Setting Up IP Whitelisting
When setting up your call forwarding to Asterisk, it is important to whitelist AVOXI’s signaling IPs within Asterisk, which can be found here. The IP addresses to whitelist are contained in the SIP AVOXI to Customer row.
Setting up a SIP End Point
Next, you’ll need to create a URI for AVOXI to use. What this looks like will depend on your business setup, how many Asterisk instances you have, and the level of resiliency you require.
Please note that this must be done at your DNS server level and will have to point to your public IP address assigned on your firewalls or SBC to your Asterisk PBX (IP address can be used directly on Genius Forward System).
Configuring a SIP URI To Point At Asterisk
Next, we will set up the SIP interface from AVOXI to Asterisk in your AVOXI Genius platform. This is done via Voice URIs.
In order to differentiate from any other Voice URIs that might point to your Asterisk, we’ll use a reserved keyword on our platform, “{E164}”, so that we can use the same voice URI for many different numbers and detect what number is called/who’s calling.
- First, log into AVOXI Genius, and navigate to the Numbers module
- Click on the SIP URIs tab located next to Manage
- Click on the ‘+ Add button to the right of the filter magnifying glass
- Clicking the ‘+ Add’ button will pop out the SIP URI configuration tool
- The SIP URI configuration tool has 3 configurable fields and 1 auto-generated preview field.
- Name: Meaningful name of the new SIP URI configuration
- Protocol: A transport protocol for your SIP URI configuration. Genius currently supports UDP ( preferred ) and TCP. For more information about the differences between these two protocols, please refer here!
-
URI: This is the configurable URI you will use to set up your forwarding. There are three components.
The defined number format expected from the dialed AVOXI Genius number (DNIS ) to your destination, the two format options are e.164 with or without a ‘+’.
- Preview: This is a preview of your full SIP URI
- Navigate to the number where you want to set your configured SIP URI
- Specify the Voice URI as {E164}@YourAsterisk.com or {E164}@YourAsteriskExternalIP
Testing Calls
Any calls placed to the numbers associated with your URI are now delivered by AVOXI Genois to your Asterisk. If you used the {E164} keyword, Asterisk should see them as calls received to URI (called AVOXI DID)@YourAsteriskDomain. You should be able to do a packet capture to capture the traffic and see the incoming call.
