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Sat 17 of May, 2008 [05:24 UTC]

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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
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VoIPUser

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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:-

http://apps.facebook.com/voipuser

Click to call will be following shortly.

RSS

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

Syndicated Feed:-

RSS

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 Dean, Last modification by Dean on Thu 28 of Jun, 2007 [13:59 UTC]

Comments

Comments Filter

Thanks...

by Dean on Wednesday 18 of August, 2004 [23:31:17 UTC]
... for the positive comments "Anonymous".

Personally, I think we have a way to go yet to provide the level of service that we actually want to provide, especially in relation to offering users a big degree of control over number management.

But we're working hard and the user feedback we've had has been amazing. We won't be resting on any laurels.

Dean
Edit

Great service!

by Anonymous on Monday 16 of August, 2004 [13:40:57 UTC]
I have used this service for sometime. And really the best UK DID provider around! (:biggrin:)

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

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