Help needed - very basic setup!


I’ve managed to install Asterisk 1.4.5 OK and now want to set up a very simple system - all internal connections on a LAN.

I’m not at all familiar with the terminology used :confused: so… which of the .conf files do I need to edit and can someone give me examples as I can’t understand the built-in example files.

I’m trying to use X-Lite softphones and could do with info on the settings required here too !

Each of my PCs has a static IP address, as does my Asterisk server.

Many thanks


Hi Mike,

You will need to make one entry per ‘phone’ in your sip.conf for each of the xlite clients.

Then you will need to make a basic dial plan to allow each phone to call one another. (in extensions.conf).

you then configure the basic sip settings in your xlite clients to match with what you put in the sip.conf.

Maybe you could do some reading / trying then come back with particular problems? is a great site.



Thanks Seabro,

I’ve tried reading until my head hurt :frowning: It’s all new to me an the terminology baffles me. I’ve been through some of the example files included with the installation too.

Strangely… I first installed Asterisk last week, tweaked the files you’ve mentioned and it seemed to work. I decided to try AsteriskNOW as it has a GUI for configuration. That confused me more and I couldn’t get anything working. So, I’ve gone back and reinstalled the ordinary Asterisk and tweaked what I thought I’d done previously but it’s not working.

I’ll go back to tweaking, trying, tweaking, trying… and if I still fail I’ll try to nail down something more specific and ask.


Mike (in Llandrndod)

Hello , here is a minimal configuration that let you do call from one sip phone to an other :

  1. just add this two context at the end of your /etc/asterisk/sip.conf file:



  1. and add this simple dialplan at the end of your /etc/asterisk/extensions.conf file :

exten => 101,1,Dial(SIP/sipuser1,)
exten => 102,1,Dial(SIP/sipuser2,)

In the first x-lite client fill like this :
username : sipuser1
domain : “@ ip address of your asterisk server”

in the second just remplace username by : sipuser2 and keep the same ip address.
you can do call from sipuser1 to sipuser2 by dialing 102 and vice versa (101).

good luck man !

Thanks Sara,

It worked, I used exactly what you suggested :smiley:

I couldn’t get the X-Lite phones to connect so I tried a Firefly. Setting that one up seemed so much easier. Only problem is that the Firefly phones seem to crash when the callee hangs up :confused:

Thanks again.


To avoid the crash when the callee hangs up , add this lines to the context [test] in your extensions.conf file ; so you will have somethings like :

exten => 101,1,Dial(SIP/sipuser1,)
[color=red]exten => 101,2,Hangup()
exten => 102,1,Dial(SIP/sipuser2,)
[color=red]exten => 102,2,Hangup()[/color]

Thanks again Sara,

It didn’t stop the Firefly crashing :frowning: But I have now managed to get the X-Lite phones working OK and I do prefer them.