With the holidays fast approaching and the exciting new things coming we thought it was time to freshen things up and give the site a nice clean look. We've been busy with CSS magic and challenges with drupal and Sledge has gotten really creative with our favorite mascot. We hope you like what you see.

There's no under construction sign up but if you run into a few glitches over the next week, that's probably why. Let us know what you like (and what you don't) and we will try to tweak things as best we can.

Over 4 years ago, FreePBX, then known as AMP, started a course to disrupt the traditional PBX market by changing the game on the existing black box telephone industry. In this short time the Open Source process has been able to surpass what has taken the industry decades to accomplish. With millions of downloads, an installed base counted in the hundreds of thousands, and almost every successful Open Source PBX distribution basing itself off of FreePBX, our project has clearly proven itself as a leader in this telephony revolution!

For those of you who keep a close eye on the development site you will notice that some data has just been deleted. I unfortunately corrupted the database in error trying to remove some profanities in some ticket comments. I restored it to the last snapshot that was taken which is about 3/4 of a day old. There should be nothing major lost.

Perhaps one of the most requested features in FreePBX is the ability to configure calling permissions. While this is a complex and costly request from a development point of view, there are some simple techniques which can be used to provide some level of outbound call control. It is said that well written software can be used in a way totally different to what its author intended. Some of the current FreePBX modules can be 'exploited' to provide just such functionality.

One of the really cool things added to the latest version of FreePBX is support for Russell's devstate backport for Asterisk 1.4. Today I decided to have a look at how it works, and I found it to be extremely simple and straightforward to set up. Obviously, you need to add the backport to asterisk. Luckily, that is extremely easy - just follow the directions in the readme.

Hi, my name is Mikael Carlsson and this is my first blog entry here on FreePBX.

It looks like our most recent Open Telephony Training Seminar at Digium's Headquarters was a success beyond our wildest dreams! Digium was so impressed with FreePBX, our community (as represented by the class attendees), the great attitude of everyone (who else has participants set up camp with their RV in the Digium parking lot), and the project as a whole that they went off and created AstriskNOW^TMwhile I was on the plane returning from Huntsville!

I must admit the announcement was not a complete surprise, having used our captive audience last week as a test bed for the new AsteriskNOW distribution which went very smoothly. If you are looking for a clean ISO with out a lot of extra fluff, this is an excellent option to consider. Although it is still in beta with a few minor known issues that are being addressed, I expect it to come out strong.

The most recent OTTS training was well received and our class was the first to get the revised material focusing on the great FreePBX 2.5 release that went final last month. We also heard much positive feedback about the quality time spent with Mark Spencer, Danny Windham and other Digium Asterisk developers during the week and over dinner. (And I didn't hear any complaints from Tony who was kidnapped by Mark for a birds eye view of Huntsville in Mark's Diamond DA40XLS!)

With their great training facility, the wonderful hospitality everyone received and the overall exceptional experience, how could we not plan on coming back soon. So if you couldn't make it last week, here's your next chance:

January 20th-23rd, 2009, again in Huntsville, AL

If you couldn’t attend this last opportunity or want to be one of the first to come to the next AsteriskNOW/FreePBX training, then sign up early while we are still offering an un-advertised EARLYBIRD discount of $300 until the end of this month.

For now, we hope to see you in Huntsville but it’s time to get back to work. One of our developers has done some nice improvements embedding the audio player into the User Portal (Recording Interface / ARI – no more pop-up!) so we are off to get that into a short lived release candidate for that component so we can get it out to you quickly. If you would like to help accelerate that activity, you can aways "press here":

Philippe - On Behalf of the FreePBX Team!

In part 1, we were discussing the basics of how the Asterisk dialplan works. To recap: asterisk is made up of contexts, which can in turn include more context, creating the whole dialplan. FreePBX takes advantage of this structure by creating a lot of contexts and then included these in each other. Until now, the easiest way to include your own custom dialplan was to put it in one of custom context that FreePBX intentionally leaves blank for the purpose of customization.

FreePBX was primarily designed to be a simple and easy to tool for programming asterisk dialplan and call flow. In the name of simplicity, however, it is sometimes necessary to sacrifice advanced features and overly complex ways of doing things. FreePBX takes a great middle ground in providing the best of both worlds: on one hand, an extremely powerful yet intuitive and simple GUI, and on the other hand a really neat way to seamlessly extend the gui into 'raw' dialplan. This is done using a combination of the Custom and Miscellaneous modules.

FreePBX 2.5 breaks down Time Conditions in to two separate modules: Time Groups and Time Conditions. The reason? To confuse you! Nay, we wouldn't do that. But in case we did, this should set the record straight.