[Asterisk-Dev] HELP: DTMF signals screwed up by asterisk

[Steven Critchfield]

On Wed, 2004-09-15 at 08:22, Asterisk Developer wrote:
> hello,
> 
> I tried to use an analog emergency-call device. that device is a one-button 
> device that calls an emergency service like 911, it's for older people or 
> people with disabilities. however, the device dials out to a remote server to 
> login and gather some information. when the device dials the number, the 
> server answers with a 2000hz tone and a DTMF sequence. that DTMF sequence is 
> however totally screwed up by asterisk. each dtmf sequence is 80ms long, but 
> asterisk cuts out 40ms from the sequence. it looks like:
> 
>  *******00000000000000*******
>  0-----20-----40-----60----80---->
>  ms                           
> 
> * = DTMF signal 
> 0 = no DTMF signal
> 
> call direction SIP<->ZAP<->E1 (TE410P). I have a WAV file and a PNG image of 
> the waveform. see http://www.jass.at/asterisk/
> 
> do we have a chance to turn off DTMF detection within zaptel/chan_zap?

You can't pass DTMF inband on the SIP leg if you use any compression.
This means the problem you are experiencing is really in your SIP leg of
the call. Check your SIP device to see if it will let you change the
length of the DTMF tones.
-- 
Steven Critchfield <critch at basesys.com>