SIP Trunking for PBX

We have introduced a new feature to the user account area allowing the user to enable/disable the SIP Trunking Service for PBX users.

This feature will provide additional flexability to our system, allowing the use of SIP accounts with standard VoIP endpoint equipment, extension telephone devices and softphones etc, as well as PBX servers such as Asterisk and FreePBX. Previously the SIP Trunking Service had to be enabled by our support engineers on request.

What is a SIP Trunking Service?
Sounds very technical, and expensive.... like a lot of the termanology that surrounds VoIP. In truth, it is very simple, and doesn't cost a thing.

SIP Trunking for PBX just sends call invites in a different form so that a PBX knows on which DID number a call arrives and can route it to the correct endpoint.

Invite for PBX : DIDNUMBER@IP:port
Invite for extension device : SIPUSERNUMBER@IP:port

It is important that the correct invite is sent for the correct equipment else incoming calls are not received.

By adding a feature to 'toggle' between invite forms we enable both possabilities.

SIP Trunks can work with any SIP-ready PBX - contact us if you have any questions.