[asterisk-dev] Zaptel DTMF regeneration

[Tony Mountifield]

In article <61575c810809170759q58a0afafw500e72c871377024 at mail.gmail.com>,
Matt Florell <astmattf at gmail.com> wrote:
> On 9/16/08, Moises Silva <moises.silva at gmail.com> wrote:
> > Take your time to read the code. The places I pointed out to you are
> >  the correct ones, at least for detection of DTMF, the routine
> >  ast_dsp_process takes care of that. If you want to see the very
> >  working of the Goertzel algorithm used you could have found so by
> >  checking that routine and seeing that calls dtmf_detect which does the
> >  actual DTMF loop detection.
> >
> >  As of the audio mute, not sure what you mean?
> 
> Not sure how much this helps, but in Meetme if there is a zap channel
> and an IAX or SIP channel then the IAX or SIP channel will not hear
> the audio tones that the zap channel is sending, but it will be
> translated into DTMF signalling out-of-band.
> 
> I do not know of any way to make the audio on the IAX/SIP channel
> "hear" the true DTMF audio within a meetme conference without
> completely breaking DTMF detection in there as well.

The dsp routines convert DTMF tones detected on zap channels (and
inband VoIP channels) into asterisk DTMF frames, and at the same time
mute the audio to suppress the inband tones.

If the Meetme has implemented the F option and is using it, it will
pass on these DTMF frames to the listening channels.

If those channels have been configured with dtmfmode=inband, I assume
Asterisk will regenerate the DTMF audio from those frames. However,
I don't think that behaviour is settable per-call, is it?

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