[Asterisk-Dev] outbound DTMF with RFC2833

[Dan Evans]

Take a look at the history of why Asterisk does what it does with 
outbound DTMF in patch 0003339.

Dan

Ed Greenberg wrote:
> Back in June, there was a thread called "outbound RFC2833 broken. Use 
> inband instead."
> 
> I also have had problems with outbound DTMF. I submitted some traces to 
> Level 3 and my engineer stated that RFC2833 DTMF, as generated by 
> Asterisk, was being discarded by the Level 3 media gateway since the 
> timing and duration was wrong.
> 
> Asterisk sends one set of RFC2833 packets in the rtp stream to start the 
> tone, then immediately sends another set of packets to end the tone. The 
> packets carry a duration of 800.
> 
> Level 3 stated that they expected a packet every 20 ms, in accordance 
> with the "agreement" made when, in sip, the rtp stream was negotiated.  
> The engineer also provided a packet capture showing a "good" session.
> 
> I rewrote ast_rtp_senddigit to send tones in accordance with Level 3's 
> requirements.  I plan to test with VoipJet as well, since the original 
> problem existed there too.
> 
> Is there anybody who is having outbound DTMF problems who would like to 
> test this code against either an ATA or some other providers? I'll be 
> happy to share it.
> 
> Although I've been programming in C for many years, I've never submitted 
> a patch to an open source project. If there is consensus that this 
> should be submitted, I'll need a bit of help (or a document to read) 
> telling me what is expected of me to do the submission.
> 
> Thanks,
> </edg>
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