Wednesday, June 27, 2007

NewMarket Technology, Inc. Partners with Digium, Inc., the Asterisk Company

DALLAS, TX--(Marketwire - June 27, 2007) - IP Global Voice Inc., the VoIP subsidiary of NewMarket Technology Inc. (OTCBB: NMKT), announces market and product expansion with Asterisk, the "open source" standard for telecommunications systems. The IP Global Voice, Inc. product line is marketed under the Xiptel brand name. IP Global Voice has entered into a contract with Digium, Inc., the Asterisk company and the original creator and primary developer of the industry's first open source telephony platform. IP Global Voice has earned the coveted "dCAP" certification from Digium, and is positioned as Digium's key certified partner located in Northern California.

IP Global Voice currently contributes approximately $10 million in annual profitable sales to NewMarket's overall revenue. Last year, NewMarket reported $77.6 million in annual revenue with $5 million in net income. The partnership with Digium can substantially accelerate the expansion of IP Global Voice's VoIP Small and Medium Business (SMB) market share.

Asterisk has many hundreds of motivated and qualified professionals and engineers contributing daily to its development. Unlike proprietary closed systems like Cisco, Avaya, Nortel, and Alcatel, which tend to be more expensive and less flexible, XIPTEL's focus on Asterisk lowers costs, and takes maximum advantage of the enormous flexibility built into the Asterisk ecosystem.

"IP Global Voice offers a wide range of hosted and premise based telephone services," said Peter Geddis, CEO for IP Global Voice. "Open source technologies and developments are changing the way businesses are built. From desk top applications to core systems including telephone and data network applications, the open source direction is taking off like wildfire. Virtually every major carrier and service provider has Asterisk working in their lab or in field tests. The large legacy companies know it's coming and they are desperately trying to figure out how to defend themselves. It's clearly a case of 'get on board or get out of the way.'"

Asterisk, coupled with other open source technologies, is a lower cost, highly flexible replacement for expensive software platforms, similar to the one on which the XIPTEL offering was originally launched. The old one will be replaced. This is not the company's first move in the open source direction. XCCP, XIPTEL's back office system designed and built internally from the ground up, is a high performance, easily scalable environment built entirely on proven open source technologies, using Linux, Apache, MySQL, and PHP. XCCP has been running XIPTEL's back office with key functions since early 2006.

To be added to NewMaket's e-mail list for shareholders and interested investors, please send an e-mail to

About IP Global Voice, Inc. (

IP Global Voice continues to expand its business service and product offerings to offer a full range of managed IT services for small, medium and large commercial customers. IPGV, with operating divisions Corsa Network Technologies and XIPTEL Communications, offers full end-to-end IP based solutions for business customers, including hosted and premise based VOIP telephone systems, IP PBX hardware and managed systems solutions, high-end network security technologies, and other cutting edge network and productivity enhancement products. IP Global Voice Inc. offers dozens of other products and services for every business need having partnerships with high profile companies such as Digium/Asterisk, 3Com, Riverbed, Juniper Networks, Ae6, Bradford Networks, Mirapoint, Tipping Point, and others.

About Digium (

Digium®, Inc., the Asterisk company, is the original creator and primary developer of Asterisk, the industry's first open source telephony platform. Digium provides hardware and software products, including AsteriskNOW™, the complete open source software appliance; Asterisk Business Edition™, the professional-grade version of Asterisk; and the Asterisk Appliance™, hardware-based telephony solution, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services, including consulting, technical support, and custom software development.

Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.

Code for Asterisk has been contributed from open source software engineers around the world. Currently boasting over two million users, Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

About NewMarket Technology Inc. (

NewMarket helps clients maintain the delicate balance between maintaining legacy systems and gaining a competitive edge from the latest technology innovations. NewMarket provides certified integration and maintenance services to support the prevailing industry standard solutions such as Microsoft, Cisco Systems, SAP, Siebel and Sun Microsystems. Concurrently, NewMarket continuously seeks to acquire emerging technology assets to incorporate into an overall product portfolio carefully packaged to complement the prevailing industry standard solutions.

NewMarket delivers its portfolio of products and services through its global network of Solution Integration subsidiaries in North America, Latin America, China and Singapore.

NewMarket maximizes shareholder return on investment by independently listing consolidated regional and emerging technology subsidiaries in order to issue subsidiary stock in shareholder dividends. Currently, NewMarket Technology is the parent company to two publicly listed regional subsidiaries, NewMarket Latin America, Inc. (PINKSHEETS: NLAI) and NewMarket China, Inc. (OTCBB: NMCH).

NewMarket ranked Number Five on Deloitte's 2006 Technology Fast 500, a ranking of the 500 fastest growing technology, media, telecommunications and life sciences companies in North America. Rankings are based on percentage revenue growth over five years, from 2001-2005. The Company grew from less than $1 million in revenue in 2001 to over $50 million in profitable revenue in 2005. In 2006, the company continued its rapid growth, reporting $77.6 million in revenue with a net income of $5.8 million.


This press release contains forward-looking statements that involve risks and uncertainties. The statements in this release are forward-looking statements that are made pursuant to safe harbor provision of the Private Securities Litigation Reform Act of 1995. Actual results, events and performance could vary materially from those contemplated by these forward-looking statements. These statements involve known and unknown risks and uncertainties, which may cause NewMarket's actual results in future periods to differ materially from results expressed or implied by forward-looking statements. These risks and uncertainties include, among other things, product demand and market competition. You should independently investigate and fully understand all risks before making investment decisions.

Monday, June 25, 2007

NeoPhonetics CEO Kicks Off ClueCon

NeoPhonetics CEO Kicks Off ClueCon Telephony Developer Conference June 26, 2007
Posted by Michael Becce on Monday, June 25, 2007 - 12:28 pm:

TINLEY PARK, IL and CHICAGO, IL — (June 25, 2007) — NeoPhonetics, a telephony provider that designs, implements and supports custom VoIP systems for enterprises, today announced that the company’s co-founder and CEO, Chad Agate, will be kicking off the upcoming ClueCon Telephony Developer Conference with his session on open source vs. proprietary telephony solutions on June 26th from 9-9:30am CDT.

“ClueCon is a great forum for developers to get together and discuss the current issues surrounding our industry today,” said Chad Agate, CEO of NeoPhonetics. “Open source telephony, especially Asterisk, has created a tremendous amount of flexibility and opportunity for developers working with IP-based solutions. I am looking forward to addressing the benefits to open source, along with specific direction on how to work with Asterisk-based systems.”

Agate is responsible for the day-to-day management, business development, infrastructure design and business partnerships for NeoPhonetics. Prior to NeoPhonetics, Chad co-founded The Cipher Group. At Cipher, Chad utilized his skills in building IT services solutions for businesses and education organizations. Within four years, Cipher grew to $1.5 million in service revenue. Previously, Chad worked as a senior network engineer at DeVry University, maintaining and supporting DeVry’s university network of over 4,000 users. Chad attended the University of Illinois and The DeVry University. Currently, Chad is a member of the DeVry Telecommunications Management Advisory Board.

ClueCon will take place June 26-28 in Chicago. Additional information is available at

About NeoPhonetics
NeoPhonetics designs, implements and supports custom VoIP telephony systems for enterprises with at least fifty employees. With a specialty in open source Asterisk® installations, NeoPhonetics creates solutions that offer more features, flexibility and cost effectiveness than traditional telephony systems. The company differentiates itself from other providers by offering on-site network design and integration as well as 24/7 support. In 2006, NeoPhonetics received the “Entrepreneurial Company of the Year Award” by Frost & Sullivan for its role as an emerging company with vast potential in the enterprise telephony equipment services market. Additional information can be found at

NeoPhonetics and the NeoPhonetics logo are registered trademarks of NeoPhonetics. Asterisk and the Asterisk Business Edition are registered trademarks of Digium Inc.

Media Contacts:

Janine Savarese
MRB Public Relations

Kristen Keller
MRB Public Relations

Wednesday, June 13, 2007

Introducing the Digium TE420 - PCIe

Digium's TE420 PCI Express card provides termination of up to 60 channels of voice or data across four E1, T1, or J1 interfaces in a PCIe x1 form factor. Selectable on a per-port or per-card basis, the TE420 allows E1 and T1 circuits to be mixed with full channel synchronization. Supporting PCIe x1, the TE420 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 without considerations for voltage selection or lane size.

The TE420 may be combined with Digium's VPMOCT128 Octasic DSP-based echo cancellation module (not shown in picture). The VPMOCT128 provides the G.168 algorithm which has been labeled a benchmark for echo cancellation and performs 128ms (1024 taps) of echo cancellation across all 128 channels in E1 mode or all 96 channels in T1/J1 modes. Bundled with the VPMOCT128, the product SKU is TE420B.

Digium has designed the TE420 to be fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk.

The TE420 supports industry standard telephony protocols including North American and European Primary Rate signaling as well as standard Robbed Bit, Channel Associated Signaling in addition to standard PPP, HDLC, and Frame Relay data modes.

Digium® Launches Line of PCI Express Cards for Use with Asterisk®-

HUNTSVILLE, Ala.--(BUSINESS WIRE)--Digium®, Inc., the Asterisk® company, today announced a new family of telephony interface cards based on the PCI Express (PCIe) format. PCIe is fast becoming the dominant form factor in expansion cards for server, workstation and desktop systems because it offers numerous performance benefits over traditional PCI and PCI Extended interface formats. Digium is rolling out PCIe cards across its full product family and will continue to support interface cards based on the PCI and PCI Extended formats.

Digium is the creator and driving force behind Asterisk, the open source voice communications software deployed by more than two million servers and serving well over 10 million people today. The company’s hardware, including the new PCIe cards, is designed to help customers realize the full power and flexibility of Asterisk by meeting precise requirements for scalability; network connectivity; and support for IP, traditional analog or mixed telephony lines.

“PCI Express is quickly becoming the industry preference for server and workstation interfaces for expansion cards,” said Bill Miller, vice president of product management and marketing at Digium. “Digium’s goal in offering a full line of PCIe cards, and in continuing to support PCI and PCI Extended formats, is to offer our customers and partners the widest array of choices in how they use and deploy Asterisk solutions.”

Digium’s four- and two-port digital T1 and E1 PCIe interface cards, the TE420 and TE220, eliminate voltage and slot considerations associated with standard PCI cards. They are available immediately from the company and its authorized resellers. The TE420 will retail for $1,195 USD and the TE220 for $695 USD. Both are compatible with Digium’s existing VPMOCT series of hardware echo cancellation modules. Digium is scheduled to release analog and other cards for PCI Express in Q2 and Q3 2007.

About Asterisk

Code for Asterisk®, originally written by Mark Spencer of Digium®, Inc., has been contributed from open source software engineers around the world. Currently boasting over two million servers, Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

About Digium

Digium®, Inc., the Asterisk® company, is the original creator and primary developer of Asterisk, the industry’s first open source telephony platform. Digium provides hardware and software products, including AsteriskNOW™, the complete open source software appliance; Asterisk Business Edition™, the professional-grade version of Asterisk; and Asterisk Appliance™, the hardware-based telephony solution, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services, including consulting, technical support and custom software development.

Used in combination with Digium’s telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium’s offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways. Additional information can be found at

The Digium logo, Digium, Asterisk, Asterisk Business Edition, AsteriskNOW, Asterisk Appliance and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.

Digium announces new line of PCIe Hardware - Digium TE220

Digium announced today that it's new line of T1 hardware will support PCIe... This new hardware answers the cries of many Asterisk users by supporting the most modern type of PCI slot.


Click here for more information on the new Digium TE220

DuSLIC (Dual-channel Subscriber Line Interface Circuit)

The latest member of its DuSLIC (Dual-channel Subscriber Line Interface Circuit) family of products for voice and VoIP-enabled devices has been announced by Infineon Technologies.

DuSLIC-xT is an energy-efficient CODEC / SLIC that offers best in class cost position for a complete voice enabled system. It reduces the overall bill-of-material for voice and VoIP systems including cable modems, analog terminal adaptors (ATA) and VoIP terminal applications by up to 50% depending on system specifications and requirements. DuSLIC-xT also helps to shrink the line interface unit footprint by up to 40% compared to current solutions.

The DuSLIC-xT includes an integrated high definition (HD) audio interface which allows the device to provide an interface between a PC or laptop and an analog phone. Such voice-integrated PC applications enable users to make low-cost, high-quality VoIP telephone calls by connecting an analog phone directly to a PC or laptop, says the company.

“As VoIP gains mass market acceptance, our customers are looking to gain additional business in voice terminal applications, while extending to new untapped markets,” said Christian Wolff, senior vice president of the Communication Solutions business group and general manager of the Wireline Access business unit at Infineon. "DuSLIC-xT simplifies system design by minimizing the development time and the bill-of-material (BOM) and paves the way for PC-based VoIP applications that will further fuel customer adoption".

DuSLIC-xT (PEF 3201, PEF 3101) is offered as either a dual-channel or single-channel solution. The product combines one dual-channel CODEC and two single channel high-voltage SLICs in a PG-TQFP-100 package. Engineering samples including system design package will be available in July 2007, says the company.

Monday, June 11, 2007

Sangoma Releases ISDN BRI Card - A500

Sangoma's A500 BRI Card Supports Asterisk(R) And Other Open Source Telephony Projects

STOCKHOLM, SWEDEN and TORONTO, ONTARIO--(Marketwire - June 11, 2007) - Sangoma Technologies Corporation (TSX VENTURE:STC), a leading provider of connectivity hardware and software products including VoIP, TDM voice, WANs and Internet infrastructure, has released its new A500 BRI card designed to support the European and global open source telephony markets.

"Our A500 BRI card was developed in response to the continuing world market demand for a quality BRI that works flawlessly when connected to any ISDN switch," says Sangoma Technologies president and CEO David Mandelstam. "The solution uses a well-tested and widely-certified ISDN code set, ensuring full resolution of connection issues that have plagued BRI connectivity on open source telephony systems up until now."

The BRI fully supports the ISDN S/T bus as either an NT or TE device using a reversed module design that avoids jumpers. Distinctive features include:

- Expandable from 2 to 24 ports of 2B+D interfaces for a total of 48 ISDN channels

- PCI or PCI Express bus support

- Optional carrier-grade echo cancellation

- Support for all open source telephony projects

"Over the years we have built our business by developing hardware that simply works, the first time," adds Mandelstam. "As an integral part of Sangoma's AFT design family, the new A500 inherits the compatibility, enhanced performance and reliability of its siblings."

The A500 supports the same digital processing and highly-compatible PCI/PCI Express interfaces as the AFT series. It provides the benefits of hardware-based echo cancellation and voice enhancement for scalability for various installations.

The card will enter full production in mid-July.

Sangoma continues to bring innovation, scalability and higher density solutions to market at the request of its enterprise and telco customers. Sangoma's current solution set can be found at:

See Sangoma's solutions at VON Europe Spring 2007, June 11-14 - Stand 703.

A more detailed technical discussion is available on Sangoma's Technical Wiki.

About Sangoma Technologies Corporation

Founded in 1984, Sangoma Technologies Corporation is a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure. The company develops and manufactures voice and data communication products including the industry-leading series of Advanced Flexible Telecommunications (AFT) PCI cards.

Sangoma Technologies Corporation is publicly traded on the TSX Venture Exchange (TSX VENTURE: STC - News).

Sunday, June 10, 2007

Sopranos Restaurant Ending-

Pure Bullshit!-

I suppose they are leaving us all hanging on for the very Soprano Christmas...


Wednesday, June 06, 2007

Introducing the Aastra 9116LP - PBX Line Powered

Please visit for all of the latest from Aastra
More flexibility and options in a value priced Business Phone

The 9116LP from Aastra is a fully featured Enterprise-grade telephone
offering a wide array of the most sought after features in a rugged dayto-
day office telephone. Based on our popular 9116 model, the 9116LP
incorporates PBX line power, improved user interfaces and a number of
other feature enhancements that can add value and flexibility to any office
environment at a price that may surprise you.

With a three line adjustable display, the 9116LP is fully compatible with
Call Display and Visual Call Waiting offering prompts in English, French
and Spanish. Additional information is provided by the LED indicator
which flashes for message waiting, new call indication, incoming call,
extension in use, feature activation and hold. Call management features
include an 80 name/number callers list and 20 name/number directory.
With speakerphone and 15 autodial positions, the 9116LP offers flawless
performance in any environment.

Flexible Deployment Capability
The 9116LP can be deployed in a wide variety of business environments. Whether
using standard PSTN service, Centrex or PBX, the 9116LP reliably delivers flawless
performance. With a choice of powering via the PBX line or with the included
DC power supply, it can be used in most business settings and will support all
common CLASS and Voltage Message Waiting indication protocols. And, with a
fail to POTS feature, the 9116LP will continue to provide basic telephone service
in the event of a power failure.
More Caller Information
The three line adjustable display with contrast control supports Caller ID and
Call Waiting Display to provide more information on incoming callers. The
bright, well-positioned LED is also used to provide a variety of information
from message waiting indication to extension in use. These features enhance
productivity by quickly providing “at a glance” information to the user.
Enhanced Call Management
The 9116LP offers storage capacity for Callers List, Personal Directory and Last
Number Redial as well as 15 “one touch” autodial positions employing 8 memory
keys to facilitate easy of use. On hook dialing using the built-in speakerphone
provides additional calling options and the speakerphone can be disabled
for office designs not suitable for speakerphones. Predefined hard keys for
functions such as Dial, Redial, Hold, Flash and Goodbye allow for efficient call
handling in any environment.

Monday, June 04, 2007

Digium Transcoder Hardware g.729 - TC400B

have you seen the new Digium TC400B?

If not look at this...

Cbeyond -VoIP Solutions Seminar

ATLANTA - Monday, June 04, 2007

Cbeyond®, Inc., a leading IP-based managed services provider to small businesses, announces CTO Chris Gatch will be a speaker at the VoIP Solutions Seminar at Emory's Goizuetta Business School held today. The event, sponsored by Digium and Asterisk, will allow technology professionals to learn about the latest trends in VoIP technology. Gatch will discuss the SIP Forum's SIP Trunking industry specification, SIPconnect and how the Asterisk Appliance complements Cbeyond's industry-leading SIP trunking service, BeyondVoice with SIPconnect.

Additionally, Mark Spencer, CTO of Digium will provide an industry update about Asterisk including one of Digium's latest innovations, a stand-alone embedded Asterisk-based PBX targeted for small to medium- sized businesses. Gatch has served as an editor of the SIPconnect technical specification published by the SIP Forum. He also served on the board of the Cisco BTS 10200 Users Group, the Service Provider Board of the International Packet Communications Consortium (IPCC) and he presently serves on the Board of the SIP Forum. The free seminar be held in the Boynton Auditorium (Room 130), located at 1300 Clifton Road NE, in Atlanta and begins at 4:30 p.m. For more information and to register for the seminar, visit: 572/ItemID/14029/Default.aspx? selecteddate=6/4/2007

About Cbeyond

Cbeyond, Inc. (NASDAQ: CBEY) is a leading IP-based managed services provider that delivers integrated packages of local and long-distance voice along with mobile and broadband Internet services to more than 29,000 small businesses in Atlanta, Chicago, Dallas, Denver, Houston, Los Angeles and San Diego. Cbeyond offers more than 20 productivity-enhancing applications including BlackBerry®, voicemail, email, Web hosting, fax-to-email, data backup, file-sharing, and VPN. Cbeyond manages these services over a private, 100-percent Voice over Internet Protocol (VoIP) facilities-based network. For more information on Cbeyond, visit ( Website:

NetVanta 150 Wireless Access Point

NetVanta 150 Wireless Access Point

Designed exclusively to co-exist with other ADTRAN NetVanta routing and switching solutions, the NetVanta 150 is a lightweight 802.11a/b/g Wireless Access Point. It extends a wireless reach to NetVanta networks. Its small form factor allows it to be mounted in discrete locations and the 802.3af compliant PoE port eliminates the need to be near a power outlet. The NetVanta 150 is centrally managed through an ADTRAN Wi-Fi access controller unit. ADTRAN Wi-Fi access controller units currently include the NetVanta 1335, NetVanta 3448, NetVanta 3430, NetVanta 3130, and NetVanta 3120. All of these Wi-Fi access controllers can manage up to eight NetVanta 150s.

802.11a/b/g Dual Radio Wireless Access Point
Robust, business-class security including WPA and WPA2 support
WMM QoS (Wi-Fi Multimedia Quality of Service)
802.1x user authentication
802.3af compliant PoE support for flexible deployment
Supports eight Virtual Access Points per radio
Centrally managed through a NetVanta Wi-Fi access controller

Download the NetVanta 150 data sheet

The NetVanta Series Goes Wireless!

For the best pricing avalible on this and other Adtran hardware please visit us online @

Introducing the new Wi-Fi enabled NetVanta 1335 Multiservice Access Router and the NetVanta 150 Wireless Access Point (WAP).
Previously, small and medium businesses did not have an affordable Wi-Fi solution designed specifically for their needs. Home systems are not secure enough and large enterprise systems are too expensive and management-intensive.

Sales, medical, retail, manufacturing, hospitality, education, and collaborative ventures are a few potential customers. This solution also works for situations where traditional cabling and powering options are too expensive. Basically, NetVanta Wi-Fi is designed for any SMB or distributed enterprise that needs converged WAN / LAN / WLAN. NetVanta solutions are ideal for site or multi-site connectivity, or to add a wireless reach to an existing NetVanta network. Help your customers un-wire with the NetVanta Series!

Linksys iPhone vs Apple iPhone

Pricing and Availability

The Linksys Wireless-G Phone for Skype allows users to make calls to other Skype users from anywhere in the world, as long as there is wireless Internet access that does not require browser authentication. It provides all the convenience of a wireless handset with an extensive list of features, including high-quality voice reception and an intuitive color display.

"The great feature of the Linksys Wireless-G Phone for Skype is that you don’t need a computer anymore to use Skype but also you can use it wherever you need it!" said Tunji Akintokun, UK and Ireland Country Manager at Linksys. "With the built-in WiFi-finder it is possible to search and connect to WiFi networks meant for free and public availability in the neighbourhood, which could theoretically be anywhere around the world."

The Wireless-G Phone for Skype makes Skype portable by integrating it into a Wireless-G handset, allowing users to place calls from any location where they can connect to a wireless network meant for free and public availability and that does not require browser authentication.

No need for a switched-on computer; the new Linksys Wireless-G Phone for Skype connects directly to wireless networks. With the help of the integrated WiFi-finder it is easy to search and connect to a suitable wireless network (802.11 b/g) meant for free and public availability. The new member of the iPhone family even can connect to WEP and WPA encrypted networks and can store preferred networks on the handset.

Some features of the Linksys Wireless-G Phone for Skype:

Use Skype without a computer, including SkypeOut™, SkypeIn™, Skype Voicemail
WiFi-finder: search and connect to any wireless network that does not require browser authentication
Intuitive 65K color display
Supports WPA and WEP
Outdoor/Indoor range: up to 200 meters / up to 75 meters
Weight: 100 grams

Linksys iPhone Family

The iPhone family of handheld devices harnesses the power of the Internet to enhance voice communications, integrate compelling information services, and deliver access to multimedia. In short, Linksys iPhone voice solutions and products give consumers the ability to do more with their phone than talk.

The Linksys iPhone family currently available in selected countries in Europe includes:

Linksys iPhone® Cordless Internet Telephony Kit for Skype - CIT200
Linksys iPhone® Dual-Mode Internet Telephony Kit for Skype - CIT300
Linksys iPhone® Dual-Mode Internet Telephony Kit with Integrated Skype - CIT400
New! Linksys iPhone® Wireless-G Phone for Skype WIP320
Linksys iPhone® Wireless-G IP Phone - WIP330

Pricing and Availability

Vyatta and Digium Partner

For information on Asterisk hardware please visit -
Vyatta and Digium, the Asterisk company, has entered a partnership to collaborate on open-source voice and data networks. Vyatta, a provider of open-source data communications, and Digium, an open-source voice communications provider, will work together to make it easier for customers to purchase, deploy, and maintain high-quality, integrated voice and data platforms, the companies said in a joint statement.

SMB and enterprise customers are increasingly seeking telecom and data communications solutions that are more flexible, efficient, and tailored to their specific needs. Open source-based products are uniquely capable of rapid integration and feature flexibility, making them an ideal choice as “unified communications” move up on the IT and business priority list.

The partnership involves both technology and marketing initiatives, and includes efforts to improve VoIP Quality of Service (QoS) and security features in both Digium and Vyatta products. The companies will also focus on making it easier for customers to install and configure a secure, integrated voice and data environment using Digium’s Asterisk and Vyatta’s open-source networking solutions that include routing, firewall, and VPN functionality.

“A partnership between Vyatta and Digium is a natural fit that will leverage our very complementary core competencies to enhance a common mission – to provide SMBs and enterprises with open-source alternatives to expensive and proprietary solutions,” said Bill Miller, vice president of marketing and business development at Digium.

“Voice over IP is now a mainstream technology, and we are poised to do the same with open-source networking,” said Dave Roberts, vice president of strategy and marketing at Vyatta. “By combining the efforts of Vyatta and Digium, the two technology leaders in our respective areas, and working together to expand our marketing reach, we can drive rapid adoption of our solutions.”

VoIP Providers to pay USF- Universal Service Fund

For more on VoIP Service Plans please visit us online @

Internet telephone companies like Vonage must contribute to the Universal Service Fund (USF), a federal appeals court ruled Friday afternoon. If VoIP providers follow the lead of local, long-distance and wireless services that already pay into the USF, the costs will be passed on to consumers.

The decision is limited to VoIP providers that interconnect with the public switched telephone network (PSTN). The USF subsidizes phone service in under-served and rural areas. Through the E-rate fund, the USF also subsidizes Internet connections in schools and libraries.

After the Federal Communications Commission (FCC) ruled last year VoIP providers were obligated to pay USF fees, Vonage challenged the decision. The Holmdel, N.J.-based company claimed the FCC exceeded its authority in the ruling, in addition to overestimating how much VoIP providers should pay.

"We conclude that the commission has statutory authority to require VoIP providers to make USF contributions," the U.S. Court of Appeals for the District of Columbia wrote in its opinion.

As for the amount VoIP providers must pay, Vonage challenged the formula used by the FCC. Traditional telephone companies pay 11.7 percent of their long distance revenue into the fund. But since Internet calls can't be broken into local and long distance categories, the FCC set a "safe harbor" estimate of 64.9 percent for VoIP providers.

The FCC analogized the VoIP safe harbor with wireless companies, which pay 37.1 percent of their long distance revenues into the USF.

"Because VoIP's functionality and customer profile differ from those of other technologies, reasoning by analogy in this way invites some inevitable imprecision," the court ruled. "We agree with Vonage that this difference in capabilities renders the VoIP/wireline toll service analogy imperfect. Perfection, however, is not what the law requires."

Brooke Schulz, senior vice president for corporate communications at Vonage, insisted in a statement that Vonage supports expanding USF contributions to VoIP carriers. "Our case simply challenged the funding methodology, not the underlying concept of USF," Schulz said.

Last year, Medley Global Advisors estimated the FCC decision would add approximately $1.30 a month to each VoIP subscriber's bill.

"I am pleased that the court has affirmed the commission's action, which ensures that USF contribution obligations are administered in a competitively and technologically neutral manner on all phone providers, including interconnected Voice over Internet Protocol (VoIP) providers," FCC Chairman Kevin Martin said in a statement.

Aspect Software- Digium and Asterisk

for more information on this and other Asterisk Call Center Solutions please visit us online @

.e4 Telephony Center

Open Source Voice Communications Technology Works with Unified Solution to Yield Cost-Savings, Reduce Complexity and Increase Flexibility

CHELMSFORD, Mass., 31 May 2007- Aspect Software, Inc., the world’s largest company solely focused on the contact center, announced today it has successfully deployed the Digium® Asterisk Business Edition™ Internet Protocol (IP) private branch exchange (PBX) at its new corporate headquarters and is now benefiting from the cost-savings and flexibility of an open source deployment. It is being used to support 500 Aspect Software business employees, as well as technical service support personnel in its mission-critical contact center environment. The industry-leading open source IP PBX is interoperating with Aspect® Unified IP™, the unified solution from Aspect Software that streamlines routing, reporting and administration by handling voice, email, and web interactions on a single, scalable and highly reliable platform.

What communication products or services does your business need? (Please check all that apply)

Business Phone Systems
Messaging/Forwarding Systems

As a reseller of Digium’s Asterisk Business Edition, Aspect Software recognizes the value that an open source IP PBX can bring to the organization, particularly when interoperating with its session initiation protocol (SIP)-based Voice over Internet Protocol (VoIP) contact center solution, Aspect Unified IP.

“We needed a reliable and flexible PBX platform that offered high-quality voice interactions for both our general business and contact center employees and have found that the Asterisk solution has delivered,” said Jamie Ryan, chief information officer at Aspect Software. ”The implementation was incredibly easy, the quality of the calls has been great and most importantly, Asterisk coupled with our unified contact center solution is the first step in our goal of creating a full fledged unified communications offering.”

Because of its SIP-based VoIP capabilities, the Asterisk Business Edition can readily interoperate with the company’s leading unified contact center solution, Aspect Unified IP, which provides the ACD, voice portal, recording and quality management capabilities to help the Aspect Technical Services organization manage its critical customer service interactions.

Mark Spencer, founder and chief technology officer at Digium, the original creator and primary developer of Asterisk®, said: ”Asterisk implementations within organizations of every size are rapidly growing because of the openness and flexibility that it brings, as validated by Aspect Software. Aspect Software, a Digium partner, brings a unique perspective based on its extensive knowledge of open and closed source voice communications technology and unified communications. Aspect Software’s choice of Digium Asterisk for their corporate headquarters further demonstrates the quality of the open source solution and shows that businesses and their contact centers can easily leverage a VoIP infrastructure and experience the benefits of SIP-based solutions.”

Aspect Software is also planning on expanding the deployment of Asterisk Business Edition to other offices in the near future.

Aspect Software provides support for the Digium open source internet protocol (IP) PBX, the Asterisk Business Edition – a professional-grade version of the industry’s first open source IP PBX – for customers of its Unified and Signature product lines. The Aspect Software packaged offering includes licenses, optional SIP phones, optional application servers and IP gateways, interoperability with Aspect contact centers products, as well as installation, deployment and post-deployment support.

About Aspect Software

Aspect Software, Inc. founded the contact center industry and is now the world’s largest company solely focused on Internet Protocol (IP) and traditional voice-based products and services for customer service, collections, and sales and telemarketing business processes. Each day, Aspect Software powers more than 125 million customer-company interactions at thousands of in-house and outsourced contact centers around the globe. Its trusted Signature product line offers automatic call distributors (ACDs), dialers, voice portals and computer telephony integration (CTI). The company’s leading Contact Center Performance Optimization product line provides workforce management, quality management, performance management and interaction optimization applications. And, its pioneering Unified IP Contact Center product line delivers a comprehensive, multichannel solution. Headquartered in Chelmsford, Mass., Aspect Software has operations across the Americas, Europe, Africa, the Middle East and Asia Pacific. For more information, visit .

Sangoma goes Platinum- ClueCon Telephony Conference

ClueCon Announces Sangoma Technologies' Platinum Sponsorship

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Sangoma Technologies become Platinum Sponsor for one of the most-awaited open source telephony conferences of the year.

Derwood, MD - June 4, 2007 - Cluecon announces the platinum sponsorship of Sangoma Technologies in the upcoming Open Source Telephony Conference. Existing for more than 20 years in the VoIP and telephony connectivity industry, Sangoma Technologies has provided infrastructure and solutions compatible with today's communications applications and protocols.

Sangoma Technologies will be represented by Chief Software Engineer, Nenad Corbic.
Nenad Corbic, a pioneer in the connectivity hardware and software development, will represent Sangoma Technologies in the said event. Nenad is a graduate of Ryerson University in Toronto and holds an Honors degree in Computer Engineering.

About Sangoma (
Sangoma's technological expertise is data and voice communications. From the earliest days, we have concentrated on delivering data across long distance networks efficiently and cost effectively. Today we are the leading provider of PC-based voice and data communications products to customers large and small in over 130 countries around the world.
Data and Voice communications can be complex, but technology can make it simple for the user. Sangoma's tools coupled with our rock-solid product reliability make complex WANs and voice networks something our customers never have to think about.
About ClueCon
ClueCon - is an annual 3-Day Telephony User and Developer Conference bringing together the entire spectrum of Telephony from TDM circuits to VoIP and everything in between. The presentations and discussions will cover several open source telephony applications such as Asterisk/OpenPBX, Bayonne, YATE and FreeSWITCH.
Each day of the conference is filled with presentations and Q&A sessions with many of the leaders in the industry including hardware engineers, programmers and project leaders. No short course or even full semester can deliver as much information and knowledge as this concentrated exposure to the front lines of Telephony.

June 26-28, 2007

Contact ClueCon:

Register at:

ClueCon will be held at:
Best Western Inn of Chicago
162 E Ohio Street
Chicago, Illinois 60611
Phone: (312) 787-3100
Fax: (312) 573-3136
Toll Free: (800) 557-2378 US/Canada

Sangoma Fax over T1

Sangoma Solves Industry-Wide Fax Quality and Reliability Issue on T1/E1 Digital Phone Systems
For Unbeatable Pricing on this or any other Asterisk Solution.

Soft PBX Users With T1/E1 Lines Can Now Use Them For Reliable Faxing
TORONTO, ONTARIO--(Marketwire - June 4, 2007) - Sangoma Technologies Corporation (TSX VENTURE:STC) a leading provider of connectivity hardware and software products including VoIP, TDM voice, WANs and Internet infrastructure, has created the industry's first cost-effective solution supporting fax over PC-based T1/E1 digital phone systems.

Until now, users of Asterisk® and other PC-based systems with T1/E1 connections needed a separate external analog line at an additional cost if they wanted reliable fax service. Fax machines connected to a PBX using T1/E1 connections would run inconsistently at best because the analog interface supporting the fax machine was not synchronized with the PSTN timing.

"Sangoma's latest firmware revisions include an option for the analog card to receive its clocking from a T1/E1 card via a simple two-connector cable," says Sangoma Technologies' CEO David Mandelstam. "Because the T1/E1 card runs on the network clocking, the analog card that connects the fax machine to the network also becomes completely synchronized, allowing for perfect, error-free faxing without any of the glitches caused by imperfect timing.

"Fax connectivity remains an absolute requirement for businesses, especially for mission critical documents like legal contracts," adds Mandelstam. "This breakthrough allows larger offices to make proper use of the T1/E1 lines that they are paying for without having to purchase and manage additional analog capacity just to support faxing."

Sangoma's fax solution is software agnostic and works with any operating system or application, including Asterisk®, FreeSwitchTM, CallWeaverTM, OPALTM and YateTM. All T1/E1 cards with hardware echo cancellation shipping after June 1, 2007 will have synchronous fax support enabled. This solution also solves similar issues with modem transmission.

About Sangoma Technologies Corporation

Founded in 1984, Sangoma Technologies Corporation is a leading provider of connectivity hardware and software products for VoIP, TDM voice, WANs and Internet infrastructure. The company develops and manufactures voice and data communication products including the industry-leading series of Advanced Flexible Telecommunications (AFT) PCI cards.

Sangoma continues to take an industry lead by providing its revolutionary and award-winning AFT Series T1/E1/J1 voice/data cards that are engineered for today's demanding soft PBX, IVR and VoIP applications.