4905 Commits (5a3826e334be96812e322f3c2ecd18ba00ec8b9d)

Author	SHA1	Message	Date
Friendly Automation	1254747497	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR." into 13	6 years ago
Friendly Automation	66970af0d3	Merge "confbridge: Add support for specifying maximum sample rate." into 13	6 years ago
Frederic LE FOLL	b07bccd0bd	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	ca35de4282	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
Stanislav	22f9c57f43	app_voicemail: warning when is compiling ... Change-Id: Ib53eba1a66e25fbeba61c620bd3edd462f699ada ASTERISK-28628 Change-Id: Ib53eba1a66e25fbeba61c620bd3edd462f699ada	6 years ago
Walter Doekes	d8063c1cf9	app_queue: Fix old confusing comment about when the members are called ... ASTERISK-28644 Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b	6 years ago
George Joseph	bb732e4292	Merge "app_senddtmf: Add receive mode to AMI Action PlayDTMF" into 13	6 years ago
Friendly Automation	7bcfcd3433	Merge "app_amd: Fixed timeout issue" into 13	6 years ago
lvl	17c393377e	app_senddtmf: Add receive mode to AMI Action PlayDTMF ... ASTERISK-28614 Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664	6 years ago
Michael Cargile	9361e32bed	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
Kevin Harwell	6ca76798e1	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	c8e38f8550	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	0834e06673	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	25918f2837	app_voicemail: Fix module unload leak. ... Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a	6 years ago
Frederic LE FOLL	9462722b17	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	ddc64ca059	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	16bc4ed93c	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
Kevin Harwell	f145b58542	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	a302d46dd8	Merge "app_amd: issue with silence suppression fixed" into 13	6 years ago
George Joseph	f22dedc597	Merge "app_confbridge: Attended transfer event fixup" into 13	6 years ago
George Joseph	41f5d15763	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	83c353c650	Merge "app_attended_transfer: new application AttendedTransfer" into 13	6 years ago
Alexei Gradinari	45a9ee4c53	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(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b	6 years ago
Alexei Gradinari	dd12e1cbd3	app_blind_transfer: new application BlindTransfer ... BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a	6 years ago
Nasir Iqbal	f7527921b6	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
Alexei Gradinari	6ded762dbf	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	6 years ago
George Joseph	4337895aee	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	6 years ago
George Joseph	6edef49525	Merge "mwi core: Move core MWI functionality into its own files" into 13	6 years ago
agupta	188b1d3e68	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	6 years ago
Antoni Goldstein	001e7762e4	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
Kevin Harwell	4ea20c9c85	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	6 years ago
Sean Bright	d2f519dcfa	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	6 years ago
Sean Bright	fd2e8d0da7	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	6 years ago
Sean Bright	4a6bd68869	app_queue: Fix documentation for QUEUE_MEMBER function. ... It was a copy/paste of the QUEUE_MEMBER_COUNT function's synopsis. ASTERISK-20986 #close Reported by: Olivier Krief Change-Id: If51ec481feb35824a4e78ab5600b197b819b10be	6 years ago
Joshua C. Colp	c7d58aec7a	Merge "stasis: Improve topic/subscription names and statistics." into 13	6 years ago
Dömsödi Gergely	92d1889813	app_queue: fix ring_entry to access nativeformats with a channel lock ... Fixes an intermittent segmentation fault which occured when accessing nativeformats of a channel which entered into a queue. ASTERISK-27964 Reported by: Francisco Seratti Change-Id: Ic87fa7a363f3b487c24ce07032f4b2201c22db9e	6 years ago
Joshua Colp	07b3253155	stasis: Improve topic/subscription names and statistics. ... Topic names now follow: <subsystem>:<functionality>[/<object>] This ensures that they are all unique, and also provides better insight in to what each topic is for. Subscriber ids now also use the main topic name they are subscribed to and an incrementing integer as their identifier to make it easier to understand what the subscription is primarily responsible for. Both the CLI commands for listing topic and subscription statistics now sort to make it a bit easier to see what is going on. Subscriptions will now show all topics that they are receiving messages from, not just the main topic they were subscribed to. ASTERISK-28335 Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d	6 years ago
Sean Bright	b4bba8f217	app_meetme: Don't mute joining admins if conference is muted ... ASTERISK-28328 #close Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b	6 years ago
Sean Bright	cb6a976656	app_queue: Handle empty 'interface' in queue member config ... While the 'interface' column is a NOT NULL, the empty string is still allowed. res_config_odbc treats the empty string as a NULL and we crash when trying to dereference. Also cleaned up an adjacent error message for consistency. ASTERISK-28168 #close Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202	6 years ago
Joshua Colp	a4d930c2ed	res_odbc: Add basic query logging. ... When Asterisk is connected and used with a database the response time of the database can cause problems in Asterisk if it is long. Normally the only way to see this problem would be to retrieve a backtrace from Asterisk and examine where things are blocked, or examine the database to see if there is any indication of a problem. This change adds some basic query logging to make it easier to investigate such a problem. When logging is enabled res_odbc will now keep track of the number of queries executed, as well as the query that has taken the longest time to execute. There is also an option which will cause a WARNING message to be output if a query takes longer than a configurable amount of time to execute. This makes it easier and clearer for users that their database may be experiencing a problem that could impact Asterisk. ASTERISK-28277 Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6	6 years ago
Friendly Automation	9514829403	Merge "app_voicemail: Add Mailbox Aliases" into 13	6 years ago
George Joseph	d0eccac3ff	app_voicemail: Add Mailbox Aliases ... You can now define an "aliases" context in voicemail.conf whose entries point to actual mailboxes. These can be used anywhere the mailbox is specified. Example: [general] aliasescontext = myaliases [default] 1234 = yadayada [myaliases] 4321@devices = 1234@default Now you can use 4321@devices to refer to the 1234@default mailbox. This can be useful to provide channel drivers with constant mailbox specifications such as <extension>@devices leaving app_voicemail to control exactly which mailbox the alias points to. Now, only voicemail has to be reloaded to make changes instead of individual channel drivers which are usually more expensive to reload. Change-Id: I395b9205c91523a334fe971be0d1de4522067b04	6 years ago
Bryan Boatright	250812df09	app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail ... If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is not updated correctly since urgent messages are in a different directory. The fix is to update the channel variable when the path to the urgent message is created. ASTERISK-28225 Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca	6 years ago
George Joseph	e137086163	app_voicemail: Don't delete mailbox state unless mailbox is deleted ... The free_user function was automatically deleting the stasis mailbox state but this only makes sense when the mailbox is actually deleted, not just the structure freed. This was causing issues where leave_voicemail would publish the mwi message to stasis and delete the state before the message could be processed by res_pjsip_mwi. * Removed the delete of state from free_user(). * Created a new free_user_final() function that both frees the data structure and deletes the state. This function is only called during module load/unload where it's appropriate to delete the state. ASTERISK-28215 Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd	7 years ago
Alexei Gradinari	6392e452f2	confbridge: announce to the marked users when they join an empty conference ... Currently the file sound_only_person is not played when a marked user (with announce_only_user=yes) joins an empty conference. This patch fixes it. ASTERISK-28201 #close Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4	7 years ago
lvl	91630834f7	app_queue: Revert broken queue channel reference patch ... Revert commit 6409e7b11a, and add NULL checks for all app_queue event handling code. Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844 ASTERISK-28125 Change-Id: I37334ea184ebb56e54471496b82937d4927815a0	7 years ago
George Joseph	363d19326b	Revert "app_voicemail: Remove need to subscribe to stasis" ... This reverts commit 1843b0e2b5. That commit closed a long standing hole which allowed subscriptions to mailboxes that weren't configured in voicemail.conf. This caused an issue with FreePBX which depdended on that behavior. The commit is being reverted until FreePBX can handle the new behavior. ASTERISK-28151 Reported by: Ronald Raikes Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15	7 years ago
Corey Farrell	0a9904e1c6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	d748ed4147	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
Joshua Colp	dc52719cb9	Merge "app_dial/queue/followme: 'I' options to block initial updates in both directions" into 13	7 years ago