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1.4.26-rc3
1.4.26-rc4
1.4.26-rc5
1.4.26-rc6
1.4.26.1
1.4.26.2
1.4.26.3
1.4.27
1.4.27-rc1
1.4.27-rc2
1.4.27-rc3
1.4.27-rc4
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1.4.28
1.4.28-rc1
1.4.29
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1.4.30
1.4.30-rc1
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1.4.31-rc1
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1.4.32-rc1
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1.4.33-rc1
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1.4.34-rc1
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1.4.35
1.4.35-rc1
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1.4.36-rc1
1.4.37
1.4.37-rc1
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1.4.38-rc1
1.4.38.1
1.4.39
1.4.39-rc1
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1.4.39.2
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1.4.40
1.4.40-rc1
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1.4.40-rc3
1.4.40.1
1.4.40.2
1.4.41
1.4.41-rc1
1.4.41.1
1.4.41.2
1.4.42
1.4.42-rc1
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1.4.44
1.4.5
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1.4.7
1.4.7.1
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1.4.9
1.6.0
1.6.0-beta1
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1.6.0.1
1.6.0.10
1.6.0.11-rc1
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1.6.0.12
1.6.0.13
1.6.0.13-rc1
1.6.0.14
1.6.0.14-rc1
1.6.0.15
1.6.0.16
1.6.0.16-rc1
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1.6.0.17
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
1.6.0.20-rc1
1.6.0.21
1.6.0.21-rc1
1.6.0.22
1.6.0.23
1.6.0.23-rc1
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1.6.0.24
1.6.0.25
1.6.0.26
1.6.0.26-rc1
1.6.0.27
1.6.0.27-rc1
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1.6.0.27-rc3
1.6.0.28
1.6.0.28-rc1
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1.6.0.3
1.6.0.3-rc1
1.6.0.3.1
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1.6.0.5
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1.8.1.1
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1.8.10.0-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
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10.12.2-digiumphones
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10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
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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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10.4.0-rc1
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10.4.0-rc3
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10.5.0
10.5.0-digiumphones
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10.5.0-rc1
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10.5.1-digiumphones
10.5.2
10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
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10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
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10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
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11.0.0
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11.1.1
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11.14.0
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11.14.1
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11.15.0
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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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11.3.0
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11.4.0
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11.5.0
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11.5.1
11.6-cert11
11.6.0
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11.6.1
11.7.0
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11.8.0
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11.8.1
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12.0.0
12.0.0-alpha1
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12.1.0
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12.1.1
12.2.0
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12.3.0
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12.3.1
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12.4.0
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12.5.0
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12.5.1
12.6.0
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12.6.1
12.7.0
12.7.0-rc1
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12.7.1
12.7.2
12.8.0
12.8.0-rc1
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12.8.1
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13.0.0
13.0.0-beta1
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13.0.0-beta3
13.0.1
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13.1-cert2
13.1.0
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13.1.0-rc2
13.1.1
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13.2.0-rc1
13.2.1
13.3.0
13.3.0-rc1
13.3.1
13.3.2
13.4.0-rc1
13.5.0-rc1
13.6.0-rc1
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13.7.0
13.7.0-rc1
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certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
certified/1.8.11-cert3-rc2
certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
certified/1.8.15-cert1-rc2
certified/1.8.15-cert1-rc3
certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
certified/11.2-cert1-rc1
certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
certified/11.6-cert2
certified/11.6-cert3
certified/11.6-cert4
certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
5401 Commits (047a1e7dcc3e89fa951c5d714c68cd7fd2d09b37)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e910dbab90 |
build: Execute ldconfig to build cache.
On some platforms a multiarch approach is used for libraries. The build system does not take this into account and still places libraries into the lib directory if no --libdir is specified to configure. On initial startup this results in libasteriskssl.so not being found, as it is not in the multiarch lib directory. This change does the minimally invasive thing and executes ldconfig so that the libraries in the lib directory are found and their location cached. By doing so Asterisk starts up fine. ASTERISK-26705 Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519 |
8 years ago |
|
0fc27fa364 |
http: Ensure capath is defined on all http creations
ASTERISK-26794 #close Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1 |
8 years ago |
|
99b40e72ae |
libasteriskssl: do nothing with OpenSSL >= 1.1
OpenSSL 1.1 requires no explicit initialization. The hacks in the library are not needed. They also happen to fail running Asterisk. ASTERISK-26109 #close Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100 |
8 years ago |
|
e97e50b68b |
tcptls: use TLS_client_method with OpenSSL 1.1
OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous version-specific methods (such as TLSv1_client_method(). Other than being simpler to use and more correct (gain support for TLS newer that TLS1, in our case), the older ones produce a deprecation warning that fails the build in dev-mode. ASTERISK-26109 #close Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07 |
8 years ago |
|
0d555f0d81 |
openssl 1.1 support: use OPENSSL_VERSION_NUMBER
Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect the openssl 1.1 API. ASTERISK-26109 #close Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458 |
8 years ago |
|
0d6c99e715 |
Merge "cli: Fix various CLI documentation and completion issues" into 13
|
8 years ago |
|
bc2104819c |
Merge "channel: Protect flags in ast_waitfor_nandfds operation." into 13
|
8 years ago |
|
6958241b3f |
Merge "core: Cleanup some channel snapshot staging anomalies." into 13
|
8 years ago |
|
ea8a610776 |
cli: Fix various CLI documentation and completion issues
* app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/asterisk: Correct and extend completions for 'core show file version.' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca |
8 years ago |
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18f1b52601 |
channel: Protect flags in ast_waitfor_nandfds operation.
The ast_waitfor_nandfds operation will manipulate the flags of channels passed in. This was previously done without the channel lock being held. This could result in incorrect values existing for the flags if another thread manipulated the flags at the same time. This change locks the channel during flag manipulation. ASTERISK-26788 Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed |
8 years ago |
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8936568515 |
manager: Restore Originate failure behavior from Asterisk 11
In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in
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8 years ago |
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2817f87d27 |
core: Cleanup some channel snapshot staging anomalies.
We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a |
8 years ago |
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7e14e086cf |
srv: Fix crash when ast_srv_lookup is used and 0 records are returned.
When performing an SRV lookup using the ast_srv_lookup function it did not properly handle the situation where 0 records are returned. If this happened it would wrongly assume that at least one record was present. This change fixes the code so it will exit early if an error occurs or if 0 records are returned. ASTERISK-26772 patches: srv_lookup.patch submitted by nappsoft (license 6822) Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351 |
8 years ago |
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95e33a1f31 |
Merge "channel.c: Fix unbalanced read queue deadlocking local channels." into 13
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8 years ago |
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ba201f71af |
Merge "res_agi: Prevent an AGI from eating frames it should not. (Re-do)" into 13
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8 years ago |
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0bb5c58cba |
Merge "Frame deferral: Revert API refactoring." into 13
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8 years ago |
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3aee199913 |
audiohooks: Muting a hook can mute underlying frames
If an audiohook is placed on a channel that does not require transcoding, muting that hook will cause the underlying frames to be muted as well. The original patch is from David Woolley but I have modified slightly. ASTERISK-21094 #close Reported by: David Woolley Patches: ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded by David Woolley Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed |
8 years ago |
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43f0ff4b69 |
channel.c: Fix unbalanced read queue deadlocking local channels.
Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6 |
8 years ago |
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a199f94908 |
res_agi: Prevent an AGI from eating frames it should not. (Re-do)
A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43 |
8 years ago |
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6bed318a66 |
Frame deferral: Revert API refactoring.
There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496 |
8 years ago |
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d2f30f0225 |
Merge "ast_careful_fwrite to support EPIPE gracefully" into 13
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8 years ago |
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56e925f96a |
Merge "libastssl/pj: libastssl/pj should have an so_version" into 13
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8 years ago |
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484e8ed5e3 |
Merge "debug_utilities: Add ast_logescalator" into 13
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8 years ago |
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456bc3c704 |
debug_utilities: Add ast_logescalator
The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543 |
8 years ago |
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54b027916a |
libastssl/pj: libastssl/pj should have an so_version
Issue introduced in
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8 years ago |
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cb504ad6ae |
Merge "ari: Implement 'debug all' and request/response logging" into 13
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8 years ago |
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555e8cd2ba |
ast_careful_fwrite to support EPIPE gracefully
When a reading end of the network socket is closed by an AMI manager, the EPIPE is signaled when writing to our end, resulting in the spurious log error message ast_careful_fwrite: fwrite() returned error: Broken pipe Previously EPIPE was handled in ast_carefulwrite() a few lines above, but not in this function. ASTERISK-26753 Change-Id: I6a67335cd6526608bb9b78f796c626b1677664b8 |
8 years ago |
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b39c04fc02 |
Merge "PJPROJECT logging: Fix detection of max supported log level." into 13
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8 years ago |
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f5b31cb2ae |
Merge "frame.c: Fix off-nominal format ref leaks." into 13
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8 years ago |
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616e98ce0a |
Merge "T.140: Fix format ref and memory leaks." into 13
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8 years ago |
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da098e2227 |
Merge "astobj2.c: Add excessive ref count trap." into 13
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8 years ago |
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c54f9d2bf0 |
T.140: Fix format ref and memory leaks.
* channel.c:ast_sendtext(): Fix T.140 SendText memory leak. * format_compatibility.c: T.140 RED and T.140 were swapped. * res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak. * res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic scheduled red_write(). * res_rtp_asterisk.c: Some other minor misc tweaks. Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb |
8 years ago |
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607b3ac736 |
astobj2.c: Add excessive ref count trap.
Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a |
8 years ago |
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ab8cb5a7ce |
main/app.c: Memory corruption from early format destruction.
* make_silence() created a malloced silence slin frame without adding a slin format ref. When the frame is destroyed it will unref the slin format that never had a ref added. Memory corruption is expected to follow. * Simplified and fixed counting the number of samples in a frame list for make_silence(). * Eliminated an unnecessary RAII_VAR associated with the make_silence() frame. Change-Id: I47de3f9b92635b7f8b4d72309444d6c0aee6f747 |
8 years ago |
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dcd8e4b1a0 |
frame.c: Fix off-nominal format ref leaks.
* ast_frisolate() could leak frame format refs on allocation failures. * Similified code in ast_frisolate() and code used by ast_frisolate(). Change-Id: I79566d4d36b3d7801bf0c8294fcd3e9a86a2ed6d |
8 years ago |
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17f4989d49 |
ari: Implement 'debug all' and request/response logging
The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf |
8 years ago |
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30cb4eb57f |
PJPROJECT logging: Fix detection of max supported log level.
The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4 |
8 years ago |
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883e7fde31 |
abstract/fixed/adpative jitter buffer: disallow frame re-inserts
It was possible for a frame to be re-inserted into a jitter buffer after it had been removed from it. A case when this happened was if a frame was read out of the jitterbuffer, passed to the translation core, and then multiple frames were returned from said translation core. Upon multiple frames being returned the first is passed on, but sebsequently "chained" frames are put back into the read queue. Thus it was possible for a frame to go back into the jitter buffer where this would cause problems. This patch adds a flag to frames that are inserted into the channel's read queue after translation. The abstract jitter buffer code then checks for this flag and ignores any frames marked as such. Change-Id: I276c44edc9dcff61e606242f71274265c7779587 |
9 years ago |
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473330983b |
taskprocessor.c: Change when high water warning logged.
The task processor queue reached X scheduled tasks message was originally intended to get logged only once per task processor to prevent spamming the log. This is no longer necessary since high and low water thresholds can better control when the message is logged. It is beneficial to generate the warning each time a task processor reaches the high water level because PJSIP stops processing new requests while any high water alert is active. Without this change you would have to enable at least debug level 3 logging to know about a repeated alert trigger. * Made generate the warning message whenever a task is pushed into the task processor that triggers the high water alert. * Appended 'again' to the warning for a repeated high water alert trigger. Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999 |
9 years ago |
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ac31233dbe |
acl.c: Improve ast_ouraddrfor() diagnostic messages.
* Made not generate strings unless they will actually be used. ASTERISK-26672 Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3 |
9 years ago |
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75a6afbec5 |
MESSAGE: Flush Message/ast_msg_queue channel alert pipe.
ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2 |
9 years ago |
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4b233675d8 |
pjproject_bundled: Fix missing inclusion of symbols
Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to the CFLAGS. Not sure how they went missing. Also fixed an uninstall problem where we weren't removing the symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was there, I fixed it for libasteriskssl as well. Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556 |
9 years ago |
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8d56016ae4 |
Merge "tcptls: Use new certificate upon sip reload" into 13
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9 years ago |
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28b76ed667 |
Merge "PJPROJECT logging: Made easier to get available logging levels." into 13
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9 years ago |
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c3a509be7e |
Merge "res_rtp: Fix regression when IPv6 is not available." into 13
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9 years ago |
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bddc60df73 |
Merge "Frame deferral: Re-queue deferred frames one-at-a-time." into 13
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9 years ago |
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662a4741e2 |
Merge "OpenSSL 1.1.0 support" into 13
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9 years ago |
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2ceb609edb |
res_rtp: Fix regression when IPv6 is not available.
The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e |
9 years ago |
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44fe4a5769 |
PJPROJECT logging: Made easier to get available logging levels.
Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389 |
9 years ago |
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17b0b91afa |
Frame deferral: Re-queue deferred frames one-at-a-time.
The recent change that made frame deferral into an API had a behavior change to it. When frame deferral was completed, we would take all of the deferred frames and queue them all onto the channel in one call to ast_queue_frame_head(). Before frame deferral was API-ized, places that performed manual frame deferral would actually take each deferred frame and queue them onto the channel. This change in behavior caused the confbridge_recording test to start failing consistently. Without going too crazily deep into the details, a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect was attempting to break it out of the sleep, but because there were more frames in the channel read queue than expected, the channel ended up being unable to break from its sleep loop. By restoring the behavior of individual frame queuing after deferral, the test starts passing again. Note, this points to a potential underlying issue pointing to an "unbalance" that can occur when queuing multiple frames at once, and so a follow-up issue is being created to investigate that possibility. Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d |
9 years ago |