SIP Broker Wiki

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SIP Broker is a free service that provides unlimited VoIP calls to millions of people around the world. No signup is necessary.

How Does It Work?

Simply find the SIP-Code for the network you wish to reach in our Network Directory and enter it into your phone followed by your friend's phone number. Your call will be connected, free of charge!

Example:

Your friend uses SIPphone as their VoIP provider. Your friend's VoIP number is 1-747-555-1212. Find SIPphone in the list, which is *747. Simply dial *747-1-747-555-1212 and you're connected!

Also, if you don't have access to a VoIP phone, you can dial any one of our PSTN access numbers available in over 20 countries around the world. At the voice prompt, just dial the VoIP number as described above.

SIP Broker setup is very easy. You don't even need to "sign up". If you have a Sipura or Linksys adapter , all you need to do is replace your current dial plan with:



For this dial plan to work correctly, press '#' after you're done dialing. This is similar to pressing the 'Send' key on your cell phone, and is also a handy way to speed up call connection time, even on old PSTN phones. For other adapters, or if you want to edit this dial plan before trying it out, see Dial Plans

Asterisk and Brekeke SIP Server setups are also very easy. Here are some snippets of Asterisk Configuration file settings and Brekeke SIP Server Configuration.

To verify that SIP Broker is working:

• Call the SIP Broker test number at *011-1-88888 (the actual keys you press are '*011188888#'). You should hear a SIP Broker welcome message. If not, SIP Broker does not work for you at all, so change your dial plan back to whatever it was before and stop here.
• Call *266-300. If you hear the monkey sounds, then domain-to-prefix mapping is working.
• Call 1-516-687-5089. After playing an ad, it tells you whether you connected via ENUM (free call) or PSTN (paid call). (If you would like the call to be "ENUM or bust", then dial it as *013-1-516-687-5089.)
• Call any PSTN number. It should go through the same way it did prior to changing your dial plan. That is, if you use a SIP-to-PSTN gateway, the call will be completed via PSTN because no ENUM entry exists for that number. (Note: If you want to know if there is an ENUM mapping prior to placing the call, see here.)
• Here are some more Phone Numbers you can call.

To see how everything fits together, see A Complete Plan for DIY VoIP.

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