[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 noreply at bugs.digium.com
Fri Dec 3 14:17:28 CST 2010


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 14:17 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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---------------------------------------------------------------------- 
 (0129304) kg4ysy (reporter) - 2010-12-03 14:17
 https://issues.asterisk.org/view.php?id=18344#c129304 
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We also run CS1000 to Asterisk.  As a "fix"  I made the following
adjustment to the incoming context for the CS1000 trunks.  This may have to
be slightly adjusted depending on what the CS1000 is sending you. 
Hopefully this is helpful for those looking for a quick "fix".

[from-trunk-sip-CS1000]
exten => _.,1,Set(GROUP()=OUT_2)
exten => _.,2,GotoIf($[ "${EXTEN:-22}" =
";phone-context=udp.cdp"]?goto1:goto2
exten => _.,n(goto1),Goto(from-trunk,${EXTEN:0:-22},1)
exten => _.,n(goto2),Goto(from-trunk,${EXTEN},1) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-03 14:17 kg4ysy         Note Added: 0129304                          
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