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FreePBX Configuration Guide

Third Party PBX Support - FreePBX Configuration Guide

Here you will find the configuration details for FreePBX which is a third party open source PBX that you can build yourself:


This is based on FreePBX (Distribution 6.12.65 FreePBX 12, Linux 6.5, Asterisk 11 or 13) available during December 2014. There are several sections to work through. This may not be exhaustive or tailored to your exact needs, and is offered as a guide only to get you started.


Open the web interface and log in.


Select Connectivity


Select Trunks


Select Add SIP (chan_sip) Trunk


For the General Settings:


Trunk Name - you could name this 'Voipfone'


Outbound CallerID - this will not be used by Voipfone, you set the CallerID in your Voipfone Control Panel. However, it can be useful to put something in here, in case you need to look through system logs. So a description such as 'by-voipfone' could be used.


For the Outgoing Settings:


Trunk Name - we suggest 'Voipfone-SIP'


Peer Details:




























For the Incoming Settings:


Don't do anything here - leave blank and continue to the next section.


For the Registration Settings:


Register String - USERID:PASSWORD@sip.voipfone.net/USERID


Remember you have to Submit Changes at the bottom, and then Apply Changes at the top.


Select Connectivity


Select Outbound Routes


Select Add Route


Route Settings:


Route Name - for example 'out-by-voipfone'


Dial Patterns that will use this Route:


There are 4 boxes; [prepend] [prefix] [match pattern] [CallerID]


Leave [prepend] [prefix] and [CallerID] alone.


We want to send all of our outbound call requests to voipfone, so we use the rule of “X.” to mean all.


Put X and the Decimal Point in the 'match pattern' box.


You should end up with the 4 boxes as [prepend] [prefix] [X.] [CallerID]


Trunk Sequence for Matched Routes Settings:


From the drop down list select the Trunk Name you created earlier, such as 'Voipfone'


Remember you have to Submit Changes at the bottom, and then Apply Changes at the top.


Select Connectivity


Select Inbound Routes


Select Add Incoming Route


Incoming Route Settings:


Description - for example 'in-from-voipfone'




Changes to other settings are not required. Continue down to:


Set Destination:


Choose how your system routes calls. Extensions, IVR, Queues, Ring Groups etc. You should have already configured the destination. For example, probably the simplest configuration - Extensions - <nnnn> First User.


Remember you have to Submit Changes at the bottom, and then Apply Changes at the top.


The USERID mentioned should be replaced with your own details - this is often the Voipfone account number, typically 8 digits long, starting 3. Take care to replace the PASSWORD also.


You don't use any of your Voipfone phone numbers in the above configuration. You don't use your registered email address in the configuration.


Select Reports


Select Asterisk Info


Select Registries


Chan_Sip Registry should show that you are sucessfully registered and online with Voipfone.


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