[VoIP] My Asterisk switch, etc.

[Steph Kerman]

I agree.  It seems impossible based on the schematics I have seen for 
the hardware to do this.  During dialing, the line towards the station 
and towards the network are isolated from each other by relay contacts, 
making it impossible to hear call progress tones or converse until the 
dialer finishes outpulsing all queued digits and joins the two lines.  
Once that occurs, AFAIK it will no longer actively intervene in a call 
and will not re-split the two lines as required to convert subsequent 
digits.  AFAIK it remains passive until it detects an on-hook condition.

Steph

john jones wrote:
> Can you share the Smart-1 config?   I could never even get close to making mine do that.
>
> Thanks!
>
> John
>
> ----- Original Message ----
> From: Donald Froula <dfroula at sbcglobal.net>
> To: voip at ckts.info
> Sent: Wednesday, September 26, 2007 4:53:42 PM
> Subject: [VoIP] My Asterisk switch, etc.
>
> <snip>
>
> The Smart-1 is used as a simple DP to DTMF converter
> for the WE 211, my "operator" phone (1-762-0000). I
> figured out how to program the Smart-1 to do
> digit-at-a-time pulse-tone conversion.
>
> Best,
>
> Don
>