5095 Commits (16.17)

Author	SHA1	Message	Date
Sean Bright	dcfcf7d70a	app_dial.c: Only send DTMF on first progress event. ... ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	4 years ago
Sean Bright	8604867857	app_page.c: Don't fail to Page if beep sound file is missing ... ASTERISK-16799 #close Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4	4 years ago
Ivan Poddubnyi	c63ab5580c	app_queue: Fix conversion of complex extension states into device states ... Queue members using dialplan hints as a state interface must handle INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE. ASTERISK-28369 Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f	4 years ago
Sebastien Duthil	435d68be97	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ... ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	4 years ago
Sean Bright	62ca911d29	app_read: Release tone zone reference on early return. ... Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a	4 years ago
Dan Cropp	a5364ac69b	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	4 years ago
Kevin Harwell	e96f744816	app_mixmonitor: cleanup datastore when monitor thread fails to launch ... launch_monitor_thread is responsible for creating and initializing the mixmonitor, and dependent data structures. There was one off nominal path after the datastore gets created that triggers when the channel being monitored is hung up prior to monitor starting itself. If this happened the monitor thread would not "launch", and the mixmonitor object and associated objects are freed, including the underlying datastore data object. However, the datastore itself was not removed from the channel, so when the channel eventually gets destroyed it tries to access the previously freed datastore data and crashes. This patch removes and frees datastore object itself from the channel before freeing the mixmonitor object thus ensuring the channel does not call it when destroyed. ASTERISK-28947 #close Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8	4 years ago
Sean Bright	2b7af3eb27	app_voicemail: Prevent deadlocks when out of ODBC database connections ... ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	4 years ago
Sean Bright	0a23296834	app_chanspy: Spyee information missing in ChanSpyStop AMI Event ... The documentation in the wiki says there should be spyee-channel information elements in the ChanSpyStop AMI event. https://wiki.asterisk.org/wiki/x/Xc5uAg However, this is not the case in Asterisk <= 16.10.0 Version. We're using these Spyee* arguments since Asterisk 11.x, so these arguments vanished in Asterisk 12 or higher. For maximum compatibility, we still send the ChanSpyStop event even if we are not able to find any 'Spyee' information. ASTERISK-28883 #close Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f	4 years ago
Joshua C. Colp	15566494f9	voicemail: add option 'e' to play greetings as early media ... When using this option, answering the channel is deferred until all prompts/greetings have been played and the caller is about to leave their message. ASTERISK-29118 #close Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb	5 years ago
George Joseph	8d8c9db618	app_queue: Fix deadlock between update and show queues ... Operations that update queues when shared_lastcall is set lock the queue in question, then have to lock the queues container to find the other queues with the same member. On the other hand, __queues_show (which is called by both the CLI and AMI) does the reverse. It locks the queues container, then iterates over the queues locking each in turn to display them. This creates a deadlock. * Moved queue print logic from __queues_show to a separate function that can be called for a single queue. * Updated __queues_show so it doesn't need to lock or traverse the queues container to show a single queue. * Updated __queues_show to snap a copy of the queues container and iterate over that instead of locking the queues container and iterating over it while locked. This prevents us from having to hold both the container lock and the queue locks at the same time. This also allows us to sort the queue entries. ASTERISK-29155 Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41	5 years ago
Alexander Traud	13b56c4be6	Compiler fixes for GCC with -Og ... ASTERISK-29144 Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62	5 years ago
Sean Bright	9ac933fbba	app_voicemail.c: Document VMSayName interruption behavior ... ASTERISK-26424 #close Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0	5 years ago
George Joseph	3b0a53f257	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
Kfir Itzhak	ae718c975a	app_queue: Fix leave-empty not recording a call as abandoned ... This fixes a bug introduced mistakenly in ASTERISK-25665: If leave-empty is enabled, a call may sometimes be removed from a queue without recording it as abandoned. This causes Asterisk to not generate an abandon event for that call, and for the queue abandoned counter to be incorrect. ASTERISK-29043 #close Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7	5 years ago
Sean Bright	53a88df076	app_voicemail: Fix pollmailboxes ... The name of the voicemail context was overwriting the name of the subscribed mailbox. Fix by simplifying how we create the MWI subscription. ASTERISK-29029 #close Change-Id: Ie8a7db6a0b68f3995b0846bbb733a21909ba44e5	5 years ago
Evandro César Arruda	aa93107444	app_queue: Member lastpause time reseting ... This fixes the reseting members lastpause problem when realtime members is being used, the function rt_handle_member_record was forcing the reset members lastpause because it does not exist in realtime ASTERISK-29034 #close Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5	5 years ago
Sean Bright	788b60d935	app_voicemail: Process urgent messages with mailcmd ... Rather than putting messages into INBOX and then moving them to Urgent later, put them directly in to the Urgent folder. This prevents mailcmd from being skipped. ASTERISK-27273 #close Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5	5 years ago
Joshua C. Colp	397aa391b7	app_stream_echo: Fix state of added streams. ... When stream support was added to Asterisk the stream state was used inconsistently, resulting in odd behavior. This was then standardized to be the state of a stream from the perspective of Asterisk. This change updates the StreamEcho dialplan application to use the correct state, send only, since we are only sending to the endpoint and not expecting them to send us multiple video streams. ASTERISK-28954 Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56	5 years ago
Walter Doekes	83b4507500	app_queue: Read latest wrapuptime instead of (possibly stale) copy ... Before this changeset, it was possible that a queue member (agent) was called even though they just got out of a call, and wrapuptime seconds hadn't passed yet. This could happen if a member ended a call _between_ a new call attempt and asterisk trying that particular member for a new call. In that case, Asterisk would check the hangup time of the call-before-the-last-call instead of the hangup time of the-last-call. ASTERISK-28952 Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3	5 years ago
Walter Doekes	e8cb8957ec	app_queue: Remove stale code in try_calling ... Because ring_entry() is not called, outgoing->chan is not touched here either. ASTERISK-28950 ASTERISK-28644 Change-Id: I564613715dfaf45af868251eb75a451f512af90f	5 years ago
Kevin Harwell	f59001d171	Compiler fixes for gcc 10 ... This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	5 years ago
George Joseph	c7406a5b48	app_confbridge: Plug ref leak of bridge channel with send_events ... When send_events is enabled for a user, we were leaking a reference to the bridge channel in confbridge_manager.c:send_message(). This also caused the bridge snapshot to not be destroyed. Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97	5 years ago
traud	a5100e7171	app_osplookup: Avoid a format truncation. ... Ensure that output buffers for the osp_convert_inout function have sufficient space for additional data such as brackets and ports. ASTERISK-28804 Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b	5 years ago
Nathan Bruning	92169e6f8a	app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions ... Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6	5 years ago
George Joseph	cf5e30dd0e	app_voicemail: Add workaround for a gcc 10 issue with -Wrestrict ... The gcc 10 -Wrestrict option was causing "overlap" errors when snprintf was copying one char[256] structure member to another char[256] member in the same structure. Using ast_alloca instead of declaring the structure inline solves the issue. Here's a link to the "enhancement": https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html We may follow up with a gcc bug report. Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534	5 years ago
Alexander Traud	302c20546a	app_fax: SpanDSP headers do not use ast_malloc; ignore that. ... Since Asterisk 14, app_fax did not compile at all because Asterisk requires that not malloc but ast_malloc is used everywhere. However, the system headers of SpanDSP use malloc. Because we cannot (and do not need to) change system headers, let us ignore this. ASTERISK-28848 Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546	5 years ago
Alexander Traud	d89968fc96	app_getcpeid: Add build-time dependency. ... ASTERISK-28838 Change-Id: I68b78e7e4718be82507247433127ce3992a5ba96	5 years ago
Joshua C. Colp	62183bc777	confbridge: Add support for disabling text messaging. ... When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	5 years ago
Jaco Kroon	d278350768	dahdiras: Only set plugin dahdi.so to pppd if we're running as root. ... Users of this should set plugin dahdi.so in their options file. ASTERISK-16676 Change-Id: I6d01ad0a10e9fea477876d0941c3f38aac357e91	5 years ago
Joshua C. Colp	c40050d350	audiohook: Don't allow audiohooks to attach to hung up channels. ... Given a scenario where MixMonitor was initiated over AMI it was possible for the channel and MixMonitor thread to remain alive past hang up of the channel. This scenario required the AMI initiated MixMonitor to retrieve the channel, a hangup to occur on the channel in another thread, and then for MixMonitor to actually start. If this occurred the MixMonitor thread would remain alive indefinitely and the channel reference would remain. This change ensures that audiohooks are never able to be attached to channels that have been hung up. An additional fix has also been done in app_mixmonitor to properly release the channel reference if this occurs. ASTERISK-28780 Change-Id: I8044c06daa06f0f16607788c596f55623be26f58	5 years ago
Walter Doekes	b12ecbe27f	app_queue: Refactor odd placement of if's around say_position ... Change-Id: Icba97905e331812f129e5966e91a59b104c7a748	5 years ago
Sean Bright	cd8b27dcc2	app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used ... When opening a file for writing, Asterisk silently converts filenames ending with 'wav49' to 'WAV.' We aren't taking that in to account when setting the MIXMONITOR_FILENAME variable in MixMonitor. * If the user wants to write to a wav49 file, make sure that it is reflected properly in MIXMONITOR_FILENAME. * Add a note to the documentation describing this behavior. * Add a note in main/file.c indicating that app_mixmonitor needs to be changed if the logic in build_filename was changed. ASTERISK-24798 #close Reported by: xrobau Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c	5 years ago
Joshua Colp	5166088628	Merge "app_voicemail, say: Fix various leading whitespace problems" into 16	5 years ago
Friendly Automation	f66885225a	Merge "app_voicemail: Prevent crash when saving message with realtime voicemail" into 16	5 years ago
Friendly Automation	23515dea5c	Merge "app_voicemail: Set globals to default values when voicemail.conf missing" into 16	5 years ago
Sean Bright	13fa33588f	app_voicemail, say: Fix various leading whitespace problems ... In af90afd90c, Japanese language support was added to app_voicemail and main/say.c, but the leading whitespace is not consistent with Asterisk coding guidelines. This patch fixes that. Whitespace only, no functional change. ASTERISK~23324 Reported by: Kevin McCoy Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87	5 years ago
Sean Bright	293600724d	app_voicemail: Prevent crash when saving message with realtime voicemail ... ast_store_realtime() is not NULL tolerant, so we need to initialize the field values we pass to it to the empty string to avoid a crash. ASTERISK-23739 #close Reported by: Stas Kobzar Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c	5 years ago
Sean Bright	e379fe48e1	app_voicemail: Set globals to default values when voicemail.conf missing ... If voicemail.conf exists but is empty, the config parsing process will default a number of global variables to non-zero values. On the other hand, if voicemail.conf is missing (arguably semantically equivalent to an empty file), this process is skipped and the globals are defaulted to 0. Set the globals to the same values they would be set to if a configuration were present. This allows voicemail configuration to be done completely by Realtime without the need to create an empty voicemail.conf file. ASTERISK-27622 #close Reported by: Jim Van Meggelen Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409	5 years ago
Sean Bright	5fded77e7f	app_queue: Deprecate the QueueMemberPause.Reason field ... The QueueMemberPause AMI event includes two fields that return the reason a member was paused. * In release branches, deprecate Reason in favor of PausedReason. * In master, remove the Reason field entirely. ASTERISK-28349 #close Reported by: Niksa Baldun Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296	5 years ago
Joshua Colp	6dc9239367	Merge "app_record: Do not hang up if beep audio is missing" into 16	5 years ago
Corey Farrell	0c07a7ee00	app_record: Do not hang up if beep audio is missing ... Additionally alter the warning to mention that it was "beep" which could not be streamed to give admins a better clue about what the warning means. ASTERISK-28682 Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e	5 years ago
Kevin Harwell	de078debab	app_agent_pool: Update XML docs for AgentLogin ... This patch fixes some wrongly formatted documentation for the AgentLogin application. A couple of "see also" links should contain only the function name, and no parameters. Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661	5 years ago
Friendly Automation	7c227aa130	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." into 16	5 years ago
Friendly Automation	077bb05528	Merge "app_dial.c: Simplify dialplan using Dial." into 16	5 years ago
Joshua Colp	5352022b2c	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message." into 16	5 years ago
Friendly Automation	ab042335f5	Merge "app_page.c: Simplify dialplan using Page." into 16	5 years ago
Richard Mudgett	a7692ce2f4	app_chanisavail.c: Simplify dialplan using ChanIsAvail. ... Dialplan has to be careful about passing an empty device list or empty positions in the list. As a result, dialplan has to check for these conditions before using ChanIsAvail. Simplify dialplan by making ChanIsAvail handle these conditions gracefully. * Made tolerate empty positions in the device list. * Simplified the code and eliminated some unnecessary indention. ASTERISK-28638 Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3	5 years ago
Richard Mudgett	144b774b85	app_dial.c: Simplify dialplan using Dial. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Dial. Simplify dialplan by making Dial handle these conditions gracefully. * Made tolerate empty positions in the dialed device list. * Reduced some message log levels from notice to verbose. ASTERISK-28638 Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9	5 years ago
Richard Mudgett	2780be334d	app_page.c: Simplify dialplan using Page. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Page. Simplify dialplan by making Page handle these conditions gracefully. * Made tolerate empty positions in the paged device list. * Reduced some warnings associated with the 's' option to verbose messages. The warning level for those messages really serves no purpose as that is why the 's' option exists. ASTERISK-28638 Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3	5 years ago