Voip Innovations Module

Overview:

Sangoma's connectivity module for VoIP Innovations is designed for simplified deployment with your PBX.  

Features:

• Auto-Configure VI endpoint in FreePBX

• Automatically create inbound route for DIDs associated with VI endpoint

• View real-time connectivity status

• Send and Receive SMS using UCP and Sangoma Connect soft clients

Module Installation:

To install Sangoma's beta add-on for VoIP Innovation, you must first log in to FreePBX Distro or PBXact via SSH and run the following command

Create VoIP Innovations Endpoint:

1. Log in to VI Back Office at dashboard.voipinnovations.com

2. Click ellipsis on left to expand VI Menu

3. Select Endpoints > Endpoint Group Management > Add Endpoint Group

4. Select "Sangoma PBX"

5. Configure Options

6. Copy Auto-configure Key

7. Click Add Endpoint Group button

Available Options:

Note: Please make sure you mark the checkbox for Enable SMS if you plan to use SMS services either with UCP, SangomaConnect or any compatible module on the PBX, otherwise SMS will fail to be sent.

Open

Add/purchase DID:

1. Click ellipsis on left to expand VI Menu

2. Select DIDs

3. Choose relevant option from DID menu

   1. Purchase DID: Buy new DID

   2. Manage DIDs: This tool will search your local, toll-free, and international DIDs based on the filters you choose.

   3. Personal Inventory: Use personal inventory to hold DIDs for future use.

4. Click Save Changes

Screenshot - new DID purchase:

Open

 
Module Configuration: (Core Menus)

Initial Setup Screen FreePBX GUI - Auto-Configuration screen:

1. Navigate to Connectivity > Metered SIP trunking

2. Paste auto-configure key related to VI endpoint

3. Click Add Key

Screenshot:

System Status:

This screen provides helpful information about the status of your system's connectivity with VoIP Innovations.  

Options:

• Refresh Asterisk Account Info

• Check Connectivity

Screenshot:

Check Connectivity

Clicking the Check Connectivity button will display the external IP address of the PBX and show the result of the Firewall test.  When a firewall test is initiated, a free port is chosen from the defined SIP RTP media range on the PBX, and an API call is made to the FreePBX mirror server using http/https. The mirror server sends back a sample SIP media payload to the chosen RTP port. If the payload is received successfully, the test passes. This test result is not definitive. It is intended to assist with configuration of external Firewall/Router devices, but may not catch all configuration edge cases for all devices.

Location Information:

Shows you information on the current SIP gateways and SIP credentials are currently being utilized.

Options:

Remove Key - Will take out any existing key configuration and disable the trunk. This can be reclaimed by modifying the endpoint in your VoIP innovations back office account.

Screenshot:

Routing Information:

Allows the adjustment of existing outbound routes and inbound routes from a single page.

Options:

• Area Code- Enter local area code to enable 7-digit dialing on VoIP Innovations trunk

• Enable Primary Gateway-  Select yes for any route where you want to make VoIP Innovations the 

• DID- Clicking the number opens the inbound route for newly created DID

• Route To - Change inbound route destination, i.e. extension IVR etc. 

• Last Call - Shows the caller id number of the last number to reach the DID

Additional Configuration Options:

Additional SMS configuration can be found in both the User Management and Inbound Routes module

• User Management

  • Edit User: Enable SMS to extension by selecting VI DID

• Inbound Routes

  • Add SMS to extensions by selecting them individually or in bulk

User Management Screenshot:

Inbound Routes Screenshot: