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POTS Line Replacement Setup

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Overview

POTS line replacement allows you to register the gateway to a SIP server and map the 6 FXS ports to specific DIDs per port. This allows you to simulate traditional analog lines that each line is a dedicated number.
The gateway itself will do a registration to your SIP server with a Username and Password at the SIP trunk level.

Each FXS port itself does not register to the SIP Server. We just map DIDs to FXS ports.

The setup guide below assumes you have already setup the 4G Sim and APN as outlined in our Initial Setup guide here

From the Menu click on Trunk > SIP Trunk
Click on the Add Button to add a new SIP Trunk
Set the following fields to setup your SIP Trunk and press the OK button when done.

Registration - Enable
Number - SIP username that you register with on you SIP Trunk
Username - SIP username that you register with on your SIP Trunk
Password - SIP password that you register with on your SIP Trunk
Concurrent Calls - Set to the number of concurrent calls you want to limit this trunk to. Usually set to the same quantity of FXS ports you plan on using.

We now need to set the server the SIP trunk registers to. Click on the Registrar Options button
Set the following fields and press the Save button when done.

Default Registrar- IP address of FQDN of your SIP Server
Local Signaling Port - What port you want to listen to SIP on for this SIP Trunk
Register Expiration - How often to re-register to the SIP Server in seconds (we recommend 60-600 Seconds)

Now that we have a SIP Server setup we need to map each DID to a FXS port on the Gateway for inbound calling so we know which port to ring for each phone line. The gateway supports 6 FXS ports.
From the Menu click on Trunk > SIP Trunk
Click on the Add Button to add a new Phone Number Mapping
Set the following fields to setup your SIP Trunk and press the OK button when done.

Registration - Disable
Number - Set to be the phone number of the DID we want to map to a port.
Username - SIP username that you register with on your SIP Trunk
Password - SIP password that you register with on your SIP Trunk
Concurrent Calls - Set to the max number of inbound calls you want on this DID at anytime.

The DID will now show up and next we will need to edit that DID entry Routing options. The default routing for new DIDs is the built in Auto Attendant but in the mode we are setting up here with POTS replacement we need to route this to a FXS port.
In this setup we want to route calls from this DID to a specific FXS analog port on the gateway. Click on the DID that we just created and edit the below fields.

Inbound Route - We want to route to a FXS port so we pick DID here.
DID Number - In our example we want to route to FXS Port 1 which internally maps to extension 200 so we will pick 200 here.

From the Menu click on Advanced > Routing
Click on the Route tab at the top
Call From - Analog Extension
Number Matched - Caller and the Any button
To - SIP Trunk
Press the + by Select Line under the To column and pick the SIP Trunk for the SIP username you have setup and press OK
You should now see a screen like this and can press the "Add to routing table" button
This will show a similar line below showing that all FXS ports when making a outbound call will route out the SIP Trunk ID 13. Press "Save" when done.

From the Menu click on Extension > Analog
Click on the FXS port you want to set the Outbound Caller ID Number on and in the CPN field provide the number.

If you are experiencing 1 way or 2 way no audio issues on your LAN or 4G you may need to setup a STUN server. See our wiki here on STUN settings

If you want to setup a hunt group that would take a inbound DID number and have it failover to other FXS ports if the main port the DID is mapped to is busy or in use. See our wiki here on Inbound Hunt Groups

If you would like to setup a Inbound number to be answered by a IVR or Auto Attendant, See our wiki here on IVRs and Auto Attendants