[Asterisk-Users] Sip to PSTN Gateway Config

[Anon]

On Sunday 09 May 2004 02:32 pm, bob mc wrote:
> 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.

You would be doing yourself a great favor by making your first configuration  
VERY simple: 1 FXO and 1 phone.  Walk before you start to run.  This method 
does delay immediate gratification, yet gets one to become a champion 
sprinter much faster in the long run.

Also, your answer is definitely in the archives.  Use 
http://www.google.com/custom?sitesearch=lists.digium.com and 
http://www.voip-info.org/tiki-index.php?page=Asterisk

These are very good resources that keep getting even better.  All newbies need 
to invest some time at these resources.

Anon