Without really knowing much about what I was doing I’ve managed to setup a few asterisk boxes (only meetme conferencing) based on *now.
I Guess it’s time I start to know what I’m doing I want to build a test setup with based on a normal asterisk install. I managed (with very little linux knowledge) to setup a debian system, with Asterisk 1.4.21, Zaptel, and the 2.0 GUI.
My system sees the 4 analogue ports, and I managed to have it anwser incomming calls and route them to a meetme conference. So far, so good.
now… I actually have very little understanding of how calls flow through dailplans, or specifically how calls enter the dailplan.
What I want to try to do is make a test setup at home which does the following:
- have 2 incomming analogue lines
- have 2 analogue phones
- have 2 sip phones
- have 1 or more VOIP providers
and to start I would like the system to do:
- when an incomming call comes in, all lines should ring, no menu’s etc.
- if no one anwsers the phone, it should get routed to a voicemail system
- all phones should be allowed to make calls to other internal phones, and outside numbers (routes should be chosen based on patern of the number)
advances options will follow:)
So… where do I start? I have many questions:
- what context do calls from ‘outside’ lines come in on, and why?
- when I pick up the horn of an internal phone, what produces the dial tone?
- when I start dialing outside, do I go through the dialplan? if so, where do I enter it, and why there?
Well… that’s enough for now as I said, I need a mentor