[Asterisk-Users] Sip to PSTN Gateway Configs

[bob mc]

Hi,

I'm trying to put together a simple gateway
configuration involving Asterisk. I have a machine
with 2 Digium X100P FXO cards installed and the
Asterisk Software, and I have 2 Sip Phones defined.

What I want to achieve is, any call arriving at FXO 1
is forwarded to Sip phone 1 only, and any call
received on FXO 2 is transferred to Sip Phone 2,
conversely any call originating from Sip Phone 1 goes
out of FXO 1, and any call originating from Sip Phone
2 goes out of FXO 2.

Some example .conf files would be greatly appreciated.

Thanks




	
	
		
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