Noob looking for mentor... bunch of beginner questions

Hi…

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 :smile: 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:

  1. what context do calls from ‘outside’ lines come in on, and why?
  2. when I pick up the horn of an internal phone, what produces the dial tone?
  3. 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 :smile: as I said, I need a mentor :smile:

No one able or willing to help me on my way?

To answer your questions.

  1. what context do calls from ‘outside’ lines come in on, and why?

If the calls are SIP, they come into the context specified on the peer in sip.conf. If the peer is not defined or could not be found, it will go to the context specified in the context parameter in the [general] section.
If they are PSTN lines coming in on hardware, it is the context specified in zapata.conf

  1. when I pick up the horn of an internal phone, what produces the dial tone?

Dialtone is generated locally on the phone. Or in the case of an analog phone connecting to an ATA, the ATA is generating the tones.

  1. when I start dialing outside, do I go through the dialplan? if so, where do I enter it, and why there?

Yes, when dial a number on an extension to the pbx, it sends the call in the context defined for the device to determine how to route the call. It is set in extensions.conf.

Hi,

Sorry for net getting back sooner, I was away for a while…

I understand the places calls enter the dialplan, thank you!

I do get new questions though.

I used the GUI to generate some sample rules for me. I do not understad the follow:

I my [outgoing] context I have the rule:
exten = 1001,1,Goto(default|1001|1)

My default context is empty.

Why does this make the sip phone on 1001 ring, when I dial 1001 on a second phone? It seems like I give a goto command, and there is nothing there? still it works???

Do

show dialplan default

that will show you whats in it