VoIP User

The Community VoIP Resource and Service Provider

Last Summer we celebrated our membership exceeding 10,000 subscribers with a site redesign and a new custom softphone built for us by SIPfoundry. In January, we reached the 20,000 mark and continue to grow at a rate of 40 new subscribers each day. Our VoIP Forum has become the busiest independent Voice over IP forum on the internet with some 12,000 active users, many of whom are industry professionals, and extensive moderated sections covering all aspects of the technology from Industry News to Asterisk Configuration.

Update in Brief

We recently rolled over 40,000 members and have now routed over 1.5 million calls to and from the PSTN. Our servers act as a testing sandbox for the openSER team and several independent developers.

Our latest gizmo has been a Facebook SIP Presence/Google Maps mashup. If you're a Facebook user, and have a VoIP User SIP account, you can install that here:-


Click to call will be following shortly.


We now have an RSS feed available. This contains administrator selected threads from the forum, reviews and news items relating to VoIP.

Syndicated Feed:-


Mission Statement

VoIP User is 100% community driven. Our aim is to introduce people to the concept of VoIP and allow potential users to experiment with VoIP by offering innovative free facilities for both inbound and outbound calling from a SIP or IAX2 device.

To meet this aim, we have developed a number of backend services which directly negotiate with our PSTN gateway servers in the UK.

All services we provide to our members are free. All that we ask in return is that users become involved in the community side of the project. That is to help others, offer feedback on products and ISP's and generally assist other members of the community in the transition from PSTN to VoIP. Membership and subscription to our services is free. We operate a clear privacy policy on members details which can be seen here.

What the community offers

  • Free UK DID (PSTN local rate access numbers) to SIP, IAX and PSTN termination Worldwide

Members get PSTN UK DID's (local rate direct dial number) to SIP, IAX2 (e.g. Asterisk) or landline termination.

  • Free community outbound service from SIP to Worldwide PSTN.

We have a cap on call routes which means our gateway will only route calls to PSTN in any Country which we can route to for 2.5p/min or less. This is mostly landlines, but does include some mobile phones. A full Country list can be seen in the VoIP Termination thread

We also have a maximum allowed call time per call set at 10 mins. That may be reduced depending on useage.

We have a fair use policy setout here.

Our software back-end has been developed by ourselves and is designed to be scalable and flexible in order to maintain a constant "pool" of available minutes for new users.

  • We have updated our Control Panel for the UK DID's to enable full "time of day" options, multiple destination "follow me" type services and free voice2email. More info on our new DID Control Panel here.

Our UK DID's

  • Work with all SIP hardware/software
  • Work with all IAX hardware/software (Asterisk PBX, IAXy etc)

See Current VoIP Services for more info

See Free outbound VoIP PSTN gateway for support info

  • New 3 pence per minute flat rate UK DID (released with our free outbound service)
  • Voicemail and voicemail to e-mail service
  • New Fax to e-mail service

Analytical Data from the VoIP User server can be seen here: VoIP Analysis

General Overview

VoIP User is a community resource providing service provision through revenue created by community use of the inbound PSTN numbers.

All services provided by VoIP User are free. More information about VoIP User can be found here

VoIP User in the Press

Created by: deanster, Last modification: Thu 28 of Jun, 2007 (13:59 UTC)
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