[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. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15966 
====================================================================== 
Reported By:                riksta
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-09-25 13:57 CDT
Last Modified:              2009-09-29 15:07 CDT
====================================================================== 
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.

====================================================================== 

---------------------------------------------------------------------- 
 (0111538) riksta (reporter) - 2009-09-29 15:07
 https://issues.asterisk.org/view.php?id=15966#c111538 
---------------------------------------------------------------------- 
I forgot to mention in the above comment that I have found a workaround by
enabling session-timers=refuse in sip.conf

Naturally, I'd like to be able to find a proper resolution to this issue
:)

R 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-29 15:07 riksta         Note Added: 0111538                          
======================================================================