Ask your self the following question: Do I have documented my dial plan? Do I know which contexts are in use and which contexts which are included where?

Do you have a document describing your dial plan? Probably not. Most of the people managing Asterisk by hand never care to document the structure of their dial plan - after all - It's a text file - documenting a text file is stupid work!

If your dial plan becomes quite extensive after a while - or if you become in charge of an previous Asterisk installation - you really wish for some documentation of the structure of the dial plan.

Even with just a few contexts it can become a daunting task to change an existing dial plan. Even if you do not need to really amend your dial plan - just having a graph of how your dial plan is organized is really a good thing to have in your PBX documentation.

There is a tool that will help you getting the big picture of your dial plan: JUNG (Java Universal Network/Graph Framework). The tool will not work with the Asterisk dial plan "out of the box", but Martin Smith of the Asterisk-Java blog have done all the hard work.

Even if the tool that Martin Smith created in May 2008 is "old news", his solution is so simple that it borders to ingenuity - it's worth repeating for new and old Asterisk manager alike.

Read his full article and feed your dial plan into his tool at Visualizing your dialplan with a graph

Some good news from the read worthy Skype Journal: Skype SILK codec in the IETF standards process.

This IS major news. As Phil Wolff points out, one of the three obstacles for Skype is now being solved. In my opinion, the codec issue has been one of the most difficult for Skype - along with the Joltid issues.

From my perspective, one of the good thing about the SILK Speech Codec being put through an ITEF track is that the codec is now becoming open and usable for more than Skype users.

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We all know the bad ugly truth: Most people do not update their PBX software to handle the latest security vulnerabilities. As long as your PBX can receive incoming client connections you are at risk. Not because you have given your user weak user name / password combinations, but because your PBX has a security flaw you did not know about.

Common solutions

Let's face it: PBX security is not as sexy as operating systems or web security. When did you last read about a security flaw in a PBX product in the main stream IT-press? Compare this to any mention of a OS or web security hole.

There are a couple of things you can do to make your PBX installation as secure as possible. The most obvious one is to have a strong password regime. There are also those who believe that strong user names are also the way to go. I will not deny that this is a bad thing per se, but it is not very user friendly.

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Nearly a month ago Steve Sokol announced that Digium is finally providing paid support for the Open Source version of the Asterisk software.

Back in March I did not want to comment on this particular news item. I would let the dust settle before commenting.

What implication will this have for the ecosystem around Asterisk?

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I consider VoIP Supply in the US one of the best VoIP on line shops on that side of the Atlantic. I really wish they had a European presence and when they decide carry a product I trust their judgment.

They also have a blog called VoIP Insider that is worth following. They have manged to walk the fine line between self promotion and really newsworthy information within the VoIP cloud.

When these guys say that they are going to sell a certain device - it's worth taking a closer look.

So when I learned that they are going to sell a IAX based hand set I became very interested for several reasons which I will address later in this posting.

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I do follow the trixbox community closely. I believe that trixbox is one reason why a lot of "PBX Built On Asterisk" companies will never have a chance in the long run.

So it is with quite some interest that I read a posting from Kerry Garrison in the trix box forum about a "fork" of the FreePBX front end.

The main reason stated is:

1. integrated GUI (so it looks professional when you sell it to your customers)
2. faster bug fixing
3. new features

Fixing the GUI so that it does at least appear more integrated with itself is paramount. Every time I work with the FreePBX GUI I do feel that it is a bunch of non-related packages thrown together.

It is also quoted that "We are utilizing what we call the patch-plus model (what we do with Asterisk, as well). This means that as future versions of Asterisk and FreePBX come out, we simply grab that code and apply our elaborate patch-plus script to it, and *wallah* you have the latest and greatest of FreePBX *plus* the latest and greatest of trixbox CE all in one interface." My question is - how long are they going to do that rather than create a clean fork. I have no reason not doubt that Fonality is trying to do the Right Thing - until proven otherwise I firmly believe that they are good guys.

For the sake of FreePBX I really hope the developers does not loose faith over this decision. For the FreePBX project this is a good thing - they get help fixing a lot outstanding issues. It is of course up to the FreePBX project to incorporate what they see fit into their own project.

In my opinion this could be a big win-win situation.