FreePBX VI Trunk Authentication Guide (PJSIP)

Setting up PJSIP authentication trunks within FreePBX is fairly straightforward and can be completed by the following steps.

First you will need to enter the “Trunks” section of the “Connectivity” menu, select “Add Trunk” then the option for PJSIP.

From this area you will need to enter a relevant trunk name, then select the “pjsip Settings” tab.

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 In this area you will need to enter the username and secret (password) of the trunk which can be located within the Back Office. SIP Server will be a combination of your endpoint group ID followed immediately followed by ‘authvi’. For example if your endpoint ID is 98765 it should be entered as “98765authvi”, the remaining information can be left as is.

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Moving to the “advanced” tab, you can leave most of these settings as is, or tailor to your specific configurations should you have them. Within “Outbound Proxy” field is where our IPs will be configured for VI origination/termination IPs. While you will still have to configure multiple trunks with the below IPs much like the “Static” default, the trunk IP should be entered in “Outbound Proxy” as ‘sip:IP_ADDRESS’ so in this example we have ‘sip:64.136.174.36’. The “Server URI” field will be entered with ‘sip:’ followed immediately by your Endpoint Group ID such as ‘sip:98765authvi’. Once completed you will need to enter in the SIP IP information in which we will be communicating with you, these can be single IPs or subnets, however the below combination of subnet and IPs should be a good copy/paste solution as this includes relevant IP space.

Match (Permit)

64.136.160.0/20,209.166.128.200,209.166.128.201,209.166.154.70,209.166.154.71

VI IPs

VoIP Innovations has multiple SIP servers for redundancy and reliability. These are the servers in which you will need to register.

We suggest making sure you have set up and are registering to each of the IP addresses listed here in order to ensure optimal uptime. Not registering on all the origination servers could possibly result in failed calls due to the nature of inbound routing.

Switching to the “Codecs” menu, we recommend limiting these to ULAW, ALAW, and G729 (license required) to not only limit the options to what is supported, but to restrict packet size to within the UCP 1500 byte limitation.

You should now be able to press “Submit” in the lower right portion of the window, then “Apply” (generally in red) in the upper right to commit the configuration. Once reloaded, you should see that there is a new trunk online via the FreePBX Dashboard.

*Please note: Any DID’s routing to username/password authenticated endpoint groups will need Inbound CNAM enabled to function properly*