Tutorial shows how to configure P2P SIP dialing on your Asterisk PBX

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 (sip.blyon.com). When the call comes into my Asterisk box, blyon is setup as a extension, and that extension is connected to a phone or a context. As a result, if someone uses something like Xten to call blyon@blyon.com, I get a normal ring and phone call. When I use my Cisco 7960 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@somecompany.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!

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Written by Dal on July 13th, 2006 with no comments.
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Tricking Out Your TrixBox

For those that thought we’d dropped off the face of the planet, good news. Not yet. If you haven’t heard, there’s a new version of TrixBox, 1.2. And we’ve given it the old college try for a week or two with about that same results pictured in this old comic book. On some platforms, it runs just fine. On others, including our VMware for Windows machines, it’s a nightmare.

The voice synthesis system is again broken, freePBX can’t reload Asterisk without completely shutting down and restarting Asterisk (amportal restart). And there appear to be all sorts of interrupt or timing problems that we’ve never seen before … going back to Asterisk@Home 1.2.

We attribute many of the problems to a new version of CentOS and Asterisk, both of which are bundled into the TrixBox 1.2 package, but who knows. What we do know is TrixBox 1.2 is a little too Bleeding Edge for our taste, and most of the Nerd Vittles goodies that depend upon the Flite speech engine no longer work on many machines...

Click Here for the Full Nerd 

Written by Dal on September 25th, 2006 with no comments.
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