[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 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-08-09 17:00 CDT
Last Modified:              2008-12-08 11:43 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.
====================================================================== 

---------------------------------------------------------------------- 
 (0095961) ptimmins (reporter) - 2008-12-08 11:43
 http://bugs.digium.com/view.php?id=13265#c95961 
---------------------------------------------------------------------- 
I'll need a couple of weeks as I can better experiment with this in our lab
and this bug was reported when we were experiencing the problem in
production before we got the lab up. But I'll need a week or so to get the
lab connected to the carrier with the sonus that was causing the issue.
Just didn't want this to be closed for lack of response. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-08 11:43 ptimmins       Note Added: 0095961                          
======================================================================