[Asterisk-Users] Sip to PSTN Gateway Configs

[Karl Brose]

Bob,
What I am going to tell you may seem arrogant or what, but I think
you would do yourself a great favor if you figured this one out yourself
by studying the info that is available and ask questions if things don't
work.  Your configuration is indeed very simple and with 100% certainty
you will want more once you have it, and you will come back for
more sample configurations, but haven't learned the basics to simply
build on what you got.
This response is really addressed to all that are starting out with 
Asterisk.
With Asterisk is pays well (in terms of satisfaction and total time 
invested)
to spend--no INVEST--some startup time to learn the basics, starting
very simple and build up a system and dial plan step by step.
I am seeing over and over here and other boards that starters want
everything right now in their first configuration and therefore download
all kinds of configuration files from the web and are confused by it all,
because nothing fits together, and the simple principles are lost  among the
complexity.
If one follows that advise, Asterisk is really very simple to set up.



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.
>
>Some example .conf files would be greatly appreciated.
>
>Thanks
>
>
>
>
>	
>	
>		
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