[asterisk-bugs] [Asterisk 0016857]: Incorrect checking of Refer-To and Referred-By SIP headers

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Feb 19 12:16:43 CST 2010


The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=16857 
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Reported By:                tomsullivan
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16857
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.2.X 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2010-02-17 20:58 CST
Last Modified:              2010-02-19 12:16 CST
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Summary:                    Incorrect checking of Refer-To and Referred-By SIP
headers
Description: 
Asterisk 1.2.39.

Within asterisk-1.2.39/channels/chan_sip.c, lines 7032 and 7039 the
Refer-To and Referred-By headers are parsed from the SIP request. 

The get_header(...) method returns empty string if the header is not
found, but the test on these lines is only for NULL, so both refer_to and
referred_by can get through as "".

This is not a problem per se for refer_to, as it is checked later on (line
7050) and -1 is returned.

However, referred_by gets set to NULL (line 7057), which (in concert with
the bristuff patches) causes a SEGFAULT when dereferenced.
====================================================================== 

---------------------------------------------------------------------- 
 (0118283) lmadsen (administrator) - 2010-02-19 12:16
 https://issues.asterisk.org/view.php?id=16857#c118283 
---------------------------------------------------------------------- 
As per davidw's note, Asterisk 1.2 does not receive bug fixes. If this is
an issue on Asterisk 1.4 or above, please open a new ticket with the
following information attachements added as text files to the ticket:

* SIP trace demonstrating the problem
* Console trace with debug level logging
* Configuration and topology in order to be reproduced or at least
understood by a developer.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-19 12:16 lmadsen        Note Added: 0118283                          
2010-02-19 12:16 lmadsen        Status                   new => closed       
======================================================================




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