[Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP extension *before* answering the PSTN line?

[Stephen Bosch]

Time Bandit wrote:
>> > [incoming]
>> > ; incoming calls from the FXO port are directed to this context from
>> zapata.conf
>> >
>> > exten => s,1,Answer()
>> > exten => s,2,Dial(SIP/polycom)
> 
> Try this
> 
> exten => s,1,Dial(SIP/polycom,20)
> exten => s,2,Hangup()
> 
> I think this way, * won't answer the line until your SIP phone
> answers. If you don't pickup the phone after 20 seconds it will just
> ignore this incoming call.

Hmn. Very nice! It works!

On the matter of timing --

Asterisk appears to wait two full PSTN rings before it dials the SIP
extension. Is there any way we can tighten up this interval? Is that
done in the Zap configuration? The driver? The dialplan?

Cheers,

-Stephen-