[asterisk-bugs] [Asterisk 0014256]: [patch] SIP Channel name is not unique

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Feb 11 03:50:12 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14256 
====================================================================== 
Reported By:                Nick_Lewis
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   14256
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
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:             2009-01-16 04:31 CST
Last Modified:              2009-02-11 03:50 CST
====================================================================== 
Summary:                    [patch] SIP Channel name is not unique
Description: 
The name of the asterisk channel that is created on an incoming sip call is
not unique

There can be two trunks with different ITSP but same username e.g.
[trunk1]
username=nicklewis
host=sip.myitsp1.com

[trunk2]
username=nicklewis
host=sip.myitsp2.net

The sip channel name of the asterisk channel that is created when a call
comes into these trunks is "SIP/nicklewis" in both cases. The sip channel
should be named after the peername instead e.g. "SIP/trunk1" and
"SIP/trunk2" 
====================================================================== 

---------------------------------------------------------------------- 
 (0099873) Nick_Lewis (reporter) - 2009-02-11 03:50
 http://bugs.digium.com/view.php?id=14256#c99873 
---------------------------------------------------------------------- 
It appears that pvt->peername and pvt->username are both populated when
there is a match in check_user_ok() or check_peer_ok(). In cases where
peername is null then username is also null so a policy of 'use username
otherwise' would never actually be applied. Note that if there is no match
then sip_new() internally requisitions fromdomain for the channel name.

If there is real worry about losing username from the channel name then
perhaps there could be a global setting legacychannelname=yes 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-11 03:50 Nick_Lewis     Note Added: 0099873                          
======================================================================




More information about the asterisk-bugs mailing list