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5045 Commits (895a66f77278f85ba406b6870a64a09ec92c23e7)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e1eb5e8dc2 |
app_queue: Fix old confusing comment about when the members are called
ASTERISK-28644 Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b |
6 years ago |
|
715e4d0e72 |
Merge "app_senddtmf: Add receive mode to AMI Action PlayDTMF" into 17
|
6 years ago |
|
e23b2856d0 |
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 |
|
6345a00228 |
app_senddtmf: Add receive mode to AMI Action PlayDTMF
ASTERISK-28614 Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664 |
6 years ago |
|
8bc6fa0fbd |
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 |
|
36b28c98dd |
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 |
|
aa0973f868 |
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 |
|
b903994987 |
Revert "app_voicemail: Cleanup stale lock files on module load"
This reverts commit
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6 years ago |
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78f329dad9 |
app_voicemail: Fix module unload leak.
Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a |
6 years ago |
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50997de887 |
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 |
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e123d1ccc5 |
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 |
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052ab9d966 |
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 |
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90a14a0a90 |
Merge "app_voicemail: Remove extra menuselect build options" into 17
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6 years ago |
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87f8ca2812 |
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 |
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a305f2fdcb |
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 |
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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 |
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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 |
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cfdb567425 |
Merge "app_amd: issue with silence suppression fixed"
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6 years ago |
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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 |
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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 |
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93ccff25c6 |
Merge "app_attended_transfer: new application AttendedTransfer"
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6 years ago |
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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(${dest}) same => n,Return() [my_transfer] include => default ;;; Change-Id: Ie5cfa455d0813cffd5c85a6fb117f07d8f0b903b |
6 years ago |
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745cbab501 |
app_blind_transfer: new application BlindTransfer
BlindTransfer redirects all channels currently bridged to the caller channel to the specified destination. This application can be useful with Custom Dynamic Features. For example to make blind transfer to a predefined number. features.conf ;;; [applicationmap] my_blindxfer => *6,self,GoSub,"my_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=my_blindxfer [my_blindxfer] exten => s,1,BlindTransfer(1234567890,default) same => n,Return() ;;; This application also can be used to completly redefine Blind transfer feature using dialplan. For example: features.conf ;;; [featuremap] blindxfer => [applicationmap] custom_blindxfer => ##,self,GoSub,"custom_blindxfer,s,1",default ;;; extensions.conf ;;; [globals] DYNAMIC_FEATURES=custom_blindxfer [custom_blindxfer] exten => s,1, same => n,Playback(pbx-transfer) same => n,Read(dest,dial,10,i,3,3) same => n,BlindTransfer(${dest},default) same => n,Return() ;;; Change-Id: I9d55e7f69ccfd4472dec00d62771d6de8803215a |
6 years ago |
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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 |
6 years ago |
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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 |
6 years ago |
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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 |
6 years ago |
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45a9ff8286 |
Merge "app_queue: Set correct value by default for shared_lastcall"
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6 years ago |
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c2326155aa |
Merge "mwi core: Move core MWI functionality into its own files"
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6 years ago |
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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 |
6 years ago |
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ed615afb7e |
app_queue: Set correct value by default for shared_lastcall
There a long history here: In commit |
6 years ago |
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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 |
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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 |
6 years ago |
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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 |
6 years ago |
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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 |
6 years ago |
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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 |
6 years ago |
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834d022da5 |
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 |
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a145f83d30 |
Merge "stasis: Improve topic/subscription names and statistics."
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6 years ago |
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48e407e506 |
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 |
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0231dd6ae7 |
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 |
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57850c7861 |
app_meetme: Don't mute joining admins if conference is muted
ASTERISK-28328 #close Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b |
6 years ago |
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f6b5b7208c |
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 |
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ce0523a57e |
app_queue: Enable set the wrapuptime from AddQueueMember application
This change add ability to set the wrapuptime per-member using the AddQueueMember application. The feature to set wrapuptime per member was include in the issue ASTERISK-27483 for static member by configuration file and was not added to set from AddQueueMember. ASTERISK-28055 #close Change-Id: I7c7ee4a6f804922cd7c42cb02eea26eb3806c6cf |
6 years ago |
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54a912b26d |
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 |
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c6980e32ae |
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 |
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b4523ef334 |
Merge "app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail"
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6 years ago |
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2c48b5d9bf |
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 |
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b7b080a0aa |
app_queue: Fix crash when using 'b' option on non-ringall queue.
When using the 'b' option to Queue with a queue that was not configured for ring all a crash would occur as the wrong pointer would be used. ASTERISK-28218 Change-Id: If1390f64e321047dff24fd2410c95dde74904980 |
6 years ago |
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c23c8d92d5 |
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 |
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cb1a08bdcb |
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 |
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140702ba2d |
app_queue: Revert broken queue channel reference patch
Revert commit
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7 years ago |