[asterisk-bugs] [JIRA] (ASTERISK-21047) asterisk crashes during an attended transfer SIP -> SIP -> DAHDI
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Thu Feb 14 19:35:58 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21047?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-21047:
------------------------------------
Assignee: piero ferraresso
Status: Waiting for Feedback (was: Triage)
Thanks for the report. Please attach a full log with VERBOSE and DEBUG set to level 5 showing the call where the crash occurs. Also, please attach sanitized sip.conf, dahdi.conf and chan_dahdi.conf files. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
> asterisk crashes during an attended transfer SIP -> SIP -> DAHDI
> ----------------------------------------------------------------
>
> Key: ASTERISK-21047
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-21047
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_dahdi, Channels/chan_sip/Transfers
> Affects Versions: 11.2.0, 11.2.1
> Environment: debian 5.0.10, opensips v.1.8.2, digium te420B, PBX Aastra MD110
> Reporter: piero ferraresso
> Assignee: piero ferraresso
> Severity: Critical
> Attachments: backtrace.txt
>
>
> I'm trying to make an attended transfer.
> Sip hardphone A call sip hardphone B (both Polycom SPIP 450);
> B press the 'transfer' softkey and compose the number of legacy hardphone C (PBX Aastra connected to asterisk with a Qsig trunk using a digium te420B card).
> After legacy hardphone C answered, B press again the 'transfer' softkey to transfer the call to phone A. At this point asterisk crashes.
> Ps The some scenario using asterisk 1.4 works
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