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

[Asterisk Bug Tracker]

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 05:31 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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 (0111581) riksta (reporter) - 2009-09-30 05:31
 https://issues.asterisk.org/view.php?id=15966#c111581 
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92.63.138.97 is the opensips balancer.

I am going to experiment with the sst and dialog module within opensips to
see if I can remedy the problem.

I will get back to you. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-30 05:31 riksta         Note Added: 0111581                          
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