[asterisk-bugs] [Asterisk 0017127]: Wrong encoding of SIP URI

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Apr 5 15:16:51 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17127 
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Reported By:                sdaniels
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17127
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.30 
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:             2010-04-01 16:37 CDT
Last Modified:              2010-04-05 15:16 CDT
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Summary:                    Wrong encoding of SIP URI
Description: 
When callerid(num) of an incoming call has a # in the number the SIP URI is
not encoded properly.

Should be encoded as %23

A lot of phones just reject the call.



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---------------------------------------------------------------------- 
 (0120142) pabelanger (manager) - 2010-04-05 15:16
 https://issues.asterisk.org/view.php?id=17127#c120142 
---------------------------------------------------------------------- 
Thank you for taking the time to report this bug and helping to make
Asterisk better. Unfortunately, we cannot work on this bug because your
description didn\'t include enough information. You may find it helpful to
read \"Asterisk Issue Guidelines\"
http://www.asterisk.org/developers/bug-guidelines. [^] We\'d be grateful if
you would then provide a more complete description of the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the
problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a
SIP trace (if this is SIP related), and configuration information such as
dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-04-05 15:16 pabelanger     Note Added: 0120142                          
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