Dec 3

My Trixbox adventures

Let’s start with the disclaimer, I am not saying that VOIP is the end all fix to everything. It is a useful tool for reducing cost or improving connectivity. What I am going to cover is the easiest (and hopefully cheapest) path between a new user and the ability to actually test a VOIP system.

What you are going to need:
A spare computer to act as the VOIP server with an ethernet port.
Minimal knowledge of linux command line, remember, when in doubt, google
trixbox 2.4, we will be going over how to get the trixbox 2.4 software.
Access to your router and internet device, namely the ability to open incoming ports.

The first thing we are going to do is download trixbox 2.4, this step is less trivial than one would think, but has been made easy by the availibility of the .iso on this site. The trixbox site only hosts the latest version (2.6 at the time of this writing) which gave me a lot of bugs upon first install.

Download the iso from either this site or another site.

Install trixbox 2.4 on the server, the details of this step are outside the scope of this writing. But can be found on the trixbox website.

Once installed, the trixbox server will be set to an automatic address on your local network, the issue with this is that getting to the web interface is more difficult. If you have a IP scanner on your computer, use it to find the IP of the trixbox server. If you still cannot find the IP, I suggest googling the commands for looking up the IP address directly from the server.

Now that we have found the ip address of the server, use your favorite web browser to access the server. Type the ip address of the server directly into the browser address. (i.e.

The web interface will give you a screen that has a link in the upper right corner called “switch”, click this and you will be prompted for a username and password. The username is ‘maint’ and the password is ‘password’. Enter these values and click OK. You should change these to prevent hacking.

Close the registration window. You are now logged in as the administrator, look for a link called ‘Asterisk’ and click on ‘FreePBX’.

This will take you to the FreePBX administaration screen. Here we can create the user accounts/extentions and configure the SIP Truck which handles calls to the outside world.

We will first start by setting up an extention in FreePBX. Click extensions, then click submit. This will set up a new SIP extension, which is what your computer will use to connect to the trixbox server. Give the first user an extension number (201 works good) to start, this is the number you use to dial this user internally. Next select give the account a display name and a password in the ‘secret’ field. For a basic extention this is all we need so click submit. Add at least one more for interoffice testing.

We now have the server set up for basic interoffice communication. Time to set up some user computers.

download X-lite from , Install it, run it. Under SIP account settings, add a new account. In the account properties, create a display name, this will be the name that will show up on caller ID. The username will be your extension, the password will be the value you entered for ‘secret’ and the domain will be the IP address of the trixbox server. click the register with domain and receive incoming calls checkbox and select domian. then click OK.

If you have done everything correct, Xlite should discover the network and prompt you saying ready and listing your extension. You can now make calls to and from extensions on the local network. Try dialing another extension you have set up and see how it works.

Next step is to add the ability to call outside your local network, this includes dialing out, and with the right customer plan, dialing in. So far we have not actually signed up for any service, all we have done is install server software and played with it to allow dave in accounting to talk to jill in marketing using their computers.

We need to get a service provider for incoming/outgoing. This tutorial will be using voicepulse so from here on all SIP trunking will be related to setting up service with voicepulse. If you do not like them, have a great time on your own. I spent two weeks stuck using vitelity and hated them. So I use voicepulse.

First, we need to sign up for an account with VP. Go to , use Trixbox CE as your PBX. Set up an account, they will give you information on how to download the voicepulse module.

Once you have downloaded the voicepulse module, get back into FreePBX, then click Tools, then Module Admin. Next click Upload module. Next find the file you downloaded and select it to be installed.

Once installed, enable the module. Once enabled, a voicepulse menu should show up in the left hand menu when you click setup. Start by clicking status, this should give you a basic idea of what they are doing for you. Next click troubleshooting and find what is and is not working. There should be at least one thing not working.

When I installed my trixbox, it said I needed to install the curl library. To do this, you need to login using SSH to the server. once logged in, follow the directions given to you by the module. (i.e. ‘yum install curl….’)

once this was installed. It all ran smoothly. I was able to make calls. This meant anywhere in the world as long as I was willing to pay. I did add the ability to block 411 calls and a dummy least cost routing mechanism. Otherwise, that is about it.

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