[asterisk-dev] Passing DTMF through MeetMe

[Tony Mountifield]

I have a system in the field which uses MeetMe to conference parties,
all of whom are connected via Zap channels on a T1 PRI.

One of the important features the client uses is the ability to press
DTMF tones on their phone and for the tones to pass out to the other
party(ies). This is for outgoing calls made using a control screen, that
arrive on an IVR at the called party. Since MeetMe is invoked without
any menus enabled, the AST_OPTION_TONE_VERIFY option is not set, and
the Zap channel allows the in-band DTMF to pass through.

I now need to replicate this functionality for SIP channels (and possibly
IAX too), where the DTMF arrives at Asterisk already out-of-band
(e.g. SIP INFO or RFC2833).

I can think of several ways to achieve this, but am not sure which is
the most correct or achievable:
- Take the incoming AST_FRAME_DTMF frames in MeetMe and queue them up
  for output on all other channels in the conference. This assumes they
  would then be converted back to inband somewhere between Meetme and
  the remote party. (what happens if you queue an AST_FRAME_DTMF on a
  Zap channel?)
- Somehow act on incoming AST_FRAME_DTMF frames by generating the DTMF
  tones and playing them into the conference.
- Have the tones converted from out-of-band to inband audio by the
  incoming channel driver (chan_sip or chan_iax).

Any pointers on the best way to approach this would be gratefully received.

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org