[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:                     acknowledged
Asterisk Version:           1.8.0 
JIRA:                       SWP-2637 
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:              2011-01-02 14:11 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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 (0130082) danimal (reporter) - 2011-01-02 14:11
 https://issues.asterisk.org/view.php?id=18344#c130082 
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this issue DOES NOT exist in 1.6.2.15

Asterisk 1.6.2.15 built by root @ localhost.localdomain on a x86_64
running Linux on 2010-12-13 03:39:01 UTC

    -- Executing [4254887565 at default:9] NoOp("SIP/360trunk-00000190",
"dialed number => 4254887565") in new stack 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-02 14:11 danimal        Note Added: 0130082                          
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