Positron Signs VoIPtel Europe Ltd for Distribution

November 27th, 2009

- Positron Telecommunication Systems Inc. announced today that VoIPtel of Norway is an official member of the Positron Partner program and will soon be reselling Positron solutions to the marketplace. This represents the first steps for VoIPtel to resell the powerful and flexible Asterisk based solutions from Positron who offer the first integrated “Blade” solution.

“VoIPtel is an established vendor in the open source marketplace who focuses on efficiency, value and green products,” said Richard McGravie, President of Positron Telecom. “They bring along a lot of experience and knowledge of the VoIP market and will allow us to not only serve Norway but the entire Scandinavian and European market place.”

“At VoIPtel we believe in sourcing and selling only quality products to our customers,” said Jan Bjorkhaug, CEO of VoIPtel. “We were impressed by the hardware build and software quality of the Positron products. Not only have they created interesting products that address the market but they are also correctly priced for the small business, a fact that many vendors miss. The Positron solutions are feature rich and come in many different form factors to address our customers’ needs.”

Positron Telecom has created a partnering program to meet the many needs of the market place. Through innovation, support and sharing of ideas, this program can be profitable and beneficial to all parties. The program is designed to establish distribution and reseller partnerships with independent software vendors (ISVs), systems integrators, consultants and other technology companies. These companies will benefit from integration with, or distribution of, Positron products globally or within a specific territory

Many of today’s telephony solutions were architected over five years ago and struggle to blend within the IP world. Positron Telecom breaks down that barrier. Traditional telephony PBX boards only provide the telephony interface, which have complicated drivers that can have operating system issues and require a complex installation process that is prone to failure. The Positron Telecom solution integrates the PBX, incorporates telephony and IP ports and installs as an Ethernet adapter, which makes its operating system independent, thus requiring no additional drivers.

About VoIPtel
VoIPtel was established as a division of Netsecur a/s in 2005 and became an independent corporation in August 2009 under the name VoIPtel Europe Ltd. The main focus of VoIPtel is Asterisk based products and a series of environmentally friendly PBX’s. Development of an improved firmware for the IP04 open source project by David Rowe was started in November 2007 and became known as the VoIPtel GUI. In December 2008, a new and greatly improved firmware named VoIPtel CE, followed by VoIPtel SE, was launched. This is now the default firmware installed on all IPxx PBX’s at the factory.

VoIPtel is now strongly focused on developing a series of communications appliances called VoIP X, which will be powered by Positron Telecommunications. Based on the 2U and 4U form factor and a capacity of up to 18 Blades, every eventuality is covered. VoIPtel Europe Ltd. is located in Bergen (Norway), Breda (the Netherlands) and London (United Kingdom)

About Positron Telecommunications System Inc.
Positron Telecommunications Systems Inc. creates, develops and markets sophisticated VoIP equipment for enterprise communication and collaboration through communication service providers. The company’s products integrate VoIP and traditional telephony in stand-alone systems that combine ease of use with powerful functionality. Its VoIP devices connect analog devices (telephone, fax and modem) to an IP-Network and gateways; connect PSTN users to an IP Network or analog PBX, as well as Key Systems to an IP-Network.

For more information, please contact:

Mireille Alvo
Marketing Specialist
Positron Inc.
Tel: (514) 345-2220 ext 2906
www.positrontelecom.com

Jan Bjorkhaug
CEO
VoIPtel Europe Ltd.
Tel: (47) 55 69 80 12
www.voiptel-pro.com

Media Contact:
Positron Telecommunication Systems Inc.
Mireille Alvo
514-345-2220

Press Release

Asterisk Developer Community Growth

November 23rd, 2009

Did you ever wonder when Mark Spencer created his first version of Asterisk? Or the number of people actually contributing code actively to Asterisk, the PBX platform that have had such a great impact on how we communicate not to mention the dwindling prices we pay for the means needed to do this? Some time ago I prepared one of my posts on this Blog, and I spent a considerable amount of time looking for the right answers. No need to do that again, Russel Bryant just gave us the whole story. If you find it interesting please visit his Blog, you find the link at the bottom of this post.

Asterisk trunk is the main development branch for Asterisk. This is where we are preparing the newest changes for the next major release. For example, new features that go into Asterisk trunk today will be first available in Asterisk 1.8 (wait, what?! 1.8?! Yeah, yeah. I’ll get back to that in a bit!) Asterisk trunk stays very busy. Here are some measurements regarding activity in trunk over the last year:

  • 2320 commits
  • 825 files changed
  • 322148 Lines of code added
  • 53251 Lines of code removed

A lot of people contribute to Asterisk. Among those writing code for Asterisk, there is a select group that has direct access to make changes to the source (committers). At the beginning of the project, there was effectively one committer, Mark Spencer. As the project has grown, we have worked very hard at scaling our development community such that we can process more code. The number of committers today is over 50 (with the number of contributors much higher than that).

In order to contribute code to Asterisk you will have to sign an agreement with Digium, and so far over 800 people have signed up to contribute to Asterisk over the past couple of years.

The history of Asterisk releases begins 10 years ago. Here are some dates on releases during the first half of the project’s lifetime:

  • 0.1 – December 1999
  • 0.2 – September 2002
  • 0.3 – February 2003
  • 0.4 – April 2003
  • 0.5 – September 2003
  • 0.7 – January 2004
  • 0.9 – April 2004

If you’re reading this and have been using Asterisk long enough to remember these releases, cool! That means you were involved in the project longer than me. I got involved in the Asterisk project in the middle of 2004. By the first Astricon in the Fall of 2004, Mark Spencer decided to release Asterisk 1.0 and asked me to maintain it.

  • Asterisk 1.0
    • Fall of 2004
    • Regular 1.0.X updates with bug fixes only
    • Eventually went into security only maintenance, and is no longer maintained today

Asterisk development continued and over the next couple of years and we released Asterisk 1.2 and Asterisk 1.4. We made some changes to development processes regarding how and when to port bug fixes and how they were merged between releases. However, the release policies regarding what went into updates and roughly how often they were released didn’t change.

  • Asterisk 1.2
    • Released November of 2005
    • Still updated with security fixes only
  • Asterisk 1.4
    • Released December of 2006
    • Still fully maintained

Source: Asterisk Project Update @ AstriCon 2009

blockquote>

Knowledge, The Past and the Present

Wanna join us and play? We got Toys for the BIG boys ;o)

November 9th, 2009

For the last three years I have been looking for parts, those very special parts that would permit me to built a very special line of PBX’s. My requirements was tough; two basic appliances should cover at least ten different models, the PBX should have the capability of growing more powerful and with more features when the need arise, quality should be as good as the BIG manufacturers, it should be based on Open Source software and last but not least, High Value For Money! Guess what! The first prototype was running this afternoon!

The prototype is a 2U appliance with dual redundant power supplies and with the capacity of 4 PBX BladesDual Power 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.Analog Blade 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.CPU Blade

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.

Knowledge, Open Source; Good Business or Dangerous Adventure?, PBX stuff, Unified Communications

They look good and sound even better!

November 7th, 2009

We have finally done it! After months of testing we are (almost) ready to launch our initial two phone models! And the heading don’t lie, at least in my opinion they both look and sound just great, and the price is probably better than any other VoIP phone out there, at least those that support both  SIP and IAX2. The only thing remaining is for me to finish writing the manuals, convert them to FLASH paper and post them online.

Our entry level phone is capable of two SIP and one IAX2 line. This is not the phone I would have chosen for a big corporation, but for a small company or a home office it fits the bill perfectly. It should be available from our shop in a week or so, at least if I am able to keep my schedule.

SOHO

I am pretty sure that the second phone will be the answer to the prayers of a huge number of VoIP integrators. Yeah, I know we have similar phones from all the major brands, but most of them (if any at all) don’t support IAX2. Do I need to mention that the majority of them also carry a much higher price-tag, and I really don’t know too many people who don’t like saving a couple of $$. It comes with or without the sidecar making it a great choice for both the receptionist as well as the office desk.

CORPORATE

They are already in stock at our DC in Breda, The Netherlands. Job has just replaced his trusty old SNOM phone with the SOHO model, and so far I have got nothing but positive feedback from him.

IAX2 Phone, Receptionist Phone or Console, SIP and IAX2 Phone, VoIP Phones

The “Danger” of using Asterisk powered PBX

November 2nd, 2009

We have been using different types of Asterisk based IP PBX’s for almost three years now, and with the exceptions of problems created by our ISP, VoIP provider or our own staff it has been smooth sailing. But just over a week ago I was really %¤##”& off; I was invited to attend a conference with the CEO of a reputable Canadian company and their UK based EMEA manager. Everything was prepared, I dialed the UK number to attend the conference and got busy line! Tried again, same result. OK, breath slowly, I had a US number for the same conference. BUSY LINE! Try again, same result, what the ¤%#& was going on?? A couple of emails later the conference was rescheduled for next day, hopefully the problems with the conference server would be solved by then.

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.

Knowledge, Open Source; Good Business or Dangerous Adventure?, PBX stuff, VoIP Providers supporting Asterisk

While waiting for Skype and Digium…

October 19th, 2009

Nick from the UK is one of our best and most active contributors both through our Forum as well as directly to me, and a few hours ago he sent me a mail that highlights an alternative to both SIP for Skype and Skype for Asterisk. Nick has spent the weekend testing out the solution and the conclusion is in:  It really works both ways! This solution is not for everybody since it requires a PC running Windows in addition to the IPxx PBX, but for the majority of companies and IT professionals this is not a major problem. Here is the letter from Nick:

Jan
I have managed to get NCH software’s skype2sip software working with the IP04. It works in both directions (dials and recieves) and required setting up a sip trunk, editing its script in the file manager and using some interesting settings on the skype2sip software. Down side of course is it has to run on a separate windows PC, not a problem if you are running a pro network and have a domain controller or other form of server to control the network. When I can work out a way of putting into writing the settings etc. I will forward them to you to play with. Tests at the weekend worked well, and it may be a temporary fix for those desperate for integration, until Digium get their act together and provide code that will run directly on the Ip04.
regards
Nick

Thanks for sharing this with us, when you have completed the instructions about how to make it work I will make sure that it is integrated in our online Admin Manual as well as posted here ASAP.
And last but certainly not least, thanks to NCHfor their great work! ;o)

UPDATE: Just received the step by step guide from Nick, please check it out here!

Knowledge, Skype

The Countdown Has Completed! WE ARE ONLINE AND OPEN!!!

October 16th, 2009

We have talked about it, planned and looked forward to this day for so long! Now it is FINALLY ready, and it is a great pleasure to thank you for your patience and wish you a heartfelt

WELCOME!!!

http://voiptel-pro.com

UPDATE!!!

We were planning several promo’s when we opened our new webshop, but unfortunately we got some problems with availability of stocks. I am happy to announce that new stocks will arrive within a week or so, most items has had their prices reduced with promo’s and we will introduce several new and very exiting “toys” within the next two weeks! ;o)

P1010267

Knowledge

IPxx and Skype – When??? UPDATE!

October 9th, 2009

We have received an unofficial feedback from a Digium representative regarding this matter. It was made clear that the following info was not necessarily that of the Board of Digium, just the personal opinion and understanding of the person speaking to us.

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!

Knowledge, Open Source; Good Business or Dangerous Adventure?, PBX stuff

IPxx and Skype – When???

October 4th, 2009

Skype for SIPIt has been five months since my last post about Skype for SIP, and I know that I speak for a huge number of people when I say that my patience is starting to run thin. Specially when you find out that Cisco and Shoretel have implemented it in their proprietary, very expensive products!

And what about Skype for Asterisk made available last month? Sorry, only available if you run anything else than anSkype for Asterisk 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!

Skype2

Knowledge, Open Source; Good Business or Dangerous Adventure?, PBX stuff, The Past and the Present

History of Zapata Telephony and how it relates to Asterisk PBX.

September 26th, 2009

With the exception of a limited group of hardcore Asterisk dudes the vast majority of us have either none or very limited knowledge about Jim Dixon, the creator of Zapata Telephony. Without him it would not have been possible to connect Asterisk based devices to the PSTN, and in good old spirit of Open Source he has made his work available to the public for free! So I have decided to republish an article written by Jim himself in the hope that more people will learn about the huge contributions of this man!

By Jim Dixon, WB6NIL

About 20-25 or so years ago, AT&T started offering an API (well, one to an extent, at least) allowing users to customize functionality of their Audix voicemail/attendant system which ran on an AT&T 3BX usually 3B10) Unix platform. This system cost thousands of dollars a port, and had very limited functionality.

In an attempt to make things more possible and attractive (especially to those who didnt have an AT&T PBX or Central Office switch to hook Audix up to) a couple of manufacturers came out with a card that you could put in your PC, which ran under MS-DOS, and answered one single POTS line (loopstart FXO only). These were rather low quality, compared with today’s standards (not to mention the horrendously pessimal environment in which they had to run), and still cost upwards of $1000 each. Most of these cards ended up being really bad sounding and flaky personal answering machines.

In 1985 or so, a couple of companies came out with pretty-much decent 4 port cards, that cost about $1000 each (wow, brought the cost down to $250 per port!). They worked MUCH more reliably then their single port predecessors, and actually sounded pretty decent, and you could actually put 6 or 8 of them in a fast 286 machine, so a 32 port system was easy to attain. As a result the age of practical Computer Telephony had begun.

As a consultant, I have been working heavily in the area of Computer Telephony ever since it existed. I very quickly became extremely well- versed in the hardware, software and system design aspects of it. This was not difficult, since I already had years of experience in non-computer based telephony.

After seeing my customers (who deployed the systems that I designed, in VERY big ways) spending literally millions of dollars every year (just one of my customers alone would spend over $1M/year alone, not to mention several others that came close) on high density Computer Telecom hardware.

It really tore me apart to see these people spending $5000 or $10000 for a board that cost some manufacturer a few hundred dollars to make. And furthermore, the software and drivers would never work 100% properly. I think one of the many reasons that I got a lot of work in this area, was that I knew all the ways in which the stuff was broken, and knew how to work around it (or not).

In any case, the cards had to be at least somewhat expensive, because they had to contain a reasonable amount of processing power (and not just conventional processing, DSP functionality was necessary), because the PC’s to which they were attached just didnt have much processing power at that time.

Very early on, I knew that someday in some “perfect” future out there over the horizon, it would be commonplace for computers to handle all of the necessary processing functionality internally, making the necessary external hardware to connect up to telecom interfaces VERY inexpensive and in some cases trivial.

Accordingly, I always sort of kept a corner of an eye out for what the “Put on your seatbelt, you’ve never seen one this
fast before” processor throughput was becoming over time, and in about the 486-66 DX2 era, it looked like things were pretty much progressing at a sort of fixed exponential rate. I knew, especially after the Pentium processors came out, that the time for internalization of Computer Telephony was going to be soon, so I kept a much more watchful eye out.

I figured that if I was looking for this out there, there *must* be others thinking the same thing, and doing something about it. I looked, and searched and waited, and along about the time of the PentiumIII-1000 (100 MHz Bus) I finally said, “gosh these processors CLEARLY have to be able to handle this”.

But to my dismay, no one had done anything about this. What I hadn’t realized was that my vision was 100% right on, I just didn’t know that *I* was going to be one that implemented it.

In order to prove my initial concept I dug out an old Mitel MB89000C “ISDN Express Development” card (an ISA card that had more or less one-of-everything telecom on it for the purpose of designing with their telecom hardware) which contained a couple of T-1 interfaces and a cross-point matrix (Timeslot- Interchanger). This would give me physical access from the PC’s ISA bus to the data on the T-1 timeslots (albeit not efficiently, as it was in 8 bit I/O and the TSI chip required MUCHO wait states for access).

I wrote a driver for the kludge card (I had to make a couple of mods to it) for FreeBSD (which was my OS of choice at the time), and determined that I could actually reliably get 6 channels of I/O from the card. But, more importantly, the 6 channels of user-space processing (buffer movement, DTMF decoding, etc), barely took any CPU time at all, thoroughly proving that the 600MHZ PIII I had at the time could probably process 50-75 ports if the BUS I/O didn’t take too much of it.

As a result of the success (the ‘mie’ driver as I called it) I went out and got stuff to wire wrap a new ISA card design that made efficient use of (as it turns out all of) the ISA bus in 16 bit mode with no wait states. I was successful in getting 2 entire T-1’s (48 channels) of data transferred over the bus, and the PC was able to handle it without any problems.

So I had ISA cards made, and offered them for sale (I sold about 50 of them) and put the full design (including board photo plot files) on the Net for public consumption.

Since this concept was so revolutionary, and was certain to make a lot of waves in the industry, I decided on the Mexican revolutionary motif, and named the technology and organization after the famous Mexican revolutionary Emiliano Zapata. I decided to call the card the “tormenta” which, in Spanish, means “storm”, but contextually is usually used to imply a “*BIG* storm”, like a hurricane or such.

That’s how Zapata Telephony started.

I wrote a complete driver for the Tormenta ISA card for *BSD, and put it out on the Net. The response I got, with little exception was “well that’s great for BSD, but what do you have for Linux?”

Personally, Id never even seen Linux run before. But, I can take a hint, so I went down to the local store (Fry’s in Woodland Hills) and bought a copy of RedHat Linux 6.0 off the shelf (I think 7.0 had JUST been released but was not
available on shelf yet). I loaded it into a PC, (including full development stuff including Kernel sources). I poked around in the driver sources until I found a VERY simple driver that had all the basics, entry points, interfaces, etc (I used the Video Spigot driver for the most part), and used it to show me how to format (well at least to be functional) a minimal Linux driver. So, I ported the BSD driver over to Linux (actually wasnt *that* difficult, since most of the general concepts are roughly the same). It didn’t have support for loadable kernel modules (heck what was that? in BSD 3.X you have to re-compile the Kernel to change configurations. The last system I used with loadable drivers was VAX/VMS.) but it did function (after you re-compiled a kernel with it included). Since my whole entire experience with Linux consisted of installation and writing a kernel module, I *knew* that it *had* to be just wrong, wrong, wrong, full of bad, obnoxious, things, faux pauses, and things that would curl even a happy Penguin’s nose hairs.

With this in mind, I announced/released it on the Net, with the full knowledge that some Linux Kernel dude would come along, laugh, then barf, then laugh again, then take pity on me and offer to re-format it into “proper Linuxness”.

Within 48 hours of its posting I got an email from some dude in Alabama (Mark Spencer), who offered to do exactly that. Not only that he said that he had something that would be perfect for this whole thing (Asterisk).

At the time, Asterisk was a functional concept, but had no real way of becoming a practical useful thing, since it didn’t, at that time, have a concept of being able to talk directly (or very well indirectly for that matter, being that there wasn’t much, if any, in the way of practical VOIP hardware available) to any Telecom hardware (phones, lines, etc). Its marriage with the Zapata Telephony system concept and hardware/driver/ library design and interface allowed it to grow to be a real switch, that could talk to real telephones, lines, etc.

Additionally Mark has nothing short of brilliant insight into VOIP, networking, system internals, etc., and at the beginning of all this had a great interest in Telephones and Telephony. But he had limited experience in Telephone systems, and how they work, particularly in the area of telecom hardware interfaces. From the beginning I was and always have been there, to help him in these areas, both providing information, and implementing code in both the drivers and the switch for various things related to this. We, and now more recently others have made a good team (heck I ask him stuff about kernels, VOIP, and other really esoteric Linux stuff all the time), working for the common goal of bringing the ultimate in Telecom technology to the public at a realistic and affordable price.

Since the ISA card, I designed the “Tormenta 2 PCI Quad T1/E1″ card, which Mark marketed as the Digium T400P and E400P, and now Varion is marketing as the V400P (both T1 and E1). All of the design files (including photo plot files) are available on the Zapatatelephony.org website for public consumption.

We have more, higher-density designs on the way.

As anyone can see, with Mark’s dedicated work (and a lot of Mine and other people’s) on the Zaptel drivers and the Asterisk software, the technologies have come a long, long way, and continue to grow and improve every day.

Footnote:
Has anyone ever taken a moment to sit back and consider the ENORMOUS responsibility that Mark has taken upon himself by doing this project? Have you ever thought of how incredibly many things that he has to concern himself with, and that it just *NEVER ENDS*! At this point, I believe that I have worked with him on this project longer that just about anyone, including some of his employees, and believe me, I have a good vantage point to see at least some of the stuff that he has to go through to accomplish this.

Personally, I would have *NEVER* taken on such a task, being that I am and was quite aware of the level of responsibility required to do so.

Yes, the task that I took on was and is quite a task, and quite a responsibility, but I did what I knew I could accomplish. Mark’s part is way larger then mine, and all I can say that I know what it takes for him to do what he is doing, and I seriously appreciate the time and dedication that he has put into all the incredibly wonderful things that he has done for it and all of us.

Furthermore, Id like to seriously thank all of the project contributors and everyone else that has done some part to help with this project. Thank you for demonstrating that you believe in it, and that you believe in us.
This article has been published in my blog according to the guidelines of this site (original story).

Last but certainly not least, have you ever thought about the possibility of interconnecting the global network of radio amateurs with Asterisk? Just that happens to be Jim’s pet project, you’ll find all about it here.

Knowledge, The Past and the Present

WordPress SEO fine-tune by Meta SEO Pack from Poradnik Webmastera