[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:              2010-11-22 15:47 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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 (0129063) schmidts (manager) - 2010-11-22 15:47
 https://issues.asterisk.org/view.php?id=18344#c129063 
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I think this is a regression cause RFC 3261 says: 
The escaping rules defined above allow a semicolon to appear unescaped in
this field.  For the purposes of this protocol, the field is opaque.  The
structure of that value is only useful to the SIP element responsible for
the resource.

See chapter 19.1.3 for the user part of a Sip-Uri and telephony subscriber
field handling (RFC RFC 2806 part 9)

btw please attach logfiles only as text files and not in your
description.


thx
Stefan 

Issue History 
Date Modified    Username       Field                    Change               
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2010-11-22 15:47 schmidts       Note Added: 0129063                          
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