There are two main parts of the change associated with this
commit. These are driven by the change in call order of
pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
when an in-dialog SUBSCRIBE is received.
First, the previous behavior was for pjproject to call
pubsub_on_rx_refresh before calling pubsub_on_evsub_state
when an in-dialog SUBSCRIBE was received that changes the
subscription state.
If that change was a termination due to a re-SUBSCRIBE with
an expires of 0, we used to use the call to pubsub_on_rx_refresh
to set the substate of the evsub to TERMINATE_PENDING before
pjproject could call pubsub_on_evsub_state.
This substate let pubsub_on_evsub_state know that the
subscription TERMINATED event could be ignored as there was
still a subsequent NOTIFY that needed to be generated and
another call to pubsub_on_evsub_state to come with it.
That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
which would see the TERMINATE_PENDING state and transition it
to TERMINATE_IN_PROGRESS before triggering another call to
pubsub_on_evsub_state (which now would clean up the evsub.)
The new pjproject behavior is to call pubsub_on_evsub_state
before pubsub_on_rx_refresh. This means we no longer can set
the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
that it can ignore the first TERMINATED event.
To handle this, we now look directly at the event type,
method type and the expires value to determine whether we
want to ignore the event or use it to trigger the evsub
cleanup.
Second, pjproject now expects the NOTIFY to actually be sent
during pubsub_on_rx_refresh and avoids the protocol violation
inherent in sending a NOTIFY before the SUBSCRIBE is
acknowledged by caching the sent NOTIFY then sending it
after responding to the SUBSCRIBE.
This requires we send the NOTIFY using the non-serialized
pubsub_on_refresh_timeout directly and let pjproject handle
the protocol violation.
ASTERISK-30469
Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7
* 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
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
`getcwd(…)` is decorated with the `warn_unused_result` attribute and
therefore needs its return value checked.
Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93
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
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
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
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
Adds 's' option to skip calling the extension and instead set the
extension as DIRECTORY_EXTEN channel variable.
ASTERISK-30405
Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299
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
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
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
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
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
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
Variable references within global variable assignments are now
expanded rather than being included literally.
ASTERISK-30406 #close
Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d
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
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
Added NULL pointer check and channel lock to prevent resource release
while the chanspy is processing.
ASTERISK-29604
Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24
In Asterisk 11, if a channel was redirected away during Playback(),
the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
(specifically commit 7d9871b394) 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
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
Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).
ASTERISK-30319 #close
Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645
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
-----------------
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
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
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
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
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
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
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=<optimized out>, 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=<optimized out>, data=<optimized out>, object=<optimized out>)
at res_http_media_cache.c:613
#6 0x0000000000487638 in bucket_file_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
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=<optimized out>, 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=<optimized out>, cmd=<optimized out>, a=0xffffca9976b8)
at media_cache.c:640
ASTERISK-30375 #close
Change-Id: I6a9433688cb5d3d4be8758b7642d923bdde6c273
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
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
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
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
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
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
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
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
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
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
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
Asterisk Development Team
Mike Bradeen
Sean Bright
George Joseph
Naveen Albert
Nick French
cmaj
sungtae kim
George Joseph
Igor Goncharovsky
Alexei Gradinari
Boris P. Korzun
Holger Hans Peter Freyther
Ben Ford
Peter Fern
Alexandre Fournier