[asterisk-bugs] [Asterisk 0018344]: regression improper sip parse when invite contains values to the left of the @ ; phone-context=+1; npdi=yes

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18344 
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Reported By:                danimal
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18344
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-21 13:58 CST
Last Modified:              2010-12-03 13:30 CST
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Summary:                    regression improper sip parse when invite contains
values to the left of the @ ;phone-context=+1;npdi=yes
Description: 
This is related to a circa 2006 issue
https://issues.asterisk.org/view.php?id=7761

when a call origionates from the pstn to a sonus and terminates to
asterisk 1.8 sonus adds values to the left side of the @ side in the
invite. Previous versions of asterisk would return only the telephone
number in ${EXTEN}. 1.8 is returning everything to the left of the @ sign
in ${EXTEN}/

steps to reproduce.
1. install centos 5.5
2. install asterisk 1.8 via the digium repository / yum install
3. my upstream provider uses a sonus GW for pstn terminaltion.
4. call from the pstn to asterisk 1.8
5. exten =>_X!,1,NOOP(dialed number => ${EXTEN})
6, notice the value of ${EXTEN}

anticipated results
asterisk will return only the DID number in ${EXTEN}

actual results
asterisk is returning everything to the left side of the @ symbol in
${EXTEN} this value is then also stored in the cdr records as well.
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 (0129303) MrHanMan (reporter) - 2010-12-03 13:30
 https://issues.asterisk.org/view.php?id=18344#c129303 
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I'm seeing this issue as well coming from 1.6 to 1.8.  We have calls coming
from a Nortel CS1000 via SIP.  Previously, the DID is set to just the
extension dialed.  Now, it is set to "8000;phone-context=cdp.udp".  I
attached a copy of the console output with SIP debugging on. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-03 13:30 MrHanMan       Note Added: 0129303                          
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