[asterisk-bugs] [JIRA] (ASTERISK-24633) Asterisk Replies : Call leg/transaction does not exist

[Sid Mason (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24633?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224213#comment-224213 ] 

Sid Mason commented on ASTERISK-24633:
--------------------------------------

If I set debug mode and recompile, using gdb and debug, shall asterisk be able to take heavy load to collect the logging information to resolve this issue?

While calls on my server are private and user confidential, how can I submit this logs to asterisk issue tracker and private which some form of NDA is agreed by asterisk? 

I have to ensure our client's phone numbers and destination calls are safe and kept private.

Regards

> Asterisk Replies : Call leg/transaction does not exist 
> -------------------------------------------------------
>
>                 Key: ASTERISK-24633
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24633
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.15.0, 13.1.0
>         Environment: CentOS 6.5 64bit
>            Reporter: Sid Mason
>            Assignee: Sid Mason
>            Severity: Critical
>         Attachments: DEBUG_LOG Asterisk 11.15.txt, DEBUG_LOG.txt
>
>
> It seems asterisk erases the information about the ongoing call through Chan_SIP 
> This happens when :
> A) This is right before the time Asterisk looses all SIP Dialog status related second leg on Bridges
> Dec 23 03:29:24 callback01my kernel: asterisk[2652]: segfault at 9d0 ip 000000000048c120 sp 00007f9463ffc858 error 4 in asterisk[400000+208000]
> Note:  At above point asterisk uptime 'core show uptime' is reset to zero, but strangely asterisk recover or continue to operate.
> B) 20 or AMI listener is attached to asterisk on port 5038 and they all have filter as below:
> C) AMI has filter of the UNIQUE ID of the channels and others:
> '!Event: AGIExec'
> '!Event: VarSet'
> '!Event: RTCPSent'
> '!Event: RTCPReceived'
> '!Event: VarSet'
> D) the call durations are usually above 10 minutes
> E) the bridge is doing unlink before the hang-up and somehow asterisk must hold the information for SIP Dialog right before the hang-up
> I tried asterisk 11.15 and the same result was there.
> At this point ANSWEREDTIME is reported as NULL as well
> F) At incident asterisk is loosing track of multiple calls at once.



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