The prototype is a 2U appliance with dual redundant power supplies and with the capacity of 4 PBX Blades and one Server Blade. Each PBX blade has a tested capacity of more than 50 concurrent VoIP calls, and we have Blades with E1/T1, Analogue and BRI technology. The PBX can operate with or without the Server Blade and with any combination of the PBX Blades. And more Blades are currently under development.

The initial release will be with 1 to 4 separate PBX Blades in the 2U cabinet, but we are working on a solution that will make it possible to combine the power of all blades to a whopping 200+ concurrent calls. And this is only the beginning, a 4U appliance with a much higher capacity is in the works as well.

As everybody with children know, finding a good name for the last member of the family is a lot more difficult than the creation itself. But I think we finally got it;
VoIPtel X Series Communication Appliance, or VoIP X for short. Cute name for a great future! ;o)

More info and pictures will be posted in the near future.

New day, new chances. I dialed the UK number again and got … BUSY LINE! US number, same result! This couldn’t be right, I decided that we better check our own equipment before we attempted to attend the conference once more! A new conference was scheduled after the weekend, at least we would have enough time to find out what was going on.

Bruce logged in to our PBX and checked the config after which he tried to call an international number, do I need to mention that he got busy line? He tried another number watching the progress of the call and discovered that the call appeared to be denied by our VoIP supplier! Strange, to my knowledge all our bills was paid, was there really any reason why a VoIP provider would block international calls from a corporate customer?

We called up their support department and explained the problem, I prefer not to describe my reaction when I was informed that they had indeed blocked our international access! The reason; we were using Asterisk and one of their customers using a very old Asterisk distribution had been hacked and his account abused for more than $30 000. And our provider had forgotten to include a statement leaving the responsibility for such incidents with the subscriber. Needless to say they panicked, closed down international calls for all Asterisk users and “forgot” to inform their customers leaving us all in the dark! It took some time to convince them that our PBX was secure, but finally we were able to get full functionality restored.

I can understand their reaction (no, not their “customer support”, it sucks!), so I have decided to point your attention to a blog post titled Seven Steps to Better SIP Security with Asterisk. Please visit it and read it all, I have only published the seven steps here:

Seven Easy Steps to Better SIP Security on Asterisk:

1) Don’t accept SIP authentication requests from all IP addresses. Use the “permit=” and “deny=” lines in sip.conf to only allow a reasonable subset of IP addresess to reach each listed extension/user in your sip.conf file. Even if you accept inbound calls from “anywhere” (via [default]) don’t let those users reach authenticated elements!

2) Set “alwaysauthreject=yes” in your sip.conf file. This option has been around for a while (since 1.2?) but the default is “no”, which allows extension information leakage. Setting this to “yes” will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks.

3) Use STRONG passwords for SIP entities. This is probably the most important step you can take. Don’t just concatenate two words together and suffix it with “1″ – if you’ve seen how sophisticated the tools are that guess passwords, you’d understand that trivial obfuscation like that is a minor hinderance to a modern CPU. Use symbols, numbers, and a mix of upper and lowercase letters at least 12 digits long.

4) Block your AMI manager ports. Use “permit=” and “deny=” lines in manager.conf to reduce inbound connections to known hosts only. Use strong passwords here, again at least 12 characters with a complex mix of symbols, numbers, and letters.

5) Allow only one or two calls at a time per SIP entity, where possible. At the worst, limiting your exposure to toll fraud is a wise thing to do. This also limits your exposure when legitimate password holders on your system lose control of their passphrase – writing it on the bottom of the SIP phone, for instance, which I’ve seen.

6) Make your SIP usernames different than your extensions. While it is convenient to have extension “1234″ map to SIP entry “1234″ which is also SIP user “1234″, this is an easy target for attackers to guess SIP authentication names. Use the MAC address of the device, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, try “md5 -s ThePassword5000″)

7) Ensure your [default] context is secure. Don’t allow unauthenticated callers to reach any contexts that allow toll calls. Permit only a limited number of active calls through your default context (use the “GROUP” function as a counter.) Prohibit unauthenticated calls entirely (if you don’t want them) by setting “allowguest=no” in the [general] part of sip.conf.

A lot of time and money has gone into the development, the license cost was to pay Skype, and Digium need to recover the cost of development. It made sense to release it for intel based systems first as this was the largest segment of the market and since their own Switchbox uses intel. They realize that the Blackfin systems are very popular and Switchbox was looking at all chipsets to produce green products. It was expected that a version for Backfin chipsets would be made available in the future.

In other words, Digium is not sleeping in class but are aware of the GREAT and GREEN potential of embedded PBX’s based on the Blackfin DSP. But I still believe that they need a little convincing, so please keep on sending mails telling how much we want to integrate Skype!

And what about Skype for Asterisk made available last month? Sorry, only available if you run anything else than an embedded, Blackfin based PBX. Digium’s own AA50 is an embedded PBX based on the Blackfin DSP, I would be quite surprised if users of the AA50 want Skype for Asterisk less than the rest of us!

I have registered once more with the Skype for SIP program, hopefully this time we will be included. But I need your help! Please send loads of mail to both Digium and Skype, ask them why the Blackfin based IPxx PBX is prevented from Skype integration! If we scream load enough maybe they’ll finally hear us!

With the recent addition of an IP-PBX in our office, we started thinking about ways electronic health records (EHR) could be integrated with IP-PBX phone systems. After our initial research, we were surprised to find that little has been developed so far in the way of medical-specific applications.

Asterisk-based Internet Protocol – Private Branch Exchange (IP-PBX) phone systems could help medical practices be more efficient and provide better patient care. It certainly has helped us reduce costs and become more efficient.

To spark interest in the Asterisk development community, we’ve decided to put together a list of seven ways IP-PBX and EHR technology can be combined. The ideas range from automatic dunning (i.e. collections) voicemails to vPrescribing.

To read the full post, visit: Seven Great Applications for IP-PBXs in the Medical Practice

Please follow the link for the entire article, it is definitely well worth the time spent reading it. We are going to seriously consider developing a firmware based on Houston’s article and ideas and sincerely hope that he will be available for future discussions.

UPDATE: the site is online and available at http://voiptel-se.com

We recently signed a new contract with our dear friends at Atcom Technology Co. Ltd., the manufacturer of the famous IPxx series of PBX’s. They are now actively involved in the development and improvement of the VoIPtel firmware. There is no doubt that this will speed up the development of new features making the IPxx the preferred choice for a growing number of VoIP integrators.

And for those with special needs; there are no less than three new models almost ready for production:
IP-4B4A              Four BRI and 4 Analog ports IP PBX
IP-PRI                 Four digital ports (E1/T1) IP PBX
IP-4G                  Four GSM module IP PBX

For a complete list of all the avaiable models please take a look here:

But finally there appears to be light in the tunnel. The recent announcement of Skype for SIP Beta Program was simply too interesting to leave alone and yesterday I got a mail with the following content:

How is the beta program developing?
We have been testing the solution across different PBX vendors and some large partners to ensure any possible interoperability issues and potential bugs are resolved prior to the next phase of the beta program. In the next phase we will open up the program to all remaining applicants. We hope to have the next release out soon, at which time we will be contacting you again to provide instructions on setting up Skype for SIP. By providing your Skype Name at this stage you’ll be able to start using Skype for SIP as soon as the next release is available.

I can hardly wait for that next release; I believe that integrating Skype with open source SIP devices is a huge step in the right direction.

Another very good resource of Asterisk knowledge is the newly published trixbox CE 2.6 written by Kerry Garrison. As one of the reviewers I have spent a couple of hours with this book, and whether you are a dedicated trixbox fan or not, it’s well worth both the price and the time.  The book is written for those who want to learn how to install and configure either trixbox CE systems or Asterisk-based PBX systems, without struggling with confusing configuration files and cryptic scripts. It is ideal for any user wishing to set up a telephony system for small business usage. No previous knowledge of trixbox or networking is required, although some basic knowledge of PBX and Linux would be an advantage.

This year we were a little late trying to get a place to stay during the Messe and ended up in Hamburg, one hour and twenty minutes drive from Hannover. It wasn’t until we arrived at CeBIT and met up with Job from our DC in Holland that he told us he had been able to find accommodations almost within walking distance from the fair. So next year I think I’ll leave the preparations to Job, it’s going to save me a lot of time and money.

The size of CeBIT is huge, I am under the impression that it covers almost the same area as the entire Bergen centrum, my hometown. So if you are planning to visit the fair, be smart and do your homework. It is simply impossible to cover everything. I won’t go into a lengthy description of the exhibit itself, it is thoroughly covered in various media. For me it is the small incidents that is interesting, incidents like bumping into the OEM manufacturer of the SNOM M3, or being offered to become the representative of a company that have stolen most of their PBX promo material directly from our web site! But more on that later. I found the meeting with my contact in Planet Corporation much more interesting, we are both interested in an even closer relationship in order to improve the support to our customers. For quite some time I have tried to locate a distributor of Grandstream IP phones in the Scandinavian countries to no avail, I believe that their products are both inexpensive as well as well designed and of relatively good quality. I was lucky enough to find their booth and ended up discussing the possibility of becoming their Scandinavian Partner! That discussion is not completely finished yet but I will post an update as soon as I have something to share.

I spent most of my time together with Peter, Edwin and Gottenphone from Atcom discussing the future of our products. Most of it is still to early to reveal, but some very interesting concepts and designs are currently under development in the PBX area. They are currently doing the final testing on two new corporate IP phones, one of them with a detachable extension console. And we have got a new Dual analog module for the IP02 and IP08 with one FXO and one FXS port. But for us the greatest news are the contract that was finally signed today, a contract where we will be responsible for the firmware installed on all IPxx PBX’s by default. And a brand new IPxx PRO series of PBX’s complete with VoIPtel SE and SEq firmware will be released in the near future. For those of you who want better security, the new AT-530P+ OpenVPN phone is currently being tested together with an IPxx with VoIPtel SEq firmware. More info will follow soon ;o)

We are preparing some very promising projects with our close friends at Azuralis as well, the same guys who convinced us that it was a good idea to integrate VoIP over VPN in our firmware. This time we are looking at the quality of the line from your provider, quite important if your business depends on VoIP trunks.

And last but certainly not least, a large firmware upgrade will be available next week for both VoIPtel CE and SE. Several new features as well as bug fixes are included so make sure you upgrade your PBX. The upgrade can be performed through the GUI.