Human rights; do we still have any???

March 6th, 2010

Some time ago I spent a couple of days in London together with Adelina and a few good friends. Despite the fact that London is huge and most people seems to be very busy I was still impressed by the care and hospitality of the average Londoner. There was only one small detail I wasn’t quite happy with. Coming from a small country (Norway) where you can go unnoticed almost anywhere I was stunned when I discovered all the CCD cameras. As a matter of fact, you will have a hard time finding any part of central London where you are not a “movie star”! An internal Metropolitan Police report was released in August 2009 that admitted less than 1 crime was solved per year for every 1000 CCTV cameras in London. This comes as a major blow to the UK police who spent £500 million between 1996 and 2006 installing 4 million cameras nationwide, with 1 million in London alone. Despite claims that each citizen might be seen on 300 cameras a day, perhaps half of all CCTV camera footage is unsuitable to convict criminals in court. The British public is crying foul, the police force is scrambling to access the problem, and everyone is watching to see what the worlds most recorded country is going to do next.

Some time ago I wrote about Police set to step up hacking of home PCs where the average British police officer can brake into your PC if he is suspecting that you might be doing something unlawful. He doesn’t need a court order, not even the approval of his superiors.

A couple of new laws has been passed lately, laws that force your Internet provider to monitor all traffic and keep the records for two years. They claim it is for our safety, yours and mine, to protect us from all kinds of terrible threats.

Have you ever been surfing the net utilizing the free services provided by your local coffee store or gasoline station or hotel or any other place where some friendly person want to give you free access to the rest of the world? Or maybe you stumbled over a private WiFi connection where the owner didn’t bother to secure it and thereby granted you free access? According to UK legislators this practice has be banned and the people providing free access to the Internet should be treated as criminals. According to the same legislators the unprotected WiFi hot-spots are being used by copyright pirates and therefore has to be banned! Don’t you just love those legislators who have nothing better to do than to deprive the rest of us from free access to the net?

The Norwegian police will be given a few billion kroner (local currency, 8 kroner= 1€) as funding for a new computer system. Nothing strange about that, even the cops has to be upgraded from time to time, right? Well, not quite. This new system is not only improving the capabilities of the Norwegian police to look for criminals, it actually takes every incident where the police is involved or informed about and makes pictures and information available to every police officer in Europe! And the Norwegian police have access to the same kind of information from their colleagues throughout the European continent! According to the Norwegian Ministry of Justice the civil rights of the individual person will be better protected with this new system!?! And maybe, just maybe they will be able to capture a criminal from time to time, just in time to defend their Orwellian system!

Knowledge, The Past and the Present

PRESS RELEASE; New GSM PBX series

February 12th, 2010

For some time now we have been eagerly waiting for a new line of Atcom PBX’s that can handle one or more GSM modules. The wait is finally over, the first few units has already come of the assembly line and will be available through our web shop in the very near future. There are currently 3 models available, all based on the same motherboard and steel chassis; IP-4G (4 GSM), IP-2G4A (2 GSM 4 Analogue) and IP-2G4B (2 GSM 4 BRI). Another model with 4 Analogue and 4 BRI (IP-4A4B) will be available in the near future.


Prototype PBX


GSM module


BRI module

UPDATE: Just got a call from Ruben, he has shared his thoughts about these PBX’s here. As always, it’s well worth reading!

Asterisk compatible GSM equipment, Knowledge, Press Release

We are growing fast, very fast!

January 23rd, 2010

PhoenixAnd for once I am not talking about the size around my waist! The last couple of months I have been working on a new concept, a network of Partners all over the world. As a small company you meet all kinds of problems such as high prices, costly freight, none or very limited influence on the products that you are marketing. While the large corporations have the resources to develop fancy equipment carrying their name and logo the small companies is left with the rest. Signing contract with Alain and PeterBut guess what, we are about to change that! With the support of a lean organization handling both ordering as well as shipment to the various Partners we are able to cut cost and grow influence. We still need more members in order to have a real impact, at the moment we have Partners in Norway, the Netherlands, United Kingdom, France and USA (east coast). And another four companies is going to join within a week or so. I believe there is strength in numbers, and we will use our strength to the benefit of our customers.

UPDATE: We have decided to participate at AstriEurop together with our  French Partner Analytel. For details please click here.

For inquiries about our global network of Retailers and Partners please go here.

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

Asterisk Embedded Mini-Appliance Roundup; Atcom’s IP-xx

January 9th, 2010

2009 was a tough year, every time I turned on the TV or sat down with a newspaper there was new disasters, bankruptcy’s, rising unemployment rates, the works. Most companies scaled back, research and development of new products was put on hold. But not every company followed the trend and put their heads in the sand hoping the international crisis would go away. I know of at least three companies that did the opposite, that invested time and money in the future of communication the Asterisk Way. In some ways their products are very similar, yet they are so different that they are the champions of different categories altogether. I got a sample of each to play with, and I’ll try to share my experience.

First one out is my old friends at Atcom Technology, the manufacturers of David Rowe’s open source IP-04 PBX. IP04Who could have known that David’s basement project would become so popular that it changed the entire price structure for large (and obsolete) PBX manufacturers and finally put the PBX within reach of even the smallest companies. From an average price of $5000 or more you can now have the same features packed in the size of an ATA for $135! ip01_1The IP-04 is no longer alone, it has grown to an entire family of PBX’s with no end in sight. And they are GREEN, with an average consumption of less than 5W it can even been run from solar power! The only negative sides of this incredible device is the fact that it has taken us way too long to make the firmware stable enough for professional use in addition to the fact that the person given the task of setting it up should have some experience with Asterisk. Or get access to direct assistance through an SE Support Contract.  But for the seasoned professional it is a dream, you can even develop your own version of the firmware.

As I mentioned earlier Atcom has developed an entire family of PBX’s sharing the same basic design and hardware with the following specifications:

Interface
1 or 2 RJ45 port(s)
1 x Power port
1 x RS232 port
1 to 8  RJ11 port (FXS/FXO interchangeable)  or 4 RJ45 (IP-BRI)
1 to 4 FXO/FXS module slot (Except IP-BRI)
1 slot for SD FLASH  (not IP-01 and IP-02)

Hardware
CPU: 400MHz Blackfin BF532 Chip  (IP-BRI use a 600Mhz BF537 chip)
NAND flash 256 M
SDRAM 64M

System
Open Source uClinux with Asterisk firmware, VoIPtel CE by default  ( Optional: VoIPtel SE, BAPs or Astfin)

Size
100 * 100 * 28mm    IP-01 and IP-02
225 * 120 * 30mm    IP-04, IP-08 and IP-BRI

The graphical user interface (GUI) is a modified version of AsteriskNOW GUI v2, so users of AsteriskNOW will feel right at home. It is unfortunately very common that open source projects often has either none or very poor user documentation, but this has been addressed in a rather unique way. After logging in you will be presented with a similar screen to the one below, and in the upper right corner there is a question-mark (?). Click on it and an Administrators Manual created on FLASH paper will be downloaded and displayed separately from an on-line server. This way of presenting the Manual has several benefits; it is easy to update it for the developers, it is always available in the latest version and it is very environmentally sound.

GUI

Assisted by the Manual it’s not too difficult to get the PBX online, but I miss a Quick-start guide outlining how to get it operational and in what order the steps should be performed. I better get my act together and write it.

There is currently no auto-provisioning system for VoIP phones in this firmware (will come soon), so I have to configure the phones manually. I have five phones available in my lab; one analogue Siemens Gigaset, two Atcom phones (AT-610P and AT-620P) and two VoIPtel Pro phones (Executive and SOHO). The analogue phone is plugged into the FXS port while the four other phones takes 3-5 minutes each to configure through their GUI.

The sound quality is very good, but there is a difference between the analogue Siemens phone compared to the VoIP phones. The VoIP phones has a clear edge when it comes to voice quality, and they are not experiencing any problems with echo on the line. The Siemens phone has a very good voice quality considering that it is an analogue device, but I have experienced problems with echo a few times. Unfortunately this is not the kind of problems that can be solved by the software developers, it has to be tuned by the owner of the PBX. But this is not a serious problem, for the large majority the OSLEC does a very good job.

I recommend this line of PBX’s for people and integrators with very limited budgets and who has the required skills or available time to learn. Even though it has a very good GUI accompanied by tool-tips and an on-line Administrators Manual it still require some degree of experience and skills.

The prices for the IPxx start at US$135.00 for an IP-01 on special offer to the IP-BRI clocking in at $500.00 For more info about prices and availability please go here.

Next: Meet PIKA WARP running FreePBX

IP BRI, IP01, IP02, IP04, IP08, Knowledge, PBX stuff

PRESS RELEASE: Problem with latest batch Atcom AX-400P cards

December 11th, 2009

We just received information from Atcom that the latest batch of their AX-400P cards has a problem that might cause a short circuit. This is an excerpt from the letter:

Today, we found a serious issue with the AX-400P new design we shipped you last time, the issue relates to potential short circuit in the screw on the bottom of the board.

When the short circuit happen, the board may be damaged, you can see there is a broken black line in the circuit of the PCB. The card will still be able to use in this case, but there is potential danger since the circuit exposures.

If you have purchased this type of card recently you should contact your supplier and ask how to find out if your card is affected as well as their policy relating to this problem.

Knowledge, Press Release

Cool Skype for SIP Upgrades, Enhanced Inbound Routing and more

December 9th, 2009

Go here for original story.

It has been a long wait for the majority of us, but finally it looks like Skype has decided to shower us with new features! May it just keep coming, we’ll never get enough when it comes to new ways of integrating communication platforms.

Skype for SIPSome important news for Skype for SIP beta “open” users I thought I’d share. Just last week Skype announced going from closed “beta” to “open” to the public, and now today they’re adding some cool new Skype for SIP features.

Once the upgrades are complete there will be cool new features, such as direct inward dialing (DID) calling over the Skype network to a specific IP-PBX extension. Similarly, if someone calls your Skype business account name, i.e. tkeatingtmc, Skype will prefix the extension number into the SIP header so that the call will be routed directly to that person’s extension. Sweet!

The upgrade will also add IP authentication, which is critical to businesses that do not support registrations. According to Skype, “For SIP enabled PBX’s and SIP gateways that do not support registrations, you can now change your authentication settings in your SIP profile to a fixed IP address/port. Based on your IP address we will work out the best settings for your SIP enabled PBX when using Skype for SIP.”

Today, Skype’s Kevin Holmes Senior Product manager, Skype for Business announced the upgrades, outage, etc. Check it out:

We are please to announce three new features for Skype for SIP open beta. We will start upgrading from today and expect this to last one week. These new features are designed to enhance the way you can integrate Skype for SIP with your SIP enabled PBX.

Upgrades are due to start at 1pm GMT on Wednesday 9th Dec until Wed 16th Dec 2009.

Skype for SIP new features include:

Inbound from Online numbers
Inbound calls will now include the Online number, which means you can now set up rules in your SIP enabled PBX to automatically route incoming calls from your Online numbers to extensions and other resources such as auto attendant or IVR systems.

Extension tagging for Business Account Skype names attached to your SIP profile. You will be able to add a Business account Skype name’s target extension, so when Skype users call your Business account Skype name, Skype for SIP will prefix the extension number in the SIP message for calls from Skype users to your SIP enabled PBX.

IP authentication. For SIP enabled PBX’s and SIP gateways that do not support registrations, you can now change your authentication settings in your SIP profile to a fixed IP address/port. Based on your IP address we will work out the best settings for your SIP enabled PBX when using Skype for SIP.

We will start upgrading Skype for SIP open beta from Wednesday 9th Dec 2009 starting at 1pm GMT, the upgrade is likely to last one week.

We will be working with our Skype for SIP certified PBX vendors to update their configuration guides for setting up Skype for SIP.

Important notes: This upgrade will not affect:

* SIP Registrations
* Outbound calls to landlines and mobile numbers

Recommendations to avoid inbound call failures
We recommend that you add an additional route as soon as possible for inbound calls from online numbers so that your SIP enabled PBX can accept and route based on:

• SIP user name (current configuration)
• Online numbers (international format)

Technical details
Please note the following changes to Inbound Invite messages from Skype: (no other features or use cases are affected)

Inbound calls from Online numbers to your SIP profile
We have changed the SIP format which now includes the Online number that was dialled via the PSTN or Skype user in the To header. With this SIP message change, you can now assign up to 99 Online numbers to your SIP profile and be able to distinguish which online was called.

An example of the change in SIP headers:

R-URI: <SIP_username> 99051000000000@pbx_ip – taken from your SIP profile
To: sip:<onlinenumber>@sip.skype.com
From: sip:<callers_cli>@sip.skype.com

The remainder of the SIP format message remains unchanged.

Extension tagging for calls to Business Account Skype names
Inbound calls from Skype users to your SIP profile (from the 10th Dec 2009) You can now add a target DDI or extension number to Business Account Skype names that are attached to your SIP profile. The reason we have done this to enable SIP enabled PBX’s to route inbound calls from Skype to PBX extensions or resources such as an Auto attendant or IVR. You will find a numeric text box for this feature in the BCP –> SIP profile –> calling tab next to the Business Account Skype name. Up to 11 characters are allowed in this field.

Extension numbers will be shown in the Invites request URI for example:

R-URI: <extension_number>@pbx_ip
To: <SIP username> 99051000000000@sip.skype.com
From: caller_id@sip.skype.com

Further information can be found from Wed 10th Dec in the Skype Business support section at the following address:

http://forum.skype.com/index.php?showtopic=486951

http://forum.skype.com/index.php?showforum=252

We hope you enjoy the new features in Skype for SIP open beta
Skype product management smile.png

Knowledge, Skype

Skype for SIP beta program opens up to all businesses!

December 3rd, 2009

Skype for SIPJust got the news, it’s finally OPEN! After months of testing  Skype for Sip is finally available to everybody with a business account! Does it really work? Is it worth the wait? And the price? We’ll definitely find out, we’ll try to get it configured and operational as soon as possible! ;o)

In the meantime I found some interesting info  on this blog:

Skype for SIP connects your company’s phone switch to Skype. SkypeIn and Skype-to-Skype calls come in to your phone system, outbound calls can go over SkypeOut. VoIP people call a connection between your phone system and a phone company a “trunk.” Some people call Skype for SIP “Skype trunking.” SFS is a limited add-on. No emergency dialing: You still need a regular phone service to dial police, fire, ambulance. No phone number portability. You need a Skype Online Number and you don?t get to use an existing number. Service levels aren?t regulated by local or government authorities or guaranteed by Skype. SFS isn?t free. US$ 6.95 per month for each channel, one call at a time per channel. You have to rent an Online Skype phone number for your business. You pay for SkypeOut at published rates. At the moment, the Skype Global Rate is 2.1¢/minute in more than 36 countries. You’ll pay more for mobiles in most places. Unlike SkypeOut for consumers, Skype doesn’t allow or offer flat-rate calling plans. Calls coming in to your phone from the Skype ecosystem are free. Many smart phone systems let you write rules for routing outbound calls. You might choose SkypeOut for international calls or if you haven’t the buying power to negotiate discounts with your phone company. Skype is building a distribution channel. They’ve partnered with PBX makers like ShoreTel, Cisco, and SIPfoundry. Together they have thousands of value added resellers (VARs) who serve local businesses. Those resellers will be eligible to earn affiliate referral commissions from Skype, although a separate program for VARs is not in place. Skype is talking with more PBX makers to make adding a Skype channel a built-in menu option. Skype for SIP is an indirect sales effort. SFS partners with PBX makers, their VARs, to reach IT and telecom departments responsible for configuring telephone systems and buying telephone services. So Skype gets to know your Phone Guy. This gives Skype a beachhead in your company, a relationship to sell more Skype products, and a champion for Skype technology.

Phil Wolff at Skype Journal talked Monday with Skype’s Matthew Jordan about the latest update. Here are the details.

And for those who want to try it out you’ll find more info and links here.

Have fun! ;o)

UPDATE: We made an attempt to make it work a few hours ago, and the first attempt disconnected all our trunks and SIP phones! Since this was on our production PBX (yes, we like to live dangerously) we removed it rather quickly. Our second attempt was on a test PBX at a different location (we like to learn the hard way), no crashes and it registered both ordinary SIP and Skype for SIP trunks. And that was all it did, no calls was able to get through. Unfortunately we were running out of time, but we will continue our little experiment as soon as we have a few minutes available. ;o)

Just found this info, could it be this simple???
The clues are in the documentation; SkypeIn and SkypeOut use G.729 for nearly all calls, so handling calls via those paths requires a G.729 transcoder on the system if the target of the call will not also be using G.729. This is why the Skype For Asterisk license includes licenses for Digium’s G.729 software transcoder as well.

It works! At least when we call out, just have to get a SkypeIn phone number for inbound calls and then we are home free (I hope)! ;o)

Knowledge, Skype

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

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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

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