FreePBX 2.6 Final, SIPSTATION Module and other progress

A lot has been happening and we have been so busy it seems the only time to escape and get some information out is getting on a plane. (And that is quickly disappearing with in flight internet, luckily not available on today's flight...).

A few things have been happening since I last blogged and it's time to give a quick update. We brought 2.6 out of beta just in time for Astricon 2009 two weeks ago. This was a particularly exciting Astricon. It was both the 10th Anniversary of Asterisk and the 5th for FreePBX. We hung out on the exhibit floor in our Open Source booth with some great props (thanks Sledge!). (I'll try to get some pictures up here in the next day or two).

The 2.6 release had over 6000 users at the time we took it out of a release candidate! (I still need to generate the final tarball and update the site in a few place, sigh, but you can update through the online Module Repository for now) It contains quite a few new features and a handful of "hidden" changes that I always refer to as internal plumbing upgrades. One change we introduced is the Extended Repository. This provides a mechanism for us to expose some of the more common “un-supported” but otherwise popular modules that are part of FreePBX but have required finding them and manually installing them in the past. There are also a handful of new modules so if you haven't upgraded yourself from 2.5 or earlier releases, several thousand of your fellow community members should have paved the way for you to have the confidence of pushing that upgrade button.

We are also excited to push out a new module that interacts with our SIP Service, SIPSTATION™ that is available for purchase online. The service has been really well received and has been a great way to get access to the best SIP Service available (powered by's purpose built VoIP network) and support the project at the same time. You can now use our new sipstation module to auto configure your services once purchasing them online from the portal. In it's first version, the module provides a very easy ability to connect to our redundant gateways, provide realtime diagnostics on both server side and client side status, and a central dashboard and control point to easily configure your service into existing outbound routes as well as setup your DIDs and outbound CIDs, all in one central location. It even has a really nifty Firewall Test diagnostic which helps you determine if your RTP media ports are properly forwarded from our firewall, since most of us put our systems properly behind a firewall and have to deal with the realities of NAT.

This is the first step in closer integration that we will build on bringing client side control of services that you obtain through our offerings. We will expand on this to bring features like PSTN failover numbers per DID directly configurable from within FreePBX and other ideas that are on the drawing board. If you haven't tried our service and you want to make your life easier, here's another reason to give it a go!

On the FreePBX v3 front, a lot of progress is being made though it continues to be in a developer release state. The Asterisk driver is well on its way and work continues on both the Asterisk and FreeSWITCH front. This has been really exciting as it's an important milestone in proving out the design of writing a GUI and framework capable of supporting multiple engines, a goal that has been in place since early 2006 when AMP was renamed to FreePBX and introduced the first generation of modular architecture present in the 2.x version.

For now, it's time to sign off (at least that's what the flight attendants are telling us, in preparation for landing...) If you haven't pulled the plug and upgraded to 2.6, now's the time! If you want to hop on the development train of v3, come chat in the #freepbx-dev IRC channel. Lastly, if you want to see a really cool module (which is just the beginning), go sign up for some service, get your keycode and activate the SIPSTATION™ module to get a glance of the ease of use that tight integration can deliver!

Philippe – on behalf of the FreePBX Team!

I setup a trunk with DID using SIPSTATION. Pretty cool. Only quibble is that it setup two trunks, primary and secondary. and the primary defaults to the west coast, and the secondary to the east coast, with no obvious way to change that in the sipstation module itself - I was able to “fix” this by editing the two trunks, but then sipstation gets worried since it is not a standard configuration.

Also, my DID does not work - calling it results in a busy signal, no indication it is even hitting my server. Sent an email to the support address an hour or so ago, with no reply. Some kind of ticket system would be nice :frowning:

For those of you who are still running 2.5 and don’t want to update yet, I just published the sipstation module that let’s you auto-configure the SIP trunks on your system.

While we still like to see our users update to the latest versions (it makes our lives simpler:) we understand the prudence of not upgrading just for the sake of upgrading so you can now have easy access to the new module and not have to upgrade right away.

The primary and secondary gateways are for SIP signaling only so the East/West coast locations are not an issue. Media will flow from the media servers which are neither of these systems and vary based on the call, unless you force the media through the servers from the portal.

I’ve been using for a month or so for trunks and DIDs - will installing this module (freepbx2.5) cause any issues? In other words, does it handle already having setup stuff? I mainly want it to help deal with monitor and maybe work with DIDs.

----Update on 11/26
After backing everything up, I went ahead and tried it out and it seems to work fine. I did have to delete my original trunks in order to let the module make its own copies, but that was no big deal.

One issue, in the list of DIDs in the module, it has a checkbox to add the outgoing caller ID which is nice. However, when used, it adds only the number, and not bracketed by <> which I thought was required? In addition, it does not add the name on the extension, such as: “Jane Doe” <4155551212>
That would be nice, even if not supporting CNAM yet.

But at least it is pointing out where staff forgot to set that field so I can have them fix it.

It doesn’t matter what you set the outgoing CNAM to - that does not get passed to the PSTN - the receiving telco has to do a LIDB dip (if at all) to provide CNAM to person receiving the call.

I am very happy - I have had two issues that impacted calls - the first was some issue with my DID that was cleared up quickly. The second was an issue that cause no audio on inbound calls. It turned out I had broken DNS on my end, so asterisk was sending a LAN IP as the contact IP - once I fixed that, all was well. I am happy enough that I have sent an email asking how to port my voipo number in - I do not expect that to be painful, since (AFAIK) the CLEC is already :slight_smile: My only negative: I really think you need to improve the ticket system - currently it is emails, and there is no way to know what is going on or get status, short of waiting for a return email.