This was definetely not what I was planning to write about! We have had a lot of things happening lately, but they all have to wait in order to make room for this article that I found through Twitter!

Just when you thought it couldn’t get any worse in the world of software patents, a reliable source sent me this response from Steve Jobs about a patent pool that’s forming and aiming to nail the open source codecs projects. It’s enough to make the weak at heart go weak in the knees and stop working on open source projects.

Here is Steve Jobs’ response to the letter from Hugo Roy:

From: Steve Jobs
To: Hugo Roy
Subject: Re: Open letter to Steve Jobs: Thoughts on Flash
Date: Fri, 30 Apr 2010 06:21:17 -0700
All video codecs are covered by patents. A patent pool is being assembled to go after Theora and other “open source” codecs now. Unfortunately, just because something is open source, it doesn’t mean or guarantee that it doesn’t infringe on others patents. An open standard is different from being royalty free or open source.

Sent from my iPad

You can read the story here!

Original letter sent to Steve Jobs

Knowledge, Licensing, Open Source; Good Business or Dangerous Adventure?

Have you ever been looking for a VoIP provider that support IAX2? Or a provider that will have complete configurations for PBX’s manufactured by Atcom, Positron, PIKA WARP or Tinyworx in their firmware? With better rates for “Pay-as-you-go” services than most companies offer for a monthly fee?

We are currently testing the services of what looks like a very promising wholesale provider. I am expecting the test period will be finished within the next few weeks after which we’ll sign up and invite you all to join!

The Basics

You connect any kind of device; server, switch, or software that supports one of the 2 protocols we offer (SIP, IAX2) and at least one of the codec’s offered to our server. You can authenticate by IP address or dynamic registration, which is supported by almost every piece of VoIP software and hardware on the globe. You can configure multiple devices with different usernames by using our sub account section. When you place calls, we terminate them for you. You can also order DID numbers. When people call these numbers, we send them to you via VoIP.

Before we make our services publicly available we will need a number of people to test it out from different parts of the world. If you would like to have one of those test accounts please send a mail with some informations about where you live, what VoIP service you use today and why we should give you one of our test accounts. Please send the email to
post@voiptel-pro.com, write “TEST ACCOUNT” in the subject field.

Knowledge, Press Release, VoIP Providers supporting Asterisk

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 to 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!

9/11 was a tragedy, just like the ongoing slaughter of civilians in countries like Iraq and Afghanistan are local tragedies. But there are a couple of differences; like the number of casualties and the global effect on these terrible situations. 9/11 was confined to a small area and ended the lives of 3000+ lives; the situation in Iraq and Afghanistan is ongoing throughout these countries and the death toll is well beyond 100 000. But the most significant effect on the global population is caused by 9/11; it gave the authorities the best possible excuse to impose all kinds of restrictions and surveillance on their population. Maybe we should read George Orwell one more time, maybe we should tell the politicians to slow down, or maybe we have to implement our own systems that not only protect us from criminals but even from our own government? Not to commit a crime but simply to prevent that the authorities jump on the wrong conclusion based on a private conversation where certain key words was used, words that was picked up by the automatic listening systems installed by the local Intelligence Agency at every ISP throughout your country.

Knowledge, Uncategorized

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 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

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.

Some 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

Knowledge, Skype

Just 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

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

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 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.

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

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