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Summary:ASTERISK-29031: Deadlock and stop making calls
Reporter:Luciano Moreira (lmoreira@logictelecom.com.br)Labels:
Date Opened:2020-08-14 13:31:20Date Closed:2020-09-02 12:00:01
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:16.11.1 Frequency of
Occurrence		Frequent
Related
Issues:
Environment:Freepbx latest version. Custom DialplanAttachments:( 0) codec_opus.manifest.xml
( 1) core.e1virtual.logictelecom.com.br-2020-08-14T11-42-15-0300-brief.txt
( 2) core.e1virtual.logictelecom.com.br-2020-08-14T11-42-15-0300-full.txt
( 3) core.e1virtual.logictelecom.com.br-2020-08-14T11-42-15-0300-info.txt
( 4) core.e1virtual.logictelecom.com.br-2020-08-14T11-42-15-0300-locks.txt
( 5) core.e1virtual.logictelecom.com.br-2020-08-14T11-42-15-0300-thread1.txt
Description:Several times per day Asterisk stops (no crash) making calls and chan_sip became no responsive. Asterisk service doesn't stop and it must to be killed before running it again.
Comments:By: Asterisk Team (asteriskteam) 2020-08-14 13:31:21.572-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
By: Luciano Moreira (lmoreira@logictelecom.com.br) 2020-08-14 13:33:24.759-0500

coredump after deadlock.
By: Kevin Harwell (kharwell) 2020-08-14 16:19:25.027-0500

The coredump is unfortunately missing symbols. Please ensure DONT_OPTIMIZE is enabled in menuselect [1], and next time it happens please obtain another backtrace and attach again here.

Also it might be useful as well to get a backtrace, wait a minute or two and then get another one so we can get a better picture of what threads are truly "stuck".

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Thanks!
By: Luciano Moreira (lmoreira@logictelecom.com.br) 2020-08-14 16:41:37.434-0500

It's a FreePBX, asterisk installed from RPM. So, It's not possible compile on this server. But, I disable opus codec for all system about 5 hours ago, and so far so good. Solid like a rock.
I'll observe couple days before give a final conclusion about opus.
By: Luciano Moreira (lmoreira@logictelecom.com.br) 2020-08-17 13:43:09.309-0500

Hi Guys,

I'm using FreePBX 15.0.16.72 with Asterisk 16.11.1
Couple days ago, I've created SIP trunk to Freeswitch 1.10.4 using opus codec.
Then, began to crash or got stuck many times a day. No calls coming in or out of the box.
We did deep investigation and only recente change was SIP trunk to FS using opus.

So, we changed codec to g722 4 days ago. Since then, no crash, no deadlock, no problem at all, as it has been working for months.
By: Kevin Harwell (kharwell) 2020-08-17 17:37:03.246-0500

When codec_opus is loaded what's the output of the following commands:
{noformat}
*CLI> module show like codec_opus.so
*CLI> module show like format_ogg_opus.so
{noformat}
Also can you attach your _codec_opus.manifest.xml_ file found in the asterisk module libs directory. For insatnce:
{noformat}
/usr/lib/asterisk/modules/codec_opus.manifest.xml
{noformat}
If you have anything in _codecs.conf_ configured for opus please post/attach that here as well.

Also if I understand opus is being used between an Asterisk and Freeswitch systems? Used with any endpoints, phones, or devices, or browser?
By: Kevin Harwell (kharwell) 2020-08-18 10:45:26.439-0500

Also here's more information about collecting debug and backtraces with symbols in freepbx:

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug
By: Luciano Moreira (lmoreira@logictelecom.com.br) 2020-08-18 12:05:50.763-0500


{code:xml}
*CLI> module show like codec_opus.so
Module                         Description                              Use Count  Status      Support Level
codec_opus.so                  OPUS Coder/Decoder                       0          Running          extended
1 modules loaded

*CLI> module show like format_ogg_opus.so
Module                         Description                              Use Count  Status      Support Level
format_ogg_opus.so             OGG/Opus audio                           0          Running              core
1 modules loaded
{code}

I attached codec_opus.manifest.xml file.
By: Kevin Harwell (kharwell) 2020-08-18 15:28:25.223-0500

Do you still have, or can you get, the Asterisk log for around when the problem happened? If so please attach it to this issue.
By: Asterisk Team (asteriskteam) 2020-09-02 12:00:01.251-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines