Interconnect XiVO with any VoIP provider

When you want to send and receive calls to the global telephony network, one option is to subscribe to a VoIP provider. To receive calls, your XiVO needs to tell your provider that it is ready and to which IP the calls must be sent. To send calls, your XiVO needs to authenticate itself, so that the provider knows that your XiVO is authorized to send calls and whose account must be credited with the call fare.

The steps to configure the interconnections are:

  • Establish the trunk between the two XiVO, that is the SIP connection between the two servers

  • Configure outgoing calls on the server(s) used to emit calls

  • Configure incoming calls on the server(s) used to receive calls

Establish the trunk

You need the following information from your provider:

  • a username

  • a password

  • the name of the provider VoIP server

  • a public phone number

On your XiVO, go on page Services ‣ IPBX ‣ Trunk management ‣ SIP Protocol, and create a SIP trunk:

Name : provider_username
Username: provider_username
Password: provider_password
Connection type: Peer
IP addressing type:
Context: Incalls (or another incoming call context)
Media server: MDS Main

Register tab:

Register: checked
Transport: udp
Name: provider_username
Username: provider_username
Password: provider_password
Remote server:


For the moment, Name and Username need to be the same value.

If your XiVO is behind a NAT device or a firewall, you should set the following:

Monitoring: Yes

This option will make Asterisk send a signal to the VoIP provider server every 60 seconds (default settings), so that NATs and firewall know the connection is still alive. If you want to change the value of this cycle period, you have to select the appropriate value of the following parameter:

Qualify Frequency:

At that point, the Asterisk command sip show registry should print a line showing that you are registered, meaning your trunk is established.

Set the outgoing calls

The outgoing calls configuration will allow XiVO to know which extensions will be called through the trunk.

Go on the page Services ‣ IPBX ‣ Call management ‣ Outgoing calls and add a route.

Tab General:

Trunks: provider_username
Extension: 418. (note the period at the end)

This will tell XiVO: if an internal user dials a number beginning with 418, then try to dial it on the trunk provider_username.

The most useful special characters to match extensions are:

. (period): will match one or more characters
X: will match only one character

You can find more details about pattern matching in Asterisk (hence in XiVO) on the Asterisk wiki.

Set the incoming calls

Now that we have calls going out, we need to route incoming calls.

To route an incoming call to the right destination in the right context, we will create an incoming call in Services ‣ IPBX ‣ Call management ‣ Incoming calls.

Tab General:

DID: your_public_phone_number
Context: Incalls (the same than configured in the trunk)
Destination: User
Redirect to: the_front_desk_guy

This will tell XiVO: if you receive an incoming call to the public phone number in the context Incalls, then route it to the user the_front_desk_guy. The destination context will be found automatically, depending on the context of the line of the given user.