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

I have never really bothered to dig into YATE. I have seen this project as a small niche product in the Open Source Telephony sphere. That said, YATE have quite a few nice features.

With the release of Freesentral I finally have a excellent reason for taking a look into YATE.

Even better, if you need a simple way to set up an IP based PBX I can really recommend the Freesentral GUI. The GUI is clear and simple - really uncluttered compared to other front ends.

Even if the project seems to be very young, the GUI is rock solid.

In my opinion, the only thing missing is a ready-to-play distribution.

If you are tired of FreePBX and it's cousins - and if you need something simple and easy to us, you should really check out Freesentral.

At first sight, using any VoIP client on the iPhone or the iPod Touch (a.k.a. iDevices) may seem like a uninteresting thing. The reason for this is that Apple does not allow 3rd party applications to run in the background. So when a user close down his iVoIP Client he will not be able to receive any calls at all, thus defeating the reason for using VoIP on these devices in the first place.

However, if we take a look at some of the VoIP clients offerings available we notice that a few of these clients have the ability to receive incoming calls, even when the software it self is not running.

At first sight this seems to be a Good Thing - however, there are severe security implications by doing this. Users will in fact willingly, put them self under a man-in-the-middle attack.

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Back in August 2009 I did a guest appearance on the VoIP Users Conference speaking about VoIP clients for mobile telephones. I briefly touched on to a few offerings for the iPhone and iPod Touch (a.k.a. i-Devices). This inspired Randy a.k.a @voipusers to do a review of a few of the available offerings on i-Devices.

His article, along with my Fianceè buying herself an iPhone, prompted me to take a closer look at a few of the offerings available.

I have given SIP clients on the iPhone and iPod Touch a generic name: iVoIP Client(s). Likewise, an iDevice is a generic term for both the iPod Touch and the iPhone.

The following clients have been tested:

• iPico from MailVision Ltd
• iSIP from VNET Corporation
• WeePhone from Justin Bray
• Acrobits Softphone from Acrobits
• Media5-fone from Media5 Corporation

For completeness I have also taken a quick look at the SIP function in both Fring and Nimbuzz, two free (as in beer) offerings available for the iDevices.

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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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Some time ago I did a posting about VoIP Supply announcing that they would begin to sell a IAX phone. At the time of release, they had a contest where one could win one of these phones. Usually I am not very into contests, but when someone asked me nicely if I was willing to write a "Why is IAX {cooler,better,more suiteable} than SIP", and submit this to their contest, I did oblige.

Given that VoIP Supply is one of the few companies around that I consider myself to be a fan of, I had no problem doing this. VoIP Supply are very good, and if they did ship their hardware with Euro-type power adapters I would probably be buying most of my stuff from them. Unfortunately they do not.

Now back to the Citel C-4110 VoIP Supply sent me back in May 2009.

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After the initial feedback on lack of Asterisk support for FreePBX, the dust have now settled.

But let's take a step back in time - approximately 15 months. Back then, the Trixbox people decided to spawn off their own version of FreePBX. I then stated that I really hoped that the FreePBX developers would not loose faith over this decision.

Fast forward to the last days announcements regarding version 3.0 of FreePBX.

Granted - it does not yet have Asterisk support, but keep in mind that this is a developer release, this means the support will eventually be there. As I will show in this article, is really does not matter if there is no Asterisk support yet.

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