[asterisk-bugs] [JIRA] (ASTERISK-24456) SIP deadlock in transfer scenario between Asterisk Servers

Peter Katzmann (JIRA) noreply at issues.asterisk.org
Fri Nov 14 08:13:29 CST 2014


     [ https://issues.asterisk.org/jira/browse/ASTERISK-24456?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Peter Katzmann updated ASTERISK-24456:
--------------------------------------

    Status: Waiting for Feedback  (was: Waiting for Feedback)

Well i'm trying to narrow down the problem.

The server skyrocks load when a calls b but was forwarded to c.
But only while a and c are connected not during ringing. Audio is delayed also.

When i now try to transfer the call back from c to b, the system even loose sip connection or crashes.

Is there any possibility to find the root cause of this high load inside asterisk ?

Channel              Location             State   Application(Data)             
SIP/asterisk-asteris (None)               Up      AppDial((Outgoing Line))      
Local/1000 at from-inte (None)               Up      AppDial((Outgoing Line))      
Local/1000 at from-inte s at to-node:9          Up      Dial(SIP/asterisk asteriskdev1
SIP/2202-00000004    s at macro-executeagi:8 Up      Dial(Local/1000 at from-internal-



> SIP deadlock in transfer scenario between Asterisk Servers
> ----------------------------------------------------------
>
>                 Key: ASTERISK-24456
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24456
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_local, Channels/chan_sip/General
>    Affects Versions: 11.13.0
>         Environment: Ubuntu precise
>            Reporter: Peter Katzmann
>            Assignee: Peter Katzmann
>            Severity: Critical
>         Attachments: asterisk-sip-loc, backtrace.txt, serverchef.txt, serversek.txt, threads.txt
>
>
> We have 3 Asterisk server (a,b and c)
> we have also 3 different user (Caller, Chef, Sek) .
> User Caller on Server a
> User Chef on Server b
> User Sek on Server c
> Now Caller Dials to Chef
> Chef has a call rule via agi to only accept direct calls from Sek, all other calls are transferred to Sek.
> So the call from Caller on server a is tranfered to Sek on server c automatically
> Sek accepts the calls
> Now Sek will do a unattended transfer of caller to Chef
> Now the server b with Chef shows no action on the dialplan anymore and blocks
> Sometimes also crashes.



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