[asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Dec 4 10:14:33 CST 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10665 
====================================================================== 
Reported By:                rjain
Assigned To:                oej
====================================================================== 
Project:                    Asterisk
Issue ID:                   10665
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 81013 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             09-07-2007 03:43 CDT
Last Modified:              12-04-2007 10:14 CST
====================================================================== 
Summary:                    [patch] SIP Session-Timers Support in Asterisk
Description: 
The Asterisk SIP stack currently does not support SIP Session-Timers (RFC
4028). This leads to defunct SIP sessions in Asterisk when calls do not
clear through normal signaling procedures due to network or end-point
failures.

John Todd recently discussed this concept on asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2007-July/028574.html

John Todd, JR Richardson and Kevin Fleming have expressed interest in
seeing this feature supported in Asterisk.

A software design document for this feature and code changes (unified diff
of chan_sip.c) are attached to this report. Digium has my code submission
agreement on file.
====================================================================== 

---------------------------------------------------------------------- 
 oej - 12-04-07 10:14  
---------------------------------------------------------------------- 
Well, the SIP_PVT is used for many things that are not calls and I don't
want to keep expanding it. Everyone that adds something says "look, it's
only 40 bytes"...

Sorry, but it needs to be encapsulated. There are previous examples on how
to make it work properly and clean it up without the risk of memory leaks,
so there's plenty of previous experience. No need to worry. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-04-07 10:14  oej            Note Added: 0074753                          
======================================================================




More information about the asterisk-bugs mailing list