"As communication services migrate to internet-based platforms, it is important that the security and dependability users expect in the current public switched networks be maintained with these new converged technologies," the group stated.

The researchers plan to conduct a security analysis of VoIP protocols and implementations, and explore issues such as VoIP authentication for dealing with voice spam, modelling of VoIP traffic and device behaviour, mobile phone security, and security of VoIP applications running on user agents. "GTISC feels strongly that VoIP security should not be an afterthought," said Mustaque Ahamad, principal investigator and director of GTISC. ISS and BellSouth have committed $300,000 to a two-year research program.

The funding will enable GTISC faculty and graduate students to work with ISS and BellSouth technologists to develop and evaluate solutions that address VoIP security. In return, BellSouth and ISS will have access to the resulting intellectual property.

Source: VNUNet

Posted by Dal at 11:29 AM | Direct Link Posted to VoIP Security

Note:  Another great installment from the Nerd Vittle's Asterisk Factory.

To add additional extensions takes about 5 seconds. This PBX features the latest build of Asterisk (1.2.12.1) and 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 ...

Click Here for the Full Nerd 

Posted by Dal at 10:26 AM | Direct Link Posted to Asterisk Help

 

“As a municipal wireless pioneer in the state, we need to keep ahead of the technology curve,” said Anthony Rodrigues, director of information technology for the City of Malden. “But, we also need to make sure that our new technology endeavors are good investments and not a threat to our operation. Bluesocket was the clear choice for our wireless security need, because it was a proven, affordable solution that was easy to manage and eliminated thousands of dollars in connectivity costs.”

Bluesocket’s BlueSecure technology, which uses WLAN authentication to overcome bandwidth limitations and detect unauthorized users, became essential when the City of Malden won a $50,000 grant from Proxim Wireless last month to upgrade its mesh network hardware. With the new security and hardware technologies currently being deployed, the city is putting into motion a plan to extend the range of Malden’s secure network, which includes putting Wi-Fi transmitters on city vehicles and in high-density areas, such as T stations, playgrounds and parks. The city’s information technology department is also looking to expand the mesh network to local school systems to promote remote learning, as well as create an active link to the local access TV studio in order to expedite news reporting and editing.

The Malden police department will also use the expanded network to stream video footage from local areas directly to the police station, making it easier for police officers to monitor and respond to crimes at those locations.

“Protecting the community is our department’s number one priority and technology plays a big role in that effort,” said Kenneth Coye, Chief of Police at the Malden Police Department. “Malden’s wireless system shows great promise for law enforcement. Cameras at high risk locations and connectivity to state and federal resources are mandatory in a post-9/11 world. We expect to develop community support for virtual crime watches at locations of concern, bringing law enforcement and our neighborhoods closer than ever.”

“Municipal wireless networks are growing at a remarkable rate because of their ability to offer more efficient communications between city departments, in addition to providing residents more affordable access to the Internet,” added Mads Lillelund, CEO at Bluesocket. “Wireless security plays a significant role in developing city-wide networks and, because of that, a number of municipalities have turned to Bluesocket. We provide reliable wireless security at an affordable price, so organizations with fixed budgets don’t have to settle for inferior quality.”

The City of Malden is one of several municipal districts that have deployed Bluesocket for wireless security. The City of Burbank, Calif., and City of Springfield, Mo., also recently deployed BlueSecure technology to manage network bandwidth, assign user roles and provide load-balancing between wireless APs. For more information on Bluesocket public access solutions, visit http://www.bluesocket.com/solutions/publicaccess.html.

About Malden, Mass.

Located five miles northwest of Boston, Malden is a vibrant and diverse community of 58,000 in the midst of an urban renaissance. Covering 4.8 square miles, Malden’s close proximity to Boston and convenient access to mass transit, make Malden a very desirable home for urban professionals, artists and students alike.

Posted by Dal at 09:39 AM | Direct Link Posted to WiFi / Wireless

Note:  Nice to see the fully SIP support predictive dialer.  Time to flex my blogger muscle and see if they will let me give it a test drive.

Interactive Intelligence Inc., a global developer of business communications software, has released a new VoIP-enabled version of its outbound dialing and campaign management software, Interaction Dialer, designed for contact centers, teleservices firms and collections operations. The latest release of Interaction Dialer, version 2.4, is now based on the SIP standard so it can operate in an all-software, all-VoIP environment to help reduce costs and simplify management. The release includes an all new SIP gateway called Interaction Gateway, an appliance connecting legacy telephone trunks (T1s) to VoIP networks. Interaction Gateway works with Interaction Dialer to give high-volume outbound contact centers accurate predictive call analysis capabilities.

“Interaction Gateway is unique because it provides advanced outbound call analysis -- such as distinction between answering machines versus live speakers, and wrong numbers versus network issues -- while taking full advantage of the open SIP standard,” said Yankee Group senior analyst, Ken Landoline. “Combined, these enhancements result in more effective dialing campaigns, simplified deployment and configuration, lower cost, better redundancy, and unbeatable flexibility compared to traditional outbound dialing solutions.”

According to Interactive Intelligence, the new SIP-based Interaction Dialer can lower costs by between 10 and 50 percent compared to traditional TDM-based dialers by reducing hardware requirements.

The company says Interaction Dialer can reduce costs further using Interaction Gateway at strategic locations to ensure it uses the most cost-effective call routing. To illustrate, in a typical configuration, ISDN PRI trunks from the public phone network are connected to Interaction Gateway. After that point, the entire configuration is voice over IP. This provides unmatched geographic independence. For example, the Interaction Dialer and Gateway servers can be in the U.S., while agents are in India.

“As an outsourced provider of CRM services, our clients drive our technology requirements, and a key requirement across the board is to offer highly customizable services at an affordable price,” said Chris Adomaitis, president of network and telecom services for Dialogue Marketing, an Interactive Intelligence customer since 1998. “Interaction Dialer’s new Gateway option lets us do just that by using a pure softswitch, thus dramatically reducing costs associated with hardware.”

Other Interaction Dialer enhancements include improvements to its patented pacing algorithm, which maintains the optimum balance between low abandon rates and maximum agent utilization; a single “Health View” interface that enables supervisors to monitor and analyze Dialer statistics from campaigns, workflows, and outbound calls; and a “Contact Import Wizard” that simplifies campaign management by providing a tool to import records into call list tables from a variety of commonly used sources, including CSV files, Microsoft Access, Microsoft Excel, SQL Server, and Oracle.

Interaction Dialer was first released in 1999 as an add-on module to the company’s contact center automation software, Customer Interaction Center (CIC). By leveraging CIC’s multi-channel routing, recording, and interactive voice response, Interaction Dialer reduces costs and simplifies infrastructure compared to traditional, standalone dialers. Interaction Dialer, in conjunction with CIC, can also work with existing PBXs and IP PBXs.

In addition to power, preview, predictive, precise, and agent-less dialing, outbound campaign management, and inbound/outbound call blending, Interaction Dialer includes capabilities for telemarketing regulatory compliance, Web-based scripting, campaign staging, real-time supervision and reporting.

A single Interaction Dialer system, along with Interaction Gateway’s “rack and stack” architecture, can support up to 1,000 outbound agents -- more when deploying multiple systems -- while giving multi-site organizations maximum load-balance and fault tolerance capabilities.

Interaction Dialer 2.4 is available immediately and is offered through the Interactive Intelligence channel composed of approximately 250 value-added resellers, and through the company’s direct sales force.

Source: Interactive Intelligence

Posted by Dal at 08:38 AM | Direct Link Posted to VoIP Software

 

Ok OK, I am kidding but really.  Why is it so damn hard these days to cancel any new service you sign up for?  Those kind of tactics just make it so I research even harder before trying out new services and in fact actually reducing the number of new things I try just order at the drop of a dime fearing the hassle and cussing I will need to do when I find out what they are offering was not what it was marketing as.  Companies please take note make either make your service better than advertised to make cancelling painless as possible.  /END RANT.  Thanks Tom, good heads up on the Vonage Elite Team

Posted by Dal at 08:20 AM | Direct Link Posted to Vonage

T220B, the first unit to become available under the Sequans/ MiTAC agreement, is a fixed WiMAX (802.16-2004) self-installable desktop unit that provides wireless broadband connectivity at the 3.5 GHz frequency. T220B features switched diversity between four built-in high gain antennas and supports space/ time coding (STC) and subchannelization. The RF chip for this device is provided by Sierra Monolithics, one of Sequans' RF partners.

MiTAC is also developing mobile WiMAX subscriber units built around Sequans' 802.16e-2005 SOCs. The mobile WiMAX units currently under development for release in late 2006 include a PCMCIA card and a desktop unit.

"Sequans' chips deliver the highest performance and flexibility and allow us to build a full range of WiMAX end user devices with advanced features and functionality," said Jay Cheng, general manager of MiTAC Technology Corporation. "Following our successful collaboration with Sequans on the high performance fixed WiMAX desktop unit, we are now developing a range of mobile devices. Sequans high performance chips allow us to create small form factor units with the lowest power consumption in the industry."

"We are very pleased to work with MiTAC, a dynamic organization with very creative designs and unparalleled expertise in manufacturing consumer electronics," said Bernard Aboussouan, vice president of marketing and business development, Sequans. "MiTAC is the first ODM delivering a WiMAX self-installable desktop unit and we believe this unit as well as the coming mobile units will set a standard in the industry for excellence and quality."

The MiTAC WiMAX end user devices are thoroughly tested and proven interoperable with WiMAX base stations using chips from Sequans or others. Sequans has been at the forefront of interoperability testing.

See Sequans at WiMAX World USA at the Boston World Trade Center, October 10-12, booth 308.

Posted by Dal at 11:16 AM | Direct Link Posted to WiMax

The voice synthesis system is again broken, freePBX can’t reload Asterisk without completely shutting down and restarting Asterisk (amportal restart). And there appear to be all sorts of interrupt or timing problems that we’ve never seen before … going back to Asterisk@Home 1.2.

We attribute many of the problems to a new version of CentOS and Asterisk, both of which are bundled into the TrixBox 1.2 package, but who knows. What we do know is TrixBox 1.2 is a little too Bleeding Edge for our taste, and most of the Nerd Vittles goodies that depend upon the Flite speech engine no longer work on many machines...

Click Here for the Full Nerd 

Posted by Dal at 10:51 AM | Direct Link Posted to Asterisk Help

My suggestion is that we start 10 AM on Monday, nov 20th, and continue to 3 PM (15:00) on Wednesday, November 22nd.

This is going to be a very practical meeting with interoperability tests between various devices, SIP debugging and coding. We need to figure out a way to add proper handling of video attributes in Asterisk and maybe look at additional features I know that you are working on out there

- Video on hold (streaming)
- Video prompts for IVRs
- T.140 text in addition to video
- Integration with 3G video
- Video conferencing

Let's investigate these areas together, trying to find solutions that we can work forward on - integration with other Open Source products or just experience in connections to commercial products.

There will be a very limited amount of seats. Mail me off list if you want me to keep a seat open for you. You will have to cover your own costs, but if everything works out we will have access to the conference center sponsored.

Have a nice weekend. I'm sure I will - going to the Asterisk beachcamp on the beautiful beach outside Malaga in Spain!

Cheers,
/Olle
 

Posted by Dal at 10:46 AM | Direct Link Posted to Asterisk News

All of these releases include substantial new functionality and performance improvements. The documentation detailing these changes is still in process, so future beta releases will contain more complete lists of the new features. There are a few functionality changes still not yet complete for Asterisk, but we expect to have them finalized in the next few days and merged into the beta release shortly.

In the specific case of Asterisk, there were quite a few changes made in this release that are not backwards compatible with prior releases.  Users are encouraged to read the UPGRADE.txt file very closely before starting to build, install and test this release, so that they will be aware of changes they may need to make on their systems to accommodate it.

We encourage everyone to try out these releases on NON-PRODUCTION systems and report their findings on the mailing lists and the issue tracker; we'd like to move through this beta process as quickly as possible and getting feedback is the fastest way for us to accomplish that.

As always, the releases are on our FTP servers, although there are no patch versions available since this is the first release from the 1.4 branches. The releases have been signed by nearly every member of the Digium development crew, for your GnuPG verification pleasure :-)

Thanks for supporting Asterisk, Zaptel and our other projects!

Posted by Dal at 10:43 AM | Direct Link Posted to Asterisk Releases

Hello AVN Readers,

This is your humble editor Dal, back in the office from vacation.  I had a great time on the Oregon coast relaxing and recharging.  I will posting the most important developments I missed the last couple days and then it will be news like normal tomorrow morning.  Hope you enjoy the last weekend of summer and now the fall season begins.  Cheers!

-Dal 

Posted by Dal at 10:40 AM | Direct Link Posted to AVN Site Updates

n January, the University of California, Santa Barbara announced it was prohibiting Skype because the license agreement it presented to users gave third parties access to the university's network. UC-Santa Barbara said it would allow other computer-calling services.

California State University Dominguez Hills has long discouraged use of all computer-calling services, including Skype, a spokesman said. Skype has also been banned by some universities in the United Kingdom.

Jennifer Caukin, a spokeswoman for eBay, said Skype was looking ``forward to having a direct dialogue with SJSU officials to discuss their concerns and educate them about how Skype works.'' Caukin declined to discuss other bans.

EBay purchased Skype last October in a deal that was then valued at $2.6 billion. The growth of Skype, which is believed to be the world's most popular computer-calling service, has been key to eBay's attempts to persuade people to invest in its future.

As of mid-July, Skype boasted 113 million registered users around the world, an increase of 156 percent from the previous year.

But the value of shares in eBay has fallen 41 percent since January on the fear that the decade-long and world-wide expansion of the online marketplace giant had ended.

Founded by the creators of KaZaA, a controversial file-sharing service, Skype uses peer-to-peer architecture to route free calls between computers. Previously, universities had banned KaZaA, along with sister services like Morpheus, iMesh, Gnutella, LimeWire, Grokster, because they were used to illegally trade online movies and music.

The problem with Skype is not that it enables illegal behavior, but that its end-user license agreement appears to permit legal use of university's networks by people outside the university and, indeed, the United States.

``It's a fairly subtle problem,'' said Kevin Schmidt, campus network programmer at the University of California, Santa Barbara. Skype users agree to run an application on their computers that is built to relay calls between third parties whenever a computer is turned on.

 Click Here to Continue Reading

Posted by Dal at 09:45 AM | Direct Link Posted to Skype

Note: Now this is a nice announcment.  It is a very bright spot for Asterisk with enterprise level equipment being released for carriers.  grats

Ranch Networks, provider of networking appliances designed to facilitate carrier and enterprise grade VoIP deployments, today announced the availability of its VoIP Matrix Technology(TM), the company's patent-pending clustering technique developed to increase scalability, reliability and security across Asterisk server farms.

The VoIP Matrix Technology enables enterprises and service providers to combine one or more Asterisk servers with a Ranch Networks' (RN) appliance - forming a virtual IP PBX that supports large number of simultaneous calls. Like a web server farm, theoretically the VoIP Matrix Technology could support tens of thousands of simultaneous calls using an Asterisk server farm. The VoIP Matrix Technology, bridged with the RN appliance ensures zero dropped calls on a VoIP network, prioritizing calls over data. The RN appliance bridges the RTP media between the SIP end points, allowing for media to continually flow without interruption, even if an Asterisk server is unplugged. All new calls then become distributed among the remaining PBXs in the cluster. This technique is a seamlessly scalable solution which enables customers to add more PBXs as their needs grow.

"Ranch Networks understands the needs of VoIP customers," said Alex Pavlovsky, president of Ranch Networks. "When using the VoIP Matrix Technology solution, customers do not have to worry about the security, quality, and scalability of their VoIP systems. Through the clustering technique, call centers, enterprises and service providers can easily increase network capabilities as their businesses grow."

Users can access the virtual IP PBX, including any one of the servers involved, using a single IP address and MAC. Each Asterisk server used to form the virtual IP PBX within the cluster is monitored using SIP health messages. When a PBX stops responding to the SIP health monitoring messages, it will be taken out of the cluster in real-time. As soon as it starts responding to the SIP health monitoring messages, it will be added back to the cluster. The VoIP Matrix Technology allows for users to also add a new IP PBX in real-time, as well as remove an existing IP PBX can for maintenance purposes.

The VoIP Matrix Technology is available with the RN 20, 40, and 41. The RN series of appliances can be purchased through leading Asterisk distributors and resellers worldwide. For a list of resellers or additional information, please visit www.ranchnetworks.com.

Posted by Dal at 11:41 AM | Direct Link Posted to Asterisk News

Posted by Dal at 11:33 AM | Direct Link Posted to AVN Site Updates

Posted by Dal at 10:11 AM | Direct Link Posted to Asterisk Releases

Note: Matt Florell sent this in for the Asterisk community.  Thanks

We've released another update to our astGUIclient suite: 2.0.1

http://astguiclient.sf.net/

The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the
VICIDIAL call center suite.
 

This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks.

For this major revision, we have reworked a lot of the server-side processes to make them more efficient as well as writing a new interactive installation script which uses a new central configuration
file. Another major new feature is the addition of a more advanced Predictive dialing algorithm for the outbound dialer portion of the system. We have also tested the suite on Asterisk versions through 1.2.12.1(cannot use 1.2.11 or 1.2.12 because of Asterisk bugs)

All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only.

Check out the project blog for more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,
MATT---
 

Posted by Dal at 10:08 AM | Direct Link Posted to Asterisk Software

In addition to Los Angeles, Vonage recently turned up E911 at more than 125 local run emergency call centers across the U.S. in less than one month -- bringing the total percentage of Vonage U.S. subscriber lines that have E911 to over 89 percent.

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

"Turning up Los Angeles is a tremendous step for Vonage, as thousands more subscriber lines in California, one of Vonage's largest markets, now have full E911 capability," said Vonage CEO Mike Snyder. "In less than one month, Vonage equipped more than 100 locally run calling centers with E911. Vonage will continue to work with the FCC, regulators Congress and public safety officials until PSAPs across the nation are equipped with E911."

Since August 21, 2006, Vonage has added the following counties to its list of those with E911 capabilities -- bringing the total number of calling centers with emergency 911 service to over 5700.

Source: Vonage 

Posted by Dal at 11:45 AM | Direct Link Posted to Vonage

 

The TPMC-4X offers direct high-speed wireless communication with any Ethernet Crestron control system, providing complete control of AV, lighting, HVAC, shades and screens from anywhere in the house. True 2-way communications allows dynamic graphics and complete system status feedback, providing an enhanced user experience.

The Crestron TPMC-4X features a Windows CE.NET 4.2 Operating Systems providing 802.11b WiFi communication throughout a facility on its wireless network. The TPMC-4X offers the perfect combination of touchpanel flexibility and tactile control, featuring a 3.5-inch touchscreen and 17 programmable, backlit pushbuttons. The 320x240 touchscreen display features 16-bit color graphics, dynamic text, dynamic graphics, translucency and streaming video capabilities.

Control all the audio and video components, lighting, thermostats, shades and more. Full 2-way WiFi communication allows users to view and set temperature in any room; confirm and adjust lighting levels and monitor lobby entrances from a sleek, lightweight handheld controller. Control an audio server or an Apple iPod remotely; network communications enable dynamic graphics such as the display of cover art. Search the music library efficiently with the on-screen virtual keyboard to enter text such as artist name or song title.

Have another set of eyes in the palm of your hand. The TPMC-4X also features embedded functionality for viewing streaming video from security cameras or other video sources. The handheld Isys i/O touchpanel can display any MJPEG format source signal - perfect for monitoring the nursery, entrances and gateways or swimming pool.

The Isys i/O TPMC-4X delivers complete 2-way WiFi communication and direct high-speed access to Ethernet Crestron control system for unparalleled system integration from a handheld touchpanel.

For 35 years Crestron has been the world’s leading manufacturer of advanced control and automation systems, innovating technology and reinventing the way people live and work. Offering integrated solutions to control audio, video, computer, IP and environmental systems, Crestron streamlines technology, improving the quality of life for people in corporate boardrooms, conference rooms, classrooms, auditoriums, and in their homes.

Crestron’s leadership stems from its dedicated people who are committed to providing the best products, programs and services in the industry.

Posted by Dal at 10:26 AM | Direct Link Posted to VoIP Hardware

Services between destinations from London and Aberdeen including York, Leeds, Newcastle, Edinburgh and Glasgow are all covered and non-stop wireless connectivity is promised. Ninty would like it noted that this is especially good for Animal Crossing fans (though of course could also be used by laptop users or - shudder - owners of the PSP).

The technology will apaprently be seamless thanks to a combination of satellite dishes on train roofs and 3G/GPRS connectivity - meaning you'll remain connected even in tunnels. Prices start from 2.95 GBP, and it is free for those in first-class, whilst we're told the fleet of trains will be ready from the end of the summer.

John Gelson, Media Relations Manager at GNER, enthused: “What otherwise might be seen as five long hours on a train is now an excuse to make the most of new wireless gaming! The combination of WiFi and the at-seat power-point is great for gamers as it means you really can play non-stop while you travel. The WiFi access also means you can surf the web and communicate with friends on the move.”

Source: Ferrago

Posted by Dal at 10:17 AM | Direct Link Posted to WiFi / Wireless

Ron Sege, President and CEO of Tropos Networks, said, "We are excited to be working with GigaBeam, the global leader in 70-80 GHz technology. Their 1 Gbps WiFiber throughput substantially increases the capacity of a WiFi mesh network, enabling service providers to deliver multi-megabit services and support bandwidth intensive applications for a very large number of subscribers on a metro scale. Also, being a wireless solution, it can be deployed faster, more economically and to more locations than terrestrial fiber. We look forward to working with GigaBeam to deliver fiber-grade connectivity to our customers' MetroMesh networks worldwide."

Lou Slaughter, GigaBeam's Chairman and CEO, stated, "We are delighted to be a preferred partner of Tropos, the leading provider of meshed WiFi networks in the US and globally. Our current WiFiber product, offering 1 Gigabit-per-second (Gbps) transport and backhaul, ensures robust network integrity in any meshed network. Unlike lower speed microwave technologies, our higher speed technology is more suitable for carrying and distributing data, VoIP and video capacity across WiFi networks. We and Tropos have found that to be precisely the case in the Mountain View network, where the addition of our WiFiber product substantially improved the capacity of the overall meshed WiFi network. We look forward to working with Tropos in the US and globally."

GigaBeam WiFiber products operate in the 71-76 GHz and 81-86 GHz radio spectrum bands. This portion of the radio frequency spectrum has been authorized by the Federal Communications Commission for wireless point-to-point commercial use. Use of these frequency bands for commercial use was pioneered by GigaBeam's founders.

GigaBeam's technology, utilizing these large blocks of authorized contiguous spectrum, enables multi-Gigabit-per-second communications through use of Gigabit Ethernet and other standard protocols. The current speed achieved by GigaBeam's WiFiber G-1.25 product series is full duplex at one Gigabit-per-second (equivalent to 647 T1 lines or 1,000 DSL connections) which supports GigE protocol. GigaBeam recently announced its WiFiber G-2.7 series, to be released this year, which will operate at 2.7 Gbps. The protocols to be supported by the G-2.7 product series include 2 x GigE (2 x 1 Gbps); OC-48 / STM-16 (2.488 Gbps); SMPTE 292M (1.485 Gbps) and both 1 and 2 Gbps fiber channel. GigaBeam also plans deployment of future products capable of 10 Gigabits-per-second utilizing either the 10 Gigabit Ethernet or OC-192 protocol standards.

Click Here to Continue Reading 

Posted by Dal at 10:09 AM | Direct Link Posted to WiFi / Wireless

Trango Broadband Wireless, an industry leader in high-performance wireless products and backhaul solutions, announced today that U.S. Wireless Online, one of the nation's largest wireless Internet broadband network providers, selected Trango to power the backhaul for Pittsburgh's Downtown Wi-Fi Network.

The Pittsburgh Downtown Partnership (PDP) commissioned U.S. Wireless Online to deploy the Wi-Fi network in order to provide free outdoor wireless access in the Central Business District. In addition, the Wi-Fi network will provide nearby municipalities with secure service for critical communications for public safety entities. "Trango Broadband Wireless has been an infrastructure partner of U.S. Wireless Online for some time now," said Tim Pisula, EVP & CTO of U.S. Wireless Online.

"Their product suite represents a significant value for us as it delivers -- day in and day out -- consistent carrier-grade performance whose operational cost of ownership is tough to beat." The Trango products used for this project were the Access5830 Access Point, the Atlas Fox Subscriber Unit, and the Atlas 5010 Point-to-point System, known for exceptional performance and reliability, offering features that continue to out-perform the competition for "last-mile" and backhaul connectivity. 

"We have found significant success using Trango wireless radios to power muni-wifi applications similar to Pittsburgh's Downtown Wi-Fi Network. Our ability to provide high-capacity 45 Mbps constant throughput and our proven reliability in the field makes Trango the vendor of choice for powering up muni-wifi mesh networks," said John Seaman, Director of Sales at Trango Broadband Wireless. "We are pleased that US Wireless has chosen Trango for the Pittsburgh project and we look forward to supporting them in any way possible."

Posted by Dal at 10:04 AM | Direct Link Posted to WiFi / Wireless

Note:  Nice little article explaining SIP in a basic sense.  Informative Read

Have you ever wondered why long distance calls cost so much? In part the reason is because telephone lines cost so much. When driving, you might occasionally see a telephone crew maintaining a telephone line, but what you may have never considered is that there are literally thousands of individuals working around the clock to maintain our telephone lines.

The telephony system works via a cog and wheel setup. What this means is that every long distance call you make is routed along a telephone wire to a central station, where your voice is routed to another central station, which is finally carried to the person with whom you are trying to communicate. For the call to be maintained, the entire time you are speaking, a space along all the lines in between you and the person you are talking with must be completely devoted to you. Because millions of people are talking at the same time, the little space along the telephone lines becomes rather desired property. And like all things desired, the price is high. Before recent innovations, however, there were no alternatives, so everyone grudgingly paid the often costly long-distance telephone bill.

SIP, or Session Initiation Protocol, has turned the telephony world upside down. Specifically, SIP refers to a protocol that allows computers to talk to each other without going through a central station. Practically, what that means for you and me is that it is no longer necessary to pay for expensive telephone lines to complete our calls. SIP technology is a relatively new development in which calls are made on a peer-to-peer rather than cog and wheel network. What that means, is that you are now able to call people directly from your SIP enabled phone to theirs. This ends up being radically cheaper than the old way of calling.

The SIP system does not require a central computer and operators like the old telephony system did. Rather, your computer, or SIP enabled phone, does all the routing for you.

SIP has been around for a number of years, but only recently has it begun to go mainstream and take off in popularity. This quick increase in interest over SIP is due to companies like Mobalex, who were aware of the fact that over the generations we have come to expect certain tones, buttons, and protocols from our phones. So what they have done is to transpose those functions onto the SIP system. Rather than forcing users to communicate in a completely new way, what these companies have done is to provide a calling experience which from the user’s perspective is completely identical to traditional telephony.

SIP is typically offered in two formats, computer based and hardware based. Computer based SIP is a system that allows you to make calls using your computer as the router and communicating via a headset on your computer. The more practical and popular version, however, actually provides you with new SIP enabled telephone handsets or converts your existing phones to SIP. By eliminating any technical requirements, modern SIP providers have made using the system as easy, or easier, than using a traditional phone. I say easier, because many companies are able to take advantage of the fact that the system is internet based to provide you with some very unique benefits. These include the ability to adjust your plan, change your calling options, and even pay your bill from the same website.

SIP technology is quite revolutionary in the world of communication. By creating a peer-to-peer network, SIP has been able to radically undercut the prices of traditional telephony, take advantage of the Internet, and still maintain the ease of traditional telephony. It is merely a matter of time before we are all using SIP for all of our telephoning needs.

Source: Click Here 

Posted by Dal at 09:52 AM | Direct Link Posted to SIP

 

The kit is aimed to help Digium customers quickly reach businesses with two to 50 users, a group for which traditional IP PBX solutions tend to be too large and expensive, and for which lower-end solutions are typically lacking in features, said Bill Miller, vice president of product management and marketing. He added that more than 87 percent of all PBX office systems still use traditional PBX phones, so the “hybrid” solution – which accepts both analog phones and Ethernet routers – offered by Digium, presents a more feature-rich and flexible solution. Because the appliance is based on Asterisk, service providers and others get all the popular PBX features already offered by the Open Source PBX solution, and they can build specialized features and applications on top of that.

And because the appliance is small and, unlike PC-based solutions, has no hard drive and no power requirements, it has a lower failure rate and takes up less real estate at the customer premises than existing popular IP PBX solutions, Miller explained.

The stackable appliance includes a full Asterisk server; eight analog ports (FXS, FXO); five Ethernet ports (four LAN, WAN); hardware echo cancellation; compact flash for voice mail or wireless; a built-in router; a craft port for debug; a commercial Asterisk license, so any applications developed on the box don’t have to go back to the open-source community; and built-in uCLinux.

Also in the kit is Release 1.4 of the Asterisk Business Edition software, which will be publicly available the first week of October. The new release will dramatically improve performance, interoperability and call quality handling; offer connectivity to GoogleTalk; have the ability to store voice mail in an IMAP server to enable unified messaging; and more, said Kevin Fleming, senior software engineer.

In addition to the appliance and Asterisk software, the kit includes 2-4 port FXS cards; 2-4 port FXO cards; multimedia add-on cards; an 8MB flash; cables for all port types; an IP phone; and a CD containing software, documentation/specs, how-to manuals and Digium support details relating to Asterisk, which Miller said had not been available in one place previously. Also new with the kit is a GUI for Asterisk, said Miller, making it more simple to use, and more easily allowing customers to specialize screens for different business verticals. He added that any changes to the GUI are reflected immediately in the configuration files, and vice versa.

The kit has two options relative to the service piece, one of which includes a day of training with Asterisk inventor Mark Spencer.

The kit will be available directly from Digium in October for $3,995. Production units are expected to be available at a later date with volumes of 10,000 for under $500.

Posted by Dal at 05:26 PM | Direct Link Posted to Asterisk News

Polycom, Inc., provider of unified collaborative communications solutions, today announced the VoIP Interoperability Partner (VIP) Program for manufacturers of SIP-based call control platforms. The Program allows vendors to demonstrate through testing and certification that their premise-based (IP PBX) or hosted (Softswitch) call control solutions are interoperable on a defined feature set with Polycom SoundPoint IP desktop and SoundStation IP conference phones and SIP software. 

The Program assures that customers moving from legacy PSTN phone systems to productivity-enhancing VoIP solutions featuring Polycom phones and certified manufacturers' equipment, enjoy seamless interoperability and reliable technical support. Currently, Digium, Comverse/Netcentrex Converged IP Communications, Objectworld, Pingtel, and Whaleback Systems have successfully passed interoperability testing and are now Polycom VIP certified.

"Interoperability testing between SIP-based call control platforms and phones deployed with them is critical to ensuring reliability and comprehensive feature support," said Paul Waadevig, industry manager for Frost & Sullivan's Conferencing & Collaboration practice. "Polycom's VIP program provides standards-based IP PBX and softswitch vendors, who do not manufacture their own phones, a process to test their systems with Polycom's VoIP phones and provide customers with a certified interoperable telephony solution."

The Program involves rigorous interoperability testing at a Polycom-approved test facility according to a Polycom test plan. To become certified as Polycom VoIP compatible, a call control platform must demonstrate 100 percent interoperability on a Polycom-predefined critical feature set. Interoperability can also be certified for features extending beyond the critical feature set, depending upon the capabilities of a particular platform. Certification in the VIP Program is available for the entire Polycom portfolio of SoundPoint VoIP desktop and SoundStation VoIP conference phones.

Upon becoming a Polycom VoIP Interoperability Partner, manufacturers of VoIP call control platforms can enjoy the following benefits:

Benefits: 

-- The right to use the "VIP Partner" and "Polycom VoIP Compatible" marks when referring to itself and to its platform, respectively.

-- Inclusion in the published list of VIP Partners on www.polycom.com

-- Eligibility of Manufacturer's resellers to apply for Polycom VoIP Reseller Certification and, if certified, officially resell Polycom VoIP phones in conjunction with the certified platform.

-- Access to Polycom marketing and technical resources, as per the VIP program agreement.

Source: Polycom Inc. 

Posted by Dal at 11:11 AM | Direct Link Posted to VoIP Hardware

Note: Sheldon Rose sent this tibit into me about snom being a preferred telephony hardware vendor for Pandora's pbx solution.

snom technology AG, developer and manufacturer of Voice-over-IP (VoIP) telephones, today announced a strategic partnership with Pandora Networks, a provider of On Demand IP communication services for small- to medium-sized businesses (SMBs). According to the agreement, Pandora will include snom telephones as part of its Worksmart communications solution allowing SMBs to enjoy the benefits of enterprise-level communications services and collaboration tools at an affordable price point.

“The partnership with Pandora signifies our first service provider agreement in the U.S.,” said Michael Knieling, CFO and Executive Vice President of Marketing and Sales for snom. “Working directly with service providers is an important part of our business model, as it allows us to expand rapidly in an already- established customer base.”

Pandora Networks delivers unified business communications and collaboration tools as a service to small- and medium-sized businesses. Pandora Network's Worksmart solution, an integrated voice, video, messaging and collaboration platform, will now include the snom 300, snom 320, and snom 360 IP phones.

“As a leader in VoIP telephones, it was an obvious choice to partner with snom,” said Walter Snell, CEO and Founder of Pandora Networks. “Our Worksmart hosted IP communication solution allows hardware manufacturers like snom to reach out to SMBs who want to avoid an investment in premise-based PBX hardware. Now our customers will be able to subscribe to a complete end-to-end communications solution, starting with snom phones.”

snom telephones support all common standards, as well as the latest technology platforms. All snom phones are compatible with SIP-based telephone systems and system components including open source platforms such as Asterisk, SER or sipXpbx, and proprietary solutions offered by companies such as Pandora (Worksmart), Kapsch (Missisipi), Objectworld (UC Server) and many more. snom's technical team works closely with the Internet Engineering Task Force (IETF), ensuring that all snom phones are equipped with the highest security standards in VoIP.

Posted by Dal at 10:33 AM | Direct Link Posted to PBX / iPBX

 

“This directly addresses the need for an affordable speech solution for the Asterisk community, and provides a significant additional benefit for Asterisk Business Edition customers,”

said Jim Webster, director of software technologies at Digium. This is the first time that Digium has offered a fully integrated commercial speech recognition technology for Asterisk. To ensure seamless integration, Digium created a unique connector bridge to the LumenVox Speech Engine.

The Speech Starter Kit includes the connector bridge and one LumenVox Speech Engine port, and can be deployed directly on any supported Asterisk server. Asterisk users can purchase the Speech Starter Kit directly from Digium's website.

Test-drive the LumenVox speech recognition software at the upcoming VON show in Boston (www.von.com), at www.LumenVox.com or call 877-977-0707 and say “Demo.”

Posted by Dal at 10:27 AM | Direct Link Posted to Asterisk Software

 

"Carriers are seeking to both increase revenue and to retain their customer base from encroachment by cable operators," said Matt Cuson, vice president of marketing for Minerva. "Converged solutions such as what we are presenting with MetaSwitch will provide new financial opportunities for the operators and at the same time revolutionize the way their customers purchase and enjoy television programming."

The combined solution seamlessly blends VoIP, legacy TDM and next generation telephony services, while adding video content. This application of converged IP telephony and video services offers service providers a significant additional revenue opportunity from existing customers as well as a clear competitive advantage in the increasingly competitive subscriber markets to new customers. Through this service offering, subscribers have access to caller ID information, high definition TV broadcasts, pay-per-view, video on demand and personal video recorder features, enabling service providers to deliver a full suite of interactive IP communications and entertainment services.

"This partnership enables our customers to actively participate today in the video over IP market, which is expected to grow ten-fold by the end of the decade, both in terms of revenue and number of subscribers," said Andrew Randall, vice president of marketing at MetaSwitch. "With competition for last mile subscribers heating up and technology rapidly moving forward, traditional telephony providers must deploy new applications, such as video, to participate in this growing opportunity."

The combined MetaSwitch and Minerva demonstration can be seen at the Fall VON Conference and Expo in the MetaSwitch booth, number 855, at the Boston Convention Center. The Expo floor is open to show attendees Tuesday, Sept. 12 through Thursday, Sept. 14.

Posted by Dal at 12:28 PM | Direct Link Posted to VoIP News

 

"The telephone remains the most critical communication tool in business, yet studies show that two-thirds of the frequencies in which the human ear is most sensitive, and 80 percent of the frequencies in which speech occurs, are beyond the capabilities of the public telephone network," said Jeffrey Rodman, co-founder of Polycom and chief technology officer of Polycom voice communications division. "People are accustomed to mediocre phone call quality because there hasn't been an alternative -- until now.

VoIP gives us a new network dynamic to deliver a broader spectrum for voice communication and dramatically improve the user experience. Polycom HD Voice smashes the quality barrier of traditional telephony by leveraging broader frequencies and incorporating leading technology developed in Polycom's labs."

Polycom HD Voice solutions incorporate leading technologies that Polycom has developed through more than 15 years in voice communications, including wideband audio, enhanced signal processing, Acoustic Clarity Technology(TM)2 (including next generation technologies for transparent full duplex, echo cancellation, dynamic noise reduction, automatic gain control and microphone management) and specialized system design to deliver unrivaled clarity and richness. This enables better clarity and improved intelligibility of information, which significantly improves comprehension and productivity and reduces listener fatigue.

"Polycom HD Voice represents the next wave in VoIP, showing customers that VoIP communications quality can be vastly superior to that of traditional phones," said Will Stofega, research manager, VoIP Services at IDC. "We have seen how VoIP offers greater flexibility, expanded integration options and long-term cost saving opportunities. Now we are beginning to see how the voice quality and user experience can be much better than we are accustomed to, and there will be no looking back."

"Polycom is delivering the UltimateHD experience that focuses on all elements of communication and collaboration to provide the best possible quality, performance and overall value to customers," said Sunil Bhalla, senior vice president and general manager of voice communications at Polycom. "Polycom HD Voice sets a new standard for voice quality by leveraging our proven, market-leading technologies. Customers want to communicate more efficiently and Polycom HD Voice is like going from AM radio to CD quality, providing enhanced clarity in communications and collaboration, and ultimately improved productivity. Polycom HD Voice is the third element in our UltimateHD strategy, which also includes currently shipping HD video bridges and an HD video conference recording and streaming server. Next quarter we will add the fourth component, HD video conferencing systems, delivering an unrivaled end-to-end solution."

SoundPoint IP 650 -- the first VoIP phone with Polycom HD Voice

Leveraging the benefits of Polycom HD Voice, the SoundPoint IP 650 desktop phone sets the new standard for IP telephones. The SoundPoint IP 650 with Polycom HD Voice delivers unprecedented voice quality, high performance, and expandability to appeal to all users, including executives who require advanced features and applications, and telephone attendants who need multiple line support and enhanced call handling capabilities.

The SoundPoint IP 650's innovative micro-browser feature gives managers and executive users the ability to access XHTML-based applications, from the latest stock quotes to the weather forecast. The phone's Polycom SIP 2.0 software fully supports Asterisk iPBX and Microsoft Live Communications Server 2005 for telephony and presence, and integrates with Microsoft Office Communicator instant messenger client. The phone also features a USB port for future applications.

The SoundPoint IP 650 accommodates six lines in a standalone mode and up to 12 lines or 24 concurrent calls when equipped with up to three SoundPoint IP Expansion Modules, as an attendant console. The phone supports shared call/bridged line appearances that enable effective phone interaction between executives and administrative assistants and the busy lamp field (BLF) functionality that allows phone attendants to monitor the on-hook and off-hook status of key contacts and dispatch incoming calls for those contacts more efficiently.

The SoundPoint IP 650 delivers all of its capabilities through an intuitive user interface, featuring a high-quality, backlit 320x160 graphical grayscale LCD display, an easy-to-navigate menu, and a combination of twenty- six dedicated hard keys and four context-sensitive soft keys for one-button access to essential telephony features. The phone is easy to install and set up, with a two-port Ethernet switch, built-in auto-sensing Power over Ethernet circuitry, and can be centrally provisioned and upgraded from an FTP, TFTP, HTTP, or HTTPS server.

Pricing and Availability

The SoundPoint IP 650 will be available for order in Q4 2006 in most countries around the world for a MSRP of US$449 through Polycom certified VoIP resellers.

Polycom reserves the right to modify future product plans at any time. Products and/or related specifications referenced in this press release are not guaranteed, and will be delivered on a when and if available basis.

Source: Polycom Inc. 

Posted by Dal at 12:11 PM | Direct Link Posted to VoIP Hardware

This release of PIKA Connect for Asterisk, available in November, allows Asterisk-based applications to use Skype to receive incoming and/or make outgoing calls, provides access to the calling Skype ID profile information (caller ID), and has touch tone (DTMF) detection capabilities.  Skype clients running on Windows based PCs are connected to the channel driver via PIKA’s AllOnHost™ (host-based) voice processing technology.  Skype clients can be distributed across an unlimited number of Windows PCs to achieve the density requirements the voice application may require. 

“Given the ever increasing use of Asterisk in business communication solutions and the rapid adoption of Skype as a business-to-business communication tool it seemed natural to do a mashup of these two very popular technologies,” said David Clarke, Business Development Manager at PIKA Technologies. “Our second generation PIKA Connect for Asterisk channel driver makes it a seamless exercise to Skype-enable your Asterisk based solution.”

PIKA has already implemented this technology in their own corporate PBX system, allowing customers to contact them directly via Skype. Using Skype, customers are able to call PIKA from anywhere in the world, free of charge, using any computer with Skype installed.  It is as simple as a click of a button on the PIKA website: www.pikatechnologies.com   The Skype call in turn is carried over the Skype network and terminated directly on PIKA’s Asterisk based auto attendant.  No extra phone lines or ports on the PBX are required.  Once answered, callers can navigate the auto attendant in the normal fashion using the Skype dial pad (touch tone digits).  All internal extensions as well as speed dials for sales and support are accepted. 

PIKA is currently enlisting Beta candidates for the second generation PIKA Connect for Asterisk.  If you are an Asterisk user or developer interested in running a Beta trial, please contact David Clarke at david.clarke@pikatech.com or skype davidclarkepika

Posted by Dal at 11:59 AM | Direct Link Posted to Asterisk Development