Summary:	ASTERISK-26904: codec silk crash asterisk on outgoing call
Reporter:	TSAREGORODTSEV Yury (tsarik)	Labels:
Date Opened:	2017-03-29 08:32:00	Date Closed:	2020-01-14 11:13:52.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Codecs/codec_silk
Versions:	13.14.0 14.3.0	Frequency of Occurrence
Related Issues:
Environment:	Test on x86_64, x86_32 same result	Attachments:
Description:	codec_silk crash asterisk with asterisk[1809]: segfault at 4 ip b3e1722d sp ae33f0e0 error 4 in codec_silk.so[b3e15000+2f000] tested on x64 and i386 architectures. Both hosts have ubuntu 16.04 CPU on both: Intel(R) Xeon(R) CPU E5-1650 Tested on asterisk 13, 14, both crash. Crash happened only if 1st host make outgoing call in SILK on 2nd host. If I do incoming call from SILK supported softphone with dummy Playback extension - everything works correctly.
Comments:	By: Asterisk Team (asteriskteam) 2017-03-29 08:32:01.427-0500 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. 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: Rusty Newton (rnewton) 2017-04-02 19:31:27.160-0500 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: Rusty Newton (rnewton) 2017-04-02 19:31:39.026-0500 Thank you for the crash report. However, we need more information to investigate the crash. Please provide: 1. A backtrace generated from a core dump using the instructions provided on the Asterisk wiki [1]. 2. Specific steps taken that lead to the crash. 3. All configuration information necesary to reproduce the crash. Thanks! [1]: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace By: Asterisk Team (asteriskteam) 2017-04-17 12:00:01.361-0500 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