Many users were requesting configuration settings to use the 3300V's FXO ports as Trunks in Asterisk/Trixbox. This was not the intended design of the ports and as such, they cannot be defined as normal Trunks in Asterisk, they are designed to function as extensions.

However, through some configuration trickery we can make them function like trunks in a manner that is transparent to the caller/callee.

The first thing to do for this setup is to configure your Asterisk/Trixbox server inside the 3300V, so that the FXO/FXS ports know what to register to, in our example we are telling the 3300V that our Asterisk/Trixbox server is at 192.168.100.20.

The next step is to configure the port settings for the chosen FXO port. In our example we have chosen port 5, the first thing to do is to enable the port and then fill in the extension details that we will setup in FreePBX in a moment. In the example I have chosen 2000 for the extension number and 0000 for the Password. Under proxy server you will be able to select the SIP server defined in the first step. We then want to enable "Call without Registration" as a failsafe measure.

Next, set the "VoIP IP Address" to LAN/VPN, if your Asterisk/Trixbox server is out on your WAN somewhere, outgoing calls to the FXO will not function with this setup as the workaround does not support it.

Finally define "Hotline Number to Internet" as a SIP URI using a second extension that we will setup in a moment, it must be a new unused/undefined extension. This will be how we handle incoming calls from the PSTN. In the example I have entered This e-mail address is being protected from spambots. You need JavaScript enabled to view it

Depending on your IVR you can simplify the incoming call setup, by specifying "Hotline to Internet" as a SIP URI of This e-mail address is being protected from spambots. You need JavaScript enabled to view it , this will simulate an incoming call directly to the IVR. If you are using Announcements and Time Conditions however, you will have more control over the incoming call using the second extension method with Follow Me settings.

Now we want to login to FreePBX and add a new SIP extension.

Here we will define extension 2000 so the FXO on the 3300V can register to it.

Additionally, change the DTMF mode to inband for this extension, so it is compatible with the 3300V's FXO port.

Now we will need to edit extensions_custom.conf to define a custom dialing string:

exten => _6.,1,Dial(SIP/2000,60,D(${EXTEN:1}))

We have now completed the setup for outgoing call functionality, we now need to define another extension to specifically handle incoming calls. I have arbitrarily chosen 1000 and defined it as below.

Now we need to define follow me settings for this extension. If you do not have follow me installed, you can do so from the Module Admin, under Inbound Call Control.

The important settings here are Initial Ring Time and Ring Time, we want to define them to the lowest possible settings of 0 and 1 respectively.

Now under the "Destination if no answer" section we can define where we want incoming calls to be routed to, in this case I have chosen our IVR MainMenu, but you can direct the call to anywhere just like a Trunk.

This completes the incoming call setup on the Asterisk/Trixbox server, save and apply the settings

We have one last configuration to do, and it is on the 3300V. Under VoIP - Tone Settings, switch to "User Defined" and change the Dial tone frequency to 30-35Hz, this will get rid of the dial tone when an incoming call hits the 3300V but before the Hotline function forwards it to the PBX.

Configuration is now complete, you should be able to dial out to the PSTN using 6, just like it was an outbound route. Incoming calls will ring once, get answered by the 3300V's FXO port, then forwarded to extension 1000 by tthe Hotline to Internet function, and then ring once more before being directed to the IVR, or whatever destination you desire.