This document assumes that you have already purchased and configured SIP trunks from the Gradwell VOIP Control Panel to point at the IP of your public WAN connection/Switchvox.
In the examples, the DDI of 01225123456 will be used, this will need to be replaced with your purchased DDI number.
The Switchvox administration and configuration is managed via a browser-based tool suite. To log in to the administrator tool suite, type the URL of the server into the navigation bar of a web browser on a computer that is part of the same network as your Switchvox Server.
The URL will be similar to https://192.168.0.100/admin
You will be prompted to enter a username and password. Switchvox is configured with a default username and password combination.
The default username is: admin
The default password is: admin
Once logged in, navigate to System Setup, and then VOIP Providers.
Next click Go next to Add New SIP provider under the VOIP Providers section.
Next, complete the boxes as per the screenshot below - this sets up the basic information in order to route calls to your Switchvox.
Once the above settings have been entered, click on Click to Show Advanced Options and complete each section as per the examples below
Once complete, click on Modify SIP Provider at the bottom of the screen.
You have now configured Switchvox to accept calls from Gradwell.
Next, you need to tell Switchvox how to route the calls that are sent to it from our network. Click on System Setup and then Incoming Calls, and select Single DID as the new call route, and click on Add Route.
You can now create a rule, as the example below shows. Any call received by Switchvox for 441225123456 from Gradwell will be sent to extension 800. The number must be entered in the international format.
Click on Save when finished. Switchvox will reload its configuration, you should now be able to make test calls to your number.
If you wish to route any SIP call sent to your Switchvox from Gradwell to an extension then you can configure your rule as the example below. A sip call could then be placed to firstname.lastname@example.org via our network
This option could be used, if you are routing a large number of DDI's to your server, and you do not have the requirement to individually route each one.
If you want to reject any unauthorised SIP calls, then you can configure a rule as below
This example can be used to stop users making SIP calls to your IP address from an unknown host.