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

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Jan 27 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:                     feedback
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:              2009-01-27 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" 
====================================================================== 

---------------------------------------------------------------------- 
 (0098824) Nick_Lewis (reporter) - 2009-01-27 03:50
 http://bugs.digium.com/view.php?id=14256#c98824 
---------------------------------------------------------------------- 
Agreed the channel name should be SIP/peername as sip specific information
such as username should not be exposed to the rest of the pbx. The patch
corrects this.

Here is an extract of conf and cli showing the problem:

[root at asterisk1 ~]# tail -15 /etc/asterisk/sip_additional.conf

[test2]
type=peer
context=from-trunk
host=sip.atltelecom.com
port=5060
outboundproxy=
outboundproxyport=
insecure=invite
username=2345
secret=2345
fromuser=2345
fromdomain=sip.atltelecom.com
fromname=PBX

[root at asterisk1 ~]# asterisk -r
asterisk1*CLI> core show channels
Channel              Location             State   Application(Data)
SIP/2345-09938190    s at app-announcement-1 Up      Wait(1)


The channel name is given as SIP/2345 instead of SIP/test2 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-27 03:50 Nick_Lewis     Note Added: 0098824                          
======================================================================




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