Using FREEPBX to configure your Trixbox server

4.1 What is FreePBX?

Asterisk Management Portal makes Asterisk configuration easier by providing a graphical method (through a web browser). FREEPBX allow you con configure the textual configuration files that Asterisk needs to function.

FREEPBX can configure the following  in asterisk:

Incoming Calls --- Specify where to send calls coming from the outside
Extensions --- Add extensions and set voicemail properties
Ring Groups --- Group extensions that should ring simultaneously
Queues --- Place calls into queues and allow them to be answered in order
Digital Receptionist --- Create voice menus to greet callers
Trunks --- Set up trunks to connect to the outside world
Outbound Routing --- Manage which trunks outbound calls go out
DID Routes --- Specify the destination for calls if their trunk supports direct inward dial
On Hold Music --- Upload MP3 files to be played while users are on hold
System Recordings --- Record or upload messages for specific extensions
Backup and Restore --- Create, back up, and restore profiles of your system
General Settings --- Set basic dialing, company directory, and fax settings    

For IP Communications configuration purposes we will need to enable some of the modules in FreePBX
Please follow these steps:

1. Open your web browser and type HTTP://YourAsteriskIPaddressHere
2. Switch to Admin Mode. (click on the switch link in the upper right corner)
3. Click on the Asterisk Menu
4. Select Free PBX
5. Click on Tool
6. Click on Module Admin
7. Enable the following
   • Core
   • Voicemail
   • IVR
   • Ring Groups
   • Recording
   • Call Forward
   • Call Waiting
   • Do-Not-Disturb
   • Info Service

4.2 Configuring an extension

1. Open your web browser and type HTTP://YourAsteriskIPaddressHere
2. Switch to Admin Mode. (click on the switch link in the upper right corner)
3. Click on the Asterisk menu and select FreePBX.
4. In the FreePBX menu click setup and select extensions.

5. From the device drop down menu select “Generic SIP device” and click submit.

Example

1. Create extension 200 and type in a password for registration like "abc123". Then enter the name of the person using this extension.
   
   
   
   
   
   
2. Select enable, and enter a voicemail password. Use something you can type on a phone keypad like '1234'. Enter an e-mail address where you would like your voice messages sent and click add extension. Then click on the red apply bar at the top of the screen.
   
   
   
   
3. Configure your extension in a soft phone for testing. Xlite is the best choice for this test. Remember to use your extension number and password in Xlite. Use your Trixbox  private IP address as the sip proxy.
   
   
4. Make a call from your phone. Try *43. This is an echo test.

NOTE: If the extension you are configuring will connect remotely (outside the Local Area Network) you will need to change the NAT option to yes.

Just create the extension, submit the changes and go back to edit it. You will see NAT=never; change it to NAT=yes

Every time you make a configuration change and click “Submit” an ORANGE button will appear at the top of the screen “Apply Configuration Changes”. This button will reload the . conf files. Click this bar in order for the changes to take effect.

4.3 Configuring trunk for inbound calls

1. Connect to your Trixbox using a PC in your network by typing HTTP://YourAsteriskIpaddress   in your web browser.
2. Select FREEPBX under the “Asterisk” Menu
3. Click Trunks then “Add SIP Trunk”.
4. Only enter the following information:
   
   Incoming Settings
   User Context = sip.ipcomms.net
   
   
   
   
   *Registration String = DO NOT ENTER REGISTRATION STRING ON THIS SCREEN.
   
   
   
   
   Leave this registration string text box empty. It will be entered in the sip_nat.conf  file.
   
   
5. Click “Submit Changes”
   
   
   
6. Use a pc on your network that has a web browser and connect to your Trixbox box using HTTP://PutYourTrixboxIpaddressHere.

If you have a public IP address, nothing needs to be added to the general section go to step 4.5. 

4.4 What do I do if my asterisk server has a private IP Address (Optional)?

1. Click on the Asterisk menu.
2. Click on Config Edit
3. Click on sip.conf
4. Enter the following information at the top of your sip.conf file in the general section:
   
   externip=WAN IP address
   localnet=192.168.1.0/255.255.255.0 (or whatever your private LAN is configured for)
   
   
5. Click UPDATE

6. Click re-Read Configs located at the top of the screen.

Note: If public, nothing needs to be added to the general section.  Depending on your setup, you might need to forward all ports to your Asterisk server's private IP address or put the private IP address of the Asterisk server on the DMZ.

4.5 Configuring Inbound Routes 

NOTE: YOU WILL NOT BE ABLE TO RECEIVE CALLS IF YOU DO NOT CONFIGURE AT LEAST ONE INBOUND ROUTE

Configuring inbound routes will allow calls from IP Comms to go someplace in your PBX.

Using FREEPBX

1. Select setup
2. Select Inbound Routes.
3. Leave the  DID number and Caller ID Number boxes empty.
4. Under “set destination”  select extension 200.
5. Click Submit
   
   
   
   
   
   Call the your IP Communications DID . Your SIP phone extension should ring.