New Release is a Great Success – Las Vegas for Training?


FreePBX 2.4.0 Looking Great

It has been about a week since FreePBX 2.4.0 went final and we already have over 3000 installations running, of which over 2200 of them have successfully upgraded with the new 2.3 to 2.4 upgrade module that was published just 5 days ago. (Not counting the hundreds of 2.4 rc1 and beta systems which have not yet upgraded but some of which are probably active.) Given the almost deadening silence in the bug tracker, we might mark this as the most successful new release to date!

For those of you wondering what we are seeing in terms of Asterisk versions these days, on 2.4 installations it is just about 70% Asterisk 1.4 and 30% 1.2, and of course those few systems living on the edge beta testing 1.6 for the community.

If you are running version 2.3 you can press the button below to purchase your guilt free upgrade pass to benefit from all our hard labor in making this upgrade so easy and so successful for you!


Open Telephony Training Seminar – West Coast

With our first training class sold out and just a week away we have gotten inquiries asking when and where our next training event will be. We are currently in discussions to see if we can bring you the next event in Las Vegas, NV the week of April 8th, 2008 which is coming up quick. We are looking at this location to help those on the West Coast who did not want to come clear across the country to Charleston. Las Vegas also has the benefit of low cost airfair and the availability of very reasonable hotel rates. Whether you end up winning the cost of your training at the craps table we can not say but from what we’ve heard from some of you, this would be a good location.

Please reply to this thread if you are considering coming and let us know if this works for you. We are always open to hearing what might work – but if we are to get this next one scheduled soon we need to move quickly!

Philippe on behalf of the FreePBX Team!

FreePBX 2.4.0 – It’s Final!


This year is starting out with a real bang for the FreePBX team as we once again come to you with a lot of great enhancements and features on the new 2.4.0 Final Release. It has been an exciting beta program that we officially launched 9 weeks ago and has been tested by over 1300 systems with fantastic feedback from our community.

The list of features, enhancements and bug fixes (most bugs from version 2.3 and prior) is much to long to share with you here but some highlights include :

  • Simultaneous support for Asterisk version 1.2, 1.4 and 1.6 (best effort for 1.6 as it is still in beta)
  • New Voicemail Blast Group Module
  • New Language Module
  • Vast improvements and many feature enhancements to Paging and Intercom Module
  • Internal improvements and several new features to Queues Module
  • DUNDi™ Trunk support with automatic dialplan integration
  • Vast improvements to handle Zap Channel Inbound routing and incorporating Zap Channel + CID routing abilities
  • System wide extension and destination registry and automatic conflict and integrity detection
  • New Custom Apps module to integrate custom applications and extensions into the above registries
  • Vast improvements to Devices and Users mode especially around adhoc devices
  • Call Confirmation support for hunt ring strategy in Ring Groups and Follow-Me

And the list goes on! Despite the many new features introduced, there has been a focus on building capabilities to help the administrator maintain internal integrity and detect issues when they occur. This effort started back in version 2.3 when we introduced the Notification Panel in the System Status Dashboard and continues to evolve as we report on more error conditions, and as we introduce more mechanisms such as the Extension and Destination registries and internal integrity checking ability as mentioned above.

So how do you get 2.4.0 onto your system when we told you that you would be able to upgrade your 2.3 systems from Module Admin? After spending all day yesterday writing and testing the upgrade module that will enable this, my family informs me that I will be taking them out to a really expensive restaurant for abandoning them (and not just yesterday!) so I’m holding it up for ransom. Once enough people use the following button we’ll get it out!


Seriously though … the donations will be much appreciated but the Update Module is on its way and will be published and available on 2.3 systems within a few days. We are delaying it just briefly for two reasons. We are monitoring the conversion rate of 2.4 beta test sites to the final release to make sure there are no issues reported in the last bug fixes that got rolled in since RC1 was released. We are also asking some of the community to beta test the upgrade module to make sure it functions smoothly when we publish it. You can get information in this forum post if you would like to help.

You can also go the traditional route to either upgrade or build new installs by using the standard install and upgrade processes listed in our site’s Upgrade Instructions.

If you’ve been following the other activities happening here at FreePBX then you probably did not miss the Open Telephony Training Seminar we have been talking about. We are thrilled at the response we have gotten on this upcoming event which has now completely SOLD OUT more than two weeks before its start date! These lucky participants will be the first to receive formal training based on the 2.4.0 release announced today, including many internal capabilities not always exposed in the GUI interface. We have had many inquiries from would be participants who could not make those dates or were looking for something a little closer to their homes and we are happy to say that we will soon be choosing and organizing the next event to accommodate those requests.


With the current training being held in Charleston, South Carolina it is only natural to plan the next event on the other Coast. We have not made any choices but are considering locations like Las Vegas (or just outside to keep the distractions during the event down a bit) or San Diego but we want to hear from you both where and when we should hold this. So feel free to reply and give your feedback to if you are hoping to attend one of our next sessions!

For now, go check out the new 2.4.0 Release and if you like what you see, or want to get the 2.3 to 2.4 Upgrade module out quicker, go press that donate button above to show your support and appreciation for all the great work and hundreds of volunteer hours that have gone into bringing you this new version!

Philippe on behalf of the FreePBX Team!

Asterisk Voice Recognition Company Directory

Along with requests for the Asterisk Voice Recognition "Magic Button", I have had numerous requests over the last couple days for the Asterisk Voice Recognition Company Directory (AVRCD). As promised before, I am releasing the source for the company directory. Drop to the bottom for links to a demo and all the source files for the Company Directory or keep reading for details.

This isn’t your Daddy’s company directory. My goal in this project was to not only create a voice-enabled company directory for Asterisk, but to also extend the functionality of the company directory in the process. It took over 200 hours and utilizes some of the most advanced web-based technology available for an interactive interface.

The AVRCD is tree-based with advanced web-based drag-and-drop technology that lets you drag the extensions you want in the company directory to a custom tree that you create. You can create sales, customer service, marketing, accounting, and whatever other "department" folders you want and have employees specifically in that department folder. When navigating the tree through voice commands, you could for example, say "support", then "Ethan Schroeder" and be connected to me. You can configure each extension to playback confirmations by either pointing an extension to the user’s voicemail name recording or system recording. If there is no audio file selected as a confirmation, the system falls back on text-to-speech using Flite.

The AVRCD also has custom entries support. These could be entries for ring groups, off site IAX or SIP users that don’t have an extension on the main PBX, etc. You can then drag these custom extensions to the tree, just as the built in extensions. These custom extensions also support custom dial strings, so if you do have a remote PBX, you can have it dial that person over
the IAX trunk through the custom dial string.

The company directory, once setup, automatically configures itself to your existing PBX users. All you have to do is drag them to where you want them, and you can drag the same person to multiple directory folders. It sets up "pronunciations" for you based on their names. These pronunciations are editable by you on a per-extension and per-tree-item basis. So if you
are getting complaints that "Michael Smith" is getting calls to "Michelle Smith", you can phonetically help the speech engine by writing custom pronunciations for each name. For example, "Mishell Smith" and "Mikall Smith".

Now, all of this said, the company directory is definitely in beta form. It’s using cutting edge javascript/ajax/ui libraries, and I’m going to be honest with you, I hadn’t worked in client-side application development for years and never OO Javascript, though I was once a Java developer. I just squashed some of the last UI bugs I could find, but there may be more. In addition, it doesn’t support voice error handling. This basically means that it isn’t threshold aware on the voice recognition level, meaning that if the application "isn’t quite sure" what was spoken, it’s probably going to make a choice anyway. Ideally, this is handled by determining threshold acceptability levels and if a recognized pattern is below that threshold it will prompt with options. But it’s free, so you are encouraged to use it as you see fit.

I’m too busy working on material for the Open Telephony Training Seminar to setup an entire demonstration PBX for everyone to try it out, but I have posted a live demo of the interface. All the source is here, which includes PHP, Javascript, dial plans, and AGI. You’ll need to setup and install LumenVox and Flite. Lumenvox has a Linux/Asterisk developer version available for around $50, or you could attend the Training at the end of February to get a free LumenVox license and the Asterisk Voice Recognition "Magic Button".

Lastly, as with the Asterisk Voice Recognition "Magic Button", this project deserves to be a FreePBX modular for integration more tightly coupled with configuration of extensions. In addition, this is a voice only directory. There is no support for dtmf entry for "the first three letters of the first or last name". This functionality is definitely on the todo list, but won’t be useful to accomplish until the FreePBX integration occurs. FreePBX needs your support to make things like this a reality, so consider attending the training at the end of the month. The training is going to reach capacity, so hurry up and register. If you miss this one, stay tuned for announcements in the coming month on subsequent training.