Help getting started -- sample configuration?

So far I have my X100P (FXO card) accepting calls from an external PSTN line and playing the Asterisk demo message after 1 ring. The card is channel Zap/1. I also have one soft SIP client that is able to dial the console using the demo extension “1234”. But now I would like to bridge those two so I can proxy the SIP phone to the outside line.

I am still learning a lot about Asterisk, but I need a small favor please… Can someone give me a sample configuration, the most basic configuration you can think of with zapata.conf, extensions.conf, sip.conf, and possibly voicemail.conf, to achieve the following simple specification (or at least advise me):

When the outside PSTN line (Zap/1) rings, Asterisk will not answer it but will ring the SIP client. If the SIP client answers, Asterisk answers the PSTN line and connects it to the SIP client and initiates conversation. If the SIP client does not answer but the line is answered by another regular phone directly connected to the line, Asterisk forgets about it. Otherwise, if more than, say, 3 rings go unanswered, Asterisk will pick up the line and play the voicemail or demo menu (or at least some simple message; doesn’t matter for now).

Also, if the SIP client picks up and dials 9, Asterisk connects it to the outside PSTN line (Zap/1) and allows making an external call (or accepts a number then initiates connection with the PSTN line, whatever).

If I can get at least that far, I can at least start really using the Asterisk server and take it from there to do more. Thanks!

at the top of the forum is a sticky from muppetmaster with a link to a free book … read that, try your own config files, then if it’s not working as you expected, post them back here for more help.

it might sound brutal, but you’re not going to learn anything if it’s all done for you. the examples are there in the book.