[asterisk-bugs] [JIRA] (ASTERISK-26904) codec silk crash asterisk on outgoing call

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26904?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236249#comment-236249 ] 

Rusty Newton commented on ASTERISK-26904:
-----------------------------------------

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



> codec silk crash asterisk on outgoing call
> ------------------------------------------
>
>                 Key: ASTERISK-26904
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26904
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_silk
>    Affects Versions: 13.14.0, 14.3.0
>         Environment: Test on x86_64, x86_32 same result
>            Reporter: TSAREGORODTSEV Yury
>
> 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.



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