asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	1d64cb99d2	Update for 20.2.1	2 years ago
Mike Bradeen	f8dfbaf225	res_pjsip_pubsub: subscription cleanup changes 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	2 years ago
Sean Bright	6e50550d28	Revert "pbx_ael: Global variables are not expanded." This reverts commit `56051d1ac5`. Reason for revert: Behavior change that breaks existing dialplan. ASTERISK-30472 #close Change-Id: I83bed3b800d36228a04ded0a6164b795f7f16bd6	2 years ago
Asterisk Development Team	e7f0440032	Update for 20.2.0	2 years ago
Asterisk Development Team	460d13e916	Update for 20.2.0-rc1	2 years ago
Asterisk Development Team	93813c9dca	Update CHANGES and UPGRADE.txt for 20.2.0	2 years ago
George Joseph	ceda5a9859	res_pjsip: Replace invalid UTF-8 sequences in callerid name * Added a new function ast_utf8_replace_invalid_chars() to utf8.c that copies a string replacing any invalid UTF-8 sequences with the Unicode specified U+FFFD replacement character. For example: "abc\xffdef" becomes "abc\uFFFDdef". Any UTF-8 compliant implementation will show that character as a � character. * Updated res_pjsip:set_id_from_hdr() to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. * Updated stasis_channels:ast_channel_publish_varset to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. ASTERISK-27830 Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8	2 years ago
Sean Bright	e5c5cd6e25	test.c: Avoid passing -1 to FD_* family of functions. This avoids buffer overflow errors when running tests that capture output from child processes. This also corrects a copypasta in an off-nominal error message. Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d	2 years ago
Naveen Albert	ede67a99be	chan_iax2: Fix jitterbuffer regression prior to receiving audio. ASTERISK_29392 (a security fix) introduced a regression by not processing frames when we don't have an audio format. Currently, chan_iax2 only calls jb_get to read frames from the jitterbuffer when the voiceformat has been set on the pvt. However, this only happens when we receive a voice frame, which means that prior to receiving voice frames, other types of frames get stalled completely in the jitterbuffer. To fix this, we now fallback to using the format negotiated during call setup until we've actually received a voice frame with a format. This ensures we're always able to read from the jitterbuffer. ASTERISK-30354 #close ASTERISK-30162 #close Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9	2 years ago
Sean Bright	827222d607	test_crypto.c: Fix getcwd(…) build error. `getcwd(…)` is decorated with the `warn_unused_result` attribute and therefore needs its return value checked. Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93	2 years ago
Nick French	200dc7d0e8	pjproject_bundled: Fix cross-compilation with SSL libs. Asterisk makefiles auto-detect SSL library availability, then they assume that pjproject makefiles will also autodetect an SSL library at the same time, so they do not pass on the autodetection result to pjproject. This normally works, except the pjproject makefiles disables autodetection when cross-compiling. Fix by explicitly configuring pjproject to use SSL if we have been told to use it or it was autodetected ASTERISK-30424 #close Change-Id: I8fe2999ea46710e21d1d55a1bed92769c6ebded9	2 years ago
Mike Bradeen	5c11d7adea	app_read: Add an option to return terminator on empty digits. Adds 'e' option to allow Read() to return the terminator as the dialed digits in the case where only the terminator is entered. ie; if "#" is entered, return "#" if the 'e' option is set and "" if it is not. ASTERISK-30411 Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5	2 years ago
cmaj	5b0e3444c3	res_phoneprov.c: Multihomed SERVER cache prevention Phones moving between subnets on multi-homed server have their initially connected interface IP cached in the SERVER variable, even when it is not specified in the configuration files. This prevents phones from obtaining the correct SERVER variable value when they move to another subnet. ASTERISK-30388 #close Reported-by: cmaj Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92	2 years ago
Mike Bradeen	2308afed8e	app_directory: Add a 'skip call' option. Adds 's' option to skip calling the extension and instead set the extension as DIRECTORY_EXTEN channel variable. ASTERISK-30405 Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299	2 years ago
Mike Bradeen	98742388b6	app_senddtmf: Add option to answer target channel. Adds a new option to SendDTMF() which will answer the specified channel if it is not already up. If no channel is specified, the current channel will be answered instead. ASTERISK-30422 Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085	2 years ago
Mike Bradeen	37e558f6ef	res_pjsip: Prevent SEGV in pjsip_evsub_send_request contributed pjproject - patch to check sub->pending_notify in evsub.c:on_tsx_state before calling pjsip_evsub_send_request() res_pjsip_pubsub - change post pjsip 2.13 behavior to use pubsub_on_refresh_timeout to avoid the ao2_cleanup call on the sub_tree. This is is because the final NOTIFY send is no longer the last place the sub_tree is referenced. ASTERISK-30419 Change-Id: Ib5cc662ce578e9adcda312e16c58a10b6453e438	2 years ago
Sean Bright	aeb16aa7d8	app_queue: Minor docs and logging fixes for UnpauseQueueMember. ASTERISK-30417 #close Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe	2 years ago
Sean Bright	aef0c0ce0e	app_queue: Reset all queue defaults before reload. Several queue fields were not being set to their default value during a reload. Additionally added some sample configuration options that were missing from queues.conf.sample. Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4	2 years ago
Mike Bradeen	58636a6ea6	res_pjsip: Upgraded bundled pjsip to 2.13 Removed multiple patches. Code chages in res_pjsip_pubsub due to changes in evsub. Pjsip now calls on_evsub_state() before on_rx_refresh(), so the sub tree deletion that used to take place in on_evsub_state() now must take place in on_rx_refresh(). Additionally, pjsip now requires that you send the NOTIFY from within on_rx_refresh(), otherwise it will assert when going to send the 200 OK. The idea is that it will look for this NOTIFY and cache it until after sending the response in order to deal with the self-imposed message mis-order. Asterisk previously dealt with this by pushing the NOTIFY in on_rx_refresh(), but pjsip now forces us to use it's method. Changes were required to configure in order to detect which way pjsip handles this as the two are not compatible for the reasons mentioned above. A corresponding change in testsuite is required in order to deal with the small interal timing changes caused by moving the NOTIFY send. ASTERISK-30325 Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f	2 years ago
Sean Bright	96d9ad51ac	doxygen: Fix doxygen errors. Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	2 years ago
Naveen Albert	88b2c741ca	app_signal: Add signaling applications Adds the Signal and WaitForSignal applications, which can be used for inter-channel signaling in the dialplan. Signal supports sending a signal to other channels listening for a signal of the same name, with an optional data payload. The signal is received by all channels waiting for that named signal. ASTERISK-29810 #close Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3	2 years ago
Mike Bradeen	70856e865f	app_directory: add ability to specify configuration file Adds option to app_directory to specify a filename from which to read configuration instead of voicemail.conf ie; same => n,Directory(,,c(directory.conf)) This configuration should contain a list of extensions using the voicemail.conf format, ie; 2020=2020,Dog Dog,,,,attach=no\|saycid=no\|envelope=no\|delete=no ASTERISK-30404 Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13	2 years ago
Naveen Albert	8a45cd7af4	func_json: Enhance parsing capabilities of JSON_DECODE Adds support for arrays to JSON_DECODE by allowing the user to print out entire arrays or index a particular key or print the number of keys in a JSON array. Additionally, adds support for recursively iterating a JSON tree in a single function call, making it easier to parse JSON results with multiple levels. A maximum depth is imposed to prevent potentially blowing the stack. Also fixes a bug with the unit tests causing an empty string to be printed instead of the actual test result. ASTERISK-29913 #close Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b	2 years ago
sungtae kim	f99849f8d5	res_stasis_snoop: Fix snoop crash Added NULL pointer check and channel lock to prevent resource release while the chanspy is processing. ASTERISK-29604 Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24	2 years ago
Sean Bright	56051d1ac5	pbx_ael: Global variables are not expanded. Variable references within global variable assignments are now expanded rather than being included literally. ASTERISK-30406 #close Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d	2 years ago
Naveen Albert	a1da8042d1	res_pjsip_session: Add overlap_context option. Adds the overlap_context option, which can be used to explicitly specify a context to use for overlap dialing extension matches, rather than forcibly using the context configured for the endpoint. ASTERISK-30262 #close Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db	2 years ago
Sean Bright	ef16eaee36	app_playback.c: Fix PLAYBACKSTATUS regression. In Asterisk 11, if a channel was redirected away during Playback(), the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 (specifically commit `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	2 years ago
George Joseph	2f5aece0c9	res_rtp_asterisk: Don't use double math to generate timestamps Rounding issues with double math were causing rtp timestamp slips in outgoing packets. We're now back to integer math and are getting no more slips. ASTERISK-30391 Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4	2 years ago
Alexei Gradinari	e86d5d7fda	format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...) Each playback of WAV files results in logging "Skipping unknown block 'LIST'". To prevent unnecessary flooding of this DEBUG log this patch replaces ast_log(LOG_DEBUG, ...) by ast_debug(1, ...). Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0	2 years ago
Igor Goncharovsky	3526441e41	res_pjsip_rfc3326: Add SIP causes support for RFC3326 Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	2 years ago
George Joseph	4710f37ef6	res_rtp_asterisk: Asterisk Media Experience Score (MES) ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	2 years ago
George Joseph	62ca063fca	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `d454801c2d`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86	2 years ago
Naveen Albert	d33bd6d67a	loader: Allow declined modules to be unloaded. Currently, if a module declines to load, dlopen is called to register the module but dlclose never gets called. Furthermore, loader.c currently doesn't allow dlclose to ever get called on the module, since it declined to load and the unload function bails early in this case. This can be problematic if a module is updated, since the new module cannot be loaded into memory since we haven't closed all references to it. To fix this, we now allow modules to be unloaded, even if they never "loaded" in Asterisk itself, so that dlclose is called and the module can be properly cleaned up, allowing the updated module to be loaded from scratch next time. ASTERISK-30345 #close Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5	2 years ago
Naveen Albert	e06fe8e344	app_broadcast: Add Broadcast application Adds a new application, Broadcast, which can be used for one-to-many transmission and many-to-one reception of channel audio in Asterisk. This is similar to ChanSpy, except it is designed for multiple channel targets instead of a single one. This can make certain kinds of audio manipulation more efficient and streamlined. New kinds of audio injection impossible with ChanSpy are also made possible. ASTERISK-30180 #close Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726	2 years ago
Naveen Albert	68e345286b	func_frame_trace: Print text for text frames. Since text frames contain a text body, make FRAME_TRACE more useful for text frames by actually printing the text. ASTERISK-30353 #close Change-Id: Ia6ce3d15cecd7a673a528d34faac86854a2bab50	2 years ago
Naveen Albert	3b3fef2347	json.h: Add ast_json_object_real_get. json.h contains macros to get a string and an integer from a JSON object. However, the macro to do this for JSON reals is missing. This adds that. ASTERISK-30361 #close Change-Id: I8d0e28d763febf27b05801cdc83b73282aa6ee7a	2 years ago
Naveen Albert	7b8f7428da	manager: Fix appending variables. The if statement here is always false after the for loop finishes, so variables are never appended. This removes that to properly append to the end of the variable list. ASTERISK-30351 #close Reported by: Sebastian Gutierrez Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199	2 years ago
George Joseph	24102ba236	res_pjsip_transport_websocket: Add remote port to transport When Asterisk receives a new websocket conenction, it creates a new pjsip transport for it and copies connection data into it. The transport manager then uses the remote IP address and port on the transport to create a monitor for each connection. However, the remote port wasn't being copied, only the IP address which meant that the transport manager was creating only 1 monitoring entry for all websocket connections from the same IP address. Therefore, if one of those connections failed, it deleted the transport taking all the the connections from that same IP address with it. * We now copy the remote port into the created transport and the transport manager behaves correctly. ASTERISK-30369 Change-Id: Ib506d40897ea6286455ac0be4dfbb0ed43b727e1	2 years ago
Boris P. Korzun	edc90c96ac	http.c: Fix NULL pointer dereference bug If native HTTP is disabled but HTTPS is enabled and status page enabled too, Core/HTTP crashes while loading. 'global_http_server' references to NULL, but the status page tries to dereference it. The patch adds a check for HTTP is enabled. ASTERISK-30379 #close Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249	2 years ago
Holger Hans Peter Freyther	3d9b9a2b16	res_http_media_cache: Do not crash when there is no extension Do not crash when a URL has no path component as in this case the ast_uri_path function will return NULL. Make the code cope with not having a path. The below would crash > media cache create http://google.com /tmp/foo.wav Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault. 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 (gdb) bt #0 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6 #1 0x0000ffff43d43a78 in file_extension_from_string (str=<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	2 years ago
George Joseph	d454801c2d	res_rtp_asterisk: Asterisk Media Experience Score (MES) This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	2 years ago
Naveen Albert	cc8d9b947b	pbx_app: Update outdated pbx_exec channel snapshots. pbx_exec makes a channel snapshot before executing applications. This doesn't cause an issue during normal dialplan execution where pbx_exec is called over and over again in succession. However, if pbx_exec is called "one off", e.g. using ast_pbx_exec_application, then a channel snapshot never ends up getting made after the executed application returns, and inaccurate snapshot information will linger for a while, causing "core show channels", etc. to show erroneous info. This is fixed by manually making a channel snapshot at the end of ast_pbx_exec_application, since we anticipate that pbx_exec might not get called again immediately. ASTERISK-30367 #close Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086	2 years ago
Naveen Albert	c7598ee947	res_pjsip_session: Use Caller ID for extension matching. Currently, there is no Caller ID available to us when checking for an extension match when handling INVITEs. As a result, extension patterns that depend on the Caller ID are not matched and calls may be incorrectly rejected. The Caller ID is not available because the supplement that adds Caller ID to the session does not execute until after this check. Supplement callbacks cannot yet be executed at this point since the session is not yet in the appropriate state. To fix this without impacting existing behavior, the Caller ID number is now retrieved before attempting to pattern match. This ensures pattern matching works correctly and there is no behavior change to the way supplements are called. ASTERISK-28767 #close Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7	2 years ago
Ben Ford	881faf544f	res_pjsip_sdp_rtp.c: Use correct timeout when put on hold. When a call is put on hold and it has moh_passthrough and rtp_timeout set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is expected to be used, but rtp_timeout is used instead. This change adds a couple of checks for locally_held to determine if rtp_timeout_hold needs to be used instead of rtp_timeout. ASTERISK-30350 Change-Id: I7b106fc244332014216d12bba851cefe884cc25f	2 years ago
Naveen Albert	20d4775d0a	app_voicemail_odbc: Fix string overflow warning. Fixes a negative offset warning by initializing the buffer to empty. Additionally, although it doesn't currently complain about it, the size of a buffer is increased to accomodate the maximum size contents it could have. ASTERISK-30240 #close Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877	2 years ago
Naveen Albert	cbb1fd2cb9	func_callerid: Warn about invalid redirecting reason. Currently, if a user attempts to set a Caller ID related function to an invalid value, a warning is emitted, except for when setting the redirecting reason. We now emit a warning if we were unable to successfully parse the user-provided reason. ASTERISK-30332 #close Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a	2 years ago
Igor Goncharovsky	115a1b4f0a	res_pjsip: Fix path usage in case dialing with '@' Fix aor lookup on sip path addition. Issue happens in case of dialing with @ and overriding user part of RURI. ASTERISK-30100 #close Reported-by: Yury Kirsanov Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13	2 years ago
Peter Fern	58404b5c22	streams: Ensure that stream is closed in ast_stream_and_wait on error When ast_stream_and_wait returns an error (for example, when attempting to stream to a channel after hangup) the stream is not closed, and callers typically do not check the return code. This results in leaking file descriptors, leading to resource exhaustion. This change ensures that the stream is closed in case of error. ASTERISK-30198 #close Reported-by: Julien Alie Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803	2 years ago
Naveen Albert	36bea9ad33	app_sendtext: Remove references to removed applications. Removes see-also references to applications that don't exist anymore (removed in Asterisk 19), so these dead links don't show up on the wiki. ASTERISK-30347 #close Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909	2 years ago
Asterisk Development Team	fefc236e7c	Update CHANGES and UPGRADE.txt for 20.1.0	2 years ago

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33537 Commits (20.2) All Branches Search

33537 Commits (20.2)

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