[asterisk-bugs] [JIRA] (ASTERISK-22380) Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Sat Aug 24 18:05:02 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22380?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-22380:
------------------------------------
Description:
To reproduce:
* see attached pjsip.conf
* Dialed from either Jitsi or any various hard phone to an extension with Playback or Dial.
Note:
Unloading res_rtp_multicast.so allows calls to Playback to function normally. Calls from pjsip to pjsip endpoints (on same LAN) have no audio.
was:
To reproduce:
* see attached pjsip.conf
* Dialed from either Jitsi or any various hard phone to an extension with Playback or Dial.
> Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c
> -----------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-22380
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22380
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip, Core/RTP, Resources/res_pjsip
> Affects Versions: 12
> Environment: SVN-branch-12-r397614
> Reporter: Rusty Newton
> Severity: Critical
> Attachments: backtrace5.txt, full5.txt, pjsip.txt
>
>
> To reproduce:
> * see attached pjsip.conf
> * Dialed from either Jitsi or any various hard phone to an extension with Playback or Dial.
> Note:
> Unloading res_rtp_multicast.so allows calls to Playback to function normally. Calls from pjsip to pjsip endpoints (on same LAN) have no audio.
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