[asterisk-bugs] [Asterisk 0011489]: During call setup signalling asterisk does not offer telephone-event (rfc2833) anymore

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11489 
====================================================================== 
Reported By:                macbrody
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11489
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.15  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!): 91736 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             12-07-2007 07:36 CST
Last Modified:              12-08-2007 12:14 CST
====================================================================== 
Summary:                    During call setup signalling asterisk does not offer
telephone-event (rfc2833) anymore
Description: 
What happened:
As of asterisk-1.4.15 (tested trunk too) asterisk, when
connected to a peer via sip trunk does not recognize dtmf
anymore (same config worked with 1.4.13).

What I found:
During signalling negotiation at call setup in the sdp body,
asterisk does not offer the rtpmap: telephone-event anymore,
even if I configure dtmfmode=rfc2833.

Therefore the other end will not send any rfc2833 specific rtp events.

====================================================================== 

---------------------------------------------------------------------- 
 eliel - 12-08-07 12:14  
---------------------------------------------------------------------- 
Could you upload the SIP debug trace of 1.4.13 in the same scenario that
you say is working fine?
What I see here is that the Cisco is not doing the INVITE with the RFC2833
capability:
[Dec  7 16:54:05] VERBOSE[7209] logger.c: Non-codec capabilities (dtmf):
us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)

Thanks 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-08-07 12:14  eliel          Note Added: 0075088                          
======================================================================