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February 28, 2007

O’Reilly E-Tel Conference Update

Note:  Found this little update on Todd Pinkerton's blog about some of the projects he has come across at the conference.

"this week I’m in San Francisco at the O’reilly Emerging Telephony conference, chaired by my good friend Surj Patel. The conference is all about how the old-school world of telephones and phone networks are now merging with the Internet. So now, anyone who can write a web application can create a voice service.  This is part of the inspiration behind Ringfo." 

Some cool stuff so far : Adhearsion, a ruby on rails interface to the Asterisk PBX. My former Colleague from Orange Sunil Vemuri has a company working on recording and searching all your phone conversations; and  Chris Sacca gave an inspiring talk about US broadband penetration and Google’s muni wifi project in Mountain View.

More details when I return…

Source: Todd Pinkerton Blog 

Columbia University Law School Professor's Paper Sparks Wireless Net Neutrality Debate

Note:  It looks like this is heating up with now Skype and a Columbia Law Professor getting into the fray.   I have taken the liberty on linking too the paper in discussion for anyone to read.

A paper published by Columbia University Law School Professor Tim Wu claims that wireless networks don’t play by the same rules that wired networks do and limit consumer choice. Skype, for one, agreed with him and petitioned the FCC to mandate that wireless network operators open their networks to more devices and applications. The CTIA fired back.

Wu stated that the FCC’s Carterfone rules “continue to affect innovation and the development of new devices and applications for wireless networks.” His comments elicited a large response from the industry and refocused the net neutrality discussion, this time on the wireless networks.

Wu went on to argue that the carriers exert too much control over the design of mobile equipment and said, "They have used that power to force equipment developers to omit or cripple many consumer-friendly features.”

His words spurred Skype to action. Its filing with the FCC looks to force open the wireless networks and allow applications such as its own to be used on the networks. CTIA President Steve Largent called the move self-interested. "Skype's filing contains glaring legal flaws and a complete disregard for the vast consumer benefits provided by the competitive marketplace," he said in a statement. “Skype's 'recommendations' will freeze the innovation and choice hundreds of millions of consumers enjoy today. The call for imposing monopoly-era Carterfone rules to today's vibrant market is unmistakably the wrong number."

The net neutrality issue has long been focused on just wired providers of Internet services, such as cable and DSL broadband. The new thinking would have the wireless networks treat all content equally, too.

House Democrats plan to look more closely at several FCC policies regarding telecommunications.

Source: Wireless Week 

February 27, 2007

Surging IP Telephony Adoption Drives Strong Growth of Enterprise Phone Systems

Note: Maybe Asterisk will make the list next year. 
 
According to new research from industry analyst firm IntelliCom Analytics, North American shipments of Call Control software licenses -- or "seats" -- grew by 9.7% in 2006 over the previous year. This growth in license shipments for the software that enables the features and functionality of Enterprise Communication platforms was driven by broad-based adoption of IP Telephony at a time when traditional TDM systems are increasingly being phased out by vendors.

During 2006, sales of Pure IP systems -- those that are primarily equipped with IP endpoints -- surged 54% year over year, while corresponding TDM system sales declined 30% over the same period.

IntelliCom measures activity in the Enterprise Communication market based on software license seats rather than traditional hardware ports. This shift in measurement metrics follows the previously announced findings from IntelliCom's Market Dashboard research program that documented the rapid evolution of the Enterprise IP Telephony market to a software-centric delivery model.

 

"Our research has demonstrated that Enterprise Communications is clearly undergoing a broad software transformation from both an architectural and business model perspective," explains Frank Stinson, director of IntelliCom's Market Dashboard and Market Performance Dashboard research programs.

IntelliCom also found that three vendors, Nortel, Avaya, and Cisco, accounted for more than half of total 2006 IPT and TDM shipments in North America. "Nortel and Avaya's long-standing battle for North American market leadership raged throughout 2006," adds Stinson. "Whether this continues as a two-way contest in 2007 remains to be seen, given that Cisco has now broken through into the top tier of providers and is moving forward with a very strong growth trajectory."

Source: IntelliCom 

Digium's AsteriskNOW Awarded "Best of Show" at Internet Telephony Conference

 
 
Digium, Inc., the Asterisk company, today announced that AsteriskNOW has been named a winner of a “Best of Show” Award at TMC's INTERNET TELEPHONY Conference and EXPO East 2007.

AsteriskNOW is the software appliance and open source distribution of the popular Asterisk PBX featuring a Digium-designed graphical user interface (GUI). AsteriskNOW includes a new setup wizard that guides users through the installation process, regardless of any previous Linux experience.

The AsteriskNOW distribution includes all the Linux components necessary to install and use Asterisk, simplifying the ease of use. Download options include ISO/CD Image, VM Player image, Xen universal guest image and LiveCD (burn and boot). Beginning March 5, AsteriskNOW will be available with a minibook reference guide.

“AsteriskNOW lets anyone who can use a computer install Asterisk and run it with the confidence of an expert,” said Mark Spencer, CTO of Digium and creator of Asterisk. “AsteriskNOW includes an easy-to-use setup wizard and user interface that allows users to configure their system within minutes, even without any previous Linux experience.”

“Digium and its innovative product AsteriskNow, are a standout indication of why so many enterprise buyers, developers, resellers and service providers flock to INTERNET TELEPHONY® Conference & EXPO,” said TMC President and Conference Chairman, Rich Tehrani. “Digium’s innovation and commitment to quality attracted many serious prospects to their booth. Attendees knew they'd find solutions in the Digium booth that would help them in their businesses today.”

The Best of Show awards are presented to companies unveiling the most impressive new products or new releases at the show. Each winner displayed and demonstrated their product on the INTERNET TELEPHONY Expo show floor.

The next INTERNET TELEPHONY Conference & EXPO will take place September 10-12, 2007

at the Los Angeles Convention Center in Los Angeles, California. For information, visit www.itexpo.com. Or call (203) 852-6800 ext. 146.

Source:  TMC Inc. 

February 26, 2007

Skype Users Have New Feature Available - Voicemail to Text

 
 
Skype users can have their voice mail messages delivered to their e-mail and SMS addresses starting Monday through a voice-mail-to-text service offered by SimulScribe. Utilizing its proprietary voice-recognition algorithms that transcribe voice mail to text, the service works with wireless, networked, and voice-over-IP services, SimulScribe says, adding that the feature typically can save a user three hours a month in wasted time that would be spent listening to voice mail.

"Carriers are already set up to provide" the service, says David Gerzof, SimulScribe's chief marketing officer. "We don't need to do deals with carriers." Gerzof cites a typical application in which a businessperson attending a meeting couldn't answer a phone, but would be able to read incoming calls via e-mail or SMS without interrupting the meeting.

Click Here to Continue Reading 

CERN Uses WiFi Meshing Networking to Solve Universal Problems

Note:  It's nice to see technologies giving each other a helping hand :]

 

The Large Hadron Collider (LHC), part of the massive particle physics lab at CERN, the European Organization for Nuclear Research (and the birthplace of the Internet), will later this year host some of the most audacious scientific experiments ever conceived. And Wi-Fi will play a vital role.

In fact, Wi-Fi already has played a vital role.

The LHC includes a 27-kilometer circular tunnel 50 to 175 meters below the surface near Geneva, Switzerland, plus an underground lab the size of a small village, as well as fantastically complex and delicate detectors -- some as big as small office blocks and weighing hundreds of tons.

 

In the experiments, slated to begin later this year, beams of protons or heavy ions rotating around the LHC tunnel in opposite directions at close to the speed of light will collide, smashing into smaller constituents – hadrons, electrons, muons, photons. The detectors will measure the results.

The object? To recreate conditions a fraction of a second after the Big Bang that scientists believe created the universe billions of years ago, to try and work out what happened -- and also to study high-energy particle interactions and, just maybe, observe new particles and phases of matter, including the basic building blocks of the material world.

This facility is riddled with Wi-Fi networks, an estimated 300 access points in all, most from Proxim Wireless. Not only do the number of APs mean that interference is a major problem, but much of the scientific equipment also interferes. Which is why CERN’s network gurus have turned to Wi-Fi mesh technology for a number of applications, including one mind-boggling maneuver during site construction.

The requirement was to move the huge detector for the ATLAS experiment – the size of a five-story building – from the surface down a 100-meter shaft to the lab. The trick was that the incredibly fragile detector could not tilt or absorb vibration, even a tiny bit. That meant that the crane lowering it had to be controlled very precisely, based on feedback on the detector’s progress down the shaft.

The solution the project team came up with involved a unique Wi-Fi mesh network that would relay data from sensors on the detector to computers that engineers could use to calculate adjustments in crane controls, which would guide the operator.

Using mesh technology was vital for a couple of reasons, says Olaf van der Vossen, the engineer responsible for network infrastructure at CERN. One is the ability -- with dual-radio mesh access points -- to switch between using 2.4GHz and 5.8GHz spectrum to avoid interference, and the ability of the network to dynamically reroute data when the path between two access points becomes “polluted” with interference. Another crucial benefit was ease of setup.

Click Here to Continue Reading the Full Article 

 

How To Configure Asterisk: Your First Installation

Note:  Found this Asterisk How-To article by Mr. NuFone himself: Jeremy McNamara

Deploying your first Asterisk system can be a frustrating task, espcially if you attempt to utilize the more advanced features and functions of Asterisk without having a solid foundation of the basic concepts. This article will walk you through the process of downloading, compiling and configuring your first instance of Asterisk that will allow you to place and receive calls between two Linksys SPA-942s and an asterisk voip provider.

To get started, if you are running a new installation of Linux, you will need to resolve a few external dependencies before attempting to compile Asterisk from source code.

voip:/usr/src# apt-get install build-essential libssl-dev zlib1g-dev
libncurses5-dev

Using Debian one can use apt-get to acquire the necessary packages. Other Liunx distros should use similar package names.

Download the latest 1.4 version of Asterisk from ftp.digium.com.

Note: Asterisk version 1.4.0 does have its issues, which can be show stoppers for larger installations. For this series of How To’s we will focus on Asterisk version 1.4.

Click Here to Continue Reading the Full How-To

 

Stealing Starbucks' WiFi Customers

Note:  I actually just got my free FON wifi router via GigaOm's giveaway

Just because you pay a premium for Starbucks coffee doesn't mean you have to pay a premium to surf the Web at Starbucks cafes. FON, a community WiFi provider headquartered in Madrid, Spain, is offering wireless Internet access to Starbucks' latte-sipping surfers for just $2 a day--versus the $10 users pay to sign onto the 5,100 T-Mobile hotspots at U.S. Starbucks.

Just how does FON plan to steal away Starbucks Internet users? By offering FON wireless routers, also known as "La Foneras," free to anyone who lives above or next to a Starbucks. The routers, which usually cost $40, split an Internet broadband connection into two wireless signals--one for personal Internet use and the second for public use, which can be accessed by anyone within range for $2 per day.
 
The routers' owners get to pocket half of the sign-on fee, and FON takes home the rest. Starbucks refused to comment directly on the FONbucks campaign, but a Starbucks spokesperson said any increase in the number of WiFi hotspots is "a good thing." T-Mobile also declined to comment on the program. The idea behind FON is to build the world's biggest WiFi network from the bottom up by encouraging the world's 300 million broadband customers to buy La Fonera routers and share their wireless access with other FON subscribers.
 
The goal: to have one million global WiFi “hotspots” by 2010 accessible to all so-called Foneros, or members of the FON community. Currently, there are over 300,000 hotspots in Europe, Asia and the U.S.
 
FON was founded in November 2005 by Argentine telecom and new media entrepreneur Martin Varsavsky, whose brainchildren also include Jazztel Telecommunications, now the second-largest publicly traded telecom company in Spain, and the Internet portal Ya.com, which Varsavsky sold to Deutsche Telecom subsidiary T-Online in 2000 for 550 million Euros ($722 million).
 
 

February 25, 2007

VoIM (Voice Over Instant Messaging) not VoIP that Telcos Need to Worry About

VoIP mimics traditional telephony in that it is means of establishing a voice conversation between, generally, two parties across a telephone network: sometimes between PCs and sometimes between telephone handsets. VoIM on the other hand provides the voice communication as one component of a real time communication system based around the PC.
The report - 'Voice Over IM (VoIM) and How it is Changing Traditional Telephony,' from Heavy Reading - argues that this creates a fundamental differentiator. "The Internet-connected PC is now firmly established as a mainstream communication tool, and it is also beginning to transform into the primary tool for real-time communication.
 
"Instant messaging (IM) is the killer application that is helping to make the PC the main medium for the widest range of communication possibilities. By adding voice services to their IM platforms, operators of IM networks are aiming to complete that transformation."

The report argues that "One can hardly separate the voice application from text, video, file sharing, picture sharing, and gaming that is possible via sophisticated IM clients. In fact, that integrated communications element of the service is itself what makes VoIM so different from traditional telephony... Although VoIM can be viewed as a subset of VoIP, in that it delivers packet-based voice to end users, there are fundamental differences between the two. While VoIP is designed as a network service that is based on traditional voice telephony, VoIM is designed as an application that runs over the Internet."

Heavy Reading's associated news service, Light Reading, also quotes the report's author, John Longo saying that: "Carriers will ultimately need to reconcile traditional telephony with VoIM as they face increasing pressure from their customers to receive the same types of flexible services" that VoIM enables."

Longo also claims that VoIM actually sounds better than traditional circuit switched voice because it uses better performing codecs that operate with double the analogue bandwidth of the original voice signal: 16kHz as opposed to 8kHz.
 
Source: IT Wire 

Cisco Allows Apple Use iPhone Name

Cisco Systems and Apple Computer have finally called a truce, and will share the 'iPhone' name in exchange for exploring 'interoperability' between the companies' products in areas such as security, consumer, and business communication. Both Apple and Cisco have said they will dismiss any pending legal actions regarding the 'iPhone' trademark, but continue to remain tight-lipped about future products that might come out of the 'interoperability' deal.

On Jan 10, 2007, just a day after Apple CEO, Steve Jobs, unveiled the company's much awaited iPhone, Cisco slapped a suit against Apple, alleging that the latter's use of the 'iPhone' name constituted a 'willful and malicious' violation of a trademark Cisco has owned since 2000. And, Cisco's Linksys division has been using the 'iPhone' trademark for a series of phones that make free long distance calls over the Internet, using Voice over Internet Protocol (VoIP).
 
In its lawsuit, Cisco said that in an era of 'convergence', the two companies' phones could eventually take on different features and end up competing head-to-head. Cisco said this would result in confusion, mistake, and deception for consumers. Apple dubbed the suit 'silly' initially, arguing that it was entitled to use the 'iPhone' name as the phones belonging to the two companies operate over different networks, and as such, would never compete with each other.
 
Meanwhile, analysts say the settlement will help both companies strengthen their positions in the increasing competition to deliver video and other applications via the network direct to consumers' homes. One network infrastructure analyst with the Yankee Group, Zeus Kerravala, voiced the view that there are ample opportunities for the two companies to dream up collaborative projects to win over consumers. He cited one possibility as the creation of a Linksys device that users could call into to record podcasts, which could then be uploaded onto iTunes automatically.
 
Kerravala said that if the two companies could actually find common ground and work together, the combination would be a formidable one unlike that in which both are continually at loggerheads with each other. Kerravala pointed out that after all, there's no company out there that understands network service like Cisco, and no other company that quite understands user experience like Apple Computer.
 
Source:  TechTree 

Interview With Encryption Advocate Phil Zimmermann Regarding VoIP

Note:  Very well executed interview about privacy, encryption and policy.  Usually I would give my opinion on a subject like this but I need to look deeper into this subject because it really goes to some core beliefs.  It is a very fine balance we must maintain in our wonderful free society.  I see where the government sees that it needs certain tools to help do their job of keeping us safe.  But on the other end this is AMERICA, the land of the free.  At a point I would say we are better off being free and maybe a little less safe and still have our basic privacy.  A free society is only as free as they assert imho. 
 
 
 
Phil Zimmermann has been an advocate of using technology to protect privacy for many years. He created Pretty Good Privacy, an email encryption program, as a tool to protect human rights. He figured that encryption was a way for people in totalitarian countries to escape government spying. He released it for free in 1991, but the U.S. government accused him of violating export control laws, which at the time restricted the use of strong encryption because it could help criminals evade law enforcement.

After the government dropped its case in 1996, Zimmermann founded PGP Inc. Network Appliance bought that company in 1997. In August, 2002, PGP was acquired by PGP Corp., where Zimmermann still works as an advisor and consultant. I spoke with him at the recent RSA conference in San Francisco.

q: Tell us about your history.

a: Most people know me for my work with PGP, or pretty good privacy, which is the world’s most widely used email encryption software. Most readers of the Mercury News who follow encryption software know about PGP. It caused a controversy in the 1990s because the government tried to incarcerate me for releasing it.

q: You released it what year?

a: 1991.

q: You had to fight with the government for how long?

a: For three years. From the beginning of 1993 to the beginnng of 1996.

q: That was the age when releasing strong encryption was against government policy.
They didn’t want to see it exported.

a: Fortunately we’re all past that now.

q: And what have you moved on to do?

a: My latest project is to encrypt voice over the Internet phone calls. I did that ten
years ago with PGP phone. But at that time, the Internet wasn’t ready. Nobody had broadband and there were no VoIP standards. But today, it’s time to do it again. And so the new project is called Zfone. You can read about it at www.philzimmermann.com.

q: Can you spell out the technology and the idea?
  
a: I encrypt VoIP phone calls with a protocol that does not depend on the phone company to help you negotiate the keys. It is something that leaves the phone company out of it. I think people would feel more comfortable with an encryption protocol that leaves the
phone company out.

q: Why is that?

a: I’m sure you’ve heard of the recent controversies about phone companies cooperating
with …

q: The National Security Agency?

a: Exactly.

q: They always leave a backdoor into the technology for the government to do wiretapping?

a: Yes, but I don’t. I’m sort of well known for that.

q: Can you explain how this works then?

a: I negotiate a cryptographic key at the beginning of the VoIP call. I do it without any communication with servers or the signaling that goes through the phone company. I do it entirely between the media packets that flow between the two parties on the call. I negotiate a key using the Diffie-Hellman protocol (named after the inventors), and the two parties can verify that there is no man in the middle listening in. They can compare a short authentication string. You read it aloud and see if it matches. If you don’t bother to take that step, it’s still pretty secure.

q: Is this pretty unique?

a: For VoIP, yes. There are other VoIP encryption protocols. But they usually involve going through servers or the phone company. Or they involve a public key infrastructure which is quite complex and bureaucratic and difficult to manage. In my system, the keys are created at the beginning of the call and destroyed at the end of the call.

q: Why should consumers care about this? They haven’t care about it with regular
phone calls.

a: That’s right. The public phone system was a pretty good system. It is physically
protected. It’s not easy to wiretap. The only people who do wiretap it in most cases are law enforcement. Of course, you could find a few isolated cases where a determined criminal got to some place and listened to calls with alligator clips attached to a line. Those cases were exceedingly rare.

q: It was risky because you could get caught.

a: But VoIP changes all that. It’s very easy for anyone to wiretap. If they were to infect one of your computers with specially designed spyware, they could wiretap all the VoIP phones in your building. Say you have a couple of thousand computers. If one of them got infected with spyware, it could intercept all the VoIP packets that it sees on the network and intercept them. It could store them to a disk. Then the person using the spyware could browse them like you would with Tivo player. You could choose which calls you want to listen to. The spyware could organize them by who is calling who. I’ve seen spyware like this. You can do it from the other side of the world through a web interface. Somebody in another country can control the spyware running on one of the computers in the office and listen to all of the calls from the CEO of your company to a CEO of another company that is an acquisition target. Or they could listen to your in-house counsel talking to outside law firm. Or, let’s look at it from law enforcement’s point of view….

q: From what you describe here, does that require much sophistication on the part of the person using the spyware?

a: Organized crime is attacking the Internet all the time. The Internet is becoming an incredibly hostile place. A few years ago, no one imagined the Internet would become as hostile as it is today. A few years ago, you worried that teen-age boys with black T-shirts and purple hair hacking into your computer and having fun with it. Now it is organized crime  hacking into your computer to do large-scale criminal enterprises like phishing attacks. Organized crime makes a lot of money from the Internet. When VoIP grows large enough to attract their attention, they will begin attacking it as well. They will be able to intercept phone calls. They will get insider trading information. The individuals doing it don’t have to be sophisticated. They aren’t the ones that wrote the software. You might have some people in Russia who write software that they sell to criminals who use it. This also means that organized crime could wiretap prosecutors and judges. They could get the names of informants and witnesses. They could listen to prosecutors and judges talking to their spouses about picking up the kids at school. This could have an enormous impact on the effectiveness of the criminal justice system. 

Click Here to Continue Reading this Interview 

February 24, 2007

Vonage Expected to Enter U.S. Wireless Market by Late 2007

Note:  Maybe now Skype might take a second look at different wireless options (bundle, own product or option X?)

 

Over the past few years, the name Vonage has become synonymous with a single service: inexpensive phone calls over a broadband connection. The company has recently entertained plans of expansion, however, and ultimately hopes to be far more than a pure-play VoIP provider.

The well-known broadband phone company is expected to become a Mobile Virtual Network Operator (MVNO) later this year, meaning that it will resell the wireless voice services of an established wireless carrier, but add some its own additional content to the package.
 
Vonage confirmed in its fourth-quarter earnings call on February 15 that it would begin selling dual-mode cellular and Wi-Fi phones under its own brand name in the second half of this year, but would not give details about any equipment or service contracts.
 
The announcement has caused some analysts to speculate that the company would re-brand itself under the name Vonage Wireless after the new services become available. Vonage has also expressed interest in reselling broadband internet services to its customers, which together with wireless and VoIP, would create a makeshift triple-play bundle of telecommunications services. This in turn would give the broadband phone provider a better chance at competing with established rivals in the cable sector.
 
Source: TeleClick 

February 23, 2007

Alcatel-Lucent joins WiMAX Forum Board of Directors

The WiMAX Forum, an industry-led non-profit organization comprising more than 430 companies committedto promoting and certifying interoperable WiMAX products, today appointed Philippe Goossens of Alcatel-Lucent to its Board of Directors. As a strong contributor to the WiMAX Forum, Alcatel-Lucent’s presence on its board of directors is indicative of the acknowledgement by wireless market leaders around the globe that WiMAX is a powerful option for delivering broadband Internet services anytime and anywhere.


Being elected to WiMAX Forum Board of Directors recognizes Alcatel-Lucent’s commitment and contribution to the fast-growing WiMAX market and a testimony to its technical leadership in WiMAX technologies.

"Alcatel-Lucent is a global leader in telecommunications, and having the company join our Board is evidence of its dedication to taking the WiMAX ecosystem forward,” said Ron Resnick, president and chairman of the WiMAX Forum. “Having one of the industry’s most comprehensive wireless portfolios, we look forward to Alcatel-Lucent’s further involvement to help accelerate the WiMAX deployment worldwide. As a leading proponent of WiMAX, Alcatel-Lucent has proven its ability to deliver standards-based products that are commercially installed by its numerous operator customers.”

Alcatel-Lucent’s strong support of open standards and device interoperability - as demonstrated by several interoperability testing (IOT) centers operational in France, the United States and Taiwan - benefits operators seeking to diversify and offer the optimum choice of terminals to their subscribers.

“Today’s announcement is a major step forward and further highlights our commitment to promote the IEEE 802.16e-2005 open standard to advance the adoption of WiMAX worldwide,” said Philippe Goossens, Strategic Alliances Director for Alcatel-Lucent’s WiMAX activities. “As part of the WiMAX Forum, Alcatel-Lucent is working to ensure interoperability of wireless broadband solutions and enable telecom operators to deliver innovative and differentiating services to their customers.”

Source: Web Wire 

Skype Petitions FCC to Enforce Open Mobile Networks

Note:  Well I am not sure if Skype will be able to defend against the telecoms presumed argument that the Skype traffic will overload their existing infrastructure.  Personally I would like to see the "walled garden" of mobile networks open up a little as longer as the consumers do not see a huge impact on the quality of service they pay for monthly.
 
In a move that could benefit end users greatly, VoIP service provider Skype has petitioned the FCC to apply the 1968 Carterfone decision to wireless phone networks, opening up the possibility of easier use of services similar to Skype on mobile handsets. The Carterfone decision allows customers to attach any device to the phone network, provided it does not harm the network itself, which Skype sees as extending to allow any application to run on any device that can access the network.

Currently, mobile operators limit the kind of data traffic permitted on their phone networks, especially in the case of applications like Skype that can steal revenue from them by allowing cheap VoIP calling. Skype's argument for opening up data networks is that doing so would offer "tremendous new sources of price competition provided by entities such as Skype."

The principal behind the Carterfone decision currently applies to the wired phone network and cable TV networks. Government regulation applying the principal to mobile phone networks would require carriers to allow any application on any compatible handset to be used on their network.

Source: Ars Technica 

Google Apps Set for IP Telephony Integration from Avaya

Internet search leader Google unveiled its latest set of online business tools Thursday, aiming to snare a bigger piece of the enterprise market dominated by Microsoft. Google already has at least one player onboard: IP and telephony firm Avaya.  Avaya, a provider of Internet phone switchboards for businesses, says it plans to link its IP Office product to Google Apps Premiere Edition to provide productivity-enhancing solutions geared toward businesses of all sizes.
"Google Apps offers the potential for us to provide our first-class communications applications Make sense of your IT infrastructure to a broader audience," Lawrence Byrd, director of IP telephony at Avaya, told CRM Buyer. Avaya's partnership with Google initially will focus on integrating the IP Office product, which targets small and medium-sized businesses.
 
The Google service offers a powerful set of APIs (application program interfaces), Byrd noted, that should enable Avaya to create new solutions geared toward the needs of small businesses. Among the potential applications are tools for improving employee productivity Get the Facts on BlackBerry Business Solutions and optimizing communications -- whether they take place via a PC, a telephone, or a mobile device.
 
The combined solutions will be sold through Avaya's network of resellers and distributors and are slated to roll out by September, according to Byrd. The companies will jointly market and support the offers according to the terms of their agreement.
 
 

February 22, 2007

Craig McCaw Getting Set to "Pwn" IPO Market in March

Note:  I know what your saying "Craig McCaw Fanboy Alert"!  Well to answer that:  I just can't help it.  Whats not to love about this guy.  I am an old school geek (1984 was when my cherry was popped with my pimped out TI-99) so I have been reading about him for some time.  I was browsing on Seeking Alpha and they have this really nice read about Clearwire and Craig McCaw in general.  I think he is positioning himself for the auction of those 2.5 Ghz licenses under the AT&T/Bell South Deal (The "New" AT&T heh).  There are alot of striking similarities in his strategy thus far and it will be exciting to see what the future holds.

Craig McCaw 

"On the week of March 5, a service provider called Clearwire (CLWR) is going public on the NASDAQ with the assistance of Morgan Stanley, Merrill Lynch, J.P. Morgan Securities, Bear Sterns, and Wachovia Capital Markets. The excitement of the Clearwire IPO has less to do with their technology and the business accomplishments to date, rather more to do with who founded the company. The excitement around the Clearwire IPO has more to do with the founder and Chairman of Clearwire: Craig McCaw (pictured)." 

There is an interesting dichotomy in the social grouping of the people who have interest in Clearwire. In the first group we find the Internet Generation (i.e. born between 1986-1999) and in the second group with find Baby Boomer Generation (i.e. born between 1946-1964). Full disclosure I am a member of Generation X. Many months ago I posted a blog on the generation topic and social networking. You might be wondering what social generations have to do with the Clearwire IPO. Let us start with the first group the Internet Generation. If you have seen a Clearwire sales team in action, you are looking at a group of twenty-something age sales people. If you read the blogs of dedicated internet junkies, you will realize that the internet generation finds Clearwire great, more great, more cool, or they are not all that impressed, or have questions about the company and the WiMAX service offered by Clearwire.

Click Here to Continue Reading <---- Recommended Reading

QueueMetrics 1.3.3 for Asterisk Released Today

Today QueueMetrics 1.3.3 was released. This release offers experimental advanced clustering support, so that one instance of QueueMetrics can monitor multiple instances of Asterisk as if they were one large server. This feature is experimental, i.e. not production-ready, but we'd like you to try it and let us know how it is going.

This release also adds a bit of functionality and fixes a number of bugs that were present in 1.3.2, and notably:

  • New data base inspector tool. If you run MySQL storage, you'll see a new label "Mysql storage information" reporting on the different queues and partitions present.
  • By popular demand, showing the detail of the queue definition van be turned on or off using the "default.showQueueComposition" configuration parameter
  • Fixed a problem where UTF-8 input would not be correctly read
  • Fixed a problem with UTF-8 encoding of the wallboard
  • Fixed a problem with UTF-8 encoding of the soutrce JSP pages that would not work on some app servers
  • Fixed a bug with duplicate sesions being shown on very weird log data (#32)
  • Fixed a bug to delete stale data from the session cache (#36)

You can download the latest version immediately from the downloads page, together with the 98-page User manual. As an alternative, if you run RHEL/CentOS/TrixBox/AAH, you can install it automatically using yum - see the installation page.

If you would like to write a language pack for your native language, it's very easy and it only takes a couple of hours' work. See the Translating QueueMetrics document from the Downloads page.

We are looking forward to version 1.4 of QueueMetrics for major improvements, including a new high performance analysis engine.

February 21, 2007

Zaptel 1.2.14 Released for Asterisk PBX

The Asterisk and Zaptel development team has released version 1.2.14 of Zaptel. This release contains only minor changes, the most important of which relates to single-port module support on Digium's TDM800P analog interface card (previously these modules were not properly recognized by the driver).
 
 
Thanks for supporting Asterisk and Zaptel!

Shimon Releases WiFi Security Solution Using Biometric Technology

Note: Fun Stuff

Shimon Systems' Bio-Netguard a, featured product at the prestigious DEMO emerging technology product showcase and winner of an INNY award from The Tech Museum in Silicon Valley, the Bio-NetGuard uses biometric fingerprint verification technology to authenticate the user. Incumbent technologies authenticate the equipment accessing the WiFi network and do not verify user identity like Bio-NetGuard.

"One of the fundamental weaknesses of most wireless networks at small-to-medium businesses and home offices is that any equipment within range can gain access to company WiFi resources," said Dr. Baldev Krishan, president and CEO of Shimon Systems. "Small, medium and large companies can now prevent unauthorized use of their WiFi networks' resources, save bandwidth, and filter out rogue access points."

Bio-NetGuard leverages the fingerprint readers that are increasingly built into laptop computers, as well as a wide range of USB and PCMCIA card sensors. Fingerprint matching is very fast and accomplished with virtually no performance penalty. Bio-NetGuard also prevents client association with WiFi access points that are not WPA/WPA 2.0 configured or authorized by the administrator.

The plug-and-play Bio-NetGuard unit can connect either to the WiFi access point or LAN router and requires about five minutes to install and set up. Most common 802.11a, 802.11b, or 802.11g wireless networks can be protected by Bio-NetGuard; the product is also fully compliant with 802.11i, the next-generation WiFi security standard also known as WPA/WPA2.0. Supported WiFi access point devices include NetGear, Linksys, Cisco, D-Link, and Bountiful, with more equipment in queue for interoperability certification.

Each Bio-NetGuard unit is capable of securing multiple WiFi access points connected to the same router, allowing authenticated users to freely and seamlessly roam between access points -- even access points from different vendors -- without having to re-authenticate! This ability not only minimizes equipment cost, but also lowers administrative overhead.

Management of Bio-NetGuard, regardless of the number of units within the enterprise LAN, is accomplished via a single, easy-to-use administrative interface. The device is available in a fingerprint-only version, as well as a two-factor fingerprint and password authentication version for maximum network security. Both versions can be purchased in user configurations supporting from five users for small application all the way to 250 users for a large deployments. More than 250 users can be accommodated as well.

Bio-NetGuard requires the Windows (2000/XP/Vista) operating system in either home or professional versions. The device is powered by algorithms from NEC, a world leader in biometric technology, and Texas Instruments' DSP chip running custom Shimon firmware.

For more information about Bio-NetGuard visit www.shimonsystems.com.

February 20, 2007

OnState Communications Announces ACD for Skype 3.0

OnState Communications, announced OnState ACD for Skype 3.0. The OnState Automatic Call Distribution (ACD) solution is a fully integrated, Skype Certified third-party application, which is available as a Skype for Business Extra.
. OnState ACD for Skype does not require installation, hardware or special software yet it delivers enterprise-class customer contact management capabilities spanning services such as skills-based routing, chat, click-to-call Web integration, interactive voice response, and reporting. OnState ACD for Skype is part of a suite of OnState Intelligent CallCenter solutions for Skype solutions to be rolled out in the coming months.

"Skype is easy to use and a very affordable calling solution," said Pat Kelly, COO at OnState. "Our OnState ACD for Skype extends Skype's inherent value proposition to the call center and customer contact management industries." With OnState, businesses can select the call center services they need for just one or up to hundreds of users at a single, per-agent, per-month fee. There are no upfront or fixed costs and no commitments. "We're offering pay-as-you-go, pay-for-what-you-need call center management, which effectively redefines how call centers are administered, how services are priced, and how profit margins are calculated," continued Kelly.

The anticipated return on investment for OnState ACD for Skype has cross-industry appeal with "OnState ACD getting immediate market traction in help desk environments, travel-related services, manufacturing, and the emerging online communities of domain experts in the automotive, technology and medical fields," said Kelly.

Depending on the number of agents, the cost of maintaining a call center has traditionally been in the hundreds of dollars per month, per agent -- even when businesses rely on near-shore and off-shore resources. "Because OnState ACD for Skype does not use traditional hosted or VoIP call center options and requires minimal agent training and management, we can lower monthly per-agent costs to a fraction of what businesses typically invest," noted Kelly.

"It's no exaggeration to say that these new price points and ease-of-use features herald a new-generation call center business model that may even help businesses become less reliant on off-shore resources so that they can maintain local call center agents and ultimately drive increased customer satisfaction, loyalty, and retention." These benefits also improve the effectiveness and investments in existing call center agents and all customer-facing functions.

"OnState is part of a growing ecosystem of third-party companies that are developing truly innovative Extras for Skype to meet specific business-user needs in both small and large enterprises. Extras for Skype are third-party applications that are accessible from the Skype 3.0 software, which gives our users new ways to do more with Skype," said Paul Amery, director of Skype Developer Programs. "We're thrilled to see applications such as OnState ACD that capitalize on the advantages of Skype 3.0, while opening the door for Skype in the call center space."

Availability and Pricing

OnState ACD for Skype 3.0 is available through Skype as a plug-in, or directly from OnState Communications with trial versions available at: http://www.on-state.com/contact.html. General Availability is scheduled by the end of March 2007.

Monthly pricing for OnState ACD for Skype 3.0 starts at $29.95/per agent.

 

February 19, 2007

Polycom RMX 2000 Platform: Video Conferencing for the Future

Polycom will try to raise the bar for next-generation videoconferencing when it launches its new RMX 2000 conferencing platform on Feb. 20. As part of the effort to move video conferencing out of the prescheduled, conference-room-based realm, Polycom with the new RMX 2000 addressed several issues that have kept the technology from being used from the desktop in an ad hoc way, the company said.

The RMX 2000 is based on the Advanced Telecom Computing architecture, which delivers greater "performance, reliability and serviceability," according to Megan Bouhamama, product marketing manager for Polycom, based in Pleasanton, Calif.

The platform supports IP communications over high bandwidth links with lower latency. "It's two-and-a-half to three times better performance than products in the market now" can offer, and the platform provides greater motion clarity and reduces talk-overs between parties, Bouhamama said.

Beta testers at W.R. Grace & Co. said they believe the platform's quality is much better than that of past videoconferencing technology. "In the old days there was video and audio stutter. Nobody wants to look at jerky, five-frames-a-second video," said Guy Welty, manager of global media networks and collaborative services for W.R. Grace & Co., in Columbia, Md. "Now you get 30 frames a second, which is more like traditional video on a TV. Now the motion is so fluent, it's much more acceptable," added Welty, who has worked with videoconferencing technology for eight years.

Click Here to Continue Reading 

February 18, 2007

Iridium Set to Announce Plans for Next-Gen IP-Based Satellites

Iridium Satellite LLC, the company that raised Motorola Inc.'s expensive space-based network from the ashes of bankruptcy, is now planning a new generation of satellites that may be able to continuously monitor the environment and take pictures of Earth.

Iridium was launched in 1998 as a go-anywhere phone service aimed at executives, with outdoor coverage almost everywhere on the planet, including the north and south poles. But its high price and bulky handsets doomed the network to financial failure.

The craft stayed aloft and the current company, which took over in 2000, has had more success selling data communications to government and industries such as shipping and aviation along with voice, according to Matt Desch, Iridium's chairman and CEO.

Satellite Phone Rental 

Next week at the Satellite 2007 conference in Washington, D.C., the company will unveil an initiative called Iridium Next to build a next-generation network, or "constellation," of satellites. Over the next two years, Iridium will consider technologies and seek partners and financing for the system, which is expected to cost more than US$2 billion to build and deploy, becoming fully operational by 2016. Future services could include environmental monitoring, photography and a geographic positioning system to complement the current GPS (Global Positioning System), Desch said.

The current system, consisting of 66 satellites that form a mesh network, provides a baseline speed of only 2.4K bps (bits per second) but supports voice calls, e-mail and exchange of some data such as a ship's position, Desch said. The next generation may have speeds up to 10M bps and provide a broadband data experience, he said.

The new satellites may be able to constantly monitor environmental factors such as temperature and the level of the oceans around the world, according to Desch. They could constantly take pictures so enterprises or consumers could monitor facilities or homes anywhere in the world. And working with GPS, they could provide location accuracy down to feet or inches, he said.

The next-generation system will be totally based on IP (Internet Protocol), making it easier for Iridium to take advantage of new technology advances and for enterprises to integrate services into their existing applications, according to the company. Iridium sells more data modems than handsets today, and with the next generation it hopes to bring those modems down to a single chip that can be built into more devices, Desch said.

Iridium Next will gradually replace the current generation of satellites, which are expected to reach the end of their useful lives starting around 2014, Desch said.

Iridium sells wholesale access to the network that is then resold by partners, Desch said. Voice calls cost end customers about US$0.80 to $1.50 per minute and data services are typically priced per packet, he said. Modems cost about $300 and handsets start at about $1,200.

Source: PC World 

 

Getting Over the Love-Hate Dynamic With Open Standards

Note:  TechWorld does it again with some great commentary about open standards and the illusion of open technology.  I was even caught up in the iPhone hype not to notice that is in fact "does not" support SIP.  You would think that the more the device could support the more valuable and versatile it was.
 
"I'm old enough to have a long memory, but young enough that it does not yet resemble Swiss cheese. So, despite having consumed certain, uh, chemicals on more than one occasion, I actually still can recall more than a few things -- one of which was the promise of open standards." 
Once the industry migrated to IP (Internet Protocol) telephony, we were told, buyers would be living in a new world -- one where they could mix and match phones and other system.  Prices would fall; life would be good.Well, things haven't turned out exactly as promised. Yes, virtually every vendor bows to the great unifier SIP (Session Initiation Protocol), but adherence to SIP is often honored more in the breach. Indeed, there are times when it's not honored at all. Apple's iPhone, for example, announced with unbelievable fanfare at last month's CES, is not SIP compliant.
 
'Disappointing but Not Surprising'
 
To be sure, SIP calls can and do go between phones and terminals from multiple vendors, but with so many vendor-specific extensions, the promise of feature transparency between multivendor systems remains largely unfulfilled. This is disappointing, but not surprising, for reasons that have less to do with technology than business models -- both the vendors' and the buyers'.
 
Everyone has a stake in being able to connect devices, no matter from which vendor; in all but a few narrow exceptions -- the more connections a network can support, the more valuable the network is to its owner and customers. So, since the carriers have designated SIP as their default signaling protocol, the manufacturers of communications have to support it.
 
At least to talk to the carriers' network; what goes on behind the demarc (demarcation point), however, is quite another matter. So, here is where both vendors and customers have common interests. The vendors want to deliver as much functionality as possible for all the obvious competitive reasons. As a result, they offer SIP and other "open" standards as part of the basic package, but reserve the real goodies for their proprietary protocols and software.
 
 

Nokia N800 Review

Note:  MobileWhack has a nice review of the Linux based Nokia N800.  After reading through it sounds like this  has all the bells and whistles of a portable handheld computer.  Only issues seems to be a recurring one of handwriting recognition/ virtual  keyboard vs.  built in hard keyboard.  Personally  think until they really advance the software in that area, keyboards even if they are small are here to stay.

 

Review:

The Nokia N800 internet tablet features a large 4.1-inch 800x480 touchscreen display, which displays both text and pictures very clearly. Other features of the tablet include WiFi connectivity, Bluetooth, a mini-USB port, and a 3.5mm headphones jack. It runs off a 320Mhz TI CPU, and 128MB RAM, and includes 256MB of flash storage and two SD slots.

Alongside the top of the tablet is the on/off button, zoom buttons, and microphone. On the front of the Nokia on the left side next to the screen, is a a 5-way joypad as well as three other buttons - one for closing applications/windows, one for application menus, and another button for task switching.
 
At the bottom of the tablet is a kickstand which folds out to prop up the device, which is great for watching videos on.On the left side of the tablet near the top is a really cool pop-out VGA video calling camera. When not in use, it retracts nicely inside the Nokia.
 
Press on it to pop it out, and then you've got a nice little camera which because it is so discreet, it looks like a spy cam. Inside the box I found the tablet, a spare stylus, stereo headphones (w/ call answer/drop buttons and mic), a cloth protective pouch, a 128MB SD card with case, a USB cord, and a power adapter. The stylus is just the right weight - not too heavy, but not too light - and slides easily into its storage slot on the back of the tablet.
 

February 15, 2007

Inter-Tel CS5000 PBX Series Review

Note:
TechWorld has a write-up about the Inter-Tel iPBX series today.  After reading it, I noticed it is linux based and supports some of the same standard and codecs Asterisk supports.  If anything we can say Asterisk is having an effect on the whole PBX industry by looking at how they are supporting more standards that more pbxs support like SIP, Audio Codecs, Open Source software, etc..
 
" The CS-5000 series(the launch of whose CS-5600 prompted this review) is a VoIP platform, based on a Linux kernel. There are three variants in the range, each of which is basically a superset of the entry-level CS-5200.

I’ve come across Inter-Tel’s Axxess range of phone systems over the last few years in my “day job” as a consultant and freelance software developer. The Axxess is what you’d call a “traditional” PBX – a chassis-based PBX that you slot feature cards into, of which the VoIP option (the “IPRC” card, as they call it) is just one. The CS-5000’s takes a different tack, though, with the emphasis firmly on VoIP at the core.

The CS-5200 is a blue 1U box with a small LCD display on the front alongside some buttons that let you do some basic functions like setting the device’s IP address or rebooting it. There’s also a socket where you insert a RAM card that acts as the voicemail repository. On the back of the unit are a pair of analogue RJ-45s; one drives a pair of analogue trunks (ie you can connect it to a pair of analogue phone lines as your external service) and the other a pair of analogue extensions (so you can hang a couple of traditional phones, or more likely modems/fax machines, off the unit).

Obviously you’ll need to break out each individual RJ-45 into two sockets on your patch panel, but that’s a simple cabling job. Next to this are the interfaces for paging services and music-on-hold, and then you have the slot for the processor card (which has an Ethernet interface for connecting to your LAN).

You then have three slots into which you can put trunk cards; each card can handle a pair of ISDN2s (i.e. four channels) or a single ISDN30, though there’s a dual ISDN30 module due for launch shortly. Oh, and if you need more analogue extensions, you can put a four-port analogue module in one of the slots. Oh, and the unit comes with a built-in 4-channel voicemail (i.e. four calls can be talking to the voicemail memory at once) and a simple licence change ups this to eight channels. There’s also built-in CTI capability via the OAI protocol.

The CS-5200 supports up to 75 IP endpoints (ie handsets). The "up to" bit needs a bit of clarification, though. When you connect an IP phone, you can set it to use either G.711 or G.729 signalling.

Click Here to Continue Reading 

 

A Guide to Understanding the VoIP Security Threat

Note:  VN has a good article about VoIP Security threats how what to make of them.

"At its heart, a VoIP system is a data network. This means VoIP deployments are vulnerable to the same internal and external threats that plague any enterprise data local area network (LAN) or wide area network (WAN).

Enterprises pondering voice over Internet protocol (VoIP) primarily focus on the technology's cost benefits. Yet, in their zeal to converge voice and data networks and shave telephony costs, many organizations are failing to adequately consider VoIP's single drawback: security.
"

Like Seinfeld's George Costanza and the cashmere sweater with the little red dot, most VoIP supporters would prefer to ignore the ugly defect that mars their otherwise stainless technology. Unfortunately, VoIP's little red dot has the potential to cripple enterprise VoIP systems. Worse yet, VoIP's security gaps threaten to wreck havoc in several different, often insidious ways.

In-Stat, a US technology research firm, predicts that the number of business IP phones sold will grow from 9.9 million in 2006 to 45.8 million in 2010. Yet, the company ominously notes that over 40 percent of the enterprises it surveyed don't have any specific plans for securing their VoIP deployments. Additionally, when asked to rate their VoIP security knowledge, most enterprise managers In-Stat contacted characterized themselves as being "somewhat knowledgeable," the lowest rating the survey offered.

Locking Down Your System

There's no such thing as a bulletproof VoIP implementation, but there are a handful of fundamental steps you can take today to ensure that your system, or the systems that you're planning, will be highly secure.

According to network vendor Cisco, preventing unauthorized access to the network is a smart first step in a voice security program. For an additional layer of protection, in case somebody does gain unauthorized access, organizations can also encrypt voice traffic. Voice and video-enabled VPN (V3PN) technology, available in many routers and security appliances, encrypts voice as well as data traffic using IP Security (IPsec) or Advanced Encryption Standard (AES). Encryption is performed in hardware so that firewall performance is not affected.

Many security experts also recommend limiting VoIP data to a single virtual local area network (VLAN). A VLAN will keep voice network traffic hidden from data network users, providing an additional layer of security. The technique can also limit the scope of damage to the VLAN in the event of an attack. An additional side benefit is that a VLAN help prioritize VoIP data over other types of network traffic.

When creating the VLAN, be sure to place its equipment behind separate firewalls. This practice will restrict traffic crossing VLAN boundaries to applicable protocols and prevent viruses and other kinds of malware from spreading from clients to servers. When looking for firewall technology, be sure to examine products that support both leading standards: Session Initiation Protocol (SIP) and the International Telecommunication Union's H.323 protocol.

Data and Physical Security

By now, just about everybody is aware of the need for packet data encryption to safeguard VoIP transmissions. Yet call signaling encryption is important as well to prevent hackers from misdirecting or otherwise interfering with call traffic.

To install multiple encryption layers, turn to Transport Level Security (TLS), which encrypts the entire call process. The Secure Real Time Protocol (SRTP) is useful as well for encrypting communication between endpoints.

A secure gateway, properly configured, is a VoIP system's cornerstone. The gateway will limit system access to authenticated and approved users while keeping hackers safely on the outside. Gateways themselves, as well as the networks that lie behind them, can be protected through the use of a stateful package inspection (SPI) firewall and network address translation (NAT) tools.
 
 
 

February 14, 2007

TowerStream Rolls Out Los Angeles (LA) WiMax Service


 
 Towerstream announced the expansion of its Los Angeles WiMAX network to include the popular commercial district, Century City. Through the addition of new Points-of-Presence (PoPs) throughout the metro area, Towerstream will offer wireless Internet coverage to businesses in the entire west side of Los Angeles, home to many important film, television and music industry businesses.
Century City is an important business district outside of downtown Los Angeles; therefore, Towerstream's reliable and flexible fixed wireless broadband service is especially needed in this densely-populated area. Media and entertainment companies there, in particular, are in need of higher bandwidth products not currently offered by legacy phone companies and Towerstream is able to provide connectivity options to meet the diverse needs of these businesses.

"Following completion of Towerstream's recent financing, we are committed to deeper penetration in our major metropolitan markets and adding additional cities throughout 2007," said Jeff Thompson, President and CEO of Towerstream. "Expansion into Century City has been one of our important near-term goals.

Our fixed wireless broadband, supporting high bandwidth video capabilities, is a tremendous asset to many businesses that could benefit from a more robust broadband solution not currently offered by traditional wireline providers. Our primary objective has been, and will always be, to offer exceptional broadband in the cities we serve. We are pleased that businesses on the west side of LA will be able to take advantage."

Source: TowerStream 

Clearwire Moves to the Nasdaq with Wimax Offering

Note: This should be interesting.  I hope them all the best in their initial offering.
 
 
Investors have a wide range of technology to choose from in the stocks they buy, and Clearwire Corp. is set to add more WiMax to the list, when it offers 20 million shares for sale at between US$23-$25.00 on the Nasdaq Stock Market.
The Kirkland, Washington wireless broadband Internet service provider Tuesday issued a preliminary prospectus for a share sale with the U.S. Securities and Exchange Commission (SEC), and said it planned to launch the public offering as soon as possible after the share sale is registered. It did not give a specific date.

Clearwire will become one of just a few WiMax companies listed on a U.S. stock market, and one of the most high profile due to its relationships with Intel Corp. and Motorola Inc. The listing may also be a sign that word of WiMax technology is spreading. The popularity of sipping coffee at Starbucks while reading e-mails on a laptop PC has grown along with wireless technologies such as Wi-Fi. WiMax aims to replace Wi-Fi as a speedier service with far wider ranging access.

Clearwire will list under the stock ticker code, CLWR.

Clearwire is also offering 3 million additional shares to the investment banking firms underwriting the sale, and if all 23 million shares sell out at the mid-range price, $24, Clearwire will pocket $513 million from the sale, it said in the SEC filing.

The money will mainly be used to finance its business. It has spent millions expanding its WiMax network, and expects to need more to continue growing.

"As Clearwire is at an early stage of development, we cannot anticipate with certainty what our earnings, if any, will be in any future period. However, we expect to incur significant net losses as we develop and deploy our network in new and existing markets, expand our services and pursue our business strategy," it said in the filing. The company has lost money ever since it opened, and estimated its total indebtedness as of Sep. 30, last year, at approximately $755.7 million.

But the company has some big backers to help keep it running, mainly companies promoting WiMax.

Intel's venture capital arm invested $600 million in Clearwire last year in one of its single biggest deals ever. Intel Capital commonly invests in companies tied to Intel strategies aimed at growing its core microprocessor business. WiMax has been a major push for Intel over the past few years. The company's line of Centrino notebook PC chips, which put Wi-Fi into portable PCs, was wildly successful on the market, and some analysts say the company could be a big winner with a similar chip package developed around WiMax.

Clearwire and Intel have also agreed to jointly develop and promote a mobile WiMax service offering as a co-branded service available only over Clearwire's WiMax in the U.S., according to the prospectus, to target users of laptops, ultramobile PCs, and other mobile devices.

Once Clearwire completes its share sale, Intel will still hold a 26.6 percent stake in the company, or 36.76 million shares, while Motorola will hold a 12.9 percent stake, or 16.67 million shares, according to the prospectus. Clearwire sold its wireless broadband equipment business to Motorola last year, and indicated in the prospectus that it will rely on Motorola to supply equipment for its network.

Clearwire currently offers broadband wireless services in the U.S. and Europe. Its subscribers rose to 206,000 as of the end of last year, from just 1,000 on Sep. 30, 2004, it said. Its subscriber base has grown despite wireline alternatives for users, including broadband cable modems and DSL (digital subscriber line) Internet service.

As of the end of last year, Clearwire services were available to 9.6 million people, including 8.6 million throughout the U.S., and one million in Brussels, Belgium and Dublin, Ireland, the company said.

WiMax base stations can send broadband Internet signals to far greater distances than the Wi-Fi technology that WiMax is meant to replace. Although estimates vary on how far WiMax signals can go, in densely populated cities, where users are not likely to be positioned within sight of access points, the distance should be between 2 km and 4 km.

A few of the other listed companies in the WiMax field include device maker Proxim Wireless Corp., service provider Alltel Corp., and WiMax network equipment maker, Alvarion Ltd. The stock ticker codes of the three companies are PROX, AT, and ALVR, respectively.
 
Source: WebWererld 
 

 

February 13, 2007

Linux-powered IP Video Phone Dials Numbers from Web Pages

Note:  Another cool device from LD.  The videophone even has a version of Mozilla built in.

 

"Iwatsu Electric Co. used Linux to build an IP videophone that lets users place calls by selecting numbers from web pages. The NR-IPKTV includes Opera Software's Opera for Devices browser along with an email client, and targets business users. " 

The NR-IPKTV business phone is one component of Iwatsu's Precot (Premium Communication Tool) product, an all-inclusive VoIP package for small offices. Computer translations from Iwatsu's Japanese-language website suggest that Precot includes an IP PBX server (pictured below) available in three options that support 16, 24, or 48 users, respectively. The Precot server appears to incorporate a gateway PHS (personal handyphone system) network gateway.
 
In addition to the browser-enabled NR-IPKTV, the Precot package appears to be available with handheld cordless phones, door phones, and several basic desktop models. The NR-IPKTV is based on a Freescale i.MX21 processor, a power-stingy ARM9 SoC (system-on-chip) generally found in mobile devices. A full multitasking Linux-based operating system runs on the processor, according to Freescale, enabling users to browse the web while videoconferencing or making calls.
 
The phone exploits the i.MX21's media engines for H.263, MPEG-4 video, JPEG, eliminating the need for separate DSPs to handle G.711 compression and echo cancellation, the chip vendor says. On the software side, the NR-IPKTV uses a customized version of Opera Software's Opera for Devices browser, which is available "pre-integrated" with Freescale's i.MX processors.
 
According to Opera, the browser was enhanced with "phone-to-tag" support said to enable users to place calls by simply selecting phone numbers from web directories. Another interesting feature of the device appears to be some kind of "desktop shopping" deal with Amazon of Japan that lets users make purchases -- possibly of office supplies -- from the interface.
 
According to Scott Hedrick, executive VP of Opera, "Iwatsu's Precot solution is breaking new ground in integrating the Web into IP telephony. Having a full Internet browser as a part of the phone allows rich integration of corporate data, Web applications, voice, and video calling. Users will be able to initiate calls directly from a Web site or the corporate intranet."
 
Availability
 
The NR-IPKTV appears to be available now, in black or silver, as part of Iwata's Precot IP Phone Package.
 
Source: Linux Devices 
 

New VoIP Call Recording Software for Mac OS X Released

Arcosoft Inc., announced the release of VONaLink SoloRecord for Mac OS X. SoloRecord works with any VoIP phone system based on the open SIP standard, such as Vonage, to record phone calls and to provide screen pops. Also available is a separate ScreenPop product, if recording is not required. SoloRecord and ScreenPop were previously available only for Windows.
With traditional phone systems, calls are recorded with either analog equipment or expensive, proprietary products from the phone company. With the latest VoIP systems built on open, standard protocols, calls can be recorded by monitoring network packets. Call recording benefits a company by allowing business transactions over the phone to be verified and disputes resolved.

VONaLink SoloRecord works with any SIP based VoIP system, such as Vonage. Soft phone, hard phone, or analog phone via an adapter are supported. The call is recorded as a stereo WAV or MP3. An inaudible watermark can be added to the recording for later verification that the file has not been changed.

Using the caller ID of the incoming call, SoloRecord searches for the caller in Address Book, or launches custom applications to search the web or company database. If the caller is found, the information is popped on the screen, thereby increasing the productivity of the user.

Unwanted callers can be added to the reject list. Integration with Vonage Click2Call allows outbound calls to be placed by clicking in the call log within the VONaLink application.
 
Price and Availability

SoloRecord and ScreenPop are universal binaries that run on Mac OS X 10.3 and later. Prices are $99.00 USD for SoloRecord, and $29.00 USD for ScreenPop. Evaluation downloads are available from: www.vonalink.com
 

10 Reasons To Switch To An IP PBX

Note:  TelephonyWorld.com has a write-up about different reasons to move to a IP based PBX Platform.

"
An IP PBX is a complete telephony system that provides telephone calls over IP data networks. All conversations are sent as data packets over the network. The technology includes advanced communication features but also provides a significant dose of worry-free scalability and robustness that all enterprises seek.

Enterprises don’t need to disrupt their current external communication infrastructure: An IP PBX is able to connect to traditional PSTN lines via a VoIP gateway - so an enterprise can keep its regular telephone numbers.
 
How It Works:
 
An IP PBX system consists of one or more SIP phones, an IP PBX server and optionally a VoIP Gateway to connect to existing PSTN lines. The IP PBX server functions in a similar manner to a proxy server: SIP clients, being either soft phones or hardware-based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection.
 
The IP PBX has a directory of all phones/users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider.
 
 

February 12, 2007

Polycom V700 Lowers the Barrier for Executive Desktop Video

Note: This is a pretty value priced video conferencing solution.  I had a chance to listen in on the announcement of this product and was excited.  I use a IP 501 with Asterisk as our office pbx solution.  Now we are just waiting on Polycom to release a Full Color LCD IP Phone.  *hint hint

 

"Polycom, Inc. released the V700, a new executive desktop video conferencing and collaboration solution designed for businesses wanting to add video communications at the desktop. The Polycom V700 is a complete, all-in-one system featuring quality video conferencing, an integrated display that can double as a PC monitor, integrated camera, microphones and speakers, and content sharing capabilities." 

"As video becomes a critical communications tool within organizations, customers now have the choice of offering more employees award-winning Polycom video systems at their desktops with these new, lower-priced, high-quality V700 systems," said Ed Ellett, senior vice president and general manager of video communications at Polycom. "Polycom offers the industry's broadest range of standards-based video solutions from high definition telepresence environments and conference room solutions to desktop solutions for virtually every worker -- all of which interoperate seamlessly for enterprise-wide and inter enterprise communication."

Desktop video conferencing is set to experience significant growth according to industry analyst firm Frost & Sullivan, which projects the shipment of personal video conferencing systems (including executive desktop video conferencing systems) to grow at a compound annual growth rate of 24 percent over the next five years(1). In addition, a recent survey of attendees at the 2006 Polycom User Group conference, the annual meeting for the independent association of Polycom users, showed that 70 percent of respondents plan to either deploy desktop video conferencing or expand their current desktop video conferencing deployments in 2007.

The V700 delivers solid business-quality video conferencing capabilities, including, full-screen, full-motion 30 frames-per-second (fps) TV like video at up to 768 kbps bandwidth speed; Siren 14 wideband stereo audio for near CD quality, clear, natural voice communication; as well as standards-based H.239 high resolution content-sharing. The system features an integrated 17-inch LCD display, built-in camera, and microphones and speakers capable of delivering 14kHz wideband audio voice quality. The V700 also includes an intuitive remote control and audio prompts in multiple languages for easy use.

The V700 is the second entry in Polycom's value line of video conferencing and collaboration systems, which currently includes the Polycom V500 set-top video calling appliance for conference rooms that works in conjunction with a standard television. The V700 and V500 systems are completely interoperable with Polycom's market-leading VSX and HDX room and desktop-based video solutions, RPX HD telepresence suites, and Polycom's video network management systems.

"The V700 gives customers a complete, integrated executive desktop system that works with all other standards-based video conferencing systems and delivers solid performance and business-critical capabilities," said Ellett. "The V700 and V500 are well suited for remote workers and remote locations that may not have as much available bandwidth as a corporate office."

Pricing and Availability

The Polycom V700 is currently available for order worldwide through qualified Polycom channel partners for a MSRP of US$2,999.

Source: Polycom Inc.

Avaya IP Office PBX adds SIP Support

Avaya Inc. announced significant enhancements to its IP Office Telephony Solution specially designed for small and mid-size businesses (SMBs). Avaya's new developments also enable companies Session Initiation Protocol (SIP) based trunking, which can cut calling costs in half. SIP is a protocol governing voice and data signals traveling together on a converged network.

A SIP trunk is similar to a phone line, but routes calls over an IP network instead of the public switched telephone network.The enhancements give even the smallest firm an easy, affordable way to take advantage of advanced, IP-based communications to improve productivity, customer service, and virtually any business process.

Based on Avaya developments in converged communications technology, the company is introducing two versions of enhanced software for Avaya IP Office, the company's flagship solution for SMBs. The Standard edition is designed for very small businesses and serves up to 32 users. The Professional edition enables firms to take advantage of enhanced mobility, multi-site networking or customer service intelligence and scales to support up to 270 users.

In addition to new IP Office software, Avaya is expanding its current portfolio of servers by introducing a new IP Office 500 communications server. Its compact, modular, and flexible design supports telephony, voice messaging, and a complete customer service suite -- all at an attractive price for small and mid-size businesses.

Together, the new IP Office software and communications server deliver a solution that enables SMBs to buy only what they need today, while easily adding users or functionality via software licenses that are also compatible with existing IP Office solutions.

"We built IP Office from the ground up for small businesses. That means it has the right entry-level price, 'big business' communications capabilities, and options for companies to add more capabilities, capacities or locations as their business grows," said Geoffrey Baird, vice president and general manager, Appliances, Mobility and Small Systems Division, Avaya. "With this latest release, we believe Avaya certified resellers will now be able to pursue a broader market, as well as better serve their existing customers. In fact, because of the easy way to upgrade for existing IP Office customers and the new features such as SIP trunking and remote diagnostic tools, our channels will be able to make more money and better serve and support the SMB market. And, in the future, we'll continue to incorporate new benefits into subsequent releases of our IP Office portfolio."

Over the last two years, Avaya and its worldwide network of certified resellers have nearly tripled IP Office sales to companies of all sizes in nearly 70 countries.

New Features:

-- An application that offers a number of benefits to SMBs:

-- It acts as a SIP gateway, whether the SMB has digital, analog or IP phones. Businesses can subscribe to new SIP trunking services and significantly reduce calling costs.

-- It introduces new remote diagnostic tools. Resellers can improve customer satisfaction by quickly and proactively identifying and resolving problems remotely.

-- It adds new multi-site hot-desking capabilities so employees can log into any digital, analog or IP phone at any of their company's networked locations and have their calls and voice messages follow them.

-- It adds new multi-site hunt group capabilities so workers at any networked location can be part of the same call routing group.

Source: Avaya 

Wireless Sensors Extend Internet's Reach (Oh Joy)

Note:  Well I can see some really great uses for this technology.  I love the pollutant monitoring solutions that could test for toxics an then send an alert when some foreign substance was found.

"To the untrained eye, the sleek, airy building constructed atop a decommissioned nuclear reactor at the University of California, Los Angeles could pass for high-tech office space.  A closer inspection of the glass-and-steel facade reveals dozens of miniature, low-resolution cameras and sensors. They're wirelessly linked to computers throughout the 6,000-square-foot space, keeping tabs on traffic flow in public areas and monitoring temperature, humidity and acoustics." 

The building serves as a testing ground for developing and perfecting wireless sensing technology to connect major chunks of the real world to the Internet. Such networks could monitor the environment for pollutants, gauge whether structures are at risk of collapse or remotely follow medical patients in real time.

"I see this as the next wave of extending the Internet into the physical world," said computer scientist Deborah Estrin, who heads the Center for Embedded Networked Sensing, a UCLA-based consortium of six schools.

The researchers at the consortium have already scattered wireless networks of nodes in the rice paddies of Bangladesh, rain forests of Costa Rica and wilderness of California's San Jacinto Mountains - all for the sake of keeping a closer eye on the world.

Once the stuff of science fiction, wireless sensor networking is quickly catching on, attracting the attention of the military, academics and corporations. Just as the Internet virtually connected people with personal computers, the prospect of wireless arrays sprinkled in buildings, farmland, forests and hospitals promise to create unprecedented links between people and physical locations.

Click Here to Continue Reading 

February 10, 2007

Ubiquiti Networks Releases ExtremeRange2 (XR2) WiFi Card for Linux with a 600mw Radio (That's Huge)

Note:  After browsing LinuxDevices.com I came across an announcement about new mini-PCI card for linux that boast a huge radio (600mw) transmitter.  Just to put it into perspective the old school Orinoco cards before that tuned down the wattage were 100mw and we did some 15km range tests without any issue.

"Ubiquiti Networks is shipping what are claimed to be the first-ever mini-PCI-based WiFi radios to boast 600mW transmit power, and the first to support operating temperatures from -45 to 95 degrees Celsius. The ExtremeRange WiFi modules have both been tested extensively under Linux, the company says." 

The ExtremeRange2 (XR2) is a 2.4GHz 802.11b/g card, while the ExtremeRange5 (XR5) is a 5GHz 802.11a card. They are designed for use in outdoor access points and bridges -- devices that often use 802.11a for backhaul, and 802.11b/g for user access.
 
Both the XR2 and XR5 are based on Atheros's sixth-genaration 802.11a/b/g chipset, and include advanced features such as 5/10/20/40MHz channel widths, compression, QoS, and the latest WPA security standards, Ubiquiti says. Ubiquiti says both have been tested with the open source Atheros MADWIFI driver, and with Linux router distributions that include Mikrotik, StarOS, Antcor Ikarus, and OpenWRT.
 
Additional claimed features include:
 
* Industry best receiver sensitivity
* Advanced filtering for improved noise immunity
* Innovative built-in RF surge protection design
 
Ubiquiti president Robert J. Pera stated, "We completed our initial XR development successfully using industry-standard throughput and radio tests, [then] decided to diligently investigate radio performance under real-world scenarios using the Linux MADWIFI driver and popular third-party routing software. It was in this development cycle that we nailed some critical radio optimizations that really makes performance exceeds all expectations in outdoor environments."
 
Availability The XtremeRange X2 and X5 are shipping now, priced at about $110.00 from online retailers such as Microcom and WISP-Router. They are also available in high volume direct from Ubiquiti. Additionally, extended frequency versions will ship in Q2 for carriers licensed to operate in the 2.3-2.7GHz and 5.0-6.1GHz frequency ranges, Ubiquiti says.
 
Source: LinuxDevices 

Experimental Skype Tweak Broadcasts You in 640x480 (w00t!)

Note:  We have this working on the new version of Skype for the Mac in the office.  It was a pretty nice improvement for such a simple edit on the xml preference file.  By the way, I love Skype.  With my Black Macbook and a $20 dollar bluetooth headset from Fry's it is a pretty sweet setup.

"Although we enjoy using Skype to talk with our parents our hos in different area codes, its lousy 320x240 quality makes even YouTube look great. Good thing some developer named Jason came up with a sweet "hack"—more like a tweak—to get your Skype video up to 640x480." 

Just grab the latest version of Skype for PC/Mac, and find a certain file and change the dimensions to 640 and 480. Easy peasy on the Mac, but the Windows version requires a bit more tweaking thanks to the various webcams available.

PC Version [Skype]
Mac Version [Skype]

Experimental High-Resolution Skype for Mac and PC [Make via Crunchgear]

February 09, 2007

Asterisk 1.2.15 Released!

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.15.
This release contains a large number of bug fixes, and some significant improvements:

* Support for Zaptel-based transcoder hardware, initially the Digium TC400B 92/96 channel transcoder.

* Handling of voicemail subdirectories when using ODBC storage has been improved, so that messages can be forwarded properly.

* A problem with forwarding voicemails from folders other than the user's INBOX has been fixed.

* The Zaptel channel driver can now support echo cancellers that provide 64ms or 128ms of echo cancellation per channel.
 
Click Here to Download

Thanks for your support of Asterisk and Zaptel!

Zaptel 1.2.13 released!

The Asterisk Development Team is pleased to announce the release of Zaptel 1.2.13.

This release contains a large number of bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware.

Release Notes:

* A modification was made to the drivers for all Digium PCI cards to improve their compatibility and performance when used in interrupt sharing environments.

* Support for the Digium TDM800P 8-port analog interface card was added.

* Support for the Digium TC400B 92/96-channel transcoder card was added.

* Support for the Digium High Performance Echo Canceller add-on software module was added.

* All drivers updated to Linux kernel 2.6.20 API changes.

* Performance improvements for multiple Astribank units.

* Astribank firmware protocol version is now 2.4.

* Astribank now supports Message Waiting light on analog telephone sets.

* Added a /proc interface to blink the leds on the Astribank to identify ports in large setups.

* fxotune is now supported by Astribank.

All users of Zaptel 1.2.x are encouraged to update to this release as soon as they can practically do so. Thanks for your support of Asterisk and Zaptel!

Telephony Mashup Contest from O'Reilly Media

O’Reilly Media and StrikeIron are proud to announce the first ever Telephony Mashup Contest. This new contest provides a stage for developers to demonstrate their creative skills using emerging telephony technologies such as VoiceXML, PBX, IVR, and Web Service APIs.

A telephony mashup is a voice, Web or mobile application (VoiceXML, PBX, IVR, VoIP, SMS, Text Messaging, etc.) that combines content from more than one source to create a new user experience. Qualifying entries must demonstrate how an application can use one or more sources of content in an inventive way to benefit users.

Any tool or platform that involves content (see StrikeIron or ProgrammableWeb) telephony (ex: VoIP, SMS, Text Messaging, PBX, IVR) can be used to create a mashup. This is uncharted territory, so there is plenty of room to use your imagination!!

The first round of the contest is open to all developers. Mashup submissions must be made by Feb 20th when finalists will be chosen. The contest is timed to conclude on the first day of the O’Reilly Emerging Telephony Conference where finalists will demonstrate their mashup at the conference. The winner will be chosen by conference attendees.
 

Proxim Packs Wi-Fi and WiMAX Access in One Mesh Networking Box

Note:  I think this reads like a good product for deploying wireless mesh networks using WiMax as the backhaul and node to node communications.  Strap this with Asterisk and you could have a robust campus or metro voip network.  Hopefully it has some Quality of Service (QoS) capabilities so a person could prioritize the voice packets?
 
"Proxim Wireless has launched a low-cost, outdoor wireless-mesh access point that combines Wi-Fi and WiMAX radios, claiming it was the first to do so.

The MeshMAX product makes it possible for many, even all of the nodes in a mesh network to have a backhaul connection to a base station and the Internet. Such an arrangement can boost the capacity of these networks dramatically because the client devices connecting to each node share a much larger broadband pipe to the Internet, claimed Proxim.

Typically, today's wireless mesh networks have nodes - access points-- that route data traffic among themselves over optimal paths and around failed or congested nodes. In addition, only a few nodes make a separate wired connection to the Internet, so mesh networks have to be planned, managed and calibrated carefully to minimise the number of hops from client device to Internet.

Some products, such as those from Wi-Fi mesh vendors including BelAir, Firetide and Tropos, will use 2.4GHz radios for client connections, and separate, dedicated 5GHz node-to-node communications. Some deployments can connect these nodes physically to a separate broadband wireless radio, including WiMAX, for the backhaul connection.

Proxim's MeshMAX product does the same thing, with one 802.11b/g radio in the 2.4GHz band and one 802.11a radio in the 5GHz band for Wi-Fi connectivity. These radios are based on the company's Orinoco mesh products, but MeshMAX adds a built-in, dedicated 802.16d, fixed, WiMAX subscriber-unit radio from Proxim's Tsunami fixed-wireless product line, which can make the backhaul connection directly to a WiMAX base station. Thus, client, mesh and backhaul connectivity are integrated into a single box. The WiMAX radio, based on an Intel chipset, works in the 3.3G to 3.6GHz licensed band or the 5.1G to 5.8GHz unlicensed band. MeshMAX also has a Layer2 Ethernet switch.

"It's cheaper to install [than two separate boxes], both capital and operating expenses are lower, and it's all managed by our network management software," said Pankaj Manglik, Proxim's president and COO.

Proxim's overall strategy isn't unique. Nearly every Wi-Fi mesh vendor has announced plans to add WiMAX radios in the future. Recently, NextWave Wireless , a supplier of WiMAX gear to carriers and operators, acquired Go Networks for its Wi-Fi mesh and radio beam-forming technologies. NextWave also plans to combine both types of radio into a single product. However, Proxim said it was the first to ship such a product.

Click Here to Continue Reading this Article 

Cisco buys social networking company Five Across

Note:  Well I hope they can make a community that is fast as well as their switches :)

Cisco Systems Inc. acquired a small social networking company that allows businesses to create MySpace-like communities on their Web sites.  Cisco said Friday that it was paying an undisclosed amount to acquire privately held Five Across Inc., an 11 person San Francisco company whose software allows companies to add user-interaction functions and multimedia-sharing capabilities to their Web sites.

Five Across' publishing platform allows users to create personal Web pages and post photos, videos and audio clips, much like the proprietary system used by News Corp.'s MySpace.

Cisco said the acquisition, its 116th since 1993, is the company's first in the social networking space but likely not the last. The deal is expected to close within the current fiscal quarter.

Analysts said the acquisition helps further Cisco's expansion beyond its role as purely a network equipment provider and into helping distribute the media that drives bandwidth consumption and even more network upgrades.

Danielle Levitas, a senior analyst at market researcher IDC, said the Five Across acquisition could help Cisco win greater access to a wide range of companies, particularly those in media and entertainment, looking to upgrade their Web sites to connect with customers.

"I actually see this as benefiting their core business — if they can promote users using their broadband more, that's huge for them," Levitas said.

Cisco has profited mightily in recent quarters from surging sales of its routers and switches as service providers and other companies scramble to upgrade their networks to prepare for the next generation of video and other bandwidth-intensive downloads.

Click Here to Continue Reading 

February 08, 2007

Jajah Mobile Web has Arrived and More...

Note: Fredrick from JaJah sent is some information about the new products rolling out of JaJah

Thanks to our users and your support it has been a good year for Jajah. We are growing quickly and working hard to create great new products. Today we are launching Jajah Mobile Web and the Jajah Dynamic Buttons Beta Enrollment.

 

### Jajah Mobile Web ###

This morning we launched an all new Jajah Mobile Web (http://mobile.jajah.com) - it is one-click access to our free or low-cost global calling service, directly from the browser on your smartphone (no application download).

 * One-Click Free (or low cost) Global Calling - Jajah Mobile Web provides one-click instant access to the contacts in your Jajah address book. Just click a name or number to make the call.

 * Bookmark Names and Numbers - Save your most called numbers as bookmarks.  Click on the bookmarked name and JAJAH instantly dials that person.

 * Avoid Roaming Charges - It gives you the ability to change your source number even to new numbers which are not yet saved within your Jajah account. This means that you can easily initiate a Jajah call to a nearby office or hotel phone when you are travelling, getting rid of roaming charges and long distance charges at the same time.

 * International Calls without a special plan - In many places, and especially the US, most phones can't make an international call without a special plan from your provider. The plan alone can cost 50.00 US a year, and then the calls can cost .20-.40 a minute above that.

 * It Knows You - JAJAH Mobile Web links directly into your regular JAJAH account. When you log on from your smartphone, it knows who you are, displays your contacts and  and displays your account information..


Little demo video: http://www.youtube.com/watch?v=LjKUDOYUtPk

Some screenshots:
http://flickr.com/photos/jajah/sets/72157594523069814/

And the press release itself:
http://blog.jajah.com/press_kit/jajah_mobile_web.html


### Jajah Dynamic Buttons ###

Secondly, I would like to invite you to join the JAJAH Dynamic Buttons Beta Program. We want you to be involved in the design process of what we think will be revolutionary new web based calling tools. Jajah Buttons will let you control how and when you want to be contacted - through an embedded button that you put in various places around the web - your personal or corporate website or blog, any social networking platform like MySpace, Friendster, hi5, any auction based website like eBay, craiglist or your email signature.

We've put up a page to get the conversation going and show you our ideas. On that page you'll find a place to enter your name to get on the Beta Program list. Please sign-up and we'll be in touch to get your opinions and feedback as we move forward.

Functional Description

Jajah Buttons enable users to make "special feature" calls from within on-site properties such as blogs, websites or emails. They can be described as portable, customizable "click to contact" buttons with user defined call and contact properties. When clicked, a small window opens which can contain
information about the button owner, or information relevant to the contact preference, interaction or transaction.

It also displays contact preferences, as defined by the button owner; phone, sms, email or voice mail. The reader/viewer of this page is able to click on one of the contact options to reach the button owner.

These buttons allow the button owner full privacy, with no numbers or addresses being visible to the caller. They are dynamic and interactive.  They have various settings to control button behavior. Settings include the
ability to restrict each function by days or hours, set button expiry date, choose who pays, change receiving number/email, block incoming numbers/names.

 * They allow you to make yourself available without compromising privacy.

 * Buttons work independently from phone numbers, location or email address.  User can change his contact specific in the jajah preferences.

 * Users can have multiple buttons for multiple purposes/audiences

 * Buttons eliminate the need for extra email addresses. Since your email address is private and non-visible to viewers, you can use a single address for multiple types of interaction.

 * User can create email signature that enables managed communication

 * User is able to block specific people from contacting him.

 * User can set the time he allows / accepts calls. For example, a person can specify that he be called between certain hours but at other hours, only be reached via sms or voice mail.

Check them out, sign-up for the Beta Program and we'll be in touch!

Beta Sign-Up: http://www.jajah.com/info/services/buttons/

Mockups/Screenshots: http://flickr.com/photos/jajah/sets/72157594523076366/

#############

 

Polycom to acquire SpectraLink for $220 million

Note:  This is an interesting purchase announcement.  I think it is safe to say we will be seeing a sweet Polycom WiFi headset being released in the near future to help bolster their SoundPoint IP Phone line.

"Polycom this week announced plans to acquire SpectraLink, a maker of Wi-Fi VoIP handsets, for $220 million. The deal gives Polycom a complete wireless IP phone product set, and also strengthens its ties with such IP PBX system makers as VoIP Avaya, Cisco and Nortel, which resell SpectraLink technology and products as part of their respective wireless VoIP offerings."

SpectraLink makes 802.11-based IP telephone handsets and gateways. SpectraLink also makes devices that prioritize SpectraLink wireless LAN VoIP signals over other 802.11-based traffic, to provide QoS.

The company's handsets are used in deployments with mobile workers, such as hospital, warehouses and public safety organizations. The company also makes dual-mode Wi-Fi/cellular handsets, which can operate on a corporate WLAN and PBX infrastructure, as well as on carrier cell phone networks.

Founded in 1990, Spectralink has 440 employees and is publicly traded. Polycom was founded in 1990 and has 1,200 employees worldwide.

Click Here for the Full Article 

 

Telrex Announces CallRex Release 3.5

Telrex, developer of VoIP call recording and monitoring software for businesses announced the release of CallRex Professional 3.5 featuring an advanced distributed-services software architecture. CallRex 3.5 delivers greater scalability, high reliability and additional security, plus new features for multi-site deployments, including enhanced multi-site call recording and monitoring, advanced file transfers, flexible storage and streamlined archiving.
CallRex was the first VoIP call-recording solution verified to record encrypted VoIP calls for Cisco CallManager 5.0, and CallRex version 3.5 now provides additional security to ensure that call recording files cannot be secretly altered.

Distributed-Services Software Architecture

The innovative CallRex 3.5 distributed-services software architecture features separate services for status, control, recording and conversion of VoIP phone calls. CallRex 3.5 services operate independently and are deployed on industry-standard Windows servers in single- or multi-server configurations. Together, the CallRex 3.5 services enable the recording, monitoring, retrieval, playback and utilization of call recordings from any location. This next-generation design provides high reliability and enables virtually unlimited scalability across multiple servers and multiple office locations.

“The CallRex 3.5 platform gives us the enterprise scalability that our growing company requires,” says Jay Whalen, Manager of I.T. Infrastructure at Global Forex Trading (GFT), a world–leading provider of real–time currency and equity derivatives services for retail and institutional traders. “With CallRex 3.5, we have a great deal of flexibility in how we deploy and manage CallRex for our worldwide office locations and remote users.”

Enhanced Multi-Site Support

Businesses and call centers with multiple office locations benefit from the enhanced multi-site support of CallRex 3.5. In addition to providing full-time and automatically triggered call recording, CallRex 3.5 enables on-demand recording and real-time monitoring of employees’ phone calls from any location across the entire organization. Through the CallRex 3.5 user interface, authorized individuals are presented with the real-time status of every licensed IP phone on the system. Supervisors and managers are able to selectively monitor and record designated users regardless of their location.

CallRex can be physically deployed across multiple sites or located centrally depending upon customers’ infrastructure requirements. CallRex software is licensed on a per-user basis resulting in no additional software costs to deploy multiple servers. CallRex 3.5 is managed as a single virtual system from any location providing for ease of administration.

Streamlined Storage and Archiving

CallRex 3.5 provides multiple options for storing recorded calls, and delivers enhanced archiving features to streamline management and administration. With CallRex 3.5, call recordings can be stored on a centralized server, stored locally across multiple servers, or stored directly on Network Attached Storage (NAS) or on a Storage Area Network (SAN).

Advanced file transfer options enable call-recording files to be transferred automatically in real time, or transferred in batch at off-peak times for efficient network bandwidth utilization. Batch transfers can be initiated automatically according to flexible predefined schedules set globally or on a per-server basis. As call recordings are transferred from primary storage to archived storage, CallRex 3.5 enables the automatic tracking of files across multiple locations over time, ensuring ready access to call recordings.

Additional Security

CallRex is the first IP-based call recording and monitoring solution verified to record encrypted VoIP calls with Cisco Unified CallManager 5.0. Previously, companies deploying VoIP call-recording solutions were forced to disable call encryption in order to record their calls. Now with CallRex, companies can both record their calls and maintain secure call encryption to meet stringent security, legal and regulatory requirements. Telrex is a partner in the Cisco Technology Developer Program and has met the interoperability criteria for Cisco Unified CallManager version 5.0.

CallRex 3.5 delivers digital watermarking security to ensure that call recordings, once created, cannot be secretly altered or edited. For each recorded call, CallRex digital watermarking generates a unique signature associated with the data file. Any change made to a call recording disrupts its watermark indicating that tampering has occurred. Authorized users are able to validate the watermark at any time. CallRex digital watermarking provides additional security, enhances audit-trail integrity and raises the standards for legal and regulatory compliance.

“CallRex 3.5 represents a significant advancement in the architecture and capabilities of the CallRex platform,” said Bob Cordes, VP Product Management and Marketing at Telrex. “As call recording and monitoring continues to gain acceptance as a standard business application for dispute resolution, regulatory compliance and training purposes, Telrex continues to set the standard for practical, affordable VoIP call recording.”

For more information about CallRex Professional 3.5 or to request a free, fully functional evaluation, please visit www.telrex.com or call 425-827-6156.

 

February 07, 2007

Solving Asterisk PBX Voice Mail to Email Name Resolution Issues

Note:  Matt Birkland (Network Engineer) is on a roll by sending in another Asterisk Help article about Email Name Resolution on internal networks.
 
One of the most popular features in Asterisk is the Voice Mail to email feature.  It's fairly straight forward to set up as far as Asterisk is concerned, but many people (including myself) have run intoo a situation where the Asterisk server is on the same internal network as the mail server.  When Asterisk resolves somebody@somecompany.com , it attempts to send the mail to the Fully Qualified Domain Name (FQDN).
In the example above this might be something like 216.39.144.X.  This works great if you have hosted email outside your network, or you have internal DNS to send the mail orginating inside the network to the correct host.  However many small to medium size businesses eithe don't have internal DNS or don't know how to set it up correctly.

The result is that the VM attachment is being sent outside the network and can't get back because of a routing loop.  This discussion isn't about routing loops, just trust me that sending out packets out of your network when the destination is on the inside causes problems.  If you want more go get your CCNA.  Moving on...

Asterisk by default uses sendmail to send the VM attachments.  Sendmail will by default resolve from it's name server BEFORE it resolves from the /etc/hosts file.  Luckily, I have a solution that may just work for you.

The first step is to install the sendmail-cf package.  Without this package we cannot reconfigure sendmail.

Example:

[root@localhost ~]# yum install sendmail-cf

Altenativly you can use the add/remove software program in Fedora Core or whatever your distribution of choice is.

The second step is to edit /etc/mail/sendmail.mc and rebuild sendmail.  In the example below I'm using the nano text editor, but any will do.  Note:  I'm only showing the portion of we need to edit.  Before we make any changes lets back up the sendmail.mc file.  Type: 'cp sendmail.mc sendmail.mc.old'.  Great job, we're ready to go.  Follow the example below.

Example:

[root@localhost ~]# nano /etc/mail/sendmail.mc

sendmail.cf:
------------------------------------------------------------------------------------
dnl # Uncomment and edit the following line if your outgoing mail needs to
dnl # be sent out through an external mail server:
dnl #
dnl  # define(`SMART_HOST',`smtp.your.provider')
dnl #
-----------------------------------------------------------------------------------

Now change the SMART_HOST name to something arbitrary like 'mymailserver' and uncomment it.

Example:
---------------------------------------------------------------------------------------
dnl # Uncomment and edit the following line if your outgoing mail needs to
dnl # be sent out through an external mail server:
dnl #
define(`SMART_HOST',`mymailserver')
dnl #

------------------------------------------------------------------------------------------

Now we have to update these changes to sendmail and restart the server.

Example:

[root@localhost mail]# make -C /etc/mail

Followed by a restart of the service.

[root@localhost ~]# service sendmail reload
reloading sendmail:                                        [  OK  ]
reloading sm-client:                                       [  OK  ]
[root@localhost ~]#

Great, if you made it this far you are either a huge nerd or your job is on the line.  Either way we're almost done.  The last step is to make an entry in your /etc/hosts file.  
Example:

/etc/hosts
-------------------------------------------------------------------
# Do not remove the following line, or various programs
# that require network functionality will fail.
127.0.0.1       localhost.localdomain   localhost
192.168.1.18    mattserv.voiceipsolutions.com

#IP Address of mail server below

192.168.1.200         mymailserver
-------------------------------------------------------------------

Thats it!  Call and leave yourself a message.
 
Source: 
Matt Birkland
Network Engineer
Seattle Business PBX

Vocalscape Releases Load Balancer for VoIP Networks

 
 
Vocalscape Networks, Inc. today announced that they have released a Load Balancer for Voice over IP (VoIP) systems.  "Vocalscape has developed the Load Balancer to meet our customers' needs," commented Ron McIntyre, President of Vocalscape.
"As our customers grow their user base, they will need to add additional servers to handle the higher volume of calls. The Vocalscape Load Balancer will allow them to evenly share the load among multiple servers."

The Vocalscape Load Balancer began as an open source project which was adopted and improved upon by Vocalscape. It was made compliant with Asterisk, a popular open source PBX, and the algorithm was revised to more evenly distribute calls.

Previously, the Load Balancer would send calls to a primary server and only when the primary server was overburdened would calls be sent to additional servers. The new algorithm balances the load by evenly distributing the calls between the servers. As an additional benefit, the Load Balancer provides failover capabilities. If a server is not responding, the Load Balancer will route all calls to servers that are functional.

Click Here for More Information about Vocalscape 

Time-Zone Processing with Asterisk - Part I

Last year, I took a trip to Asia. To stay in touch, I carried a GSM world phone, capable of receiving telephone calls in the countries I was visiting. The capability to receive calls with the same mobile phone number I use at home while halfway across the world seemed incredibly cool-at least until the first call came in!
 
Mobile phones hide the location of the phone, which cuts both ways. A colleague had decided to call me in the middle of the day on a Friday, which had awakened me very early on Saturday morning, because the phone "hid"my faraway location from him.
After returning home, I asked several people why my phone company could not simply play a message to warn callers when my time zone changes by more than four or five hours, letting them know the call might be inconvenient. Nobody could come up with a technical reason, but we all suspected it was because the mobile phone company to which I subscribed charged several dollars per minute to connect calls.
 
As part of the process of attaching a GSM phone to a network, the home network needs to learn where the phone is visiting, and that information conceivably could include a time zone. I returned to my idea once I started using Asterisk, because it provides an extensive toolkit for designing PBX-hosted services. Anything that can be coded in a computer can become an Asterisk service.
 
After I understood the basics of Asterisk, I sat down to implement a feature that kept track of the time of day where I visited and prevented calls from coming in at inconvenient times. The system I built on top of Asterisk to handle this feature has two major parts. The key to the system is maintaining a time-zone offset from the time in London.
 
(My code implements offsets only of whole hours, though it could be extended to use either half or quarter hours.) When a device first connects to Asterisk, its IP address is used to guess the location and, therefore, the time offset. After the offset is programmed into the system, incoming calls are then checked against the time at the remote location. Before the phone is allowed to ring, the time at the remote location is checked, and callers can be warned if they are trying to complete a call at an inconvenient time.
 
 

Toss your PBX: Why Asterisk may be the VoIP future of your network

Note:  ComputerWorld has this nice write about Asterisk and what its benefits and limitations currently are.  What I really like about this article is the only real limitations they mentioned was that some companies are leery of open source software and the fact that the Asterisk integrator in the article did not has enterprise level experience.  Personally I know of a couple companies that do these types of installation.  It is true it is not for the faint of heart but if you have been working with Asterisk since 2004-05 you should be fine.
 
"Here's your network's dirty little secret: Your private branch exchange (PBX) is old and outdated, and if you want to bring it into the modern era with IP telephony and voice over IP (VoIP), you're going to have to spend a bundle. Specialized switches and hardware and proprietary systems don't come cheap, and they might not even offer all the telephony features you're looking for." 

But there is an alternative, as thousands of businesses and network administrators have discovered. The open-source Asterisk PBX has been gaining a big following, offering surprisingly powerful telephony features on inexpensive hardware. Not only has it been saving companies money, but it has also been able to integrate telephony with network applications in ways that previously might not have been possible.

But Asterisk isn't for everyone. And there are issues you need to confront if you plan a move to Asterisk. So here, in a nutshell, is what you need to know about Asterisk, along with advice from those who have already deployed it.

What is Asterisk?

Let's start with the basics: Asterisk is open-source PBX software that runs on a wide variety of operating systems, including Windows, Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris. It can run on inexpensive, off-the-shelf hardware, and it includes high-end features such as interactive voice response, voice mail, conference calling, and automatic call distribution and routing that have until now only been available on proprietary PBXs.

It's also exceedingly flexible. New functions can be created by writing scripts in Asterisk's language, by writing modules in C, and by writing scripts in Perl or other languages.

Particularly important is that it handles VoIP calls and works with a variety of VoIP protocols, including the Session Initiation Protocol (SIP) and H.323. It also functions as a gateway between IP phones and the public switched telephone network.

All this means that it can be used to create powerful, programmable PBXs at a low cost, says Joshua Stephens, CEO of Switchvox, a San Diego-based integrator and provider of PBX systems, including many built using Asterisk.

Click Here to Continue Reading 

February 06, 2007

Jabber XCP 5.2 Now Available

Jabber, Inc. today announced the general availability of the 5.2 release of the Jabber Extensible Communications Platform (Jabber XCP).
The highly programmable, scalable, and secure presence platform enhanced its industry leadership position with the addition of:
  • A Session Initiation Protocol (SIP)/SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) Gateway delivered on a new code base providing for transparent federation with SIP/SIMPLE-based messaging systems such as IBM Lotus Sametime, Microsoft Live Communications Server, and the AOL Instant Messenger service. A cornerstone of the company’s multi-protocol approach to real-time messaging, the new SIP/SIMPLE implementation is now available on the Microsoft Windows Server platform in addition to Solaris and Linux.
  • Sun Solaris 10 support, the platform on which recent loads tests (http://www.jabber.com/index.cgi?CONTENT_ID=1080) confirmed Jabber XCP’s ability to scale past a million concurrent users. Jabber XCP is also available for the Solaris 9, Red Hat Enterprise Linux 3.0 and 4.0, Microsoft Windows Server 2000 and 2003 platforms.
  • Increased security integrity of Jabber XCP through obscured password storage in encrypted files.
  • Stanza optimization, through the implementation of XMPP Extension Protocol 0033 (XEP-0033), which provides a method for both clients and servers to send a single stanza and have it delivered to multiple recipients. The benefit is reduced network traffic in situations such as multi-user chat and load balancing among clustered Jabber XCP routers.
  • Auto-include Special Interest Groups (SIGs)/Schemas, which simplifies the job of developers by enabling new code to run on Jabber XCP without editing configuration files.

“Jabber XCP continues to lead with the world’s most scalable, programmable, and interoperable messaging and presence platform,” said Dave Uhlir, vice president of marketing and product management at Jabber, Inc. “Today’s release is a significant milestone towards delivering a truly multi-protocol platform that can surpass a million concurrent SIP, XMPP, and/or IMPS users on modest hardware infrastructure.”

"Most people are aware of presence information because of instant messaging. However, over the next several years, presence information will be used to identify when different types of resources including people, conference rooms, vehicles, deliveries, inventories, etc. are available, now or at an estimated time in the future," said Mark Levitt, vice president for collaborative computing and the enterprise workplace at IDC. "In preparing for this future presence-aware real-time work environment, organizations must look for a presence engine that has enterprise class reliability, scalability, security, and extensibility."

Click Here for More Information

 

BorderWare Releases New Class of Enterprise VoIP Security

Note:  I came across this product today.  I like how they explain their approach using a self-proclaimed "non-proprietary solution" and open source operating system.  It seems the more we look at true security on our networks, the only way I can see us truly securing hardware is using a open source model so we can have alot of eyeballs examining the code and releasing patches faster so it stays secure.  People need to realize that securing a network is a goal and moving target.  The second you rest on your laurels and think you are secure thats the moment your most vulnerable because you have taken your eye off the ball.  Anyways here is the release we found:

"BorderWare Technologies has announced what it is calling a new approach to securing and servicing VoIP that is cost effective and highly scalable to meet the needs of today's service provider and enterprise markets. As VoIP security evolves, service providers and enterprises are realizing the need to secure their converged applications including VoIP, Video and other real-time applications, to protect from attacks such as toll fraud, service disruptions and Spam over Internet Telephony (SPIT)." 

BorderWare's new SIPassure, SIP Security Gateway, is an evolution of the traditional SBC (session border controller), that leverages standard platforms to deliver superior security, flexible deployment options and seamless integration - all at a lower cost of ownership than current proprietary solutions on the market.

SIPassure is a software based, non-proprietary solution, combining features of an SBC with the security of a traditional firewall and the application awareness of an Application Layer Gateway (ALG). SIPassure is architected to run on standard operating systems such as Solaris and Linux, enabling enterprises and service providers to deploy using standard off-the-shelf hardware from vendors including Sun and HP instead of the proprietary purpose built platforms used by traditional SBC vendors.

"Our new approach to securing VoIP solves today's business enablement challenges without re-inventing the wheel," said Tim Leisman, CEO of BorderWare. "The software-based and flexible nature of SIPassure has driven the OEM market to rapidly adopt and integrate it as part of their platforms, providing further proof that adding security to the mix shouldn't require companies to implement an entirely new architecture and solution."

BorderWare SIPassure has key partnerships with Sun Microsystems and Radware to provide high performance and a resilient architecture to meet the most demanding needs of service providers and enterprises.

Click Here for More Information

Source:  EChannelline 

February 05, 2007

Asterisk: Dealing with IRQ's on T1/PRI Lines

The one constant in every Asterisk Integrators life is the pain of dealing with telco's.  The main problem with telco's is the disparity between the tech support. 
 
Sometimes you open a ticket with the guy that can detect an issue in hours.  Usually it takes days of bitching and finger pointing to actually get someone on site.  Many times a Telco insist their PRI/T1's are funtioning, but your Asterisk server is dropping frames.  As a consequence, popping, cracking, lost calls, static and echo occur.  This is a really bad place to be, if you spent hours trying to convince your boss about the merits of Asterisk.  
Then after weeks of argument the telco's admit some kind of 'crosstalk' on the line and they dispatched a Qwest technician to fix it.  This happens so often that I don't even get mad anymore, it's like a bad joke affecting an entire Industry.  However...  as I learned recently, sometimes symptoms that look like a telco issue are actually IRQ issues.


Most Asterisk Integrators are aware that IRQ issues are huge problem and take steps to secure their Digium 110P (T1) or TDM40B (analog lines).  However on most modern OS's (like Linux) IRQ's are handled logically by the kernel through APIC (Advanced Programmable Interrupt Controller).  The great thing about APIC is that you can have multiple interrupt controllers to deliver all the IRQ's necessary for running todays boundless devices.  The downside is that the output from 'cat /proc/interrupts' is not always the same as scanning the PCI bus with the 'lspci -bv' command.  So while it may seem like your devices are on separate IRQ's, in some cases they are not.  Lets look at the example below.

[root@localhost asterisk]# cat /proc/interrupts
          CPU0
 0:     44302787        IO-APIC-edge  timer
 7:               0            IO-APIC-edge  parport0
 8:              1             IO-APIC-edge  rtc
 9:              1             IO-APIC-level  acpi
14:     113862           IO-APIC-edge  libata
15:      1588284        IO-APIC-edge  ide1
50:              0            IO-APIC-level  uhci_hcd:usb3
58:      1904417         PCI-MSI  eth0
169:              0           IO-APIC-level  uhci_hcd:usb4
225:  177207785       IO-APIC-level  uhci_hcd:usb2, wcfxo
233:              0           IO-APIC-level  uhci_hcd:usb1, ehci_hcd:usb5
NMI:              1
LOC:   44304635
ERR:              0
MIS:              0


Seems pretty straight forward; this is how I have been checking IRQ's for years.  Now take a look from lspci command below.  
Note:  This is not the full output.  I only cut and paste the Digium card and the PCI devices it shares.


[root@localhost asterisk]# lspci -bv

0a:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
       Subsystem: Unknown device 8085:0003
       Flags: bus master, medium devsel, latency 32, IRQ 10
       I/O ports at 6000
       Memory at d8300000 (32-bit, non-prefetchable)
       Capabilities: [40] Power Management version 2

0a:03.0 VGA compatible controller: ATI Technologies Inc ES1000 (rev 01) (prog-if 00 [VGA])
       Subsystem: Giga-byte Technology Unknown device 515e
       Flags: bus master, stepping, medium devsel, latency 66, IRQ 10
       Memory at d0000000 (32-bit, prefetchable)
       I/O ports at 6400
       Memory at d8310000 (32-bit, non-prefetchable)
       Capabilities: [50] Power Management version 2

05:00.0 Ethernet controller: Intel Corporation 82573V Gigabit Ethernet Controller (Copper) (rev 03)
       Subsystem: Giga-byte Technology Unknown device 108b
       Flags: bus master, fast devsel, latency 0, IRQ 10
       Memory at d8200000 (32-bit, non-prefetchable)
       I/O ports at 5000
       Capabilities: [c8] Power Management version 2
       Capabilities: [d0] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable-
       Capabilities: [e0] Express Endpoint IRQ 0

00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01)
       Subsystem: Giga-byte Technology Unknown device 27da
       Flags: medium devsel, IRQ 10
       I/O ports at 3080



In short, if you don't have an IRQ reserved at the BIOS level remember to scan your PCI bus with the 'lspci -bv' command.  With this many devices on the same IRQ all sorts of errors and strange behavior is possible.  If the IRQ is not saved for that specific device the best course of action to switch the  PCI slot your Digium card resides in.


Matt Birkland
Network Engineer
Seattle Business Phone Systems

AsteriskNow Opens a "Can of Whoop Ass" on Telecom World

Note: Steve Burke of CRN posted this story about AsteriskNow.  Personally I have been playing with AsteriskNow for our office system.  I wasn't too hard to setup once you knew how the installer worked.   I hope there will be a  Asterisk Now "How-To" coming out soon.  Here is some of Steve's article:

"Solution providers searching "Mark Spencer" on YouTube will find not only a bizarre clip about a homemade cake and humorous downloads regarding British department store Marks & Spencer, but a sure-fire way to increase sales in the lucrative IP PBX and VoIP market courtesy of Asterisk open-source PBX creator Mark Spencer."

Spencer, founder, chairman and CTO of Asterisk provider Digium, Huntsville, Ala., is a onetime Linux service provider who has taken to YouTube to promote AsteriskNow, a new GUI-powered version of the popular open-source PBX aimed at attracting a new wave of solution providers.
 
In the 4-minute, 13-second clip, Spencer takes viewers through the quick and easy AsteriskNow install. That's no small matter. What the new single-disk GUI install does is dramatically lower the cost of entry for solution providers looking to get into the open-source telecom game. Before AsteriskNow, solution providers would have to add some high-priced, open-source technical talent with Asterisk expertise to their staff at a cost of $100,000 to $150,000, Spencer said.
 
Now solution providers can get their feet wet in a couple of minutes and begin using the software in SMB solutions without a deep technical dive. For Spencer, the new software marks a stepped-up march of his vision to bring the open-source PBX and VoIP system to the masses. He said 2006 was all about making progress with VARs and customers in the early adopter stage. "2007 is going to be where we start making inroads into the early majority," he said. "We've now got a way to get VARs into this much faster. You don't have to be afraid of having to load Linux or Asterisk or anything like that. We're delivering that with all the power of open source behind it."
 
 

TECORE Unveils SIP/VoIP Enabled GSM Base Station

TECORE Wireless Systems, announced the launch of its pre-IMS VoIP/SIP enabled GSM base station to be showcased at this year's 3GSM World Congress. The BTS-4000RM delivers up to 64 high powered GSM/GPRS/EDGE voice and data channels occupying as little as 5U of a standard 19” telecommunications rack. The BTS-4000RM base station eliminates the need for traditional BSC, TRAU and PCU elements, embedding these functions within the base station platform.

The BTS-4000RM may be integrated directly with TECORE’s SoftMSC as well as other mobile core network solutions over IP, TDM, or IP over TDM. Moreover, the BTS-4000RM is enabled to support TECORE’s exclusive
 
AirSite Backhaul Free base stations, allowing operators to deploy ‘plug and play’ coverage extensions without the need for wired backhaul links. Due to its small size and flexible interface options, the BTS-4000RM is also ideally suited for remote towns and villages that may require satellite backhaul as well as other applications where macro RF coverage and performance is needed.

With the launch of the BTS-4000RM, TECORE also introduces network interface enhancements to its GSM base station product line. This includes the internal translation from a traditional GSM RAN to an all IP SIP-enabled network interface.

Thus, the BTS-4000RM supports direct connection to a pre-IMS network. Added capabilities include transcoder-free operation to minimize network bandwidth, and direct routing of local IP call traffic to optimize overall backhaul utilization. Specifically targeting applications requiring compact radio access solutions, the BTS-4000RM offers flexible network integration and deployment capabilities while providing macro RF coverage, capacity and performance.

“The operators to whom we’ve delivered and demonstrated the BTS-4000RM are very excited about the flexibility this solution offers,” said Terry Williams, Chief Technology Officer for base station products. “Large capacity, flexible interfaces, and more functionality in a smaller, easy to deploy platform are key to many specialized mission critical applications, and this platform delivers.”

“We are very pleased with customer response to this new product development and technology innovation,” said Doss McComas, Vice President, Business Development at TECORE. “This product builds upon our company’s ongoing commitment to advanced, smaller, more powerful base station solutions for the market.”

The BTS-4000RM will be on display at the 3GSM World Congress 2007 in Barcelona, Spain, February 12-15, 2007, at the TECORE Wireless Systems pavilion located in Hall 8, stand 8C78.

Kensington Vo200 Bluetooth Internet Phone = Small Form Factor

 
 
Kensington Has released their new Vo200 Bluetooth VoIP phone.  According to their website it fits into your pcima slot on your laptop and actually recharges via that slot (Very nice, I hate extra cords).  It has built-in support for the following VoIP service:  Skype, GoogleTalk, MSN and Yahoo!.

Comes with enough battery time for 3 hours of continuous talk time and over 30 hours of standby.  It retails for about $90.00 USD.  MobileMag has a review of the phone itself.

Click Here for more Information 

February 04, 2007

ITU set to make WiMAX a 3G standard

Note:  For wireless broadband this is a good move.  It would be great to have a few providers or atleast equipment that a person could take with them and have broadband access on their travels.
 
There is a good chance that the International Telecommunication Union (ITU) will allow mobile WiMAX to be included in its range of 3G technologies, collectively known as IMT-2000.  Such a move would bring considerable benefits by allowing WiMAX to operate in globally allocated frequency bands, allow it to be used to complement other 3G technologies, enable global roaming and reduce equipment costs.
According to the WiMAX Forum, in a new White Paper "WiMAX and IMT-2000" "including IP-OFDMA (mobile WiMAX) within the IMT-2000 family of radio transmission technologies will put mobile WiMAX on a comparable worldwide footing with EV-DO, HSPA, and other recent and planned enhancements to 3G technology. This will offer operators an additional migration path to consider as they strive to add network capabilities to support a larger suite of value-added broadband services."

The White paper explains: "Supporting IP-OFDMA in the IMT-2000 family enables significant flexibility in network deployment options and service offerings. For example, service providers can provide access in low density environments or enhance access capacity in metropolitan and suburban areas to support value-added services.
 
The inclusion of this additional radio interface would provide added value by giving service providers additional flexibility in selecting an IMT-2000 technology for a more optimal fit to their business model, and a more logical stepping stone to the delivery of Next Generation Mobile Internet services...The WiMAX Forum Global Roaming Working Group expects to announce a specification and business model to assure 'roaming' within WiMAX as well as other networks designated in IMT-2000. Service providers also benefit from open global standards that foster vendor interoperability and lower equipment costs."

WiMAX Day, a newsletter published by the WiMAX Spectrum Owners Alliance, reported that, ITU Working Party 8F met in Cameroon at the end of January to look at IMT-2000 and systems beyond IMT-2000
 

February 03, 2007

Explosive Growth in WiMax Patent Activity Promises Surge in Industry

A new report from WTRS finds dramatic growth surge in WiMax patent activity. Completing intensive study of almost 500 WiMax related patents, WTRS now predicts phenomenal increase in WiMax industry, but IP litigation activities will affect market leaders.

WTRS announces 2007 edition of its unique WiMax Patent Directory. This Report demonstrates the enormous growth in innovative WiMax activity; last year's Directory covered about 50 new patents and this year's new Report studies over 475!

The Report offers analysis of current and pending WiMAX-related patent litigation, and analysis of IP and Patents with registered IEEE Letters of Assurance.

Patents are tracked for mobile and fixed WiMAX and related technologies such as WiBRO. Both US and worldwide patents are evaluated by company with a focus on fundamental network architecture, enabling software, and RF chipsets. Detail is extensive and the 500 page Report is both comprehensive and accessible, organized with an easy-to-use interface; patents to claims are hyperlinked within the document.

"The Wireless Triple Play industry will be dominated by WiMax in both fixed and mobile forms," according to Principal Analyst Kirsten West. "We are starting to see examples of WiMax used to solve actual business pain and that is the prerequisite for broad adoption in the Wireless Triple Play market."

In the past this Report has been purchased by OEM's, but also by a diverse Group, including a law firm and even a University Department. A previous version was even purchased by another Market Research firm! This is a unique publication, without a comparable offering from any other firm, but also is a tool allowing purchasers to identify those individuals making significant contributions to the WiMax universe.

Using only publicly available information, meticulously gathered from public patent sources, company information, and other research this report is truly an invaluable resource for any company participating in the WiMAX sector, or planning a potential entry into this market.

Source: WiMaxxed

TrustEli Annouces WiMax and Cellular Router with Managed Security Interface

TrustELI Announces Cellular and WiMAX Router Providing a Managed Security Interface. Eli Everywhere combines enterprise grade Internet security with an Internet access router to allow secure connections of any type, including cellular, WiFi, WiMAX and FiOS.
Cities around the world are already delivering Internet access to the masses with WiMAX, while cellular service is beginning to reach remote areas not currently served by traditional broadband connections. Eli Everywhere can secure these growing alternative connection types with its new, onboard PCMCIA slot. The slot allows for the insertion of any type of PCMCIA access card, including cellular, WiFi, WiMAX, or even a traditional modem that can be used as an Internet failover connection.

"Eli Everywhere is the Swiss Army Knife of Internet routers, offering a secure connection to the Internet no matter what type of connection a user may have," said Robert Smith, Chief Technology Officer of TrustELI. "With next-generation access becoming a reality, users need Eli Everywhere to be assured that their security is maintained no matter how they connect to the Internet."

Eli Everywhere also solves the limitation of many next-generation access types to connect to only a single PC. With Eli Everywhere, up to 25 PCs can share a single access to the Internet. This allows remote or new offices to access secure Internet connections immediately without the delay of access line installation. As an additional option, Eli Everywhere can be configured to provide failover for a traditional Internet connection (such as a DSL) to a next generation technology, allowing always-on Internet connectivity for mission critical locations.

Eli Everywhere will be available everywhere Eli is sold in Q3. For more information, visit: TrustELI.

Vonage Nearly at 100% with E911 Service

Vonage America Inc. announced this week that now more than 94% of their US subscriber lines are equipped with Enhanced 911 service, a feature that automatically brings up a physical address with the associated calling party’s telephone number.

“Vonage’s nomadic E911 solution gives customers the ability to reach a Public Safety Answering Point (PSAP) or 911 centers, through the dedicated 911 network infrastructure. With Vonage’s nomadic E911 solution, a customer’s call is automatically routed to the appropriate 911 center, with the caller’s registered street address and telephone number appearing on the dispatchers screen — regardless of where or what exchange they are calling from. Vonage will continue to turn up and test new PSAPs that are VoIP-ready every day.”

Today more than 2 million of our U.S. subscriber lines have full E911 capability, which is a tremendous step for Vonage,” said Vonage CEO Mike Snyder. “Vonage will continue to work with the FCC, regulators, Congress and public safety officials until PSAPs across the nation are equipped with E911.”

Source: VoIP-News 

The Ultimate WiFi Base Station Covers 1km Range

Note:  Found this piece at MobileMag covering this new high-end wireless access point that has a pretty impressive range. 
 
I'd imagine ideas like this would be great for cities like San Francisco that are getting city-wide WiFi coverage, because the Etri access point boasts a range of up to 1 kilometer (that's about 3/5 of a mile). Keep in mind that this is a range, meaning that the radius is 1 kilometer, and -- if my high school math is correct -- the effective coverage area is the neighbourhood of 3.14 square kilometers, or about 2 square miles."
That said, I'm sure this higher end of their claim (Etri is saying that the access point gives a range of between 100m and 1km) is based on having no interference whatsoever in the form of concrete buildings, mountains, valleys, and so forth. The other wireless product that they revealed earlier today is a router offering speeds of up to 240Mbps (MIMO-OFDM).
 
 
Source:  MobileMag 

February 02, 2007

Cisco Boosts Catalyst Switch Portfolio

Cisco has recently expanded its catalyst switch portfolio that offers twice the Power over Ethernet (PoE) support as its earlier offerings and also come bundled with expanded 10 Gigabit Ethernet uplink options.
According to the company, these new offerings are expected to offer highly secure wired and wireless user access. Cisco Catalyst 3750-E and 3560-E Series switches are in its enterprise-class line of 10/100/1000 wiring closet switches with 10 Gigabit Ethernet uplinks and full PoE configurations.
 
These switches have been designed to facilitate the deployment of highly secure converged applications. Featuring StackWise Plus, the Catalyst 3750-E Series is backward-compatible with StackWise and can be integrated with Catalyst 3750 Series Switches, the Cisco Catalyst 3750 Integrated Wireless LAN Controller, and Cisco Integrated services routers.
 
Cisco's catalyst 3560 and catalyst 2960 compact switches are small form-factor switches with Gigabit Ethernet and Fast Ethernet and PoE configurations designed for meeting rooms, and other wiring and space-constrained environments. Its Catalyst 4500 series full image In-Service Software Upgrade (ISSU) apparently helps in ensuring business continuity by enabling a full Cisco IOS Software upgrade without downtime.
 
Meanwhile, its Redundant Power System (RPS) 2300 offer users uninterrupted network services in the event of an internal power-supply failure for Cisco PoE switches. The company has also boosted its power supplies on its Catalyst 6500 series chassis to power as many as 420 ports in a chassis with PoE.
 
Source: CXO Today 

Moto Promises WiMax Phones in 2008

Unstrung reports that Motorola Inc. is teaming with chipmaker Texas Instruments Inc. to develop its first mobile WiMax handsets, to launch in 2008. TI already works with Motorola developing chipsets for 3G devices. The firms say that the extension of this deal will focus on "802.16e mobile WiMAX functionality supporting voice, video and data for low-power mobile applications." TI is readying WiMax chipsets for "mobile devices that Motorola plans to launch during 2008."

Bringing WiMax handsets to market in a timely fashion is key for Motorola, which is one of the three major infrastructure suppliers named by Sprint Nextel Corp. for its forthcoming mobile WiMax network in the U.S. The Reston, Va.-based operator is working with Motorola to launch a 1,000-site mobile WiMax deployment in Chicago by the end of this year; a nationwide rollout is planned in 2008.

Motorola is clearly intending to have suitable handsets available as soon as possible -- even if the company doesn't want to give away too much detail at the moment. "I'd rather not pre-announce any products," a spokesman told Unstrung in response to questions. "Motorola will announce details about specific products either through news releases or launch events."

The Schaumburg, Ill.-based company will face competition in the race to be the first with commercial handsets using the new technology from Sprint's two other major infrastructure suppliers for the WiMax network -- Nokia Corp. and Samsung Electronics Co. Ltd. -- all of which are tasked with supplying gadgets as well as radio gear.

Source:  Wimaxxed 

February 01, 2007

GigaOm and Forbes: Information Super Traffic Jam??? (Must Read)

Note:  As I was browsing Om Malik's Blog and I came across this piece which I thought was very entertaining piece of mis-information on the behalf of Forbes.com.  Now I am not saying that this is not a complex problem that has multiple solutions.  What I am saying is that EVERYONE of these broadband providers get this thing called "monthly access fees" from their customers. 
 
So I came up with idea that I wanted to share with these Large Corporations.  Maybe you could take some of the billions of dollars you are getting from your "customers' and spend it on upgrading your network so you continue to be competitive in the coming years.  Now back to the Forbes article, Philip Kerpen says that the networks need to not have net neutrality so they can get premium fees to justify upgrading their network?!??!??  Well it looks like Verizon's FIOS is going to take the cake because the first person that brings Fiber to my home is going to get these so-called "premium fees" from this blogger. 

Here is a snippet from the Forbes.com article:

 "

Robert Kahn and David Farber, the technologists known respectively as the father and grandfather of the Internet, have both been highly critical of network neutrality mandates. In a recent speech, Kahn pointed out that to incentivize innovation, network operators must be allowed to develop new technologies within their own networks first, something that network neutrality mandates could prevent. Farber has urged Congress not to enact network neutrality mandates that would prevent significant improvements to the Internet.

Without enormous new investments to upgrade the Internet's infrastructure, download speeds could crawl to a standstill. It would be unfortunate if network neutrality proponents successfully saved the rapidly aging, straining Internet by freezing out the technological innovations and infrastructure investments that would enable next generation technologies to be developed and deployed.

The video-heavy, much vaunted Web 2.0 advances of the last couple of years were made possible at low prices to consumers because the speculative overbuilding during the bubble era created massive overcapacity that made bandwidth cheap and abundant. It's now all being consumed.

One solution suggested by network operators is to prioritize traffic based on service tiers and use revenue from content providers in the premium tiers to subsidize the high costs of infrastructure deployment. The MoveOn.org crowd denounces this solution for creating Internet fast lanes and relegating everything else to the slow lane. But as the Deloitte report shows, the likely alternative is that there will be only slow lanes, potentially very slow lanes as soon as later this year. Call it the information super traffic jam.

Advanced networks cost billions of dollars to deploy and need to generate predictable revenue to make business sense. The infrastructure companies are unanimous in their belief that offering premium services with guaranteed bandwidth will be necessary for them to justify their investments. Quality-of-service issues alone are likely to require tiering, because in a world of finite bandwidth, people won't want high-value services like video and voice if they can be degraded by the peer-to-peer applications of teenage neighbors.

"

Picture of Phil Kerpen:


 

Click Here to Read Om Malik's  and Paul K Take on Super Traffic Jam

Click Here to Read the Original Forbes.com Article 

Skype Wars Episode III : The revenge of the SIP


Note:  This is insightful.  I guess if you want to register your Name and Mark you need to enforce it.  If you let people use your name without getting permission from you and instructions you can lose your claim with the trademark agency.

"Few days ago I attended a conference on Google adsense (by Google) in Kuala Lumpur, Malaysia. I asked why I could not startup my a Google adsense campaign with the word of skype in it. The answer is that that is not allowed by Skype and Google due to copyright and trademark protection of the Skype brand." 

That is why you see mostly other VoIP brands and other website popping up on many blog where the word VoIP and hence Skype (on many SKype blog are mentioned)… Therefore the more I write about the Skype / VoIP thing the more click will go to the competition. What a joke… 

I am not even going to mention the fact that on a website which contains the word Skype in the domain-name you are not allowed to do any commercial activity that Skype does not endorse. Something that was decided by the Commission Junction that manages the Skype affiliation project it seems. This is also why you will not see any phones of Skype being sold on this website and also not the easy Ebay-store plugin. As far as I understand it is simply not allowed… So far the usefulness of the easy Typepad Ebay plugin store… Lot’s of inconsistencies in this whole concept if you ask me. Remember, I am not a Skype affiliate. Not anymore. www.cj.com disabled that… some months ago.

Back to the google adsense story… Yet another way of protectionism, proprietary obfuscationism that leads to nothing but more advertising and visibility for the others… Just look at this blog. I have put some google adsense on it. It would be good for the skype blog and skype store to see something else that SIP / VOIP in the google adsense. Why does Skype / Ebay Corporation not understand that ? Or is it Google with the Google talk product who prefer not too much early global advertiser for their competitor… Talking about net neutrality…

The same goes for the open p2p Skype cloud. I can’t tell you how many I get calls from people that are trying to sell me SIP/Asterisk or other PBX solution… Talking about being cannibalized by others… Maybe it’s the revenge of SIP.

There is one that slipped through the mazes of the net though..  See Image below for the Skyper IM…

Click Here to Continue Reading:  Skype Wars Episode III : The revenge of the SIP

Seagate puts 20GB of wireless storage in your pocket

Note: Can't wait  I like the fact of wireless devices making it so we have less cords.  I hope they setup some good security features so someone can't come by and hack my drive and take my important information. 
 
Seagate has formally announced its Digital Audio Video Experience (D.A.V.E) technology – the project previously code-named "Crickett" – at the DEMO 07 Conference in Palm Desert, California. The DAVE platform offers 10-20 GB of wireless storage in an accessory smaller than many common slim-line mobile phones. It is designed to store, play and share digital files on mobile phones, PCs and other wireless-enabled devices.
Using Bluetooth or WiFi wireless connections with a nine metre range, DAVE is the size of a centimetre-thick credit card and weighs in at 70gm. It is designed to live in your shirt pocket, backpack, or purse. Utilising Seagate's Storage Management Module power-saving technology unveiled at CES, the rechargeable lithium ion battery delivers up to 10 hours of media-streaming and up to 14 days of standby power.

The mobile storage platform is open source, enabling third party software developers to create new applications for the mobile phone utilising the hard drive. It is built on a set of open API platform accessible to developers using current development tools. A DAVE Software Developers Kit will be available in March 2007.

Seagate is expected to sell DAVE through phone vendors and telcos, and an add-on accessory, rather than as a Seagate-branded stand alone device.
 
Source: IT Wire 
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