_status on + TLS-only + Reported by: Boris P. Korzun + * [edc90c96ac] Boris P. Korzun -- http.c: Fix NULL pointer dereference + bug + + Category: Core/ManagerInterface + + ASTERISK-30351: manager: Originate variables are not added when setvar + used in manager.conf + Reported by: Sebastian Gutierrez + * [7b8f7428da] Naveen Albert -- manager: Fix appending variables. + + Category: Core/PBX + + ASTERISK-30367: pbx: Fix outdated channel snapshots with pbx_exec + Reported by: N A + * [cc8d9b947b] Naveen Albert -- pbx_app: Update outdated pbx_exec + channel snapshots. + + Category: Documentation + + ASTERISK-30347: xmldocs: Remove references to removed applications + Reported by: N A + * [36bea9ad33] Naveen Albert -- app_sendtext: Remove references to + removed applications. + + Category: PBX/pbx_ael + + ASTERISK-30406: pbx_ael: Global variables are not expanded. + Reported by: Sean Bright + * [56051d1ac5] Sean Bright -- pbx_ael: Global variables are not + expanded. + + Category: Resources/res_http_media_cache + + ASTERISK-30375: res_http_media_cache: Crash when URL has no path + component. + Reported by: Sean Bright + * [3d9b9a2b16] Holger Hans Peter Freyther -- res_http_media_cache: Do + not crash when there is no extension + + Category: Resources/res_phoneprov + + ASTERISK-30388: res_phoneprov: Stale SERVER variable when multi-homed + Reported by: cmaj + * [5b0e3444c3] cmaj -- res_phoneprov.c: Multihomed SERVER cache + prevention + + Category: Resources/res_pjsip + + ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they + shouldn't be + Reported by: Joshua C. Colp + * [24102ba236] George Joseph -- res_pjsip_transport_websocket: Add + remote port to transport + ASTERISK-30100: res_pjsip: Path is ignored on INVITE to endpoint + Reported by: Yury Kirsanov + * [115a1b4f0a] Igor Goncharovsky -- res_pjsip: Fix path usage in case + dialing with '@' + + Category: Resources/res_pjsip_caller_id + + ASTERISK-28767: chan_pjsip: Caller ID not used when checking for + extension, callerid supplement executed too late + Reported by: Oleg + * [c7598ee947] Naveen Albert -- res_pjsip_session: Use Caller ID for + extension matching. + + Category: Resources/res_pjsip_pubsub + + ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13 + Reported by: Ross Beer + * [37e558f6ef] Mike Bradeen -- res_pjsip: Prevent SEGV in + pjsip_evsub_send_request + + Category: Resources/res_pjsip_sdp_rtp + + ASTERISK-30350: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when + moh_passthrough has call on hold + Reported by: Benjamin Keith Ford + * [881faf544f] Ben Ford -- res_pjsip_sdp_rtp.c: Use correct timeout when + put on hold. + + Category: Resources/res_rtp_asterisk + + ASTERISK-30391: res_rtp_asterisk: Issue with transcoding g722 after MES + changes + Reported by: George Joseph + * [2f5aece0c9] George Joseph -- res_rtp_asterisk: Don't use double math + to generate timestamps + * [4710f37ef6] George Joseph -- res_rtp_asterisk: Asterisk Media + Experience Score (MES) + + Category: pjproject/pjsip + + ASTERISK-30424: pjproject_bundled: cross-compilation broken when ssl + autodetected + Reported by: Nick French + * [200dc7d0e8] Nick French -- pjproject_bundled: Fix cross-compilation + with SSL libs. + ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13 + Reported by: Ross Beer + * [37e558f6ef] Mike Bradeen -- res_pjsip: Prevent SEGV in + pjsip_evsub_send_request + + Improvement + + Category: Applications/app_directory + + ASTERISK-30405: app_directory: Add 's' option to skip channel call + Reported by: Michael Bradeen + * [2308afed8e] Mike Bradeen -- app_directory: Add a 'skip call' option. + ASTERISK-30404: app_directory: Add reading directory configuration from + custom file + Reported by: Michael Bradeen + * [70856e865f] Mike Bradeen -- app_directory: add ability to specify + configuration file + + Category: Applications/app_read + + ASTERISK-30411: app_read: add option to include terminating digit on + empty, terminated strings + Reported by: Michael Bradeen + * [5c11d7adea] Mike Bradeen -- app_read: Add an option to return + terminator on empty digits. + + Category: Applications/app_senddtmf + + ASTERISK-30422: app_senddtmf: add the option for senddtmf to answer + Reported by: Michael Bradeen + * [98742388b6] Mike Bradeen -- app_senddtmf: Add option to answer target + channel. + + Category: Core/General + + ASTERISK-30361: json.h: Add missing ast_json_object_real_get + Reported by: N A + * [3b3fef2347] Naveen Albert -- json.h: Add ast_json_object_real_get. + + Category: Functions/General + + ASTERISK-29913: func_json: Adds multi-level and array parsing to + JSON_DECODE + Reported by: N A + * [8a45cd7af4] Naveen Albert -- func_json: Enhance parsing capabilities + of JSON_DECODE + ASTERISK-30353: func_frame_trace: Print text for text frames + Reported by: N A + * [68e345286b] Naveen Albert -- func_frame_trace: Print text for text + frames. + + Category: Functions/func_callerid + + ASTERISK-30332: func_callerid: Warn if invalid redirecting reason provided + Reported by: N A + * [cbb1fd2cb9] Naveen Albert -- func_callerid: Warn about invalid + redirecting reason. + + Category: Resources/res_pjsip + + ASTERISK-30325: Upgrade Asterisk to bundled pjproject 2.13 + Reported by: Stanislav Abramenkov + * [58636a6ea6] Mike Bradeen -- res_pjsip: Upgraded bundled pjsip to 2.13 + + Category: Resources/res_rtp_asterisk + + ASTERISK-30280: Create capability to assign a Media Experience Score to + RTP streams + Reported by: George Joseph + * [d454801c2d] George Joseph -- res_rtp_asterisk: Asterisk Media + Experience Score (MES) + + ---------------------------------------------------------------------- + + Commits Not Associated with an Issue + + [Back to Top] + + This is a list of all changes that went into this release that did not + reference a JIRA issue. + + +------------------------------------------------------------------------+ + | Revision | Author | Summary | + |------------+----------------------+------------------------------------| + | 93813c9dca | Asterisk Development | Update CHANGES and UPGRADE.txt for | + | | Team | 20.2.0 | + |------------+----------------------+------------------------------------| + | e5c5cd6e25 | Sean Bright | test.c: Avoid passing -1 to FD_* | + | | | family of functions. | + |------------+----------------------+------------------------------------| + | 827222d607 | Sean Bright | test_crypto.c: Fix getcwd(…) build | + | | | error. | + |------------+----------------------+------------------------------------| + | aef0c0ce0e | Sean Bright | app_queue: Reset all queue | + | | | defaults before reload. | + |------------+----------------------+------------------------------------| + | 96d9ad51ac | Sean Bright | doxygen: Fix doxygen errors. | + |------------+----------------------+------------------------------------| + | ef16eaee36 | Sean Bright | app_playback.c: Fix PLAYBACKSTATUS | + | | | regression. | + |------------+----------------------+------------------------------------| + | | | format_wav: replace | + | e86d5d7fda | Alexei Gradinari | ast_log(LOG_DEBUG, ...) by | + | | | ast_debug(1, ...) | + |------------+----------------------+------------------------------------| + | 62ca063fca | George Joseph | Revert "res_rtp_asterisk: Asterisk | + | | | Media Experience Score (MES)" | + +------------------------------------------------------------------------+ + + ---------------------------------------------------------------------- + + Diffstat Results + + [Back to Top] + + This is a summary of the changes to the source code that went into this + release that was generated using the diffstat utility. + + .lastclean | 1 + .version | 1 + ChangeLog |105422 ---------- + asterisk-20.1.0-summary.html | 279 + asterisk-20.1.0-summary.txt | 768 + b/CHANGES | 66 + b/UPGRADE.txt | 13 + b/apps/app_broadcast.c | 619 + b/apps/app_directory.c | 36 + b/apps/app_mf.c | 1 + b/apps/app_playback.c | 3 + b/apps/app_queue.c | 14 + b/apps/app_read.c | 23 + b/apps/app_senddtmf.c | 31 + b/apps/app_sendtext.c | 4 + b/apps/app_signal.c | 471 + b/apps/app_voicemail.c | 7 + b/channels/chan_iax2.c | 17 + b/channels/chan_pjsip.c | 114 + b/channels/pjsip/dialplan_functions.c | 67 + b/configs/samples/pjsip.conf.sample | 2 + b/configs/samples/queues.conf.sample | 10 + b/configure |18025 - + b/configure.ac | 13 + b/contrib/ast-db-manage/config/versions/f261363a857f_add_overlap_context.py | 21 + b/formats/format_wav.c | 2 + b/funcs/func_callerid.c | 1 + b/funcs/func_frame_trace.c | 1 + b/funcs/func_json.c | 232 + b/include/asterisk/autoconfig.h.in | 124 + b/include/asterisk/channel.h | 4 + b/include/asterisk/crypto.h | 12 + b/include/asterisk/file.h | 1 + b/include/asterisk/json.h | 9 + b/include/asterisk/pbx.h | 6 + b/include/asterisk/res_aeap.h | 8 + b/include/asterisk/res_aeap_message.h | 3 + b/include/asterisk/res_geolocation.h | 4 + b/include/asterisk/res_pjsip.h | 66 + b/include/asterisk/res_stir_shaken.h | 2 + b/include/asterisk/rtp_engine.h | 54 + b/include/asterisk/time.h | 88 + b/include/asterisk/utf8.h | 53 + b/include/asterisk/xml.h | 18 + b/main/bridge_basic.c | 2 + b/main/file.c | 4 + b/main/http.c | 10 + b/main/loader.c | 25 + b/main/manager.c | 6 + b/main/pbx_app.c | 2 + b/main/rtp_engine.c | 74 + b/main/stasis_channels.c | 33 + b/main/test.c | 13 + b/main/utf8.c | 544 + b/menuselect/autoconfig.h.in | 22 + b/menuselect/configure | 3476 + b/res/ael/pval.c | 14 + b/res/res_aeap/transaction.h | 4 + b/res/res_aeap/transport.h | 2 + b/res/res_geolocation/geoloc_eprofile.c | 14 + b/res/res_http_media_cache.c | 9 + b/res/res_phoneprov.c | 20 + b/res/res_pjsip.c | 337 + b/res/res_pjsip/pjsip_config.xml | 10 + b/res/res_pjsip/pjsip_configuration.c | 5 + b/res/res_pjsip/pjsip_manager.xml | 3 + b/res/res_pjsip/pjsip_transport_events.c | 2 + b/res/res_pjsip_caller_id.c | 227 + b/res/res_pjsip_path.c | 73 + b/res/res_pjsip_pubsub.c | 101 + b/res/res_pjsip_rfc3326.c | 31 + b/res/res_pjsip_sdp_rtp.c | 4 + b/res/res_pjsip_session.c | 24 + b/res/res_rtp_asterisk.c | 547 + b/res/res_speech_aeap.c | 51 + b/res/res_stasis_snoop.c | 10 + b/res/res_stir_shaken.c | 2 + b/tests/test_crypto.c | 32 + b/tests/test_res_rtp.c | 189 + b/third-party/pjproject/configure.m4 | 7 + b/third-party/pjproject/patches/0000-remove-third-party.patch | 6 + b/third-party/pjproject/patches/0010-Make-sure-that-NOTIFY-tdata-is-set-before-sending-it_new-129fb323a66dd1fd16880fe5ba5e6a57.patch | 46 + contrib/realtime/mysql/mysql_cdr.sql | 41 + contrib/realtime/mysql/mysql_config.sql | 1402 + contrib/realtime/mysql/mysql_voicemail.sql | 35 + contrib/realtime/postgresql/postgresql_cdr.sql | 45 + contrib/realtime/postgresql/postgresql_config.sql | 1524 + contrib/realtime/postgresql/postgresql_voicemail.sql | 39 + third-party/pjproject/patches/0100-allow_multiple_auth_headers.patch | 413 + third-party/pjproject/patches/0200-potential-buffer-overflow-in-pjlib-scanner-and-pjmedia.patch | 306 + third-party/pjproject/patches/0201-potential-stack-buffer-overflow-when-parsing-message-as-a-STUN-client.patch | 44 + third-party/pjproject/pjproject-2.12.1.tar.bz2.md5 | 1 + 92 files changed, 13653 insertions(+), 122894 deletions(-) diff --git a/contrib/realtime/mysql/mysql_cdr.sql b/contrib/realtime/mysql/mysql_cdr.sql new file mode 100644 index 0000000000..0d48377e85 --- /dev/null +++ b/contrib/realtime/mysql/mysql_cdr.sql @@ -0,0 +1,41 @@ +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL, + CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num) +); + +-- Running upgrade -> 210693f3123d + +CREATE TABLE cdr ( + accountcode VARCHAR(20), + src VARCHAR(80), + dst VARCHAR(80), + dcontext VARCHAR(80), + clid VARCHAR(80), + channel VARCHAR(80), + dstchannel VARCHAR(80), + lastapp VARCHAR(80), + lastdata VARCHAR(80), + start DATETIME, + answer DATETIME, + end DATETIME, + duration INTEGER, + billsec INTEGER, + disposition VARCHAR(45), + amaflags VARCHAR(45), + userfield VARCHAR(256), + uniqueid VARCHAR(150), + linkedid VARCHAR(150), + peeraccount VARCHAR(20), + sequence INTEGER +); + +INSERT INTO alembic_version (version_num) VALUES ('210693f3123d'); + +-- Running upgrade 210693f3123d -> 54cde9847798 + +ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL; + +ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL; + +UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d'; + diff --git a/contrib/realtime/mysql/mysql_config.sql b/contrib/realtime/mysql/mysql_config.sql new file mode 100644 index 0000000000..c08b26a890 --- /dev/null +++ b/contrib/realtime/mysql/mysql_config.sql @@ -0,0 +1,1408 @@ +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL, + CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num) +); + +-- Running upgrade -> 4da0c5f79a9c + +CREATE TABLE sippeers ( + id INTEGER NOT NULL AUTO_INCREMENT, + name VARCHAR(40) NOT NULL, + ipaddr VARCHAR(45), + port INTEGER, + regseconds INTEGER, + defaultuser VARCHAR(40), + fullcontact VARCHAR(80), + regserver VARCHAR(20), + useragent VARCHAR(20), + lastms INTEGER, + host VARCHAR(40), + type ENUM('friend','user','peer'), + context VARCHAR(40), + permit VARCHAR(95), + deny VARCHAR(95), + secret VARCHAR(40), + md5secret VARCHAR(40), + remotesecret VARCHAR(40), + transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp'), + dtmfmode ENUM('rfc2833','info','shortinfo','inband','auto'), + directmedia ENUM('yes','no','nonat','update'), + nat VARCHAR(29), + callgroup VARCHAR(40), + pickupgroup VARCHAR(40), + language VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + insecure VARCHAR(40), + trustrpid ENUM('yes','no'), + progressinband ENUM('yes','no','never'), + promiscredir ENUM('yes','no'), + useclientcode ENUM('yes','no'), + accountcode VARCHAR(40), + setvar VARCHAR(200), + callerid VARCHAR(40), + amaflags VARCHAR(40), + callcounter ENUM('yes','no'), + busylevel INTEGER, + allowoverlap ENUM('yes','no'), + allowsubscribe ENUM('yes','no'), + videosupport ENUM('yes','no'), + maxcallbitrate INTEGER, + rfc2833compensate ENUM('yes','no'), + mailbox VARCHAR(40), + `session-timers` ENUM('accept','refuse','originate'), + `session-expires` INTEGER, + `session-minse` INTEGER, + `session-refresher` ENUM('uac','uas'), + t38pt_usertpsource VARCHAR(40), + regexten VARCHAR(40), + fromdomain VARCHAR(40), + fromuser VARCHAR(40), + qualify VARCHAR(40), + defaultip VARCHAR(45), + rtptimeout INTEGER, + rtpholdtimeout INTEGER, + sendrpid ENUM('yes','no'), + outboundproxy VARCHAR(40), + callbackextension VARCHAR(40), + timert1 INTEGER, + timerb INTEGER, + qualifyfreq INTEGER, + constantssrc ENUM('yes','no'), + contactpermit VARCHAR(95), + contactdeny VARCHAR(95), + usereqphone ENUM('yes','no'), + textsupport ENUM('yes','no'), + faxdetect ENUM('yes','no'), + buggymwi ENUM('yes','no'), + auth VARCHAR(40), + fullname VARCHAR(40), + trunkname VARCHAR(40), + cid_number VARCHAR(40), + callingpres ENUM('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib'), + mohinterpret VARCHAR(40), + mohsuggest VARCHAR(40), + parkinglot VARCHAR(40), + hasvoicemail ENUM('yes','no'), + subscribemwi ENUM('yes','no'), + vmexten VARCHAR(40), + autoframing ENUM('yes','no'), + rtpkeepalive INTEGER, + `call-limit` INTEGER, + g726nonstandard ENUM('yes','no'), + ignoresdpversion ENUM('yes','no'), + allowtransfer ENUM('yes','no'), + dynamic ENUM('yes','no'), + path VARCHAR(256), + supportpath ENUM('yes','no'), + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX sippeers_name ON sippeers (name); + +CREATE INDEX sippeers_name_host ON sippeers (name, host); + +CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port); + +CREATE INDEX sippeers_host_port ON sippeers (host, port); + +CREATE TABLE iaxfriends ( + id INTEGER NOT NULL AUTO_INCREMENT, + name VARCHAR(40) NOT NULL, + type ENUM('friend','user','peer'), + username VARCHAR(40), + mailbox VARCHAR(40), + secret VARCHAR(40), + dbsecret VARCHAR(40), + context VARCHAR(40), + regcontext VARCHAR(40), + host VARCHAR(40), + ipaddr VARCHAR(40), + port INTEGER, + defaultip VARCHAR(20), + sourceaddress VARCHAR(20), + mask VARCHAR(20), + regexten VARCHAR(40), + regseconds INTEGER, + accountcode VARCHAR(20), + mohinterpret VARCHAR(20), + mohsuggest VARCHAR(20), + inkeys VARCHAR(40), + outkeys VARCHAR(40), + language VARCHAR(10), + callerid VARCHAR(100), + cid_number VARCHAR(40), + sendani ENUM('yes','no'), + fullname VARCHAR(40), + trunk ENUM('yes','no'), + auth VARCHAR(20), + maxauthreq INTEGER, + requirecalltoken ENUM('yes','no','auto'), + encryption ENUM('yes','no','aes128'), + transfer ENUM('yes','no','mediaonly'), + jitterbuffer ENUM('yes','no'), + forcejitterbuffer ENUM('yes','no'), + disallow VARCHAR(200), + allow VARCHAR(200), + codecpriority VARCHAR(40), + qualify VARCHAR(10), + qualifysmoothing ENUM('yes','no'), + qualifyfreqok VARCHAR(10), + qualifyfreqnotok VARCHAR(10), + timezone VARCHAR(20), + adsi ENUM('yes','no'), + amaflags VARCHAR(20), + setvar VARCHAR(200), + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX iaxfriends_name ON iaxfriends (name); + +CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host); + +CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port); + +CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port); + +CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port); + +CREATE TABLE voicemail ( + uniqueid INTEGER NOT NULL AUTO_INCREMENT, + context VARCHAR(80) NOT NULL, + mailbox VARCHAR(80) NOT NULL, + password VARCHAR(80) NOT NULL, + fullname VARCHAR(80), + alias VARCHAR(80), + email VARCHAR(80), + pager VARCHAR(80), + attach ENUM('yes','no'), + attachfmt VARCHAR(10), + serveremail VARCHAR(80), + language VARCHAR(20), + tz VARCHAR(30), + deletevoicemail ENUM('yes','no'), + saycid ENUM('yes','no'), + sendvoicemail ENUM('yes','no'), + review ENUM('yes','no'), + tempgreetwarn ENUM('yes','no'), + operator ENUM('yes','no'), + envelope ENUM('yes','no'), + sayduration INTEGER, + forcename ENUM('yes','no'), + forcegreetings ENUM('yes','no'), + callback VARCHAR(80), + dialout VARCHAR(80), + exitcontext VARCHAR(80), + maxmsg INTEGER, + volgain NUMERIC(5, 2), + imapuser VARCHAR(80), + imappassword VARCHAR(80), + imapserver VARCHAR(80), + imapport VARCHAR(8), + imapflags VARCHAR(80), + stamp DATETIME, + PRIMARY KEY (uniqueid) +); + +CREATE INDEX voicemail_mailbox ON voicemail (mailbox); + +CREATE INDEX voicemail_context ON voicemail (context); + +CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context); + +CREATE INDEX voicemail_imapuser ON voicemail (imapuser); + +CREATE TABLE meetme ( + bookid INTEGER NOT NULL AUTO_INCREMENT, + confno VARCHAR(80) NOT NULL, + starttime DATETIME, + endtime DATETIME, + pin VARCHAR(20), + adminpin VARCHAR(20), + opts VARCHAR(20), + adminopts VARCHAR(20), + recordingfilename VARCHAR(80), + recordingformat VARCHAR(10), + maxusers INTEGER, + members INTEGER NOT NULL, + PRIMARY KEY (bookid) +); + +CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime); + +CREATE TABLE musiconhold ( + name VARCHAR(80) NOT NULL, + mode ENUM('custom','files','mp3nb','quietmp3nb','quietmp3'), + directory VARCHAR(255), + application VARCHAR(255), + digit VARCHAR(1), + sort VARCHAR(10), + format VARCHAR(10), + stamp DATETIME, + PRIMARY KEY (name) +); + +INSERT INTO alembic_version (version_num) VALUES ('4da0c5f79a9c'); + +-- Running upgrade 4da0c5f79a9c -> 43956d550a44 + +CREATE TABLE ps_endpoints ( + id VARCHAR(40) NOT NULL, + transport VARCHAR(40), + aors VARCHAR(200), + auth VARCHAR(40), + context VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + direct_media ENUM('yes','no'), + connected_line_method ENUM('invite','reinvite','update'), + direct_media_method ENUM('invite','reinvite','update'), + direct_media_glare_mitigation ENUM('none','outgoing','incoming'), + disable_direct_media_on_nat ENUM('yes','no'), + dtmf_mode ENUM('rfc4733','inband','info'), + external_media_address VARCHAR(40), + force_rport ENUM('yes','no'), + ice_support ENUM('yes','no'), + identify_by ENUM('username'), + mailboxes VARCHAR(40), + moh_suggest VARCHAR(40), + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + rewrite_contact ENUM('yes','no'), + rtp_ipv6 ENUM('yes','no'), + rtp_symmetric ENUM('yes','no'), + send_diversion ENUM('yes','no'), + send_pai ENUM('yes','no'), + send_rpid ENUM('yes','no'), + timers_min_se INTEGER, + timers ENUM('forced','no','required','yes'), + timers_sess_expires INTEGER, + callerid VARCHAR(40), + callerid_privacy ENUM('allowed_not_screened','allowed_passed_screened','allowed_failed_screened','allowed','prohib_not_screened','prohib_passed_screened','prohib_failed_screened','prohib','unavailable'), + callerid_tag VARCHAR(40), + `100rel` ENUM('no','required','yes'), + aggregate_mwi ENUM('yes','no'), + trust_id_inbound ENUM('yes','no'), + trust_id_outbound ENUM('yes','no'), + use_ptime ENUM('yes','no'), + use_avpf ENUM('yes','no'), + media_encryption ENUM('no','sdes','dtls'), + inband_progress ENUM('yes','no'), + call_group VARCHAR(40), + pickup_group VARCHAR(40), + named_call_group VARCHAR(40), + named_pickup_group VARCHAR(40), + device_state_busy_at INTEGER, + fax_detect ENUM('yes','no'), + t38_udptl ENUM('yes','no'), + t38_udptl_ec ENUM('none','fec','redundancy'), + t38_udptl_maxdatagram INTEGER, + t38_udptl_nat ENUM('yes','no'), + t38_udptl_ipv6 ENUM('yes','no'), + tone_zone VARCHAR(40), + language VARCHAR(40), + one_touch_recording ENUM('yes','no'), + record_on_feature VARCHAR(40), + record_off_feature VARCHAR(40), + rtp_engine VARCHAR(40), + allow_transfer ENUM('yes','no'), + allow_subscribe ENUM('yes','no'), + sdp_owner VARCHAR(40), + sdp_session VARCHAR(40), + tos_audio INTEGER, + tos_video INTEGER, + cos_audio INTEGER, + cos_video INTEGER, + sub_min_expiry INTEGER, + from_domain VARCHAR(40), + from_user VARCHAR(40), + mwi_fromuser VARCHAR(40), + dtls_verify VARCHAR(40), + dtls_rekey VARCHAR(40), + dtls_cert_file VARCHAR(200), + dtls_private_key VARCHAR(200), + dtls_cipher VARCHAR(200), + dtls_ca_file VARCHAR(200), + dtls_ca_path VARCHAR(200), + dtls_setup ENUM('active','passive','actpass'), + srtp_tag_32 ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_endpoints_id ON ps_endpoints (id); + +CREATE TABLE ps_auths ( + id VARCHAR(40) NOT NULL, + auth_type ENUM('md5','userpass'), + nonce_lifetime INTEGER, + md5_cred VARCHAR(40), + password VARCHAR(80), + realm VARCHAR(40), + username VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_auths_id ON ps_auths (id); + +CREATE TABLE ps_aors ( + id VARCHAR(40) NOT NULL, + contact VARCHAR(40), + default_expiration INTEGER, + mailboxes VARCHAR(80), + max_contacts INTEGER, + minimum_expiration INTEGER, + remove_existing ENUM('yes','no'), + qualify_frequency INTEGER, + authenticate_qualify ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_aors_id ON ps_aors (id); + +CREATE TABLE ps_contacts ( + id VARCHAR(40) NOT NULL, + uri VARCHAR(40), + expiration_time VARCHAR(40), + qualify_frequency INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_contacts_id ON ps_contacts (id); + +CREATE TABLE ps_domain_aliases ( + id VARCHAR(40) NOT NULL, + domain VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id); + +CREATE TABLE ps_endpoint_id_ips ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40), + `match` VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id); + +UPDATE alembic_version SET version_num='43956d550a44' WHERE alembic_version.version_num = '4da0c5f79a9c'; + +-- Running upgrade 43956d550a44 -> 581a4264e537 + +CREATE TABLE extensions ( + id BIGINT NOT NULL AUTO_INCREMENT, + context VARCHAR(40) NOT NULL, + exten VARCHAR(40) NOT NULL, + priority INTEGER NOT NULL, + app VARCHAR(40) NOT NULL, + appdata VARCHAR(256) NOT NULL, + PRIMARY KEY (id), + UNIQUE (context, exten, priority), + UNIQUE (id) +); + +UPDATE alembic_version SET version_num='581a4264e537' WHERE alembic_version.version_num = '43956d550a44'; + +-- Running upgrade 581a4264e537 -> 2fc7930b41b3 + +CREATE TABLE ps_systems ( + id VARCHAR(40) NOT NULL, + timer_t1 INTEGER, + timer_b INTEGER, + compact_headers ENUM('yes','no'), + threadpool_initial_size INTEGER, + threadpool_auto_increment INTEGER, + threadpool_idle_timeout INTEGER, + threadpool_max_size INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_systems_id ON ps_systems (id); + +CREATE TABLE ps_globals ( + id VARCHAR(40) NOT NULL, + max_forwards INTEGER, + user_agent VARCHAR(40), + default_outbound_endpoint VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_globals_id ON ps_globals (id); + +CREATE TABLE ps_transports ( + id VARCHAR(40) NOT NULL, + async_operations INTEGER, + bind VARCHAR(40), + ca_list_file VARCHAR(200), + cert_file VARCHAR(200), + cipher VARCHAR(200), + domain VARCHAR(40), + external_media_address VARCHAR(40), + external_signaling_address VARCHAR(40), + external_signaling_port INTEGER, + method ENUM('default','unspecified','tlsv1','sslv2','sslv3','sslv23'), + local_net VARCHAR(40), + password VARCHAR(40), + priv_key_file VARCHAR(200), + protocol ENUM('udp','tcp','tls','ws','wss'), + require_client_cert ENUM('yes','no'), + verify_client ENUM('yes','no'), + verifiy_server ENUM('yes','no'), + tos ENUM('yes','no'), + cos ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_transports_id ON ps_transports (id); + +CREATE TABLE ps_registrations ( + id VARCHAR(40) NOT NULL, + auth_rejection_permanent ENUM('yes','no'), + client_uri VARCHAR(40), + contact_user VARCHAR(40), + expiration INTEGER, + max_retries INTEGER, + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + retry_interval INTEGER, + forbidden_retry_interval INTEGER, + server_uri VARCHAR(40), + transport VARCHAR(40), + support_path ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_registrations_id ON ps_registrations (id); + +ALTER TABLE ps_endpoints ADD COLUMN media_address VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN redirect_method ENUM('user','uri_core','uri_pjsip'); + +ALTER TABLE ps_endpoints ADD COLUMN set_var TEXT; + +ALTER TABLE ps_endpoints CHANGE mwi_fromuser mwi_from_user VARCHAR(40) NULL; + +ALTER TABLE ps_contacts ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN path TEXT; + +ALTER TABLE ps_aors ADD COLUMN maximum_expiration INTEGER; + +ALTER TABLE ps_aors ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_aors ADD COLUMN support_path ENUM('yes','no'); + +UPDATE alembic_version SET version_num='2fc7930b41b3' WHERE alembic_version.version_num = '581a4264e537'; + +-- Running upgrade 2fc7930b41b3 -> 21e526ad3040 + +ALTER TABLE ps_globals ADD COLUMN debug VARCHAR(40); + +UPDATE alembic_version SET version_num='21e526ad3040' WHERE alembic_version.version_num = '2fc7930b41b3'; + +-- Running upgrade 21e526ad3040 -> 28887f25a46f + +CREATE TABLE queues ( + name VARCHAR(128) NOT NULL, + musiconhold VARCHAR(128), + announce VARCHAR(128), + context VARCHAR(128), + timeout INTEGER, + ringinuse ENUM('yes','no'), + setinterfacevar ENUM('yes','no'), + setqueuevar ENUM('yes','no'), + setqueueentryvar ENUM('yes','no'), + monitor_format VARCHAR(8), + membermacro VARCHAR(512), + membergosub VARCHAR(512), + queue_youarenext VARCHAR(128), + queue_thereare VARCHAR(128), + queue_callswaiting VARCHAR(128), + queue_quantity1 VARCHAR(128), + queue_quantity2 VARCHAR(128), + queue_holdtime VARCHAR(128), + queue_minutes VARCHAR(128), + queue_minute VARCHAR(128), + queue_seconds VARCHAR(128), + queue_thankyou VARCHAR(128), + queue_callerannounce VARCHAR(128), + queue_reporthold VARCHAR(128), + announce_frequency INTEGER, + announce_to_first_user ENUM('yes','no'), + min_announce_frequency INTEGER, + announce_round_seconds INTEGER, + announce_holdtime VARCHAR(128), + announce_position VARCHAR(128), + announce_position_limit INTEGER, + periodic_announce VARCHAR(50), + periodic_announce_frequency INTEGER, + relative_periodic_announce ENUM('yes','no'), + random_periodic_announce ENUM('yes','no'), + retry INTEGER, + wrapuptime INTEGER, + penaltymemberslimit INTEGER, + autofill ENUM('yes','no'), + monitor_type VARCHAR(128), + autopause ENUM('yes','no','all'), + autopausedelay INTEGER, + autopausebusy ENUM('yes','no'), + autopauseunavail ENUM('yes','no'), + maxlen INTEGER, + servicelevel INTEGER, + strategy ENUM('ringall','leastrecent','fewestcalls','random','rrmemory','linear','wrandom','rrordered'), + joinempty VARCHAR(128), + leavewhenempty VARCHAR(128), + reportholdtime ENUM('yes','no'), + memberdelay INTEGER, + weight INTEGER, + timeoutrestart ENUM('yes','no'), + defaultrule VARCHAR(128), + timeoutpriority VARCHAR(128), + PRIMARY KEY (name) +); + +CREATE TABLE queue_members ( + queue_name VARCHAR(80) NOT NULL, + interface VARCHAR(80) NOT NULL, + uniqueid VARCHAR(80) NOT NULL, + membername VARCHAR(80), + state_interface VARCHAR(80), + penalty INTEGER, + paused INTEGER, + PRIMARY KEY (queue_name, interface) +); + +UPDATE alembic_version SET version_num='28887f25a46f' WHERE alembic_version.version_num = '21e526ad3040'; + +-- Running upgrade 28887f25a46f -> 4c573e7135bd + +ALTER TABLE ps_endpoints MODIFY tos_audio VARCHAR(10) NULL; + +ALTER TABLE ps_endpoints MODIFY tos_video VARCHAR(10) NULL; + +ALTER TABLE ps_endpoints DROP COLUMN cos_audio; + +ALTER TABLE ps_endpoints DROP COLUMN cos_video; + +ALTER TABLE ps_endpoints ADD COLUMN cos_audio INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN cos_video INTEGER; + +ALTER TABLE ps_transports MODIFY tos VARCHAR(10) NULL; + +ALTER TABLE ps_transports DROP COLUMN cos; + +ALTER TABLE ps_transports ADD COLUMN cos INTEGER; + +UPDATE alembic_version SET version_num='4c573e7135bd' WHERE alembic_version.version_num = '28887f25a46f'; + +-- Running upgrade 4c573e7135bd -> 3855ee4e5f85 + +ALTER TABLE ps_endpoints ADD COLUMN message_context VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN user_agent VARCHAR(40); + +UPDATE alembic_version SET version_num='3855ee4e5f85' WHERE alembic_version.version_num = '4c573e7135bd'; + +-- Running upgrade 3855ee4e5f85 -> e96a0b8071c + +ALTER TABLE ps_globals MODIFY user_agent VARCHAR(255) NULL; + +ALTER TABLE ps_contacts MODIFY id VARCHAR(255) NULL; + +ALTER TABLE ps_contacts MODIFY uri VARCHAR(255) NULL; + +ALTER TABLE ps_contacts MODIFY user_agent VARCHAR(255) NULL; + +ALTER TABLE ps_registrations MODIFY client_uri VARCHAR(255) NULL; + +ALTER TABLE ps_registrations MODIFY server_uri VARCHAR(255) NULL; + +UPDATE alembic_version SET version_num='e96a0b8071c' WHERE alembic_version.version_num = '3855ee4e5f85'; + +-- Running upgrade e96a0b8071c -> c6d929b23a8 + +CREATE TABLE ps_subscription_persistence ( + id VARCHAR(40) NOT NULL, + packet VARCHAR(2048), + src_name VARCHAR(128), + src_port INTEGER, + transport_key VARCHAR(64), + local_name VARCHAR(128), + local_port INTEGER, + cseq INTEGER, + tag VARCHAR(128), + endpoint VARCHAR(40), + expires INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id); + +UPDATE alembic_version SET version_num='c6d929b23a8' WHERE alembic_version.version_num = 'e96a0b8071c'; + +-- Running upgrade c6d929b23a8 -> 51f8cb66540e + +ALTER TABLE ps_endpoints ADD COLUMN force_avp ENUM('yes','no'); + +ALTER TABLE ps_endpoints ADD COLUMN media_use_received_transport ENUM('yes','no'); + +UPDATE alembic_version SET version_num='51f8cb66540e' WHERE alembic_version.version_num = 'c6d929b23a8'; + +-- Running upgrade 51f8cb66540e -> 1d50859ed02e + +ALTER TABLE ps_endpoints ADD COLUMN accountcode VARCHAR(20); + +UPDATE alembic_version SET version_num='1d50859ed02e' WHERE alembic_version.version_num = '51f8cb66540e'; + +-- Running upgrade 1d50859ed02e -> 1758e8bbf6b + +ALTER TABLE sippeers MODIFY useragent VARCHAR(255) NULL; + +UPDATE alembic_version SET version_num='1758e8bbf6b' WHERE alembic_version.version_num = '1d50859ed02e'; + +-- Running upgrade 1758e8bbf6b -> 5139253c0423 + +ALTER TABLE queue_members DROP COLUMN uniqueid; + +ALTER TABLE queue_members ADD COLUMN uniqueid INTEGER NOT NULL; + +ALTER TABLE queue_members ADD UNIQUE (uniqueid); + +ALTER TABLE queue_members MODIFY uniqueid INTEGER NOT NULL AUTO_INCREMENT; + +UPDATE alembic_version SET version_num='5139253c0423' WHERE alembic_version.version_num = '1758e8bbf6b'; + +-- Running upgrade 5139253c0423 -> d39508cb8d8 + +CREATE TABLE queue_rules ( + rule_name VARCHAR(80) NOT NULL, + time VARCHAR(32) NOT NULL, + min_penalty VARCHAR(32) NOT NULL, + max_penalty VARCHAR(32) NOT NULL +); + +UPDATE alembic_version SET version_num='d39508cb8d8' WHERE alembic_version.version_num = '5139253c0423'; + +-- Running upgrade d39508cb8d8 -> 5950038a6ead + +ALTER TABLE ps_transports CHANGE verifiy_server verify_server ENUM('yes','no') NULL; + +UPDATE alembic_version SET version_num='5950038a6ead' WHERE alembic_version.version_num = 'd39508cb8d8'; + +-- Running upgrade 5950038a6ead -> 10aedae86a32 + +ALTER TABLE sippeers MODIFY directmedia ENUM('yes','no','nonat','update','outgoing') NULL; + +UPDATE alembic_version SET version_num='10aedae86a32' WHERE alembic_version.version_num = '5950038a6ead'; + +-- Running upgrade 10aedae86a32 -> 371a3bf4143e + +ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone ENUM('yes','no'); + +UPDATE alembic_version SET version_num='371a3bf4143e' WHERE alembic_version.version_num = '10aedae86a32'; + +-- Running upgrade 371a3bf4143e -> 15b1430ad6f1 + +ALTER TABLE ps_endpoints ADD COLUMN moh_passthrough ENUM('yes','no'); + +UPDATE alembic_version SET version_num='15b1430ad6f1' WHERE alembic_version.version_num = '371a3bf4143e'; + +-- Running upgrade 15b1430ad6f1 -> 945b1098bdd + +ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic ENUM('yes','no'); + +UPDATE alembic_version SET version_num='945b1098bdd' WHERE alembic_version.version_num = '15b1430ad6f1'; + +-- Running upgrade 945b1098bdd -> 45e3f47c6c44 + +ALTER TABLE ps_globals ADD COLUMN endpoint_identifier_order VARCHAR(40); + +UPDATE alembic_version SET version_num='45e3f47c6c44' WHERE alembic_version.version_num = '945b1098bdd'; + +-- Running upgrade 45e3f47c6c44 -> 23530d604b96 + +ALTER TABLE ps_endpoints ADD COLUMN rpid_immediate ENUM('yes','no'); + +UPDATE alembic_version SET version_num='23530d604b96' WHERE alembic_version.version_num = '45e3f47c6c44'; + +-- Running upgrade 23530d604b96 -> 31cd4f4891ec + +ALTER TABLE ps_endpoints MODIFY dtmf_mode ENUM('rfc4733','inband','info','auto') NULL; + +UPDATE alembic_version SET version_num='31cd4f4891ec' WHERE alembic_version.version_num = '23530d604b96'; + +-- Running upgrade 31cd4f4891ec -> 461d7d691209 + +ALTER TABLE ps_aors ADD COLUMN qualify_timeout INTEGER; + +ALTER TABLE ps_contacts ADD COLUMN qualify_timeout INTEGER; + +UPDATE alembic_version SET version_num='461d7d691209' WHERE alembic_version.version_num = '31cd4f4891ec'; + +-- Running upgrade 461d7d691209 -> a541e0b5e89 + +ALTER TABLE ps_globals ADD COLUMN max_initial_qualify_time INTEGER; + +UPDATE alembic_version SET version_num='a541e0b5e89' WHERE alembic_version.version_num = '461d7d691209'; + +-- Running upgrade a541e0b5e89 -> 28b8e71e541f + +ALTER TABLE ps_endpoints ADD COLUMN g726_non_standard ENUM('yes','no'); + +UPDATE alembic_version SET version_num='28b8e71e541f' WHERE alembic_version.version_num = 'a541e0b5e89'; + +-- Running upgrade 28b8e71e541f -> 498357a710ae + +ALTER TABLE ps_endpoints ADD COLUMN rtp_keepalive INTEGER; + +UPDATE alembic_version SET version_num='498357a710ae' WHERE alembic_version.version_num = '28b8e71e541f'; + +-- Running upgrade 498357a710ae -> 26f10cadc157 + +ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout_hold INTEGER; + +UPDATE alembic_version SET version_num='26f10cadc157' WHERE alembic_version.version_num = '498357a710ae'; + +-- Running upgrade 26f10cadc157 -> 154177371065 + +ALTER TABLE ps_globals ADD COLUMN default_from_user VARCHAR(80); + +UPDATE alembic_version SET version_num='154177371065' WHERE alembic_version.version_num = '26f10cadc157'; + +-- Running upgrade 154177371065 -> 28ce1e718f05 + +ALTER TABLE ps_registrations ADD COLUMN fatal_retry_interval INTEGER; + +UPDATE alembic_version SET version_num='28ce1e718f05' WHERE alembic_version.version_num = '154177371065'; + +-- Running upgrade 28ce1e718f05 -> 339a3bdf53fc + +ALTER TABLE ps_endpoints MODIFY accountcode VARCHAR(80) NULL; + +ALTER TABLE sippeers MODIFY accountcode VARCHAR(80) NULL; + +ALTER TABLE iaxfriends MODIFY accountcode VARCHAR(80) NULL; + +UPDATE alembic_version SET version_num='339a3bdf53fc' WHERE alembic_version.version_num = '28ce1e718f05'; + +-- Running upgrade 339a3bdf53fc -> 189a235b3fd7 + +ALTER TABLE ps_globals ADD COLUMN keep_alive_interval INTEGER; + +UPDATE alembic_version SET version_num='189a235b3fd7' WHERE alembic_version.version_num = '339a3bdf53fc'; + +-- Running upgrade 189a235b3fd7 -> 2d078ec071b7 + +ALTER TABLE ps_aors MODIFY contact VARCHAR(255) NULL; + +UPDATE alembic_version SET version_num='2d078ec071b7' WHERE alembic_version.version_num = '189a235b3fd7'; + +-- Running upgrade 2d078ec071b7 -> 26d7f3bf0fa5 + +ALTER TABLE ps_endpoints ADD COLUMN bind_rtp_to_media_address ENUM('yes','no'); + +UPDATE alembic_version SET version_num='26d7f3bf0fa5' WHERE alembic_version.version_num = '2d078ec071b7'; + +-- Running upgrade 26d7f3bf0fa5 -> 136885b81223 + +ALTER TABLE ps_globals ADD COLUMN regcontext VARCHAR(80); + +UPDATE alembic_version SET version_num='136885b81223' WHERE alembic_version.version_num = '26d7f3bf0fa5'; + +-- Running upgrade 136885b81223 -> 423f34ad36e2 + +ALTER TABLE ps_aors MODIFY qualify_timeout FLOAT NULL; + +ALTER TABLE ps_contacts MODIFY qualify_timeout FLOAT NULL; + +UPDATE alembic_version SET version_num='423f34ad36e2' WHERE alembic_version.version_num = '136885b81223'; + +-- Running upgrade 423f34ad36e2 -> dbc44d5a908 + +ALTER TABLE ps_systems ADD COLUMN disable_tcp_switch ENUM('yes','no'); + +ALTER TABLE ps_registrations ADD COLUMN line ENUM('yes','no'); + +ALTER TABLE ps_registrations ADD COLUMN endpoint VARCHAR(40); + +UPDATE alembic_version SET version_num='dbc44d5a908' WHERE alembic_version.version_num = '423f34ad36e2'; + +-- Running upgrade dbc44d5a908 -> 3bcc0b5bc2c9 + +ALTER TABLE ps_transports ADD COLUMN allow_reload ENUM('yes','no'); + +UPDATE alembic_version SET version_num='3bcc0b5bc2c9' WHERE alembic_version.version_num = 'dbc44d5a908'; + +-- Running upgrade 3bcc0b5bc2c9 -> 5813202e92be + +ALTER TABLE ps_globals ADD COLUMN contact_expiration_check_interval INTEGER; + +UPDATE alembic_version SET version_num='5813202e92be' WHERE alembic_version.version_num = '3bcc0b5bc2c9'; + +-- Running upgrade 5813202e92be -> 1c688d9a003c + +ALTER TABLE ps_globals ADD COLUMN default_voicemail_extension VARCHAR(40); + +ALTER TABLE ps_aors ADD COLUMN voicemail_extension VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN voicemail_extension VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN mwi_subscribe_replaces_unsolicited INTEGER; + +UPDATE alembic_version SET version_num='1c688d9a003c' WHERE alembic_version.version_num = '5813202e92be'; + +-- Running upgrade 1c688d9a003c -> 8d478ab86e29 + +ALTER TABLE ps_globals ADD COLUMN disable_multi_domain ENUM('yes','no'); + +UPDATE alembic_version SET version_num='8d478ab86e29' WHERE alembic_version.version_num = '1c688d9a003c'; + +-- Running upgrade 8d478ab86e29 -> 65eb22eb195 + +ALTER TABLE ps_globals ADD COLUMN unidentified_request_count INTEGER; + +ALTER TABLE ps_globals ADD COLUMN unidentified_request_period INTEGER; + +ALTER TABLE ps_globals ADD COLUMN unidentified_request_prune_interval INTEGER; + +ALTER TABLE ps_globals ADD COLUMN default_realm VARCHAR(40); + +UPDATE alembic_version SET version_num='65eb22eb195' WHERE alembic_version.version_num = '8d478ab86e29'; + +-- Running upgrade 65eb22eb195 -> 81b01a191a46 + +ALTER TABLE ps_contacts ADD COLUMN reg_server VARCHAR(20); + +ALTER TABLE ps_contacts ADD CONSTRAINT ps_contacts_uq UNIQUE (id, reg_server); + +UPDATE alembic_version SET version_num='81b01a191a46' WHERE alembic_version.version_num = '65eb22eb195'; + +-- Running upgrade 81b01a191a46 -> 6be31516058d + +ALTER TABLE ps_contacts ADD COLUMN authenticate_qualify ENUM('yes','no'); + +UPDATE alembic_version SET version_num='6be31516058d' WHERE alembic_version.version_num = '81b01a191a46'; + +-- Running upgrade 6be31516058d -> d7e3c73eb2bf + +ALTER TABLE ps_endpoints ADD COLUMN deny VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN permit VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN acl VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN contact_deny VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN contact_permit VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN contact_acl VARCHAR(40); + +UPDATE alembic_version SET version_num='d7e3c73eb2bf' WHERE alembic_version.version_num = '6be31516058d'; + +-- Running upgrade d7e3c73eb2bf -> a845e4d8ade8 + +ALTER TABLE ps_contacts ADD COLUMN via_addr VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN via_port INTEGER; + +ALTER TABLE ps_contacts ADD COLUMN call_id VARCHAR(255); + +UPDATE alembic_version SET version_num='a845e4d8ade8' WHERE alembic_version.version_num = 'd7e3c73eb2bf'; + +-- Running upgrade a845e4d8ade8 -> ef7efc2d3964 + +ALTER TABLE ps_contacts ADD COLUMN endpoint VARCHAR(40); + +ALTER TABLE ps_contacts MODIFY expiration_time BIGINT NULL; + +CREATE INDEX ps_contacts_qualifyfreq_exp ON ps_contacts (qualify_frequency, expiration_time); + +CREATE INDEX ps_aors_qualifyfreq_contact ON ps_aors (qualify_frequency, contact); + +UPDATE alembic_version SET version_num='ef7efc2d3964' WHERE alembic_version.version_num = 'a845e4d8ade8'; + +-- Running upgrade ef7efc2d3964 -> 9deac0ae4717 + +ALTER TABLE ps_endpoints ADD COLUMN subscribe_context VARCHAR(40); + +UPDATE alembic_version SET version_num='9deac0ae4717' WHERE alembic_version.version_num = 'ef7efc2d3964'; + +-- Running upgrade 9deac0ae4717 -> 4a6c67fa9b7a + +ALTER TABLE ps_endpoints ADD COLUMN fax_detect_timeout INTEGER; + +UPDATE alembic_version SET version_num='4a6c67fa9b7a' WHERE alembic_version.version_num = '9deac0ae4717'; + +-- Running upgrade 4a6c67fa9b7a -> c7a44a5a0851 + +ALTER TABLE ps_globals ADD COLUMN mwi_tps_queue_high INTEGER; + +ALTER TABLE ps_globals ADD COLUMN mwi_tps_queue_low INTEGER; + +ALTER TABLE ps_globals ADD COLUMN mwi_disable_initial_unsolicited ENUM('yes','no'); + +UPDATE alembic_version SET version_num='c7a44a5a0851' WHERE alembic_version.version_num = '4a6c67fa9b7a'; + +-- Running upgrade c7a44a5a0851 -> 3772f8f828da + +ALTER TABLE ps_endpoints MODIFY identify_by ENUM('username','auth_username') NULL; + +UPDATE alembic_version SET version_num='3772f8f828da' WHERE alembic_version.version_num = 'c7a44a5a0851'; + +-- Running upgrade 3772f8f828da -> 4e2493ef32e6 + +ALTER TABLE ps_endpoints ADD COLUMN contact_user VARCHAR(80); + +UPDATE alembic_version SET version_num='4e2493ef32e6' WHERE alembic_version.version_num = '3772f8f828da'; + +-- Running upgrade 4e2493ef32e6 -> 7f3e21abe318 + +ALTER TABLE ps_endpoints ADD COLUMN preferred_codec_only ENUM('yes','no'); + +UPDATE alembic_version SET version_num='7f3e21abe318' WHERE alembic_version.version_num = '4e2493ef32e6'; + +-- Running upgrade 7f3e21abe318 -> a6ef36f1309 + +ALTER TABLE ps_globals ADD COLUMN ignore_uri_user_options ENUM('yes','no'); + +UPDATE alembic_version SET version_num='a6ef36f1309' WHERE alembic_version.version_num = '7f3e21abe318'; + +-- Running upgrade a6ef36f1309 -> 4468b4a91372 + +ALTER TABLE ps_endpoints ADD COLUMN asymmetric_rtp_codec ENUM('yes','no'); + +UPDATE alembic_version SET version_num='4468b4a91372' WHERE alembic_version.version_num = 'a6ef36f1309'; + +-- Running upgrade 4468b4a91372 -> 28ab27a7826d + +ALTER TABLE ps_endpoint_id_ips ADD COLUMN srv_lookups ENUM('yes','no'); + +UPDATE alembic_version SET version_num='28ab27a7826d' WHERE alembic_version.version_num = '4468b4a91372'; + +-- Running upgrade 28ab27a7826d -> 465e70e8c337 + +ALTER TABLE ps_endpoint_id_ips ADD COLUMN match_header VARCHAR(255); + +UPDATE alembic_version SET version_num='465e70e8c337' WHERE alembic_version.version_num = '28ab27a7826d'; + +-- Running upgrade 465e70e8c337 -> 15db7b91a97a + +ALTER TABLE ps_endpoints ADD COLUMN rtcp_mux ENUM('yes','no'); + +UPDATE alembic_version SET version_num='15db7b91a97a' WHERE alembic_version.version_num = '465e70e8c337'; + +-- Running upgrade 15db7b91a97a -> f638dbe2eb23 + +ALTER TABLE ps_transports ADD COLUMN symmetric_transport ENUM('yes','no'); + +ALTER TABLE ps_subscription_persistence ADD COLUMN contact_uri VARCHAR(256); + +UPDATE alembic_version SET version_num='f638dbe2eb23' WHERE alembic_version.version_num = '15db7b91a97a'; + +-- Running upgrade f638dbe2eb23 -> 8fce4c573e15 + +ALTER TABLE ps_endpoints ADD COLUMN allow_overlap ENUM('yes','no'); + +UPDATE alembic_version SET version_num='8fce4c573e15' WHERE alembic_version.version_num = 'f638dbe2eb23'; + +-- Running upgrade 8fce4c573e15 -> 2da192dbbc65 + +CREATE TABLE ps_outbound_publishes ( + id VARCHAR(40) NOT NULL, + expiration INTEGER, + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(256), + server_uri VARCHAR(256), + from_uri VARCHAR(256), + to_uri VARCHAR(256), + event VARCHAR(40), + max_auth_attempts INTEGER, + transport VARCHAR(40), + multi_user ENUM('yes','no'), + `@body` VARCHAR(40), + `@context` VARCHAR(256), + `@exten` VARCHAR(256), + UNIQUE (id) +); + +CREATE INDEX ps_outbound_publishes_id ON ps_outbound_publishes (id); + +CREATE TABLE ps_inbound_publications ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40), + `event_asterisk-devicestate` VARCHAR(40), + `event_asterisk-mwi` VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_inbound_publications_id ON ps_inbound_publications (id); + +CREATE TABLE ps_asterisk_publications ( + id VARCHAR(40) NOT NULL, + devicestate_publish VARCHAR(40), + mailboxstate_publish VARCHAR(40), + device_state ENUM('yes','no'), + device_state_filter VARCHAR(256), + mailbox_state ENUM('yes','no'), + mailbox_state_filter VARCHAR(256), + UNIQUE (id) +); + +CREATE INDEX ps_asterisk_publications_id ON ps_asterisk_publications (id); + +UPDATE alembic_version SET version_num='2da192dbbc65' WHERE alembic_version.version_num = '8fce4c573e15'; + +-- Running upgrade 2da192dbbc65 -> 1d0e332c32af + +CREATE TABLE ps_resource_list ( + id VARCHAR(40) NOT NULL, + list_item VARCHAR(2048), + event VARCHAR(40), + full_state ENUM('yes','no'), + notification_batch_interval INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_resource_list_id ON ps_resource_list (id); + +UPDATE alembic_version SET version_num='1d0e332c32af' WHERE alembic_version.version_num = '2da192dbbc65'; + +-- Running upgrade 1d0e332c32af -> 86bb1efa278d + +ALTER TABLE ps_endpoints ADD COLUMN refer_blind_progress ENUM('yes','no'); + +UPDATE alembic_version SET version_num='86bb1efa278d' WHERE alembic_version.version_num = '1d0e332c32af'; + +-- Running upgrade 86bb1efa278d -> d7983954dd96 + +ALTER TABLE ps_endpoints ADD COLUMN notify_early_inuse_ringing ENUM('yes','no'); + +UPDATE alembic_version SET version_num='d7983954dd96' WHERE alembic_version.version_num = '86bb1efa278d'; + +-- Running upgrade d7983954dd96 -> 39959b9c2566 + +ALTER TABLE ps_endpoints ADD COLUMN max_audio_streams INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN max_video_streams INTEGER; + +UPDATE alembic_version SET version_num='39959b9c2566' WHERE alembic_version.version_num = 'd7983954dd96'; + +-- Running upgrade 39959b9c2566 -> 164abbd708c + +ALTER TABLE ps_endpoints MODIFY dtmf_mode ENUM('rfc4733','inband','info','auto','auto_info') NULL; + +UPDATE alembic_version SET version_num='164abbd708c' WHERE alembic_version.version_num = '39959b9c2566'; + +-- Running upgrade 164abbd708c -> 44ccced114ce + +ALTER TABLE ps_endpoints ADD COLUMN webrtc ENUM('yes','no'); + +UPDATE alembic_version SET version_num='44ccced114ce' WHERE alembic_version.version_num = '164abbd708c'; + +-- Running upgrade 44ccced114ce -> f3d1c5d38b56 + +ALTER TABLE ps_contacts ADD COLUMN prune_on_boot ENUM('yes','no'); + +UPDATE alembic_version SET version_num='f3d1c5d38b56' WHERE alembic_version.version_num = '44ccced114ce'; + +-- Running upgrade f3d1c5d38b56 -> b83645976fdd + +ALTER TABLE ps_endpoints ADD COLUMN dtls_fingerprint ENUM('SHA-1','SHA-256'); + +UPDATE alembic_version SET version_num='b83645976fdd' WHERE alembic_version.version_num = 'f3d1c5d38b56'; + +-- Running upgrade b83645976fdd -> a1698e8bb9c5 + +ALTER TABLE ps_endpoints ADD COLUMN incoming_mwi_mailbox VARCHAR(40); + +UPDATE alembic_version SET version_num='a1698e8bb9c5' WHERE alembic_version.version_num = 'b83645976fdd'; + +-- Running upgrade a1698e8bb9c5 -> 20abce6d1e3c + +ALTER TABLE ps_endpoints MODIFY identify_by ENUM('username','auth_username','ip') NULL; + +UPDATE alembic_version SET version_num='20abce6d1e3c' WHERE alembic_version.version_num = 'a1698e8bb9c5'; + +-- Running upgrade 20abce6d1e3c -> de83fac997e2 + +ALTER TABLE ps_endpoints ADD COLUMN bundle ENUM('yes','no'); + +UPDATE alembic_version SET version_num='de83fac997e2' WHERE alembic_version.version_num = '20abce6d1e3c'; + +-- Running upgrade de83fac997e2 -> 041c0d3d1857 + +ALTER TABLE ps_endpoints ADD COLUMN dtls_auto_generate_cert ENUM('yes','no'); + +UPDATE alembic_version SET version_num='041c0d3d1857' WHERE alembic_version.version_num = 'de83fac997e2'; + +-- Running upgrade 041c0d3d1857 -> e2f04d309071 + +ALTER TABLE queue_members ADD COLUMN wrapuptime INTEGER; + +UPDATE alembic_version SET version_num='e2f04d309071' WHERE alembic_version.version_num = '041c0d3d1857'; + +-- Running upgrade e2f04d309071 -> 52798ad97bdf + +ALTER TABLE ps_endpoints MODIFY identify_by VARCHAR(80) NULL; + +UPDATE alembic_version SET version_num='52798ad97bdf' WHERE alembic_version.version_num = 'e2f04d309071'; + +-- Running upgrade 52798ad97bdf -> d3e4284f8707 + +ALTER TABLE ps_subscription_persistence ADD COLUMN prune_on_boot ENUM('yes','no'); + +UPDATE alembic_version SET version_num='d3e4284f8707' WHERE alembic_version.version_num = '52798ad97bdf'; + +-- Running upgrade d3e4284f8707 -> 0be05c3a8225 + +ALTER TABLE ps_systems ADD COLUMN follow_early_media_fork ENUM('yes','no'); + +ALTER TABLE ps_systems ADD COLUMN accept_multiple_sdp_answers ENUM('yes','no'); + +ALTER TABLE ps_endpoints ADD COLUMN follow_early_media_fork ENUM('yes','no'); + +ALTER TABLE ps_endpoints ADD COLUMN accept_multiple_sdp_answers ENUM('yes','no'); + +UPDATE alembic_version SET version_num='0be05c3a8225' WHERE alembic_version.version_num = 'd3e4284f8707'; + +-- Running upgrade 0be05c3a8225 -> 19b00bc19b7b + +ALTER TABLE ps_endpoints ADD COLUMN suppress_q850_reason_header ENUM('yes','no'); + +UPDATE alembic_version SET version_num='19b00bc19b7b' WHERE alembic_version.version_num = '0be05c3a8225'; + +-- Running upgrade 19b00bc19b7b -> 1d3ed26d9978 + +ALTER TABLE ps_contacts MODIFY uri VARCHAR(511) NULL; + +UPDATE alembic_version SET version_num='1d3ed26d9978' WHERE alembic_version.version_num = '19b00bc19b7b'; + +-- Running upgrade 1d3ed26d9978 -> fe6592859b85 + +ALTER TABLE ps_endpoints MODIFY mwi_subscribe_replaces_unsolicited VARCHAR(5) NULL; + +ALTER TABLE ps_endpoints MODIFY mwi_subscribe_replaces_unsolicited ENUM('0','1','off','on','false','true','no','yes') NULL; + +UPDATE alembic_version SET version_num='fe6592859b85' WHERE alembic_version.version_num = '1d3ed26d9978'; + +-- Running upgrade fe6592859b85 -> 7f85dd44c775 + +ALTER TABLE ps_endpoints CHANGE suppress_q850_reason_header suppress_q850_reason_headers ENUM('yes','no') NULL; + +UPDATE alembic_version SET version_num='7f85dd44c775' WHERE alembic_version.version_num = 'fe6592859b85'; + +-- Running upgrade 7f85dd44c775 -> 465f47f880be + +ALTER TABLE ps_transports MODIFY protocol ENUM('udp','tcp','tls','ws','wss','flow') NULL; + +ALTER TABLE ps_auths MODIFY auth_type ENUM('md5','userpass','google_oauth') NULL; + +ALTER TABLE ps_registrations ADD COLUMN support_outbound ENUM('0','1','off','on','false','true','no','yes'); + +ALTER TABLE ps_registrations ADD COLUMN contact_header_params VARCHAR(255); + +ALTER TABLE ps_auths ADD COLUMN refresh_token VARCHAR(255); + +ALTER TABLE ps_auths ADD COLUMN oauth_clientid VARCHAR(255); + +ALTER TABLE ps_auths ADD COLUMN oauth_secret VARCHAR(255); + +UPDATE alembic_version SET version_num='465f47f880be' WHERE alembic_version.version_num = '7f85dd44c775'; + +-- Running upgrade 465f47f880be -> 2bb1a85135ad + +ALTER TABLE ps_globals ADD COLUMN use_callerid_contact ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='2bb1a85135ad' WHERE alembic_version.version_num = '465f47f880be'; + +-- Running upgrade 2bb1a85135ad -> 1ac563b350a8 + +ALTER TABLE ps_endpoints ADD COLUMN trust_connected_line ENUM('0','1','off','on','false','true','no','yes'); + +ALTER TABLE ps_endpoints ADD COLUMN send_connected_line ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='1ac563b350a8' WHERE alembic_version.version_num = '2bb1a85135ad'; + +-- Running upgrade 1ac563b350a8 -> 0838f8db6a61 + +ALTER TABLE ps_globals ADD COLUMN send_contact_status_on_update_registration ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='0838f8db6a61' WHERE alembic_version.version_num = '1ac563b350a8'; + +-- Running upgrade 0838f8db6a61 -> f3c0b8695b66 + +ALTER TABLE ps_globals ADD COLUMN taskprocessor_overload_trigger ENUM('none','global','pjsip_only'); + +UPDATE alembic_version SET version_num='f3c0b8695b66' WHERE alembic_version.version_num = '0838f8db6a61'; + +-- Running upgrade f3c0b8695b66 -> 80473bad3c16 + +ALTER TABLE ps_endpoints ADD COLUMN ignore_183_without_sdp ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='80473bad3c16' WHERE alembic_version.version_num = 'f3c0b8695b66'; + +-- Running upgrade 80473bad3c16 -> 3a094a18e75b + +ALTER TABLE ps_globals ADD COLUMN norefersub ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='3a094a18e75b' WHERE alembic_version.version_num = '80473bad3c16'; + +-- Running upgrade 3a094a18e75b -> fbb7766f17bc + +CREATE TABLE musiconhold_entry ( + name VARCHAR(80) NOT NULL, + position INTEGER NOT NULL, + entry VARCHAR(1024) NOT NULL, + PRIMARY KEY (name, position) +); + +ALTER TABLE musiconhold_entry ADD CONSTRAINT fk_musiconhold_entry_name_musiconhold FOREIGN KEY(name) REFERENCES musiconhold (name); + +ALTER TABLE musiconhold MODIFY mode ENUM('custom','files','mp3nb','quietmp3nb','quietmp3','playlist') NULL; + +UPDATE alembic_version SET version_num='fbb7766f17bc' WHERE alembic_version.version_num = '3a094a18e75b'; + +-- Running upgrade fbb7766f17bc -> 79290b511e4b + +ALTER TABLE ps_systems ADD COLUMN disable_rport ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='79290b511e4b' WHERE alembic_version.version_num = 'fbb7766f17bc'; + +-- Running upgrade 79290b511e4b -> b80485ff4dd0 + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_incoming_offer VARCHAR(128); + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_outgoing_offer VARCHAR(128); + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_incoming_answer VARCHAR(128); + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_outgoing_answer VARCHAR(128); + +UPDATE alembic_version SET version_num='b80485ff4dd0' WHERE alembic_version.version_num = '79290b511e4b'; + +-- Running upgrade b80485ff4dd0 -> 61797b9fced6 + +ALTER TABLE ps_endpoints ADD COLUMN stir_shaken ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='61797b9fced6' WHERE alembic_version.version_num = 'b80485ff4dd0'; + +-- Running upgrade 61797b9fced6 -> 1ae0609b6646 + +ALTER TABLE ps_contacts MODIFY reg_server VARCHAR(255) NULL; + +UPDATE alembic_version SET version_num='1ae0609b6646' WHERE alembic_version.version_num = '61797b9fced6'; + +-- Running upgrade 1ae0609b6646 -> e658c26033ca + +ALTER TABLE ps_endpoints ADD COLUMN send_history_info ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='e658c26033ca' WHERE alembic_version.version_num = '1ae0609b6646'; + +-- Running upgrade e658c26033ca -> 8915fcc5766f + +ALTER TABLE queue_members ADD COLUMN ringinuse ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='8915fcc5766f' WHERE alembic_version.version_num = 'e658c26033ca'; + +-- Running upgrade 8915fcc5766f -> c20d6e3992f4 + +ALTER TABLE ps_endpoints ADD COLUMN allow_unauthenticated_options ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='c20d6e3992f4' WHERE alembic_version.version_num = '8915fcc5766f'; + +-- Running upgrade c20d6e3992f4 -> f56d79a9f337 + +ALTER TABLE ps_aors ADD COLUMN remove_unavailable ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='f56d79a9f337' WHERE alembic_version.version_num = 'c20d6e3992f4'; + +-- Running upgrade f56d79a9f337 -> a06d8f8462d9 + +ALTER TABLE ps_endpoints ADD COLUMN t38_bind_udptl_to_media_address ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='a06d8f8462d9' WHERE alembic_version.version_num = 'f56d79a9f337'; + +-- Running upgrade a06d8f8462d9 -> 8f72185e437f + +ALTER TABLE ps_resource_list ADD COLUMN resource_display_name ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='8f72185e437f' WHERE alembic_version.version_num = 'a06d8f8462d9'; + +-- Running upgrade 8f72185e437f -> 0bee61aa9425 + +ALTER TABLE ps_globals ADD COLUMN allow_sending_180_after_183 ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='0bee61aa9425' WHERE alembic_version.version_num = '8f72185e437f'; + +-- Running upgrade 0bee61aa9425 -> 18e0805d367f + +ALTER TABLE ps_registrations ADD COLUMN max_random_initial_delay INTEGER; + +UPDATE alembic_version SET version_num='18e0805d367f' WHERE alembic_version.version_num = '0bee61aa9425'; + +-- Running upgrade 18e0805d367f -> 58e440314c2a + +ALTER TABLE ps_transports ADD COLUMN allow_wildcard_certs ENUM('yes','no'); + +UPDATE alembic_version SET version_num='58e440314c2a' WHERE alembic_version.version_num = '18e0805d367f'; + +-- Running upgrade 58e440314c2a -> 7197536bb68d + +ALTER TABLE ps_endpoints ADD COLUMN geoloc_incoming_call_profile VARCHAR(80); + +ALTER TABLE ps_endpoints ADD COLUMN geoloc_outgoing_call_profile VARCHAR(80); + +UPDATE alembic_version SET version_num='7197536bb68d' WHERE alembic_version.version_num = '58e440314c2a'; + +-- Running upgrade 7197536bb68d -> 9f3692b1654b + +ALTER TABLE ps_endpoints ADD COLUMN incoming_call_offer_pref ENUM('local','local_first','remote','remote_first'); + +ALTER TABLE ps_endpoints ADD COLUMN outgoing_call_offer_pref ENUM('local','local_merge','local_first','remote','remote_merge','remote_first'); + +ALTER TABLE ps_endpoints ADD COLUMN stir_shaken_profile VARCHAR(80); + +UPDATE alembic_version SET version_num='9f3692b1654b' WHERE alembic_version.version_num = '7197536bb68d'; + +-- Running upgrade 9f3692b1654b -> 539f68bede2c + +ALTER TABLE ps_endpoints MODIFY `100rel` ENUM('no','required','peer_supported','yes') NULL; + +UPDATE alembic_version SET version_num='539f68bede2c' WHERE alembic_version.version_num = '9f3692b1654b'; + +-- Running upgrade 539f68bede2c -> 417c0247fd7e + +ALTER TABLE ps_endpoints ADD COLUMN security_negotiation ENUM('no','mediasec'); + +ALTER TABLE ps_endpoints ADD COLUMN security_mechanisms VARCHAR(512); + +ALTER TABLE ps_registrations ADD COLUMN security_negotiation ENUM('no','mediasec'); + +ALTER TABLE ps_registrations ADD COLUMN security_mechanisms VARCHAR(512); + +UPDATE alembic_version SET version_num='417c0247fd7e' WHERE alembic_version.version_num = '539f68bede2c'; + +-- Running upgrade 417c0247fd7e -> ccf795ee535f + +ALTER TABLE ps_globals ADD COLUMN all_codecs_on_empty_reinvite ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='ccf795ee535f' WHERE alembic_version.version_num = '417c0247fd7e'; + +-- Running upgrade ccf795ee535f -> 5a2247c957d2 + +ALTER TABLE ps_endpoints ADD COLUMN send_aoc ENUM('0','1','off','on','false','true','no','yes'); + +UPDATE alembic_version SET version_num='5a2247c957d2' WHERE alembic_version.version_num = 'ccf795ee535f'; + +-- Running upgrade 5a2247c957d2 -> f261363a857f + +ALTER TABLE ps_endpoints ADD COLUMN overlap_context VARCHAR(80); + +UPDATE alembic_version SET version_num='f261363a857f' WHERE alembic_version.version_num = '5a2247c957d2'; + diff --git a/contrib/realtime/mysql/mysql_voicemail.sql b/contrib/realtime/mysql/mysql_voicemail.sql new file mode 100644 index 0000000000..b30456e41c --- /dev/null +++ b/contrib/realtime/mysql/mysql_voicemail.sql @@ -0,0 +1,35 @@ +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL, + CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num) +); + +-- Running upgrade -> a2e9769475e + +CREATE TABLE voicemail_messages ( + dir VARCHAR(255) NOT NULL, + msgnum INTEGER NOT NULL, + context VARCHAR(80), + macrocontext VARCHAR(80), + callerid VARCHAR(80), + origtime INTEGER, + duration INTEGER, + recording BLOB, + flag VARCHAR(30), + category VARCHAR(30), + mailboxuser VARCHAR(30), + mailboxcontext VARCHAR(30), + msg_id VARCHAR(40) +); + +ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum); + +CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir); + +INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e'); + +-- Running upgrade a2e9769475e -> 39428242f7f5 + +ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL; + +UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e'; + diff --git a/contrib/realtime/postgresql/postgresql_cdr.sql b/contrib/realtime/postgresql/postgresql_cdr.sql new file mode 100644 index 0000000000..79380171a8 --- /dev/null +++ b/contrib/realtime/postgresql/postgresql_cdr.sql @@ -0,0 +1,45 @@ +BEGIN; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL, + CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num) +); + +-- Running upgrade -> 210693f3123d + +CREATE TABLE cdr ( + accountcode VARCHAR(20), + src VARCHAR(80), + dst VARCHAR(80), + dcontext VARCHAR(80), + clid VARCHAR(80), + channel VARCHAR(80), + dstchannel VARCHAR(80), + lastapp VARCHAR(80), + lastdata VARCHAR(80), + start TIMESTAMP WITHOUT TIME ZONE, + answer TIMESTAMP WITHOUT TIME ZONE, + "end" TIMESTAMP WITHOUT TIME ZONE, + duration INTEGER, + billsec INTEGER, + disposition VARCHAR(45), + amaflags VARCHAR(45), + userfield VARCHAR(256), + uniqueid VARCHAR(150), + linkedid VARCHAR(150), + peeraccount VARCHAR(20), + sequence INTEGER +); + +INSERT INTO alembic_version (version_num) VALUES ('210693f3123d'); + +-- Running upgrade 210693f3123d -> 54cde9847798 + +ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80); + +ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80); + +UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d'; + +COMMIT; + diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql new file mode 100644 index 0000000000..aa14f92a81 --- /dev/null +++ b/contrib/realtime/postgresql/postgresql_config.sql @@ -0,0 +1,1530 @@ +BEGIN; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL, + CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num) +); + +-- Running upgrade -> 4da0c5f79a9c + +CREATE TYPE type_values AS ENUM ('friend', 'user', 'peer'); + +CREATE TYPE sip_transport_values AS ENUM ('udp', 'tcp', 'tls', 'ws', 'wss', 'udp,tcp', 'tcp,udp'); + +CREATE TYPE sip_dtmfmode_values AS ENUM ('rfc2833', 'info', 'shortinfo', 'inband', 'auto'); + +CREATE TYPE sip_directmedia_values AS ENUM ('yes', 'no', 'nonat', 'update'); + +CREATE TYPE yes_no_values AS ENUM ('yes', 'no'); + +CREATE TYPE sip_progressinband_values AS ENUM ('yes', 'no', 'never'); + +CREATE TYPE sip_session_timers_values AS ENUM ('accept', 'refuse', 'originate'); + +CREATE TYPE sip_session_refresher_values AS ENUM ('uac', 'uas'); + +CREATE TYPE sip_callingpres_values AS ENUM ('allowed_not_screened', 'allowed_passed_screen', 'allowed_failed_screen', 'allowed', 'prohib_not_screened', 'prohib_passed_screen', 'prohib_failed_screen', 'prohib'); + +CREATE TABLE sippeers ( + id SERIAL NOT NULL, + name VARCHAR(40) NOT NULL, + ipaddr VARCHAR(45), + port INTEGER, + regseconds INTEGER, + defaultuser VARCHAR(40), + fullcontact VARCHAR(80), + regserver VARCHAR(20), + useragent VARCHAR(20), + lastms INTEGER, + host VARCHAR(40), + type type_values, + context VARCHAR(40), + permit VARCHAR(95), + deny VARCHAR(95), + secret VARCHAR(40), + md5secret VARCHAR(40), + remotesecret VARCHAR(40), + transport sip_transport_values, + dtmfmode sip_dtmfmode_values, + directmedia sip_directmedia_values, + nat VARCHAR(29), + callgroup VARCHAR(40), + pickupgroup VARCHAR(40), + language VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + insecure VARCHAR(40), + trustrpid yes_no_values, + progressinband sip_progressinband_values, + promiscredir yes_no_values, + useclientcode yes_no_values, + accountcode VARCHAR(40), + setvar VARCHAR(200), + callerid VARCHAR(40), + amaflags VARCHAR(40), + callcounter yes_no_values, + busylevel INTEGER, + allowoverlap yes_no_values, + allowsubscribe yes_no_values, + videosupport yes_no_values, + maxcallbitrate INTEGER, + rfc2833compensate yes_no_values, + mailbox VARCHAR(40), + "session-timers" sip_session_timers_values, + "session-expires" INTEGER, + "session-minse" INTEGER, + "session-refresher" sip_session_refresher_values, + t38pt_usertpsource VARCHAR(40), + regexten VARCHAR(40), + fromdomain VARCHAR(40), + fromuser VARCHAR(40), + qualify VARCHAR(40), + defaultip VARCHAR(45), + rtptimeout INTEGER, + rtpholdtimeout INTEGER, + sendrpid yes_no_values, + outboundproxy VARCHAR(40), + callbackextension VARCHAR(40), + timert1 INTEGER, + timerb INTEGER, + qualifyfreq INTEGER, + constantssrc yes_no_values, + contactpermit VARCHAR(95), + contactdeny VARCHAR(95), + usereqphone yes_no_values, + textsupport yes_no_values, + faxdetect yes_no_values, + buggymwi yes_no_values, + auth VARCHAR(40), + fullname VARCHAR(40), + trunkname VARCHAR(40), + cid_number VARCHAR(40), + callingpres sip_callingpres_values, + mohinterpret VARCHAR(40), + mohsuggest VARCHAR(40), + parkinglot VARCHAR(40), + hasvoicemail yes_no_values, + subscribemwi yes_no_values, + vmexten VARCHAR(40), + autoframing yes_no_values, + rtpkeepalive INTEGER, + "call-limit" INTEGER, + g726nonstandard yes_no_values, + ignoresdpversion yes_no_values, + allowtransfer yes_no_values, + dynamic yes_no_values, + path VARCHAR(256), + supportpath yes_no_values, + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX sippeers_name ON sippeers (name); + +CREATE INDEX sippeers_name_host ON sippeers (name, host); + +CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port); + +CREATE INDEX sippeers_host_port ON sippeers (host, port); + +CREATE TYPE iax_requirecalltoken_values AS ENUM ('yes', 'no', 'auto'); + +CREATE TYPE iax_encryption_values AS ENUM ('yes', 'no', 'aes128'); + +CREATE TYPE iax_transfer_values AS ENUM ('yes', 'no', 'mediaonly'); + +CREATE TABLE iaxfriends ( + id SERIAL NOT NULL, + name VARCHAR(40) NOT NULL, + type type_values, + username VARCHAR(40), + mailbox VARCHAR(40), + secret VARCHAR(40), + dbsecret VARCHAR(40), + context VARCHAR(40), + regcontext VARCHAR(40), + host VARCHAR(40), + ipaddr VARCHAR(40), + port INTEGER, + defaultip VARCHAR(20), + sourceaddress VARCHAR(20), + mask VARCHAR(20), + regexten VARCHAR(40), + regseconds INTEGER, + accountcode VARCHAR(20), + mohinterpret VARCHAR(20), + mohsuggest VARCHAR(20), + inkeys VARCHAR(40), + outkeys VARCHAR(40), + language VARCHAR(10), + callerid VARCHAR(100), + cid_number VARCHAR(40), + sendani yes_no_values, + fullname VARCHAR(40), + trunk yes_no_values, + auth VARCHAR(20), + maxauthreq INTEGER, + requirecalltoken iax_requirecalltoken_values, + encryption iax_encryption_values, + transfer iax_transfer_values, + jitterbuffer yes_no_values, + forcejitterbuffer yes_no_values, + disallow VARCHAR(200), + allow VARCHAR(200), + codecpriority VARCHAR(40), + qualify VARCHAR(10), + qualifysmoothing yes_no_values, + qualifyfreqok VARCHAR(10), + qualifyfreqnotok VARCHAR(10), + timezone VARCHAR(20), + adsi yes_no_values, + amaflags VARCHAR(20), + setvar VARCHAR(200), + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX iaxfriends_name ON iaxfriends (name); + +CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host); + +CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port); + +CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port); + +CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port); + +CREATE TABLE voicemail ( + uniqueid SERIAL NOT NULL, + context VARCHAR(80) NOT NULL, + mailbox VARCHAR(80) NOT NULL, + password VARCHAR(80) NOT NULL, + fullname VARCHAR(80), + alias VARCHAR(80), + email VARCHAR(80), + pager VARCHAR(80), + attach yes_no_values, + attachfmt VARCHAR(10), + serveremail VARCHAR(80), + language VARCHAR(20), + tz VARCHAR(30), + deletevoicemail yes_no_values, + saycid yes_no_values, + sendvoicemail yes_no_values, + review yes_no_values, + tempgreetwarn yes_no_values, + operator yes_no_values, + envelope yes_no_values, + sayduration INTEGER, + forcename yes_no_values, + forcegreetings yes_no_values, + callback VARCHAR(80), + dialout VARCHAR(80), + exitcontext VARCHAR(80), + maxmsg INTEGER, + volgain NUMERIC(5, 2), + imapuser VARCHAR(80), + imappassword VARCHAR(80), + imapserver VARCHAR(80), + imapport VARCHAR(8), + imapflags VARCHAR(80), + stamp TIMESTAMP WITHOUT TIME ZONE, + PRIMARY KEY (uniqueid) +); + +CREATE INDEX voicemail_mailbox ON voicemail (mailbox); + +CREATE INDEX voicemail_context ON voicemail (context); + +CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context); + +CREATE INDEX voicemail_imapuser ON voicemail (imapuser); + +CREATE TABLE meetme ( + bookid SERIAL NOT NULL, + confno VARCHAR(80) NOT NULL, + starttime TIMESTAMP WITHOUT TIME ZONE, + endtime TIMESTAMP WITHOUT TIME ZONE, + pin VARCHAR(20), + adminpin VARCHAR(20), + opts VARCHAR(20), + adminopts VARCHAR(20), + recordingfilename VARCHAR(80), + recordingformat VARCHAR(10), + maxusers INTEGER, + members INTEGER NOT NULL, + PRIMARY KEY (bookid) +); + +CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime); + +CREATE TYPE moh_mode_values AS ENUM ('custom', 'files', 'mp3nb', 'quietmp3nb', 'quietmp3'); + +CREATE TABLE musiconhold ( + name VARCHAR(80) NOT NULL, + mode moh_mode_values, + directory VARCHAR(255), + application VARCHAR(255), + digit VARCHAR(1), + sort VARCHAR(10), + format VARCHAR(10), + stamp TIMESTAMP WITHOUT TIME ZONE, + PRIMARY KEY (name) +); + +INSERT INTO alembic_version (version_num) VALUES ('4da0c5f79a9c'); + +-- Running upgrade 4da0c5f79a9c -> 43956d550a44 + +CREATE TYPE yesno_values AS ENUM ('yes', 'no'); + +CREATE TYPE pjsip_connected_line_method_values AS ENUM ('invite', 'reinvite', 'update'); + +CREATE TYPE pjsip_direct_media_glare_mitigation_values AS ENUM ('none', 'outgoing', 'incoming'); + +CREATE TYPE pjsip_dtmf_mode_values AS ENUM ('rfc4733', 'inband', 'info'); + +CREATE TYPE pjsip_identify_by_values AS ENUM ('username'); + +CREATE TYPE pjsip_timer_values AS ENUM ('forced', 'no', 'required', 'yes'); + +CREATE TYPE pjsip_cid_privacy_values AS ENUM ('allowed_not_screened', 'allowed_passed_screened', 'allowed_failed_screened', 'allowed', 'prohib_not_screened', 'prohib_passed_screened', 'prohib_failed_screened', 'prohib', 'unavailable'); + +CREATE TYPE pjsip_100rel_values AS ENUM ('no', 'required', 'yes'); + +CREATE TYPE pjsip_media_encryption_values AS ENUM ('no', 'sdes', 'dtls'); + +CREATE TYPE pjsip_t38udptl_ec_values AS ENUM ('none', 'fec', 'redundancy'); + +CREATE TYPE pjsip_dtls_setup_values AS ENUM ('active', 'passive', 'actpass'); + +CREATE TABLE ps_endpoints ( + id VARCHAR(40) NOT NULL, + transport VARCHAR(40), + aors VARCHAR(200), + auth VARCHAR(40), + context VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + direct_media yesno_values, + connected_line_method pjsip_connected_line_method_values, + direct_media_method pjsip_connected_line_method_values, + direct_media_glare_mitigation pjsip_direct_media_glare_mitigation_values, + disable_direct_media_on_nat yesno_values, + dtmf_mode pjsip_dtmf_mode_values, + external_media_address VARCHAR(40), + force_rport yesno_values, + ice_support yesno_values, + identify_by pjsip_identify_by_values, + mailboxes VARCHAR(40), + moh_suggest VARCHAR(40), + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + rewrite_contact yesno_values, + rtp_ipv6 yesno_values, + rtp_symmetric yesno_values, + send_diversion yesno_values, + send_pai yesno_values, + send_rpid yesno_values, + timers_min_se INTEGER, + timers pjsip_timer_values, + timers_sess_expires INTEGER, + callerid VARCHAR(40), + callerid_privacy pjsip_cid_privacy_values, + callerid_tag VARCHAR(40), + "100rel" pjsip_100rel_values, + aggregate_mwi yesno_values, + trust_id_inbound yesno_values, + trust_id_outbound yesno_values, + use_ptime yesno_values, + use_avpf yesno_values, + media_encryption pjsip_media_encryption_values, + inband_progress yesno_values, + call_group VARCHAR(40), + pickup_group VARCHAR(40), + named_call_group VARCHAR(40), + named_pickup_group VARCHAR(40), + device_state_busy_at INTEGER, + fax_detect yesno_values, + t38_udptl yesno_values, + t38_udptl_ec pjsip_t38udptl_ec_values, + t38_udptl_maxdatagram INTEGER, + t38_udptl_nat yesno_values, + t38_udptl_ipv6 yesno_values, + tone_zone VARCHAR(40), + language VARCHAR(40), + one_touch_recording yesno_values, + record_on_feature VARCHAR(40), + record_off_feature VARCHAR(40), + rtp_engine VARCHAR(40), + allow_transfer yesno_values, + allow_subscribe yesno_values, + sdp_owner VARCHAR(40), + sdp_session VARCHAR(40), + tos_audio INTEGER, + tos_video INTEGER, + cos_audio INTEGER, + cos_video INTEGER, + sub_min_expiry INTEGER, + from_domain VARCHAR(40), + from_user VARCHAR(40), + mwi_fromuser VARCHAR(40), + dtls_verify VARCHAR(40), + dtls_rekey VARCHAR(40), + dtls_cert_file VARCHAR(200), + dtls_private_key VARCHAR(200), + dtls_cipher VARCHAR(200), + dtls_ca_file VARCHAR(200), + dtls_ca_path VARCHAR(200), + dtls_setup pjsip_dtls_setup_values, + srtp_tag_32 yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_endpoints_id ON ps_endpoints (id); + +CREATE TYPE pjsip_auth_type_values AS ENUM ('md5', 'userpass'); + +CREATE TABLE ps_auths ( + id VARCHAR(40) NOT NULL, + auth_type pjsip_auth_type_values, + nonce_lifetime INTEGER, + md5_cred VARCHAR(40), + password VARCHAR(80), + realm VARCHAR(40), + username VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_auths_id ON ps_auths (id); + +CREATE TABLE ps_aors ( + id VARCHAR(40) NOT NULL, + contact VARCHAR(40), + default_expiration INTEGER, + mailboxes VARCHAR(80), + max_contacts INTEGER, + minimum_expiration INTEGER, + remove_existing yesno_values, + qualify_frequency INTEGER, + authenticate_qualify yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_aors_id ON ps_aors (id); + +CREATE TABLE ps_contacts ( + id VARCHAR(40) NOT NULL, + uri VARCHAR(40), + expiration_time VARCHAR(40), + qualify_frequency INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_contacts_id ON ps_contacts (id); + +CREATE TABLE ps_domain_aliases ( + id VARCHAR(40) NOT NULL, + domain VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id); + +CREATE TABLE ps_endpoint_id_ips ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40), + match VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id); + +UPDATE alembic_version SET version_num='43956d550a44' WHERE alembic_version.version_num = '4da0c5f79a9c'; + +-- Running upgrade 43956d550a44 -> 581a4264e537 + +CREATE TABLE extensions ( + id BIGSERIAL NOT NULL, + context VARCHAR(40) NOT NULL, + exten VARCHAR(40) NOT NULL, + priority INTEGER NOT NULL, + app VARCHAR(40) NOT NULL, + appdata VARCHAR(256) NOT NULL, + PRIMARY KEY (id), + UNIQUE (context, exten, priority), + UNIQUE (id) +); + +UPDATE alembic_version SET version_num='581a4264e537' WHERE alembic_version.version_num = '43956d550a44'; + +-- Running upgrade 581a4264e537 -> 2fc7930b41b3 + +CREATE TYPE pjsip_redirect_method_values AS ENUM ('user', 'uri_core', 'uri_pjsip'); + +CREATE TABLE ps_systems ( + id VARCHAR(40) NOT NULL, + timer_t1 INTEGER, + timer_b INTEGER, + compact_headers yesno_values, + threadpool_initial_size INTEGER, + threadpool_auto_increment INTEGER, + threadpool_idle_timeout INTEGER, + threadpool_max_size INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_systems_id ON ps_systems (id); + +CREATE TABLE ps_globals ( + id VARCHAR(40) NOT NULL, + max_forwards INTEGER, + user_agent VARCHAR(40), + default_outbound_endpoint VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_globals_id ON ps_globals (id); + +CREATE TYPE pjsip_transport_method_values AS ENUM ('default', 'unspecified', 'tlsv1', 'sslv2', 'sslv3', 'sslv23'); + +CREATE TYPE pjsip_transport_protocol_values AS ENUM ('udp', 'tcp', 'tls', 'ws', 'wss'); + +CREATE TABLE ps_transports ( + id VARCHAR(40) NOT NULL, + async_operations INTEGER, + bind VARCHAR(40), + ca_list_file VARCHAR(200), + cert_file VARCHAR(200), + cipher VARCHAR(200), + domain VARCHAR(40), + external_media_address VARCHAR(40), + external_signaling_address VARCHAR(40), + external_signaling_port INTEGER, + method pjsip_transport_method_values, + local_net VARCHAR(40), + password VARCHAR(40), + priv_key_file VARCHAR(200), + protocol pjsip_transport_protocol_values, + require_client_cert yesno_values, + verify_client yesno_values, + verifiy_server yesno_values, + tos yesno_values, + cos yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_transports_id ON ps_transports (id); + +CREATE TABLE ps_registrations ( + id VARCHAR(40) NOT NULL, + auth_rejection_permanent yesno_values, + client_uri VARCHAR(40), + contact_user VARCHAR(40), + expiration INTEGER, + max_retries INTEGER, + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + retry_interval INTEGER, + forbidden_retry_interval INTEGER, + server_uri VARCHAR(40), + transport VARCHAR(40), + support_path yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_registrations_id ON ps_registrations (id); + +ALTER TABLE ps_endpoints ADD COLUMN media_address VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN redirect_method pjsip_redirect_method_values; + +ALTER TABLE ps_endpoints ADD COLUMN set_var TEXT; + +ALTER TABLE ps_endpoints RENAME mwi_fromuser TO mwi_from_user; + +ALTER TABLE ps_contacts ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN path TEXT; + +ALTER TABLE ps_aors ADD COLUMN maximum_expiration INTEGER; + +ALTER TABLE ps_aors ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_aors ADD COLUMN support_path yesno_values; + +UPDATE alembic_version SET version_num='2fc7930b41b3' WHERE alembic_version.version_num = '581a4264e537'; + +-- Running upgrade 2fc7930b41b3 -> 21e526ad3040 + +ALTER TABLE ps_globals ADD COLUMN debug VARCHAR(40); + +UPDATE alembic_version SET version_num='21e526ad3040' WHERE alembic_version.version_num = '2fc7930b41b3'; + +-- Running upgrade 21e526ad3040 -> 28887f25a46f + +CREATE TYPE queue_autopause_values AS ENUM ('yes', 'no', 'all'); + +CREATE TYPE queue_strategy_values AS ENUM ('ringall', 'leastrecent', 'fewestcalls', 'random', 'rrmemory', 'linear', 'wrandom', 'rrordered'); + +CREATE TABLE queues ( + name VARCHAR(128) NOT NULL, + musiconhold VARCHAR(128), + announce VARCHAR(128), + context VARCHAR(128), + timeout INTEGER, + ringinuse yesno_values, + setinterfacevar yesno_values, + setqueuevar yesno_values, + setqueueentryvar yesno_values, + monitor_format VARCHAR(8), + membermacro VARCHAR(512), + membergosub VARCHAR(512), + queue_youarenext VARCHAR(128), + queue_thereare VARCHAR(128), + queue_callswaiting VARCHAR(128), + queue_quantity1 VARCHAR(128), + queue_quantity2 VARCHAR(128), + queue_holdtime VARCHAR(128), + queue_minutes VARCHAR(128), + queue_minute VARCHAR(128), + queue_seconds VARCHAR(128), + queue_thankyou VARCHAR(128), + queue_callerannounce VARCHAR(128), + queue_reporthold VARCHAR(128), + announce_frequency INTEGER, + announce_to_first_user yesno_values, + min_announce_frequency INTEGER, + announce_round_seconds INTEGER, + announce_holdtime VARCHAR(128), + announce_position VARCHAR(128), + announce_position_limit INTEGER, + periodic_announce VARCHAR(50), + periodic_announce_frequency INTEGER, + relative_periodic_announce yesno_values, + random_periodic_announce yesno_values, + retry INTEGER, + wrapuptime INTEGER, + penaltymemberslimit INTEGER, + autofill yesno_values, + monitor_type VARCHAR(128), + autopause queue_autopause_values, + autopausedelay INTEGER, + autopausebusy yesno_values, + autopauseunavail yesno_values, + maxlen INTEGER, + servicelevel INTEGER, + strategy queue_strategy_values, + joinempty VARCHAR(128), + leavewhenempty VARCHAR(128), + reportholdtime yesno_values, + memberdelay INTEGER, + weight INTEGER, + timeoutrestart yesno_values, + defaultrule VARCHAR(128), + timeoutpriority VARCHAR(128), + PRIMARY KEY (name) +); + +CREATE TABLE queue_members ( + queue_name VARCHAR(80) NOT NULL, + interface VARCHAR(80) NOT NULL, + uniqueid VARCHAR(80) NOT NULL, + membername VARCHAR(80), + state_interface VARCHAR(80), + penalty INTEGER, + paused INTEGER, + PRIMARY KEY (queue_name, interface) +); + +UPDATE alembic_version SET version_num='28887f25a46f' WHERE alembic_version.version_num = '21e526ad3040'; + +-- Running upgrade 28887f25a46f -> 4c573e7135bd + +ALTER TABLE ps_endpoints ALTER COLUMN tos_audio TYPE VARCHAR(10); + +ALTER TABLE ps_endpoints ALTER COLUMN tos_video TYPE VARCHAR(10); + +ALTER TABLE ps_endpoints DROP COLUMN cos_audio; + +ALTER TABLE ps_endpoints DROP COLUMN cos_video; + +ALTER TABLE ps_endpoints ADD COLUMN cos_audio INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN cos_video INTEGER; + +ALTER TABLE ps_transports ALTER COLUMN tos TYPE VARCHAR(10); + +ALTER TABLE ps_transports DROP COLUMN cos; + +ALTER TABLE ps_transports ADD COLUMN cos INTEGER; + +UPDATE alembic_version SET version_num='4c573e7135bd' WHERE alembic_version.version_num = '28887f25a46f'; + +-- Running upgrade 4c573e7135bd -> 3855ee4e5f85 + +ALTER TABLE ps_endpoints ADD COLUMN message_context VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN user_agent VARCHAR(40); + +UPDATE alembic_version SET version_num='3855ee4e5f85' WHERE alembic_version.version_num = '4c573e7135bd'; + +-- Running upgrade 3855ee4e5f85 -> e96a0b8071c + +ALTER TABLE ps_globals ALTER COLUMN user_agent TYPE VARCHAR(255); + +ALTER TABLE ps_contacts ALTER COLUMN id TYPE VARCHAR(255); + +ALTER TABLE ps_contacts ALTER COLUMN uri TYPE VARCHAR(255); + +ALTER TABLE ps_contacts ALTER COLUMN user_agent TYPE VARCHAR(255); + +ALTER TABLE ps_registrations ALTER COLUMN client_uri TYPE VARCHAR(255); + +ALTER TABLE ps_registrations ALTER COLUMN server_uri TYPE VARCHAR(255); + +UPDATE alembic_version SET version_num='e96a0b8071c' WHERE alembic_version.version_num = '3855ee4e5f85'; + +-- Running upgrade e96a0b8071c -> c6d929b23a8 + +CREATE TABLE ps_subscription_persistence ( + id VARCHAR(40) NOT NULL, + packet VARCHAR(2048), + src_name VARCHAR(128), + src_port INTEGER, + transport_key VARCHAR(64), + local_name VARCHAR(128), + local_port INTEGER, + cseq INTEGER, + tag VARCHAR(128), + endpoint VARCHAR(40), + expires INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id); + +UPDATE alembic_version SET version_num='c6d929b23a8' WHERE alembic_version.version_num = 'e96a0b8071c'; + +-- Running upgrade c6d929b23a8 -> 51f8cb66540e + +ALTER TABLE ps_endpoints ADD COLUMN force_avp yesno_values; + +ALTER TABLE ps_endpoints ADD COLUMN media_use_received_transport yesno_values; + +UPDATE alembic_version SET version_num='51f8cb66540e' WHERE alembic_version.version_num = 'c6d929b23a8'; + +-- Running upgrade 51f8cb66540e -> 1d50859ed02e + +ALTER TABLE ps_endpoints ADD COLUMN accountcode VARCHAR(20); + +UPDATE alembic_version SET version_num='1d50859ed02e' WHERE alembic_version.version_num = '51f8cb66540e'; + +-- Running upgrade 1d50859ed02e -> 1758e8bbf6b + +ALTER TABLE sippeers ALTER COLUMN useragent TYPE VARCHAR(255); + +UPDATE alembic_version SET version_num='1758e8bbf6b' WHERE alembic_version.version_num = '1d50859ed02e'; + +-- Running upgrade 1758e8bbf6b -> 5139253c0423 + +ALTER TABLE queue_members DROP COLUMN uniqueid; + +ALTER TABLE queue_members ADD COLUMN uniqueid INTEGER NOT NULL; + +ALTER TABLE queue_members ADD UNIQUE (uniqueid); + +UPDATE alembic_version SET version_num='5139253c0423' WHERE alembic_version.version_num = '1758e8bbf6b'; + +-- Running upgrade 5139253c0423 -> d39508cb8d8 + +CREATE TABLE queue_rules ( + rule_name VARCHAR(80) NOT NULL, + time VARCHAR(32) NOT NULL, + min_penalty VARCHAR(32) NOT NULL, + max_penalty VARCHAR(32) NOT NULL +); + +UPDATE alembic_version SET version_num='d39508cb8d8' WHERE alembic_version.version_num = '5139253c0423'; + +-- Running upgrade d39508cb8d8 -> 5950038a6ead + +ALTER TABLE ps_transports ALTER COLUMN verifiy_server TYPE yesno_values; + +ALTER TABLE ps_transports RENAME verifiy_server TO verify_server; + +UPDATE alembic_version SET version_num='5950038a6ead' WHERE alembic_version.version_num = 'd39508cb8d8'; + +-- Running upgrade 5950038a6ead -> 10aedae86a32 + +CREATE TYPE sip_directmedia_values_v2 AS ENUM ('yes', 'no', 'nonat', 'update', 'outgoing'); + +ALTER TABLE sippeers ALTER COLUMN directmedia TYPE sip_directmedia_values_v2 USING directmedia::text::sip_directmedia_values_v2; + +DROP TYPE sip_directmedia_values; + +UPDATE alembic_version SET version_num='10aedae86a32' WHERE alembic_version.version_num = '5950038a6ead'; + +-- Running upgrade 10aedae86a32 -> 371a3bf4143e + +ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone yesno_values; + +UPDATE alembic_version SET version_num='371a3bf4143e' WHERE alembic_version.version_num = '10aedae86a32'; + +-- Running upgrade 371a3bf4143e -> 15b1430ad6f1 + +ALTER TABLE ps_endpoints ADD COLUMN moh_passthrough yesno_values; + +UPDATE alembic_version SET version_num='15b1430ad6f1' WHERE alembic_version.version_num = '371a3bf4143e'; + +-- Running upgrade 15b1430ad6f1 -> 945b1098bdd + +ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic yesno_values; + +UPDATE alembic_version SET version_num='945b1098bdd' WHERE alembic_version.version_num = '15b1430ad6f1'; + +-- Running upgrade 945b1098bdd -> 45e3f47c6c44 + +ALTER TABLE ps_globals ADD COLUMN endpoint_identifier_order VARCHAR(40); + +UPDATE alembic_version SET version_num='45e3f47c6c44' WHERE alembic_version.version_num = '945b1098bdd'; + +-- Running upgrade 45e3f47c6c44 -> 23530d604b96 + +ALTER TABLE ps_endpoints ADD COLUMN rpid_immediate yesno_values; + +UPDATE alembic_version SET version_num='23530d604b96' WHERE alembic_version.version_num = '45e3f47c6c44'; + +-- Running upgrade 23530d604b96 -> 31cd4f4891ec + +CREATE TYPE pjsip_dtmf_mode_values_v2 AS ENUM ('rfc4733', 'inband', 'info', 'auto'); + +ALTER TABLE ps_endpoints ALTER COLUMN dtmf_mode TYPE pjsip_dtmf_mode_values_v2 USING dtmf_mode::text::pjsip_dtmf_mode_values_v2; + +DROP TYPE pjsip_dtmf_mode_values; + +UPDATE alembic_version SET version_num='31cd4f4891ec' WHERE alembic_version.version_num = '23530d604b96'; + +-- Running upgrade 31cd4f4891ec -> 461d7d691209 + +ALTER TABLE ps_aors ADD COLUMN qualify_timeout INTEGER; + +ALTER TABLE ps_contacts ADD COLUMN qualify_timeout INTEGER; + +UPDATE alembic_version SET version_num='461d7d691209' WHERE alembic_version.version_num = '31cd4f4891ec'; + +-- Running upgrade 461d7d691209 -> a541e0b5e89 + +ALTER TABLE ps_globals ADD COLUMN max_initial_qualify_time INTEGER; + +UPDATE alembic_version SET version_num='a541e0b5e89' WHERE alembic_version.version_num = '461d7d691209'; + +-- Running upgrade a541e0b5e89 -> 28b8e71e541f + +ALTER TABLE ps_endpoints ADD COLUMN g726_non_standard yesno_values; + +UPDATE alembic_version SET version_num='28b8e71e541f' WHERE alembic_version.version_num = 'a541e0b5e89'; + +-- Running upgrade 28b8e71e541f -> 498357a710ae + +ALTER TABLE ps_endpoints ADD COLUMN rtp_keepalive INTEGER; + +UPDATE alembic_version SET version_num='498357a710ae' WHERE alembic_version.version_num = '28b8e71e541f'; + +-- Running upgrade 498357a710ae -> 26f10cadc157 + +ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout_hold INTEGER; + +UPDATE alembic_version SET version_num='26f10cadc157' WHERE alembic_version.version_num = '498357a710ae'; + +-- Running upgrade 26f10cadc157 -> 154177371065 + +ALTER TABLE ps_globals ADD COLUMN default_from_user VARCHAR(80); + +UPDATE alembic_version SET version_num='154177371065' WHERE alembic_version.version_num = '26f10cadc157'; + +-- Running upgrade 154177371065 -> 28ce1e718f05 + +ALTER TABLE ps_registrations ADD COLUMN fatal_retry_interval INTEGER; + +UPDATE alembic_version SET version_num='28ce1e718f05' WHERE alembic_version.version_num = '154177371065'; + +-- Running upgrade 28ce1e718f05 -> 339a3bdf53fc + +ALTER TABLE ps_endpoints ALTER COLUMN accountcode TYPE VARCHAR(80); + +ALTER TABLE sippeers ALTER COLUMN accountcode TYPE VARCHAR(80); + +ALTER TABLE iaxfriends ALTER COLUMN accountcode TYPE VARCHAR(80); + +UPDATE alembic_version SET version_num='339a3bdf53fc' WHERE alembic_version.version_num = '28ce1e718f05'; + +-- Running upgrade 339a3bdf53fc -> 189a235b3fd7 + +ALTER TABLE ps_globals ADD COLUMN keep_alive_interval INTEGER; + +UPDATE alembic_version SET version_num='189a235b3fd7' WHERE alembic_version.version_num = '339a3bdf53fc'; + +-- Running upgrade 189a235b3fd7 -> 2d078ec071b7 + +ALTER TABLE ps_aors ALTER COLUMN contact TYPE VARCHAR(255); + +UPDATE alembic_version SET version_num='2d078ec071b7' WHERE alembic_version.version_num = '189a235b3fd7'; + +-- Running upgrade 2d078ec071b7 -> 26d7f3bf0fa5 + +ALTER TABLE ps_endpoints ADD COLUMN bind_rtp_to_media_address yesno_values; + +UPDATE alembic_version SET version_num='26d7f3bf0fa5' WHERE alembic_version.version_num = '2d078ec071b7'; + +-- Running upgrade 26d7f3bf0fa5 -> 136885b81223 + +ALTER TABLE ps_globals ADD COLUMN regcontext VARCHAR(80); + +UPDATE alembic_version SET version_num='136885b81223' WHERE alembic_version.version_num = '26d7f3bf0fa5'; + +-- Running upgrade 136885b81223 -> 423f34ad36e2 + +ALTER TABLE ps_aors ALTER COLUMN qualify_timeout TYPE FLOAT; + +ALTER TABLE ps_contacts ALTER COLUMN qualify_timeout TYPE FLOAT; + +UPDATE alembic_version SET version_num='423f34ad36e2' WHERE alembic_version.version_num = '136885b81223'; + +-- Running upgrade 423f34ad36e2 -> dbc44d5a908 + +ALTER TABLE ps_systems ADD COLUMN disable_tcp_switch yesno_values; + +ALTER TABLE ps_registrations ADD COLUMN line yesno_values; + +ALTER TABLE ps_registrations ADD COLUMN endpoint VARCHAR(40); + +UPDATE alembic_version SET version_num='dbc44d5a908' WHERE alembic_version.version_num = '423f34ad36e2'; + +-- Running upgrade dbc44d5a908 -> 3bcc0b5bc2c9 + +ALTER TABLE ps_transports ADD COLUMN allow_reload yesno_values; + +UPDATE alembic_version SET version_num='3bcc0b5bc2c9' WHERE alembic_version.version_num = 'dbc44d5a908'; + +-- Running upgrade 3bcc0b5bc2c9 -> 5813202e92be + +ALTER TABLE ps_globals ADD COLUMN contact_expiration_check_interval INTEGER; + +UPDATE alembic_version SET version_num='5813202e92be' WHERE alembic_version.version_num = '3bcc0b5bc2c9'; + +-- Running upgrade 5813202e92be -> 1c688d9a003c + +ALTER TABLE ps_globals ADD COLUMN default_voicemail_extension VARCHAR(40); + +ALTER TABLE ps_aors ADD COLUMN voicemail_extension VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN voicemail_extension VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN mwi_subscribe_replaces_unsolicited INTEGER; + +UPDATE alembic_version SET version_num='1c688d9a003c' WHERE alembic_version.version_num = '5813202e92be'; + +-- Running upgrade 1c688d9a003c -> 8d478ab86e29 + +ALTER TABLE ps_globals ADD COLUMN disable_multi_domain yesno_values; + +UPDATE alembic_version SET version_num='8d478ab86e29' WHERE alembic_version.version_num = '1c688d9a003c'; + +-- Running upgrade 8d478ab86e29 -> 65eb22eb195 + +ALTER TABLE ps_globals ADD COLUMN unidentified_request_count INTEGER; + +ALTER TABLE ps_globals ADD COLUMN unidentified_request_period INTEGER; + +ALTER TABLE ps_globals ADD COLUMN unidentified_request_prune_interval INTEGER; + +ALTER TABLE ps_globals ADD COLUMN default_realm VARCHAR(40); + +UPDATE alembic_version SET version_num='65eb22eb195' WHERE alembic_version.version_num = '8d478ab86e29'; + +-- Running upgrade 65eb22eb195 -> 81b01a191a46 + +ALTER TABLE ps_contacts ADD COLUMN reg_server VARCHAR(20); + +ALTER TABLE ps_contacts ADD CONSTRAINT ps_contacts_uq UNIQUE (id, reg_server); + +UPDATE alembic_version SET version_num='81b01a191a46' WHERE alembic_version.version_num = '65eb22eb195'; + +-- Running upgrade 81b01a191a46 -> 6be31516058d + +ALTER TABLE ps_contacts ADD COLUMN authenticate_qualify yesno_values; + +UPDATE alembic_version SET version_num='6be31516058d' WHERE alembic_version.version_num = '81b01a191a46'; + +-- Running upgrade 6be31516058d -> d7e3c73eb2bf + +ALTER TABLE ps_endpoints ADD COLUMN deny VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN permit VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN acl VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN contact_deny VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN contact_permit VARCHAR(95); + +ALTER TABLE ps_endpoints ADD COLUMN contact_acl VARCHAR(40); + +UPDATE alembic_version SET version_num='d7e3c73eb2bf' WHERE alembic_version.version_num = '6be31516058d'; + +-- Running upgrade d7e3c73eb2bf -> a845e4d8ade8 + +ALTER TABLE ps_contacts ADD COLUMN via_addr VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN via_port INTEGER; + +ALTER TABLE ps_contacts ADD COLUMN call_id VARCHAR(255); + +UPDATE alembic_version SET version_num='a845e4d8ade8' WHERE alembic_version.version_num = 'd7e3c73eb2bf'; + +-- Running upgrade a845e4d8ade8 -> ef7efc2d3964 + +ALTER TABLE ps_contacts ADD COLUMN endpoint VARCHAR(40); + +ALTER TABLE ps_contacts ALTER COLUMN expiration_time TYPE BIGINT USING expiration_time::bigint; + +CREATE INDEX ps_contacts_qualifyfreq_exp ON ps_contacts (qualify_frequency, expiration_time); + +CREATE INDEX ps_aors_qualifyfreq_contact ON ps_aors (qualify_frequency, contact); + +UPDATE alembic_version SET version_num='ef7efc2d3964' WHERE alembic_version.version_num = 'a845e4d8ade8'; + +-- Running upgrade ef7efc2d3964 -> 9deac0ae4717 + +ALTER TABLE ps_endpoints ADD COLUMN subscribe_context VARCHAR(40); + +UPDATE alembic_version SET version_num='9deac0ae4717' WHERE alembic_version.version_num = 'ef7efc2d3964'; + +-- Running upgrade 9deac0ae4717 -> 4a6c67fa9b7a + +ALTER TABLE ps_endpoints ADD COLUMN fax_detect_timeout INTEGER; + +UPDATE alembic_version SET version_num='4a6c67fa9b7a' WHERE alembic_version.version_num = '9deac0ae4717'; + +-- Running upgrade 4a6c67fa9b7a -> c7a44a5a0851 + +ALTER TABLE ps_globals ADD COLUMN mwi_tps_queue_high INTEGER; + +ALTER TABLE ps_globals ADD COLUMN mwi_tps_queue_low INTEGER; + +ALTER TABLE ps_globals ADD COLUMN mwi_disable_initial_unsolicited yesno_values; + +UPDATE alembic_version SET version_num='c7a44a5a0851' WHERE alembic_version.version_num = '4a6c67fa9b7a'; + +-- Running upgrade c7a44a5a0851 -> 3772f8f828da + +ALTER TYPE pjsip_identify_by_values RENAME TO pjsip_identify_by_values_tmp; + +CREATE TYPE pjsip_identify_by_values AS ENUM ('username', 'auth_username'); + +ALTER TABLE ps_endpoints ALTER COLUMN identify_by TYPE pjsip_identify_by_values USING identify_by::text::pjsip_identify_by_values; + +DROP TYPE pjsip_identify_by_values_tmp; + +UPDATE alembic_version SET version_num='3772f8f828da' WHERE alembic_version.version_num = 'c7a44a5a0851'; + +-- Running upgrade 3772f8f828da -> 4e2493ef32e6 + +ALTER TABLE ps_endpoints ADD COLUMN contact_user VARCHAR(80); + +UPDATE alembic_version SET version_num='4e2493ef32e6' WHERE alembic_version.version_num = '3772f8f828da'; + +-- Running upgrade 4e2493ef32e6 -> 7f3e21abe318 + +ALTER TABLE ps_endpoints ADD COLUMN preferred_codec_only yesno_values; + +UPDATE alembic_version SET version_num='7f3e21abe318' WHERE alembic_version.version_num = '4e2493ef32e6'; + +-- Running upgrade 7f3e21abe318 -> a6ef36f1309 + +ALTER TABLE ps_globals ADD COLUMN ignore_uri_user_options yesno_values; + +UPDATE alembic_version SET version_num='a6ef36f1309' WHERE alembic_version.version_num = '7f3e21abe318'; + +-- Running upgrade a6ef36f1309 -> 4468b4a91372 + +ALTER TABLE ps_endpoints ADD COLUMN asymmetric_rtp_codec yesno_values; + +UPDATE alembic_version SET version_num='4468b4a91372' WHERE alembic_version.version_num = 'a6ef36f1309'; + +-- Running upgrade 4468b4a91372 -> 28ab27a7826d + +ALTER TABLE ps_endpoint_id_ips ADD COLUMN srv_lookups yesno_values; + +UPDATE alembic_version SET version_num='28ab27a7826d' WHERE alembic_version.version_num = '4468b4a91372'; + +-- Running upgrade 28ab27a7826d -> 465e70e8c337 + +ALTER TABLE ps_endpoint_id_ips ADD COLUMN match_header VARCHAR(255); + +UPDATE alembic_version SET version_num='465e70e8c337' WHERE alembic_version.version_num = '28ab27a7826d'; + +-- Running upgrade 465e70e8c337 -> 15db7b91a97a + +ALTER TABLE ps_endpoints ADD COLUMN rtcp_mux yesno_values; + +UPDATE alembic_version SET version_num='15db7b91a97a' WHERE alembic_version.version_num = '465e70e8c337'; + +-- Running upgrade 15db7b91a97a -> f638dbe2eb23 + +ALTER TABLE ps_transports ADD COLUMN symmetric_transport yesno_values; + +ALTER TABLE ps_subscription_persistence ADD COLUMN contact_uri VARCHAR(256); + +UPDATE alembic_version SET version_num='f638dbe2eb23' WHERE alembic_version.version_num = '15db7b91a97a'; + +-- Running upgrade f638dbe2eb23 -> 8fce4c573e15 + +ALTER TABLE ps_endpoints ADD COLUMN allow_overlap yesno_values; + +UPDATE alembic_version SET version_num='8fce4c573e15' WHERE alembic_version.version_num = 'f638dbe2eb23'; + +-- Running upgrade 8fce4c573e15 -> 2da192dbbc65 + +CREATE TABLE ps_outbound_publishes ( + id VARCHAR(40) NOT NULL, + expiration INTEGER, + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(256), + server_uri VARCHAR(256), + from_uri VARCHAR(256), + to_uri VARCHAR(256), + event VARCHAR(40), + max_auth_attempts INTEGER, + transport VARCHAR(40), + multi_user yesno_values, + "@body" VARCHAR(40), + "@context" VARCHAR(256), + "@exten" VARCHAR(256), + UNIQUE (id) +); + +CREATE INDEX ps_outbound_publishes_id ON ps_outbound_publishes (id); + +CREATE TABLE ps_inbound_publications ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40), + "event_asterisk-devicestate" VARCHAR(40), + "event_asterisk-mwi" VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_inbound_publications_id ON ps_inbound_publications (id); + +CREATE TABLE ps_asterisk_publications ( + id VARCHAR(40) NOT NULL, + devicestate_publish VARCHAR(40), + mailboxstate_publish VARCHAR(40), + device_state yesno_values, + device_state_filter VARCHAR(256), + mailbox_state yesno_values, + mailbox_state_filter VARCHAR(256), + UNIQUE (id) +); + +CREATE INDEX ps_asterisk_publications_id ON ps_asterisk_publications (id); + +UPDATE alembic_version SET version_num='2da192dbbc65' WHERE alembic_version.version_num = '8fce4c573e15'; + +-- Running upgrade 2da192dbbc65 -> 1d0e332c32af + +CREATE TABLE ps_resource_list ( + id VARCHAR(40) NOT NULL, + list_item VARCHAR(2048), + event VARCHAR(40), + full_state yesno_values, + notification_batch_interval INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_resource_list_id ON ps_resource_list (id); + +UPDATE alembic_version SET version_num='1d0e332c32af' WHERE alembic_version.version_num = '2da192dbbc65'; + +-- Running upgrade 1d0e332c32af -> 86bb1efa278d + +ALTER TABLE ps_endpoints ADD COLUMN refer_blind_progress yesno_values; + +UPDATE alembic_version SET version_num='86bb1efa278d' WHERE alembic_version.version_num = '1d0e332c32af'; + +-- Running upgrade 86bb1efa278d -> d7983954dd96 + +ALTER TABLE ps_endpoints ADD COLUMN notify_early_inuse_ringing yesno_values; + +UPDATE alembic_version SET version_num='d7983954dd96' WHERE alembic_version.version_num = '86bb1efa278d'; + +-- Running upgrade d7983954dd96 -> 39959b9c2566 + +ALTER TABLE ps_endpoints ADD COLUMN max_audio_streams INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN max_video_streams INTEGER; + +UPDATE alembic_version SET version_num='39959b9c2566' WHERE alembic_version.version_num = 'd7983954dd96'; + +-- Running upgrade 39959b9c2566 -> 164abbd708c + +CREATE TYPE pjsip_dtmf_mode_values_v3 AS ENUM ('rfc4733', 'inband', 'info', 'auto', 'auto_info'); + +ALTER TABLE ps_endpoints ALTER COLUMN dtmf_mode TYPE pjsip_dtmf_mode_values_v3 USING dtmf_mode::text::pjsip_dtmf_mode_values_v3; + +DROP TYPE pjsip_dtmf_mode_values_v2; + +UPDATE alembic_version SET version_num='164abbd708c' WHERE alembic_version.version_num = '39959b9c2566'; + +-- Running upgrade 164abbd708c -> 44ccced114ce + +ALTER TABLE ps_endpoints ADD COLUMN webrtc yesno_values; + +UPDATE alembic_version SET version_num='44ccced114ce' WHERE alembic_version.version_num = '164abbd708c'; + +-- Running upgrade 44ccced114ce -> f3d1c5d38b56 + +ALTER TABLE ps_contacts ADD COLUMN prune_on_boot yesno_values; + +UPDATE alembic_version SET version_num='f3d1c5d38b56' WHERE alembic_version.version_num = '44ccced114ce'; + +-- Running upgrade f3d1c5d38b56 -> b83645976fdd + +CREATE TYPE sha_hash_values AS ENUM ('SHA-1', 'SHA-256'); + +ALTER TABLE ps_endpoints ADD COLUMN dtls_fingerprint sha_hash_values; + +UPDATE alembic_version SET version_num='b83645976fdd' WHERE alembic_version.version_num = 'f3d1c5d38b56'; + +-- Running upgrade b83645976fdd -> a1698e8bb9c5 + +ALTER TABLE ps_endpoints ADD COLUMN incoming_mwi_mailbox VARCHAR(40); + +UPDATE alembic_version SET version_num='a1698e8bb9c5' WHERE alembic_version.version_num = 'b83645976fdd'; + +-- Running upgrade a1698e8bb9c5 -> 20abce6d1e3c + +ALTER TYPE pjsip_identify_by_values RENAME TO pjsip_identify_by_values_tmp; + +CREATE TYPE pjsip_identify_by_values AS ENUM ('username', 'auth_username', 'ip'); + +ALTER TABLE ps_endpoints ALTER COLUMN identify_by TYPE pjsip_identify_by_values USING identify_by::text::pjsip_identify_by_values; + +DROP TYPE pjsip_identify_by_values_tmp; + +UPDATE alembic_version SET version_num='20abce6d1e3c' WHERE alembic_version.version_num = 'a1698e8bb9c5'; + +-- Running upgrade 20abce6d1e3c -> de83fac997e2 + +ALTER TABLE ps_endpoints ADD COLUMN bundle yesno_values; + +UPDATE alembic_version SET version_num='de83fac997e2' WHERE alembic_version.version_num = '20abce6d1e3c'; + +-- Running upgrade de83fac997e2 -> 041c0d3d1857 + +ALTER TABLE ps_endpoints ADD COLUMN dtls_auto_generate_cert yesno_values; + +UPDATE alembic_version SET version_num='041c0d3d1857' WHERE alembic_version.version_num = 'de83fac997e2'; + +-- Running upgrade 041c0d3d1857 -> e2f04d309071 + +ALTER TABLE queue_members ADD COLUMN wrapuptime INTEGER; + +UPDATE alembic_version SET version_num='e2f04d309071' WHERE alembic_version.version_num = '041c0d3d1857'; + +-- Running upgrade e2f04d309071 -> 52798ad97bdf + +ALTER TABLE ps_endpoints ALTER COLUMN identify_by TYPE varchar(80) USING identify_by::text::pjsip_identify_by_values; + +DROP TYPE pjsip_identify_by_values; + +UPDATE alembic_version SET version_num='52798ad97bdf' WHERE alembic_version.version_num = 'e2f04d309071'; + +-- Running upgrade 52798ad97bdf -> d3e4284f8707 + +ALTER TABLE ps_subscription_persistence ADD COLUMN prune_on_boot yesno_values; + +UPDATE alembic_version SET version_num='d3e4284f8707' WHERE alembic_version.version_num = '52798ad97bdf'; + +-- Running upgrade d3e4284f8707 -> 0be05c3a8225 + +ALTER TABLE ps_systems ADD COLUMN follow_early_media_fork yesno_values; + +ALTER TABLE ps_systems ADD COLUMN accept_multiple_sdp_answers yesno_values; + +ALTER TABLE ps_endpoints ADD COLUMN follow_early_media_fork yesno_values; + +ALTER TABLE ps_endpoints ADD COLUMN accept_multiple_sdp_answers yesno_values; + +UPDATE alembic_version SET version_num='0be05c3a8225' WHERE alembic_version.version_num = 'd3e4284f8707'; + +-- Running upgrade 0be05c3a8225 -> 19b00bc19b7b + +ALTER TABLE ps_endpoints ADD COLUMN suppress_q850_reason_header yesno_values; + +UPDATE alembic_version SET version_num='19b00bc19b7b' WHERE alembic_version.version_num = '0be05c3a8225'; + +-- Running upgrade 19b00bc19b7b -> 1d3ed26d9978 + +ALTER TABLE ps_contacts ALTER COLUMN uri TYPE VARCHAR(511); + +UPDATE alembic_version SET version_num='1d3ed26d9978' WHERE alembic_version.version_num = '19b00bc19b7b'; + +-- Running upgrade 1d3ed26d9978 -> fe6592859b85 + +CREATE TYPE ast_bool_values AS ENUM ('0', '1', 'off', 'on', 'false', 'true', 'no', 'yes'); + +ALTER TABLE ps_endpoints ALTER COLUMN mwi_subscribe_replaces_unsolicited TYPE VARCHAR(5); + +ALTER TABLE ps_endpoints ALTER COLUMN mwi_subscribe_replaces_unsolicited TYPE ast_bool_values USING mwi_subscribe_replaces_unsolicited::ast_bool_values; + +UPDATE alembic_version SET version_num='fe6592859b85' WHERE alembic_version.version_num = '1d3ed26d9978'; + +-- Running upgrade fe6592859b85 -> 7f85dd44c775 + +ALTER TABLE ps_endpoints ALTER COLUMN suppress_q850_reason_header TYPE yesno_values; + +ALTER TABLE ps_endpoints RENAME suppress_q850_reason_header TO suppress_q850_reason_headers; + +UPDATE alembic_version SET version_num='7f85dd44c775' WHERE alembic_version.version_num = 'fe6592859b85'; + +-- Running upgrade 7f85dd44c775 -> 465f47f880be + +CREATE TYPE pjsip_transport_protocol_values_v2 AS ENUM ('udp', 'tcp', 'tls', 'ws', 'wss', 'flow'); + +ALTER TABLE ps_transports ALTER COLUMN protocol TYPE pjsip_transport_protocol_values_v2 USING protocol::text::pjsip_transport_protocol_values_v2; + +DROP TYPE pjsip_transport_protocol_values; + +CREATE TYPE pjsip_auth_type_values_v2 AS ENUM ('md5', 'userpass', 'google_oauth'); + +ALTER TABLE ps_auths ALTER COLUMN auth_type TYPE pjsip_auth_type_values_v2 USING auth_type::text::pjsip_auth_type_values_v2; + +DROP TYPE pjsip_auth_type_values; + +ALTER TABLE ps_registrations ADD COLUMN support_outbound ast_bool_values; + +ALTER TABLE ps_registrations ADD COLUMN contact_header_params VARCHAR(255); + +ALTER TABLE ps_auths ADD COLUMN refresh_token VARCHAR(255); + +ALTER TABLE ps_auths ADD COLUMN oauth_clientid VARCHAR(255); + +ALTER TABLE ps_auths ADD COLUMN oauth_secret VARCHAR(255); + +UPDATE alembic_version SET version_num='465f47f880be' WHERE alembic_version.version_num = '7f85dd44c775'; + +-- Running upgrade 465f47f880be -> 2bb1a85135ad + +ALTER TABLE ps_globals ADD COLUMN use_callerid_contact ast_bool_values; + +UPDATE alembic_version SET version_num='2bb1a85135ad' WHERE alembic_version.version_num = '465f47f880be'; + +-- Running upgrade 2bb1a85135ad -> 1ac563b350a8 + +ALTER TABLE ps_endpoints ADD COLUMN trust_connected_line ast_bool_values; + +ALTER TABLE ps_endpoints ADD COLUMN send_connected_line ast_bool_values; + +UPDATE alembic_version SET version_num='1ac563b350a8' WHERE alembic_version.version_num = '2bb1a85135ad'; + +-- Running upgrade 1ac563b350a8 -> 0838f8db6a61 + +ALTER TABLE ps_globals ADD COLUMN send_contact_status_on_update_registration ast_bool_values; + +UPDATE alembic_version SET version_num='0838f8db6a61' WHERE alembic_version.version_num = '1ac563b350a8'; + +-- Running upgrade 0838f8db6a61 -> f3c0b8695b66 + +CREATE TYPE pjsip_taskprocessor_overload_trigger_values AS ENUM ('none', 'global', 'pjsip_only'); + +ALTER TABLE ps_globals ADD COLUMN taskprocessor_overload_trigger pjsip_taskprocessor_overload_trigger_values; + +UPDATE alembic_version SET version_num='f3c0b8695b66' WHERE alembic_version.version_num = '0838f8db6a61'; + +-- Running upgrade f3c0b8695b66 -> 80473bad3c16 + +ALTER TABLE ps_endpoints ADD COLUMN ignore_183_without_sdp ast_bool_values; + +UPDATE alembic_version SET version_num='80473bad3c16' WHERE alembic_version.version_num = 'f3c0b8695b66'; + +-- Running upgrade 80473bad3c16 -> 3a094a18e75b + +ALTER TABLE ps_globals ADD COLUMN norefersub ast_bool_values; + +UPDATE alembic_version SET version_num='3a094a18e75b' WHERE alembic_version.version_num = '80473bad3c16'; + +-- Running upgrade 3a094a18e75b -> fbb7766f17bc + +CREATE TABLE musiconhold_entry ( + name VARCHAR(80) NOT NULL, + position INTEGER NOT NULL, + entry VARCHAR(1024) NOT NULL, + PRIMARY KEY (name, position) +); + +ALTER TABLE musiconhold_entry ADD CONSTRAINT fk_musiconhold_entry_name_musiconhold FOREIGN KEY(name) REFERENCES musiconhold (name); + +ALTER TYPE moh_mode_values RENAME TO moh_mode_values_tmp; + +CREATE TYPE moh_mode_values AS ENUM ('custom', 'files', 'mp3nb', 'quietmp3nb', 'quietmp3', 'playlist'); + +ALTER TABLE musiconhold ALTER COLUMN mode TYPE moh_mode_values USING mode::text::moh_mode_values; + +DROP TYPE moh_mode_values_tmp; + +UPDATE alembic_version SET version_num='fbb7766f17bc' WHERE alembic_version.version_num = '3a094a18e75b'; + +-- Running upgrade fbb7766f17bc -> 79290b511e4b + +ALTER TABLE ps_systems ADD COLUMN disable_rport ast_bool_values; + +UPDATE alembic_version SET version_num='79290b511e4b' WHERE alembic_version.version_num = 'fbb7766f17bc'; + +-- Running upgrade 79290b511e4b -> b80485ff4dd0 + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_incoming_offer VARCHAR(128); + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_outgoing_offer VARCHAR(128); + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_incoming_answer VARCHAR(128); + +ALTER TABLE ps_endpoints ADD COLUMN codec_prefs_outgoing_answer VARCHAR(128); + +UPDATE alembic_version SET version_num='b80485ff4dd0' WHERE alembic_version.version_num = '79290b511e4b'; + +-- Running upgrade b80485ff4dd0 -> 61797b9fced6 + +ALTER TABLE ps_endpoints ADD COLUMN stir_shaken ast_bool_values; + +UPDATE alembic_version SET version_num='61797b9fced6' WHERE alembic_version.version_num = 'b80485ff4dd0'; + +-- Running upgrade 61797b9fced6 -> 1ae0609b6646 + +ALTER TABLE ps_contacts ALTER COLUMN reg_server TYPE VARCHAR(255); + +UPDATE alembic_version SET version_num='1ae0609b6646' WHERE alembic_version.version_num = '61797b9fced6'; + +-- Running upgrade 1ae0609b6646 -> e658c26033ca + +ALTER TABLE ps_endpoints ADD COLUMN send_history_info ast_bool_values; + +UPDATE alembic_version SET version_num='e658c26033ca' WHERE alembic_version.version_num = '1ae0609b6646'; + +-- Running upgrade e658c26033ca -> 8915fcc5766f + +ALTER TABLE queue_members ADD COLUMN ringinuse ast_bool_values; + +UPDATE alembic_version SET version_num='8915fcc5766f' WHERE alembic_version.version_num = 'e658c26033ca'; + +-- Running upgrade 8915fcc5766f -> c20d6e3992f4 + +ALTER TABLE ps_endpoints ADD COLUMN allow_unauthenticated_options ast_bool_values; + +UPDATE alembic_version SET version_num='c20d6e3992f4' WHERE alembic_version.version_num = '8915fcc5766f'; + +-- Running upgrade c20d6e3992f4 -> f56d79a9f337 + +ALTER TABLE ps_aors ADD COLUMN remove_unavailable ast_bool_values; + +UPDATE alembic_version SET version_num='f56d79a9f337' WHERE alembic_version.version_num = 'c20d6e3992f4'; + +-- Running upgrade f56d79a9f337 -> a06d8f8462d9 + +ALTER TABLE ps_endpoints ADD COLUMN t38_bind_udptl_to_media_address ast_bool_values; + +UPDATE alembic_version SET version_num='a06d8f8462d9' WHERE alembic_version.version_num = 'f56d79a9f337'; + +-- Running upgrade a06d8f8462d9 -> 8f72185e437f + +ALTER TABLE ps_resource_list ADD COLUMN resource_display_name ast_bool_values; + +UPDATE alembic_version SET version_num='8f72185e437f' WHERE alembic_version.version_num = 'a06d8f8462d9'; + +-- Running upgrade 8f72185e437f -> 0bee61aa9425 + +ALTER TABLE ps_globals ADD COLUMN allow_sending_180_after_183 ast_bool_values; + +UPDATE alembic_version SET version_num='0bee61aa9425' WHERE alembic_version.version_num = '8f72185e437f'; + +-- Running upgrade 0bee61aa9425 -> 18e0805d367f + +ALTER TABLE ps_registrations ADD COLUMN max_random_initial_delay INTEGER; + +UPDATE alembic_version SET version_num='18e0805d367f' WHERE alembic_version.version_num = '0bee61aa9425'; + +-- Running upgrade 18e0805d367f -> 58e440314c2a + +ALTER TABLE ps_transports ADD COLUMN allow_wildcard_certs yesno_values; + +UPDATE alembic_version SET version_num='58e440314c2a' WHERE alembic_version.version_num = '18e0805d367f'; + +-- Running upgrade 58e440314c2a -> 7197536bb68d + +ALTER TABLE ps_endpoints ADD COLUMN geoloc_incoming_call_profile VARCHAR(80); + +ALTER TABLE ps_endpoints ADD COLUMN geoloc_outgoing_call_profile VARCHAR(80); + +UPDATE alembic_version SET version_num='7197536bb68d' WHERE alembic_version.version_num = '58e440314c2a'; + +-- Running upgrade 7197536bb68d -> 9f3692b1654b + +CREATE TYPE pjsip_incoming_call_offer_pref_values AS ENUM ('local', 'local_first', 'remote', 'remote_first'); + +CREATE TYPE pjsip_outgoing_call_offer_pref_values AS ENUM ('local', 'local_merge', 'local_first', 'remote', 'remote_merge', 'remote_first'); + +ALTER TABLE ps_endpoints ADD COLUMN incoming_call_offer_pref pjsip_incoming_call_offer_pref_values; + +ALTER TABLE ps_endpoints ADD COLUMN outgoing_call_offer_pref pjsip_outgoing_call_offer_pref_values; + +ALTER TABLE ps_endpoints ADD COLUMN stir_shaken_profile VARCHAR(80); + +UPDATE alembic_version SET version_num='9f3692b1654b' WHERE alembic_version.version_num = '7197536bb68d'; + +-- Running upgrade 9f3692b1654b -> 539f68bede2c + +CREATE TYPE pjsip_100rel_values_v2 AS ENUM ('no', 'required', 'peer_supported', 'yes'); + +ALTER TABLE ps_endpoints ALTER COLUMN 100rel TYPE pjsip_100rel_values_v2 USING 100rel::text::pjsip_100rel_values_v2; + +DROP TYPE pjsip_100rel_values; + +UPDATE alembic_version SET version_num='539f68bede2c' WHERE alembic_version.version_num = '9f3692b1654b'; + +-- Running upgrade 539f68bede2c -> 417c0247fd7e + +CREATE TYPE security_negotiation_values AS ENUM ('no', 'mediasec'); + +ALTER TABLE ps_endpoints ADD COLUMN security_negotiation security_negotiation_values; + +ALTER TABLE ps_endpoints ADD COLUMN security_mechanisms VARCHAR(512); + +ALTER TABLE ps_registrations ADD COLUMN security_negotiation security_negotiation_values; + +ALTER TABLE ps_registrations ADD COLUMN security_mechanisms VARCHAR(512); + +UPDATE alembic_version SET version_num='417c0247fd7e' WHERE alembic_version.version_num = '539f68bede2c'; + +-- Running upgrade 417c0247fd7e -> ccf795ee535f + +ALTER TABLE ps_globals ADD COLUMN all_codecs_on_empty_reinvite ast_bool_values; + +UPDATE alembic_version SET version_num='ccf795ee535f' WHERE alembic_version.version_num = '417c0247fd7e'; + +-- Running upgrade ccf795ee535f -> 5a2247c957d2 + +ALTER TABLE ps_endpoints ADD COLUMN send_aoc ast_bool_values; + +UPDATE alembic_version SET version_num='5a2247c957d2' WHERE alembic_version.version_num = 'ccf795ee535f'; + +-- Running upgrade 5a2247c957d2 -> f261363a857f + +ALTER TABLE ps_endpoints ADD COLUMN overlap_context VARCHAR(80); + +UPDATE alembic_version SET version_num='f261363a857f' WHERE alembic_version.version_num = '5a2247c957d2'; + +COMMIT; + diff --git a/contrib/realtime/postgresql/postgresql_voicemail.sql b/contrib/realtime/postgresql/postgresql_voicemail.sql new file mode 100644 index 0000000000..db7d09a930 --- /dev/null +++ b/contrib/realtime/postgresql/postgresql_voicemail.sql @@ -0,0 +1,39 @@ +BEGIN; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL, + CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num) +); + +-- Running upgrade -> a2e9769475e + +CREATE TABLE voicemail_messages ( + dir VARCHAR(255) NOT NULL, + msgnum INTEGER NOT NULL, + context VARCHAR(80), + macrocontext VARCHAR(80), + callerid VARCHAR(80), + origtime INTEGER, + duration INTEGER, + recording BYTEA, + flag VARCHAR(30), + category VARCHAR(30), + mailboxuser VARCHAR(30), + mailboxcontext VARCHAR(30), + msg_id VARCHAR(40) +); + +ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum); + +CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir); + +INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e'); + +-- Running upgrade a2e9769475e -> 39428242f7f5 + +ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA; + +UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e'; + +COMMIT; +