What is a SIP URI?

There are thousands of people that operate their own Asterisk based PBX systems, yet they do not enable any method to allow for p2p sip URI dialing. These sip “targets” are very easy to enable and allow you to dial anyone that has also enabled the function. Dialing with SIP URI completely avoids toll calling and forces your Asterisk server to create P2P sip connections when you dial someone’s SIP URI. It makes a less complex phone call without a system administrator configuring a peer and best of all: It gets rid of phone numbers and your telco!

How does it work?

By creating a SRV record in DNS for your domain you can help remote PBX systems establish P2P calls for a specific extensions. For example, when someone calls me, my URI is resolved to my PBX (gateway.vsip.co.uk). When the call comes into my Asterisk box, I am is setup as an extension, and that extension is connected to a phone or a context. As a result, if someone uses a softphone to call me@vsip.uk, I get a normal ring and phone call. When I use my SIP desk phone and dial someone’s SIP URI it completes like a normal phone call.

Why is this cool?

This is great because it takes away any central control for locating people. The ENUM standard is nice, but gives someone else control over the mapping database and it keeps an ugly old phone numbers in place. I really don’t want to dial phone numbers 10 years from now, I much rather just give someone my email address and have that map to my phone. If I need to call a business, I much rather just call pbx@companyname.com then find some obscure phone number.
If more people adopt this as a standard, it will be the method of choice for calling people and it puts power into the end user’s hands!