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1.4.27-rc3
1.4.27-rc4
1.4.27-rc5
1.4.27.1
1.4.28
1.4.28-rc1
1.4.29
1.4.29-rc1
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1.4.3
1.4.30
1.4.30-rc1
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1.4.31
1.4.31-rc1
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1.4.40
1.4.40-rc1
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1.4.41-rc1
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1.4.42
1.4.42-rc1
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1.4.6
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1.4.7.1
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1.4.9
1.6.0
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1.6.0.11-rc1
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1.6.0.16-rc1
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1.6.0.18
1.6.0.18-rc1
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1.6.0.19
1.6.0.2
1.6.0.20
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1.6.0.21
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1.6.0.23
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1.6.0.24
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1.6.0.26
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1.6.0.27-rc1
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10.0.0
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10.10.0
10.10.0-digiumphones
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10.11.1-digiumphones
10.12.0
10.12.0-digiumphones
10.12.0-digiumphones-rc1
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10.12.1
10.12.1-digiumphones
10.12.2
10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
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10.2.1
10.3.0
10.3.0-rc1
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10.3.1
10.4.0
10.4.0-digiumphones-rc1
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10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
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10.5.0-rc1
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10.5.1-digiumphones
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10.6.0
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10.7.0
10.7.0-digiumphones
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10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
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10.9.0
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11.0.0
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11.1.1
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11.10.0
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11.16.0
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11.2.1
11.2.2
11.20.0-rc1
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11.21.0
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12.0.0
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12.6.0
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certified/1.8.11-cert1
certified/1.8.11-cert10
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certified/1.8.11-cert3-rc1
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certified/1.8.11-cert6
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certified/1.8.15-cert1
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certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.6-cert1
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certified/11.6-cert6
certified/11.6-cert7
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certified/13.1-cert1
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certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
4879 Commits (13.27)
Author | SHA1 | Message | Date |
---|---|---|---|
|
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 |
|
6edef49525 |
Merge "mwi core: Move core MWI functionality into its own files" into 13
|
6 years ago |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
c7d58aec7a |
Merge "stasis: Improve topic/subscription names and statistics." into 13
|
6 years ago |
|
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 |
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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 |
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b4bba8f217 |
app_meetme: Don't mute joining admins if conference is muted
ASTERISK-28328 #close Change-Id: I4f6069fb34923b7521520c2a205a1e56227e592b |
6 years ago |
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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 |
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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 |
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9514829403 |
Merge "app_voicemail: Add Mailbox Aliases" into 13
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6 years ago |
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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 |
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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 |
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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 |
6 years ago |
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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 |
6 years ago |
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91630834f7 |
app_queue: Revert broken queue channel reference patch
Revert commit
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7 years ago |
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363d19326b |
Revert "app_voicemail: Remove need to subscribe to stasis"
This reverts commit
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7 years ago |
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0a9904e1c6
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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 |
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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 |
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dc52719cb9 |
Merge "app_dial/queue/followme: 'I' options to block initial updates in both directions" into 13
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7 years ago |
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46e3dbf1fd |
Merge "app_dial/app_queue: Update application option documentation" into 13
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7 years ago |
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0e7b91a22b |
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 |
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badf0adf86 |
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 |
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01716f70e6 |
taskprocessor: Warn on unused result from pushing task.
Add attribute_warn_unused_result to ast_taskprocessor_push, ast_taskprocessor_push_local and ast_threadpool_push. This will help ensure we perform the necessary cleanup upon failure. Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d |
7 years ago |
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ec52409a53 |
app_queue.c: Fix json ref leak
Declining the queue_member_status_type stasis message in stasis.conf causes these messages to leak json objects. * Add missing ast_json_unref() if the type is NULL in queue_publish_member_blob(). ASTERISK-28084 Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc |
7 years ago |
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74c5c1cd1b |
app_queue: Fix Attended transfer hangup with removing pending member.
This issue related to setting of holdtime, announcements, member delays. It works well if we set the member delays to "0" and no announcements and no holdtime.This issue will happen if we set member delays to "1", "2"... or announcements or holdtime and hangs up the call during processing it. And here is the reason: (At the step of answering a phone.) It takes care any holdtime, announcements, member delays, or other options after a call has been answered if it exists. Normally, After the call has been aswered, and we wait for the processing one of the cases of the member delays or hold time or announcements finished, "if (ast_check_hangup(peer))" will be not executed, then queue will be updated at update_queue(). Here, pending member will be removed. However, after the call has been aswered, if we hangs up the call during one of the cases of the member delays or hold time or announcements, "if (ast_check_hangup(peer))" will be executed. outgoing = NULL and at hangupcalls, pending members will not be removed. * This fixed patch will remove the pending member from container before hanging up the call with outgoing is NULL. ASTERISK-27920 Reported by: Cao Minh Hiep Tested by: Cao Minh Hiep Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855 |
7 years ago |
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594bbbea57 |
Merge "app_voicemail: Fix stack overrun in append_mailbox" into 13
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7 years ago |
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656b3e85cf |
app_voicemail: Fix stack overrun in append_mailbox
The append_mailbox function wasn't calculating the correct length to pass to ast_alloca and it wasn't handling the case where context might be empty. Found by the Address Sanitizer. Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161 |
7 years ago |
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7bdf1d3c67 |
app_voicemail: Cleanup mailbox topic and cache
app_voicemail wasn't properly cleaning up the stasis cache or the mwi topic pool when the module was unloaded or when a user was deleted as a result of a reload. This resulted in leaks in both areas. * app_voicemail now calls ast_delete_mwi_state_full when it frees a user structure and ast_delete_mwi_state_full in turn now calls the new stasis_topic_pool_delete_topic function to clear the topic from the pool. Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8 |
7 years ago |
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1843b0e2b5 |
app_voicemail: Remove need to subscribe to stasis
app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee |
7 years ago |
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d226458c5b |
optional_api: Remove unused nonoptreq fields
As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc |
7 years ago |
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eda1af091e |
app_queue: Update realtime queuemembers after wait_a_bit(), not before
This ensures the most up-to-date information is used for the next call attempt. ASTERISK-28032 Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce |
7 years ago |
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17040d1ce3 |
app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done
Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8 |
7 years ago |
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b9d9c0a8b9 |
app_queue: Silence GCC 8 compiler warning
I'm only seeing an error in 14+, so I assume it is due to different compiler options: app_queue.c: In function ‘handle_queue_add_member’: app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11 bytes into a region of size 3 [-Werror=format-overflow=] sprintf(num, "%d", state); ^~ app_queue.c:10234:18: note: directive argument in the range [-2147483648, 99] sprintf(num, "%d", state); ^~~~ Compiler: gcc version 8.0.1 20180414 (experimental) [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10 |
7 years ago |
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f48761907a |
app_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE
When a call leaves a queue on leaveempty condition, QUEUESTATUS must be set to LEAVEEMPTY, no matter whether Queue was executed with or without the "c" (continue) option. The regression was introduced in the fix for ASTERISK_25665. The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was overwritten in case when "c" is set, regardless of what was the cause for leaving the queue. ASTERISK-27973 #close Reported-by: Valentin Safonov Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c |
7 years ago |
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447ec4e472 |
app_confbridge: Bridge and announcers not removed if conference ends quickly
If a conference is ended very quickly after it was created (i.e., the first user immediately hangs up) then the conference bridge and announcer channels are not removed. When a conference is created, the push_announcer() function is added to the playback queue task processor and the conference object reference is bumped. If a conference is ended while the push_announcer() function is still going then the ao2_cleanup(conference) at the end of push_announcer() will call the destructor function - destroy_conference_bridge(). The destroy_conference_bridge() function will then add the hangup_playback() task to the playback queue and will wait for it to end. Since it is already a current task of the playback queue it will wait forever. This patch makes the conference thread call push_announcer() directly. This way the conference object reference bump is not needed. Since the playback queue task processor is only used by the conference thread itself, there is no danger of trying to play announcements before the announcer is pushed to the bridge. ASTERISK-27870 #close Change-Id: I947a50fb121422d90fd1816d643a54d75185a477 |
7 years ago |
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6a1626c265 |
AMI PlayDTMF Action: Make not compete with channel's media thread.
There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905 |
7 years ago |
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bb0ce22b2b |
app_mp3: remove 10 seconds of silence after mp3 playback
This patch changes the way asterisk polls output from mpg123, instead of waiting for 10 seconds(when playing an http url) it now uses a timeout of one second and iterates 10 times using this same timeout. The main difference is that for every timeout asterisk receives it now checks if mpg123 is still running before poll again. ASTERISK-27752 Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620 |
7 years ago |
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499867d006 |
Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response" into 13
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7 years ago |
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f17d09ae63 |
Merge "app_meetme: Fix manager event documentation for several events." into 13
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7 years ago |
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db2413b446 |
app_sendtext: Allow content types other than text/plain
There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9 |
7 years ago |
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9ff4779f03 |
app_confbridge: Add talking indicator for ConfBridgeList AMI response
When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6 |
7 years ago |
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071232244a |
app_meetme: Fix manager event documentation for several events.
The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and MeetmeTalkRequest AMI events were documented with sending out a Usernum header when the User header was actually output. * Change the online documentation to match reality. ASTERISK-27873 ASTERISK-25261 Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39 |
7 years ago |
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6dbecc2319 |
Merge "app_voicemail: Fix data-type mismatch between app_voicemail and database" into 13
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7 years ago |
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835cbbe38c |
Merge "app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail" into 13
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7 years ago |
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436d17fa50 |
app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail
Correct the log warning message shown when ODBC voicemail retrieve_file is called and there is a null value in the category column. A more meaningfull message is now written at debug level. ASTERISK-27853 Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4 |
7 years ago |