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Summary:ASTERISK-25785: PJSIP Segmentation fault.
Reporter:Ross Beer (rossbeer)Labels:
Date Opened:2016-02-12 05:21:31.000-0600Date Closed:2016-03-02 07:14:12.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Resources/res_pjsip_multihomed
Versions:13.7.2 Frequency of
Occurrence
Frequent
Related
Issues:
Environment:Centos 6.6Attachments:( 0) backtrace_2016-02-12T11_04_49.txt
( 1) backtrace_2016-02-22_1128.txt
Description:Asterisk crashes when using PJ SIP channel diver.

Issue lies in 'multihomed_on_tx_message' however the test environment isn't multi-homed.
Comments:By: Asterisk Team (asteriskteam) 2016-02-12 05:21:33.514-0600

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.

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].

By: Joshua C. Colp (jcolp) 2016-02-12 08:25:41.425-0600

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

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

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



By: Ross Beer (rossbeer) 2016-02-22 05:43:28.466-0600

There is nothing special happening here, just incoming and outgoing calls.

The issue looks to be related to:

#0  0x00007fed6b5863c6 in pj_strcmp () from /usr/lib64/libpj.so.2

I am uploading a new backtrace which hasn't been optimised.

By: Ross Beer (rossbeer) 2016-02-22 05:59:50.592-0600

I'm wondering if this issue is related to pjproject SVN commits after 5240 which has depreciated a function as it was causing a crash.

If I install the latest pjproject from trunk, asterisk starts and then crashes when started. Is this because Asterisk is attempting to use the method that has been depreciated?

By: Ross Beer (rossbeer) 2016-03-02 04:05:38.172-0600

Can this ticket be closed please as the issue is fixed in trunk when using the latest commit of PJSIP and Asterisk.

By: Joshua C. Colp (jcolp) 2016-03-02 07:14:12.416-0600

Closed per reporter.