VoIP Monitor - Hardware

PCN Introduces UltraEdge VoIP System for Any Grade Legacy Copper

VoIP Monitor — Mon, 26 Nov 2012 19:02:41 GMT

PCN announces the introduction and availability of its UltraEdge system for rapid cost effective deployment of VoIP and Unified Communication data networks.

With a single Ethernet connection to access the Internet or outside data networks, the UltraEdge system requires limited IT knowledge for installation and enables access and transport for IP device connectivity at every edge. Using any grade copper the UltraEdge VoIP system easily interfaces to existing junction boxes, legacy PBX consoles, or other wiring infrastructure already in place.

Providing maximum capabilities and feature rich applications, the system is a true IP networked solution leveraging PCN's high-performance 19" Rack Mount Multi-Channel Server (PCN3485-MCS4) for 10/100 IP transport to every UltraEdge module (PCN3485-SCC1) which enables access for VoIP enabled phones or other devices. Unlike media converters, each UltraEdge module is a true Ethernet switch where each end point has access for up to three Ethernet RJ45 ports.

Unlike DSL extenders and converter technologies, the PCN UltraEdge solution is built on PCN's patented "Dynamic Adaptive Channeling" data prioritization algorithms which run over 500 times per second in real time scanning, monitoring and managing the physical layer of legacy copper looking for anomalies and moving IP data to its most optimum spot on the wire. This not only provides 3X the noise immunity of DSL, but enables distance stretching. When combined together this not only allows more reliability and robustness of your VoIP system, but provides access to IP at distances up to 1500 feet and far greater than other solutions.

Across the world there is an abundance of legacy copper that is unusable for IP enabled Ethernet systems and where running new structured cabling is either too expensive, or requires significant shut down and disruption to business operations. This is true for government buildings, hospitals, historic buildings, factories, college campuses and office buildings that have Cat-3 or older wiring.

Without having to run any new structured cabling, the UltraEdge system allows owners and integrators to instantly leverage and take advantage of legacy wiring as if their facility was just wired with new Cat 5/6 cabling.

The UltraEdge VoIP system supports high-speed 10/100Base-T Ethernet-on-Demand across any grade copper wiring topology. This includes multi-drop, daisy chain, point to point and others allowing access in the far reaches of a building or installation.

Davis also commented, "Customers love the fact that they can easily install the multi-channel router at their old PBX locations or at other junction wiring; and then simply drop a plug-and-play UltraEdge module anywhere to have immediate access to high-bandwidth VoIP networks."

The UltraEdge VoIP system supports third party standard Layer 2 and Layer 3 Ethernet switching, QoS and a variety of network security features including port security, multi-level passwords, DoD protections, SSH and SSL for encryption. When combined with PCN's physical layer technologies the result is a comprehensive VoIP and Unified Communications system that is more reliable, more robust and one that goes further.

Grandstream Announces 1-Line IP Telephone with LCD Display

VoIP Monitor — Tue, 23 Oct 2012 21:22:58 GMT

Grandstream Networks introduces the GXP1160/1165. Along with a graphical LCD display that supports multiple languages, this low cost, single-line IP phone features high quality audio, integrated Power over Ethernet (GXP1165 model only), 2 call appearances, rich telephony functionalities, and Electronic Hook Switch capability with Plantronics headsets. The GXP1160/1165 is fully interoperable with leading IP Telephony platforms (such as Asterisk, Broadsoft, Metaswitch, 3CX, etc.) and various Internet Telephony Service Provider networks, and can be easily mass deployed and securely managed from a central location.

GXP1160/1165 Feature Highlights

128x40 pixel graphical LCD display with support for 11 languages
Single SIP account, up to 2 call appearances, 3 XML programmable context-sensitive soft keys, 3-way conference
Phonebook with up to 500 contacts and call history with up to 200 records
Automated personal information service (e.g., local weather), music ring tone/ring back tone
Dual switched auto-sensing 10/100Mbps network ports, integrated PoE (GXP1165 only)
Automated provisioning using TR-069 or AES encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
Electronic Hook Switch capability with a number of Plantronics headsets

Pricing and Availability

The GXP1160/GXP1165 will be generally available by the end of November 2012 through Grandstream's worldwide distribution channels at a list price of US$55 and US$59 respectively.

snom m9 Cordless DECT Phone is Now Interoperable with BroadSoft's BroadWorks

VoIP Monitor — Mon, 22 Oct 2012 21:09:57 GMT

snom announces the completion of interoperability between the snom m9 cordless DECT phone and BroadSoft's BroadWorks hosted Unified Communications platform. The snom m9 has completed interoperability testing with BroadWorks Release 18.SP1.

snom's m9 is now available to BroadSoft's more than 500 global telecommunications service provider customers in order to deliver enhanced, personalized communications and entertainment services to their businesses and residential subscribers. BroadSoft serves 18 of the top 25 largest telecommunications service providers, based on revenue, in more than 65 countries. With the completion of interoperability testing, BroadWorks powered end-users can benefit from the latest features and functionality when choosing a snom m9 DECT phone for their business.

"snom is committed to interoperability with a broad array of SIP-based platforms and is pleased to have completed the recent testing with BroadWorks," said Michael Knieling, chief operating officer of snom technology AG. "BroadSoft is at the forefront of the IP telephony industry and offers a mature, reliable platform, so we are extremely pleased to make our snom m9 cordless DECT phone widely available to BroadSoft customers around the globe."

The snom m9 combines snom's highly-interoperable IP communications firmware with an advanced DECT handset, giving business users an affordable, cordless VoIP phone. The m9 offers interference-free communication through a dedicated DECT frequency band, provides 100+ hours of stand-by battery time, features wideband audio, supports up to nine handsets and four simultaneous calls per base station and delivers superior speech quality, demanded by small business and residential users alike.

Standard features include hands-free mode and calling line identification, as well as typical mobile phone features, such as an on-board address book, calendar, calculator and alarm functions. Advanced features include IPv6 support, voice encryption, pre-installed security and color picture caller ID. The snom m9 is also fully functional in any open standard SIP environment.

Additional products in the snom portfolio will complete interoperability testing with BroadWorks in the coming months.

Edgewater Networks Certifies Panasonic for its Plug & Dial Alliance Program

VoIP Monitor — Thu, 18 Oct 2012 20:45:49 GMT

Edgewater Networks announces that Panasonic’s SIP communications-based phones have been certified as interoperable with Edgewater’s Plug & Dial solution. The Plug & Dial solution uses the EdgeMarc ESBC and the EdgeView VoIP Support System to automate the provisioning of a wide variety of IP phones. This automation reduces operating expenses and improves the end-user experience for service providers delivering cloud communications services.

Panasonic phones that have been Plug & Dial certified on EdgeView version 11.7.2 include KX-UT113B, KX-UT123B, KX-UT133B, and KX-UT136B.

The Plug & Dial Alliance program provides interoperability testing for multi-vendor VoIP networking environments and automated setup of many leading brands of SIP-based IP phones. Service providers use Edgewater Networks’ Plug & Dial solution to significantly shorten hosted PBX installation times and simplify ongoing moves, adds and changes. The solution uses intuitive voice prompts provided to the end-user so they can “self provision,” completely eliminating pre-staging or manual configuration of IP phones. The solution also provides notification to existing OSS or billing systems at the completion of the automated IP phone configuration. The level of automation provided by the Plug & Dial solution reduces installation times from hours to minutes.

The EdgeView VoIP Support System is used for the ongoing maintenance and management of IP phones. Qualifying phones report call quality scores to EdgeView where they are combined with results from other EdgeMarc monitoring points in a VoIP network. This greatly reduces problem-resolution times and enables service providers to deliver an improved customer experience. EdgeView is also used to remotely administer IP Phones and includes features such as the modification and backup of IP phone configuration files.

The EdgeMarc ESBC and EdgeView VoIP Support System are a part of a comprehensive solution from Edgewater Networks that connect, protect, optimize and manage IP-based communications.

Edgewater Networks and Panasonic will be exhibiting at BroadSoft Connections October 21 – 24, 2012, at the Westin Kierland Resort and Spa in Scottsdale, Ariz. Visit Edgewater Networks at booth #16 and Panasonic at booth #13 and 17.

netTALK Announces Free Advanced Privacy Features Added to its DUO VoIP Devices

VoIP Monitor — Thu, 12 Jul 2012 20:23:01 GMT

netTALK.com announces the launch of new privacy features: Caller ID Block and Call Blocking. These are being added to netTALK’s long list of free features, which is already overflowing with value and convenience for netTALK DUO digital phone users. All you need is a high speed internet connection and any standard home phone.

The new privacy features are now fully-integrated into all netTALK DUO digital phone service accounts, and using them could not be more simple or easy:

For caller ID blocking on your outgoing calls, simply dial *77, followed by the number and your call will come up as “UNKNOWN” on the recipient’s caller ID.
To block incoming calls from a specific number, log into your account and click on 'Phone Numbers' and enter a phone number you want to block including callers that use UNKNOWN or anonymous on their caller ID.

Patton Announces Microsoft Lync Certification for SmartNode VoIP Gateways

VoIP Monitor — Thu, 21 Jun 2012 20:37:34 GMT

Patton announced that SmartNode VoIP media gateways (release 6.T and newer) have been tested and certified in Microsoft's Unified Communications Open Interoperability Program for Lync Communications Server.

Microsoft lists Patton and SmartNode in the Direct SIP category, while Patton cites Microsoft as a Patton-Certified Enterprise Communication Partner for Lync and OCS.

The certification assures enterprise customers and system integrators that SmartNode VoIP equipment interoperates reliably and seamlessly with Microsoft's UC software, so they can select and deploy the products with absolute confidence.

SmartNode gateway-routers enable organizations to reap the benefits of Lync-based unified communications without replacing their existing phones or PBX systems, and without sacrificing connectivity with the traditional PSTN.

By interconnecting Lync software with legacy voice technologies and the PSTN, SmartNode preserves past investment in still-operational systems, while adding rich IP-telephony features including voice-and-data survivability and least-cost call routing—with the legendary set-it-and-forget-it reliability of SmartNode.

For more information about Lync solutions incorporating SmartNode VoIP gateways, register for Patton's educational webinar entitled "Gateway to Link" scheduled for Wednesday 27 June 10:00 AM EDT.

To facilitate trouble-free installation, Patton has published a free eleven-page application note entitled Microsoft Lync and Patton SmartNode. The publication provides set-up tips, valuable background information, and a sample SmartNode configuration file. Patton also provides free lifetime support with free software upgrades for its entire catalog of over 1,000 products.

Scaling from 2 to 2048 voice channels, the SmartNode product line is ideal for the small-to-medium, enterprise. SmartNode offers an exhaustive selection of telephony interfaces including PRI, BRI, FXS, FXO, T1, E1, and others.

Swiss-engineered and manufactured in the USA at Patton's world headquarters, SmartNode is the industry's first Microsoft Lync Certified VoIP gateway that offers models with integrated BRI interfaces.

Grandstream Adds VoIP Mobility with New Digital Cordless IP Phone

VoIP Monitor — Tue, 19 Jun 2012 21:20:23 GMT

Grandstream Networks introduced its new DP715/710 DECT Cordless IP Phone. The DP715/710, Grandstream’s first VoIP cordless phone, is SIP/DECT compliant, works with various major IP PBXs and supports wide range indoor/outdoor radio coverage based on the popular DECT standard, thus giving users the ability to enjoy the benefits of mobility and VoIP for minimum investment. Grandstream is showcasing the DP715/710 at Booth ID2-04 at CommunicAsia this week in Singapore.

The DP715, which includes the DECT handset and the VoIP base station with integrated charger, supports up to 5 distinct SIP accounts and is expandable to a total of 5 handsets and 4 concurrent calls. The DP710 or expansion unit, which includes a DECT phone and charger unit, is used in conjunction with the DP715 base station unit and is situated throughout the home or office for added user convenience.

“The new DP715/710 DECT Cordless IP Phones are an exciting extension of Grandstream’s market leading VoIP product portfolio,” said David Li, CEO of Grandstream. “These new phones provide residential and small-to-medium business users with a flexible, secure, and powerful wireless IP telephony solution that offers affordable mobility beyond the desktop and doesn’t compromise on voice quality or features.”

With an outdoor coverage range of nearly 1,000 feet (about 300 meters) and indoor range of nearly 150 feet (50 meters), the DP715/710 features a 102x80 pixel backlit graphic display, 10-hour talk time, 80-hour standby-time, full duplex speakerphone with acoustic echo cancellation, and rich features associated with VoIP including calling ID, call waiting, transfer, 3-way conference, do-not-disturb, hunt group, 18 polyphonic ringtones, 200 phone book entries (per handset), message waiting, multi-language voice prompt and many more. The DP715/DP710 DECT IP phones also support TLS and SRTP security protection which safeguards account information and encrypts voice conversations thus eliminating eavesdropping concerns typical of legacy analog cordless phones. In addition, the DP715/710 can be automatically and securely provisioned for easy deployment using TFTP, HTTP/HTTPS, and soon TR069 (pending).

Pricing and Availability

The DP715 (list price US$85) includes a DECT handset and VoIP base station with built-in charger. The expansion unit or DP710 (list price US$49) is shipped with a DECT handset and charging unit. The European version of the new DECT IP phone is generally available in Europe and some parts of Asia immediately through Grandstream’s EMEA and Asia Pac distribution channels. The North American version (DECT 6.0) of the product will be generally available by mid-July through Grandstream’s North American distribution channels.

Sangoma Commences Shipments of SBC

VoIP Monitor — Thu, 17 May 2012 20:43:36 GMT

Sangoma announces that commercial shipments of its highly-anticipated Vega 400 SBC standalone Session Border Controller appliance commenced today. The solution is designed to provide seamless connectivity between VoIP networks and SIP Trunking connections, while delivering security and policy enforcement to ensure call quality and reliability.

Bridging the gap between disparate IP networks or systems, the Vega 400 SBC provides encryption of media and signaling, as well as call access control, secure network logon and secure management interfaces. In addition to its security functions, the SBC provides a broad range of intermediation and core VoIP capabilities necessary to ensure proper call handling and call quality. This includes a wide array of audio codecs, and the capacity to perform real-time transcoding when necessary to facilitate call media handling. Key VoIP call quality measures and features enable the device to provide many VoIP gateway functions, such as adaptive jitter removal, comfort noise generation and silence suppression, QoS statistics reporting and hardware-based echo cancellation. In addition, Sangoma offers customers a lifetime guarantee on Vega 400 SBC appliances, further enhancing the value of this offering.

Sangoma Enhances IP Security Offering with SBC Release

VoIP Monitor — Tue, 08 May 2012 20:45:30 GMT

Sangoma announces the release of its Vega 400 SBC stand-alone Session Border Controller appliance,. The solution is designed to provide seamless connectivity between VoIP networks and SIP Trunking connections, while delivering security and policy enforcement to ensure call quality and reliability.

Located between disparate IP networks or systems, the SBC helps customers contain and control vulnerabilities that can be exploited in SIP-based networks to misappropriate or misuse VoIP resources and systems. The Vega 400 SBC provides encryption of media and signaling, as well as call access control, secure network logon and secure management interfaces. In addition to the security functions, the SBC provides a broad range of intermediation and core VoIP capabilities necessary to ensure proper call handling and call quality.

Supporting a wide range of audio codecs, the Vega 400 SBC performs transcoding when necessary to facilitate call media handling. Key VoIP call quality measures and features enable the device to provide many VoIP gateway functions, such as adaptive jitter removal, comfort noise generation and silence suppression, QoS statistics reporting, and hardware-based echo cancellation.

Sangoma Releases Transcoding Gateway Appliance

VoIP Monitor — Wed, 02 May 2012 21:12:07 GMT

Sangoma announces the release of its NetBorder Transcoding Gateway, the latest product in the growing range of gateway appliances to serve enterprises worldwide. This new gateway facilitates the interconnection of VoIP devices and networks with support for transcoding between a wide range of codecs-from low-bandwidth to HD-that are employed throughout the rapidly growing enterprise communications landscape.

The new NetBorder Transcoding Gateway provides bandwidth savings in interoffice VoIP and SIP trunking deployments with support for narrowband G.723.1 and G.729 codecs. Additionally, the appliance enables seamless connections between various SIP implementations such as those using different RTP packet sizes.

Beyond delivering bandwidth efficiency, the Transcoding Gateway offers many options for consistently high voice quality. In addition to the wide variety of narrowband codecs, support for HD voice codecs such as G.722 and G.722.1 makes it possible for HD and non-HD compliant devices to interoperate. On networks that are susceptible to packet loss, and the attending degradation of voice quality, more resilient codecs such as iLBC can be used to improve the performance and deliver solid call quality.

Available as a 1U appliance, the NetBorder Transcoding Gateway allows transcoding between any supported codec from a wide range including, iLBC, GSM-FR and as well as all common SIP and HD voice formats. These can be selected on a per call basis allowing for complete flexibility and the gateway can support up to 4000 simultaneous SIP sessions. The appliance provides very low latency processing to ensure low-jitter and excellent call quality.

snom Phones Now Interoperable with Schmooze Com's Unified Communications Systems

VoIP Monitor — Tue, 24 Apr 2012 21:49:27 GMT

snom announces interoperability with Schmooze Com Inc.'s PBXact phone system and PBXtended hosted PBX. snom VoIP phones can now be auto provisioned via the Schmooze Com Endpoint Manager for quick and easy installation and no-hassle management directly from the PBX Graphical User Interface. The following six snom models are currently certified for the Schmooze solution: snom 300, snom 320, snom 370, snom 821 and snom 870 desktop phones and the snom M9 wireless handset.

The combination of snom phones and PBXact and PBXtended systems gives small and medium-size businesses and call centers a complete, end-to-end voice solution. Value Added Resellers will appreciate the seamless auto provisioning and implementation using the Endpoint Manager GUI tool. Schmooze customers can take advantage of the full range of snom 300 and 800 series desktop phones to meet any staff requirement, as well as the snom m9 wireless handset that allows mobile workers to stay in touch while away from their desks. Whether a business chooses the PBXact premises-based PBX system or the hosted version, PBXtended, they will enjoy the same snom endpoint functionality.

The PBXact business telephone system and PBXtended hosted PBX provide all the standard features of a traditional PBX plus a unique set of features for the modern workforce, including visual voicemail, custom IVR auto attendants, find-me-follow-me and convenient voice recognition capabilities that allow employees to access the company directory, check messages and perform call control using simple voice commands. snom phones have been specially engineered with Schmooze's "magic button" that has been programmed for voice commands and dialing. Users simply speak commands such as 'call Mary,' 'transfer to Steve' or 'check messages' rather than using the keypad and entering an extension number or code. For VARs, training is a breeze with no star codes or cheat sheets needed to remember the features.

The range of snom phones suit any business requirement. The entry-level snom 300 is a good fit for general office workers, while the 320 adds more line keys for office workers with high call volumes and the snom 370, with its large graphical, high-definition display, is ideal for business executives. The advanced snom 821 and 870 (touch-screen) feature a large, high-resolution color display, an integrated XML browser, wideband audio for enhanced audio quality and the built-in Gigabit Ethernet switch for the latest network installations. The snom M9 cordless IP DECT handset targets the mobile workforce, providing coverage of 50 meters indoors or 300 meters outdoors.

snom to Donate 1 Dollar to Earth Day International for Every Phone Purchased Without Additional Power Supply

VoIP Monitor — Tue, 17 Apr 2012 21:27:25 GMT

snom announced that it will no longer be including external power supplies as part of the standard package for its 3xx and 7xx series IP phones, focusing instead on powering phones via Power-over-Ethernet. PoE is ideal for centralized and uninterrupted power supply and easy to implement. To encourage greater environmental awareness, snom will donate one dollar for every IP phone from its 3xx or 7xx series purchased in the nine weeks between April 22 and June 22, 2012. All donations will be made to the environmental initiative “Earth Day International” in Germany.

“Feedback from our customers has revealed that the vast majority use PoE to power their phones,” explains snom’s COO, Dr. Michael Knieling. “This means that many thousands of power supplies have probably been produced and never been used: In our eyes an unnecessary burden on the environment. With the donation to Earth Day International, we are providing our customers with a really easy way to help conserve the environment.” snom’s promotion is another addition to its ‘Green VoIP’ initiative and the low levels of energy consumption which snom IP phones represent. Those who rely on the external power supply unit and are unable to take advantage of the benefits of PoE will of course continue to be able to request them from their distributor or reseller.

Earth Day takes place annually on April 22 in more than 150 countries worldwide. Everyone can be involved, and every day is a day for the environment. Earth Day International’s initiators are not only concerned with the current well-being of humanity but also with future sustainability on a global level.

VoIP Supply Offers Best-in-Class Bundles for LifeSize Connections

VoIP Monitor — Thu, 12 Apr 2012 22:01:05 GMT

VoIP Supply announces the immediate availability of LifeSize Connections HD Bundles, LifeSize's cloud-based, HD video conferencing infrastructure as a service offering, complete with Logitech HD Pro C920 Webcams and compatible speakerphones and headsets.

VoIP Supply offers four bundles, each of which combine the cloud-based LifeSize Connections video software client—the fastest way for businesses to add to and extend their HD video conferencing capabilities—with matching hardware that's been tested for full compatibility, delivering the highest quality, seamless end-to-end video conferencing experience. VoIP Supply's four HD bundles include the HD standard - for people on-the-go; HD Popular - for the in-crowd; and HD Elite – for those who demand the very best. There is also a software-only option.

Cloud-based video conferencing reduces hardware costs, simplifies access for any sized business, and can be deployed in just a few minutes. LifeSize Connections' built-in firewall traversal ensures secure face-to-face collaboration that includes data sharing and multi-party HD video calls. The LifeSize Connections service gives each user the ability to call up to nine participants simultaneously, including up to two guests. Unlimited guest invitations are included with the service, as well as instant messaging and seamless integration among meeting rooms, PCs or Mac computers.

"The LifeSize Connections HD Bundles from VoIP Supply provide video conferencing from zero to video in under five minutes with an uncomplicated setup," said Benjamin P. Sayers, President and CEO of VoIP Supply. "With businesses rapidly migrating to video conferencing, we're focusing our bundles on affordability and greater customer support; which is why when you call for help you actually get to speak to someone who understands these technologies."

VoIP Supply's LifeSize HD Connections bundles:

The LifeSize Connections Elite Solution from VoIP Supply includes the LifeSize Connections Client, and features the stunning 720p HD clarity of the Logitech HD Pro Webcam C920, and the Jabra Speak410 USB full-duplex speakerphone, and the Jabra Biz 2400 3-in-1 professional headset.
The LifeSize Connections Popular Solution from VoIP Supply includes the Logitech HD Pro Webcam C920, the Jabra Speak410 USB full-duplex speakerphone, and the Jabra WAVE corded headset.
The LifeSize Connections Standard Solution from VoIP Supply includes the LifeSize Connections Client, the Logitech HD Pro Webcam C920, and the Jabra UC Voice 250.
The LifeSize Connections HD Software-Only Solution from VoIP Supply.

New Digium IP Phones Now Available

VoIP Monitor — Mon, 09 Apr 2012 20:47:34 GMT

Digium announces the general availability of its new family of IP phones. The Digium phones are the first that are specifically engineered to fully leverage the power of Asterisk, the world's most widely adopted open source communications software, and Switchvox, Digium's award-winning Unified Communications system. Business customers will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system.

The phones come with a number of standard applications, including interactive voicemail, visual call parking, one-touch call recording, searchable contact directory and presence management. The Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create applications that interface directly with core phone features.

Early reviews confirm that Digium phones are the easiest to install and manage with Asterisk or Switchvox:

"Even someone who has never provisioned an IP phone can literally have their Digium phones working within five minutes. We've never experienced this before with any other phone hardware," said Corey McFadden, managing partner, Infradapt. "With the upcoming API and application engine, the Digium phones offer nearly unlimited possibilities for implementing features for demanding clients. We have immediate plans to roll out features like a phone-based time clock for recording employee attendance, 'hot-desking' for desk sharers, call accounting for professionals and many others."

"These new products change the phone market by bringing a set of features unparalleled in the IP-PBX world. The API and app engine will allow us to offer a truly tailored end-to-end solution for our customers," said Robert Shubow, president of Ultracom-Intelesys.

"IP telephones specifically designed for Asterisk and Switchvox are exactly what we've been looking for," said Michael LeBlanc, CEO of LeBlanc Communications Group. "Our customers will benefit from the tight integration and will be able to do even more with their desktop phones, which means we can deliver greater value to our customers."

"I am really excited about the new Digium phones," said Ken Stauffer, CEO of Stauffer Technologies. "Our initial set-up time has decreased significantly thanks to the no-touch configuration."

Both Asterisk and Switchvox have been updated to offer enhanced functionality and direct integration when used with the new Digium phones. The phone-ready release of Asterisk is based on the 1.8 long-term support release of the Asterisk open source platform. A free add-on module called the Digium Phone Module for Asterisk (DPMA) acts as a bridge between the Asterisk server and the phones. The DPMA handles provisioning, firmware updates and application integration.

Switchvox version 5.5 is available as a free upgrade for all Switchvox customers with active subscriptions. This new version provides integration with Digium phones, making many of the popular Switchvox features available on the desktop phone. In addition to the new capabilities developed for Digium phones, Switchvox version 5.5 also includes customizable directories, and call rules that respond to status.

The Digium phones include the following models:

D70 - An executive-level HD IP phone with 6-line keys and 10 rapid dial/busy lamp field keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.
D50 - A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field keys with an easy-to-print paper label strip for the user's most important contacts. This model is ideal for users who spend a lot of time on the phone.
D40 - An entry-level HD IP phone with 2-line keys. This is Digium's best value phone, designed for any employee in the company.

The MSRP for these models is as follows: D70 - $279 USD, D50 - $179 USD and the D40 - $129 USD.

VoIP Supply Adds Revolabs FLX Wireless VoIP Conferencing Systems

VoIP Monitor — Thu, 29 Mar 2012 22:48:58 GMT

VoIP Supply is pleased to announce the addition of innovative Revolabs FLX wireless VoIP conferencing systems. Revolabs is the premier audio solutions provider of wireless unified communications products for free-flowing workspaces uninhibited by cords.

The Revolabs FLX VoIP conference system is the only wireless IP conference phone available today. Used for both audio and video communications, the Revolabs FLX now benefits from Revolabs' revolutionary audio technology that was once reserved for the largest of conference rooms and board rooms. Now, with the FLX, small to medium-sized conference rooms, executive offices, or home offices can leverage the best audio quality with the freedom of wireless mobility.

Not your traditional conference phone, the Revolabs FLX system is a dual-purpose solution eliminating the need for a separate desktop and conference phone. The flexibility of the included wireless FLX dialer is not only used to set up a conference call but it acts just like a regular telephone handset for private calls before, during, or after a conference call.

The Revolabs FLX may be sleek and compact but unlike the small conference phones you're used to, your group no longer has to huddle around the device or shout across the room to be heard on the other end. Instead, FLX users can enjoy a more natural conference room setting because with the FLX's separate wireless components you have more freedom to place the speaker, dialer, and microphones where they make the most sense. The FLX comes standard with two interchangeable microphones in your choice of directional, omnidirectional, or a wearable model so that a presenter can move freely about the room.

The Revolabs FLX is also compatible with leading video conferencing system brands, is Bluetooth enabled for connection to your cell phone or computer, has an encrypted voice signal for secure calls, and the FLX system will not steal Wi-Fi bandwidth or interfere with other radio signals.

"Revolabs is bringing a whole new dimension to VoIP conferencing," said Garrett Smith, Chief Marketing Officer at VoIP Supply. "Adding the only wireless VoIP conference phone on the market with the Revolabs FLX allows VoIP Supply to provide so much more flexibility for audio and video project collaboration, webinars, Skype, and distance learning."

Digium Simplifies Communications With Advanced Asterisk-based VoIP Gateways

VoIP Monitor — Mon, 26 Mar 2012 21:13:40 GMT

Digium introduces the G100 and G200, the first in a family of cost-effective VoIP gateways that simplify the process of deploying converged media networks. Built on a powerful combination of the Asterisk open source communications engine and a state-of-the-art embedded platform, the new gateways provide the best value for Asterisk communications solutions.

Digium’s gateways are built to support both TDM-to-SIP and SIP-to-TDM applications. In a TDM-to-SIP deployment, the gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.

The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. The gateways feature a power-saving embedded design with a highly efficient digital signal processor handling all media-related operations. The combination of an intuitive user interface, the flexibility of Asterisk and the purpose-built media processing capabilities of the DSP results in a gateway platform that outperforms the dated designs in the market today.

Digium beta testers agree. “Setting up the G200 was extremely easy compared to doing it with other gateways. I'm spoiled now!” said Tim Banks of Project Resource Solutions, an Illinois-based Digium Select partner. “Digium has really set the bar high. Their new gateways make it incredibly easy to connect older TDM phone systems with SIP services.”

Digium’s new gateways represent a solution to one of the challenges associated with running Asterisk applications in virtualized environments. TDM interface cards require a card slot – something distinctly missing from virtual servers. By converting the media and signaling from TDM to SIP on a dedicated external device, Asterisk users can migrate applications to virtualized, hosted or cloud environments.

The G100 includes a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. The G200 doubles the capacity with two T1/E1/PRI interfaces and up to 60 concurrent calls. Both models have integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.

The single-span G100 lists for $1,195 USD while the dual-span G200 model lists for $1,995 USD. The gateways are currently available worldwide through Digium’s network of distribution and integration partners.

Patton Fills VoIP-Market Gap with SmartNode BRI/FXS/FXO Gateway-Router

VoIP Monitor — Wed, 07 Mar 2012 22:16:10 GMT

Patton announces it is now taking orders for the 8-to-24-call SmartNode 4660 BRI/FXS/FXO VoIP Gateway Router for mid-sized enterprises and service providers.

With today's announcement Patton addresses the demand for VoIP-enabling solutions in European markets where incumbent PSTNs offer 8-BRI services alongside FXS/FXO lines often supplied by CLECs. Initial shipments are expected during May 2012.

Patton's SN4660 fills the void between high-cost modular VoIP equipment (overkill for most mid-size businesses) and kludgey multi-box solutions (cobbled together from lower-quality, smaller-port-count devices).

Offering flexible port combinations of 2-to-8 ISDN BRIs, a standard 4-port IP-Ethernet switch, and optional 4-to-8 analog FXS/FXO interfaces, the SmartNode 4660 IP-enables ISDN-PBX systems and POTS phone, fax, answering machine, and audio-intercom systems with a cost-effective, one-box solution for 8-to-24 voice or fax calls over IP.

Four 10/100 Ethernet ports provide seamless WAN access via any existing cable, ADSL, VDSL, EFM, or fiber-optic modem, as well as enterprise LAN connectivity.

As a mid-sized enterprise itself, Patton majors on flexible manufacturing combined with responsive product engineering. Since the company houses sales, marketing, engineering and manufacturing—all under one roof—in its US-based bricks-and-mortar headquarters, Patton can respond to market feedback and deliver orders faster than most other VoIP or Telecom equipment suppliers.

During March and April, Patton offers resellers and channel partners promotional discounts ranging from 5 to 20 percent on SN4660 pre-orders, depending on order quantities and timing (contact sales@patton.com for details).

Next-up from Patton and SmartNode... watch for an 8-BRI FXS/FXO VoIP IAD with an on-board G.SHDSL internet-access modem.

Grandstream Introduces New GXP Enterprise HD Telephone

VoIP Monitor — Mon, 05 Mar 2012 21:08:16 GMT

Grandstream Networks introduces the GXP2124 Enterprise HD IP Telephone for enterprise customers looking for a high performance HD telephone with numerous programmable keys and Electronic Hook Switch support for high call volume applications such as call centers, customer support and reception areas.

The GXP2124 is Grandstream’s first HD IP telephone with EHS support for Plantronics headsets. Users can answer and end calls using only the button on the headset – eliminating the need to touch the desktop phone. In addition, the GXP2124 features 4 line keys with up to 4 SIP accounts, 24+4 programmable speed-dial/BLF keys, broad interoperability with major SIP platforms such as Broadsoft/Asterisk/etc, superior HD audio, and 5-way conference. Grandstream is showcasing the GXP2124 and the entire family of award-winning GXP Enterprise IP Telephones at Stand B76, Hall 13 at CeBIT being held this week in Hanover, Germany.

Key Features of Grandstream’s GXP2124 Desktop HD Telephone:

4 lines with up to 4 SIP accounts, 24+4 speed-dial/BLF extension keys with dual-color LED, 4+4 context sensitive XML programmable keys, up to 32 call appearances, support for Electronic Hook Switch with Plantronics headsets
240x120 backlit graphical LCD with up to 8 level grayscale, dual 10M/100Mbps Ethernet ports with integrated PoE
Support for multiple native languages (including English, German, French, Spanish, Italian, Portuguese, Chinese, Korean, Japanese, Russian, and Greek)
HD wideband audio, full-duplex high performance hands-free speakerphone with advanced acoustic echo cancellation, 5-way conference by leveraging the superb audio performance of DSP Group’s XciteR chipset.
Integrated real-time web applications (weather, stock, currency, etc.), large phonebook (up to 2,000 contacts) and call history (up to 500 records)
Advanced security protection (TLS/SRTP/HTTPS/802.1x) and auto provisioning (TR-069, HTTPS, and AES encrypted XML configuration file)

Pricing and Availability

The GXP2124 will be generally available for purchase by the end of March 2012 through Grandstream’s distribution channels at a list price of US$169 (North America). For other regions, please contact your local Grandstream distributors/resellers.

VoIP Supply Chosen as First Distributor for AudioCodes HTTPS Fax Adapter

VoIP Monitor — Tue, 14 Feb 2012 23:45:37 GMT

VoIP Supply announced at IT Expo 2012 a new generation of advanced fax telephone adapters, the AudioCodes Fax ATA. This new HTTPS enabled adapter is a cost-effective, advanced fax product, which allows the connection of ordinary fax machines and Multi Function Printers to cloud based fax service providers such as eFax and to premise-based fax servers. The Fax ATA still retains the ability to connect voice calls, provide fully real-time audio feedback on fax call connections along with E911 support. The new Fax ATA product, which is making its debut through VoIP Supply, is available immediately.

Combining superior fax reliability, security and cutting-edge features for end users and service providers alike, the HTTPS Fax enabled MP-202B, preserves the easy and familiar experience of the fax machine, so that users can get rid of dedicated phone lines to their fax machine. No matter what type of data connection is used, including open Internet, Satellite and Cellular, this solution compliments VoIP deployments of all kinds allowing users to keep their fax machine. Traditionally those who used eFax® services could not use their existing fax machine to send faxes; they needed to scan and send paper-based documents. With this new Fax ATA device, users of these services can send directly in the traditional manner through their existing eFax® account.

"VoIP Supply continues to source and offer the new and innovative products that meet the needs of businesses today. The new AudioCodes Fax ATA demonstrates that commitment," said Benjamin P. Sayers, President and CEO of VoIP Supply. "We see more and more businesses migrating to Internet based communications. The AudioCodes Fax ATA removes a major hurdle by enabling them to send faxes using services like eFax® with their legacy fax machines."

Dialog Semiconductor VoIP Technology Adopted by VTech for S-Series Desktop Phones

VoIP Monitor — Tue, 07 Feb 2012 23:36:03 GMT

Dialog Semiconductor announces that its ultra low power Green VoIP chip family has been adopted by VTech. Under the terms of the partnership VTech is using Dialog’s SC14452 and SC14461 VoIP processors and Rhea software suite to produce a series of VoIP cordless and corded phones. The first of which, the S-series of single and dual-line phone systems, has been designed for the hotel market and went into mass production in the fourth quarter of 2011. The design win successfully extends Dialog’s relationship with VTech, which is already using the company’s DECT IC technology in its digital cordless phones.

VTech’s S-series VoIP phones incorporate multi-call and multi-handset capabilities, speakerphone functionality, IEEE802.3af power over Ethernet, and a USB charging facility, together with an array of other advanced hotel phone features. Furthermore, the VTech SIP cordless desktop phones incorporate DECT, CAT-iq and DECT 6.0 with an adjustable coverage area to deliver improved clarity and installation management.

The Dialog SC14452 is an award-winning VoIP IC with built-in DECT support and has the flexibility to meet the diverse needs of users and manufacturers. It features a 16-bit CompactRISC processor and two user-programmable Gen2DSPs to deliver 240-MIPS of computational power. The device uses a low clock frequency to achieve the industry’s lowest power consumption figures, saving as much as 2W per desktop phone compared with other solutions. It integrates a 32KHz wideband audio codec, Class D amplifier and PAEC (Perfect Acoustic Echo Canceller) to deliver exceptional audio clarity. Its sister device, the SC14461, has all the capabilities of the SC14452 but without DECT support and is aimed at corded desktop models.

Digium Introduces World’s First Phones Designed for Asterisk

VoIP Monitor — Wed, 01 Feb 2012 23:01:33 GMT

Digium introduces a new family of high-definition IP phones. They are the first that are engineered to fully leverage the power of Asterisk, the world’s most widely adopted open source communications software, and Switchvox, Digium’s award-winning unified communications system. With Digium technology on both the server and the phone, users will benefit from the best possible performance, unprecedented integration and a uniquely customizable phone system.

Asterisk has always been about flexibility, allowing integrators and developers to create highly customized solutions. Likewise, Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren’t simply XML pages; Digium phone apps can interface directly with core phone features.

Digium has leveraged this unique programming interface of the phones to create a suite of productivity applications that work with both Asterisk and Switchvox. Switchvox includes a unique web interface called Switchboard that gives each system user control of their personal communications environment. Digium has extended the capabilities of the Switchboard to the phone, putting advanced features like presence management, searchable contact directory, queue monitoring, recording and voicemail control, all at the user’s fingertips.

The Digium phones include the following models:

D40—An entry-level HD IP phone with 2-line keys. This is Digium’s best value phone, designed for any employee in the company.
D50—A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field keys with an easy to print paper label strip for the user’s most important contacts. This model is perfect for users who spend a lot of time on the phone.
D70—An executive-level HD IP phone with 6-line keys and 10 rapid dial/BLF keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.

Digium plans to have general availability of these new phones in April 2012. The MSRP for these models is as follows: D70 - $279, D50 - $179 and the D40 - $129.

RTX Launches the DUALphone 4088 Cordless Skype Phone

VoIP Monitor — Wed, 01 Feb 2012 22:52:39 GMT

RTX is launching a combined SkypeT and landline phone. The new cordless phone, the DUALphone 4088, is being launched at ITEXPO in Miami.

The main benefit of the DUALphone 4088 is that it does not require the user to be physically connected to a PC to make VoIP calls. The caller can simply connect the DUALphone base station to a broadband connection and make Skype calls free of charge to other Skype users or have the choice of conventional paid-for landline calls worldwide.

At a time when the number of Skype calls is soaring, RTX says it expects the new handset to bring free and low cost calls to consumers worldwide. According to a recent report by telecom market research firm TeleGeography, cross-border Skype-to-Skype calls grew 48 percent in 2011, to 145 billion minutes.

Jesper Mailind CEO RTX commented, "Our new elegant DUALphone 4088 offers consumers a unique way to use Skype, bringing the benefits of wireless low cost or free calls to households across the globe at an affordable price-point. We believe there is a growing demand for dual phones from the existing 170 million Skype account holders. As more and more people use Skype our user-friendly and intuitive phone will be an attractive solution. It will also provide a business opportunity for resellers around the world."

The new handset has a colour TFT screen and superior sound quality through the use of high definition Audio. The phone system can switch between two separate Skype accounts and can be extended to four handsets per household. Standard features including call waiting, voicemail and a phone contact list of 200 numbers have been enhanced by software to show the Skype status of the user and all contacts. The design of DUALphone 4088 is suitable for the modern living room.

Revolabs Unveils FLX VoIP

VoIP Monitor — Tue, 31 Jan 2012 22:02:34 GMT

Revolabs introduces the Revolabs FLX VoIP, the first wireless conference phone designed for VoIP networks. Supporting a wide variety of IP switches, the FLX VoIP is the only conference phone that supports the audio clarity of HD audio while providing the freedom of wireless microphones and speakers. The feature set that has been available through the Revolabs FLX for analog phone lines is now also available for IP telephone networks, providing unprecedented conference call clarity and flexibility.

The FLX VoIP integrates directly with most IP telephone switches following the SIP standard. Through this integration, new features only available through digital switch environments, such as voice mail alerts and "do not disturb," can now be offered with the FLX VoIP. The phone's wireless capabilities allow it to be used in small and midsize conference rooms without running any cables. As with the FLX for analog phone lines, this allows for a clean look while requiring less space on the conference table. The independent microphones, speaker, and dialer of the FLX VoIP give the user freedom and flexibility that other conference phone systems cannot offer.

Combining wireless operation, high-quality wideband audio, 128-bit encryption, and integrated Bluetooth®, the FLX VoIP redefines the conference speakerphone. Unlike the single-component design of previous solutions, Revolabs FLX VoIP evolves the conference phone into several distinct components, giving users unprecedented freedom with respect to placement and accessibility of the speaker, microphones, and dial pad.

Available with a variety of compatible Revolabs microphones, the FLX VoIP supports a lapel microphone worn by one person; an omnidirectional tabletop microphone that captures the sound of six to 10 participants; and a directional tabletop microphone that enables audio capture from two to three people. Because the FLX VoIP dialer operates like a telephone for handset calls and enables the set up of conference calls, there is no need for a separate desk and conference phone.

The Revolabs FLX VoIP can also serve as the audio interface for virtually any major brand of video conferencing equipment, making it the ideal unified communication technology for small to medium-sized conference rooms, executive offices, and small office/home office environments. FLX VoIP's integrated Bluetooth technology provides a single collaboration device no matter which communication channel is used, allowing users to connect speakers and microphones to their Bluetooth-enabled mobile phones or computers.

The Revolabs FLX VoIP will be available worldwide in February 2012, and sold through major distributors, dealers, and resellers.

Ooma Unveils HD2 VoIP Handset at CES

VoIP Monitor — Wed, 11 Jan 2012 23:35:37 GMT

Ooma announced at the 2012 International CES in Las Vegas, Nevada, a new cordless handset with superior HD Voice call clarity and smartphone features made possible by the Ooma cloud-enabled platform. The new cordless Ooma HD2 Handset features a two-inch color screen and picture caller-ID with the ability to automatically display Facebook profile pictures and online contact lists from Facebook, Google and Yahoo. Picture caller-ID and contact lists provide the ability to see a picture of the caller as the phone rings, download and scroll through contacts, and easily manage contacts using the My Ooma web portal.

The combination of the Ooma Telo and new Ooma HD2 Handset provides cutting-edge HD Voice call clarity by capturing twice the voice data to double the fidelity of standard phone calls, for richer, more natural-sounding conversations. The cordless handset offers superior security and range afforded by the latest DECT technology without interfering with home Wi-Fi networks or other home electronics. Up to four handsets can be used with each Ooma Telo.

One-touch voicemail access lets users check messages anywhere in the home, and there’s an intercom to talk between handsets or transfer calls. The handset’s speakerphone and headset port keep hands free for multitasking during calls. Another useful feature is the ability to configure a handset so it can be used as a baby monitor.

Ooma HD2 handset users who subscribe to Ooma Premier service (optional, $9.99/month) can enjoy even more advanced features such as an Instant Second Line, Multi-Ring to simultaneously ring another phone, three-way conferencing, a second personal number anywhere in the U.S., call screening via the built-in speakerphone, and the ability to send calls directly to voicemail with the touch of a button.

The Ooma HD2 Handset will be available in March 2012 at http://www.ooma.com and select retailers with an M.S.R.P of $59.99.

Telchemy Announces Powerful New Active Test Tool for VoIP, Videoconferencing and Network Test

VoIP Monitor — Fri, 18 Nov 2011 17:45:52 GMT

Telchemy announces DVQattest Version 2.0, a powerful new active test system for unified communications. DVQattest generates Voice over IP and Videoconferencing calls with SIP signaling, synthetic HTTP, POP3, SMTP, DNS and DHCP transactions and a range of IP network tests. This advanced test product supports pre-deployment testing, SLA monitoring and troubleshooting for converged networks and services.

DVQattest Agents are compact but highly featured software applications that can be installed on a range of operating systems and hardware platforms, including Linux servers and appliances, Android mobile phones and directly integrated into network equipment and CPE. Tests can be run on-demand or scheduled to run indefinitely. Agents can run multiple concurrent tests to other Agents or to IP phones, Web sites, Email sites and other network-based services. DVQattest Agents support complex networks with overlapping IP address spaces, VLANs and a range of SIP infrastructure configurations.

VoIP and Videoconferencing tests verify the performance of both SIP signaling and the media stream. Voice and Video quality is measured using Telchemy’s market-leading VQmon technology, providing MOS scores and a wide range of diagnostic data. Voice over IP tests support configurable codec, packet size and jitter buffer configuration; Videoconferencing tests support configuration of codec, image size, GoP, frame rate, bit rate and smoothing.

Network tests include Agent-to-Agent tests that measure loss, jitter and available bandwidth in each direction, industry standard network tests and advanced route diagnostics. DHCP and DNS tests verify correct operation of vital network functions and HTTP/POP3/SMTP tests measure performance of key applications.

The DVQattest Controller provides an easy-to-use management application that supports test definition, remote DVQattest Agent management and test result collection and reporting. For larger networks, the SQmediator performance management application provides a scalable and intuitive solution for multiple concurrent users. DVQattest Controller and SQmediator support key security requirements and maintain AES encrypted connections to DVQattest Agents.

DVQattest provides dependable, accurate and detailed performance metrics and has already been deployed in critical network applications. When the US Department of Defense needed accurate tools for measuring performance for Internet Routing in Space (IRIS) satellite based router project, DVQattest provided a key element of their measurement infrastructure.

DVQattest is available in a wide range of configurations, suitable for mid-large enterprise, hosted and cloud based services and tier one service providers.

Digium Releases Octal-Span Digital Card; Connects Traditional Telephony Services with Asterisk Communications Systems

VoIP Monitor — Tue, 15 Nov 2011 22:28:44 GMT

Digium announces the availability of the TE820 Octal-Span digital card. This new high-density solution compliments Digium’s existing broad suite of telephony card offerings designed specifically for Asterisk-based communications systems. The TE820 enables Asterisk integrators and OEMs to build large scale telephony deployments that are both high performance and cost-effective.

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. The combination of Digium hardware and Asterisk software provides a cost-effective platform for building numerous communications solutions, from PBX systems and VoIP gateways to IVR servers, call centers and complete unified communications suites. The TE820 supports up to 192 channels (in T1/J1 mode) or 240 channels (in E1 mode) and is available with or without hardware echo cancellation.

The TE820 card supports industry standard telephony protocols, including multiple variants of Primary Rate ISDN. Each span can be configured as either CPE or network for optimal flexibility. The optional VPMOCT256 hardware echo cancellation module, based on the industry-leading Octasic chipset, offloads the task of echo cancellation from the CPU, increasing overall system performance and call quality.

The Octal-Span digital card will be available on November 18, 2011 from Digium and Digium partners.

snom Unveils New Class of SIP Phones Designed for SMBs with Big Business Tastes

VoIP Monitor — Mon, 07 Nov 2011 22:03:52 GMT

snom introduces a new line of business VoIP phones – the snom 7xx series designed for both small and mid-sized businesses requiring an enterprise-class desktop phone on an SMB budget. The snom 720 and snom 760 business phones bring together the multiple programmable buttons and popular standard business functionality of the snom 3xx series with the advanced functionality, sleek styling and Gigabit Ethernet switch found in the snom 8xx series to create an advanced desktop phone at a value-driven price.

Advanced Features and Elegant Design for Next Generation Business

Both the snom 720 and 760 offer a Gigabit Ethernet switch, automatic provisioning, wireless LAN connectivity and snom’s superior wideband high definition voice quality. In addition, thanks to a Gigabit Ethernet switch, both phones can transfer data at a speed of 1000Mbits/s without slowing down the network or a connected PC. Both phones also feature Bluetooth connectivity via optional USB stick, allowing users the freedom to use a compatible Bluetooth headset with their snom 7xx series phone. The snom 760 features a high-resolution color display and two USB ports for a variety of connectivity options, as well as a newly designed handset grip that increases user friendliness by providing silent pickup and return of the handset. The snom 760 also includes a 16-key programmable busy lamp field and 4 context-sensitive keys complemented by the large, easy to read display.

The snom 760 also offers the standard desktop feature set of any snom phone, and is ideal for business environments that require a greater level of visual functions, such as the use and delivery of XML-based data. The large display also supports caller images, uploaded by the caller or included in the user’s address book.

Traditional Phone Features for Everyday Business

The snom 720 builds off the elegant and functional simplicity of the snom 3xx series business phone, featuring an easy to read, four-line monochrome graphical display. The snom 720 offers 18 fully configurable function keys and four variable keys, ideal for managing and contacting large groups of people.

The snom 720 also supports all standard VoIP calling features, including an address book with 1,000 possible entries, speed dialing, URL dialing, ringtone selection and LED call indication. In addition, the snom 720 and 760 also feature wireless LAN (WLAN) connectivity via optional USB stick.

Both the snom 720 and snom 760 are available for order today by distributors and resellers worldwide. snom 720 MSRP is $219.00 US and snom 760 MSRP is $329.00 US.

Panasonic Announces Interoperability for Full Lineup of SIP Phones With CoreDial

VoIP Monitor — Thu, 03 Nov 2011 21:16:31 GMT

Panasonic announces that CoreDial has certified Panasonic's line of SIP Phones, the KX-TGP500, KX-TGP550, KX-UT113-B, KX-UT123-B, KX-UT133-B and KX-UT136-B, for use with their Hosted PBX telephony platform. This alliance leverages Panasonic's global leadership in the DECT cordless telephone market and CoreDial's leading private label VoIP cloud software platform, adding up to a winning combination.

Ideal for both home office and business environments, Panasonic's SIP phone systems offer the flexibility of cordless or corded models while supporting a wide range of business class features provided by the CoreDial platform. The KX-TGP500/550 systems offer convenient, cordless designs that eliminate the need to run dedicated network wiring to each employee work station while incorporating DECT 6.0 to ensure no interference with wireless networks. The new KX-UT Series is designed to complement a company's existing communication infrastructure and offer end-user savings with features including two data ports, PoE and lower power consumption while in ECO mode. All Panasonic SIP models are HD Voice enabled, allowing for outstanding voice quality, and offer flexible system expandability.

With flexible configuration options, it has never been easier to deploy and expand a Panasonic SIP-based phone system with CoreDial's Hosted PBX platform. The reduced hardware costs and simplicity of routing calls over an Internet connection can add up to huge savings on monthly telephone bills, thus enabling all business environments to take advantage of a larger variety of business-class features such as call forwarding, intercom and conferencing, voicemail and more.

TGP500 Series Details:

KX-TGP500: The system features a wall-mountable base unit and one cordless handset. It is expandable up to 6 DECT 6.0 cordless handsets and supports up to 8 phone numbers and 3 simultaneous calls. It boasts Wide Band Audio (G.722) and 5 hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset Speakerphone, 2.5mm headset jack and belt clip.

KX-TGP550: The KX-TGP550 has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD and 5 hours Talk Time, 10 days Standby, plus a Hands-Free Speakerphone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

KX-TPA50: The TGP500 systems can be expanded up to a total of 6 cordless handsets by adding the KX-TPA50 cordless handset.

KX-UT136 and KX-UT133: These models are user-friendly and easy to operate, with 24 programmable feature/functionality keys. The KX-UT136-B features a six-line backlit graphical LCD and 2 Ethernet ports, while the KX-UT133-B offers a three-line backlit graphical LCD and 2 Ethernet ports.

KX-UT123 and KX-UT113: These standard models are breaking barriers by offering HD Voice, PoE and two-year warranty. The UT123 features a three-line backlit graphical LCD and two Ethernet ports, while the UT113-B has a three-line graphical LCD and one Ethernet port. They both offer ease-of-use at a competitive price for excellent return on investment.

Panasonic Showcases Award Winning SIP Telephony Solutions at AstriCon 2011

VoIP Monitor — Tue, 25 Oct 2011 21:25:13 GMT

Panasonic is showcasing an extensive range of IP telephony solutions at AstriCon in Panasonic booth #204 at the Westin Westminster in Denver, Colorado, October 25-27.

As an AstriCon 2011 Gold sponsor, Panasonic is demonstrating a broad range of SIP telephones including SIP Cordless Phone Systems and KX-UT series which are interoperable with Asterisk's open source PBX. Panasonic's SIP telephones work with the Asterisk platform which offers both classical PBX functionality and advanced UC features.

In its seventh year, AstriCon is the longest running conference devoted to the Digium Asterisk communications platform. AstriCon brings together open source enthusiasts, from coders and system integrators to service providers and enterprise IT professionals, who are looking for an in-depth understanding of Asterisk open source technology.

Panasonic's SIP Phone Systems:

The Panasonic SIP Cordless Phone System is a small business communication solution that offers the flexibility of convenient expansion as a company grows. The KX-TGP500 system features a wall-mountable base unit and one cordless handset. Expandable up to six DECT 6.0 cordless handsets, the model supports up to eight phone numbers and three simultaneous calls. It boasts Wide Band Audio (G.722) and five hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.

The KX-TGP550 model has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

Also on display, the Panasonic KX-UT series offers a cost-effective communications solution for businesses of all sizes that leverages the latest developments in Hosted and Open Source PBX technologies and is designed to complement a company's existing communication infrastructure. Most models feature two data ports so users can connect a second network device without the time and expense of running an additional Ethernet cable. The KX-UT series models are Power over Ethernet ready which eliminates the need for additional electrical adaptors. Wide-band, high-definition audio (G.722 codec) coupled with echo cancellation and an expanded acoustic chamber allows the KX-UT series to offer crisp sound quality for crystal clear conversation.

Panasonic is also previewing the new KX-UT670, a highly expandable corded SIP phone with a seven-inch color LCD touch screen function that will help to transform business communication. Additional key features include HD Voice (G.722), two Ethernet ports, 3-way conference calling, IP camera integration, full duplex speakerphone, 100 entry phonebook and PoE ready.

VoIP Supply Adds Dual Mode Polycom SoundStation Duo Conference Phone

VoIP Monitor — Fri, 21 Oct 2011 21:16:41 GMT

VoIP Supply is happy to expand Polycom device offerings with the addition of the Polycom SoundStation Duo IP Conference Phone.

Whether your business is planning a migration to VoIP service or is already enjoying the benefits, the latest Polycom conference phoneworks in dual environments. The Polycom SoundStation Duo is a dual mode analog and VoIP conference phone offering support for both telephony platforms.

Need investment protection? The Polycom SoundStation Duo offers best-in-class ROI. Use the SoundStation Duo with traditional analog phone service or, switch it over to VoIP when you’re ready. And when you do switch to VoIP, this SoundStation conference phone is compatible with leading SIP-based PBX and softswitch platforms.

“Customers needing more board room device choices can look to innovative industry leader Polycom ,” said Garrett Smith, Chief Marketing Officer at VoIP Supply. “The flexible SoundStation Duo works with the service you have now and what you plan to have; future proofing your investment.”

Ooma Free Home Phone Service Cuts the Cord with New Wireless Offerings

VoIP Monitor — Wed, 14 Sep 2011 21:20:52 GMT

Ooma announces that it is extending even more freedom and flexibility to its customers with new wireless offerings.

With the first-of-kind introduction of the new Ooma Telo Air wireless adapter, the Ooma Telo can now be placed anywhere in the home within range of a Wi-Fi network. The company also announced that it is extending the availability of its Ooma Bluetooth service to all Ooma Telo subscribers to broaden the integration of mobile phones and Ooma home phone systems. Ooma Bluetooth service was previously only available to Ooma Premier service subscribers.

Ooma's first-to-market support for home Wi-Fi VoIP connectivity is made possible by Ooma's PureVoice HD technology with advanced voice compression and adaptive redundancy to overcome any signal degradation within the wireless network. More information on Ooma PureVoice HD technology is available at www.ooma.com/products/ooma-purevoice.

By using the new wireless adapter to untether the Ooma Telo from the modem or router, customers have more convenient access to their cordless phone base station, voicemail messages and other advanced features that Ooma offers. They also have the option of using the Ooma Telo as a wireless bridge for laptops and other devices throughout the home. Ooma is delivering broad home coverage and a high-quality calling experience using the latest 802.11n wireless standard combined with Ooma PureVoice HD technology, which delivers crystal-clear call quality and up to twice the fidelity of standard phone calls for a richer, more natural sounding conversation.

By adding an Ooma Bluetooth adapter to the Ooma Telo, customers gain the convenience of answering mobile phone calls on any home phone without worry of poor reception or dropped calls. It provides transmission range of up to 30 feet and supports up to seven Bluetooth devices.

Further integrating the mobile and home phone experience, Ooma also offers the Ooma Mobile HD App for Android and Apple mobile devices so customers can save money on domestic and international calls by tapping their Ooma account on the go over any Wi-Fi or 3G network.

Pricing
the Ooma Telo Air wireless adapter is available for $49.99. The Ooma Bluetooth Adapter is available for $29.99.

Ooma Premier is an optional level of service offering a suite of more than 25 advanced telephony services, including the new 911 Notification feature, and costs $9.99 per month. New customers signing-up for one-year of Ooma Premier service receive their choice of a free number port, Ooma Telo Handset, Ooma Bluetooth adapter or extended warranty.

VoIP Supply Revamps SpectraLink Store with Recertification from Polycom

VoIP Monitor — Tue, 13 Sep 2011 22:45:43 GMT

VoIP Supply is pleased to announce our refreshed SpectraLink store and device offerings including a re-certification from Polycom as a Certified Channel Partner for the line.

Polycom states, “As Partners working together, we are proud to recognize the commitment to quality and excellence this Polycom Certified Channel Partner provides.” VoIP Supply is certified by Polycom as an expert in SpectraLink phones, accessories, and infrastructure.

SpectraLink phones are a wireless solution for industries needing a mobile workforce to maintain operational efficiency and better serve customers. From entry level business grade to high-frequency usage, there’s a variety of SpectraLink handsets to choose from. SpectraLink phone accessories keep your wireless phone charged and within reach to stay as mobile as you are.

“Streamlining and revamping VoIP Supply’s SpectraLink store will empower customers with better choices,” said Garrett Smith, Chief Marketing Officer at VoIP Supply. “It’s great to be certified once again for our high level of SpectraLink service that mirrors Polycom’s VoIP solutions standards.”

2600hz and CloudTC Partner to Launch Integrated VoIP Application Suite and Smart IP Phone for the Enterprise

VoIP Monitor — Mon, 12 Sep 2011 20:43:22 GMT

2600hz and CloudTC announce the integration of Whistle, the distributed scalable VoIP platform, with the CloudTC Glass 1000, the industry's smartest IP phone. The combined solution offers enterprise users enhanced business productivity apps, such as teleconferencing and call transcription, alongside advanced telephony features, such as one-touch dialing and instant screen sharing. The two companies will demonstrate this powerful business communications solution at the upcoming ITEXPO West in Austin, Texas, at booth #210, from September 13-15. In addition, 2600hz is now an authorized distributor of CloudTC Glass smart IP phones, and will begin offering this integrated solution through a beta program to its VoIP carrier customers.

Whistle is a carrier-grade, distributed VoIP applications platform that provides the security and scalability of a hosted solution for organizations ranging from small businesses to the largest enterprises. The CloudTC Glass 1000 incorporates the benefits of the Android operating system, combining voice calls with advanced productivity features. VoIP software providers like 2600hz have enthusiastically embraced Glass because of its benefits of requiring less time to market, lower investment in R&D, and the ability to customize phone features as well as application suites for business users. An example is the 2600hz Hosted Conferencing whApp (Whistle App), which allows the conference call host to view a pop-up display on the Glass 1000 8.9 inch high-resolution color touchscreen LCD. The display lists all conference call participants and enables the host to manage the call by muting speakers, splitting calls into subgroups, or removing disruptive callers.

DSP Group Chipset Powers Grandstream Enterprise HD Phones

VoIP Monitor — Mon, 12 Sep 2011 20:37:01 GMT

DSP Group and Grandstream Networks announce that DSP Group's XciteR VoIP chipset is powering Grandstream's new high-definition voice IP desktop phones.

As the total cost of ownership of high-quality IP telephony devices nears that of legacy systems, the XciteR solution is ideal for developing innovative multiple VoIP end points. Providing the market's best price performance as well as myriad embedded hardware/software features, XciteR enables Grandstream to offer fully featured, multi-line IP phones that enhance productivity and deliver a rapid return on investment for enterprise customers.

The highly integrated XciteR chipset solution offers superior high-definition voice quality and clarity, as well as best-in-class acoustical echo cancellation, enabling the development of a high quality full-duplex, hands-free speakerphone. A state-of-the-art and mature solution, XciteR is powering an array of award-winning Grandstream HD IP phones, including the GXP21xx/14xx/11xx series. Known for their ease of use, rich telephony features, automated provisioning capabilities, and advanced security protection, the DSP Group-powered Grandstream HD IP phones are interoperable with most third-party SIP devices and platforms.

XciteR is anchored by a high-performance ARM926T application processor running Linux OS and strong DSP processor embedded in DSP Group's DVFD818x digital chip, along with a fully integrated power management and audio front-end IC. The solution also features a comprehensive software development kit to ensure short development cycles. Maintaining its small system footprint, XciteR offers an integrated DECT baseband and RF transceiver for easy implementation of wireless connectivity with DECT/CAT-iq handsets and/or headsets to enable full office mobility.

Velocity Networks Selects Sonus NBS5200

VoIP Monitor — Wed, 07 Sep 2011 21:18:56 GMT

Velocity Networks has selected the Sonus NBS5200 Network Border Switch to expand their service offering and provide improved end-user experience. Velocity Networks, which provides hosted PBX and SIP trunks, decided to replace its legacy first-generation SBC gear in favor of the NBS5200 largely because of its onboard transcoding capability. Prior to this decision, Velocity had to overcome the lack of onboard transcoding in its previous vendor's equipment by customizing work-around solutions.

Built on Sonus' ConnexIP platform, the NBS5200 represents the second generation in session border control. It is the highest density, highest scalability SBC solution in its class with integrated transcoding and media interworking as well as native support for IPv6 and IPSec encryption.

Also as part of the deployment, Velocity has selected Sonus' NetScore package, which is a powerful standalone solution that automatically captures, scores and analyzes Call Detail Records from Sonus network elements, then transforms that data into actionable information through visual reports and alerts.

Zeller Helps Canadians Save on Phone Bills, Adds netTALK DUO In All 273 Zellers Stores

VoIP Monitor — Wed, 07 Sep 2011 21:02:29 GMT

netTALK.com announces the netTALK DUO VoIP device and digital phone service is now being sold by Zellers in all of its stores across Canada. Zellers is Canada’s second-largest chain of mass merchandise discount stores, with 273 locations covering all 10 Canadian provinces.

“Having the netTALK DUO available for sale at all Zellers stores across Canada marks a major milestone in our drive into bricks n’ mortar and into this key market,” commented Anastasios ‘Takis’ Kyriakides, President and CEO of netTALK. “Canadian demand for our products is overwhelming, and we are proud to offer all Canadians the opportunity to enjoy free calls nationwide and to the US, via this easy-to-use telephone device and digital phone service.”

The netTALK DUO is a revolutionary VoIP device and digital phone service that enables free nationwide calls to any phone in Canada and the U.S. from anywhere in the world, as well as rock-bottom international rates and a slew of other features, detailed at www.netTALK.ca. No computer is necessary to use the netTALK DUO, as it simply plugs directly into a router or modem (or computer). The suggested retail price in Canada is CDN $79.95, including the entire first year of phone service and all other features, and only $39.95 for each additional year, with no additional fees or long-term contracts.

A Zellers store locator is here.

VoIP Supply Adds Portable VoIP Speakerphones from Phoenix Audio

VoIP Monitor — Thu, 18 Aug 2011 21:45:55 GMT

VoIP Supply is happy to announce the addition of new Phoenix Audio VoIP Speakerphones for expanding audio coverage wherever you need it.

Small footprint speakerphones for use with both PC’s and Mac’s from Phoenix Audio provide reliable loudspeaker output for virtually any sized room. Powerful features are not compromised in Phoenix Audio’s lightweight and sleek designs.

Phoenix Audio offers their Duet models for desktop conferencing and the Quattro2 models for conference rooms. All models are standard equipped with a USB interface for connectivity, power source, and VoIP recording and conferencing. Unique ability for daisy chaining units is available for near limitless coverage.

“VoIP Supply is excited to provide Phoenix Audio’s latest conferencing solutions,” said Garrett Smith, Chief Marketing Officer at VoIP Supply. “Phoenix Audio speakerphones don’t just look great they also pack a lot of advanced features into a small, portable package that maintains Phoenix Audio’s hallmark user-friendly, plug-and-play philosophy.”

VoIP Supply Adds Grandstream IP Surveillance Line

VoIP Monitor — Tue, 09 Aug 2011 21:02:35 GMT

VoIP Supply is pleased to announce a refreshed Grandstream Networks product line and the addition of a brand new category, Grandstream IP Surveillance.

Making customer research easier by revamping and reorganizing Grandstream products with newly segmented Grandstream Video Phones and Grandstream IP PBX appliance categories, VoIP Supply has also added the Grandstream IP Surveillance line featuring IP cameras for network-based surveillance.

Just like Grandstream’s VoIP devices, Grandstream IP Surveillance products offer the same broad interoperability, enhanced features, and flexibility at price-performance competitiveness allowing easy entry into IP network-based security systems.

“VoIP Supply’s Grandstream line just got a lot stronger with Grandstream IP Surveillance products,” said Garrett Smith, Chief Marketing Officer at VoIP Supply. “VoIP and IP camera technology are intertwined so it’s great to give customers a single stop to research all of Grandstream’s cost effective network-based solutions.”

SPIRIT DSP Signs Fifth Licensing Agreement with Polycom

VoIP Monitor — Thu, 07 Jul 2011 20:14:18 GMT

SPIRIT DSP has signed its fifth licensing agreement with Polycom. Under terms of the agreement, Polycom has licensed SPIRIT's product to support quality videoconferencing on the Polycom CX5000 Unified Conference Station. As a Polycom-branded version of the Microsoft RoundTable collaboration and conferencing device, the CX5000 system provides engaging group telepresence and also doubles as an analog conference phone.

The Polycom CX5000 Unified Conference Station adds a compelling and engaging group video experience to Microsoft Live Meeting session or to Microsoft Office Communicator conversation. With five cameras and six microphones, the CX5000 is primarily a USB peripheral device that delivers a unique, engaging 360-degree group video experience to Live Meeting 2007 applications, and when used only with Microsoft Office Communication 2007, the client functions as a video switched webcam. It also allows participants who are not within the organization's boundaries or managed video environment, meaning anyone with a PC can download the Live Meeting client and view the meeting. The device has TI C54CST processor inside that powers it, which also runs on SPIRIT communication software.

SPIRIT's engine SDKs are designed for real-time, multipoint, videoconferencing. The software uniquely combines scalable audio with scalable video, a necessary marriage to ensure the highest quality conferencing experience. The voice engine includes highly optimized standard voice codecs, and a patent-free wideband and highly scalable SPIRIT IP-MR voice codec. The video engine includes an H.264 SVC video codec that addresses video packet loss and video quality improvements, including a network adaptation module that compensates for network jitter and packet loss.

netTALK DUO Now Available from Dell Canada

VoIP Monitor — Wed, 15 Jun 2011 17:14:56 GMT

netTALK.COM announces that the netTALK DUO VoIP device and digital phone service is now being sold by Dell Canada.

“Canadians clearly recognize the unbeatable value of the netTALK DUO, which lets them say ‘Goodbye’ to their phone company, forever.”

“There is surging demand for our products in Canada, so making the netTALK DUO - available at Dell.com since April - also available for sale with Dell Canada, is a natural next step,” said Anastasios ‘Takis’ Kyriakides, President of netTALK. “Canadians clearly recognize the unbeatable value of the netTALK DUO, which lets them say ‘Goodbye’ to their phone company, forever.”

The netTALK DUO is a revolutionary VoIP device and digital phone service that enables free nationwide calls to any phone in Canada and the U.S. from anywhere in the world[FN1], as well as rock-bottom international rates and a slew of other features, detailed at www.nettalk.com. No computer is necessary to use the netTALK DUO, as it simply plugs directly into a router or modem (or computer). The suggested retail price in Canada is CDN $79.99, including the entire first year of phone service and all other features, and only CDN $39.95 for each additional year, with no additional fees or long-term contracts.

“We are very pleased to offer the netTALK DUO to our Canadian customers,” commented Phil Bryant, Vice President and General Manager, North America at Dell. “It is a nice complement to our product line-up, and offers an unmatched value to Canadians who wish to lower their phone bills.”

snom Phones Validated for Use with Interactive Intelligence Unified Business Communications Solutions

VoIP Monitor — Tue, 07 Jun 2011 15:59:06 GMT

snom announced that its snom 3xx series IP desktop phones have passed interoperability testing for use with the Interactive Intelligence all-in-one IP communications software suite, Customer Interaction Center.

The combination of snom phones and CIC gives contact centers and enterprises an advanced, cost-effective, and easily deployed SIP-based unified communications solution from which to manage dozens to hundreds of individual endpoints.

snom’s suite of VoIP phones includes advanced IP phones such as the snom 3xx series, full-color touchscreen desktop phones such as the snom 870, wireless DECT phones such as the m9 and related endpoints, such as the MeetingPoint conference phone. snom developed one of the industry’s first SIP-based solutions for endpoints and as a result, all snom endpoints are built on one of the most mature SIP stacks available, allowing for simple installation, industry-wide interoperability and crystal-clear sound quality. As unified communications systems have proliferated, the advanced technology and open standards in every snom product has proven to be an ideal combination for providing enterprises with cost-effective and feature-rich IP telephony.

Interactive Intelligence designed CIC as a scalable, standards-based, single-platform solution that delivers comprehensive multichannel applications minus the cost and complexity introduced by multipoint products.

3CX Selects Grandstream IP Phones as Preferred Phones for 3CX Phone System v10

VoIP Monitor — Tue, 07 Jun 2011 15:56:19 GMT

3CX announces they completed interoperability testing for 3CX Phone System version 10 and Grandstream’s latest line of IP Phones – the new executive series GXP 2110, GXP 2120, GXP 2100 and the GXP 1450. The integration between 3CX and Grandstream products means customers are assured of an integrated and fully supported PBX solution. 3CX has created special support for Grandstream phones which includes Plug and Play provisioning, network wide updating of Grandstream phones and automatic provisioning.

3CX Phone System V10 is the latest 3CX release and allows complete network wide management of Grandstream phones from the 3CX Management Console, saving considerable time for administrators. Features include:

Plug and Play Provisioning – To connect Grandstream phones to 3CX, they just need to be plugged in to the network. The phones will automatically show up in the 3CX Management Console, allowing the administration to assign an existing extension or create a new one with a few mouse clicks.
Network Wide Firmware Updating – Administrators can see the firmware versions of Grandstream phones, and apply new firmware in batch mode to all or selected Grandstream IP phones on the network.
Network Wide Configuration – Language, codec preference and time zone can be remotely configured from the 3CX Management Console.
Assured Interoperability – Grandstream and 3CX undertake to test each release of their software PBX and IP Phone firmware and fix any interop issues. This assures customers of a fully tested and supported solution.

The combined solution is supported by both 3CX and Grandstream and the support teams of both companies will assist customers in the configuration of each others product.

Grandstream Announces New HD IP Phones the GXP1400/1405

VoIP Monitor — Thu, 02 Jun 2011 19:29:50 GMT

Grandstream announced two new additions to the GXP line of Enterprise HD IP Phones, the GXP1400 and GXP1405. The newest additions to the award-winning GXP family of HD IP Phones, the GXP1400/1405 bring top-quality HD audio and comprehensive functionality to the market at an extremely competitive price.

The GXP1400/1405 are next generation small-to-medium business IP phones that feature 2 line keys with dual-color LED, single SIP account, a 128x40 pixel graphical LCD display with support for multi-languages (including English, German, French, Spanish, Italian, Russian, Chinese, Korean, Japanese, etc.), 3 XML programmable context-sensitive soft keys, dual network ports, integrated PoE (GXP1405 only), and 3-way conferencing. The GXP1400/1405 delivers superior wideband HD audio quality, high performance full duplex speakerphone with advanced acoustic echo cancellation, rich and leading edge telephony features, auto provisioning for easy and secure deployment using TR069 and AES encrypted XML configuration file, advanced security protection for privacy using TLS/SRTP/HTTPS, and broad interoperability with most third-party SIP devices and leading SIP/NGN/IMS platforms as well as IPPBX systems. It is a perfect choice for small-to-medium businesses looking for a high quality, feature rich IP phone with an extremely affordable cost.

For more information on the GXP1400/1405.

Pricing and Availability

The GXP1400/1405 will be commercially available through Grandstream's worldwide distribution channels at a list price of US$59 and US$65 respectively, by the end of May, 2011.

Fonality Announces Trade-In Program to Help Growing Businesses Transition to Cloud-based VoIP Solutions

VoIP Monitor — Wed, 25 May 2011 17:43:10 GMT

Fonality announces the launch of its “Turn it In/Turn it Up” trade-in program, providing small and mid-size businesses with a cost-effective solution to upgrade their business communications capabilities. Through this program, SMBs can trade their outdated, legacy phone systems for fair market value and receive credit toward Fonality’s cloud-based Unified Communications, VoIP and contact center solutions.

Specifically designed for SMBs, Fonality’s six-time award-winning communications solutions are simple to use, easy to manage and affordable to deploy. The company’s cloud-based solutions deliver Fortune 500 features without the costly hardware, infrastructure or lengthy implementation cycles associated with legacy on-premise IP systems. Productivity-enhancing features, such as unified messaging with email, secure chat and Microsoft Outlook contact integration, are combined with audio conferencing, photo caller ID, visual voicemail, email/text, ring-back and on-the-fly call recording. Fonality solutions start at a cost of $30 per user per month, which includes calls, and offer a total cost of ownership up to 50 percent less than legacy phone solutions.

New Fonality customers can experience substantial productivity gains, as a recent Webtorials “State-of-the-Market” report indicated that Unified Communications can help SMBs regain hours of lost employee productivity each week. Fonality Heads Up Display is an award-winning UC platform that connects phones, desktops and important business applications into a single, user-friendly interface.

The “Turn it In/Turn it Up” program is designed to valuate trade-in inventory according to brand, model, quantity and quality. To qualify for the program, all equipment must be in working order, without damage beyond normal wear and tear. Legacy solutions from Cisco, Avaya, Shoretel, Mitel, Nortel, NEC, Panasonic, Fujitsu and Siemens are eligible and additional competitive displacement offers may be available.

In partnership with a leading buyer for used telephony equipment, the seller will receive account credit to apply to a new Fonality solution within 30 days of the buyer’s point of receipt. The “Turn it In/Turn it Up” program also facilitates environmentally responsible recycling, refurbishment or disposal of used equipment.

VoIP Supply Adds Ruckus Wireless “Smart Wi-Fi” Products

VoIP Monitor — Thu, 19 May 2011 19:16:10 GMT

VoIP Supply is pleased to announce the addition of Ruckus Wireless products that specialize in Wi-Fi for both enterprise and service provider markets.

Routing Wi-Fi signals can be unreliable due to interference, obstacles, and the sometimes erratic behavior of wireless signals themselves. Ruckus Wireless has developed “Smart Wi-Fi” to combat those barriers to a secure wireless network and restore faith in what companies love about Wi-Fi.

“VoIP Supply is excited to offer revolutionary Wi-Fi products from a manufacturer with such an impressive track record,” said Garrett Smith, Chief Marketing Officer at VoIP Supply. “Today’s Wi-Fi systems face increasing levels of interference and obstacles. Ruckus delivers the unthinkable - Reliability, range and speed by automatically adapting to the surrounding environment.”

For additional information about the Ruckus Wireless systems available through VoIP Supply, call toll-free 1-800-398-8647, or visit http://www.voipsupply.com/manufacturer/ruckus-wireless.

Report: The Taiwanese Enterprise VoIP Equipment Industry, 1Q 2011

VoIP Monitor — Tue, 10 May 2011 15:39:11 GMT

Research and Markets has announced the addition of the "The Taiwanese Enterprise VoIP Equipment Industry, 1Q 2011" report to their offering.

This research report presents shipment volume and value forecast and recent quarter review of the Taiwanese enterprise VoIP equipment industry. The report includes IP phone and VoIP gateway shipment volume, shipment value and ASP, as well as shipment volume and share by maker, solution provider, production location, shipment destination, business type, and customer portfolio. The content of this report is based on primary data obtained through interviews with enterprise VoIP makers. The report finds that in the fourth quarter of 2010, Taiwanese IP phone shipment volume and value saw double-digit year-on-year growth while Taiwanese enterprise VoIP gateway shipment volume and value saw year-on-year declines. Taiwanese IP phone and enterprise VoIP gateway shipment volume is expected to grow upwards steadily and record year-on-year growth in the first quarter of 2011. Overall shipment value of the Taiwanese enterprise VoIP equipment industry is also projected to experience year-on-year growth in full-year 2011.

Norango Enriches its Portfolio with VoIP Handsets from Snom

VoIP Monitor — Fri, 06 May 2011 16:15:03 GMT

Norango has become a value-added reseller for snom, allowing it to offer its customers the full range of snom IP phones. The Norango packages use Hosted-PBX and handsets provided by snom, allowing them to offer a feature rich solution at a competitive price.

The advanced features of snom handsets combined with the scalability and security of a cloud-based PBX solution sees Norango broaden its customer offering and harness a cloud-based approach to telephone systems. In addition to selling snom products, Norango are part of the globally accredited Certified Engineer programme, meaning their customers are safe in the knowledge that Norango can provide technical support with certified training behind them.

Telephone answering remains the core of Norango’s business, but with the addition of the snom range of products and hosted PBX, small and medium size businesses will no longer have to go to multiple suppliers to fulfil their telecom requirements. The addition of snom to Norango’s product range comes on the back of the launch of snom Channel UK, and EFL’s appointment as UK distributor earlier this year.

netTALK DUO Now Available on Dell.com

VoIP Monitor — Mon, 18 Apr 2011 19:47:11 GMT

netTALK.com announces that the netTALK DUO VoIP device and digital phone service is now being sold at Dell.com.

The netTALK DUO is a revolutionary VoIP device and digital phone service that enables free nationwide calls to any phone in the U.S. and Canada from anywhere in the world, as well as rock-bottom international rates and a slew of other features, detailed at www.nettalk.com. No computer is necessary to use the netTALK DUO, as it simply plugs directly into a router or modem (or computer). The suggested retail price for the devise is $69.95, including the entire first year of phone service and only $29.95 each year after.

The netTALK DUO is available from Dell.com via the following link: http://accessories.us.dell.com/sna/productdetail.aspx?c=us&l=en&s=dhs&cs=19&sku=A4889205&baynote_bnrank=0&baynote_irrank=0&~ck=baynoteSearch

netTALK DUO Now Sold at Giant E-tailer NewEgg

VoIP Monitor — Tue, 12 Apr 2011 17:37:09 GMT

netTALK.COM announces that the company’s flagship product, the netTALK DUO VoIP device and digital phone service, is now available for sale at www.NewEgg.com.

The netTALK DUO is a revolutionary VoIP device and digital phone service that enables free nationwide calls to any landline or mobile phone in the U.S. and Canada from anywhere in the world, as well as rock-bottom international rates and a slew of other features, detailed at www.nettalk.com. No computer is necessary to use the netTALK DUO, as it simply plugs directly into a router or modem (or computer). The netTALK DUO has a suggested retail price of $69.95, including the entire first year of phone service and only $29.95 each year after (no monthly fees or long-term contracts).

The netTALK DUO is now available from at NewEgg.com via the link.

snom Teams with Telchemy to Provide an Integrated VoIP Performance

VoIP Monitor — Mon, 11 Apr 2011 17:25:16 GMT

snom and Telchemy announce that, after a battery of tests, snom’s full portfolio of IP phones and endpoints has been approved as interoperable with Telchemy’s award-winning DVQattest and SQmediator performance management products.

The combination of Telchemy’s DVQattest and SQmediator applications and snom’s IP desktop phones provides network managers, enterprises and end users with a full suite of pre-deployment and in-call performance management tools for business VoIP networks, allowing for faster and more efficient deployments and troubleshooting.

Telchemy’s SQmediator is an advanced VoIP performance management application that collects, correlates, analyzes, and displays call quality reports received from a wide range of endpoints, such as snom desktop phones, the snom MeetingPoint conference phone or the snom m9 wireless DECT phone. DVQattest is a scalable, cost-effective application for both pre-deployment testing and proactive service level monitoring of enterprise VoIP and IP video networks, allowing users to actively monitor networks and troubleshoot VoIP problems.

snom’s suite of VoIP phones includes advanced IP phones such as the snom 3xx series, full-color touchscreen desktop phones such as the snom 870, wireless DECT phones such as the m9 and related endpoints, such as the MeetingPoint conference phone. All built with open SIP firmware allowing for simple installation, industry-wide interoperability and crystal-clear sound quality. As unified communications systems have proliferated, the advanced technology and open standards in every snom product has proven to be an ideal combination for providing enterprises with cost-effective and feature-rich IP telephony.