Welcome! Sign in Or Register
  Home > Support > FreePBX / Trixbox guide  
 
Bullet Phone functions guide
Bullet FreePBX / Trixbox guide
Bullet How fast is your connection
 
FreePBX / Trixbox guide
 
Using FreePBX you are able to do most of Asterisk's configuration without editing the individual configuration files. Installing the account details given to you by us at sign up, will allow you to send and receive calls using us as your provider.

The setup information below is based on FreePBX 2.1.3; although most other older and newer version will look very similar.

Reference Information

Main Project Pages:
FreePBX - http://www.freepbx.org
Trixbox - http://www.trixbox.org

Help / Support:
FreePBX support page
FreePBX Forum


Setup Guides:
VoIP-info.org trixbox Wiki
Nerd Vittles trixbox and FreePBX 2.1.1 Guide


Configuring the Asterisk PBX using the FreePBX interface
This guide assumes that you have installed FreePBX using either the FreePBX package, trixbox or a method of your choice. This guide also assumes that the FreePBX install steps were completed properly and that you have administrative access to the FreePBX administration interface.

We recommend that you read each step through in its entirety before performing the action indicated in the step.

STEP 1 - Trunk Configuration
In the context of this guide a trunk is used to route calls between your Asterisk PBX and your desired VSP(Voice Service Provider), in this case PBXme. In this section we will configure a SIP trunk.

>> Login to FreePBX administrative interface

>> Click on Setup in top right of page

>> Click on Trunks in left side navigation

>> Click Add SIP Trunk in middle of page

>> Scroll to Outgoing Settings and enter PBXme into Trunk Name field

>> Copy and paste the following into the PEER Details field:
-----------------------------------------
context=from-pstn
fromdomain=fax.PBXme.com
fromuser=YourUserName
host=fax.PBXme.com
insecure=very
secret=YourPassword
type=peer
username=YourUserName
-----------------------------------------
Where YourUserName and YourPassword were given to you by PBXme at time of registration. Optionally you may change YourPassword in your preferences.

>> Scroll down to Registration

>> Enter your registration string in this format:
-----------------------------------------
YourUserName:YourPassword@fax.PBXme.com/YourUserName
-----------------------------------------
Where YourUserName and YourPassword were given to you by PBXme at time of registration. Optionally you may change YourPassword in your preferences.

>> Click on Submit Changes to add your new SIP trunk to your Asterisk server

>> Click on the red bar at the top of the screen to apply the changes you just made

>> Now you will want to edit your sip.conf file and enter, or modify, the following lines:
-----------------------------------------
context=from-pstn
srvlookup=yes
-----------------------------------------
If using Trixbox this will have to be done through the web interface to edit your config files.
If using FreePBX you will need to log in to your server and edit the /etc/asterisk/sip.conf file manually, usually with an editor such as Putty / Nano or such.


STEP 2 - Outbound Route Configuration
An outbound route sends calls which are dialed in a certain pattern to your provider, PBXme.

>> Click on Outbound Routes to configure your Asterisk box to send calls to PBXme.

>> Enter to-PBXme into Route Name field

>> Scroll to Trunk Sequence and select the SIP/PBXme trunk from the drop down list

>> Click on Submit Changes to add your new route to your Asterisk server

>> Click on the red bar at the top of the screen to apply the changes you just made


STEP 3 - Extension Configuration
An extension is an account on your Asterisk PBX which provides an account number which another device (software or hardware) can connect to in order to make and receive calls. There are a few types of extensions, here we will create a SIP extension.

If you have already configured an extension then you may skip this step. Then in the next step (Inbound Route Configuration) you may use your pre-configured extension.

>> Click on Extensions to add a new extension which will connect to your Asterisk server

>> Choose SIP as the extension type

>> Enter 1000 for the extension number

>> For now we will use a generic identifier for this extension. Enter First Extension for the Display Name field. Later you may enter a unique identifier of your choice

>> Enter your desired password in the Secret field. You will use this password when configuring your desired UA later in order to connect to your Asterisk PBX

>> Click on Submit Changes to add your new extension to your Asterisk server

>> Click on the red bar at the top of the screen to apply the changes you just made


STEP 4 - Inbound Route Configuration
In this section we are going to setup an inbound route which will handle incoming calls on ANY number, and route those calls to an extension (1000).

If you have already configured an extension then you may substitute your pre-configured extension for point 4 below.

>> Click on Inbound Routes to configure the routing of calls to your PBXme account

>> If there isn't a default inbound route called any DID / any CID then click on Add Incoming Route. Make sure leave the DID Number, Caller ID Number and Zaptel channel blank in order to match any incoming call. This is useful if you wish to receive all calls

>> Scroll down to Set Destination

>> Choose First Extension (1000) from the Core dropdown box

>> Click on Submit Changes to add your new inbound route to your Asterisk server

>> Click on the red bar at the top of the screen to apply the changes you just made


STEP 5 - Configure and test device

>> Choose your desired device.

>> Use the IP address or hostname for your Asterisk box along with 1000 (the extension created earlier) and password for the 1000 extension to connect to your Asterisk box

>> Next you will want to try placing test calls to and from your Asterisk PBX using the device currently connected to your newly created extension (1000).


STEP 6 - Placing Test Calls

>> View our FAQ, "What are the needed dialing rules when calling UK, USA or other countries?"

>> Try calling PBXme's main phone number at 9707000 (should be dialed as 9707000).

 
 
Quick Start
 
Register to service
Opening an account is easy. Start improving your bottom line today. Click here to register
 
Have a question?
For more information about our solutions, please fill our contact form.
 
Test run our service
Want to try our service for free? Test run our service.
 
 
Bookmark and Share
 
© 2005 - 2010 - all rights reserved



Numbers
Technical
Call rates
Register by email
Guides
Skype FAQ
Local Numbers
Test-run
Dynamic forwarding
Register by fax
Phone functions guide
General FAQ
Multi-channel
Conference rooms
Phone functions
FreePBX / Trixbox guide
Contact us
Toll-Free
SMS solutions
Voice mail
Test your connection
OneSIM
SMS rates
Fax2mail
Iphone
Overview
Skype solutions
Online billing
Softphone setup
Functionality
SkypeIn Numbers
Bring your own device
Telephone adaptors
Call rates
Skype for companies
Downloads
Troubleshoot
Calls termination
Cellular related
User manuel
Projects
Nokia
Coverage
Calling cards
Fring
IPBX SIP Trunk
Voice mail systems
GSM network codes
Features
Voice servers
OneSIM FAQ