5060 Commits (0c2bf1664c0ce47e68e09846fa7bddcfcfa91e22)

Author	SHA1	Message	Date
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	6 years ago
George Joseph	a4fd89536d	Merge "app_bridgeaddchan.c: Make BridgeAdd be more like Bridge"	6 years ago
George Joseph	6b7334a311	Merge "app_chanisavail.c: Simplify dialplan using ChanIsAvail."	6 years ago
Friendly Automation	5b815fe1ac	Merge "app_dial.c: Simplify dialplan using Dial."	6 years ago
Friendly Automation	5050c45e06	Merge "app_page.c: Simplify dialplan using Page."	6 years ago
Joshua Colp	b2664fd3a4	Merge "app_softhangup.c: Reduce unnecessary warning to verbose message."	6 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	6 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	6 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	6 years ago
Richard Mudgett	d86a6ac5ce	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	0d1f3d9bf3	app_chanspy.c: Reduce log message level from notice to verbose. ... Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a	6 years ago
Richard Mudgett	a457947198	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	d21427cadd	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR."	6 years ago
Frederic LE FOLL	a83625b366	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
Joshua C. Colp	89b7144fbd	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
Walter Doekes	0e750cdd10	app_queue: Fix old confusing comment about when the members are called ... ASTERISK-28644 Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b	6 years ago
George Joseph	6f82430b03	Merge "app_senddtmf: Add receive mode to AMI Action PlayDTMF"	6 years ago
Michael Cargile	5bda460300	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	772b59034f	app_senddtmf: Add receive mode to AMI Action PlayDTMF ... ASTERISK-28614 Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664	6 years ago
Kevin Harwell	bdd785d31c	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
George Joseph	a47cb71bb1	Build: Fix compile issues with seldom used modules ... The following modules needed tweaks for API changes. addons/cdr_mysql.c addons/chan_ooh323.c apps/app_meetme.c ASTERISK-28604 Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d	6 years ago
cmaj	2d67dbfef5	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	7362647e2f	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	863fe2225f	app_voicemail: Fix module unload leak. ... Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a	6 years ago
Frederic LE FOLL	2d0eee5418	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	6 years ago
Sean Bright	64906c4c9b	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	6 years ago
Alexei Gradinari	15624d9a7a	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	6 years ago
George Joseph	53c9e7962f	Merge "app_voicemail: Remove extra menuselect build options"	6 years ago
Sean Bright	9d07d5a6d6	app_voicemail: Remove extra menuselect build options ... You now select voicemail backends like normal dialplan applications, so there is no longer a need for their own menuselect category. Reported by snuff-work in #asterisk-dev Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005	6 years ago
Kevin Harwell	3656c42cb0	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	6 years ago
Kevin Harwell	c93c579190	app_voicemail: Remove dependency on the stasis cache ... app_voicemail utilized the stasis cache when polling mailboxes for MWI. This caused a memory leak (items were not being appropriately removed from the cache), and subsequent slowdown in system processing. This patch removes the stasis cache dependency, thus alleviating the memory leak. It does this by utilizing the new MWI API that better manages state lifetime. ASTERISK-28443 ASTERISK-27121 Change-Id: Ie89fedaca81ea1fd03d150d9d3a1ef3d53740e46	6 years ago
Chris-Savinovich	6b1f6ea2c4	app_voicemail.c: Build all three variants for app_voicemail at the same time ... Changes made to apps/Makefile to optionally build all three app_voicemail variations at the same time: 1) file (default), 2) odbc, and 3) imap. This functionality was requested by users. modules.conf.sample warns the user to make sure only one voicemail is loaded at a time. Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7	6 years ago
Kevin Harwell	cfdb567425	Merge "app_amd: issue with silence suppression fixed"	6 years ago
Nasir Iqbal	29bc7cf6b3	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	6 years ago
George Joseph	f3e5419d41	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	6 years ago
George Joseph	93ccff25c6	Merge "app_attended_transfer: new application AttendedTransfer"	6 years ago
Alexei Gradinari	3eaeb3e6c4	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(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	6 years ago
Alexei Gradinari	745cbab501	app_blind_transfer: new application BlindTransfer ... BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	6 years ago
Alexei Gradinari	408210bd4c	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	c5c953c1f1	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	80dba268ea	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	45a9ff8286	Merge "app_queue: Set correct value by default for shared_lastcall"	7 years ago
Friendly Automation	c2326155aa	Merge "mwi core: Move core MWI functionality into its own files"	7 years ago
agupta	7ce6d960d4	app_amd: Fix infinite loop on silent calls ... The total time logic will now be executed on calls which do not pass any media. ASTERISK-28143 Change-Id: I24726bd29d7e467fc721ca265363417234b22855	7 years ago
Rodrigo Ramírez Norambuena	ed615afb7e	app_queue: Set correct value by default for shared_lastcall ... There a long history here: In commit dd1e62c095 has introduce by default shared_lastcall = true by default but this now only happen is there not [general] directive in queues.conf After that, the commit 4b50e3f1ee fix the sample file. We'll need to keep the same setting if there a general or not section in configuration file since the shared_lastcall is by a long time in sample files as default value to 'no'. Change-Id: Id44faec370136df8d57902b453ad4059ed21b94c	7 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	7 years ago
Kevin Harwell	ff0d0ac23a	mwi core: Move core MWI functionality into its own files ... There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	7 years ago
Sean Bright	d58d7d4500	app_voicemail: Don't split mailbox options on comma ... Because the per-mailbox options are the last thing on a line, don't look for or stomp on any subsequent commas. ASTERISK-27935 #close Reported by: Sébastien Duthil Change-Id: I07b2eb4a33c303d0c7114d5b906f8c067c60a153	7 years ago
Sean Bright	63f86cac09	app_voicemail: Cleanup stale lock files on module load ... If Asterisk crashes while a VM directory is locked, lock files in the VM spool directory will not get properly cleaned up. We now clear them on module load. ASTERISK-20207 #close Reported by: Steven Wheeler Change-Id: If40ccd508e2f6e5ade94dde2f0bcef99056d0aaf	7 years ago
Sean Bright	e8cf3693f6	app_queue: Fix a few member pause bugs ... * Always set member->lastpause when setting member->paused * Fixed typo (using member->lastcall instead of member->lastpause) in 'queue show' output. * Use a constant 'now' in 'queue show' output for a better point-in-time view of time based stats. ASTERISK-27541 #close Reported by: César Benjamín García Martínez Change-Id: Ib41ced90cfdb66f9bb1e7b263d0f6fc1ac6e18fa	7 years ago