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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26904?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236560#comment-236560 ] 

Asterisk Team commented on ASTERISK-26904:
------------------------------------------

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

> 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
>            Assignee: 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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)