Celliax.Org

So, you want Celliax on the last Trixbox Beta, with Asterisk 1.4, FreePBX 2.3, and all the new bell and whistles.

OK, it's very easy.

Let's begin:

1. Download, burn, and install the 2.3.0.4 ISO image from Trixbox.org Downloads (the file is called trixbox-2.3.2.64-i386-DVD.iso but is a 600MB CD image)
2. When installation of Trixbox is completed, login into the Trixbox machine (at the keyboard, or via ssh, it's the same) as root
3. Now is the moment to download and run one of the scripts to install Celliax on your Trixbox. That script comes in two flavors, the results are exactly the same for both flavors, you need only one of them.

    

   -- If you want to add Celliax binary to Trixbox (the script will download 2MB) execute the following three lines:
   
wget http://www.celliax.org/celliax-on-trix-2.3.0.4-only-binary
chmod +x celliax-on-trix-2.3.0.4-only-binary
./celliax-on-trix-2.3.0.4-only-binary


   -- If you like better to compile Celliax on your Trixbox machine, installing all the Development Tools (the script will download 115MB) execute instead:
   
wget http://www.celliax.org/celliax-on-trix-2.3.0.4
chmod +x celliax-on-trix-2.3.0.4
./celliax-on-trix-2.3.0.4


4. The script will install and configure Celliax on your Trixbox. It explains what it does, and you can see it for yourself here or here (save the file and open it with a text editor to read it). When installation is complete, the script will ask you to reboot Trixbox.
5. After Trixbox reboot, point your browser to the Trixbox machine, in this case http://192.168.1.15 (substitute the ip address of your trixbox), and configure it to use Celliax. The following steps are an example configuration with one SIP Extension, one Celliax Trunk, one Inbound Route, one Outbound Route (you can call out via Celliax from the SIP phone, and the calls that income into Celliax will end up into the SIP phone). You can try it also if you do not have an hardware SIP phone, using a SIP softphone.

   At the bottom of this page, you will find the screenshots of Alsamixer, to regulate the volumes of the Trixbox soundcard.

First screen, you will click at top right on "switch":

After giving login credentials (user=maint password=password), you'll be in the administrative screen. Here you will click on the menu "Asterisk" and choose "FreePBX":

You'll arrive at the FreePBX initial screen, where you will click on the orange label ("apply configuration changes") at top to initialize it:

FreePBX applying configuration changes:

FreePBX has applied configuration changes. From the menu on the left, choose "Extensions":

Let's add a SIP extension. Click "Submit":

Insert "200" as User Extension, your name as Display Name, a password in Secret, then you scroll down (go to next screenshot):

Enable Voicemail and Directory, check Email Attachment, Play CID and Envelope, click on Submit:

Click on the orange label ("apply configuration changes") at top:

What happens when you over the mouse on the orange label ("apply configuration changes") at top:

FreePBX applying configuration changes:

The result. You will click on Trunks (menu at left):

Let's add a Custom Trunk:

Set Maximum Channels to 1, and Custom Dial String to: Celliax/nicephone:$OUTNUM$

Please note that the vertical bar at the end of $OUTNUM$ is the cursor, not the "|" character. Then click Submit Changes

Click on the orange label ("apply configuration changes") at top, and after the changes has been applied, from the left menu choose Inbound Routes:

Insert a name in Description, then scroll down (go to the next screenshot):

In Set Destinations choose Extensions 200, then click Submit:

Click on the orange label ("apply configuration changes") at top, and after the changes has been applied, from the left menu choose Outbound Routes:

On top right, click on 0 9_outside:

On Trunk Sequence, choose the Celliax trunk from the dropdown menu:

The result. Click on Submit Changes:

Click on the orange label ("apply configuration changes") at top, and after the changes has been applied, from the top label, choose "Panel":

You will arrive at the Flash Operator Panel (FOP), shown here with no SIP phone connected:

Here is how it looks when the SIP (soft)phone is connected:

The screenshot from "alsamixer -V all" (note that it continues to the right, go to the next screenshot):

the continuation (to the right) of the previous screenshot:

the result of clicking F4 (show details on Capture):