[asterisk-bugs] [Asterisk 0012245]: [patch] Support for RFC2833 DTMF for dumb SIP proxies

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12245 
====================================================================== 
Reported By:                bamby
Assigned To:                otherwiseguy
====================================================================== 
Project:                    Asterisk
Issue ID:                   12245
Category:                   Channels/NewFeature
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-03-18 03:25 CDT
Last Modified:              2008-12-05 13:48 CST
====================================================================== 
Summary:                    [patch] Support for RFC2833 DTMF for dumb SIP
proxies
Description: 
My recent upgrade from asterisk 1.2 to 1.4 brought in a problem.

Let me describe my setup. Calls from Asterisk are coming through the dumb
SIP proxy that doesn't announce the support for the RFC2833 telephone-event
but relays all the RTP packets regardless of RTP payload type so remote
IVRs can receive the DTMFs.

The Asterisk 1.4 handles this situation more correctly than 1.2, it drops
the telephone-events if peer didn't announce support for them in SDP. But
the problem is that this dumb SIP proxy cannot be neither avoided nor
fixed.

I've added a couple of configuration options that helps to recover the
previous behavior and also allows an administrator to choose the payload
type value for the telephone-event payload.

I believe I'm not alone with this problem so IMO this feature would be
helpful for some people.
====================================================================== 

---------------------------------------------------------------------- 
 (0095885) otherwiseguy (administrator) - 2008-12-05 13:48
 http://bugs.digium.com/view.php?id=12245#c95885 
---------------------------------------------------------------------- 
1.4 has been around for quite a while, and people haven't been cursing this
behavior yet.  I'm not sure if this warrants inclusion or not--especially
since we try to avoid adding new features to 1.4.  I'm also a bit confused
by a "dumb SIP proxy" that is relaying RTP and also must be modifying the
phones' SDP since they should be sending that they understand rfc2833 dtmf.
 This sounds like a pretty specific set of circumstances that I don't think
would affect too many users. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-05 13:48 otherwiseguy   Note Added: 0095885                          
======================================================================