[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. 
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http://bugs.digium.com/view.php?id=11489 
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Reported By:                macbrody
Assigned To:                
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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:              
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Date Submitted:             12-07-2007 07:36 CST
Last Modified:              12-20-2007 09:43 CST
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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.

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---------------------------------------------------------------------- 
 macbrody - 12-20-07 09:43  
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ok, checked out /svn/asterisk/trunk, made the changes
and then 'svn diff'.

If you need it against /svn/asterisk/branch/1.4
please let me know. 

Issue History 
Date Modified   Username       Field                    Change               
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12-20-07 09:43  macbrody       Note Added: 0075757                          
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