5065 Commits (268b0a308534eb0d5d66bcee9042cf7c8a545591)

Author	SHA1	Message	Date
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	6 years ago
Walter Doekes	b12ecbe27f	app_queue: Refactor odd placement of if's around say_position ... Change-Id: Icba97905e331812f129e5966e91a59b104c7a748	6 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	6 years ago
Joshua Colp	5166088628	Merge "app_voicemail, say: Fix various leading whitespace problems" into 16	6 years ago
Friendly Automation	f66885225a	Merge "app_voicemail: Prevent crash when saving message with realtime voicemail" into 16	6 years ago
Friendly Automation	23515dea5c	Merge "app_voicemail: Set globals to default values when voicemail.conf missing" into 16	6 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	6 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	6 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	6 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	6 years ago
Joshua Colp	6dc9239367	Merge "app_record: Do not hang up if beep audio is missing" into 16	6 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	6 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	6 years ago
Friendly Automation	7c227aa130	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail." into 16	6 years ago
Friendly Automation	077bb05528	Merge "app_dial.c: Simplify dialplan using Dial." into 16	6 years ago
Joshua Colp	5352022b2c	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message." into 16	6 years ago
Friendly Automation	ab042335f5	Merge "app_page.c: Simplify dialplan using Page." into 16	6 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	6 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	6 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	6 years ago
Richard Mudgett	00e745066c	app_chanspy.c: Reduce log message level from notice to verbose. ... Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a	6 years ago
Richard Mudgett	198f4cbdbf	app_softhangup.c: Reduce unnecessary warning to verbose message. ... Why log a warning for something your dialplan explicitly asked for? Change-Id: I167b90daf4c7d75dd4b7ef94849f6cef05aa43a7	6 years ago
Joshua C. Colp	3e0a2485c4	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR." into 16	6 years ago
Joshua C. Colp	5622df0a94	confbridge: Add support for specifying maximum sample rate. ... ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Frederic LE FOLL	aa06c6ea29	app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR. ... Temporary channel lifespan is very short and CDR deactivation request through ast_cdr_set_property() may happen when CDR is not available yet. Use CDR_PROP() dialplan function instead, it will first wait for pending CDR insertion requests to be processed. ASTERISK-28636 Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074	6 years ago
Walter Doekes	161e762742	app_queue: Fix old confusing comment about when the members are called ... ASTERISK-28644 Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b	6 years ago
Friendly Automation	17f52679b2	Merge "app_senddtmf: Add receive mode to AMI Action PlayDTMF" into 16	6 years ago
Michael Cargile	cf5b7f3a0b	app_amd: Fixed timeout issue ... ASTERISK_28143 attempted to fix an issue where calls with no audio would never timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its calculations. Unfortunately these frames seem to show up at irregular time intervals. This resulted in app_amd returning prematurely most of the time. * Removed AST_FRAME_NULL from the calculations * Added a check to see how much time has actually passed since app_amd began ASTERISK-28608 Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42	6 years ago
lvl	8894a56452	app_senddtmf: Add receive mode to AMI Action PlayDTMF ... ASTERISK-28614 Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664	6 years ago
Kevin Harwell	30c0af7257	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
cmaj	b41a8fd0f3	app_voicemail.c: Support multiple file formats for forwarded messages. ... If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav" and are using realtime ODBC backend, only the first format gets stored in the database. So when you forward a message later on, there is a bug generating the email, related to the stored format (GSM) being different than the desired email format (WAV) specified for the user. Sox can handle this, but Asterisk needs to tell sox exactly what to do. ASTERISK-22192 Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd	6 years ago
Sean Bright	68855f6a7b	Revert "app_voicemail: Cleanup stale lock files on module load" ... This reverts commit fd2e8d0da7. Reason for revert: Problematic for users who store their voicemail on network storage devices, or share voicemail storage between multiple Asterisk instances. ASTERISK-28567 #close Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41	6 years ago
Corey Farrell	256db889f2	app_voicemail: Fix module unload leak. ... Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a	6 years ago
Frederic LE FOLL	c8cf3ad389	ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf. ... ChanIsAvail() creates a temporary channel with ast_request() to test resource availability. It should not generate a CDR when it hangs up this temporary channel. This patch disables CDR generation for the temporary channel with ast_cdr_set_property(). ASTERISK-28527 Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1	7 years ago
Sean Bright	51fd43206b	audiohook.c: Substitute silence for unavailable audio frames ... There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e	7 years ago
Alexei Gradinari	ff180a5bfc	app_voicemail/IMAP: check mailstream not NULL in leave_voicemail ... The function leave_voicemail checks if expungeonhangup is set, but does not check if IMAP stream is closed, so it could call imap function with NULL stream. This leads to segfault. ASTERISK-28505 #close Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c	7 years ago
Kevin Harwell	6bb14150c4	various modules: json integer overflow ... There were still a few places in the code that could overflow when "packing" a json object with a value outside the base type integer's range. For instance: unsigned int value = INT_MAX + 1 ast_json_pack("{s: i}", value); would result in a negative number being "packed". In those situations this patch alters those values to a ast_json_int_t, which widens the value up to a long or long long. ASTERISK-28480 Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1	7 years ago
Kevin Harwell	d2a696cc04	Merge "app_amd: issue with silence suppression fixed" into 16	7 years ago
Nasir Iqbal	52a3d4a761	app_amd: issue with silence suppression fixed ... Now AMD algorithm will not ignore AST_FRAME_NULL, As I think using manual wait time instead of `framelength` is enough to fix timeout / TOOLONG issue. ASTERISK-28419 #close Change-Id: I16ea2d6295bc99b975e8c092e5f9fbd9214debdb	7 years ago
George Joseph	ccc92b6ecb	app_confbridge: Attended transfer event fixup ... When a channel already in a conference bridge is attended transfered to another extension, or when an existing call is attended transferred into a conference bridge, we now generate ConfbridgeJoin and ConfbridgeLeave events for the entering and departing channels. Change-Id: Id7709cfbceb26fbcb828b2d0d2a6b2fbeaf028e1	7 years ago
George Joseph	ca462f6e15	Merge "app_attended_transfer: new application AttendedTransfer" into 16	7 years ago
George Joseph	5c2bf122ed	Merge "app_blind_transfer: new application BlindTransfer" into 16	7 years ago
Alexei Gradinari	86cd77ec0a	app_attended_transfer: new application AttendedTransfer ... AttendedTransfer queues up attended transfer to the given extension. This application can be useful with Custom Dynamic Features. For example to make attended transfer to a predefined number. features.conf ;;; [applicationmap] my_atxfer => *7,self,GoSub,"my_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_atxfer TRANSFER_CONTEXT=my_transfer [my_atxfer] exten => s,1,AttendedTransfer(1234567890) same => n,Return() [my_transfer] include => default ;;; This application also can be used to completly redefine Attended transfer feature using dialplan. For example: features.conf ;;; [featuremap] atxfer => *7 [applicationmap] custom_atxfer => *2,self,GoSub,"custom_atxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_atxfer TRANSFER_CONTEXT=my_transfer [custom_atxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,AttendedTransfer(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	7 years ago
Alexei Gradinari	db5bc0fabf	app_blind_transfer: new application BlindTransfer ... BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	7 years ago
Alexei Gradinari	9516fb64c9	app_readexten: new option 'p' to stop reading on '#' key ... This patch adds the 'p' option. The extension entered will be considered complete when a # is entered. Change-Id: If77c40c9c8b525885730821e768f5dea71cf04c1	7 years ago
George Joseph	e7734476c6	Fixes for GCC 9 ... Various fixes for issues caught by gcc 9. Mostly snprintf trying to copy to a buffer potentially too small. ASTERISK-28412 Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e	7 years ago
Joshua Colp	8357ab7e9a	Merge "app_confbridge: Add "all" variants of REMB behavior." into 16	7 years ago
Joshua Colp	d861ebdca8	app_confbridge: Add "all" variants of REMB behavior. ... When producing a combined REMB value the normal behavior is to have a REMB value which is unique for each sender based on all of their receivers. This can result in one sender having low bitrate while all the rest are high. This change adds "all" variants which produces a bridge level REMB value instead. All REMB reports are combined together into a single REMB value that is the same for each sender. ASTERISK-28401 Change-Id: I883e6cc26003b497c8180b346111c79a131ba88c	7 years ago
Friendly Automation	f2cb892d7c	Merge "mwi core: Move core MWI functionality into its own files" into 16	7 years ago
Friendly Automation	a6863e5beb	Merge "app_amd: Fix infinite loop on silent calls" into 16	7 years ago