[asterisk-bugs] [Asterisk 0010481]: SIP with canreinvite=yes through multiple Asterisk instances fails

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Nov 15 06:05:21 CST 2007


The following issue has been set as RELATED TO issue 0009431. 
====================================================================== 
http://bugs.digium.com/view.php?id=10481 
====================================================================== 
Reported By:                mavetju
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10481
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.10.1  
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 79553 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-17-2007 08:50 CDT
Last Modified:              10-11-2007 14:52 CDT
====================================================================== 
Summary:                    SIP with canreinvite=yes through multiple Asterisk
instances fails
Description: 
The story at http://www.mavetju.org/~edwin/asterisk-sip-reinvite.html
describes a problem I experienced with calls coming from one of our
providers where during the SIP handshake our equipment was reinviting
the SIP session: The RTP stream was never setup. We experienced
this after the upgrade from 1.2 to 1.4 (the latest SVN version),
before that it always has worked.

To simulate this problem, I have setup one SIP phone, three identical
Asterisk instances and a connection towards the end-point: A Cisco
Call Manager. The only varying factor in the experiments was the
option "canreinvite": When using "canreinvite=no", it always worked
fine, but when using "canreinvite=yes", it broke down after two
hops.

I have written down the whole setup, the configurations, the scenarios
and the results at http://www.mavetju.org/~edwin/c2-flow.txt.
Attached to each scenario are the SIP packets (captured with ngrep
and processed into a flow visualiser).
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0010449 Maximum retries for seqno 102 when re-i...
related to          0009431 Modify connection: Response 491 not han...
====================================================================== 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-15-07 06:05  oej            Relationship added       related to 0009431  
======================================================================




More information about the asterisk-bugs mailing list