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

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Dec 12 02:00:46 CST 2007


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:              12-12-2007 02:00 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 - 12-12-07 02:00  
---------------------------------------------------------------------- 
file,

I understand that. Let's consider the situation as is:

we fixed a bug but thereby broke rfc2833 interoperability with cisco
equipment.
Currently (1.4.15+) only inband dtmf is possible with cisco equipment ->
which is not usable in large environments.

Would one of the following options be acceptable?

1) A config option for manually setting/forcing the usage of rfc2833
wheter the
   other side offers it or not.
2) Accepting the X-NSE parameter as equal to the telephone-event
parameter.

In asterisk-1.4.15+ the option dtmfmode=rfc2833 could be deprecated anyway
as it is no longer used if I understand that correctly. Instead
dtmfmode=auto does the same job.

To keep rfc2833 interoperability to cisco equipment what way do you
suggest
is acceptable? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-12-07 02:00  macbrody       Note Added: 0075251                          
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