[asterisk-bugs] [Asterisk 0010571]: Crash within app_voicemail

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10571 
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Reported By:                dtyoo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10571
Category:                   Channels/chan_local
Reproducibility:            unable to reproduce
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:            1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-27-2007 08:43 CDT
Last Modified:              08-27-2007 19:59 CDT
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Summary:                    Crash within app_voicemail
Description: 
We are getting crashes in app_voicemail on a fairly regular basis.  I'm
still working on steps to re-produce, but I thought I would post the
backtraces here in case someone could glean anything from them.  I will
update if I can figure out the steps to re-produce.
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---------------------------------------------------------------------- 
 dtyoo - 08-27-07 19:59  
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putnopvut-

You are probably right that this is an issue with the local channel. 
Since upgrading our servers from 1.2 to 1.4 we have been having other
issues related to server hangs (asterisk stops taking calls) where the
common thread is that the source channel for the calls is a local channel. 
Running "show channels" on a server in this state will go into an infinite
loop and won't return control to the console.  I'm still trying to get
re-producible scenarios for those issues as well.  Maybe its all related?

Corydon76-

See upload that has the info you requested.  The unusual thing about the
way these calls are getting to voicemail is that they are falling out of a
sip dialing loop.  The Dial command is dialing the extension at the
currently running server.  E.g.

Dial(SIP/1231231234 at this_server,70)

This is followed on the console by a:

Got SIP response 482 "Loop Detected" from this_server

Then a:

Now forwarding SIP/SOURCE_SIP_PEER-b7bc2938 to 'Local/1231231234 at pstn-in'
(thanks to SIP/this_server-08cec0c8)

Then into a voicemail macro that does little more than call VoiceMail with
appropriate arguments.

I'm not passing any explicit flags to the LOCAL channel as it seems that
asterisk is creating it for me based on the sip loop.

I understand that this is a less then ideal way to get a call delivered to
a different part of the dialplan on the same server.  It didn't seem to be
causing issues under 1.2 and doing it this way allowed us to simplify our
dialplan considerably. 

Issue History 
Date Modified   Username       Field                    Change               
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08-27-07 19:59  dtyoo          Note Added: 0069531                          
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