5211 Commits (fee9012fe16cb87ccdf23ea0abdc1885e811ae4b)

Author	SHA1	Message	Date
Naveen Albert	b9c031c1f8	app_voicemail: Fix missing email in msg_create_from_file. ... 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	3 years ago
Naveen Albert	b365ea8601	app_if: Adds conditional branch applications ... 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	3 years ago
Naveen Albert	691178c48e	app_mixmonitor: Add option to use real Caller ID for voicemail. ... 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	3 years ago
Naveen Albert	2efa290d3c	sla: Prevent deadlock and crash due to autoservicing. ... 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	3 years ago
Naveen Albert	6e59b01e1a	app_mixmonitor: Add option to delete files on exit. ... 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	3 years ago
Naveen Albert	dfe2f38642	app_stack: Print proper exit location for PBXless channels. ... 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	3 years ago
Naveen Albert	1e29607b5c	app_bridgewait: Add option to not answer channel. ... 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	3 years ago
Naveen Albert	8c791f9a65	app_amd: Add option to play audio during AMD. ... 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	3 years ago
Naveen Albert	205c7c8d21	app_confbridge: Add end_marked_any option. ... 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	3 years ago
Naveen Albert	e2e049e473	general: Very minor coding guideline fixes. ... Fixes a few coding guideline violations: * Use of C99 comments * Opening brace on same line as function prototype ASTERISK-30163 #close Change-Id: I07771c4c89facd41ce8d323859f022ddbddf6ca7	3 years ago
Naveen Albert	dc7ec11c26	app_confbridge: Fix memory leak on updated menu options. ... 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	3 years ago
Naveen Albert	c654486547	general: Improve logging levels of some log messages. ... 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	3 years ago
Naveen Albert	165368bf0b	general: Remove obsolete SVN references. ... 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	3 years ago
Naveen Albert	2d8f2696b2	app_confbridge: Add missing AMI documentation. ... Documents the ConfbridgeListRooms AMI response, which is currently not documented. ASTERISK-30020 #close Change-Id: Id6fff7a936244bae7b52686301eb740c1169cdea	3 years ago
Naveen Albert	4af881506e	app_meetme: Add missing AMI documentation. ... The MeetmeList and MeetmeListRooms AMI responses are currently completely undocumented. This adds documentation for these responses. ASTERISK-30018 #close Change-Id: Id93135b7edf01de6f8fba266e2122989dc8996b8	3 years ago
Naveen Albert	2843e5678d	app_confbridge: Always set minimum video update interval. ... 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	3 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Naveen Albert	626fefdf7d	app_dial: Fix dial status regression. ... 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	3 years ago
Naveen Albert	ae8a36a7d9	app_dial: Propagate outbound hook flashes. ... 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	3 years ago
Naveen Albert	4a11ae7ecf	pbx: Add helper function to execute applications. ... 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	3 years ago
Naveen Albert	cc8e098e1d	app_voicemail: Add option to prevent message deletion. ... 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	3 years ago
Naveen Albert	51d262af12	xmldocs: Improve examples. ... Use example tags instead of regular para tags where possible. ASTERISK-30090 Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5	3 years ago
Naveen Albert	432a1d2d7e	app_confbridge: Add function to retrieve channels. ... 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	3 years ago
George Joseph	4aa541683b	GCC12: Fixes for 16+ ... 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	3 years ago
Naveen Albert	892c06564f	chan_dahdi: Document dial resource options. ... 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	3 years ago
Michael Cargile	a2679b0ee2	apps/confbridge: Added hear_own_join_sound option to control who hears sound_join ... 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	3 years ago
Naveen Albert	b90650d8f4	app_meetme: Don't erroneously set global variables. ... 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	3 years ago
Naveen Albert	0c70d497bc	documentation: Adds versioning information. ... 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	3 years ago
Maximilian Fridrich	53a3af6321	app_dial: Flip stream direction of outgoing channel. ... 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	3 years ago
Naveen Albert	b87c5f5124	app_mf, app_sf: Return -1 if channel hangs up. ... 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	4 years ago
Naveen Albert	ede4e2099f	app_queue: Add music on hold option to Queue. ... 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	4 years ago
Naveen Albert	da44b848f5	app_meetme: Emit warning if conference not found. ... 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	4 years ago
Naveen Albert	1e87cadf8e	app_dial: Document DIALSTATUS return values. ... Adds documentation for all of the possible return values for the DIALSTATUS variable in the Dial application. ASTERISK-25716 Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0	4 years ago
Kfir Itzhak	2be01ba40b	app_queue: Add QueueWithdrawCaller AMI action ... 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	4 years ago
Naveen Albert	c35e205bef	documentation: Add since tag to xmldocs DTD ... 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	4 years ago
Naveen Albert	39820e3561	app_voicemail: Emit warning if asking for nonexistent mailbox. ... 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	4 years ago
Naveen Albert	c9ef2b3b86	app_mp3: Document and warn about HTTPS incompatibility. ... 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	4 years ago
Naveen Albert	0da713168d	app_mf: Add max digits option to ReceiveMF. ... 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	4 years ago
Alexei Gradinari	b41440a179	app_queue: load queues and members from Realtime when needed ... 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	4 years ago
Mark Petersen	93d090147f	app_queue.c: Queue don't play "thank-you" when here is no hold time announcements ... 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	4 years ago
Sean Bright	0d62735f99	utils.c: Remove all usages of ast_gethostbyname() ... 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	4 years ago
Mark Petersen	dc7bcd68e4	app_queue.c: Support for Nordic syntax in announcements ... 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	4 years ago
Naveen Albert	80766059ef	app_mp3: Throw warning on nonexistent stream ... 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	4 years ago
Naveen Albert	70bc0ff9d0	documentation: Add missing AMI documentation ... Adds missing documentation for some channel, bridge, and queue events. ASTERISK-24427 ASTERISK-29515 Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64	4 years ago
Naveen Albert	f7c4a3800c	app_sf: Add full tech-agnostic SF support ... 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	4 years ago
Steve Davies	a2ea233a6d	app_queue: Fix hint updates, allow dup. hints ... 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	4 years ago
Mark Petersen	92cb1c0a59	app_queue.c: added DIALEDPEERNUMBER on outgoing channel ... 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	4 years ago
Mark Petersen	4f06de7cf8	app_voicemail.c: Support for Danish syntax in VM ... 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	4 years ago
Naveen Albert	54761a41cd	app_sendtext: Add ReceiveText application ... 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	4 years ago
Naveen Albert	ee9eef492c	app_mf: Add full tech-agnostic MF support ... 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	4 years ago