[Asterisk-Dev] outbound DTMF with RFC2833

[Ed Greenberg]

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>