618 Commits (a80179bfe7063d98913f7fded201f5c48c3e022c)

Author	SHA1	Message	Date
George Joseph	a80179bfe7	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml (cherry picked from commit 85a4ab8390)	3 months ago
Naveen Albert	1b31ced994	app_dial: Fix progress timeout calculation with no answer timeout. ... If to_answer is -1, simply comparing to see if the progress timeout is smaller than the answer timeout to prefer it will fail. Add an additional check that chooses the progress timeout if there is no answer timeout (or as before, if the progress timeout is smaller). Resolves: #821 (cherry picked from commit 6a07e8e0f1)	5 months ago
Naveen Albert	c8d1c29a52	app_dial: Fix progress timeout. ... Under some circumstances, the progress timeout feature added in commit 320c98eec8 does not work as expected, such as if there is no media flowing. Adjust the waitfor call to explicitly use the progress timeout if it would be reached sooner than the answer timeout to ensure we handle the timers properly. Resolves: #821 (cherry picked from commit 97dfe4cd40)	5 months ago
Naveen Albert	6e5a6c176d	app_dial: Add dial time for progress/ringing. ... Add a timeout option to control the amount of time to wait if no early media is received before giving up. This allows aborting early if the destination is not being responsive. Resolves: #588 UserNote: The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received. (cherry picked from commit bdf9327a33)	1 year ago
Maximilian Fridrich	b3cff31e1a	app_dial: Add option "j" to preserve initial stream topology of caller ... Resolves: #462 UserNote: The option "j" is now available for the Dial application which uses the initial stream topology of the caller to create the outgoing channels. (cherry picked from commit 366dc1e99f)	1 year ago
Naveen Albert	3b7a0d617f	app_dial: Fix infinite loop when sending digits. ... If the called party hangs up while digits are being sent, -1 is returned to indicate so, but app_dial was not checking the return value, resulting in the hangup being lost and looping forever until the caller manually hangs up the channel. We now abort if digit sending fails. ASTERISK-29428 #close Resolves: #281 (cherry picked from commit c032d38d3f)	2 years ago
Naveen Albert	95a41d231f	app_dial: Fix DTMF not relayed to caller on unanswered calls. ... DTMF frames are not handled in app_dial when sent towards the caller. This means that if DTMF is sent to the calling party and the call has not yet been answered, the DTMF is not audible. This is now fixed by relaying DTMF frames if only a single destination is being dialed. ASTERISK-29516 #close Change-Id: Iafd7430ac2915126d42dc48f0b73b262452ee027	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	2 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ... ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	2 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	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
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	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	3 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	3 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	3 years ago
Mark Petersen	a8b2692836	apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL_CLEARING ... changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING ASTERISK-28053 Reported by: roadkill Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89	3 years ago
Alexander Traud	196c24df22	apps: Fix for Doxygen. ... ASTERISK-29740 Change-Id: Icb6fbcfea0a5f1c82caa5001902b6a786adbf307	3 years ago
Josh Soref	eb03b18ff9	apps: Spelling fixes ... Correct typos of the following word families: simultaneously administrator directforward attachfmt dailplan automatically applicable nouns explicit outside sponsored attachment audio spied doesn't counting encoded implements recursively emailaddress arguments queuerules members priority output advanced silencethreshold brazilian debugging argument meadmin formatting integrated sneakiness ASTERISK-29714 Change-Id: Ie5ecaec91c00b26309da4e51cfc0991a5bb7d092	3 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
Sean Bright	8987de270f	app_dial.c: Only send DTMF on first progress event. ... ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	4 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
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
Richard Mudgett	abcb4ab321	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
Antoni Goldstein	8e21c25ce5	app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings ... Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled at the earliest received PROGRESS or RINGING. Added millisecond versions of DIALEDTIME and ANSWEREDTIME. Added millisecond versions of ast_channel_get_up_time and ast_channel_get_duration in channel.c. ASTERISK-28363 Change-Id: If95f1a7d8c4acbac740037de0c6e3109ff6620b1	6 years ago
Alexei Gradinari	4a567cee3a	app_dial/queue/followme: 'I' options to block initial updates in both directions ... The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates from the called parties to the caller. This patch also blocks updates in the other direction before call is answered. ASTERISK-27980 Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01	7 years ago
Richard Mudgett	9838a5e57a	app_dial/app_queue: Update application option documentation ... * Update the post-answer documentation and example. The Dial example was incorrect and misleading for the post-answer subroutine useage. * Fix note and warning paragraphs in option descriptions. They don't show up in the wiki. Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14	7 years ago
Rodrigo Ramírez Norambuena	1a3115d1c5	app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done ... Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8	7 years ago
Jenkins2	5843a19797	Merge "loader: Convert reload_classes to built-in modules."	7 years ago
Florian Floimair	ecc846b26b	app_dial: Enable early-media video ... Certain applications (e.g. door-phone) require that also video is transmitted before a call is accepted. Change-Id: I9842e1dc2f6e1c2c49dc33fe615255007d2f821e	7 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	7 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	7 years ago
Corey Farrell	955a891a84	app_macro deprecation. ... * Mark the module deprecated. * Disable the module by default. * Produce a warning the first time a macro is used. * Note deprecation related options in app_dial and app_queue. ASTERISK-27350 Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc	8 years ago
Joshua Colp	5a7af00e80	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	8 years ago
Joshua Colp	2b22c3c84b	channel: Add ability to request an outgoing channel with stream topology. ... This change extends the ast_request functionality by adding another function and callback to create an outgoing channel with a requested stream topology. Fallback is provided by either converting the requested stream topology into a format capabilities structure if the channel driver does not support streams or by converting the requested format capabilities into a stream topology if the channel driver does support streams. The Dial application has also been updated to request an outgoing channel with the stream topology of the calling channel. ASTERISK-26959 Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6	8 years ago
Joshua Colp	5d938045d4	channel: Remove old epoll support and fixed max number of file descriptors. ... This change removes the old epoll support which has not been used or maintained in quite some time. The fixed number of file descriptors on a channel has also been removed. File descriptors are now contained in a growable vector. This can be used like before by specifying a specific position to store a file descriptor at or using a new API call, ast_channel_fd_add, which adds a file descriptor to the channel and returns its position. Tests have been added which cover the growing behavior of the vector and the new API call. ASTERISK-26885 Change-Id: I1a754b506c009b83dfdeeb08c2d2815db30ef928	8 years ago
Richard Mudgett	16fdb11bc3	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	8 years ago
Joshua Colp	4de5454ef1	app_dial: Fix incorrect device state when channel is picked up. ... Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f	9 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
George Joseph	86e8716952	app_dial: Add the "Q" option to set the cause on unanswered channels ... The "Q" option will set the cause on the unanswered channels when another channel answers. It overrides the default of ANSWERED_ELSEWHERE. NOTE: chan_sip does not support setting the cause on a CANCEL to anything other than ANSWERED_ELSEWHERE. ASTERISK-26446 #close Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47	9 years ago
zuul	cc7e978149	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option"	9 years ago
Matt Jordan	730cb3b0b7	apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option ... In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d	9 years ago
Matt Jordan	6e1a3b924e	apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 ... If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358	9 years ago
Matt Jordan	9202ca34a8	app_dial: Improve documentation ... * Add some helpful <literal> and other embedded paragraph tags * Document some of the lesser known channel variables set by Dial * Add examples for some common Dial uses, along with some more challenging but useful options Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1	9 years ago
Joshua Colp	5c949d009e	Merge "Fixes to include signal.h"	9 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h ... POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc	9 years ago
Alexei Gradinari	3e8d523d88	core/dial: New channel variable FORWARDERNAME ... Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2	9 years ago
Mark Michelson	205a31f86c	Expand the scope of Dial Events ... Dial events up to this point have come in two flavors * A Dial event with no status to indicate that dialing has begun * A Dial event with a status to indicate that dialing has ended With this change, Dial events have been expanded to also give intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS". This is especially useful for ARI dialing, as it gives the application writer the opportunity to place a channel into an early bridge when early media is detected. AMI handles these in-progress dial events by sending a new event called "DialState" that simply indicates that dial state has changed but has not ended. ARI never distinguished between DialBegin and DialEnd, so no change was made to the event itself. Another change here relates to dial forwards. A forward-related event was previously only sent when a channel was successfully able to forward a call to a new channel. With this set of changes, if forwarding is blocked, we send a Dial event with a forwarding destination but no forwarding channel, since we were prevented from creating one. This is again useful for ARI since application writers can now handle call forward attempts from within their own application. ASTERISK-25925 #close Reported by Mark Michelson Change-Id: I42cbec7730d84640a434d143a0d172a740995543	9 years ago