[asterisk-bugs] [Asterisk 0014250]: [patch] Incoming calls matched to the wrong peer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14250 
====================================================================== 
Reported By:                Nick_Lewis
Assigned To:                oej
====================================================================== 
Project:                    Asterisk
Issue ID:                   14250
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-15 05:58 CST
Last Modified:              2009-01-23 07:39 CST
====================================================================== 
Summary:                    [patch] Incoming calls matched to the wrong peer
Description: 
If there are multiple sip trunks with the same ITSP then an incoming call
is arbitarily matched to the last peer with the same host IP address. This
is not a serious problem because the DID is still correct but it does have
many insidious effects due to the incorrect channel name
====================================================================== 

---------------------------------------------------------------------- 
 (0098532) Nick_Lewis (reporter) - 2009-01-23 07:39
 http://bugs.digium.com/view.php?id=14250#c98532 
---------------------------------------------------------------------- 
The patch does not break existing usage but augments it so that an asterisk
sip trunk can operate as a proper UAC as defined in RFC3261. It only
matches the request uri to the peername if all other matches fail so it is
fully backward compatible.

The OEJ proposal to match on domain does not address this issue. This
issue occurs when multiple sip trunks have the same source ip and the same
domain. For example an ITSP may provide two sip lines each connected to a
different PSTN number. Asterisk is currently unable to distinguish between
incoming calls on these lines. Adding matching to the domain in addition to
the source ip address would leave the problem unresolved.

A simple two line sip phone has no problems with supporting two sip lines
from one ITSP and nor should asterisk. This patch fixes the problem. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-23 07:39 Nick_Lewis     Note Added: 0098532                          
======================================================================