[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:              2009-09-29 15:00 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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 (0111537) riksta (reporter) - 2009-09-29 15:00
 https://issues.asterisk.org/view.php?id=15966#c111537 
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Thanks guys.... Your feedback helped to track down the source of the
issue.

We are using OpenSIPs load balancer which uses the dispatch module to send
the calls to an asterisk node in a round robin manner.

In the attached file the call id we are concerned with is Call-ID:
66B6355D-AC4511DE-9169D31F-EFD16753 at 213.166.5.140


Looking at the CLI SIP debug output file now attached to this bug, (from
my limited knowledge) I can see that Asterisk is i think generating a
RE-INVITE (despite canreinvite=no being explicitly set both globally and on
the peer) at 900 seconds and inside the header you see  X-asterisk-Info:
SIP re-invite (Session-Timers)


At this point I do not have enough knowledge to work out WHY the actual
call is dropped and why asterisk creates the BYE.

I am hoping that now I have managed to attach a proper log someone can
kindly shed some light on this.

I have also reported this to digium under our support contract; case
LWE-506524_cli

Cheers! 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-29 15:00 riksta         Note Added: 0111537                          
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