[Asterisk-Users] Why use 'Answer'?

[Rich Adamson]

Inline...

> > As in many cases with *, there are usually multiple ways to accomplish
> > a task. Here's a couple that you'll need to tailor to your environment.
> > 
> >  [in5100]
> >  exten => s,1,Dial(SIP/sip1,20,tr)
> > 
> > The above assumes the pstn line is _not_ sending any digits to you. If
> > it does, then replace "s" with whatever they are sending to you.
> >
> If the PSTN line is just an analog line connected to the TDM400P FXO
> interface, is there any way it will be sending me digits?

Not normally, however in past years telco's would provide DID over
about any type of loop. If you didn't order DID or anything special,
then you are not going to receive any digits from the telco.

> > Or, if you don't want to maintain multiple instances of the Dial command,
> > then do something like:
> > 
> >  [in5100]
> >  exten => s,1,Goto(local-extns,sip1,1)
> > 
> > The above assumes your sip phone dialplan reside in the [local-extns]
> > context, that sip1 is a valid extn number within that context, and "1" 
> > is the first priority in that sip phone definition. 
> >
> so I would need something like:
> [local-extns]
> exten => sip1,1,Dial(SIP/sip1,,tr)
> 
> ??
>  
> > When someone answers the sip phone, the zap pstn line will be answered.
> > If no one answers the sip phone, the "20" second timer will expire and
> > ordinarily it would jump to the next priority. So, you probably want to
> > remove the timeout altogether from that exten definition and just let 
> > it ring forever.
> >
> If I want to timeout to voicemail then what would the
> line(s) in [local-extns] look like?

Something like:
 [in5100]
 exten => sip1,1,Dial(SIP/sip1,20,tr)
 exten => sip1,2,Voicemail(usip1)
 exten => sip1,102,Voicemail(bsip1)
 exten => sip1,103,Hangup