New MP-11X Analog VoIP Gateways from AudioCodes Shipping

April 19, 2006 by Garrett Smith

AudioCodes has released a new series of flexible analog gateways, the MediaPack MP-11X series, in FXS/FXO spans ranging from 2-8 ports.

The MediaPack ™ Series Analog VoIP Gateways off affordable, feature-rich solutions for connecting legacy telephones, fax machines and PBX systems with IP-based telephony networks, and integrate seamlessly with new IP-based PBX architecture. MediaPack MP11X Series gateways are designed for interoperability with leading Softswitches, H.323 Gatekeepers and SIP servers.

New Models Include:

MediaPack Series (11x) Features Include:

  • Spans ranging from 2 to 8 analog ports
  • Selectable, multiple LBR coders per channel
  • T.38 compliant
  • Echo canceller, Jitter Buffer, VAD and CNG
  • Complies with MGCP, H.323 (v4) and SIP control protocols
  • Comprehensive support for supplementary services
  • Enhanced capabilities including MWI, long-haul, metering, CID and outdoor protection
  • Web management for easy configuration and installation
  • EMS for comprehensive management operations (FCAPS)


Linksys Ships PAP2T-NA Dual FXS Analog VoIP Adapter

April 13, 2006 by Garrett Smith

The Linksys PAP2T Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analogue phones or use one of the ports for a fax machine.

Each phone port operates independently, with separate phone service and phone numbers — like having two telephone lines. You’ll get clear reception and a reliable fax connection, even while using the Internet at the same time.

With Internet telephony, along with low domestic and international phone rates, an impressive array of special telephone features are available. Choose your preferred free local dialing US area code, regardless of where you live. Or add a virtual telephone number in any area code, forwarded to your Internet phone. You can even add a toll-free number.

The Linksys Internet VoIP Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephony service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and much more.

Features on the new Linksys PAP2T-NA Includes:

  • Two voice ports (RJ11) for analog phones or Fax machines with two independent telephone numbers
  • One RJ-45 port for 10/100 Mbps Ethernet connection
  • Supports Dynamic Host Configuration Protocol (DHCP)
  • Supports Session Initiation Protocol (SIP)
  • Supports multiple voice compression standards: G.711, G.726, G.729, and G.723.1
  • Supports Simultaneous calls with G.729 codec
  • Web-based configuration through a built-in web server
  • Supports DTMF tone detection and generation
  • Supports FSK Caller ID, DTMF Caller ID and FSK VMWI
  • Supports Echo Cancellation and Voice Activity Detection (VAD)
  • Password protected access and configuration
  • Supports auto-provisioning with remote firmware upgrade

Digium Announces New Hardware Products at VON

Digium Announces New Hardware Products at VON
New Transcoder and Echo Cancellations Cards Improve Call Quality and Communication between PSTN and VoIP Gateways

HUNTSVILLE, AL and SAN JOSE, CA — (March 14, 2006) – Source: Digium Press Release — Digium Inc., the creator of Asterisk™, and pioneer of open source telephony, today announced the availability of new hardware solutions to enhance Asterisk transcoding and echo cancellation performance for VoIP and PSTN gateways. These new products include the TC400P VoIP transcoding card and the TE420P and TE415P four-port T1/E1/J1/PRI cards with onboard hardware echo cancellation.

“Our product team is always working to develop solutions like these that ultimately further the open source movement in VoIP,” said Mark Spencer, president of Digium. “Not only are we constantly striving to improve Asterisk’s performance, but we also want to contribute to the overall VoIP experience, while keeping costs low.”

The TC400P provides hardware transcoding of VoIP codecs; decreasing Asterisk’s work load and providing improved CPU efficiency and an increase in channel density over a software-only solution. The TC400P provides Asterisk with full transcoding support and hardware acceleration for the G.723.1 and G.729A codecs.

The TE420P and TE415P improve upon Digium’s existing TE411P and TE406P. These premium interface cards provide carrier-grade echo cancellation, Voice Quality Enhancement (VQE), DTMF decoding, and tone recognition. Both cards minimize loads on the processor and PCI bus, and are designed to perform in the most difficult environments. The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels.

Digium designed these solutions to be fully compatible with existing software applications and is completely integrated with the Asterisk platform. Additionally, the open source driver supports an API interface for custom application development. All new solutions support industry standard telephony and data protocols, including Primary Rate ISDN (both North American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC and Frame Relay data modes.

About Digium

Digium is the original creator and primary developer of Asterisk, the industry’s first open source PBX and Asterisk Business Edition, the professional-grade version of Asterisk. Used in combination with Digium’s PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures.

Digium provides quality hardware and software products that enable telephony applications including legacy PBX, IVR, auto attendant, next generation gateways, media servers and application servers. Digium also offers a full range of professional services including consulting, technical support and customer software development services.

About Asterisk

Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed to from open source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, and VoIP packet protocols such as IAX, SIP and H.323. It supports US and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

The Digium logo, Digium, Asterisk, and the Asterisk logo are trademarks of Digium Inc. All other trademarks are property of their respected owners.

More from: Asterisk Garrett Smith

Cisco Introduces New Unified Communications System to Streamline Business Processes, Drive Productivity

Cisco Introduces New Unified Communications System to Streamline Business Processes, Drive Productivity

ORLANDO, Florida (VoiceCon Spring 2006) – March 6, 2006 – Source: Cisco Systems Press Release – Cisco Systems, Inc. today announced the Cisco® Unified Communications system, a new suite of voice, data and video products and applications specifically designed to help organizations of all sizes to communicate more effectively. It will allow customers to integrate their communications system with their IT infrastructure, streamlining business processes for the way effective businesses need to work today.

Based on the Cisco Service-Oriented Network Architecture (SONA) announced in December 2005, the Cisco Unified Communications system is an open and extensible platform for real-time communications based on presence, mobility and the intelligent information network. By using the IT data network as the service delivery platform, the system helps workers to reach the right resource the first time by delivering presence and preference information to an organization’s employees.

“The Cisco Unified Communications system is the first true second-generation Internet Protocol (IP) Communications system providing not just telephone services, but rather a rich communications environment that seamlessly integrates voice, video and data collaboration in one system. It is also the first new Cisco system to fully support Cisco SONA, announced in December 2005,” said Charles Giancarlo, chief development officer, Cisco Systems, Inc. “Cisco SONA extends the power of the network to optimize applications, processes and resources to deliver greater business benefits to enterprises. By building on Cisco SONA, Cisco Unified Communications leverages network intelligence to greatly simplify the day-to-day challenges of collaboration with colleagues.”

The Cisco Unified Communications system is based on Cisco’s industry-leading IP Communications portfolio including Cisco CallManager, Cisco Unity, Cisco MeetingPlace and Cisco IP Contact Center and now includes additional innovative products, applications, features and capabilities. New to the Cisco Unified Communications system are Cisco Unified Personal Communicator, Cisco Unified Presence Server and Customer Interaction Analyzer. Current customers will be able to upgrade their existing systems to take advantage of the new capabilities.

Cisco Unified Personal Communicator simplifies the way workers share information by helping them to communicate in real time. Its user-friendly GUI (Graphical User Interface) makes it easy to move through multiple communications applications. The Unified Personal Communicator bridges the gap between the stand-alone applications on the desktop, telephone and network. Using dynamic presence information, employees can search existing directories to locate contacts and simply “click to call” using voice and video, allowing them to exchange ideas face-to-face. The virtual nature of IP networks allows remote or traveling employees to securely access these tools from wherever they are.

The Cisco Unified Presence Server collects information about a user’s status, such as whether or not they are using a device such as a telephone, personal computer or video terminal at a particular time. Using this information, applications such as Cisco Unified Personal Communicator and Cisco Unified CallManager can help users connect with colleagues more efficiently by determining the most effective method of communication. The Cisco Unified Presence Server aggregates presence information from the network as well as Cisco Unified CallManager and third-party devices using SIP and SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) and then publishes that information to Cisco Unified IP Phones, Cisco Unified Personal Communicator and third-party services and applications such as IBM Lotus Sametime and Microsoft Live Communications Server (LCS) 2005.

The Customer Interaction Analyzer is being introduced to maximize effective communications with customers, a new approach to analytics in the contact center. It uses information from customer interactions, including self service and agent assisted interactions, to determine things like customer distress, agent distress, silence and word patterns. The data helps to give the conversations business context and can help a business to coach and train agents, make changes to processes and self service scripts based upon findings – ultimately creating better customer relationships and growth for the business.

Additional new features of the Cisco Unified Communications system include the following:

Cisco Unified CallManager 5.0 and Cisco Unified CallManager Express 3.4 and Survivable Remote Site Telephony (SRST) 3.4 now natively support SIP, effectively opening up the system to an emerging standards-based developer community while retaining the current security and resiliency features. A new program, SIP Verified, provides third-party verification for voice, data and video SIP endpoints. An initial set of vendors who have completed this testing is also announced.

Cisco Unified CallManager 5.0 is now available in a choice of operating models based on customer and channel partner preference. A new appliance model version based on Linux is available now and a version based on the existing open operating system model is scheduled to be available within 12 months.

“Miercom has exercised and reviewed key components of the entire Cisco Unified Communications system and after seeing it in action, we believe that Cisco has leapfrogged their competition in a number of areas,” said Ed Mier, principal, Miercom. “Cisco’s native implementation of SIP, which is interoperable with Skinny Client Control Protocol (SCCP) helps give customers investment protection for their system so that it can adapt as quickly as the standard does.”

Cisco continues to bridge the communications islands with innovative solutions building on the enterprise Wi-Fi (802.11) networks and the GSM public networks. In conjunction with leading wireless handset suppliers such as Nokia and RIM, Cisco will soon bring to market single and dual mode Smartphone solutions which drive enhanced productivity of mobile enterprise employees both inside and outside the office. These single device products allow users to reduce the communications complexity and help companies manage costs without losing the productivity.

“Because of their expertise in network infrastructure, Cisco was really the only vendor we considered when we decided to implement an end-to-end IP Communications solution,” said Mike DeDecker, network administrator at Warner Pacific Insurance Services. “As we move forward with our implementation and look for new ways to reduce costs and streamline processes, Cisco Unified Communications is at the top of the list. Cisco’s long-term networking background gives us the assurance we need when we’re looking to put applications in place.”

Cisco and its partners provide a lifecycle services approach to deploy and manage the Cisco Unified Communications system. New Cisco Operate Services for Unified Communications combine technical support services capabilities such as server replacement, application software updates and hardware and software problem resolution into one service that covers the entire system. To ensure proper deployment, Cisco is also offering Planning and Design Service Bundles and Optimization Services that are packaged for easy ordering.

Cisco also today introduced a number of new phones and updates to existing applications, as well as announcing Cisco Unified CallManager and Cisco Unified IP Phone are localized for China, Korea and Japan. For more information on the Cisco Unified Communications system visit:

About Cisco Systems
Cisco Systems, Inc. (NASDAQ: CSCO) is the worldwide leader in networking for the Internet. Information about Cisco can be found at For ongoing news, please go to:

Linksys Shipping WRTP54G-NA Wireless G Broadband Router

The Linksys WRTP54G-NA Wireless-G Broadband Router is really four devices in one box. First, there’s the Wireless Access Point, which lets you connect both screaming fast Wireless-G (802.11g at 54Mbps) and Wireless-B (802.11b at 11Mbps) devices to the network. There’s also a built-in 4-port full-duplex 10/100 Switch to connect your wired-Ethernet devices together. Connect four PCs directly, or attach more hubs and switches to create as big a network as you need. The Router function lets your whole network share a high-speed cable or DSL Internet connection.


  • All-in-one Internet-sharing Router, 4-port Switch, and 54Mbps Wireless-G (802.11g) Access Point
  • Shares a single Internet connection and other resources with Ethernet wired and Wireless-G and B devices
  • Two standard phone jacks enable feature-rich telephone service over your cable or DSL Internet connection
  • High security: Wi-Fi Protected Access? (WPA), wireless MAC address filtering, powerful SPI firewall

The fourth function is the phone adapter which enables high-quality feature-rich telephone service through your high-speed connection even while you’re surfing the Internet. There are two standard telephone jacks, each operating independently — like having two phone lines. With Vonage Service, you’ll get low domestic and international phone rates, Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and lots of other available special phone features. Choose any free local dialing US area code, regardless of where you live. Add a virtual phone number in any area code, or even a US-wide toll-free number.

To protect your data and privacy, the Wireless-G Broadband Router can encode all wireless transmissions with up to 128-bit encryption, and supports both Wired Equivalent Privacy (WEP) and the industrial-strength wireless security of Wi-Fi Protected Access? (WPA). The Router can serve as a DHCP Server, has a powerful SPI firewall to protect your PCs against intruders and most known Internet attacks, supports VPN pass-through, and can be configured to filter internal users’ access to the Internet. Configuration is a snap with the web browser-based configuration utility.

With the Linksys Wireless-G Broadband Router at the center of your home or office network, you can share a high-speed Internet connection, files, printers, and multi-player games, and turn that Internet connection into a high-quality, high-value telephone service!

The Linksys WRTP54G-NA is available at a retail price of $154.95 at

Harrison Study Misses Key Barrier to Consumer VoIP Adoption

April 12, 2006 by Garrett Smith

A Harrison Interactive study released today highlighted the growing awareness of Voice over IP technology within the consumer market place. The study, which featured sections on awareness, reasons to adopt, and barriers to adoption was the result of a survey conducted online between October 12th and the 16th, 2005 involving over 2,200 individuals. Although the study was a great numerical statement of a trend that many of us within the industry already see, the study fell short in one particular area; barriers to adoption.

The number one barrier to consumer VoIP adoption is a lack of broadband internet, not the fact that consumers prefer a landline or a mobile/cell phone. One of the primary reasons the survey did not uncover this fact is that it was it was done online, and I think it is safe to say the vast majority of those participating in paid online surveys are surely already on a broadband internet connection.

Due to the way VoIP service providers are marketing their services as the “low cost alternative” to traditional phone service, VoIP service providers are attracting droves of low-cost consumers, who more than likely have low-cost (read: dial-up) internet service. Because VoIP service requires broadband internet, most of these low-cost consumers are not going to upgrade their internet service and pay more for it, just to save money on their phone bill. Hence the reason why the lack of broadband internet is the largest barrier to consumer adoption of Voice over IP.


April 11, 2006 by Garrett Smith has recently added the IPEVO line of Skype Certified products to our robust product catalog. The IP EVO Free-1 is a stylish, advanced USB phone specially made for active Skype users.

With lightweight and stylish design, it is a lot more fun using Skype! Programmable buttons provides users with higher level of convenience. It allows users to have better control of Skype and make SkypeOut calls via the keypad. Ten different ring tones alert users of incoming calls. Other functions include volume control, mic mute, speed dial, and LED light indication. Free-1 allows you to contact people around the world for free. The IPEVO FREE-1 comes in both BLACK and WHITE.

The IPEVO FREE-1 retails for $39.95ea. or in a share pack for $69.95.

The IPEVO TOUCH-1 is the first product launched with Skype certification.


It is a corded USB phone which can be easily connected to a computer via USB interface. Skype’s API is built into this USB Phone which allows Skype users to control Skype with unique function buttons and numeric keypad. There are 4 different ring tones build into Touch-1 to alert users of incoming calls, as well as make SkypeOut calls by using the numerical keypad. Other functions include microphone mute, speed dial, and blue LED for status indication. This stylish USB Phone makes calling your Skype friends and colleagues even easier.

Linksys SPA-921 Single Line Sip Phone Now Available

VoIP Supply has just added the latest addition to the Linksys handset product line to it’s online catalog! The Linksys SPA-921, a single line, SIP compatible IP phone is the ideal solution for residential or soho applications utilizing an IP Centrex solution. Stylish and functional in design, the Linksys SPA-921 is also an ideal handset for the service provider, utilizing Linksys’ SPA-9000 PBX platform.

The Linksys SPA-921 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support. Standard features on the Linksys SPA-921 include a high resolution graphical display, speakerphone, and a 2.5 mm head-set port.

For more information, or to be one of the first to get their hands on the Linksys SPA-921, call 800.398.VOIP or CLICK HERE.


April 10, 2006 by Garrett Smith

This is the first of many exciting new partnerships that VoIPSupply will be announcing over the next few weeks. We are thrilled to have partnered with Vonage on this initiative and look forward to the continued success of our partnership and the kiosk.


For more information, contact:
Mitchell Slepian
[email protected]

Garrett Smith
VoIP Supply, LLC
[email protected]


Offering its Service in a Mall Kiosk gives Consumers more Options when Seeking a Feature-Rich, Flat-Rate Internet Telephony Plan

Holmdel, NJ, April 10, 2006 – Vonage Marketing, a subsidiary of Vonage Holdings Corp. a leading Internet telephony company, today announced that the company has launched a trial sales run of its devices at a kiosk in the Walden Galleria, Buffalo, N.Y. The kiosk, located on the lower level, adjacent to Dick’s Sporting Goods and Victoria’s Secret will be managed by VoIP Supply, LLC, a division of B2 Technologies, LLC, and a leading supplier of Voice over IP hardware, software, and services.

High-speed Internet customers can sign up for Vonage service by purchasing one of several Vonage-enabled starter kits sold at the kiosk, featuring devices from but not limited to Linksys, VTech and Uniden. Devices configured with Vonage’s phone service include telephone adaptors (TA), wired and wireless routers with TAs and broadband enabled cordless phone systems. Consumers who sign up with Vonage will be able to make unlimited calls throughout the U.S., Canada and Puerto Rico for $24.99 per month. Vonage offers its retail customers a $50 to $60 mail-in-rebate after 60 days in service. The rebate amount varies based upon which device is purchased.

“The launch of Vonage’s Internet telephony products in the VoIP Supply kiosk in the Walden Galleria is a great opportunity for us, as it will allow us to further develop our retail sales channel,” stated Philip Giordano, vice president of Sales and Business Development of Vonage Marketing. “We are looking forward to this sales offering in an untapped market. Vonage’s service sold in a mall kiosk now offers consumers another outlet to purchase services making Internet telephony more readily available to the general consumer market.”

“A key to greater consumer acceptance of Internet telephony is consumer education about the benefits of the technology,” stated Garrett Smith, Director of Business Development at “This kiosk allows consumers to speak with representatives who can answer questions and respond to any misconceptions about internet telephony. This is something no advertisement or mailer can effectively deliver on.”

About Vonage®
Vonage is a pioneer in the Internet telephony industry. Vonage’s award winning service is sold on the web and through national retailers. Vonage Holdings Corp. is headquartered in Holmdel, New Jersey. For more information about Vonage’s products and services, please visit or call 1-VONAGE-HELP. Vonage(R) is a trademark of Vonage Marketing, a subsidiary of Vonage Holdings Corp.

About is the leading supplier for Voice over IP hardware, software, and services. In addition to a comprehensive selection of IP phones, Analog Telephone Adaptors, Media Gateways, networking equipment and software platforms, the company offers provisioning and fulfillment, custom configuration, technical support, extended warranties and logistical services for end-to-end customer solutions. For additional information on, please contact Garrett Smith at 716.250.3408.

Vegastream Ships New Vega 50 6X4 Flexible Analog VoIP Gateways

April 5, 2006 by Garrett Smith

Vegastream Ships New Vega 50 6X4

VoIP Gateway vendor Vegastream has begun shipment of their new Vega 50 6X4 Series Analog VoIP Gateway.

The Vega 50 6×4 enables enterprises and service providers to connect sites with a legacy of mixed analog and digital applications to the IP network via a single, cost effective and manageable The Vega 50 6×4 connects a range of legacy telephony equipment, including PBX’s, ISDN telephones, ISDN, analog phones and the PSTN to IP networks.

The Vega 50 6×4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities.

The Vega50 6×4 is available factory configured in a number of different configurations, other configurations are available to special order.The following gateway configurations are available as standard:

  • 4 FXS + 2 FXO
  • 8 FXS + 2 FXO
  • 24 FXS + 2 FXO
  • 8 BRI Channels (4 interfaces)
  • 8 BRI Channels (4 interfaces) + 4 FXO

VegaStream’s VoIP gateways enable service providers and business customers to rapidly deploy and profit from lower telephony costs and improved productivity across their organisations’ HQs and remote offices.

The award winning Vega gateways are based on international communications standards, including SIP and H.323 to deliver an open and non-proprietary VoIP solution that can be seamlessly integrated alongside existing communications investments.

Established in 1998, VegaStream operates globally with sales and support centres in Europe (Bracknell, UK), America (California) and Asia-Pacific (Sydney, Australia). VegaStream investors include MTI Partners, Pace Micro Technology and VegaStream management.

VegaStream’s mission is to enable seamless interoperability between the wide and varied range of proprietary telephone systems and the new IP networks.
As one of the earliest suppliers of VoIP gateways supporting SIP, VegaStream’s portfolio of gateways are recognized as best-in-class SIP VoIP gateways for products such as Nortel Networks MCS5100, MCS5200 and Siemens OpenScape.

VegaStream’s partners with other VoIP technology leaders in areas such as voice recognition, and IP mobility to enable connectivity between their products and traditional PSTN networks. The company’s products are deployed by global service provider networks such as Primus Communications and Orbitel and demanding end-users such as the US Navy.

VegaStream products are sold through a network of systems integrators and service providers where they are becoming commonplace within mainstream enterprise networks to reduce the cost of everyday business VoIP communications.

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