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

[noreply at bugs.digium.com]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=12245 
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Reported By:                bamby
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12245
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     new
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:              
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Date Submitted:             03-18-2008 03:25 CDT
Last Modified:              03-18-2008 03:25 CDT
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Summary:                    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.
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Issue History 
Date Modified   Username       Field                    Change               
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03-18-08 03:25  bamby          Asterisk Version          => 1.4.18          
03-18-08 03:25  bamby          SVN Branch (only for SVN checkou => N/A          
  
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