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

[Asterisk Bug Tracker]

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:                
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Project:                    Asterisk
Issue ID:                   14256
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-01-16 04:31 CST
Last Modified:              2009-01-16 05:01 CST
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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" 
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 (0098007) Nick_Lewis (reporter) - 2009-01-16 05:01
 http://bugs.digium.com/view.php?id=14256#c98007 
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This patch also helps the flash operator panel to work. The line state of a
fop button is determined by the channel name (as shown in 'core show
channels' etc) but the click-to-dial action of a fop button is determined
by the peername (as in dialplan command Dial(SIP/peername)). Without this
patch it is not possible to select a button name that gets both fop
features to work 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-16 05:01 Nick_Lewis     Note Added: 0098007                          
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