[asterisk-dev] Passing DTMF through MeetMe

[Imran Ahmed]

On 9/27/06, Tony Mountifield <tony at softins.clara.co.uk> wrote:
> 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.
>

Tony,

If you plan to convert dtmf frames to inband audio, it will not work
with other channels in the conference that are not set to inband dtmf
mode, since the dsp code does not work on the outgoing stream to
detect the tone and convert to the appropriate mode.
I suggest queuing dtmf frames to all channels in a conference like a broadcast
Hope that helps.

Imran