[Asterisk-Dev] DTMF Pass through (Transparently)

[Chris Lee]

Kevin P. Fleming wrote:
> Chris Lee wrote:
> 
>> It may have been me, I have asked about similar features before, though
>> I think on -users. My app does not require exact timing though it does
>> use the approximate length of a DTMF tone to adjust pump speeds (2 = 
>> faster, 8 = slower).
>> Makes site visit engineers jobs easier and gives a paper trail of who 
>> did what when.
> 
> 
> As best we can tell, unless you have a VPM module on a TE4XXP board, it 
> should _already_ be working this way for Zap<->Zap bridges. In that 
> case, DTMF processing is completely bypassed, and it's left in the audio 
> stream untouched.
> 
> Is your environment involving something other than Zap<->Zap bridges?

Unfortunately we have moved to a windows CAPI/TAPI solution;
We originally planned to have Asterisk centrally sending SIP to each 
site and running the app there, but I could not make it work.
We then moved to a ISDN line locally with Asterisk passing the call to 
an AGI app, this exceeded my skills as a programmer.
So we called in a coder who knew Windows TAPI and CAPI and thats where 
we ended up.
This was all way back, beginning of last year, so maybe things are 
easier now.

Still use asterisk for normal telephony though.

Chris.