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

[Asterisk Bug Tracker]

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:                     closed
Target Version:             1.6.3.0
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-01-16 04:31 CST
Last Modified:              2009-04-17 09:50 CDT
====================================================================== 
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" 
====================================================================== 

---------------------------------------------------------------------- 
 (0103368) svnbot (reporter) - 2009-04-17 09:50
 http://bugs.digium.com/view.php?id=14256#c103368 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 188950

_U  branches/1.6.2/
U   branches/1.6.2/channels/chan_sip.c

------------------------------------------------------------------------
r188950 | file | 2009-04-17 09:50:43 -0500 (Fri, 17 Apr 2009) | 29 lines

Merged revisions 188947 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
  
  Merged revisions 188946 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15
lines
    
    Fix a bug where a value used to create the channel name was bogus.
    
    This commit fixes the scenario where an incoming call is authenticated
    using a peer entry. Previously the channel name was created using
either
    the username setting from the sip.conf entry or the IP address that
the
    call came from. Now the channel name will be created using the peer
name
    itself. This commit will not change the way the channel name is
generated
    for users or friends.
    
    (closes issue http://bugs.digium.com/view.php?id=14256)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-chname.patch uploaded by Nick (license 657)
    Tested by: Nick_Lewis, file
  ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=188950 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-17 09:50 svnbot         Checkin                                      
2009-04-17 09:50 svnbot         Note Added: 0103368                          
======================================================================