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1.4.36-rc1
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1.4.40-rc1
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1.4.42-rc1
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1.6.0
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1.6.0.11-rc1
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1.6.0.13-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
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${ noResults }
4704 Commits (56f24345f7f0ccf44d3edec69a07493503dec4fe)
Author | SHA1 | Message | Date |
---|---|---|---|
|
b5c471b339 |
followme: delete the right recorded name file
FollowMe with the option a records the name of the caller and plays it to the callee. However it has failed to clean up that recorded file as it tried to delete the file name without the '.sln' extension. ASTERISK-26008 #close Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com> |
9 years ago |
|
b4e10e7c90 |
Merge "app_confbridge: Add a regcontext option for confbridge bridge profiles."
|
9 years ago |
|
8923c9ac96 |
app_confbridge: Add a regcontext option for confbridge bridge profiles.
This patch allows for having app_confbridge register the name of the conference as an extension into a specific context, similar to regcontext for chan_sip. This variant is not quite as involved as the one in chan_sip and doesn't allow for multiple contexts or custom extensions, you can only specify the context and the conference name will always be used as the extension to register. ASTERISK-25989 #close Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f |
9 years ago |
|
122895f864 |
Merge "app_chanspy: fix audiohook options in non read-only mode"
|
9 years ago |
|
0c9faaee47 |
app_chanspy: fix audiohook options in non read-only mode
When option 'o' was not set, ChanSpy created its audiohook with the flag AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one direction only. ASTERISK-25866 #close Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0 |
9 years ago |
|
080c6216b6 |
app_voicemail: always copy dynamic struct to avoid race condition
Voicemail email addresses can be corrupt or voicemail emails can end up being sent to the wrong email address if asterisk is reading voicemail.conf during a reload and processing an email at the same time. This patch always copies the struct that would otherwise only be copied once. ASTERISK-24463 #close Reported by: John Campbell Tested by: Etienne Lessard Tested by: Andrew Nagy Change-Id: I3a0643813116da84e2617291903d0d489b7425fb |
9 years ago |
|
7281770710 |
app_chanspy: reduce audio loss on the spying channel.
ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when queues grow too large or when read and write queues go out of sync. Now these flags are set conditionally: - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not be set on the audiohook ASTERISK-25866 Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd |
9 years ago |
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8ae69cffef |
app_queue: Fix crash when unloading module.
When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b |
9 years ago |
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811e24f595 |
Merge "Bridge system: Fix memory leaks and double frees on impart failure."
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9 years ago |
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456600a641 |
Merge "app_queue: queue members can receive multiple calls"
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9 years ago |
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f99ec857c8 |
Fix case sensitive actions in AMI QueueSummary and QueueStatus
ASTERISK-25954 #close
Reported by: Javier Acosta
Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256
(cherry picked from commit
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9 years ago |
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30ab21d5fa |
app_queue: queue members can receive multiple calls
It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48 |
9 years ago |
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a63656b419 |
Bridge system: Fix memory leaks and double frees on impart failure.
You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 |
9 years ago |
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d95512a7dd |
app_talkdetect: Make the module core supported.
This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88 |
9 years ago |
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dbb47e0a47 |
app_queue: Frequent segfaults in function can_ring_entry()
ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117 |
9 years ago |
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f7a434fa6d |
Merge "app_voicemail/IMAP: function 'save_to_folder' creates wrong folder"
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9 years ago |
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5a0534dc62 |
app_voicemail: Fix test_voicemail_notify_endl test.
The test_voicemail_notify_endl test checks the end-of-line characters of an email message to confirm that they are consistent. The test wrongfully assumed that reading from the email message into a buffer will always result in more than 1 character being read. This is incorrect. If only 1 character was read the test would go outside of the buffer and access other memory causing a crash. The test now checks to ensure that 2 or more characters are read in ensuring the test stays within the buffer. ASTERISK-25874 #close Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710 |
9 years ago |
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c00c298a0e |
app_voicemail/IMAP: function 'save_to_folder' creates wrong folder
If try to move message to Cust1 (number 5) the function 'save_to_folder' tries to create Greeting folder instead of Cust1. This patch fixed it by setting GREETINGS_FOLDER = -1 ASTERISK-24927 #close Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51 |
9 years ago |
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c3339816f1 |
Merge "app_voicemail/IMAP: IMAP access FATAL error: Out of memory"
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9 years ago |
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f9dab80816 |
app_voicemail/IMAP: IMAP access FATAL error: Out of memory
Sometimes uw-imap function 'mail_fetchbody' returns huge len which then pass to uw-imap function 'rfc822_base64'. uw-imap tries to allocate huge memory and abort() on fail. This patch check the len. If the len more than max size (128 Mbytes) log error. This patch also set variables len, newlen to avoid uninizialezed len. This patch also check pointer returned by rfc822_base64. ASTERISK-25899 #close Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca |
9 years ago |
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6138a75e8e |
pbx.h: Make ast_state_cb_type take more const.
This eliminates some casts that I made a note saying v10 and above would no longer need them. Better late than never :) Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572 |
9 years ago |
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9785e8d090 |
app_echo: forward and generate VIDUPDATE frames
When using app_echo via WebRTC with VP8 video the video would appear only after a few minutes, because there would be nothing to request a full reference frame. This fixes the problem in both ways: - echos any VIDUPDATE frames received on the channel - sends one such frame when first video frame is to be forwarded This makes the echo work with Firefox and Chrome WebRTC implementation. ASTERISK-25867 #close Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e |
9 years ago |
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894071ea2c |
config: fix flags in uint option handler
The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e |
9 years ago |
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7964e260d3 |
app_stasis: Don't hang up if app is not registered
This prevents pbx_core from hanging up the channel if the app isn't registered. ASTERISK-25846 #close Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce |
9 years ago |
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dcb25bb057 |
app_chanspy: Fix occasional deadlock with ChanSpy and Local channels.
Channel masquerading had a conflict with autochannel locking. When locking autochannel->channel, the channel is fetched from the autochannel and then locked. During the fetch, the autochannel -- which has no locks itself -- can be modified by someone who owns the channel lock. That means that the value of autochan->channel cannot be trusted until you hold the lock. In practice, this caused problems with Local channels getting masqueraded away while the ChanSpy attempted to get info from that channel. The old channel which was about to get removed got locked, but the new (replaced) channel got unlocked (no-op). Because the replaced channel was now locked (and would never get unlocked), it couldn't get removed from the channel list in a timely manner, and would now cause deadlocks when iterating over the channel list. This change checks the autochannel after locking the channel for changes to the autochannel. If the channel had been changed, the lock is reobtained on the new channel. In theory it seems possible that after this fix, the lock attempt on the old (wrong) channel can be on an already destroyed lock, maybe causing a crash. But that hasn't been observed in the wild and is harder induce than the current deadlock. Thanks go to Filip Frank for suggesting a fix similar to this and especially to IRC user hexanol for pointing out why this deadlock was possible and testing this fix. And to Richard for catching my rookie while loop mistake ;) ASTERISK-25321 #close Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def |
9 years ago |
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15aeb78c66 |
app_queue: fix Calculate talktime when is first call answered
Fix calculate of average time for talktime is wrong when is completed the first call beacuse the time for talked would be that call. ASTERISK-25800 #close Change-Id: I94f79028935913cd9174b090b52bb300b91b9492 |
9 years ago |
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1bc54aee80 |
app_confbridge: Only use b_profile options from the conference.
A user cannot set new bridge options after the conference is created by the first user. Attempting to do so is documented as undefined behavior. This patch ensures that the bridge profile options used are from the conference and not what a subsequent user may have tried to set. Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266 |
9 years ago |
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0aea293450 |
Merge "app_confbridge.c: Replace inlined code with existing function."
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9 years ago |
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e541d9cf34 |
Merge topic 'ASTERISK-20987'
* changes: app_confbridge: Add ability to get the muted conference state. app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. app_confbridge: Make non-admin users join a muted conference muted. |
9 years ago |
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af6b15976d |
app_confbridge.c: Replace inlined code with existing function.
Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51 |
10 years ago |
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7932336a3d |
app_confbridge: Add ability to get the muted conference state.
* Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. * Added Muted header to AMI ConfbridgeListRooms action response list events to indicate the muted conference state. * Added Muted column to CLI "confbridge list" output to indicate the muted conference state and made the locked column a yes/no value instead of a locked/unlocked value. ASTERISK-20987 Reported by: hristo Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1 |
10 years ago |
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894045e7cf |
app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7 |
10 years ago |
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12c93e8f81 |
app_confbridge: Make non-admin users join a muted conference muted.
ASTERISK-20987 #close Reported by: hristo Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38 |
10 years ago |
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f299dc0d76 |
app_queue: Add Lastpause field of queue member
Add time when started a the last pause for a queue member for QueueMemberStatus ami event. Also show accumulate time in seconds when started a pause for a queue member to CLI command 'queue show'. ASTERISK-16394 #close Change-Id: I4b12aa3b2efa8d02939db3e13712510b4879865c |
10 years ago |
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ef293354dc |
Merge "app_queue: fix some tab format"
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9 years ago |
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fe059b3534 |
Merge "app_queue: Fix preserved reason of pause when Asterisk is restared"
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9 years ago |
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527057ae24 |
Merge "app_queue.c: remove include for core_unreal.h not used in code."
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9 years ago |
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40da6434c1 |
build_system: Fix some warnings highlighted by clang
Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd |
9 years ago |
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8c664da0ff |
app_queue: fix some tab format
Change-Id: I2734392b131f1fb0949515d538f83f30fbc15d8c |
10 years ago |
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d3969d09ae |
app_queue.c: remove include for core_unreal.h not used in code.
Change-Id: Idc2ae8a6bd869a66544916906744a5678622262d |
10 years ago |
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378fed4900 |
app_queue: Fix preserved reason of pause when Asterisk is restared
When the Asterisk is restared is not preseved reason paused of members. This patch fixed this cases, retain data on astdb and set when Asterisk is started. ASTERISK-25732 #close Report by: Rodrigo Ramírez Norambuena Change-Id: Id3fb744c579e006d27cda4a02334ac0e4bed9eb5 |
10 years ago |
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90b06d1a3c |
app_queue: Add member flag "in_call" to prevent reading wrong lastcall time
Member lastcall time is updated later than member status. There was chance to check wrapuptime for available member with wrong (old) lastcall time. New boolean flag "in_call" is set to true right before connecting call, and reset to false after update of lastcall time. Members with "in_call" set to true are treat as unavailable. ASTERISK-19820 #close Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500 |
10 years ago |
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0c62e0a55c |
Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime"
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10 years ago |
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6d18fe151c |
voicemail: Move app_voicemail / res_mwi_external conflict to runtime
The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247 |
10 years ago |
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f88b952093 |
app_dial: Immediately exit dial if the caller is already hung up.
If a caller hangs up before dial is executed within an AGI then the AGI has likely eaten all queued frames before executing the dial in DeadAGI mode. With the caller hung up and no pending frames from the caller's read queue, dial would not know that the call has hung up until a called channel answers. It is rather annoying to whoever just answered the non-existent call. Dial should not continue execution in DeadAGI mode, hangup handlers, or the h exten. * Added a check early in dial to abort dialing if the caller has hungup. ASTERISK-25307 #close Reported by: David Cunningham Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418 |
10 years ago |
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ca394161cf |
app_amd: Correct maximum_number_of_words functionality & documentation
- The maximum_number_of_words was previously documented as being the number of words that when exceeded, would result in the AMD application returning that the audio represents a machine. This was inconsistent with its actual functionality - it was a number of words that when REACHED, would result in determination as a machine. This update corrects the functionality to match the previously documented functionality. This is a backwards incompatible change in configuration file, and has been added to UPGRADE.txt as a result. The sample configuration file and application defaults have been updated so that the default value is now 2, which reflects the same default functionality as previous versions. - Update documentation for silence_threshold, which previously implied that it was measuring time, rather than noise averages in the sample. - Update the comments in amd.conf.sample. ASTERISK-25639 #close Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093 |
10 years ago |
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3e7522533c |
app_queue: update RT members when the 1st call joins a queue with no agents
If a call enters on a queue and the members on that queue are updated in realtime (ex: using mysql inserting a new agent) the queue members are never refreshed and the call will stay in the queue until other event occurs. This happens only if this is the first call of the queue and there is no agents servicing. This patch prevent this issue, ensuring realtime members are updated if there is one call in the queue and no available agents ASTERISK-25442 #close Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682 |
10 years ago |
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fcaebb0e43 |
app_meetme: Set default value for audio_buffers.
The default value was never set for audio_buffers, causing bad audio quality. This ensures the default is always set. ASTERISK-25569 #close Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44 |
10 years ago |
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f2a84b500d |
app_queue: Show reason of pause on CLI
Add value of pause reason when is paused on CLI command "queue show" ASTERISK-25581 #close Report by: Rodrigo Ramírez Norambuena Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864 |
10 years ago |
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b52b494004 |
Merge "StatsD: Add sample rate compatibility"
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10 years ago |