[asterisk-bugs] [Asterisk 0013265]: RFC2833 mangled from Sonus when RTP stream passes through asterisk rather than being reinvited

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13265 
====================================================================== 
Reported By:                ptimmins
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13265
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.21.2 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2008-08-09 17:00 CDT
Last Modified:              2009-02-05 12:07 CST
====================================================================== 
Summary:                    RFC2833 mangled from Sonus when RTP stream passes
through asterisk rather than being reinvited
Description: 
We were/are using 1.4.11. We connect to Global Crossing over SIP. They use
Sonus softswitches. We use ulaw and rfc2833. We also use Cisco AS5400
gateways and Adtran Total Access 900 IADs. Under 1.4.11, RFC2833 DTMF
passed through asterisk was properly replicated. When we upgraded
(directly) to 1.4.21.2, asterisk began manipulating the DTMF such that when
it went from Global, through asterisk, to the Cisco or Adtran gateways, the
touchtones were mangled to the point they simply make a chirping silence
noise. If canreinvite=yes is set, they work fine. However, that Global
doesn't allow you to reinvite to arbitrary destinations, so this won't work
as a long term solution. Attached is a copy of a working SIP+RTP stream
going from Global to Asterisk, and from Asterisk to the as5400, as well as
a copy of a failing one, as well as the dtmf debug.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0013209 DTMF RFC2833 via SIP is not working
====================================================================== 

---------------------------------------------------------------------- 
 (0099511) ptimmins (reporter) - 2009-02-05 12:07
 http://bugs.digium.com/view.php?id=13265#c99511 
---------------------------------------------------------------------- 
The weird part is that it works perfectly under 1.4.11 (where I'm currently
stuck). Of course, it crashes randomly, and it might be related (often I
get strings of:

[Dec 23 12:59:15] WARNING[17311] app.c: No audio available on
SIP/REDACTED-b4f7e2f8??
) just before it stops responding. There's a general bug I've been finding
in 1.4.11 where the sip stack hangs and it still sends RTP. I wonder if
it's related.

Either way it's 1.4.11 and sucks implicitly. But it's weird that it works
here. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-05 12:07 ptimmins       Note Added: 0099511                          
======================================================================