I have an analog 2 FXS 2 FXO Digium card.
It lights up all four ports and I have the power supply connection
connected.
The cheap corded phone I plugged into one of the FXS ports is powered.
I want to get a dialtone and I want to integrate this handset with the
2 Linksys PAP2's I have talking to each other.

;!
;! Automatically generated configuration file
;! Filename: sip.conf (/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Sat Mar  5 01:13:34 2011
;!
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
subscribecontext = default

[2000]; Eventually line 1 in tack room.
type = friend
context = my-phones
secret = 1234
host = dynamic

[2001]; Eventually line 2 in tack room.
type = friend
context = my-phones
secret = 1234
host = dynamic

[2002]; Line 1 in the well house.
type = friend
context = my-phones
secret = 1234
host = dynamic


[2003]; Line 2 in the well house.
type = friend
context = my-phones
secret = 1234
host = dynamic

The pasted information above is the content in sip.conf.  What do I add
for the FXS ports on the Digium card connected to the Asterisk server?
What file do I program on the Asterisk server to give the FXS ports
say extensions 2004 and 2005?  Two phone lines are connected to the
FXO ports, what do I tweak to work with those?  First I want to
establish a completely private phone system, calling out is something
for later and I may want to do things like block long distance,
etcetera.

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