[asterisk-bugs] [Asterisk 0016973]: multiple values for "dtmfmode=" per trunk

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=16973 
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Reported By:                AndrewZ
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16973
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   feature
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.17 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-03-05 13:58 CST
Last Modified:              2010-03-05 13:58 CST
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Summary:                    multiple values for "dtmfmode=" per trunk
Description: 
Currently it's possible to configure only a single DTMF transmission method
per SIP trunk by setting "dtmfmode=" to rfc2833 or info or inband, but not
to combination of them. We can also use a dialplan command
SIPDtmfMode(inband|info|rfc2833) to change the method "on the fly". Both
those ways are blind or static in terms of tracking the other side
capabilities.
The problem appears when we're sending the calls to a service provider who
is sending the calls further to the third-party gateways or other carriers
for termination and those carriers/gateways are not supporting the same
DTMF processing methods.

Suggested Asterisk behavior:
Multiple values per "dtmfmode=" line within the peer configuration,
similar to the "allow=" for codecs, something like:
dtmfmode=rfc2833&info

On outgoing SIP call Asterisk should advertise it's preferred (the 1st
configured) method in SDP by putting there something like this:
(m): audio 11234 RTP/AVP 0 101
(a): rtpmap:101 telephone-event/8000

If it will be possible to negotiated this with the other side - then this
method will be used, means we will send DTMF according to RFC2833.
If rfc2833 could not be negotiated then we will check for
"application/dtmf-relay" in "Accept:" line in the response. If it's there -
we will send DTMF as SIP INFO. If not - we may fallback to inband.
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Issue History 
Date Modified    Username       Field                    Change               
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2010-03-05 13:58 AndrewZ        New Issue                                    
2010-03-05 13:58 AndrewZ        Asterisk Version          => 1.6.1.17        
2010-03-05 13:58 AndrewZ        Regression                => No              
2010-03-05 13:58 AndrewZ        SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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