[asterisk-dev] Zaptel DTMF regeneration

[Matt Florell]

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.

Are you using a conferencing/meetme application?

MATT---

>  On Mon, Sep 15, 2008 at 6:07 PM, drtester at netzero.com
>  <drtester at netzero.com> wrote:
>  >
>  >
>  > -- "Moises Silva" <moises.silva at gmail.com> wrote:
>  > I assume by "regenerate" DTMF tones you mean pass them along to the
>  > Asterisk core. If so, check channels/chan_dahdi.c dahdi_read()
>  > routine, and there search for ast_dsp_process(). If you have Asterisk
>  > 1.4 then channels/chan_zap.c and zt_read() routine.
>  >
>  > Well, specifically, I mean where the code:
>  >  (a) detects the tones, and
>  >  (b) mutes the audio for the duration of the tones
>  >
>  > Are these two in different places?
>  >
>  >
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