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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Apr 17 09:44:57 CDT 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14256 
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Reported By:                Nick_Lewis
Assigned To:                file
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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 09:44 CDT
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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" 
====================================================================== 

---------------------------------------------------------------------- 
 (0103365) svnbot (reporter) - 2009-04-17 09:44
 http://bugs.digium.com/view.php?id=14256#c103365 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 188947

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r188947 | file | 2009-04-17 09:44:56 -0500 (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
........

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http://svn.digium.com/view/asterisk?view=rev&revision=188947 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-17 09:44 svnbot         Checkin                                      
2009-04-17 09:44 svnbot         Note Added: 0103365                          
======================================================================




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