Setting Up a VoIP Lab

OK, you've tested out a few IP phones or set up a softphone and made calls using a VoIP provider.

What Next?

The next thing you should explore is setting up your own PBX. Maybe you have in mind replacing an existing Office PBX, or having several extensions at home and handling calls more slickly. Either way there is one piece of software which coupled with the right hardware provides a comprehensive PBX solution, and that is Asterisk.

Obviously you don't rip out an established system without trying it yourself and making sure it works for you. So here goes with your first Asterisk experiment! It should take an hour or two to set everything up and make your first calls.

You will need a machine to act as a server. This doesn't have to be very powerful, for testing purposes a Pentium III processor will do with at least 256 Mb of RAM. The machine needs a 10 gigabyte hard drive, a CD drive and an ethernet interface. Be warned that the entire contents of the hard drive will be destroyed!!

You will need at least two phones, these can be any combination of IP phones, traditional analogue phones with an ATA, like the Linksys PAP2T, or softphones on a suitable laptop PC with microphone headsets.

To link everything together you will need a small router and patch leads.

Asterisk comes bundled in a distribution called Trixbox which is available as a downloadable iso image or you can get it from the Emporium here. Trixbox provides an automated installation of Asterisk, the FreePBX Management Tool, a flash operator panel and the SugarCRM contact management software.

If you plan on using hard phones you will need a networked machine with a browser in order to make changes to the Asterisk/Trixbox set-up.

Trixbox uses the CentOS GNU/Linux distribution. If you aren't familiar with Linux you shouldn't worry too much as most of the administration of Asterisk can be accomplished from FreePBX.

Installing Trixbox.

All you have to do to install Trixbox is to boot from the CD and after you have responded to questions about your keyboard timezone and password, the process is completely automated.

After a final reboot you will be presented with a login prompt. The default password for the root user is "password". You should change this to something more secure by logging in and typing the command passwd at the # prompt and follow the prompts to create a new password.

By default the server will use dhcp to get a network address if it can. You will probably want to set a static address with the netconfig command, which will give you screens to enter the network details. Then type in

# /etc/init.d/network restart

to make the changes effective.

Set up phones.

Now is the time to connect up your phones or client PC's and the server to the router and configure IP addresses so that everything is on the same sub-net. On most SIP phones you can make a direct call to another phone just by entering the ip address from the keypad - so why not try it!!

Use FreePBX to configure extensions.

Use a web browser on a test machine and enter http://[the address of the server]. You should immediately see the FreePBX Web management page. Click on "Asterisk Management Portal (AMP)" to log in. The default AMP user is "maint", and the default password is "password".

Click the Setup Tab. Take a quick look at the General Settings. At this stage you might want to change the country.

Now move on to setting up extensions for the two test phones. Click the Extensions button, then select SIP. For the first phone set the extension number to 201 (say), and use the display name of test1, set secret to 1234, enable voicemail and set the voicemail password to 1234.

Remember to press the 'Submit' button. A large red bar will appear at the top of the screen - with the advice that you should click it to apply the changes that you have made. When you do this the Asterisk server is restarted so that the changes take immediate effect.

Make similar changes for the second phone.

Try it!!

Your phones should now be registered with the server. Try calling the other extension and leaving voice mail. Then retrieve your voicemail with *98.

What Next??

Full configuration of Trixbox is a lengthy process. A good introduction to Trixbox is the book: "Trixbox made Easy", which will take you through setting up the Digital Receptionist, Ring Groups and Incoming Calls.