FreePBX Distro Release tracks- The state of the FreePBX Distro

Lately there has been some confusion on the different FreePBX Distro versions. I am going to take a moment here and try and explain how this all works. The FreePBX Distro is a CentOS Distribution that includes a specific version of CentOS, Asterisk and FreePBX.

Until this week we offered the following 3 tracks. Think of each track as their own release cycle.

1.8xx.210.58 – STABLE This was the first track we released and includes
[list] [*]Asterisk 1.8.x
[*]FreePBX 2.10
[*]CentOS 5.8
[/list]

1.10xx.210.58 – STABLE Includes
[list] [*]Asterisk 10.x
[*]FreePBX 2.10
[*]CentOS 5.8
[/list]

1.10xx.210.62 – BETA Includes
[list] [*]Asterisk 10.x
[*]FreePBX 2.10
[*]CentOS 6.2
[/list]

Within each track we offer [url=http://www.freepbx.org/forums/freepbx-distro/distro-discussion-help]Upgrade Scripts[/url] that generally upgrade published CentOS packages, asterisk, DAHDi and FreePBX. Generally you can not upgrade between tracks but we do make exceptions if the track is on the same main CentOS release such as CentOS 6.x or 5.x and we have set a track to be End of Life. We specifically don’t offer upgrade scripts to go from a CentOS 5.x to 6.x released distro such as 1.10xx.210.58 to 1.10xx.210.62 since CentOS has not provided any such upgrade and states the only way to upgrade to CentOS 6 from 5 is to do a new install.

Now that Asterisk 11 is out we have decided to make some changes to our tracks, with community input [url=http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/distro-input-needed-by-development-team]Which can be see here.[/url]

The new tracks are:

1.8xx.210.62- STABLE Includes
[list] [*]Asterisk 1.8.x
[*]FreePBX 2.10
[*]CentOS 6.2
[/list]

1.10xx.210.62- STABLE Includes
[list] [*]Asterisk 10.x
[*]FreePBX 2.10
[*]CentOS 6.2
[/list]

1.11xx.210.63- BETA Includes
[list] [*]Asterisk 11.x
[*]FreePBX 2.10
[*]CentOS 6.3
[/list]

We have removed support for both of the CentOS 5.8 based track releases. The plan is to keep offering upgrade scripts on the 1.8xx.210.58 track for asterisk and DAHDi for the next 6 months since we can not provide any clear upgrade path to the 1.8xx.210.62 track that is based on CentOS 6.2

As we watch the 1.11xx.210.63 track stabilize we will be dropping the 1.10xx.210.62 track but will provide an upgrade path from 1.10xx.210.62 track to 1.11xx.210.63 track since they are in the same CentOS 6.x family. With Asterisk 10 not being a Long Term Release (LTE) from Digium we have no plans on supporting it very long and moving everyone to the Asterisk 11 track since this is their LTE.

As a recap the current state of the 5 different FreePBx Distro release tracks are;

1.8xx.210.58 – STABLE – END OF LIFE. Receiving upgrade scripts for new Asterisk 1.8 and DAHDi versions only. Will not be building any new ISOs for this version. Recommend using FreePBX backup and restore to move to a Newer Centos 6.x track of the ISO long term.

1.10xx.210.58 – STABLE – END OF LIFE. Will not be receiving any Upgrade scripts and recommend using FreePBX backup and restore to move to a newer supported version track.

1.8xx.210.62 – STABLE – PRODUCTION- Recommended track for Asterisk 1.8

1.10xx.210.62 – STABLE – PRODUCTION- Recommended track for Asterisk 10

1.11xx.210.63 – BETA – PRODUCTION- Recommended track for Asterisk 11

You can find the upgrade scripts for each version track under the Distro Forums as stickies at the top. [url=http://www.freepbx.org/forums/freepbx-distro/distro-discussion-help]Upgrade Scripts[/url]

Off to Astricon 2012

Yet another year has flown by and the annual event of all things Asterisk is upon us.

Once again I have the distinct pleasure of joining the FreePBX team to have the opportunity to share about our great project.

It’s been a busy year, 2.11 development in full force and four versions of the Distro project.

One of the new items is Tony Lewis’s “Wall of phones”. In the FreePBX booth will be working examples of every commercially available (and some discontinued) endpoints form Aastra, Cisco, Panasonic, Yealink, SNOM, Mitel and a host of others. All of these phones are powered up and talking to a live FreePBX system. This is a great opportunity to see all of these endpoints in one place.

We also have the largest team assembled. Venerable leader and software engineer Philippe Lindheimer will be heading up the group along with Tony Lewis, President of Schmooze communication, Bryan Walters (gamergamer) Developer Schmnooze Communications, Andrew Nage developer of the FreePBX endpoint manager module and provisioner.net fame and lastly myself Scott Holtzman (SkykingOH) CTO of Micro Advantage.

Please stop by and say hello. Be sure and attend Philippe’s “State of FreePBX” presentation.

Look to the site for updates from the show and other attendees blogs.

Update from the road. The team is tired, the jokes are stale, Tony just set the brakes on fire descending out of the Smokies with the trailer. Bryan and Luke have been digesting their meal in a most ungraceful manner and we are almost out of Red Bull and Marlboro’s. While back at Our NW development center fearless leader (Philippe) is resting comfortably and preparing for his First Class flight to Atlanta. What we won’t do for the project.

We would have taken FreePBX One to Atlanta but we could not get the booth tied to the landing gear.

[Img]http://www.freepbx.org/files/images/CLE-Plane-Outside.jpg[/img]

More Routing and Trunking Enhancements in 2.11

Back again with a few more features being added to the Routing and Trunks category though this time targeted at 2.11. Tony told you about the [url=/news/2012-09-20/freepbx-extension-routing-module]Extension Routing module[/url] a week or so ago which resulted in a lot of positive feedback and happy community members who have wanted a simple solution to this problem. While we were messing around with this part of the code I thought I’d address a handful of other requests that have been outstanding in this area!

To recap [url=/news/2012-09-20/freepbx-extension-routing-module]Extension Routing[/url], this was the introduction of a module available on 2.10 that allows you to restrict extensions to certain routes in a simple and easy to understand way. [url=/news/2012-09-20/freepbx-extension-routing-module]Tony’s blog post[/url] goes into a lot more detail if you didn’t get a chance to read it.

[b]Outbound Route Destinations[/b]

The first of today’s highlighted features is the addition of an optional [i]Destination[/i] that can be chosen for an Outbound Route. This dialplan destination would be followed if all the trunks configured reported some form of CONGESTION and you wanted to do something more creative [float=right]Route Destinations[/float]with the call then simply playing one of the messages configurable from the Route Congestion Messages module. A simple use case for this might be a custom announcement for all 900 phone numbers informing your users that these numbers are not allowed. You can route a call to any other destination you have on your system where I’m sure our user base will come up with all sorts of creative uses for this feature!

[b]Outbound Route Recording[/b]

When we re-engineered Call Recording in 2.10 we added the ability to force a call to be recorded based on the inbound DID it came in, within Modules like Ring Groups, Queues and Conferences or though a specific call flow directive. The last loose end was forcing all calls out a specific route to be recorded, just like the setting with inbound routes. That’s now been implemented in 2.11.

[b]Trunk Fail-Over on Busy[/b]

Something that comes up repeatedly in the forums are users running into carriers that don’t know how to signal properly. The carrier will send back a BUSY when they should have been sending back a CONGESTION. A BUSY is suppose to mean the far end destination you just tried does not want or can’t be bothered at the moment. Given this ‘proper’ interpretation FreePBX does not bother to fail over [float=left]Busy Trunk as Congested[/float]to the next trunk since the message was clear, [b]THEY ARE BUSY![/b], and another trunk is not going to tell you something differently! In order to get around these carrier issues, we’ve added a per-trunk feature so you can configure any one or multiple trunks to ‘always’ fail over to the next trunk if they can’t get the far end ringing. This is not limited to the BUSY scenario, your carrier might be signaling a number as invalid because their switch is programmed improperly or for other reasons. When configured, this will always overflow to the next trunk or configured destination on an un-successfrul call attempt.

These new features will all require 2.11 to take advantage of and with Astricon fast approaching, I’ll try to get a proper 2.11 beta tarball rolled this week so you don’t have to pull these from SVN if you want to get started with them early. Of course don’t let me stop you from grabbing the code now!

Speaking of Astricon, a bunch of us plan on being there this year and we’ll have a booth as well so come by and say hi and see what we are up to!

For now, give us feedback on these features or other other ideas this might trigger since it’s always a good time to make sure “long ignored features” show up on our radar when we are in a push to get a release finished!

[b]Philippe[/b] on behalf of the FreePBX Team!

[b]P.S.[/b] We haven’t touched on the [url=/news/2012-08-16/seeking-feedback-on-new-website-design]New Website Design[/url] in a while. We’ve been looking for an experienced Drupal developer to help with the implementation of the new design. This includes both the Drupal backend configuration and migration as well as Drupal Theme design for the new look we are shooting for. If you know someone you can recommend, can you please PM me with that information? We have funds to do this so it doesn’t have to be free though it isn’t going to be a ‘huge’ project either. Thanks!