[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)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22380?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-22380:
------------------------------------

    Attachment: pjsip.txt
                full5.txt
                backtrace5.txt
    
> 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.

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