[asterisk-bugs] [Asterisk 0015966]: Asterisk generates BYE at EXACTLY 900 seconds (15 min) and terminates call

Wed Sep 30 05:34:15 CDT 2009

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15966 
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Reported By:                riksta
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15966
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.5 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-09-25 13:57 CDT
Last Modified:              2009-09-30 05:34 CDT
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Summary:                    Asterisk generates BYE at EXACTLY 900 seconds (15
min) and terminates call
Description: 
I have an incoming SIP call, which then dials out to another SIP trunk and
the calls are bridged via asterisk.

After exactly 900 seconds there is a BYE generated and the call completely
drops.

I have canreinvite=no specified in both the sip.conf general and for the
actual trunk stanza

http://office.encompassmedia.co.uk/dump.tgz has a full SIP/RTP media pcap
dump for both legs of the call which you can merge within wireshark.

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 (0111582) davidw (reporter) - 2009-09-30 05:34
 https://issues.asterisk.org/view.php?id=15966#c111582 
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I suspect what is happening is that the re-invite is not going to the same
worker as the original invite dialogue.

The 92.63.138.97 came from the Via header and may well be one step away
from the actual responder. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-30 05:34 davidw         Note Added: 0111582                          
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