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
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
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
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 <mjordan@digium.com>
Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333
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
This fixes a small typo in the from_domain documentation on the endpoint documentation
ASTERISK-30328 #close
Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21
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
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
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
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
The commit that rearchitected media formats,
a2c912e997 (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
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
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
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
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
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
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
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
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
When decoding q.931 encoded calling/called number
now checking for length being less than minimum required.
ASTERISK-30103
Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7
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
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
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
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
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 <jaco@uls.co.za>
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
Removes the function mkstemp_file and uses
ast_file_mkftemp from file.h instead.
ASTERISK-30295 #close
Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609
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
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
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
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
(cherry picked from commit 12d18b0a40)
Fixed the specification of "outputdir" when calling ast_coredumper
so the txt files are saved in the correct place.
ASTERISK-30282
Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6
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
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
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
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
Fix compilation errors caused by using size_t
instead of uintmax_t and non-portable format
specifiers.
ASTERISK-30273 #close
Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee
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
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 <sean@seanbright.com>
Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f
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
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
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
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
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
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
Adds support for detecting audible ringback tone
to the TONE_DETECT function using the p option.
ASTERISK-30254 #close
Change-Id: Ie2329ff245248768367d26749c285fbe823f6414
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
"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
Also added a note to the geolocation.conf.sample file
and added a README to the res/res_geolocation/wiki
directory.
Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc
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
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
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
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
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
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
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
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
Adds the EXPORT function, which allows write
access to variables and functions on other
channels.
ASTERISK-29432 #close
Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf
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
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 <jaco@uls.co.za>
Adds TRIM, LTRIM, and RTRIM, which can be used
for trimming leading and trailing whitespace
from strings.
ASTERISK-30222 #close
Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554
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
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
This change adds an option, answeredonly, that will prevent music on
hold on channels that are not answered.
ASTERISK-30135
Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08
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
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
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
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
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
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
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
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
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:
"<prefer_config>"
"<discard_config>"
"<prefer_incoming>"
"<discard_incoming>"
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. "<discard_config>" is actually the
best one to use in this situation.
ASTERISK-30182
Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a
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
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
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
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
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
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
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
The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set()
for us, so no need to declare it separately.
Change-Id: I4813a884ada475ddc62bca480bceb4a53b3ec59a
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
Fixes a few coding guideline violations:
* Use of C99 comments
* Opening brace on same line as function prototype
ASTERISK-30163 #close
Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7
* 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
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
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
The MeetmeList and MeetmeListRooms AMI
responses are currently completely undocumented.
This adds documentation for these responses.
ASTERISK-30018 #close
Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8
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
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
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
Documents the ConfbridgeListRooms AMI response,
which is currently not documented.
ASTERISK-30020 #close
Change-Id: Id6fff7a936244bae7b52686301eb740c1169cdea
Adds missing documentation for the field parameter
for the SRVRESULT function.
ASTERISK-30151
Reported by: Chris Young
Change-Id: I4385a2e0892a07e30dea1a8a0588e2c1bea2b1f1
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
* 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
These new functions allow retrieving information from headers on 200 OK
INVITE response.
ASTERISK-29999
Change-Id: I264a610a9333359297a0825feb29a1bb4f4ad144
Switched res_pjsip_outbound_registration.so dep to optional. Added
module loaded check before using it.
ASTERISK-30101 #close
Change-Id: Ia34f1684d984e821fbdd4de8911f930337703666
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
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
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
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
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
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
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
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
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
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
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
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
A recent change, ASTERISK_30027, now exposes a channel driver's unique
id to ARI channel resources so bump the ARI version.
Change-Id: Ic9da84b7b0eca6f3ff5b6a738b2be8362237951e
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
If there is scheduled notification, we must delete it
to avoid using destroyed subscriptions.
ASTERISK-29906
Change-Id: I1c644e5e15a8fe43eed8e4f9112f113cbf87a40f
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
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
An m option to Park and ParkAndAnnounce now allows
specifying a music on hold class override.
ASTERISK-30087
Change-Id: I03de8d97b100e451b2611b5a621d48750f5d6a9e
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
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
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
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
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
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
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
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
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
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
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
Adjusts the pjsip show registration(s) commands to show
the amount of seconds remaining until a registration
expires.
ASTERISK-29845 #close
Change-Id: Ic4fea15a1a1056c424416def49d1ca8e776c0483
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
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
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
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
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
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
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
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
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
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
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
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
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
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
This adds the EVAL_EXTEN function, which may be used to retrieve
the variable-substituted data at any extension.
ASTERISK-29486
Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432
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
If Asterisk receives a SIP REFER with Session-Timers UAC
maintain Session-Timers when sending UPDATE"
ASTERISK-29843
Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c
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
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
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
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
added new global config option "allow_sending_180_after_183"
that if enabled will preserve 180 after a 183
ASTERISK-29842
Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18
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
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
Patch provided inline by Yury Kirsanov on the linked issue and
approved by Josh Colp.
ASTERISK-29253 #close
Change-Id: I5b9ccc67ebf06e875ed061d9e7fc21f47b0a4e1f
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 <id> 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
(cherry picked from commit 45a1977de4)
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
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
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
Chrome has added more attributes, causing the limit to be
exceeded. This raises it up some more.
ASTERISK-30015
Change-Id: I964957c005c4e6f7871b15ea1ccd9b4659c7ef32
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
(cherry picked from commit 37ece75677)
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
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
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 <philipp@redfish-solutions.com>
Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56
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 <list> 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 <resource>
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
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 <philipp@redfish-solutions.com>
Change-Id: Iaa49fbbab029c64ae3d95e4b18270e0442cce170
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
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
Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.
ASTERISK-25716
Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0
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
Omit "unsupported column type 'text'" warning in logs while
using text-type column in the PgSQL backend.
ASTERISK-29924 #close
Change-Id: I48061a7d469426859670db07f1ed8af1eb814712
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
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
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
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
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
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":
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
<docs>
<configInfo>
...
</configInfo>
</docs>
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
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/<mainline_branch>/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
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
Adds the lastcontext and lastexten channel fields to allow users
to access previous dialplan execution locations.
ASTERISK-29840 #close
Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
This code was needlessly complex and would fail to properly delimit
the response message if LOW_MEMORY was defined.
Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e
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
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
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
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
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
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
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
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
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
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
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
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
Enable the Linux rdtsc implementation on NetBSD as well. The assembly
works correctly there.
ASTERISK-29851
Change-Id: I460ad9b4d971913420ecb84186f5ba5ab03f6f37
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
Implement the ast_get_tid() function for NetBSD system. NetBSD supports
getting the TID via _lwp_self().
ASTERISK-29850
Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c
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
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 <stdlib.h>.
ASTERISK-29818
Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7
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
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
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
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
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
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
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
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
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
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
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
Fixes 12pm noon incorrectly returning 0/a.m.
Also fixes a misspelling typo in the config.
ASTERISK-29695 #close
Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406
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
Adds the macro DTMF_MATRIX_SIZE to replace
the magic number 4 sprinkled throughout
dsp.c.
ASTERISK-29815 #close
Change-Id: Ie3bddb92c6b16204ece0f758009e9490eb33b9ba
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
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
Adds missing documentation for some channel,
bridge, and queue events.
ASTERISK-24427
ASTERISK-29515
Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING
ASTERISK-28053
Reported by: roadkill
Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89
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
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
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
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
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
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 <jaco@uls.co.za>
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
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
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
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
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
Correct typos of the following word families:
multiplication
potentially
iteration
interaction
virtual
synthesis
convolve
initializes
overlap
ASTERISK-29714
Change-Id: Ia40f1aca8f2996ab407c6ed9d24cb10a67c6684b
(cherry picked from commit 2a8fb4695e)
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
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
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
Correct typos of the following word families:
command-line
immediately
extensions
momentarily
mustn't
numbered
bytes
caching
ASTERISK-29714
Change-Id: I8b2b125c5d4d2f9e87a58515c97468ad47ca44f8
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
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
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
Correct typos of the following word families:
dependency
unless
random
dependencies
delimited
randomly
modules
ASTERISK-29714
Change-Id: I3920603a8dc7c0a1852d2f885e06b1144692d40e
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
Correct typos of the following word families:
voiced
denumerator
codeword
upsampling
constructed
residual
subroutine
conditional
quantizing
courtesy
number
ASTERISK-29714
Change-Id: I471fb8086a5277d8f05047fedee22cfa97a4252d
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
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
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
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
* 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
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
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
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
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
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
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
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
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=<dir> or --with-crypto=<dir> 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
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
Some ast_stun_request users do not provide a destination address when
sending to a connection-mode socket.
ASTERISK-29691
Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc
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
Discovered while looking at ASTERISK~29684. Usage was removed in change
I3c77c7b00b2ffa2e935632097fa057b9fdf480c0.
Change-Id: Iaf2f7a16ea5a7eee6375319347e4b40b8e7b10e3
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
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
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
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
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
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
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
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
Adds a function to check for the existence of a channel by
name or by UNIQUEID.
ASTERISK-29656 #close
Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb
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
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
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
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
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 <thomas-forum@orgis.org> for giving the key!
ASTERISK-29635 #close
Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816
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
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
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
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
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
We can't rely on RAII_VAR(...) to properly clean up data that is
allocated within a loop.
ASTERISK-27176 #close
Change-Id: Ib575616101230c4f603519114ec62ebf3936882c
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
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
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
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
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
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
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
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
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
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
Adds an information element for ANI2 so that
Originating Line Information can be transmitted
over IAX2 channels.
ASTERISK-29605 #close
Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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.
cdr_mysql was deprecated in 1.8, to be removed in 19.
app_mysql was deprecated in 1.8, to be removed in 19.
app_ices was deprecated in 16, to be removed in 19.
app_macro was deprecated in 16, to be removed in 21.
app_fax was deprecated in 16, to be removed in 19.
app_url was deprecated in 16, to be removed in 19.
app_image was deprecated in 16, to be removed in 19.
app_nbscat was deprecated in 16, to be removed in 19.
app_dahdiras was deprecated in 16, to be removed in 19.
cdr_syslog was deprecated in 16, to be removed in 19.
chan_oss was deprecated in 16, to be removed in 19.
chan_phone was deprecated in 16, to be removed in 19.
chan_sip was deprecated in 17, to be removed in 21.
chan_nbs was deprecated in 16, to be removed in 19.
chan_misdn was deprecated in 16, to be removed in 19.
chan_vpb was deprecated in 16, to be removed in 19.
res_config_sqlite was deprecated in 16, to be removed in 19.
res_monitor was deprecated in 16, to be removed in 21.
conf2ael was deprecated in 16, to be removed in 19.
muted was deprecated in 16, to be removed in 19.
ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29554
ASTERISK-29555
ASTERISK-29557
ASTERISK-29558
ASTERISK-29559
ASTERISK-29560
ASTERISK-29561
ASTERISK-29562
ASTERISK-29563
ASTERISK-29564
ASTERISK-29565
ASTERISK-29566
ASTERISK-29567
ASTERISK-29568
ASTERISK-29569
ASTERISK-29570
ASTERISK-29571
ASTERISK-29572
ASTERISK-29573
ASTERISK-29574
Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131
Adds function to selectively drop specified frames
in the TX or RX direction on a channel, including
control frames.
ASTERISK-29478
Change-Id: I8147c9d55d74e2e48861edba6b22f930920541ec
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
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
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
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
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
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
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
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
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
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
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
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
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
Use the URI parsing functions 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
This appears to just have been a copy/paste error from 6258bbe7. Fix
suggested by Ross Beer in ASTERISK~29166.
Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37
Add check that data parameter specified when audiosocket used for externalMedia.
ASTERISK-29514 #close
Change-Id: Ie562f03c5d6c3835a3631f376b3d43e75b8f9617
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
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
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
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
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
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
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
When the MessageSend destination is in the form
PJSIP/<number>@<endpoint> and the endpoint's contact
URI already has a user component, that user component
will now be replaced with <number> when creating the
request URI.
ASTERISK_29404
Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5
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
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
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 <jaco@uls.co.za>
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 <jaco@uls.co.za>
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 <jaco@uls.co.za>
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 <jaco@uls.co.za>
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
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
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
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
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
* 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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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 <info> 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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function.
ASTERISK-29293 #close
Change-Id: If5a6d4b1072ea2e6e89059b21139d554a74b34f5
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
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 <jaco@uls.co.za>
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
This partially reverts commit 3d1bf3c537,
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 <jaco@uls.co.za>
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
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
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 <jaco@uls.co.za>
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
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
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
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
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
Provided a support of a MIME-type for wav16. Added new MIME-type
for classic wav.
ASTERISK-29275 #close
Change-Id: I749bda287ba1ab20c1e0af5e4c0153817d47873b
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
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
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
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
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
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
Provided a support of variuos URL-schemes for res_musiconhold,
registered by ast_bucket_scheme_register().
ASTERISK-29262 #close
Change-Id: If0ea8697587353dce358a70035d82649fd4632b6
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
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
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 <jaco@uls.co.za>
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
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
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
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
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
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
This reverts commit 860e40dd80.
Reason for revert: Too many issues reported. Need to research and correct.
ASTERISK-29230
ASTERISK-29231
Reported by: Michael Maier
Change-Id: I9011e2eecda4e91e1cfeeda6d1a7f1a0453eab41
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 <jaco@uls.co.za>
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 <jaco@uls.co.za>
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
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
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
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
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
AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an
unsigned long on all machines.
Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338
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
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
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
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
This adds support for both Digium and Sangoma user agent strings
for the Sangoma specific body supplement.
Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482
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
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, but leaves
it at /tmp by default, to ensure backwards compatibility.
A future commit that only targets master could change the default
location to something more sensible such as /var/tmp/asterisk. At
that point, the cachedir could be created and cleaned up during
uninstall by the Makefile script.
ASTERISK-29143
Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d
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
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
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
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
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
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
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
* 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
* 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
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
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
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
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
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
* Wildcards in #includes are now properly expanded
* Implement operators for Section class to allow sorting
ASTERISK-29142 #close
Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df
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
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 else<TAB>if with else<SP>if while in the vicinity.)
Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a
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
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
Commit 44bb0858cb ("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
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
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
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
(cherry picked from commit 56028426de)
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
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
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
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
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
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
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
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
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
Sometimes not play MOH on bridge.
ASTERISK-29081
Reported-by: Michal Hajek <michal.hajek@daktela.com>
Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232
ast_trace() was always emitting messages when it's level was set to -1
because it was ignoring scope_level.
Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506
The module description needs to be on the same line as the
AST_MODULE_INFO or it is not parsed correctly.
Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21
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
Implemention of History-Info capable of interworking with Diversion
Header following RFC7544
ASTERISK-29027 #close
Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6
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
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
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
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
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
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
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
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
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 <coredump>.output instead
of just <coredump> so if you expand the tarball it won't
overwrite the coredump.
Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea
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
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
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
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
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
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
Properly bump reference on format object to avoid memory corruption on double free
ASTERISK-29040 #close
Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3
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
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
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
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
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
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
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
* 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
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
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
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
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
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
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
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
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
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
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 deaa3742dc)
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
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
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
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 <bridge id>'
cli.
ASTERISK-28987
Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295
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
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
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
By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2019 Digium, Inc. and other copyright holders.
Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
```
## SECURITY
@ -47,7 +47,7 @@ and the BSD variants.
ANY special hardware, not even a sound card) to install and run Asterisk.
Supported telephony hardware includes:
* All Analog and Digital Interface cards from [Digium]
* All Analog and Digital Interface cards from [Sangoma]
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
* any full duplex sound card supported by ALSA, OSS, or PortAudio
* any ISDN card supported by mISDN on Linux
@ -91,7 +91,10 @@ guides in the [configs] directory.
2. Run `./configure`
Execute the configure script to guess values for system-dependent
variables used during compilation.
variables used during compilation. If the script indicates that some required
components are missing, you can run `./contrib/scripts/install_prereq install`
to install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need
to rerun `./configure`.
3. Run `make menuselect` _\[optional]_
@ -251,7 +254,7 @@ Welcome to the growing worldwide community of Asterisk users!
---
Asterisk is a trademark of Digium, Inc.
Asterisk is a trademark of Sangoma Technologies Corporation
[home page]: https://www.asterisk.org
[support]: https://www.asterisk.org/support
@ -261,7 +264,7 @@ Asterisk is a trademark of Digium, Inc.
/* don't need to back up a priority, because we don't actually need to execute Else, just jump to the priority after. Directly executing Else will exit the conditional. */
/* If is false, and Else exists, so jump to Else */
ast_verb(3,"Taking absolute false branch, jumping to priority %d\n",pri);
ast_channel_priority_set(chan,pri);
}else{
pri=endifpri;
if(pri>0){
ast_verb(3,"Exiting conditional, jumping to priority %d\n",pri);
ast_channel_priority_set(chan,pri);
}elseif(end==4){/* Condition added because of end > 0 instead of end == 4 */
ast_log(LOG_WARNING,"Couldn't find matching EndIf? (If at %s@%s priority %d)\n",ast_channel_context(chan),ast_channel_exten(chan),ast_channel_priority(chan));
if((maxdigits&&digits_read>=maxdigits)||digits_read>=(buflen-1)){/* we don't have room to store any more digits (very unlikely to happen for a legitimate reason) */
/* This result will probably not be usable, so status should not be START */
@ -5530,7 +5416,7 @@ static int sendmail(char *srcemail,
ast_debug(3,"Attaching file '%s', format '%s', uservm is '%d', global is %u\n",attach,format,attach_user_voicemail,ast_test_flag((&globalflags),VM_ATTACH));
/* Make a temporary file instead of piping directly to sendmail, in case the mail
res=ast_say_date_with_format(chan,t,AST_DIGIT_ANY,ast_channel_language(chan),"'vm-received' Ad 'digits/pt-de' B 'digits/pt-de' Y 'digits/pt-as' HM ",NULL);
}elseif(!strncasecmp(ast_channel_language(chan),"se",2)){/* SWEDISH syntax */
res=ast_say_date_with_format(chan,t,AST_DIGIT_ANY,ast_channel_language(chan),"'vm-received' dB 'digits/at' k 'and' M",NULL);