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

TSAREGORODTSEV Yury (JIRA) noreply at issues.asterisk.org
Wed Mar 29 08:40:09 CDT 2017


     [ https://issues.asterisk.org/jira/browse/ASTERISK-26904?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

TSAREGORODTSEV Yury updated ASTERISK-26904:
-------------------------------------------

    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.

  was:
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.

        Summary: codec silk crash asterisk on outgoing call  (was: codec silk crash asterisk)

> 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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