[asterisk-bugs] [Asterisk 0015621]: [patch] session-expires default timer wrong

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15621 
====================================================================== 
Reported By:                fnordian
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   15621
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for review
Target Version:             1.6.1.6
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 209451 
Request Review:              
====================================================================== 
Date Submitted:             2009-07-31 04:13 CDT
Last Modified:              2009-09-03 10:49 CDT
====================================================================== 
Summary:                    [patch] session-expires default timer wrong
Description: 
There is a bug in chan_sip which causes asterisk to set invalid
session-expires-header in 200 OK after invites. My guess is that it's been
there for a long time but got activated with bugfix 15403. 

The problem is, that for invites without session-expires-headers a reply
is generated with Session-Expires: -1;refresher=uas causing the uac to
hangup the call with BYE immediately. 

A workaround is setting session-timers=refuse in sip.conf
====================================================================== 

---------------------------------------------------------------------- 
 (0110139) atis (reporter) - 2009-09-03 10:49
 https://issues.asterisk.org/view.php?id=15621#c110139 
---------------------------------------------------------------------- 
Attached two SIP debug logs for Asterisk 1.6.1.5 and Audiocodes MP-124 FXS
calling itself without re-INVITEs. 

Initially Asterisk 1.6.1.5 is broken and no voice is passed through, after
using this patch everything seems to work. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-03 10:49 atis           Note Added: 0110139                          
======================================================================