Asterisk SIP / LAN Question

Okay, my setup is a Digium X100P/X101P analogue PSTN FXO interface running on slack. I’m using a plain old telephone line that just connects to the asterisk box(which is in my living room). I’d like to have a phone in my bedroom that connects via my lan (can be a wireless phone, or usb hard phone that connects to a machine in my bedroom) to the asterisk box, and does all its business via the PSTN. Is this possible? What sort of phone do I need (SIP)? Any help / advice is appreciated.

Add a sip entry in /etc/asterisk/sip.conf

Something like

callerid=“502” <502>

Enter the registration info to your sip (soft)phone

At the extension in extensions.conf that “catch” the incoming call add something like

exten => ,1,Answer()
exten => ,2,Dial(SIP/502,20,rt)

And when a call comes in the (soft)phone will ring.

In the [outbound] context there has to be a line that matches with the number you call and uses. The easies way is to do this numbermatching

exten => _9X.,1,Dial(ZAP/1/${EXTEN:1},20,rt) ; 9 as prefix for outgoing line

If you start on … sions.conf you will find a starting point for reading lots of usefull info