[Asterisk-Users] Speeding up the dial of DTMF's in SIP channel

[Imran Ahmed]

Please Ignore if you cannot edit the code.

You will have to modify app_dial.c in apps directory.
Look for code that calls ast_dtmf_stream(chan, ..., timeout)
The last parameter is the inter digit timeout, it can be set to as low
as 1 (1 millisec) a value of  0 it will default to 100millisecs.
The solution is to add an option to dial application for the timeout
which defaults to the current value(250ms) in app_dial which will
provide for custom timeouts through the dialplan.
Also note that too small timeouts like below 100ms will mess up inband
dtmf tones for example in some zap channels.

Imran

On 3/15/06, Álvaro Palma <apalma at opschile.cl> wrote:
> I'm dialing DTMF's in a SIP channel using the options:
>
> [sip.conf]
> dmtfmode=info
>
> [extensions.conf]
> exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN}))
>
> (this is a custom SIP gateway, which receives the DTMF's sent from
> softphones through Asterisk, and based on them, build the destination
> PSTN number).
>
> My problem is that Dial send the DTMF's to the SIP/gateway user at a
> rate of about 1 DTMF each 300ms. I'd like to know if it's possible to
> speed up that rate, or even, if it's possible to send the entire
> extension as a single DTMF string.
>
> Does anybody has a clue about how to do this? I was looking the options
> for the Dial command, and nothing like that appears on it.
>
> Thanks a lot for your help.
>
> --
> Atly.
> Álvaro Palma
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