[asterisk-bugs] [Asterisk 0015586]: [patch] Failure to negotiate T.38

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15586 
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Reported By:                globalnetinc
Assigned To:                kpfleming
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Project:                    Asterisk
Issue ID:                   15586
Category:                   Channels/chan_sip/T.38
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     assigned
Target Version:             1.6.2.0
Asterisk Version:           SVN 
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:             2009-07-26 15:57 CDT
Last Modified:              2009-10-05 15:04 CDT
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Summary:                    [patch] Failure to negotiate T.38
Description: 
To implement T.38 on most ATAs their is a reinvite required.  In the
process of gatewaying the T.38 negotiations the Asterisk server is not
doing this correctly.  On versions past 1.6.0.10 it does not even send the
same ports on the RTP streams to both parties.  

Every version past 1.6.0.10 fails
1.6.0.11
1.6.1.0
1.6.1.1
1.6.2.0-rc

This also includes the new T.38 stack that is is being introduced. in the
SVN tree of 1.6.1.1 and 1.6.2.0
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Relationships       ID      Summary
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has duplicate       0015886 T38 udptl.c bufferoverflow
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 (0111874) kpfleming (administrator) - 2009-10-05 15:04
 https://issues.asterisk.org/view.php?id=15586#c111874 
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It's apparent I've not been specific enough with my requests for how to
generate the log information I need, so I'll try again. At this point, the
patch has been committed to all the 1.6.x branches and trunk, so you will
not need any patches to try this. Please follow these steps, in order:

1) Update to the latest 1.6.0 SVN branch code, with no patches applied.
2) Edit /etc/asterisk/logger.conf, and add the line below:

digium => notice,warning,error,verbose,debug

3) Start Asterisk.
4) Type 'core set verbose 10'.
5) Type 'core set debug 10'.
6) Type 'sip set debug on'.
7) Type 'sip show peer myfax', and capture the output of that command (it
will not be sent to the log files).
8) Type 'sip show peer 4065877430', and capture the output of that command
(it will not be sent to the log files).
9) Place the same FAX test call from the '4065877430' peer to 'myfax' as
you did previously.
10) Once the call is complete, type 'core stop now'.
11) Upload the /var/log/asterisk/digium log file, along with the text
captures from steps 7 and 8. There is no need for a packet capture.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-05 15:04 kpfleming      Note Added: 0111874                          
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