FreePBX Configuration

FreePBX Peer Configuration for SIP Trunks

Our SIP Trunking package offers IP Authentication instead of Registration like many other providers offer. The good thing about IP Authentication is that it enables you to have your PBX server more secure, since you won't be needing to enter a password and username to connect to our servers. Here's a base configuration on how to setup your FreePBX server to use IP authentication with our service.

1. Go to the IP address of where your FreePBX server is located, and click on "FreePBX Administration".

2. You will be taken to the login page. Enter your credentials. (Note: If this is your first time logging in, the username and password are both "admin").

3. Once your are logged in, hover over "Connectivity" and click on the drop-down item "Trunks"

4. Once you are at the landing page, click on "Add SIP Trunk".

5. The landing page is where all of the important information goes. Look at the picture on the left and I will explain the settings:

•Trunk Name: This is how FreePBX identifies your trunk. Usually it's the provider name.

•Outbound Caller ID: This is usually the trunk ID or phone number (DID) in your account. AN extension can override this if you have multiple DIDs.

•Maximum Channels: This field refers to how many concurrent calls or 'ports' you are allowed on this specified trunk.

•Trunk Name (Field 2): This is how Asterisk(FreePBX Framework) identifies your trunk.

•Peer Details: This is where the provider's information goes.

Host=206.15.150.13
Bindport=5060
Type=peer
Disallow=all
Allow=ulaw&g729
Dtmfmode=rfc2833
Qualify=yes


Host=206.15.150.13 is Phonepower's IP address you need to peer with. Bindport=5060 is the port where your trunk is located(usually 5060). Type=peer is the way you connect to the trunk, it can either be user or peer; but since FreePBX is a peer of Phonepower, we set it to that. Disallow=all means you are disallowing all codecs to be used at the time of that line being executed. Allow=ulaw&g729 means that now you are allowing only those two codecs through. Dtmfmode=rfc2833 is the way you will transmit DTMF tones.

Configuring Inbound Routes

*You need to have an Extension created. Inbound routes are configured to control call flow when an incoming call comes in. Please look at the image below and copy the settings.

Configuring Outbound Routes

*You need to have an Extension created.Outbound Routes are used when you are making an outgoing call. This has to be configured if you are going to be making calls. Please follow the configuration in the image below and you will be able to make calls.

Server Redundancy

What we have configured so far is not very redundant. If the server 208.64.8.13 goes offline on our end, your outgoing calls will fail. So to remedy this, we will need to configure another trunk and add that newly created trunk to the outbound routes - "Trunk sequence for matched routes".
Steps:
1. Create a new trunk.
2. Add the Trunk Name, Outbound Caller ID, and Trunk Name(2).
3. For peer details add this information:

Host=206.15.130.13
Bindport=5060
Type=peer
Disallow=all
Allow=ulaw&g729
Dtmfmode=rfc2833
Qualify=yes


4. Save Trunk.
5. Go to your outbound routes and add the new trunk to the list of "Trunk sequence for matched routes".
6. Save and you're done.
With this now you will have redundant connectivity to our servers. Meaning if a server goes down on our end, you will still have service.