asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	244491f9b2	app_reload: New Reload application Adds an application to reload modules from within the dialplan. ASTERISK-29454 Change-Id: Ic8ab025d8b38dd525b872b41c465c999c5810774	4 years ago
Naveen Albert	c01b4e0d4b	app_waitforcond: New application 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	4 years ago
Naveen Albert	1e5a2cfe30	app_dial: Expanded A option to add caller announcement 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	4 years ago
Naveen Albert	b742514553	app_originate: Allow setting Caller ID and variables 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	4 years ago
Naveen Albert	35437879e5	app_confbridge: New ConfKick() application 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	4 years ago
Naveen Albert	5f8cabc232	app_confbridge: New option to prevent answer supervision 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	4 years ago
Naveen Albert	567ea5abf8	app_voicemail: Configurable voicemail beep 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	4 years ago
Sean Bright	aac442eecd	app_queue.c: Remove dead 'updatecdr' code. 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	4 years ago
Sean Bright	8d3d7bdb82	app_queue.c: Don't crash when realtime queue name is empty. ASTERISK-27542 #close Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a	4 years ago
Joshua C. Colp	a8a08bcd1e	app_queue: Only send QueueMemberStatus if status changes. 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	4 years ago
Joshua C. Colp	149e5e5b86	xml: Embed module information into core XML documentation. 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	4 years ago
Joshua C. Colp	7438586d8e	documentation: Fix non-matching module support levels. 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	4 years ago
Sean Bright	8987de270f	app_dial.c: Only send DTMF on first progress event. ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	4 years ago
Sean Bright	932eae69ab	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	4d8fc97e4a	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	6e695c867f	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	4 years ago
Sean Bright	4a71b08091	app_read: Release tone zone reference on early return. Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a	4 years ago
Dan Cropp	55891227e8	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	3bcf483373	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	44d68bd56b	app_voicemail: Prevent deadlocks when out of ODBC database connections ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	4 years ago
Sean Bright	357510cec3	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	eda3679c1c	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	73f458b1e0	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	57ee79a563	Compiler fixes for GCC with -Og ASTERISK-29144 Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62	5 years ago
George Joseph	773f424c7f	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
Sean Bright	4b5ed817bd	app_voicemail.c: Document VMSayName interruption behavior ASTERISK-26424 #close Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0	5 years ago
Kfir Itzhak	c3a3ab8628	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	c925ed0eb9	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
Evandro César Arruda	b2bd38a4f0	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
George Joseph	64ca2d48da	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	647c53c41f	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

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5125 Commits (785e4afc20be77f33d024c2b13bc2f8afa6627b0)