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

[Colin Anderson]

the caller is out his money anyway when you call any phone and voicemail
kicks in, although i think on a payphone they give you a 2 or 3 second
window to hang up. 

Suggest you implement i'm here / i'm away dialplan logic or set the do not
disturb button that way when someone calls and the guy is away it hits
voicemail right away and the caller can hear this and still have the 2 or 3
second window to hang up  and get his $$ back. This emulates PSTN behavior
as close as possible but you have to train your users to hit the DnD button
when they walk away from the phonw. 

-----Original Message-----
From: Stephen Bosch [mailto:posting at vodacomm.ca]
Sent: Monday, June 12, 2006 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] TDM-400 and dialplan -- how to ring a SIP
extension *before* answering the PSTN line?


Hi, folks:

Okay, so here's an idea.

I have a TDM-400 card with an FXO card in it connected to the PSTN and a
Polycom IP 501 phone.

Observe the following simple dialplan for illustration:

> [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)

And zapata.conf:

> [trunkgroups]
> ; define any trunk groups
> 
> [channels]
> ; hardware channels
> ; default
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echotraining=yes
> callprogress=yes
> 
> ; define channels
> context=incoming
> signalling=fxs_ks
> channel => 4

Pretty straightforward stuff -- a call comes in on the PSTN line, the
Asterisk answers the call, then rings the extension. The caller hears a
ring tone throughout the entire process.

The rub is that Asterisk has, in reality, taken the PSTN line off hook.
Not great if the caller is at a payphone. What if nobody answers the
extension? The caller is out his money (50 cents in most of the US, 35
cents in Alberta and 25 cents in the rest of Canada ;) )

So I had the idea of doing things a bit differently, like so:

> [incoming]
> ; incoming calls from the FXO port are directed to this context from
zapata.conf
> 
> exten => s,1,Dial(SIP/polycom)
> exten => s,2,Answer()

This way, Asterisk dials the extension first, the idea being that when
the SIP extension is answered, Asterisk answers the PSTN line and
connects the channels.

This did not have the expected result -- when I tried this, my SIP
extension rang, but answering the extension did not result in Asterisk
picking up the PSTN line.

There is a way of doing this, isn't there? How can I make this work?

Cheers,

-Stephen-

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