5102 Commits (18.1)

Author	SHA1	Message	Date
Alexander Traud	e0ee53dc9c	Compiler fixes for GCC with -Og ... ASTERISK-29144 Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62	5 years ago
Sean Bright	abee490639	app_voicemail.c: Document VMSayName interruption behavior ... ASTERISK-26424 #close Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0	5 years ago
George Joseph	4a049ad510	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	c83e4821e5	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
Evandro César Arruda	36dd15c659	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	b575868000	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
George Joseph	c4c72d55a2	scope_trace: Added debug messages and added additional macros ... The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages at the same level as the scope level. This allows the same messages to be printed to the debug log when AST_DEVMODE isn't enabled. Also added a few variants of the SCOPE_EXIT macros that will also call ast_log instead of ast_debug to make it easier to use scope tracing and still print error messages. Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21	5 years ago
George Joseph	6faf76308d	ACN: Changes specific to the core ... Allow passing a topology from the called channel back to the calling channel. * Added a new function ast_queue_answer() that accepts a stream topology and queues an ANSWER CONTROL frame with it as the data. This allows the called channel to indicate its resolved topology. * Added a new virtual function to the channel tech structure answer_with_stream_topology() that allows the calling channel to receive the called channel's topology. Added ast_raw_answer_with_stream_topology() that invokes that virtual function. * Modified app_dial.c and features.c to grab the topology from the ANSWER frame queued by the answering channel and send it to the calling channel with ast_raw_answer_with_stream_topology(). * Modified frame.c to automatically cleanup the reference to the topology on ANSWER frames. Added a few debugging messages to stream.c. Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c	5 years ago
Walter Doekes	312c23b0e1	app_queue: (Breaking change) shared_lastcall and autofill default to no ... If your queues.conf had _no_ [general] section, they would default to 'yes'. Now, they always default to 'no'. (Actually, commit ed615afb7e already partially fixed it for shared_lastcall.) ASTERISK-28951 Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6	5 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	5 years ago
Joshua C. Colp	00a52b4752	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	db012e8cc6	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
Walter Doekes	0fb6738314	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
George Joseph	b9f42a717e	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
Kevin Harwell	3d1bf3c537	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
traud	527e4f6542	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	f217fcdc62	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	7baf2c4bf1	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	26b8c99963	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
Joshua C. Colp	6cfc6ff53c	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
Alexander Traud	5c2b8fdeca	app_getcpeid: Add build-time dependency. ... ASTERISK-28838 Change-Id: I68b78e7e4718be82507247433127ce3992a5ba96	5 years ago
Jaco Kroon	4f92dcd66b	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	98d10d0a16	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
Kevin Harwell	2d9ecd9cd1	Merge "app_queue: Refactor odd placement of if's around say_position"	5 years ago
Kevin Harwell	999fdef335	Merge "app_mixmonitor: Turn on synchronization by default"	5 years ago
Walter Doekes	680e6b9774	app_queue: Refactor odd placement of if's around say_position ... Change-Id: Icba97905e331812f129e5966e91a59b104c7a748	5 years ago
Sean Bright	8dcdce42a9	app_mixmonitor: Turn on synchronization by default ... The optional synchronization behavior created in 64906c4c9b is now the default for MixMonitor. * Add a new flag 'n' that allows for this behavior to be turned off * Add a notice when the 'S' option is used indicating that it is no longer necessary Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4	5 years ago
Sean Bright	ddfb60ac2c	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
Friendly Automation	95c6fbeae0	Merge "app_voicemail: Remove MessageExists and MESSAGE_EXISTS()"	5 years ago
Joshua Colp	64debbd13f	Merge "app_voicemail, say: Fix various leading whitespace problems"	5 years ago
Joshua Colp	058e9f735e	Merge "app_voicemail: Prevent crash when saving message with realtime voicemail"	5 years ago
Joshua Colp	2d17e25015	Merge "app_voicemail: Set globals to default values when voicemail.conf missing"	5 years ago
Sean Bright	f09cf4da44	app_voicemail: Remove MessageExists and MESSAGE_EXISTS() ... * The MailboxExists dialplan application was deprecated on 2006-09-26 in Asterisk 1.6.0 (commit ec83b11183) * The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in Asterisk 11.0.0 (commit fd64bb66f9) Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f	5 years ago
Sean Bright	5cbf47714a	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	ba8ccb9132	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
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	5 years ago
Friendly Automation	c665878e92	Merge "app_queue: Deprecate the QueueMemberPause.Reason field"	5 years ago
Sean Bright	9be89d9913	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
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. ... This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	5 years ago
Joshua Colp	f6f678fe7d	Merge "app_record: Do not hang up if beep audio is missing"	5 years ago
Sean Bright	9522390a69	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
Corey Farrell	2f8b20b949	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	00a7432156	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
George Joseph	a4fd89536d	Merge "app_bridgeaddchan.c: Make BridgeAdd be more like Bridge"	5 years ago
George Joseph	6b7334a311	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail."	5 years ago
Friendly Automation	5b815fe1ac	Merge "app_dial.c: Simplify dialplan using Dial."	5 years ago
Friendly Automation	5050c45e06	Merge "app_page.c: Simplify dialplan using Page."	5 years ago
Joshua Colp	b2664fd3a4	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message."	5 years ago
Richard Mudgett	fe3cce816c	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	19069f7db7	app_bridgeaddchan.c: Make BridgeAdd be more like Bridge ... * Made BridgeAdd not hangup the call if there is a problem. * Reduced message level from warning to verbose for normal exception cases. * Added a loop safety check to BridgeAdd. * Made BridgeAdd set BRIDGERESULT with the status when dialplan is resumed. Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426	5 years ago