Note: I have to say it.   "This one time in Asterisk Camp".  Anyways, grats to Jared and the Sokol crew for helping bring this together.  This is exactly what any emerging technology needs if it is to really take hold and go main stream.  Heck if I had an extra $3500.00 I would sign-up right now. 

Big Nerd Ranch, Inc. announced today the newest addition to its line-up of classes, Asterisk Bootcamp, November 6-10, 2006. Asterisk Bootcamp, taught by instructor Jared Smith and developed by Sokol & Associates, highlights one of the most talked about emerging technologies in the telephony and programming industries. Asterisk is the perfect solution for a spectrum of telephony needs from voice over IP (VoIP), teleconferencing and switchboard functionality.

As the global marketplace demands greater flexibility for meetings, conferencing, and sophisticated voice systems, Asterisk, the open source contender, provides a flexible, economical and scaleable design environment ideal for cutting-edge business, military and governmental powerhouses. This bootcamp is ideal for developers, Linux/Unix consultants, interconnect vendors and ITSP system administrators looking for an enhanced and more elegant PBX solution that rises above its more pedestrian competitors.

The innovative bootcamp highlights the instruction prowess of Asterisk guru and author, Jared Smith. Jared is the Senior Consultant and instructor for Sokol & Associates and the creator of the IAX2 trunking technique. He is featured as a co-author of the seminal Asterisk text, Asterisk: The Future of Telephony, and one of the co-creators of the Asterisk Docs project.

"When you're dealing with emerging technologies, finding the right instructor with the requisite experience and street credentials to teach a class can be challenging," said Aaron Hillegass, Big Nerd Ranch founder. "Having someone with the knowledge base, reputation, and vigor of Jared Smith is like winning the lottery. His finger has been at the pulse of Asterisk design almost since its inception, and his ability to translate his knowledge into practical instruction for students is an incredible asset."

The Asterisk Bootcamp is a five day course which covers in-depth the keys points of Asterisk installation, configuration and administration. The course begins with an in-depth review of Asterisk as an application, a project and a community. It goes on to teach the student to download, compile, install and tune Asterisk; to connect it with both end-user devices (phones) and the outside world. Students will learn to create dialplans, to implement applications, and to make use of the long list of features which are included with Asterisk. At the end of the course, students should be able to create a working Asterisk system from a standard Linux computer, configure the system to support end users connected via multiple technologies, and to handle all necessary adds, moves and changes. If you are wanting to learn how to implement a working Asterisk system, this class is for you.

The bootcamp assumes previous experience with programming; especially with Linux.

Read more about Asterisk Bootcamp (including the complete syllabus) or our instructor Jared Smith.

The Big Nerd Ranch incorporates intensive training classes for Unix and Mac OS X programmers in a retreat setting outside Atlanta, GA. Class price of $3500 includes lodging, all meals, original instruction materials, 24-hour lab access, and transportation to and from the Atlanta airport. Students are encouraged to bring independent projects to class, allowing for input from classmates and individual instructor attention. For more information, call 678-595-6773 or visit:

www.bignerdranch.com

Posted by Dal at 11:18 AM | Direct Link | Comments (0) Posted to Asterisk Help

Note: Dear Motorola,  Please release some cheap commerical WiMAX access points along with notebook PCIMA cards so I can setup a high BW node at my house.  Also please release Linux and OSX drivers for the card also.  Thank You.  I have been following the WiMAX movement lately.  I am lucky enough to be in range of the roll out of Craig McCaw's Clearwire coverage in Seattle.   Sooner I can cut the cord the better.

Motorola has announced a strategic initiative to develop mobile WiMAX chipsets for use in Motorola's next-generation WiMAX devices.  This investment in chipset design demonstrates Motorola's on-going commitment to WiMAX and expands the company's role as a leader in end-to-end WiMAX solutions beyond infrastructure and handsets to the core implementation of WiMAX in chipsets.

Ron Garriques, President of Motorola's Mobile Devices Business, said, "For some time now Motorola has been investing in the development and deployment of WiMAX -- from infrastructure to advanced silicon. In our vision of seamless mobility, WiMAX will help transform the mobile communications experience for everyone. Enhanced speed for data, as well as cost efficiencies and network optimization for wireless operators are just some of the benefits that are fueling excitement and accelerated adoption of WiMAX across the industry."

Motorola's initial chipset will focus on core 802.16e mobile WiMAX functionality supporting voice, video, and data for low power mobile applications in handsets and modules. These first chipsets are scheduled to support commercial Motorola WiMAX devices in 2008 for carriers in North America, Japan and around the world including Sprint and others. Motorola is working with its silicon vendors on the overall fabrication of the new chipsets.

Source: Motorola 

Posted by Dal at 11:03 AM | Direct Link | Comments (0) Posted to WiMax

"It was a good quarter for carrier next-gen voice, with all segments showing gains except session border controllers (SBCs), which were down a bit for the quarter, but up significantly year-over-year," said Stephane Teral, principal analyst for service provider VoIP at Infonetics Research. "SBCs have become a strategic component of any next-gen voice equipment offering these days, and they are reshuffling the vendor ecosystem. Many vendors will likely shift from a standalone to an integrated SBC architecture, including Cisco, Juniper, and AudioCodes due to its acquisition of Nuera and Netrake."

Infonetics' Enterprise Telephony report shows Cisco had a record quarter in the worldwide PBX/KTS market, which grew 3% sequentially to $2.2 billion (10% higher than a year ago).

"Vendor results in the PBX/KTS market were very mixed this quarter, with a net result of a small sequential gain," said Matthias Machowinski, directing analyst for enterprise telephony at Infonetics Research. "We expected Cisco to do well since it was the end of their fiscal year, but they blew right past our expectations, for the first time breaking 1 million lines per quarter. Nortel had a great quarter, too, while the other major players were more or less flat. As a consequence, worldwide market share positions were shuffled around this quarter, with Nortel shooting to the top for IP PBX line share (up from third last quarter), and Cisco jumping from fifth to second."

Click Here for more Statistics 

Posted by Dal at 10:53 AM | Direct Link | Comments (0) Posted to VoIP News

Note: This will be a major undertaking.  Looks like there are going to a redundant failover solutions and goes to PSTN lines if it detect the IP network is down.  There solution also has "less cost" routing also so that it will direct either through there internal network to reduce long distance charges to zero.  This company must have there deployment together, according to the article they did the first 100+ branches in only 30 days.

The Agricultural Bank of China (ABC), China's largest commercial bank, with revenues exceeding $250 billion, was looking to decrease its long-distance phone expenses between its more than 50,000 domestic and international branches and affiliate offices.

These locations were in all of China's major provinces, cities and counties. ABC's goal was to consolidate multiple regional call centers to one centralized call center through a Voice over IP (VoIP) network.

Of foremost concern to ABC among its many requirements to implementing VoIP was that it had to have minimum impact on the existing IP network and voice system performance. The bank did not want to have to upgrade the existing IP network, reprogram its PBX or purchase any PBX upgrades or interface cards.

Most importantly, it wanted assurances that there would be no degradation to voice quality compared to wire line calls.

The bank required that its voice network be compatible with international standards (i.e., H.323), and also needed to be scalable for nationwide deployment and growth, and interoperable with other vendors. ABC wanted to minimize the impact on its end-users so they did not have to change their dial plans or habits. Overall, ABC required high reliability, even under extreme and unusual circumstances.

Public Information Technology Co. (PiTech), a large Chinese distributor, recommended Quintum's Tenor VoIP multiPath switches. The switches had the technologies that ABC required, the diverse applications that ABC's environment necessitated and the customer support that put ABC at ease.

Click Here for the Full Article 

Posted by Dal at 10:34 AM | Direct Link | Comments (0) Posted to VoIP News

 

"

Hilton was one of more than 50 customers whose accounts were suspended because they had been using SpoofCard.com's Caller ID spoofing service to hack into voicemail accounts, according to Mark Del Bianco, SpoofCard.com's attorney. Many of the accounts that were hacked via the spoofing service belonged to well-known celebrities, including Lohan, he said.

SpoofCard.com has not actually accused Hilton of hacking into Lohan's voicemail. But celebrity gossip sheets, already abuzz with the rivalry between the two divas, have jumped on the story.

The New York Post reported last month that someone had stolen the password to Lohan's BlackBerry and sent her friends "disgusting and very mean messages that everyone thought were coming from Lindsay." Lohan's representatives hinted that Hilton may have been behind the hack, the Post said.

Hilton could not be reached for comment, but her spokesman Elliot Mintz told E! Online that the alleged hacking "just didn't happen," and suggested that someone else may have opened the SpoofCard.com account in Hilton's name.

Both the Cingular Wireless LLC and T-Mobile USA Inc. telephone networks use Caller ID to identify voicemail users without requiring passwords. So users on either network could have been vulnerable to the misuse alleged by SpoofCard.com, said Lance James chief scientist with security vendor Secure Science Corp.

The scandal illustrates how the telephone industry has been affected by inexpensive telephony software, like the open-source Asterisk telephone system. Recently phishers have been using this software to set up inexpensive phone networks that give their fake e-mails an added air of authenticity, for example.

And with less than 10 employees, SpoofCard.com was able to use Asterisk and Linux to create a line of business that would have been far too expensive just ten years ago. The fake Caller ID vendor sells US$10, 60-minute calling cards that let users call a toll-free number and type in whatever Caller ID number they want their call to display."

Click Here for the Full Article 

Posted by Dal at 01:32 PM | Direct Link | Comments (0) Posted to VoIP Security

AT&T has decided to get in on the action, signing an agreement with Springfield, Illinois to develop and deploy a WiFi network that will cover the 25 square mile area around the city center. Residents outside that zone will be able to connect to the network via external antennas. While the network will initially use 802.11g, plans are in place to incorporate WiMAX over the next couple of years as that technology becomes more widely deployed in the US.

Springfield residents will have their choice of free, low-speed access and faster, paid access. As is the case with other municipal networks, the free access will be ad-supported. The faster tier of service will be available on a daily or monthly basis, with pricing undetermined. AT&T will also market the service as an add-on feature for its DSL customers.

Click Here for more Information 

Posted by Dal at 09:15 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

FSS is the clear leader in the SIP stack market with 32% share, according to VDC’s Telecom Core Infrastructure Market Intelligence Service 2005-2006. SIP is an application layer control protocol that can establish, modify and terminate multimedia sessions or calls.

“SIP is a dynamic and rapidly growing software sector where scalability and interoperability are critical,” said Robert Johnson, Senior Telecom Analyst of Venture Development Corporation. “Hardware manufacturers, software vendors and service providers have increasingly high expectations of companies providing SIP protocol stacks and toolkits. FSS’ top ranking is a very strong statement from the market,” continued Mr. Johnson.

“FSS’ products’ proven ability to reduce time-to-market and their superior scalability, interoperability and quality are the top factors our customers cite when they choose our SIP products,” according to Suresh Kabra, Assistant Vice President and Head of Products of Flextronics Software Systems. “We will continue to accelerate the development of our customers’ products and services through FSS products that include SIP-powered IMS offerings, SIP User Agent Toolkits, our SIP Server Framework, our Back-to-Back User Agent and our SIGCOMP stack,” added Mr. Kabra.

FSS protocol stacks, frameworks and toolkits have been critical to the delivery of products from over 200 Original Equipment Manufacturers (OEMs) worldwide. Flextronics Software Systems offers innovative software products for:

-- wireless, wireline, broadband and satellite infrastructures;
-- carrier-grade and enterprise class communications equipment; and
-- devices, server-side communications applications and the network core.

Source: BWI 

Posted by Dal at 09:01 AM | Direct Link | Comments (0) Posted to SIP

Editor's Note:  Thank you very much for this mention on your list.  It is very humbling to make it with such a great crowd. Here is this list to see if your favorite writers or geek blogger made it.  I am glad I didn't have to choose those top 10.

Smith On VoIP's Top 30 VoIP Blogs  

Posted by Dal at 01:26 PM | Direct Link | Comments (0) Posted to AVN Site Updates

Brekeke Software, Inc., an emerging provider in voice and data communications technology, today announced a partnership with Solegy LLC, creators of a managed platform that allows network operators, service providers, and content developers to achieve their business objectives more effectively. 

Brekeke Software offers SIP-based IP-PBX and SIP Server software for creating VoIP telephony solutions. Managed through a web-based administration tool, Brekeke PBX is scalable and flexible to meet basic to advanced telephony configurations.

Solegy's ServicePDQ platform is a multi-service software system that offers a single point from which to develop, deploy, manage and bill for network services across heterogeneous networks. At the center of ServicePDQ is the Real Time Billing Engine (RTBE), an efficient and customizable web-based IP billing solution which helps service providers to manage all their customer accounts and billing information. ServicePDQ is a high performance, fault-tolerant software platform for developing and deploying revenue-generating network services.

According to Stu Slepping, president of Solegy, "Brekeke software is everything it should be. We have explored many IP PBX vendors in our search to find a product that fits the needs of Solegy customers wanting an on-premises solution. With its many features, ease of use, and super technical support, selecting Brekeke PBX was an easy decision."

"Brekeke's IP-PBX features are perfect for combining with Solegy's ServicePDQ platform," said Shin Yamade, CEO of Brekeke Software, Inc. "Integrating Brekeke and Solegy products to develop next generation VoIP services like Billing Servers can give businesses an advantage over competitors."

Posted by Dal at 10:45 AM | Direct Link | Comments (0) Posted to VoIP News

The same phone works outside the home as a normal mobile phone that automatically seeks out a mobile network. Users will need to acquire a special telephone to use the Unlicensed Mobile Access (UMA) technology, which will be operational in November. In the future, the technology "could also be used an IP phone if the user is in a Wi-Fi hotspot outdoors, such as an airport, cafe, or conference centre for example.

But we chose to concentrate first on usage at home," TeliaSonera spokesman Rune Fick Hansen said. "We may in a later phase extend this solution outside the home," he added. "We are introducing a product that combines the best of three worlds: mobile, fixed-line and Internet," the head of TeliaSonera in Denmark, Jesper Broekner, said in a statement. "Danish families will now be able to cancel their fixed line subscriptions at home without losing their home phone numbers, and at the same time save more than 20% on their phone bill," Broekner said. South Korean mobile phone-maker Samsung is the only manufacturer that currently sells UMA phones. It launched the SGH-P200 this summer.

Source: AFP 

Posted by Dal at 10:36 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Note: Being that I live in Washington this is very exciting.  I hope that with this happening the city of Seattle will get over this dispute with Qwest (Dominate Telco) and build our municpal wifi network to cover the metro core.

Most rest areas along Washington state highways will become WiFi hot spots, the state Department of Transportation said Monday. Road conditions and other traveler information on the WSDOT's Web site are available at no cost. Subscribers to Road Connect can visit other Web sites.

A new WSDOT partnership with Parsons Transportation Group and Road Connect Inc. provides wireless Internet access in 28 of 42 safety rest areas along Washington State highways.

Posted by Dal at 10:06 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

Out of the box, there are two things you can do with your TrixBox system and GabCast. First, you can create podcasts. And, second you can host conference calls. Today Nerd Vittles adds the third leg to the stool. With the new GabCast Player for Asterisk, you'll be able to listen to any GabCast channel feed using a garden-variety, touchtone phone.
 

Click Here for the Full Nerd 

Posted by Dal at 09:47 AM | Direct Link | Comments (0) Posted to Asterisk Help

Note: Having talking with the Ranch Network people, I am very happy to hear this.  They have some nice high end hardware devices.

Ranch Networks, provider of networking appliances designed to facilitate carrier and enterprise grade VoIP deployments, today announced it has received $1 million in funding from the New Jersey Economic Development Authority's (NJEDA's) Techniuum Initiative. The funding further establishes Ranch Networks' credibility and continued growth in the IP telephony industry.

"It is always exciting to watch an emerging business develop and succeed in New Jersey and we are pleased that Ranch Networks will use our Techniuum investment to meet its growth needs and move its technology forward," said NJEDA Chief Executive Officer Caren S. Franzini. "The broad resources the NJEDA makes available under Techniuum support Governor Corzine's initiatives to encourage the location and growth of technology and life sciences companies and related jobs in New Jersey. Early-stage and advanced companies can learn more about Techniuum by visiting www.njedatechniuum.com."

Ranch Networks' technology is designed to work with leading IP PBXs and supports all sizes of enterprise and carrier deployments. In January, Ranch Networks launched its RN series of appliances and the first security code for Asterisk, the open source PBX. Ranch Networks' IP PBX controlled appliances provide dynamic, protocol independent, per-call authenticated network access. This unique approach both simplifies and increases network security, scalability, and reliability.

The NJEDA is an independent, self-sustaining state financing and development agency that works to promote economic growth, job creation and the revitalization of New Jersey's communities with financing assistance, technical support and entrepreneurial training, and real estate development activities. Techniuum provides a continuum of customized, streamlined, long-term support to technology and life sciences companies as they advance through their life cycle of growth and expand their products and services. Through this innovative initiative, the NJEDA can assist with financing, angel and lease guarantees, access to venture capital and incentive tools as well as affordable office, laboratory and production facilities.

"The NJEDA recognized the potential in VoIP," said Ram Ayyakad founder and CEO of Ranch Networks. "As a result, NJEDA identified Ranch Networks as an integral part of the IP telephony industry. The funding will help us continue to advance our product offerings and expand our team."

Posted by Dal at 10:13 AM | Direct Link | Comments (0) Posted to Asterisk News

"With high consumer demand, poor DSL access and long customer connection times Ghana is absolutely ready for mass market, rapid install, broadband wireless services," said Leslie Tamakloe, CEO, Internet Ghana. "We will be first to market across the whole of Ghana, beginning in the capital Accra followed by the regional capitals and commercial towns."

Internet Ghana's plans are to cover the capital Accra and the ten largest commercial cities such as Tamale, Kumasi and Tema in the first phase. The CPE's will be offered at retail outlets and Internet Ghana will also have direct sales.

"The ability to easily access broadband services plus the freedom that personal broadband offers will be a winning combination to customers in Ghana," said Roger Dorf, president and chief executive officer of Navini Networks. "Our leadership in 802.16e with Smart WiMAX(TM) will also ensure that Internet Ghana maintains a competitive advantage in the market."

Navini is the only vendor to offer Smart WiMAX: Mobile WiMAX with smart beamforming enabled. Navini's Ripwave(TM) MX platform has a seamless upgrade to mobile WiMAX 802.16e, the next-generation global wireless standard, enabling consumers to be able to enjoy all the benefits of true personal broadband.

About Internet Ghana:

INTERNET GHANA is a registered and licensed Internet Service Provider (ISP), incorporated in 1995 for the provision of Internet services in Ghana and the West African Sub-Region. This was achieved in partnership with InternetMCI, NY and UUNET, which made the company a leading Internet Service Provider in the country in 1996.

Posted by Dal at 10:07 AM | Direct Link | Comments (0) Posted to WiMax

 

Chumby's team of hardware hackers wanted this device to be fun and open, the anti-iPod. While iPod has a clean look and expensive molded plastic, it's not very accessible. The Chumby is meant to be personalized. If you're crafty, you can redesign it with a seam ripper; if you're a hacker, it's all open source inside.

Click Here for more Information 

Posted by Dal at 10:00 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

The use of Skype application is constantly increasing in corporate environments and is having a dramatic impact on carriers’ revenue streams. But perhaps more importantly, the very nature of Skype traffic is raising security concerns as Skype’s protocol is proprietary and acts as a complete black box. Security administrators are currently unable to monitor and secure it efficiently.

Skype uses a peer-to-peer technology and several obfuscation techniques, making it challenging for network operators to identify associated traffic. Skype encrypts data transmitted over the Internet between peers and is particularly gifted when it comes to circumvent security limitations. Entering via uncommon channels like https (Web) port, Skype is usually very successful at passing corporate firewalls. Furthermore, Skype designers are making the software even more furtive at every new version.

The network research group at Lynanda has come with a solution to identify Skype’s traffic on-the-fly. As expected the solution is not based on common firewall practices, but on statistical data-mining techniques. In fact the method used to filter Skype is a two-steps process.

First, the firewall is exposed to its target environment to “learn” the particularities of Skype’s traffic. Then, it uses the information collected together with pattern-matching techniques to actually identify Skype’s related traffic. Various technologies like neural networks, distributed statistical calculus, and pattern recognition through machine learning are involved in the methodology developed by Lynanda. These techniques are very similar to the ones currently used in financial statistics to discover regularities and typical patterns in apparently chaotic data like stock quotes.

The originality of the method is that it not only looks at the content of the network packets exchanged, it pays also attention to the timing at which they are sent and received. Given all this data, it is quite easy to get a footprint of the Skype application and drop its related traffic.

In experiments, the filter was able to detect and block a Skype call less than 30s after it started, making it a reasonably efficient Skype blocker. A regulator can drop the call by shutting down the pipe. The number of false positives was very low, though it is expected to rise in more complex environments like large corporate networks, especially under heavy network load. The solution appears to be fully scalable and doesn’t require much human intervention or monitoring.

Though this filtering technology needs financial and technical commitment, quoting Ivan Chollet, Solution Architect at Lynanda, it could be incorporated in large organizations networks very soon:

“The only drawback of this technology is its computational expensiveness. In fact one challenge facing traffic-signature techniques on telecom networks is the high speed at which such pattern matching algorithms must be executed. Therefore, this filtering solution involves massively parallel computational capabilities as well as expensive database clusters. However, as these technologies are becoming increasingly affordable, we might see in the near future a large number of small to medium-sized companies using it.”

About Lynanda CS

Lynanda CS (www.lynanda.com) is a European software publisher and technological consultancy services provider. Lynanda CS has been founded in 2003 and now serves customers of all sizes worldwide, both directly and through its network of partners. Lynanda’s area of expertise of Lynanda is applied statistics, data-mining as well as scientific calculus. Lynanda headquarters are located in Paris, France.

Posted by Dal at 12:02 PM | Direct Link | Comments (0) Posted to Skype

The IP phone has additional features such as built-in PPPoE/DHCP clients, password-protected machine management, LCD menu display, speed-dial keys, hands-free speakerphone, last number redial, incoming message indicator, and user-
intuitive web administration system.

The dual ethernet interfaces on the IP phone allows users to install in an existing network location without interfering with desktop PC network connections. When installing the VIP-153M, SIP IP phone with the IPX-2000 (Planet IP PBX system), the IP phone can be easily integrated in office via the auto-configuration support for the IPX-2000.

Simple installation and administration, IP PBX system integration, feature-rich keypad, dynamic IP address assignment, and voice communication and compliance with various standards and filed applications make the product more attractive.

Contact Info:

TVS Interconnect Systems Ltd
# 14, Domlur Extension
Second Stage, Third Phase
Bangalore - 560071
Tel: 51254490

Posted by Dal at 11:59 AM | Direct Link | Comments (0) Posted to VoIP Hardware

The 0.3-m1 milestone release focuses on ease of use and provides the new org.asteriskjava.live package that takes care of the lowlevel action and event handling of the Manager API and offers an intuitive API for Java developers. Asterisk-Java has been updated to take advantage of the new features of Java 5.0 and therfore requires a Java Virtual Machine of at least version 1.5.0.

Asterisk-Java is used in several commercial environments and by the following Open Source projects:

    * Asterisk-JTAPI
      JTAPI implementation for Asterisk.
      http://asterisk-jtapi.sf.net/
    * Asterisk-IM
      A plugin for the Jive Messenger XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients.
      http://www.jivesoftware.org/asterisk-im/
    * Asterisk Desktop Manager (ADM)
      A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more.
      http://adm.hamnett.org/

Asterisk-Java is available under Apache 2.0 license at:
http://asteriskjava.org
 

Posted by Dal at 05:43 PM | Direct Link | Comments (0) Posted to Asterisk Development

Like most startups, Digium has humble beginnings. Founder and CEO Mark Spencer had launched a small company to provide Linux support services, and the story goes that he wrote Asterisk to save money on his phone system. "It wasn't like he had this grandiose opinion of open source," Miller said.

Spencer made all of Asterisk's technology available to the open source crowd. Today it boasts a community of developers 400 to 500 strong, and Asterisk receives about 1,000 downloads per day. "It's … unique," Miller said, "in the fact that we have such a following whose entire lives and businesses revolve around Asterisk."

In fact, echoing that phone system strategy, startup Switchvox, before it began making its own Switchvox PBX, changed from cell phones to a phone system based on Asterisk because it was a low-cost solution. The question was: What advantages does an open source PBX provide? The answer -- other than the usual one of lower cost of ownership -- was: New features make it into the product faster.

"We are able to leverage the open source community," said Joshua Stephens, president and CEO of Switchvox. "If someone wants to put in some crazy feature, we take it and make it presentable."

Digium has accrued 130 partners that have production-class solutions around Asterisk. For its part, Digium sells Asterisk Business Edition, the Digium-certified, professional-grade version of open source Asterisk, along with hardware and software products that enable telephony applications including legacy PBX, IVR, auto attendant, gateways, media servers and application servers.

The company also offers a full range of professional services, including consulting, technical support and custom software development services. In addition, Digium leverages VARs to help distribute its products.

Digium's challenge today is to move into the enterprise, where it hasn't yet done much damage. SIPBox, a Digium VAR located in Tinley Park, Ill., primarily counts education and municipalities among its customers.

"I don't have anything at the enterprise level right now," said Chad Agate, co-founder and CEO of SIPBox, a provider of end-to-end telephony solutions for companies with 200-plus users. "Asterisk is a good option for SMBs [small and midsized business] or branch offices."

One of Digium's most recent customer wins was the University of Pennsylvania, which is deploying a campus-wide unified messaging system using Asterisk.

Digium has its eye on the enterprise prize, however. According to Miller, Matrix's stake will be used for developing applications and enhancements to Asterisk to make it more appealing to the enterprise audience.

"We took [the investment] on because we want to take the company to the next level and use that cash to fuel our growth and to scale into the enterprise," Miller said. "It will create more of an opportunity for enterprise-class applications like email."

To that extent, Digium faces a challenge among picky enterprise shoppers who want to shop by brand and also match products to technology skills within the company, according to Yankee Group's Vanessa Alvarez.

"Only the most tech-savvy companies can do open source, otherwise the cost structure is too high," Alvarez said. "But the question is: Has Digium achieved enough customer base so that if you're not a brand shopper, but you are the type to do due diligence, should you also consider Asterisk?"

Source: Amanda Mitchell

Posted by Dal at 02:26 PM | Direct Link | Comments (0) Posted to Asterisk News

Note:  If you haven't already noticed, we have added a way for our user to submit stories to there favorite bookmarking or community content site.  If I am missing your favorite site please send me and email (comments[at]asteriskvoipnews[dot]com) and we add it also.  enjoy!

Sites Supported:

 blinklist

 BoingBoing

 del.icio.us

 digg

 furl

 shadows

 simpy

 Slashdot

 spurl

 yahoo! 

Posted by Dal at 01:16 PM | Direct Link | Comments (0) Posted to AVN Site Updates

Here are the main features:

- Simpler build system, no messy patching!
- CDR record handling should work correctly now
- Will set the DIALSTATUS variable
- Change pitch during conversation with * and #
- Voice effect can be applied to peer channel instead with 'p' option

The simpler build system is partially due to help from anthm for wrapping my modifications to SoundTouch in to a separate library found here: http://www.lobstertech.com/code/libsoundtouch4c/

Please note that this version is a total rewrite and I haven't had a whole lot of time to valgrind and test it rigorously.  If you have ANY problems, even if you figure out how to work around them, please let me know so I can fix it so others don't have the same problem.  I usually respond pretty quick to email (in the 5 minutes to one day range)

Plans for 0.6:

- Change voice pitch via manager api and command line
- Open to suggestions

Have fun!

- Justin Tunney
 

Posted by Dal at 09:55 AM | Direct Link | Comments (0) Posted to Asterisk Software

If you have another specific topic and would like to set up a developer meeting, please mail me off list and I will see what I can do. I am planning a SIP specific meeting too, of course ;-)

Looking forward to meeting you in Boston!

/Olle

---
* Olle E. Johansson - oej@edvina.net
* Asterisk Training http://edvina.net/training/

 

Posted by Dal at 04:30 PM | Direct Link | Comments (0) Posted to Asterisk News

Posted by Dal at 01:27 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

The product is targeted at broadband wireless access providers that have a licence to operate services in the 3.5GHz spectrum in this country.  Both companies said they were in discussion with local operators but declined to provide specifics.

The MP.16 is certified to be interoperable with at least three chipset vendors, said Palette Multimedia executive director Sukhdev Singh, referring to chipsets made by Intel, Sequans and Waveset.  

He explained that this eliminates the problem of single vendor lock-in if an operator chooses to set up a WiMAX network using non-certified or proprietary base stations. 

Lionel Chmilewsky, senior vice-president of Proxim International, said the MP.16 is based on the Intel (Rosedale) chipset which ensures maximum WiMAX compatibility.  “Also, we have more than 25 years experience in wireless technology. We spend about US$4mil (about RM15mil) on research and development per quarter, most of which are WiMAX-related,” he said. 

An MP.16 base station has a 15km line-of-sight range, and an up to 5km non-line-of-sight range, it was announced.  The retail starter kit, consisting one MP.16 base station and eight subscriber units, would cost about RM35,000, according to Sukhdev. 

Palette Multimedia, Proxim’s regional partner in Asia, will also provide training to MP.16 customers, to assist them in deploying their WiMAX networks.  

“We are offering Proxim’s Wireless Certification Training, which includes two courses,” Sukhdev said.

Source: Star TechCentral 

Posted by Dal at 01:21 PM | Direct Link | Comments (0) Posted to WiMax

As a customer of Alvarion's MGW solution for multi-residential voice and data services, Netia began WiMAX rollout in 20 cities on receiving its national licence in July 2005; Netia is one of four carriers to receive such a concession.

According to Wojciech Madalski, president and CEO of Netia, Alvarion was chosen because it could ensure that ‘we can provide WiMAX services to subscribers in the first 20 cities by the end of August.’

Posted by Dal at 01:19 PM | Direct Link | Comments (0) Posted to WiMax

Businesses can now benefit from Skype's low cost call rates with the VoSKY Exchange which plugs into the PBX on one end and a Windows XP computer on the other and connects to a PBX and a broadband connection, adding up to four Skype lines to a PBX.

This system then enables every phone extension in the office to make and receive Skype calls without any changes to existing PBX equipment, phones or user PCs, yielding major savings on calls.

WIth VoSKY Exchange a business can buy and use Skype-in numbers for inbound calls to their PABX and reduce existing 1-300 / 1-800 type call costs from their customers. They can also use VoSKY Exchange to make low cost calls overseas or between interstate offices. 

More Info: 

IW Distribution

Posted by Dal at 01:16 PM | Direct Link | Comments (0) Posted to Skype

Don't forget to peek around in SVN for all kinds of goodies. Especially trunk - the Gumstix is now a direct target for builds.  That's right, build AstLinux for a Gumstix just as easily as a Soekris!

--
Kristian Kielhofner

Download: 

http://sourceforge.net/projects/astlinux/

Posted by Dal at 10:50 AM | Direct Link | Comments (0) Posted to Asterisk Software

Calling Circles Desktop (Asterisk Edition) v 1.0.6 has been released. Please make sure you uninstall prior versions (including release candidates before installing)!

Summary of improvements in version 1.0.6:

* Smart Tags enable dialing from Microsoft Office applications. Make sure Smart Tags are enabled in your office applications to take advantage of this new feature.

* Dialing in Internet Explorer is enabled. Highlight a phone number on a web page, right click, and select Calling Circles Dial.

* A new protocol will be added to your computer called "phone:" You can dial a phone number from the Windows Start menu run dialog. Select Start->Run and type phone: where is the phone number you'd like to dial. Now you can integrate dialing from any application without the need to learn COM!

All of the Calling Circles Desktop dialing rules will be applied when using any of the above methods to make a call. Thank you to all of our early adopters who provided fantastic feedback! To download this release, click here. If you need a new evaluation key, please send me an e-mail! elliot.otchet@callingcircles.com

Click Here for more Information 

Posted by Dal at 12:52 PM | Direct Link | Comments (0) Posted to Asterisk Software

Interoperability validation of Brekeke's OnDO PBX and AudioCodes TP-260 PCI VoIP gateway board has been successfully completed, which will allow users and OSN partners to quickly develop scalable, high-quality VoIP telephony systems for small to medium businesses up to the enterprise level.

Brekeke's OnDO PBX is a SIP-based telephony system well known for its easy installation and intuitive design. In addition to traditional PBX features, OnDO PBX includes advanced features such as call conference, call recording, Automatic Route Selection (ARS) failover, call forwarding, voicemail, and much more.

Fully scalable for the small to medium sized business owner up to the enterprise level, Brekeke's PBX provides multi-platform support and professional technical support to its users. AudioCodes TP-260 PCI board is an ideal solution for trunking gateways to the PSTN and integrated gateways for IP-PBXs and all-in-one communication servers. Combined AudioCodes and Brekeke solutions offer essential features of a legacy PBX on an IP-PBX, and provide a combination of telephony networking and data networking for the office. The TP-260 PCI gateway board allows Brekeke to offer a one-box solution, and enables Brekeke customers to utilize a T1 line connection for their VoIP telephony systems.

Source: Brekeke

Posted by Dal at 12:49 PM | Direct Link | Comments (0) Posted to VoIP News

Generally I don't like to get involved but I "did" change the URL for the "Zaptel 1.2.8, Asterisk-Addons 1.2.4 and Asterisk 1.2.11 have been released"  to call attention to this obvious advertising to his site by adding Daily Asterisk News" and the fact that he posted it when obviously Asterisk.org is the correct url: http://www.asterisk.org/node/101

 To be perfectly honest this was personal because of certain comments made to me by Mr. Riddell.  I think that seeing the above post was the tipping point.  I apologize if anyone else was offended and I am refraining from ever doing that again.  I do this site for my love of Asterisk and VoIP technology.  I would also like to add that Mr. Riddell is correct about Sineapps being ad-free but I would have to say his "asterisk consulting" business most likely pays the "bills". Asterisk VoIP News is ad-supported and thankfully pays for it's hosting.  Thanks for you support.

-Dal 

Last Update(08/24/2006):

Well I for one have way too many more tasks to do than go back and forth.  If I am not doing anything for Asterisk than our traffic logs must be lying.  It's funny how Matt asks about "has Dal done anyhting for Asterisk"?  Personally I think if you have following this site for the last 19 months than you would see AVN is just another source to get information about the entire Asterisk Family.  Well this seem time better than any to announce that I will be starting a Asterisk Self-Help and Common Issuse article series continung from where we left off. (Voicemail.conf I think??).  If anyone would like to submit ideas or draft aritcles please contact me (email addy below).  To close, I am not going to deny that when I was first learning to blog I didn't know proper online etiquette.  Everyone have a great day and please feel free to interact with the site or tell me if something is missing.

-Dal 

Posted by Dal at 12:10 PM | Direct Link | Comments (0) Posted to Asterisk News

For several years now, many PC users have been saving themselves a small fortune by using their PC to make VoIP calls instead of using the tradional telephone network (especially when calling internationally). PDAs together with the growth in WiFi coverage now allows this saving to be made while you are on the move. With the features offered by VoIP you can even take the same number used on your PC with you on to your PDA - you only need one contact number.

There is a huge selection of VoIP providers worldwide with very cheap rates when compared with cell phones; it is often free to call some destinations and many do not charge a monthly fee. Even international calls over VoIP can cost lower than 3 cents/min.

Articulation for PalmOS, from Hampton Software Ltd, offers users of Palm PDAs the opportunity to communicate on the go using the VoIP service of their choice.

Features:
- Ability to make PSTN calls with your PDA
- Select the SIP VoIP provider of your choice
- Lookup phone numbers direct from you contacts
- Touch-tones (DTMF) for 'phone menus'
- Supports WiFi, Bluetooth and EVDO connections
- Call timer
- Silence suppression (only transmit your voice)
- Echo cancellation for 'speakerphone' use

Technical Features:
- Secure account authentication with SIP MD5 Authentication
- Supports GSM and G711 codecs (GSM is suited to low bandwidth connections)
- NAT support through RFC 3581 and ability to fix ports for symmetric NATs
- Symmetric RTP support
- Low latency
- Low memory usage

Some features depend upon the VoIP service provider selected.

More Info: http://www.hamptonsoftware.co.uk/articulation/index.shtml

Posted by Dal at 09:27 AM | Direct Link | Comments (0) Posted to VoIP Software

As always, the release files are available on the Digium FTP servers at http://ftp.digium.com, in both tarball and patch file form. All of the release files have been signed with our GPG keys and the signature files are available in the same directories as the release files.

Thanks for using and supporting Asterisk!
 

Posted by Dal at 09:10 AM | Direct Link | Comments (0) Posted to Asterisk Releases

Asterisk Jobs (www.asterisk-jobs.com) is always upgrading and changing the site. Look forward to more announcements in the near future. The next planned release is a complete revamp of the site to include tags and other fancy stuff that will make searching for that dream job involving open source telephony a reality - quicker, and easier!

For more information or to start looking for Open source Asterisk VoIP employment head over to: http://www.asterisk-jobs.com

Thanks,
Asterisk Jobs Staff
 

Posted by Dal at 10:45 AM | Direct Link | Comments (0) Posted to Asterisk News

Over the past three months, Vonage has equipped more than 500 locally-run emergency call centers across the U.S. with E911 -- bringing the total number of calling centers with emergency 911 service to over 5400.

In June, the FCC gave Vonage, and all Internet-based phone service companies, 120 days to create an E911 system from scratch, and provide all of its customers, wherever they are in the U.S., with E911 service. In that amount of time, Vonage worked closely with local 911 centers, also known as Public Safety Answering Points (PSAPs), across the country to turn on E911 for as many customers as it could. Currently, Vonage continues to test and turn up new PSAPs which are VoIP-ready every day and will continue to do so until Vonage customers have access to E911.

"Today over 85 percent of our U.S. subscriber lines have full E911 capability, which is a tremendous step for Vonage," said Jeffrey A. Citron, Vonage's chairman and chief strategist. "Vonage will continue to work with the FCC, regulators, Congress and public safety until PSAPs across the nation are equipped with E911."

Posted by Dal at 10:42 AM | Direct Link | Comments (0) Posted to Vonage

Posted by Dal at 10:38 AM | Direct Link | Comments (0) Posted to WiMax

Ingate Systems, which develops firewall technology and products that enable SIP-based live communication for the enterprise while maintaining control and security at the network edge, showcases traversal and security solutions for enterprises connecting to SIP trunks, with Ingate Firewal and Ingate SIParator products at the VoiceCon show.

SIP trunking is a service offered by Internet Service Providers (ISPs) which permits businesses to adopt Voice-over-IP (VoIP) with its attendant benefits and remain connected to others who rely on the PSTN. For the enterprise SIP trunks can offer significant cost savings, as they eliminate the need to purchase either the local PSTN gateway or costly ISDN BRIs (Basic Rate Interfaces) or PRIs (Primary Rate Interfaces). The local IP-PBX can be connected to the service provider's PSTN gateways over the Internet.

Ingate Firewalls and SIParators solve the Network Address Translation (NAT) traversal issues that are faced by businesses using a SIP trunk. All voice traffic (as well as data traffic) must traverse the enterprise firewall/NAT. However, SIP traffic cannot traverse traditional enterprise firewalls and NAT devices. This is because traditional firewalls do not differentiate between SIP traffic and unwanted traffic. As a result, the firewall/NAT device blocks all SIP traffic, which includes VoIP. Ingate products resolve this issue, enabling enterprises to utilize SIP trunks.

Ingate's fully SIP-capable, proxy-based Firewalls and SIParators also provide flexibility to interoperate with carrier-specific requirements like numbering plans and authentication. They establish least cost routing rules, which offers businesses the ability to use multiple service providers, and to switch between them depending on which offers the best possible rates (which may vary by time, day or location). Long distance calls cost the same as a local call, reducing expenses for businesses as well as their customers, partners, etc. trying to reach, for example, the corporate sales force.

SIP trunks further reduce costs as they eliminate the need for separate voice and data connections, expanding the potential for communications convergence using both voice and data together. SIP trunking also offers scalability so that, as a company grows, all necessary infrastructure to handle additional voice/data traffic is already in place.

Security Over The Public Internet

Ingate products are specifically designed to leverage all the security benefits available with SIP communications. Ingate's enterprise-class Firewalls secure data and SIP traffic, while SIParators secure SIP media while leaving the traditional firewall in place (working in parallel to the SIParator) to secure data traffic.

Both products feature Ingate's full SIP proxy technology, which allows for advanced filtering, verification, authentication and routing, as well as dynamic control of the opening and closing of media ports. Encryption of the signaling is done using Transport Layer Security (TLS) and of the media (voice, video, etc.) using Secure RTP (SRTP). With encryption, the sessions are kept private with no chance of eavesdropping.

They also secure full VoIP redundancy, as traffic can be routed to a back-up carrier if the primary carrier is unavailable. And Ingate's proxy-based solutions can be used to provide local call management if the service provider cannot be reached; and, if the enterprise has a local gateway installed, outgoing calls can be routed to the PSTN if all connections to the hosting provider are unavailable.

"SIP trunking offers enterprises the benefits of converged communications and saves substantial expense by eliminating the need to purchase their own PSTN gateway," said Steven Johnson, President, Ingate Systems. "NAT/firewall traversal is a critical issue in any SIP trunk deployment. Ingate's SIP-capable Firewalls and SIParators offer an elegant solution to this problem, while also securing voice traffic."

Connecting Remote Workers To The SIP Trunk Without VPN

SIP trunking, when used in conjunction with Ingate's Remote SIP Connectivity software module, allows remote users to traverse most SIP-unaware residential firewalls and NAT devices and use all the IP-PBX functions installed in the enterprise. Remote SIP Connectivity complements SIP trunks, as remote workers can use the software to interface with their corporate IP-PBX and, if they were making calls to outside the enterprise, do so using the SIP trunk.

For more information, please visit us at Ingate's VoiceCon booth #639, or online at www.ingate.com.

Posted by Dal at 10:32 AM | Direct Link | Comments (0) Posted to SIP

The two companies have enjoyed a long, productive relationship, beginning with Skype's licensing of GIPS VoiceEngine PC in the summer of 2003. The result was the first PC-to-PC VoIP deployment to reach critical mass, allowing at first hundreds and then millions of users to connect and make free, high-quality phone calls. The rapid uptake of this technology signaled the beginning of a telecommunications revolution.

The agreement will help further accelerate VoIP migration as the North American market for IP phones is expected to double between 2006 and 2009, according to Infonetics Research.

"Skype has clearly revolutionized global personal communications with its offering," said Gary Hermansen, President and CEO of Global IP Sound. "This latest step in our relationship helps to solidify this growing market by expanding the options for Skype-enabled hardware devices and the ability for consumers to have a consistent, high-quality user experience regardless of platform."

Posted by Dal at 10:26 AM | Direct Link | Comments (0) Posted to Skype

As a result of this agreement, Digium and BandTel will partner in marketing initiatives to resellers interested in the cost and flexibility benefits of an open source telephony solution.

The alliance provides Digium and BandTel's channel partners worldwide to expand their portfolio of VoIP service-offerings and provide customers with a highly fault-tolerant, cost-effective and feature-rich VoIP solution. BandTel provides SIP-based VoIP solutions that fully support the Digium open source platform.

Source: Digium 

Posted by Dal at 10:24 AM | Direct Link | Comments (0) Posted to Asterisk News

For small businesses, two AT-WA1104G devices can be configured in bridge mode to link together 2 remote LANs. Unit can also be deployed as wireless access point and for wireless distribution system support. Built-in firewall protects network from incoming intrusions, while outgoing access to undesirable web sites can be limited by URL filters.

The product with an attractive price point under 100 euros, can be used in several different modes, including as a broadband wireless router and a wireless point-to-point bridge. It can also be deployed as a wireless access point and for wireless distribution system (WDS) support. The AT-WA1104G is so versatile that small companies can use this single cost-effective device to meet all of their wireless communications needs while driving costs down.

With little training needed, the AT-WA1104G includes a web-based setup wizard that is easy to use for non-IT specialists, contributing to fast deployment as well as reduced operating costs. The AT-WA1104G also provides peace of mind for small businesses with its state-of-the-art security features.

Posted by Dal at 10:21 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

With this improvement in detail, voices have more richness and clarity improving the overall experience of a phone call. This week the software has brought the first two elements together and topped it off with a new capability that may change the way we interface to our phones.

GoogleTalk has recently released a new version of their client capable of transmitting audio at 16 kilohertz making it possible to call FreeSWITCH and interact in a conference bridge or listen to a text-to-speech engine read you your favorite news story all in high definition audio.

Now for the twist, not only can you interact with the system on the phone by listening to the audio and dialing a few digits, now you can send and receive text messages with the system at the same time.

Imagine being asked out loud for your account information and being asked to type your name into the chat box and actually have the other end of the call able to intercept the information and react accordingly. This may break the paradigm of the auto-attendant altogether.

Posted by Dal at 10:14 AM | Direct Link | Comments (0) Posted to VoIP Software

As part of the agreement, Fusion will market its Efonica brand of VoIP services throughout the Jinti website. In addition to category exclusivity, the Efonica service offerings will be integrated into Jinti’s consumer registration process. As visitors register for Jinti services, they will be prompted to register for Efonica’s offering. “We are very pleased to enter into this relationship with Jinti, a key player in the burgeoning Chinese Internet marketplace.

Jinti’s unique community services website continues to experience dramatic growth through unique service offerings and the execution of a dynamic business model. With a growing Internet community, presently numbering over 120 Million people online, China represents a key market for Fusion in the Asian Region and a tremendous opportunity for the Company to continue to grow our worldwide community of users” said Matthew Rosen, President and CEO of Fusion.

“Fusion’s unique Efonica VoIP services are a very good complement to our community services website, giving our customers a chance to speak with one another and connect with friends and relatives for a significant savings. With over 400,000 subscribers registered in less than two months, Efonica represents a rapidly growing, cutting-edge service we are confident will appeal to the Chinese marketplace.

We also believe that Jinti’s 500,000 registered users with over 3 million unique visitors a month offers US firms like Fusion a unique opportunity to rapidly establish themselves in the China market.” said Jin Zhao, President and CEO of Jinti. Efonica offers consumers the ability to call each other for free using their existing landline or mobile telephone numbers.

Calls can be made to and from any combination of PCs, Internet phones and regular telephones (with a SIP adapter), connected to either a wireless, broadband or dial-up Internet connection. The service was introduced incorporating Fusion’s patent-pending Worldwide Internet Area CodeTM, which combined with a subscriber’s existing telephone number, further simplifies the process of making a call. Consumers subscribe to Efonica by visiting www.efonica.com and completing a free and easy registration process.

Posted by Dal at 11:58 AM | Direct Link | Comments (0) Posted to VoIP News

Hello, 

We are releasing a beta version of our java eclipse based asterisk voicemail client called "Tycho". You can download the client binary at: http://sip-syndication.com.

Please note that the client needs a set of php scripts installed on the server side. The server side scripts, called "vmxml" are open source and available on the same website.

Screenshots:
MS Windows
http://sip-syndication.com/images/stories/tycho/msw-tycho-overview-65.png
Mac OS X
http://sip-syndication.com/images/stories/tycho/tycho_macosx_ppc_overview-65.png
Linux
http://sip-syndication.com/images/stories/tycho/lnx-ubuntu-tycho-overview-65.png

If you have any questions or problems installing/using the client please mail: t y c h o at sip-syndication dot com or post to the support forum on our website.

cheers,
Arnd
 

Posted by Dal at 09:40 AM | Direct Link | Comments (0) Posted to Asterisk Software

Editor's Note:  This was sent in from one of our readers Kerry Garrison with Tech Data Pros.  It's a interesting read.  Please subimit any questions or comments about this article.

The actual number of things that can cause echo and how to resolve them all could almost fill a book, instead we are going to look at the most basic echo canceling using the Zaptel drivers. This will solve "most" echo problems but will not work in every case, regardless it is the first tier approach to solving to problem. The procedures outlined in this article will work with Digium TDM analog or TE PRI cards.

Click Here for the Full Article 

Posted by Dal at 09:33 AM | Direct Link | Comments (0) Posted to Asterisk Help

To add additional extensions takes about 5 seconds. This PBX is just the ticket for a small business or a school or even a fraternity or sorority house. It's also perfectly suited for your home. You get every imaginable PBX telephony feature including music on hold, call forwarding, and call transfer as well as a preconfigured AutoAttendant which lets your friends and colleagues direct an incoming call to any of your extensions or even your cellphone.

For those with the magic password, you can even dial in and get dialtone to make five hours of free calls each week to dozens of countries around the world including all of the U.S. and Canada, most of Europe, South and Central America, Australia and all your Far East favorites including China, Taiwan, Russia, and Japan. And the total cost: about $12.50 for each three months of service.

All incoming calls are free, and you even get your very own area code and phone number to pass out to your friends that are still chained to plain old telephones or cellphones. And, yes, all your favorite Nerd Vittles applications are preinstalled and ready to go including weather forecasts for 1,000 airports, MailCall for Asterisk to read you your email messages, NewsClips for Asterisk to read you the news, and the AsteriDex robodialer complete with a web interface to place your outbound calls and to serve up customized CallerID for your incoming calls.

Last but not least, you get all of the bundled TrixBox applications including freePBX, SugarCRM, Samba for Windows networking, FTP and SSH support, WebMin, PHP, MySQL, Perl, Apache, SendMail, integrated fax-to-email support, calling card billing, and more. It slices, it dices ... You get the idea.

Click Here for the Full Nerd 

Posted by Dal at 10:32 AM | Direct Link | Comments (0) Posted to Asterisk Help

Joshua Colp joined our development team a few months ago. Josh (file on IRC/Mantis) has been working on Asterisk development for quite some time and had contributed many features and bug fixes as a volunteer community member, along with being very active on the IRC channels and issue tracker.

Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements.

Jason Parker joined our development team at the beginning of this week. Jason (qwell on IRC/Mantis) has been maintaining the chan_skinny driver for Cisco SCCP phones as well acting as a bug marshal and fixing various bugs in Asterisk for the past year or more.

Russell Bryant has been a Digium part-time employee and an active Asterisk maintainer since before I got involved with Asterisk :-) His contributions are innumerable, and he has worked far more than the 'ten to twenty hours per week' he claims to have available outside of his school work! Russell (russellb on IRC/Mantis) will be joining us full time in Huntsville after the winter semester is complete, when he expects to graduate.

Please join me in welcoming all these new members of our development team; they are helping to make Asterisk (and our other software products) better every day and will enable us to accelerate our products into the future.

--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
 

Posted by Dal at 05:51 PM | Direct Link | Comments (0) Posted to Asterisk Development

The company originally launched a free rPath Builder Online service earlier this year. Now, it is launching its first major revision in a bid to equip ISVs that want to use Linux as their operating system base.

Instead of an ISV or an application vendor relying on a user with the necessary operating system, the Linux as an appliance model that rPath is pioneering enables ISVs to bundle their application together with the Linux operating system in a complete stack.

"ISVs don't want to be the operating system vendor, with an open source operating system like Linux they don't have to be," rPath CTO founder Erik Troan.

The first version of rPath Builder enabled users to build distributions as ISO images for CD/DVDs, as well as VMware images that would run inside of the VMware virtualization player.

rPath Builder 2.0 extends the distribution paradigm by allowing users to build Live CDs as well. With a Live CD, end users boot and run the whole appliance from their media drive without the need to physically install to a hard drive.

The new version also makes managing and deploying an rPath-built appliance easier with the rPath Appliance Agent(RAA). With RAA, users can control setup, configuration and updates for the rPath-based Linux appliance.

"With rPath Appliance Agent we're trying to make it easier for developers to provide a pre-built system and make it easier for people to build Linux appliances," Troan explained. "Instead of logging in and editing a config file, RAA is the initial thing that comes up when you load the system and you can connect to it over SSL."

RAA is a starting point for building Web consoles and it can be branded by ISVs such that the end user won't know that it rPath Linux.

Though the rPath approach to building Linux appliance may serve to remove barriers for ISVs and others, it's the appliance approach itself which may present the biggest obstacle to rPath adoption.

Troan wouldn't disclose how many people are currently using rPath. Some well-known applications are already offering rPath Linux-built appliances.

Among them is open source VoIP pioneer Digium, which is currently offering its Asterisk Business Edition as an all-in-one Linux appliance. SugarCRM also offers an rPath Linux appliance; database vendor Ingress today announced its Ingress Database appliance.

Troan is betting that, as the market for virtualization matures, so will the concept of virtualized appliances with customers.

"These industry efforts are making our selling job, and objections [such as] 'do I need a virtualized software appliance' and 'I don't get it,' easier and easier every time we go out."

Posted by Dal at 09:07 AM | Direct Link | Comments (0) Posted to PBX / iPBX

The Motorola PTP 49400 delivers increased bandwidth for applications such as on-scene streaming video, Internet and database access, and transfers of large files such as maps, blueprints, medical files and missing-person images.

The PTP 49400 also handles backhaul for wireless networks and temporary fixed point-to-point links, providing a cost-effective solution for T1 replacements and a 4.9 GHz solution for backhauling fixed video surveillance networks.

Motorola’s MOTOwi4 portfolio of innovative wireless broadband solutions and services create, complement and complete IP networks. Delivering IP coverage to virtually all spaces, the MOTOwi4 portfolio includes Fixed Broadband, Mesh, Broadband over Powerline, and WiMAX solutions for private and public networks. 

Posted by Dal at 09:02 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Nearly half of contact centres in North America will use voice over IP (VoIP) systems by the end of next year, according to new research.  The new report from the Yankee Group says that despite VoIP adoption lagging behind earlier predictions, the call centre market is embracing the technology.

Forty-seven per cent of call centres are expected to roll out VoIP by 2007, compared to just 17 per cent in 2005.

The key reasons for using the new technology are to manage multiple call centre sites cheaply and flexibly and to be able to place agents anywhere, including at home.

The largest contact centres - those with over 500 seats - will see the greatest uptake of VoIP from now until the end of 2007, the analyst predicts.

Though VoIP for the enterprise has been much-hyped in recent years, Yankee Group said adoption has not met its predictions.

The report said: "This lacklustre performance of VoIP products and services in the enterprise - and more specifically in the contact centre marketplace - indicates that much more than technology and end-user perception are involved in driving the market for VoIP applications to a higher level."

The biggest concerns for contact centres looking to roll out VoIP are costs, including fears of high upfront costs. Overall respondents believed they would see savings of six to 15 per cent from using the technology.

Other concerns include voice quality - a worry for 46 per cent of the market - and security and reliability.

Call centres are looking to buy VoIP systems from telcos and telephony hardware/software vendors over system integrators and value-added resellers, and though data networking vendors are still in the minority, they've made progress in capturing market share, according to the analyst.

Source: Yankee Group 

Posted by Dal at 08:57 AM | Direct Link | Comments (0) Posted to VoIP News

For the Asterisk@von agenda, see http://www.pulver.com/asterisk/ - there will be additions coming up soon.

Mark Spencer, the creator of Asterisk, will speak on Wednesday, September 13th:
----------------
Industry Perspective: An Open Source Carol: The Ghost of Open Source; Past, Present and Future
Wednesday, September 13, 2006, 4:45pm - 5:15pm

As the creator of Asterisk, the industry's first open source telephony platform, Mark Spencer, president of Digium, will discuss the phenomenal growth and industry acceptance of open source 
telephony since last year's Fall VON. Companies (from the enterprise to the SMB) as well as carriers and developers have come to realize the benefits of open source solutions go far beyond cost savings. In fact, flexibility and competitive advantage are two of the main drivers behind moving to an open source solution. Taking a glimpse at the past, present and future of open source telephony, Mark will discuss the role this industry has and will play in the development 
of next generation VoIP services.

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


For the full agenda of the conference and the exhibition, see: www.von.com/index.html


I am looking forward to meeting you at Von - the premium VoIP and Asterisk conference & Trade show!

/Olle
 

Posted by Dal at 08:50 AM | Direct Link | Comments (0) Posted to Asterisk News

Both classes are bootcamp-level classes with dCAP oppurtunities.
Visit our web site for more information or send e-mail to:
info@edvina.net

Asterisk@von - Voice On The Net in Boston
---------------------------------------------

For those of you going to Von Boston there will be a series of Asterisk seminars at von, labelled Asterisk@von. I will be covering the coming release, 1.4 as well as run developer meetings and a meeting for the Asterisk Video Task Force. See the von web site at http://www.von.com for more information.

Regards,
/Olle
 

Posted by Dal at 08:47 AM | Direct Link | Comments (0) Posted to Asterisk Help

PieNetworks says it believes the device will "revitalise the use of payphones thanks to its combination of lower cost calls and other products." It suggests the device also has the potential to become a component of future mobile telephone networks as manufacturers enable their mobile phones to operate over WiFi hotspots and VoIP as an alternative to high cost cellular networks.

PieNetworks says it has already commenced discussions with interested parties in Australia and has received expressions of interest  from telecommunication companies overseas.

Posted by Dal at 10:08 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

This agreement tightly integrates Zfone with BorderWare's SIPassure VoIP Security Gateway, bringing a new level of security and ease of use to VoIP systems.

VoIP and related real-time communication applications, such as video conferencing and instant messaging, continue to attract considerable interest worldwide, with millions of active private and business VoIP users today. Carriers and enterprises are increasingly seeing the benefits of VoIP services that allow voice messaging and video conferencing to be conducted securely, like email, as communications are transferred freely over traditional phone networks and the Internet.

Together BorderWare and Phil Zimmermann have the security knowledge and IP expertise to provide secure VoIP solutions to the marketplace. BorderWare has a long standing history of being in the security industry for over 12 years protecting email, IM, web and VoIP, for the most demanding networks. Zimmerman has over 20 years experience specializing in cryptography and data security, and is the creator of Pretty Good Privacy, an email encryption software package that was originally made available on the Internet in 1991.

By integrating Zfone media encryption with SIPassure, BorderWare is extending the VoIP security provided to organizations from threats such as Spam to Denial-of-Service attacks to include eavesdropping, spying and wiretapping, while delivering the same level of security and convenience that organizations have come to expect from email.

"We're pleased to be the first security company to license Zfone, as it is poised to become the global standard for VoIP media encryption," said Tim Leisman, CEO of BorderWare. "Phil's reputation and proven track record in email and data encryption offer a very logical foundation for VoIP media encryption, and our vast experience in security makes an unbeatable and reputable combination to deliver VoIP security through the SIPassure offering."

"As VoIP grows into a replacement for the PSTN, we will absolutely need to protect it, or organized crime will be attacking it as intensively as they attack the rest of the Internet today," commented Phil Zimmerman. "VoIP is far more vulnerable to interception than the PSTN. Corporate VoIP calls can be captured and organized on disk for convenient point-and-click wiretapping by criminals half a world away. The combination of Zfone and BorderWare makes it easy to convert an entire installed base of office phones into secure phones in just one stroke, without having to replace them all with ZRTP-enabled phones."

Zfone is secure Voice over IP. It uses a new protocol called ZRTP, which is better than other approaches to secure VoIP because it achieves security without the reliance on a PKI (Public Key Infrastructure), key certification, trust models, certificate authorities or key management complexity that bedevils the email encryption world. It does not rely on SIP signaling for the key management, and in fact, does not rely on any servers at all. It performs its key agreements and key management in a pure peer-to-peer manner over the RTP packet stream. It interoperates with any standard SIP phone, but naturally only encrypts the call if you are calling another ZRTP enabled device or phone. This new protocol has been submitted to the IETF as a proposal for a public standard, to enable interoperability of SIP endpoints from different vendors.

SIPassure is the industries first VoIP Security Gateway - a new class of product that offers the best features of an enterprise firewall, an Application layer gateway (ALG) and a Session Border Controller (SBC) to take security for VoIP applications to the next level. SIPassure is designed to secure all Session Initiation Protocol (SIP) based applications including VoIP services, video conferencing and other messaging applications. SIPassure ensures that organizations using SIP-based applications are safe from abuse and service disruption from internal and external malicious attacks, interference spam and other related threats. SIPassure sets a new standard for price performance for both the enterprise and carrier markets.

About Phil Zimmermann

Mr. Zimmermann received his bachelor's degree in computer science from Florida Atlantic University in 1978. He is a member of the International Association of Cryptologic Research, the Association for Computing Machinery, and the League for Programming Freedom. He serves on the Roundtable on Scientific Communication and National Security, a collaborative project of The National Academies and The Center for Strategic and International Studies. He is Chairman of the OpenPGP Alliance, serves on the Board of Directors for Computer Professionals for Social Responsibility, and is on the Advisory Boards for Anonymizer.com, Hush Communications, PGP Corporation, and Qualys.

Posted by Dal at 10:03 AM | Direct Link | Comments (0) Posted to VoIP Security

Highlights for the quarter include:
" The 2nd quarter 2006 results represent a more than 55% reduction in losses with a reported net loss of $6.1 million as compared to the 1st quarter 2006 loss of $13.8 million. 

" The Company has demonstrated a continuing trend of reduction in its overall loss for three consecutive quarters, beginning with the 4th quarter 2005 net loss of $14.5 million.

" Approximately $10 million (50%) of the six-month loss was non-cash (including stock/option compensation expense, depreciation, amortization and other non-cash expenses).

" 2nd quarter 2006 produced positive gross margin of $230,000 (+3% of sales) - first positive gross margin reported by company in past four (4) quarters.

" The 2nd quarter 2006 results show a continuing improvement trend in gross margins for three consecutive quarters with current 2nd quarter gross margins at +3% versus -6% of sales (-$619,000) in 1st quarter 2006 and  -7.2% of sales (-$708,000) in 4th quarter 2005.

"It is gratifying to see that our profit improvement actions implemented over the past several quarters are now taking hold," said Gary Post, VoIP, Inc.'s President and CEO.  "We now have our sights set firmly ahead toward break-even," Mr. Post added.

"We have developed a lot of useful technologies at VoIP, Inc., and we are now on course to commercializing these technologies.  These positive results will be reflected in customer growth and our future financial statements," said VoIP, Inc.'s CTO/COO Shawn Lewis.

VoIP, Inc.'s full 2006 2nd quarter financial statements will be included in its Quarterly Report on Form 10-Q, expected to be filed on Wednesday, August 16, 2006.
 

Posted by Dal at 09:56 AM | Direct Link | Comments (0) Posted to VoIP News

AgileMesh incorporated Firetide's multi-service wireless mesh technology into its video surveillance systems to create the portable video surveillance network. The network enables instant deployment of wireless multi-camera networks at crisis locations and has been utilized in a number of SWAT missions to maintain up-to-the-minute updates on criminal activity.

For instance, the wireless network was a valuable tool in apprehending carjacking suspects in a case that had escalated to a hostage situation. When the suspects escaped the scene on foot, the SWAT team quickly deployed an AgileMesh wireless video surveillance network to secure the building and its perimeter. The wireless video solution allowed field teams to maintain constant visual contact without entering crossfire range.

"It sounds like something from a Hollywood action thriller, but the ability to instantly set up a network of camera systems that communicate wirelessly with each other is already here," said Mark Satinsky, corporate spokesperson for AgileMesh. "Firetide's wireless mesh infrastructure is like an intelligent highway for AgileMesh's video surveillance systems, which can visually patrol situations too dangerous for human entry. Our combined technologies help keep rescue teams safe while giving them better tools for making critical decisions."

The AgileMesh solution uses portable devices with connectors for both analog video cameras and Ethernet-based IP digital cameras. The AgileMesh units contain Firetide mesh network hardware and software allowing them to transmit video streams from one unit to another until it reaches a central monitoring location. As soon as power is applied, operators can set up their units by simply turning knobs located on the front panel to the desired frequency and channel.

This feature eliminates the need to connect a laptop computer to manage the mesh or rely on a remote network operations center to set up the system. As soon as each unit is set, it automatically links to other nearby units to form the mesh network. "AgileMesh has removed all of the complexity and made it incredibly simple to set up a sophisticated wireless surveillance network in a matter of minutes without a trained technician," said Bo Larsson, CEO of Firetide.

"This is a real example of how mesh networks are enhancing public safety. In addition to working with any kind of camera including infrared and analog, the AgileMesh units can also use our mesh technology to carry voice over IP and Wi-Fi data on the same network as the video.

This will enable a complete multi-service communications system for police and other first responders making it easier and safer for them to get ahead of criminal activity." The AgileMesh systems are available for unlicensed 2.4 GHz and 5 GHz and licensed 4.9 GHz public safety operation.

Source: AgileMesh 

Posted by Dal at 09:39 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

Combining stylish design with state-of-the art technology, the GN 9300 series uniquely addresses the evolving telephony marketplace, offering a fit for today, tomorrow and the transition in-between.

"The GN 9300 series provides hands-free convenience in the office and gives users freedom to roam -- a benefit being realized outside the office due to Bluetooth wireless headset use with mobile phones," said David Wood, senior vice president and general manager for GN Netcom's North American operations. "The GN 9350 also provides a future-proof solution for companies who have not yet migrated entirely to IP telephony. Our new family of products is focused on providing solutions to users that meet all of their business needs, regardless of the organizational size or communications platform."

Demonstrating the migration to IP telephony (VoIP) is a recent study by Infonetics Research, User Plans for VoIP: North America 2006, which states that while IP Telephony is still in the early phases of adoption, it is starting to go mainstream. "By our estimates almost half of small and two-thirds of large organizations in North America will be using VoIP products and services by 2010. Many of them will be deploying softphones as part of these IP telephony roll-outs, and will need headsets that can operate with their desk and softphones. The GN 9300 series is capable of operating in both environments, and adds wireless capability," said Matthias Machowinski, directing analyst at Infonetics Research.

The flagship product of the series, the GN 9350, is also the first wireless headset to offer Digital Signal Processing (DSP) and, for IP telephony, wideband audio with an impressive 6.8 kHz audio bandwidth that is more than twice that of conventional telephony. Not only does the DSP filter out incoming background noise, it also adjusts incoming audio levels to maintain a safe consistent volume throughout the call, providing users with crisp, clear conversations. In addition, the GN 9350's noise-canceling microphone means that users can be assured that their voice is transmitted clearly even in noisy environments. To expand communication, users can roam up to 300 feet from the base station and take advantage of the headset's multi-unit conferencing capability.

The three headsets utilize the latest wireless standard, DECT 6.0, and operate on a WiFi-friendly frequency of 1.9 GHz. With 64-bit digital encryption, conversations are kept secure. The GN 9330 and GN 9330 USB have a range of up to 200 feet and come in the same stylish form factor as the GN 9350. Each of the headsets connects effortlessly and quickly to the landline telephone and/or computer. In addition, their simplistic, intuitive headset controls make them easy to operate.

The GN 9330 works with traditional office desk phones, bringing hands-free convenience and superior audio quality to conventional office telephony. Its IP counterpart, the GN 9330 USB, is for use with PC-based solutions such as IP softphones. Like the GN 9350, the GN 9330 USB is the industry's first wireless headset to offer 6.8 kHz wideband USB audio quality. Both headsets offer affordable innovative headset technology with simple 'plug and play' installation as well as all the benefits of hands-free convenience, including increased efficiency, comfort and mobility.

Pricing & Availability

The GN 9300 series is now available through specialty distribution channels and authorized resellers across North America. A list of resellers for the GN 9300 series can be found at www.gnnetcom.com/gn9300wtb. MSRP of the GN 9350 is $349. The GN 9330 and the GN 9330 have an MSRP of $249.

Posted by Dal at 09:37 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. “Most commercial IVR applications such as call centers are Windows-based.

This open source initiative will free developers from reliance on hard-to- use, proprietary solutions based on expensive hardware,” says Sangoma Technologies President and CEO David Mandelstam. “Recognizing the inherent strengths of the Yate model, we have offered our technical and financial assistance to offer this project to the large market that is dominated by Windows.

We expect to take a leadership role as we support the project for the developer community.” “Yate plus Sangoma are a natural fit which allow integrators to build inexpensive, robust and flexible applications for telephony on the Windows platform,” adds Diana Cionoiu, Null Team CEO.

“Since Yate supports widely-used VoIP protocols like H.323 and SIP, integrators can build IP call centers, IVRs or any other telephony applications using technologies available under Windows platform. Yate can be used with the new YateClient 1 and Mozilla Firefox embedded browser that allows integration with various databases.”

Click Here for More Info about Yate 

Posted by Dal at 09:27 AM | Direct Link | Comments (0) Posted to VoIP Software

Some of the potential applications include storing medical records on a hospital patient's wristband; providing audio-visual supplements to postcards and photos; helping fight counterfeiting in the pharmaceutical industry; adding security to identity cards and passports; and supplying additional information for printed documents.

The experimental chip, developed by the "Memory Spot" research team at HP Labs, is a memory device based on CMOS (a widely used, low-power integrated circuit design) and about the size of a grain of rice or smaller (2 mm to 4 mm square), with a built-in antenna. The chips could be embedded in a sheet of paper or stuck to any surface, and could eventually be available in a booklet as self-adhesive dots.

"The Memory Spot chip frees digital content from the electronic world of the PC and the Internet and arranges it all around us in our physical world," said Ed McDonnell, Memory Spot project manager, HP Labs.

The chip has a 10 megabits-per-second data transfer rate -- 10 times faster than Bluetooth(TM) wireless technology and comparable to Wi-Fi speeds -- effectively giving users instant retrieval of information in audio, video, photo or document form. With a storage capacity ranging from 256 kilobits to 4 megabits in working prototypes, it could store a very short video clip, several images or dozens of pages of text. Future versions could have larger capacities.

Information can be accessed by a read-write device that could be incorporated into a cell phone, PDA, camera, printer or other implement. To access information, the read-write device is positioned closely over the chip, which is then powered so that the stored data is transferred instantly to the display of the phone, camera or PDA or printed out by the printer. Users could also add information to the chip using the various devices.

"We are actively exploring a range of exciting new applications for Memory Spot chips and believe the technology could have a significant impact on our consumer businesses, from printing to imaging, as well as providing solutions in a number of vertical markets," said Howard Taub, HP vice president and associate director, HP Labs.

The chip incorporates a built-in antenna and is completely self-contained, with no need for a battery or external electronics. It receives power through inductive coupling from a special read-write device, which can then extract content from the memory on the chip. Inductive coupling is the transfer of energy from one circuit component to another through a shared electromagnetic field. A change in current flow through one device induces current flow in the other device.

Memory Spot chips have numerous possible consumer and business-based applications.

Some examples are:

Medical records: Embed a Memory Spot chip into a hospital patient's wrist band and full medical and drug records can be kept securely available.

Audio photo: Attach a chip to the prints of photographs and add music, commentary or ambient sound to enhance the enjoyment of viewing photos.

Digital postcards: Send a traditional holiday postcard to family and friends with a chip containing digital pictures of a vacation, plus sounds and even video clips.

Document notes: A Memory Spot chip attached to a paper document can include a history of all the corrections and additions made to the text, as well as voice notes and graphical images.

Perfect photocopies: A Memory Spot chip attached to a cover sheet eliminates the need to copy the original document. Just read the perfect digital version into the photocopier and the result will be sharp output every time, no matter how many copies are needed, and avoiding any possibility of the originals jamming in the feeder.

Security passes: Add a chip to an identity card or security pass for the best of both worlds -- a handy card with secure, relevant digital information included.

Anti-counterfeit tags: Counterfeit drugs are a significant problem globally. Memory Spot chips can contain secure information about the manufacture and quality of pharmaceuticals. When added to a drug container, this can prove their authenticity. A similar process could be used to verify high-value engineering and aviation components.

Posted by Dal at 06:12 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

NEW FEATURES HAVE BEEN ADDED, YOU WILL FIND THE DETAILS BELOW:

- New Reporting for Earnings : Viewing of Profits, Sells, purchase...
- English correction
- Better support for Internet Explorer
- Signup mail activation
- on Customer interface, new "forget password" feature
- DID import tool
- CARD import tool
- Voip Provider & trunk filtering in call reports
- Add markup & margin on CDR report
- Delete option after selection into card & rates browsing
- New recurring service "auto refill"
- New languages Russian & Portuguese


Project page: http://asterisk2billing.org

WIKI -> http://wiki.asterisk2billing.org/
FORUM -> http://forum.asterisk2billing.org/
DEMO -> http://demo.asterisk2billing.org/


Kind regards,
/Areski
 

Posted by Dal at 06:07 PM | Direct Link | Comments (0) Posted to Asterisk Software

The mesh network utilizes a Radio Optimized Routing Protocol (RORP) to interconnect each unit. Additionally, the integrated storage capacity of each unit can be utilized to remotely and securely store local data files. The low latency, broadband mesh network provides an ideal solution for demanding data intensive real-time applications.

The encryption protocols and web-based authentication ensures that only authorized users are permitted to access the networkThe rugged aluminum weatherproof enclosure, solar module and batteries are capable of operation twenty four hours a day. The unit mounts easily on poles, towers, or rooftops that face the sun. Where there are trees and dense foliage the 913MHz signals will provide much longer range connections compared to WiFi based systems.

Posted by Dal at 01:01 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

AirTight Networks provides a family of solutions to prevent this exploit.

 SpectraGuard Enterprise and SpectraGuard Sentry both prevent this exploit by detecting the connection being established between the hacker’s device and the victim’s laptop - and shutting down this connection, protecting the laptop. SpectraGuard Enterprise addresses the needs of enterprise facilities, while SpectraGuard Sentry is targeted at small branch/remote offices, or homes/home offices. SpectraGuard SAFE is a security agent for endpoints that can be deployed on laptops to prevent the hacker’s machine from successfully establishing a connection with the victim’s laptop. All of these products are shipping and available immediately.
 
“We’ve spent the last few days getting more details and validating that our equipment can indeed prevent these exploits,” said Pravin Bhagwat, chief technology officer at AirTight Networks. “The weaknesses that these two researchers demonstrated are actually not new. Wireless creates a whole new category of security risks and exposures - both for individuals at home and for enterprises. AirTight was founded 4 years ago – to address and prevent these threats. We now have customers around the world who can rest easy knowing that they are protected from these threats.”
 
AirTight Networks' SpectraGuard Enterprise has established its industry leadership position by delivering the key elements of an effective wireless intrusion prevention system (WIPS):
 
detecting and correctly classifying wireless threats - to catch all threats, while minimising false alarms preventing (multiple, simultaneous) wireless threats - while continuing to scan for new ones accurately locating the wireless threats on a floor map - so they can be removed.
 
As shown in the February 2006 Tolly Group report, the March 2006 Information Security Magazine review of WIPS solutions, and the May 2006 Tower Group report - AirTight’s SpectraGuard Enterprise outperforms its competitors in wireless intrusion prevention functionality.

Posted by Dal at 10:30 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

Onvoy Corporate
300 S. Highway 169, Suite 700
Minneapolis, MN 55426
http://maps.google.com/maps?q=Onvoy,+300+Highway+169+S,+55426

Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online
http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda

This month we'll be discussing telephone set options for Asterisk VoIP systems.

Please bring your telephone sets in to share with other members. I'm hoping we all get a change to touch and feel most of the telephones the market has to offer.

Don't miss this meeting!
DOOR PRIZES: I know we're going to be giving away atleast one Polycom telephone set. These baby's are a wonderful addition to your Asterisk hardware collection.

If you're having a problem with Asterisk, bring your questions to a meeting for free help. We love helping new users!

Come to a meeting to meet other asterisk users, see asterisk solutions, win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING!

New visitors can help themselves to FREE FXO Interface cards (So you can connect your phone line, and have a timing source for meetme and IAX protocols). Some members have been known to swap hardware at the meetings.  Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently?

Please come and share your own ideas and learn from others. As always,we'll have food.

We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything.

Meeting starts at 11:30am and parking is available everywhere. Meetings run about 2 hours.

Look forward to seeing you there.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA
http://www.tcaug.net/

If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor.

PS: Eric Osterberg will have a dozen or so Polycom IP500 series telephone sets at a onetime special price only available at this meeting. So bring your checkbooks. This will be a blowout sale on used but refurbished hardware. He has 30 sets available, so contact me if you'd like to have more at the meeting. (New handsets, new cords, new keycaps, lastest software from an authorized polycom dealer)
 

Posted by Dal at 03:35 PM | Direct Link | Comments (0) Posted to Asterisk Development

The network will deliver advanced voice and broadband services throughout MyTel's region, which serves millions of residents, businesses and government
entities.

MyTel has begun its RedMAX deployment in the city of Peshawar, and will expand its WiMAX network to an additional thirteen regions in northern
Pakistan. The network will improve the delivery of voice and broadband services to its existing customers and extend its network to reach more
businesses, residents and municipal organizations.

"By joining the ranks of leading telecommunications companies that are choosing Redline's RedMAX products to roll out WiMAX services, MyTel is
clearly demonstrating its commitment to meeting the communications requirements of its vast customer base," said Dino Bakakis, Redline's Managing
Director for the Middle East. "With our RedPATH architecture roadmap, MyTel can leverage its RedMAX investment to deliver the services its customers need
today and evolve to introduce new WiMAX technologies and devices over time."

"MyTel has a vision to serve the masses with state-of-the-art, innovative and affordable integrated services. For that we choose Redline Communications
as our technological partner," said Nawab Ali, Manager, Business Development, MyTel. "Redline's WiMAX products proved to be the most cost-effective,
flexible, quick to deploy and rugged solution. We are pleased with its performance and reliability and have confidence that Redline will continue to
push the limits of technology with its comprehensive and optimized approach.  Redline has always been very prompt and flexible in analyzing our requirements
and offering a suitable solution."
     

The Redline RedMAX and award winning wireless IP transport products being deployed for the MyTel project are being provided through SARCORP, a Redline
Certified Partner in Pakistan.

Redline's RedPATH Architecture Roadmap
Redline's RedPATH strategic architecture roadmap drives success for WiMAX network operators, enterprises and government organizations. With RedPATH,
Redline is the first to articulate a clear strategy that enables network operators to use WiMAX to deliver the advanced high-capacity and personal
broadband services their customers want today, and to evolve their networks to incorporate new WiMAX technologies and devices as they become available.

Source: Redline 

Posted by Dal at 12:27 PM | Direct Link | Comments (0) Posted to WiMax

Posted by Dal at 12:19 PM | Direct Link | Comments (0) Posted to Vonage

 

This product is designed for people who use instant messaging as a primary form of communication and networking for their social life. The name mylo stands for "my life online" and the communicator lets you use instant messaging, browse the Internet, listen to music, send emails and view photos concurrently.

The device, available in black or white, comes embedded with popular instant messaging services: the Google Talk instant messaging service, Skype and Yahoo! Messenger. These services are free and the product does not require initial computer setup or a monthly service contract.

"The mylo personal communicator puts the fun parts of a computer in the palm of your hand," said John Kodera, director of product marketing for personal communication devices at Sony Electronics. "It's ideal for people who want to stay connected to their online friends and family, but not be weighed down by a PC or buffeted by charges for IM and texting on cell phones."

Posted by Dal at 12:13 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

Astribank-32 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-32 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially “Plug & Play”, unlike any other solution on the market.

Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank.

Click Here for more Information 

Posted by Dal at 12:02 PM | Direct Link | Comments (0) Posted to Asterisk Hardware

The company claims that TETRA-G removes traditional scalability and cost barriers to TETRA deployment for customers deploying either single site or multisite digital radio systems. It features integrated connectivity with land-line telephony and IP network traffic, giving maximum operational flexibility.

The first TETRA-G solution, the TETRA-G Base Station, is the World's first truly scalable single-carrier TETRA solution with integral PBX teleconnect as standard. The Base Station can expand seamlessly from a single base station to a national network.

According to Team Simoco, until now it has not been cost-effective to add PBX connectivity to traditional single-carrier TETRA solutions because the costs of the required switch card are prohibitive. Furthemore, expanding beyond a single-site deployment can often mean the original hardware is rendered obsolete.

The new single-carrier TETRA-G Base Station provides a control channel and three traffic channels, and is supplied complete with a basic rate ISDN telephone interface supporting 2 concurrent telephone-to-radio calls, which can be routed via IP. The Base Station can easily be expanded as more channels are required, to support multiple carriers through the addition of further TETRA-G units.

Team Simoco managing director Mike Norfield said: "This is a key development in the digital radio market. By integrating PBX connectivity and switching into a base station unit, TETRA-G enables easy and progressive expansion from ruggedised, single-site solutions to country-wide deployments while safeguarding customers' investments at every step. It further underlines the progress we are making in developing radio communications systems that deliver total flexibility and real price / performance benefits to users."

TETRA-G system management is through a Windows XP programming package, providing a range of reporting and control options including call logging, fast database downloads and online or offline configuration changes.

The introduction of TETRA-G follows the 2005 launch of Team Simoco's Xfin technology, which converges professional radio and IP-based voice and datacomms. With Xfin, Team Simoco won the Federation of Communications Services (FCS) 2005 Gerald David OBE Award for innovation in business radio.

TETRA-G and Xfin make Team Simoco the leader in analogue and digital radio solutions which allow seamless, fully scalable expansion from one base station to complete multisite systems.

Further TETRA-G solutions will be announced through 2006 and into 2007.

Posted by Dal at 11:54 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Posted by Dal at 11:50 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Digium is the latest amongst a growing list of VoIP-related venture investments, and perhaps the most prestigious. David Skok, a general partner at Matrix Partners who won his VC stripes by backing JBoss, an early Open Source software company joins the board of the company. Digium is one of the early pioneers of the open source telecom movement.

This also marks the latest chapter in the amazing story of Digium founder Mark Spencer and by extension, the Asterisk PBX software. In 1999, Spencer started the Huntsville, Alabama-based operation with his savings and a few thousand dollars in investment from his parents.

Since then, the simple little PBX he wrote has become a rebel yell for telecom hackers, and Spencer a rock-star at VoIP conferences such as VoN. While it is not the only open source PBX, it certainly is one of the most popular.

The company turned profitable in 2002, and remains so. Digium has experienced break neck growth over past few years. Today Asterisk boasts over one million users. Nearly, 1000 copies of the software are downloaded every day. The profitable status of the company, however made me wonder why does Digium need VC dollars?

Spencer explains that the company will use these funds to expand its operations by selling a variety of Asterisk-based IP-PBX systems to small business and large corporations. In doing so, the company might end up competing with others who base their products on Asterisk. Fonality, a Los Angeles-based company, for instance sells Asterisk-based PBX systems to the very same small and medium sized businesses.

“To me, Digium has the same, perhaps even better characteristic than that of JBoss,” says Skok, “It has a highly seasoned product, market leadership, and is profitable.” And it is also one of the hottest VoIP plays around.

Skok may have scored a coup, even if it might have cost Matrix some serious dollars in terms of valuation. Digium, he says has everything to make it one of Matrix’s most successful open source companies. With many old-styled PBX phone systems being replaced by new VoIP-based systems, a low cost offering based on Asterisk can be very attractive for buyers. Fonality, for instance, has quickly garnered 10,000 users for its ultra low-cost Asterisk-based devices.

Though he only has known Spencer for a few months, Skok says his big epiphany came when he downloaded and installed Asterisk on his computer, went overseas, and plugged into his broadband connection. He was soon making calls all over the US, as if he never left his Boston office. The Asterisk became his extension, and he was hooked. “Digium is definitely in a position to become the next big open source company, behind Red Hat, JBoss and MySQL,” says Skok, clearly smitten by his latest investment.

Source: GigaOM 

Posted by Dal at 11:30 AM | Direct Link | Comments (0) Posted to Asterisk News

By implementing the Newfound IP Call Recorder in the Angel.com Call Analyzer feature, Angel.com has delivered one of their customer's most frequently requested features. Angel.com's Call Analyzer provides fully-integrated call recording, capturing caller interactions with IVR phone applications or live agent sessions. With its flexible recording and reporting features, Angel.com customers can record and listen to an entire call, specific portions of a call, browse, search and archive recordings.

"The Newfound IP Call Recorder was a clear choice for our customers.
Newfound's product was the most flexible, robust and cost-effective call recording solution for Angel.com," said Michael Zirngibl, CEO of Angel.com.

"Be it a conference call, an agent assisted call, certain IVR dialogs, or for a specific caller, the Newfound IP Call Recorder delivers full-duplex audio recordings flexibly and reliably. The breadth and capabilities of Newfound's recording solution is impressive."

Now generally available, the Newfound IP Call Recorder 1.0 reliably records any phone call in a carrier, enterprise or contact center telephony environment. Supporting multi tenant architecture, the IP Call Recorder provides ad-hoc (partial call) recording without requiring dedicated telephony ports. Ad-hoc call recording enables the capture and indexing of recordings at any point of a call and for any duration.

Through VoiceXML and web standards, the IP Call Recorder provides developers the means by which to capture and display each recording accompanied by call related data such as caller ID, destination number, account number, DTMF or speech recognition inputs, call errors, hold times or transfer number. The Newfound IP Call Recorder is a software-based solution working on standard Intel-based Windows servers in both legacy PSTN and VoIP (SIP) telephony environments.

"We are pleased that Angel.com has selected our most recent addition to Newfound's advanced telephony product line in order to support its on-demand call center and IVR solutions," said Kris Hopkins, CEO of Newfound Communications, Inc. "We look forward to supporting Angel.com as they continue to deliver innovative, reliable and easy to use services to their customers."

About Angel.com
Angel.com is a leading provider of on-demand call center and Interactive Voice Response (IVR) solutions, which enable organizations of all sizes to quickly deploy powerful telephony applications. More than 1,500 customers turn to Angel.com's patented Voice Site technology to power customer service and marketing phone numbers using intelligent speech recognition that can automate most phone-based interactions. With an innovative Internet-based solution that requires no investment in hardware, software, or human resources, Angel.com balances the need for high quality communications with affordable pay-as-you-go pricing.

About Newfound Communications, Inc.
Newfound Communications, Inc. offers technology, tools, and expertise to fuel the speech recognition revolution. By working with Newfound, companies and developers bypass the largest impediments to creating reliable and successful IVR applications. Newfound is the leading source for the most innovative speech solutions utilizing VoiceXML & VoIP. With the MediaMixer [VoiceXML edition, Outbound Call Queue Engine and IP Call Recorder products, automated customer service has never been easier. For more information visit, the Newfound website: www.newfoundcomm.net.

Posted by Dal at 11:22 AM | Direct Link | Comments (0) Posted to VoIP Software

The software is ready for download at: http://www.escaux.com/netpbx

ESCAUX net.PBX runs directly from CD, requires no installation and does not overwrite your harddisk. Your configuration changes are centrally stored on our servers and become instantly available at every system reboot. Simply place our Smart Live CD in your PC, follow the Configuration Wizard and show-off to friends and colleagues with your "Web controlled, Asterisk based, Business IP PBX".

The net.PBX Free Edition is a full featured IP PBX system suitable for Business use. You can install and run the Free Edition on your own servers and benefit from a close to zero cost IP PBX system for your company. On Demand commercial support packages are available to assist you with installation or configuration but you can also refer to our net.PBX Community Forum.

The ESCAUX Free Edition is an Open Source project hosted at SourceForge.

At any moment in time you can decide to migrate towards a Commercial version of our product and benefit from our unmatched Service Level Agreement, Guaranteed response and Repair times.

Best regards,

the ESCAUX development team
 

Posted by Dal at 11:19 AM | Direct Link | Comments (0) Posted to Asterisk Software

Call up your contacts using your favorite web browser from anywhere and click on the contact you want to call. Then, presto! AsteriDex first calls you and then connects you to your contact through an outbound call made using your Asterisk server.

AsteriDex also can automatically look up CallerID Names in your MySQL database for incoming calls. And, it can be used as a garden-variety speed dialer from any telephone by simply spelling up to five characters of any contact's name. It'll even tell you who is being called. And, yes, it's FREE!

Click Here for the Full Nerd 

Posted by Dal at 10:56 AM | Direct Link | Comments (0) Posted to Asterisk Help

Called the iQ Voice Management Suite, small and mid- size companies can now be assured that their VoIP- enabled voice service will continually operate at the quality expected to conduct business. Designed to assess and monitor services over converged networks or dedicated circuits, the iQ Voice Management Suite delivers higher voice quality and QoS management than ordinary IP Centrex services.

Pandora offers three service options to fit the right network and budget: iQ Assess, iQ Manage, and iQ Direct.

iQ Assess is designed for converging VoIP on top of an existing data network. It performs automated network assessments on customer premise networks as well as their Internet service providers to ensure a high-quality available connection direct to Pandora Networks' data center. This assures best practices when coupling Worksmart services to third-party carriers and customer premise equipment. iQ Assess is ideal for small companies that cannot justify the cost of dedicated T1 circuits but want the same quality of service that dedicated circuits offer.

“Unlike typical VoIP assessments, iQ Assess provides insight into both local and wide area networks to ensure end-to-end quality of service all the way to Pandora's data center,” says Will Lombard, of Seventhman, a Pandora reseller. “Since the assessment appliance is always on, additional assessments are available whenever it's needed.”

iQ Manage not only performs on- demand network assessments, but also includes real- time QoS monitoring services and a 24/7 technical support team to offer assistance if a problem is detected. An important advantage since many business networks evolve rapidly and can experience congestion.

“No other hosted VoIP service provider offers this level of QoS management or technical support at any price,” says Walter Snell, CEO of Pandora Networks. “Since today's business networks are so dynamic, iQ Manage is ideal for rapidly recognizing any topology or traffic changes that will have an immediate effect on VoIP service quality.”

iQ Direct goes one step further by providing dedicated T1 circuits specifically for VoIP communications. This private circuit does not route to the public Internet, but instead moves IP traffic directly to Pandora Networks' nearest data center for the highest service level possible. iQ Direct is best suited for larger organizations with fifty or more employees. Powered by XO Communications' network infrastructure, iQ Direct can be delivered in nearly every market in the United States.

All services are sold on an annual contract basis with prices starting at $495 per year.

Source: Pandora Networks 

Posted by Dal at 10:55 AM | Direct Link | Comments (0) Posted to VoIP Software

"The role of commercial and municipal wireless networks is expanding beyond basic Internet access and e-mail services to include more real-time applications such as video surveillance, voice over IP and mobility," said Bo Larsson, CEO of Firetide. "Our new access points deliver a new level of performance and features that are now becoming real requirements for large scale, multi-service networks."

The HotPoint access points enable highly flexible deployment with or without a Firetide mesh network. The modular architecture allows Firetide Mesh nodes and access points to be mounted independently to achieve optimal radio coverage for both mesh backhaul and client access. High powered, 400 mW radios extend reach and penetration, while a turbo-mode provides data rates up to 108 Mbps. A virtual access point function provides layered service and security levels for individual APs and virtual AP groups.

Firetide is currently shipping both indoor and outdoor HotPoint access point models. The outdoor model has a weatherproof enclosure that can be mounted to a HotPort mesh node or used as a stand-alone access point. Power can also be supplied directly by the mesh node eliminating the need for a separate power supply. Both models share the same access point features to enable consistent service coverage inside buildings as well as in outdoor areas. These features include:

    -- Advanced controls for HotSpot service providers with customizable
       group-based service levels, user-based rate limiting, and inter/intra-
       cell blocking.
    -- Comprehensive security, including standards based encryption and
       authentication controls including 802.11i, WPA2, 802.1x, and RADIUS for
       interoperability with any .11b/g Wi-Fi client.
    -- Additional access control via MAC address filtering, VPN tunneling and
       filtering, multiple VLAN support, ESSID suppression, and firewall
       application control.
    -- Auto-discovery and access point groupings for efficient configuration
       and management of networks with large numbers of access points.
    -- Single, centralized management for both mesh and access point networks
       with complete HotView and HotView Pro integration, web based GUI, SNMP,
       or CLI interfaces.

Current Firetide access point customers include North American Midway which uses the access points for wireless ticket sales for their traveling amusement parks and Sky Catcher which provides high-speed broadband solutions for rural and under-served areas.

"We chose Firetide's access points because they support all of our government and enterprise access requirements," said Peter Gaulin, CTO of Sky Catcher. "The tight integration of the access points, mesh network, and centralized management creates a very solid and secure wireless network environment that is also very easy to deploy and manage."

Posted by Dal at 10:53 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

Posted by Dal at 10:51 AM | Direct Link | Comments (0) Posted to SIP

Switchvox SMB is built from open source software and uses open standards. The IP PBX allows companies to make calls using traditional analog lines, as well as VoIP services. The system can be peered to allow free VoIP calls between office locations, in addition to offering comprehensive reporting tools to monitor the solution's performance in real time.

The agreement will allow VoIP Supply to deliver Switchvox to end users and resellers in a faster and more flexible way than other solutions because of Switchvox's turnkey software architecture and instantaneous software delivery infrastructure. VoIP Supply will also be able to use any handsets its customers request and further leverage its distribution arrangements with leading IP handset manufacturers. The new VoIP Supply site provides SMBs with IP PBX systems like Switchvox SMB.

"VoIP Supply has one of the largest online reseller communities in the industry and this new sales channel will open the door for increased revenues for Switchvox," said Joshua Stephens, CEO of Switchvox. "We are excited to be involved with the launch of PBXselect.com and look forward to developing a long-term relationship with VoIP Supply to provide innovative IP PBX solutions tailored for the SMB market."

"We found the Switchvox product to be one of the best IP PBX solutions for the SMB audience," said Garrett Smith, Director of Marketing and Business Development at VoIP Supply. "The addition of the Switchvox SMB to our product catalog will allow us to continue our trend of delivering the most innovative VoIP products and services to our customers."

Switchvox products are sold as turnkey solutions that include the server hardware and pre-installed Switchvox software. Switchvox SMB is available starting at $2,495. Further information on Switchvox SMB can be found at www.switchvox.com. Information on how to purchase the product can be found at www.VoIPSupply.com.

Posted by Dal at 10:48 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Talks at the show explained just how easily an Asterisk-based PBX can be used to launch attacks, notably "vishing" attacks, in which hackers use VoIP calls instead of phony Web links to steal personal and financial information.

Asterisk has become the hacker's favored tool because it's free, easy to use, and works with cheap, off-the-shelf hardware. Install Asterisk on an inexpensive PC, do a little tweaking, and you've got a full-blown PBX, something that previously would have been extremely expensive and time-consuming to do.

A vishing attack is simple to launch using Asterisk. War-dial using an Asterisk-based PBX, and send a recorded message to thousands of people, telling them their credit card number has been stolen, and that they need to call a phone number to solve the problem.

The number, of course, is the Asterisk-based PBX set up by the hacker. An automated message tells them to enter their credit card number and other personal information, for verification purposes. The PBX records the number and information, and the hacker now has a credit card to use.

Other hacks can be launched from Asterisk as well. There's the "man-in-the-middle" attack, in which a PBX-initiated call lures someone into calling a bank, credit card company, or other financial institution. The PBX answers, and forwards the caller to the real customer service number --- and then listens in and records the entire call. Again, the hacker comes away with personal and financial information he can use.

The upshot? Just as you shouldn't trust any unsolicited email, you also shouldn't trust any unsolicited phone calls. Asterisk-based vishing and similar attacks make fraud too easy these days.

Posted by Dal at 10:46 AM | Direct Link | Comments (0) Posted to Asterisk News

Digital Hollywood is the premier entertainment and technology conference in the country, with over 15,000 top executives attending each year.  Building Blocks 2006 is expected to attract more than 2,000 top industry executives to the event.

Lewis, who wrote the patent for the first Softswitch and SS7 Media Gateway, is a recognized industry pioneer and technology innovator. He co-founded CLEC XCOM Technologies, Inc., at the time of the Telecommunications Act in 1996 and directed its acquisition by Level 3 in 1998. His next venture, set-top box vendor River Delta, sold to Motorola. His third successful venture, Caerus, Inc. and its three
subsidiaries: Volo Communications, Caerus Networks, Inc., and Caerus Billing & Mediation, Inc. empowered carriers and service providers to begin selling advanced services and realizing revenues and profits immediately. The market enthusiastically responded to Caerus' approach and service offerings, and VoIP, Inc. acquired Caerus, Inc. in 2005.

"We've always pushed the envelope to develop the most advanced technology and to create applications with exciting features and functionality that revolutionize the industry," Lewis stated. "For instance, we just launched the world's first free web-click phone-to-phone calling service, and currently develop leading-edge applications for high-profile customers in this space," he added.

Interested parties are invited to test some of VoiceOne's latest applications -- free of charge -- in the company's Virtual Lab, http://Labs.Voiceone.com. Labs.Voiceone.com offers an open forum for users to review, test free of charge and provide feedback on new IP communications services developed by VoIP, Inc's subsidiary, VoiceOne, and to discuss their ideas with VoiceOne product and service developers.
 

Posted by Dal at 10:42 AM | Direct Link | Comments (0) Posted to VoIP News

Although plenty of carriers and networking providers are in the midst of WiMAX trials, all of which are promising, the future of the technology, in North American markets at least, is still an open question. "There's just no infrastructure supporting WiMAX today," said Craig Mathias, an analyst with research firm Farpoint Group. Plus, there's lots of existing capacity from cellular providers and metro-scale Wi-Fi networks already in the air, said the self-described WiMAX curmudgeon. "Converged Wi-Fi and cellular networks also give the best of both worlds."

Others, such as Dave Park, vice president of product development for BelAir Networks, are staying on the fence about it for now. "You have to have a network to get users, but if you have no users then it's hard to [build] a network," he said. The WiMAX Forum trade group has another take. For starters, WiMAX is no newbie technology. The trade group counts more than 150 fixed WiMAX (802.16d) deployments and trials worldwide. Unlike Wi-Fi, which operates over the unlicensed communications spectrum and is designed for networking at short distances, WiMAX operates on licensed bandwidths over multiple communications frequencies. It is designed for wide scale broadband deployments.

The latest version is a mobile protocol called 802.16e. It provides broadband-class wireless coverage across large areas where user congestion is an issue. But Wi-Fi can also operate over wide areas when deployed as part of a "mesh network" that ties together multiple Wi-Fi systems and access points. However, Wi-Fi's performance can be impacted by the number of users on a network and physical barriers -– moreso than WiMAX, experts say. WiMAX can also be more aggressive than its smaller cousin Wi-Fi about signal congestion. "Wi-Fi fundamentally handles congestion in a very brute force way," explained Greg Caltabiano, president and chief operating officer at SOMA Networks, a broadband services solutions provider.

Whereas Wi-Fi is designed to back off and randomly try again when it bumps into conflicting radio signals, WiMAX bullies its way through such communications congestion. After all, WiMAX operates a lot like cellular, providing every subscriber with a specific set of frequencies instead of time slots. That's why it's called frequency division multiplex technology.

Ultimately, cellular technology may be hardest hit by mobile WiMAX, since 3G technology operates at much slower transmission speeds than WiMAX, said Ellen Kirk, vice president of marketing for Tropos Networks, a provider of mesh (define) networking technology. For example, the latest version of CDMA 2000 promises speeds of up to 1.8 megabits per second, she noted. "But you won't get that and will probably average 1 mbps.

It's fast, but still not as fast as the promised 5 mbps average speed of WiMAX, she added. "Clearly the speed of 3G mobile networks is going up, but speeds and bandwidth are not enough to hit broadband performance," noted SOMA's Caltabiano. This is why "mobile WiMax is really a direct competitor to mobile cellular, and we will soon be seeing a clash of titans," he added. For all the WiMAX testing underway, however, timetables for commercial deployments are hard to peg.

Right now, experts say, fixed WiMAX technology is also not compatible with the coming mobile WiMAX protocol. This is a major reason why Motorola decided to skip right over 802.11d fixed WiMAX deployments and turn its attention to the mobile flavor. The company has also put its money where its mouthpiece is by kicking in about $300 million to invest in Clearwire Corp., a wireless broadband service provider.

Intel upped he ante by adding $600 million to that investment, with the idea of positioning Clearwire as a first user of its mobile WiMAX chips, which also support fixed deployments. "Consumers and operators can use these initially for a fixed application, while they debate whether they want to provide full mobility as well," said Raghu Rau, Motorola's senior vice president of networking and enterprise strategy. Motorola is also planning a mobile WiMAX trial in partnership with Japan broadband service provider Softbank, slated for that country in September.

The goal of the five-month project is to eventually offer wireless WiMAX as an option to Softbank's 26 million fixed line and mobile communications customers. "The purpose of the trial is to see how WiMAX performs in a mobile environment and a dense urban deployment," Rau told Internetnews.com. Motorola has even built a handful of prototype mobile WiMAX handsets for use in the trial, a first for the fledging industry.

Still, WiMAX has a ways to go before it's ready for prime time as a commercial broadband alternative. The general formula for success is to first establish a fixed WiMAX network, then attract users to that network and build out the infrastructure, and finally use that revenue and infrastructure to launch a mobile WiMAX system, according to networking experts interviewed for this story.

This can be a complicated and risky approach. "Why would we build a network if there isn't anything to connect to it?" said Tropos Networks' Kirk. "That's the advantage of Wi-Fi." In order to offer a high-bandwidth pathway to lots of users across a wide area, WiMAX needs many small cell sizes that work together. This is what Wi-Fi is already doing with mesh networks, said Kirk. "We're not anti-WiMAX, just pro-reality."

Source: InternetNews 

Posted by Dal at 04:05 PM | Direct Link | Comments (0) Posted to WiMax

VoIP security is a key theme in this week's Las Vegas events. In a last minute change to DEFCON scheduling, Phil Zimmermann will give an update on his Zfone secure VoIP software.  Zimmermann, the creator of the Pretty Good Privacy (PGP) e-mail software encryption tool, noted that there's a lot of demand for a secure VoIP client these days. "News about government eavesdropping tends to generate interest [in Zfone]," he said in off-the cuff remarks made in the BlackHat press room. "I prefer to talk about the need to secure enterprise communications from criminal elements." However, he referred to Zfone as a "psychic drain" due to the amount of time and energy it is taking.

A number of vendors are likely to get heartburn when Hacking VoIP Exposed hits bookshelves in December 2006. Authors David Endler,  Director of Security Research at TippingPoint and Mark Collier, of CTO SecureLogix, exposed a variety of techniques for identifying weaknesses in VoIP networks. In some cases , there's no protection: attackers can listen to the pre-recorded messaging embedded in a typical IP PBX to identify the manufacturer, then plan their next actions accordingly.

IP phones from one manufacturer have the ability to capture and store packets, a useful feature for debugging but also one that can be exploited to remotely record conversations if the phone password is compromised. Another manufacturer recommends that, for ease of use, IP phone passwords be set to the reverse of their assigned numeric extension. Needless to say, such elementary passwords schemes are among the first to be tested by an attacker.

Posted by Dal at 04:03 PM | Direct Link | Comments (0) Posted to VoIP Security

The high-speed service features average download speeds of 450 to 800 kbps and average upload speeds of 300 to 400 kbps, bringing the Sprint service into competition with DSL technology.

Sprint said it expects to reach over 200 million people in 220 major metropolitan areas across the U.S. by the end of 2006 with the service, which it has dubbed Power Vision. Sprint said it expects the entire Power Vision network to be completely upgraded to EV-DO Rev A by the third quarter of 2007.

Sprint said it completed the first EV-DO Rev A data connection in San Diego last week. Sprint noted that it has demonstrated a range of applications including IP video telephony, high performance push-to-talk service, multi-service video conferencing, real time gaming and video streaming.

"We have taken this technology out of the laboratory and are currently integrating it into our network," said Kathy Walker, Sprint's chief network officer, in a statement.

Source: TechWeb 

Posted by Dal at 03:59 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

CallRex Agent Evaluation is specifically focused on maximizing the bottom-line of informal call centers through agent coaching and continuous learning; until CallRex Agent Evaluation, similar tools have simply been too complex and expensive for smaller businesses.

CallRex Agent Evaluation - Three-Step Process

Evaluate: The three-step CallRex Agent Evaluation process begins with a manager evaluating an agent's phone call against pre-defined criteria, also known as a standard. Each standard identifies the key agent activities that accomplish a call center's various objectives. As managers evaluate a call, they can attach excerpts of the recorded call to the evaluation, providing an agent with the ability to listen to specific examples of their performance.

Coach: Next, the manager and agent review the evaluation together in an interactive coaching session. The coaching session is critical for the agent in understanding their strengths as well as areas requiring improvement. In the coaching session managers use CallRex Agent Evaluation to highlight actual examples from calls that illustrate key learning points. The coaching session ends with a customized coaching agreement between the agent and manager setting specific performance goals. Through the use of measurable feedback and specific goals, CallRex Agent Evaluation requires agents to actively invest in their own training and growth. The coaching agreement becomes the basis of the next regularly scheduled coaching session as the three-step process repeats itself.

Measure: As the three-step CallRex Agent Evaluation process is repeated regularly across all agents in the call center, improvements in agent call handling skills quickly become visible. By analyzing results over time and across multiple agents, call center managers gain unique insight into performance trends and agent training needs. This knowledge is particularly important to informal call centers that typically do not have a high ratio of managers to agents.

A police department in the United Kingdom uses CallRex Agent Evaluation's three-step process to evaluate their agents' ability to listen carefully to callers and quickly establish the nature of a call; these abilities are critical to responding effectively to emergency calls. By analyzing their results with CallRex Agent Evaluation's reports, the police department subsequently identified and addressed a need to train agents in remaining calm during stressful calls.

"In the past, evaluating agents' performance was a complex and time consuming process. With CallRex Agent Evaluation the entire process has been optimized to three simple steps," says Robert Kapela, president of Telrex. "CallRex Agent Evaluation is another example of how Telrex is bringing high-value applications that were once only available to large call centers to small and medium businesses."

CallRex Agent Evaluation is integrated with CallRex Professional and joins CallRex Multi-Media in expanding the power of CallRex Professional in the formal or informal call center. CallRex Multi-Media enables computer/Internet monitoring, giving call center managers the ability to monitor and record everything agents do on their computers in addition to what they say on the phone.

For more information on CallRex Agent Evaluation and a free evaluation download, please visit www.telrex.com/callrex_ae.htm.

Posted by Dal at 03:56 PM | Direct Link | Comments (0) Posted to PBX / iPBX

Barix Instreamer and Exstreamer systems offer unparalleled scalability and cost-effectiveness for IP audio decoding, distribution, encoding and monitoring compared to many competitive systems by using IP links and standard protocols for audio streaming and delivery.

“The Barix Instreamer and Exstreamer hardware platforms offer an inexpensive, reliable and proven way of distributing real-time and on-demand audio over standard IP network infrastructure for a variety of audio applications.” said Johannes Rietschel, CEO of Barix. “From live broadcasting to in-store streaming to audio distribution through large building, hotels and airports, a growing number of professionals are leveraging mature IP technologies to achieve inexpensive, reliable audio transmission over an existing IP infrastructure. This allows businesses to vastly reduce costs incurred using satellite or leased-line based technologies, and offers far better control and monitoring.”

The Instreamer-100 will receive its worldwide debut at IBC and be displayed in an end-to-end audio over IP streaming with the Exstreamer-100. Like previous Instreamer products, the Instreamer-100 encodes audio from analog and digital devices into the MP3 format in real-time for point-to-point or multipoint delivery over the Internet or IP-based systems. The serial port in the Instreamer-100 allows the operator to remotely control additional devices over the same network connection from any location.

The Exstreamer-100 decodes the audio at the receiving address. Like previous Exstreamer models, the Exstreamer-100 is an intelligent, network based audio decoder that pulls digital audio from an IP network and converts it to music or voice. Unlike previous Exstreamer models, the Exstreamer-100 offers redundancy through automatic switchover to backup streams or USB playback if the main stream is unavailable.

The flexibility of the end-to-end Instreamer/Exstreamer platform means that any number of commercial businesses can benefit from this at a fraction of the cost compared to competitive alternatives. Professional radio broadcast groups can cost-effectively encode, monitor, and decode audio for on-air operations at multiple stations, or prepare podcasts to reach new audiences.

The Instreamer-100 and Exstreamer-100 are extremely valuable in both point-to-point and multi-room applications at a single location, such as a hotel, restaurant, cruiseship or airplane, or point-to-multipoint situations for a large chain of stores. For the latter, store chains can use the end-to-end platform to broadcast audio and advertisements from a central location to the entire chain, without PC infrastructure in the stores or studio. Since every store can receive its own real-time audio feed, the performance and function of the system can be easily monitored – features not possible with the less flexible and more expensive satellite-based streaming.


Barix also offers fully customer programmable software versions of the Instreamer-100 and Exstreamer-100. This allows system integrators and OEMs to develop their own applications in a network-, audio-, or USB-flash environment such as scheduling announcers, call sign generators, or network or I/O triggered alarming devices with supervision. Barix products feature open standards and integrate well with other software and equipment such as distribution servers, media players and IP phone systems for paging and background music applications.

Posted by Dal at 11:03 AM | Direct Link | Comments (0) Posted to VoIP Hardware

In related news, Alpha Networks expects its revenues to record sequential growth in the second half of this year, up 15-25% from the NT$9.91 billion (US$301 million) it posted in the first half, company chairman John Lee said yesterday, at an investor conference.

Sales of the company's Gigabit switches, digital phone products, Gigabit Ethernet-passive optical network (GE-PON) devices and VDSL 2 customer premise equipment (CPE) products are expected to sustain significant growth in the second half, Lee asserted.

However, the company said that its after-tax profits of NT$144 million for the second quarter of this year were down 32% sequentially from the NT$212 million recorded in the first quarter due mainly to rising raw materials costs and price competition in the market.

Alpha Networks' gross margin stood at 13.39% in the second quarter, down from the first quarter's 15.3%, while the company recorded an EPS (earnings per share) of NT$0.43 in the second quarter, as compared to NT$0.64 for the first quarter, the company said.

Posted by Dal at 10:53 AM | Direct Link | Comments (0) Posted to WiFi / Wireless

Labs.Voiceone.Com offers an open forum for users to preview, test free of charge, and provide feedback on new IP communications services developed by VoiceOne. VoIP, Inc.'s CTO Shawn Lewis said, "Cick4Me is designed to meet the needs of active, on-the-go businesspeople. It integrates the World Wide Web, VoIP calling, mobile telephony and landline functionality all within one product. It even enables users to connect with others without giving out their phone number."

To access the Click4Me.Net service, simply log on and register at http:// Labs.Voiceone.Com or at www.Click4Me.Net Users of the service can connect from their phone to other users' phones free of charge simply by entering their e-mail address or Click4Me.Net user name. Users can then review, provide feedback on the Click4Me service or discuss additional IP communications services with VoIP Inc.'s development team. Added Lewis, "Users are not required to sign up for any type of VoIP phone service or use Instant Messaging or any special programs to make and receive phone calls through this service.

It's also customizable and has numerous exciting features and functionality, including working with cell phones and the ability for the service to be embedded into e-mails." VoIP, Inc.'s President and CEO Gary Post commented, "Click4Me.Net is one example of the many ground-breaking services being developed by our VoiceOne subsidiary that make creative use of VoIP technology and demonstrate our ability to deliver innovative solutions tailored to meet the fast-changing needs of our customers.

The services being developed by VoiceOne represent the future of telephony, and we are making them available today to consumers free of charge through our Virtual Lab." Registered Click4Me users maintain their settings online and can set their availability status to Away, Do Not Disturb and Available.

Registered users can also change the unpublished number at which they wish to be reached. Additional features allow users to apply custom filters for blocking specific callers or require special PINs to complete calls. Cell phone users can also connect to Click4Me users by navigating to the web interface for cell phones, http://www.click2me.us, and entering the user's e-mail address or Click4Me ID. There is no charge for the call.

Click4Me technology can be integrated into emails, web pages, web advertisements and more by embedding a simple hyperlink tag like "Call Me Now." Users who click on the tag are automatically directed to a web page directing them to the intended party. VoiceOne plans to add additional Click4Me.Net features over the coming weeks, including voicemail, text message notifications, and text message dialing.

Posted by Dal at 12:37 PM | Direct Link | Comments (0) Posted to VoIP News

New analysis from global growth consulting company Frost & Sullivan , Asia Pacific Residential Telephony Market, reveals that revenues in this industry -- covering 13 major Asia Pacific economies -- reached its peak in 2005 at USD51.58 billion, with a subscriber base of 415.1 million. While subscriber base is forecasted to grow at a CAGR (compound annual growth rate) of 2.3 percent (2005-2012) to reach 488.1 million in 2012, total revenues from residential telephony expects to wane henceforth to be worth only USD45.0 billion by end-2012.

If you are interested in a virtual brochure, which provides service providers, vendors/manufacturers, end users and other industry participants with an overview of the latest analysis of the Asia Pacific Residential Telephony Market, then send an email to Letticia Leopold, Corporate Communications at letticia.leopold@frost.com with your full name, company name, title, telephone number, fax number, and e-mail address. Upon receipt of the above information, an overview will be sent to you by e-mail.

"While most Asia-Pac countries are past their peak in residential telephony, selected markets characterized by large population and low household penetration rates, such as China, India, Indonesia, the Philippines and Thailand are still experiencing growth," explains Sin Siew Teyew, head of telecoms research at Frost & Sullivan. "These countries are however typically restrained by underdeveloped infrastructure and relatively low ARPU (average revenue per user) levels." On the other hand, certain pockets for revenue growth still exist in the more mature residential telephony markets across the region.

"While revenues from voice are declining, there is still potential for growth in other portions of the business such as VAS and interconnection. Service providers would however need to understand market peculiarities and demands well," adds Teyew. The threat of fixed-mobile substitution (FMS) is significant in advanced broadband markets where it affects the ARPU, while IP telephony affects the subscriber base.

However, in markets with high mobile and low broadband affordability, FMS usually affects ARPU and subscriber base, resulting in revenue loss from end users and hampering the potential for higher growth. "End users in high mobile and broadband penetration markets rely less on public switched telephony network (PSTN). Users would still however maintain the subscriptions for broadband access," explains Teyew. "In countries with low broadband penetration, end users seem to have higher preference for mobile phones for voice communications and are prepared to do without traditional residential telephony altogether." Service providers need to invest significant effort to prevent residential telephony from declining.

Improving VAS or increasing the interconnection rates is likely to impede the decline in revenues or subscribers in the short- to medium-term. In the long-run however, service providers would almost certainly need to accept IP telephony as well. The Asia Pacific Residential Telephony Market study is part of the Communications Services subscription. It evaluates the growth and sustainability of PSTN- and integrated services digital network (ISDN)-based technology. The study also examines related aspects in the residential telephony market in the region, such as the impact of IP telephony or other VoIP services, FMS, and the impact of regulatory and government policies on the residential telephony services market. Analyst interviews are available to the press.

Posted by Dal at 11:12 AM | Direct Link | Comments (0) Posted to VoIP Politics

This gateway is a strong addition to the IP Gear/Boscom range of products and means we can now offer all types of PBX connections from analogue, Basic Rate, Primary Rate and now VoIP'

The CelluNet provides a bridge between the VoIP SIP world into the mobile GSM network. When users make a number of mobile calls from an IP PBX then there are considerable cost savings to be made by using a GSM gateway. Users can use a tariff for GSM to GSM calls which can give a cost saving of up to 40% over landline to GSM calls and in some cases for a user group calls can be free for mobile to mobile.

The CelluNet offers other advantages as it can also be used as a backup for ISDN lines or can be used in a temporary environment such as a building site office where lines may take a considerable time to be provisioned, the gateways support incoming or outgoing calls and callers can dial direct to a user.

The CelluNet is part of the IP Gear range of products and is actually the first product that bridges their two product ranges of VoIP and GSM gateways and opens the market to all sorts of possibilities.

Posted by Dal at 11:09 AM | Direct Link | Comments (0) Posted to VoIP Hardware

Currently, it has been used (making and receiving phone calls) and  developed in an ITU SS7 environment, but I have a good chunk of the code included which is required for ANSI support as well.  I think I'm going to get an ANSI link in a few weeks, so hopefully I'll have that  tested and working relatively soon.

It supports MTP2, MTP3, and ISUP.  After I get these layers fleshed  out, I'm planning on starting on SCCP and the layers above that with the eventual goal of database-lookup and SMS support.

To test, you must have a T1/E1 card as well as an SS7 link.  You also need to have zaptel installed on your system.

Here are the instructions for checking it out of subversion and getting it working:

`svn co http://svn.digium.com/svn/libss7/trunk libss7`
`cd libss7`
`make install`

Right now, the changes to chan_zap are implemented in a special developer branch of asterisk. These are the instructions to check it out
`svn co http://svn.digium.com/svn/asterisk/team/mattf/asterisk-ss7
asterisk-ss7`
`cd asterisk-ss7`

If you haven't compiled trunk yet, you may have to run `make` a few times so that the configure script runs and sets things up properly.  It should find libss7, and compile chan_zap with support for it.  The link is brought up automatically when Asterisk starts.

Configuration in zaptel.conf is similar to that of a PRI.  Your signalling channel will be set as a "dchan" and the bearer channels are set as "bchan".  For information about setting up zapata.conf, see the sample zapata.conf in the configs/zapata.conf.sample in the asterisk-ss7 branch.  There also is a libss7 project section on Mantis now for any bugs that you might encounter.

Matthew Fredrickson
 

Posted by Dal at 10:55 AM | Direct Link | Comments (0) Posted to Asterisk Development

He joins co-founders, Walter Snell, CEO and Sridhar Muppidi, CTO. Both co-founders have deep experience in IP convergence solutions before they founded Pandora Networks in 2001.

Turner brings extensive telecommunications-industry experience to the company and as Executive Chairman, will lead the company's aims and policies. Turner is also responsible for providing leadership to the company's management team to ensure effectiveness in setting and implementing corporate direction, product strategy and meeting fiscal goals.

Yankee Group anticipates the business VoIP market will reach $3.3 billion in service revenue by 2010. Businesses are migrating to VoIP because it provides cost savings, a converged platform for voice and data, and improves the manageability of communications.

“Hal is a seasoned veteran in global telecom with a proven track record in growing successful companies,” said Walter Snell, CEO of Pandora Networks. “We are poised for explosive growth as more and more resellers and service providers discover the benefits of reselling services to their SMB customers who demand affordable and easy-to- use IP communication systems. Hal's expertise and leadership will be invaluable to our team.”

A telecommunications industry veteran with almost three decades experience, Turner founded and oversees Turner Telecom Holdings, an executive management and board services consultancy, where he utilizes his international management experience to accelerate the growth of his clients, which have included global technology and telecom companies, startups, turnarounds, high-growth ventures and multinational corporations.

“Pandora's integrated voice, video, messaging and collaboration platform is unprecedented in the industry and delivers tremendous benefits to its users, unable to be serviced by monolithic providers,” said Turner. “I'm thrilled to be part of a company that makes unified communications a reality to the SMB customer and I look forward to the opportunity to guide the company toward this bright and promising future.”

Prior to joining Pandora Turner was Executive Chairman of LastMile Communications, a London- based provider of secure, edge of the network wireless content delivery solutions. There he was successful in setting the company's go-to-market strategy as well as secured series A and B financing rounds. Prior, Turner led the merger of IP Deliver with Proficient Networks to form InfiniRoute, the industry's first carrier-neutral, fully-managed service, managing VoIP routing for wireline, wireless and emerging carriers. There as chief Executive of the Board, he was instrumental in the firm's three rounds of raising capital.

Turner also served as Chairman and Group Chief Executive, of Davnet Limited, and has held executive positions with EON Corp, TeleZone Inc., PTT Telecom Netherlands, US, Inc., and BellSouth Communications, Inc., where he was President and Chief Operating Officer. Turner began his career at AT&T Corporation, where he held senior roles in sales and marketing.

Turner earned a BS and an MBA from University of South Carolina.

Posted by Dal at 10:36 AM | Direct Link | Comments (0) Posted to VoIP News

In order to provide customers the best solution for VoIP, video surveillance and Wi-Fi over mesh, Firetide has included the following features and capabilities to support a growing number of applications:

-- Performance: The Firetide mesh now features flow-based routing for up to 32 Mbps throughput. A comprehensive set of new bandwidth metrics, including link quality and capacity, link type, and hop count enable flexible management and intelligent routing decisions. Other performance enhancements to ensure consistent service include load balancing, static routes, and broadcast containment.

-- Quality of Service: Expanded QoS and prioritization, including Layer 2 802.1p, optimize the network for multi-service offerings including latency sensitive voice and video applications.

-- Network Security: In addition to its existing encryption, packet encapsulation and filtering features, Firetide has added WPA2 encryption, digital certificates, firmware signatures and expanded password protection for multi-layer enterprise-class security.

-- Mobility: Mobile mesh nodes can now roam seamlessly throughout a Firetide mesh with zero-packet loss and zero-handoff delay. Firetide's new 3.3 software, coupled with the HotView Pro 3.3 management tool, provides a flexible, secure and easy to manage network

-- regardless of scale and size. Beta customers such as North American Midway, AdSpace Networks and Rockford Housing Authority have tested Firetide's wireless mesh with the new software offering: -

- Rockford Housing Authority in Rockford, Illinois uses Firetide's mesh with HotView Pro 3.3 software for wireless video surveillance in high- crime areas. The network cameras have already provided significant decreases in criminal activity for the community. -- North American Midway, a company that specializes in traveling amusement parks utilizes Firetide's mesh with 3.3 software as a park- wide network for wireless ticket sales.

-- AdSpace Networks adopted Firetide's mesh with 3.3 software for audio/video digital advertising networks deployed in shopping malls in Chicago and Connecticut. "We are continually setting up the network in new towns with unknown conditions," said John Gallant, IT manager for North American Midway. "Version 3.3 has made our system a lot faster in addition to being extremely reliable. The new software has a link elimination feature that allows me to tweak and tune the Firetide network to achieve peak throughput in crowded RF environments." "What our customers need and are now looking for is something beyond basic Wi-Fi data mesh networks," said Firetide CEO Bo Larsson. "They need secure, flexible, wireless solutions that are easy to install and manage and work with even more demanding applications such as high speed video and voice over IP."

Posted by Dal at 03:45 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

Led by Norfolk County Council and funded by the East of England Development Agency (EEDA) the GBP1.1m project is the largest community wireless broadband network in the UK and will evaluate the impact and potential of mobile technology, offering free mobile Internet access for public sector employees, the business community and the general public.

The network covers a large area of Norwich city centre, as well as key sites around the city including business parks, the county hospital and the University of East Anglia - in all up to 30 square kilometres.

"This is the most significant outdoor Wi-Fi network in the UK to date," said Jim Baker, Telabria CEO and founder. "Not only does the sheer scale dwarf all other UK deployments, but the project aims to deliver fully mobile Internet access to both the public sector and the private citizen for free, a departure from network models elsewhere. We're delighted that our mSystem APM-300 mesh access point was chosen by Synetrix to be at the heart of the Norfolk wireless network, providing street-level Wi-Fi coverage on the scale required."

More than two hundred Telabria APM-300 dual radio mesh devices have been attached to street furniture such as lampposts to create the network, with the main network gateway on the roof of County Hall in the centre of the city. Nine 'clusters' of APM-300s are backhauled over WiMAX-class point-to-multipoint infrastructure to a 40Mbps Internet feed at County Hall. The network can be accessed by any Wi-Fi-equipped device including laptops, PDAs and web-enabled mobile phones.

"After careful assessment of the mesh product market, we chose the Telabria APM-300 as it not only presented the most technically competent solution for this project, but was also priced highly competitively," said Alex Jadavji, Chief Executive at Synetrix. "Working in close partnership with vendors on a project of this size is critical to its success, and Telabria has shown a clear ability to respond to both our needs and those of our client."

Public sector workers will be able to access the system at speeds up to 1Mbps and organisations including health, education, and emergency services will be considering a range of projects to help evaluate the network. The project is not allowed to compete with commercial broadband providers, the speed at which the general public may connect is rate limited to 256Kbps. Later this year, OpenLink will be extended to twenty rural locations in the district of South Norfolk .

Patrick Hacon, Chairman Norfolk County Council said: "We are very excited about the possibilities that Norfolk Open Link presents for the county. The pilot project will aim to harness the potential wireless technology can have in enhancing the delivery of our public services and stimulating business and private use of wireless technologies."

The Telabria mSystem APM-300 is a ruggedised, outdoor dual radio mesh device; one radio provides network access for any Wi-Fi-enabled device within range, while the second radio communicates with other neighbouring APM-300s to create a self-forming, self-healing mesh network of 'nodes' over which traffic passes to the nearest WiMAX backhaul point. The number of links, or 'hops', between mesh nodes is kept a minimum to reduce latency that can affect real-time applications such as VoIP and IP-CCTV applications.

Telabria has seen an increasing number of contract wins based on its innovative mesh products, and the company is currently building a network of distributors and resellers to service the rapidly growing number of international metro hotzone projects worldwide.

Posted by Dal at 03:32 PM | Direct Link | Comments (0) Posted to WiFi / Wireless

Under this initiative, Voxeo will sponsor development of new open source telephony applications, distribute select applications with its VoIP and IVR offerings, and increase overall awareness of the benefits of open source telephony applications to its customers, partners, and the over 21,000 registered members of its Evolution developer program.

"We believe there’s a huge opportunity to leverage the power of open source at the telephony application layer," said RJ Auburn, CTO of Voxeo. "We hope to start a chain reaction of new contributions and projects that will spark innovation and lower the cost of voice application development."

To kick off RocketSource, three new open source applications developed by Voxeo partners will be posted to Sourceforge.net -- the world’s largest open source development site -- and distributed with downloads of Voxeo’s Prophecy Voice Platform.

The three initial RocketSource applications include:

- Vox-Attendant: a VoiceXML-based, speech-driven auto attendant that can connect callers with any person or group with an enterprise, via their desk, cellular, or VoIP phones.

- Vox-Mail: a VoiceXML-based, speech-driven voicemail application that can store messages locally or integrate with any IMAP email server to provide basic unified messaging.

- Voice Conference Manager: A VoiceXML and CCXML-based, speech-driven conference manager that features both phone and web-based conference call creation, access, and management.

"These aren’t just reusable application components. These are complete, working applications that can be deployed immediately. At the same time, enterprise customers will have the freedom to extend and customize as they see fit without the time and cost associated with custom-developed applications," said John Hibel, Vice President of Marketing at Voxeo.

"Until now, the open source community developed voice applications in isolation from each other," said Moshe Yudkowsky, President of Disaggregate and lead developer of Voice Conference Manager. "With RocketSource a new community will emerge, and we will start to create applications that use common user interfaces, common themes, common databases, and common modules. If we succeed, the result will be revolutionary -- an avalanche of open source applications that provide a wide range of functionality and become part of everyone’s basic tools, just like email."

RocketSource is the latest initiative in Voxeo’s longstanding commitment to foster VoIP and IVR application development. Voxeo has offered Evolution -- its free hosted VoIP and IVR developer program for nearly seven years. In early 2006 Voxeo also introduced a free version of its premise-based Prophecy Voice Platform -- the world’s first feature-rich, easy, VoIP and IVR platform with built-in speech recognition and synthesis engines. Unlike other extremely large and complex speech platforms, Prophecy can be downloaded and installed in less than two minutes. Going forward, Voxeo will soon announce other initiatives to sponsor and promote partner offerings to Prophecy and Evolution users.

Posted by Dal at 03:25 PM | Direct Link | Comments (0) Posted to VoIP Software