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 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.

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.

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 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 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 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.

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 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.”

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.

One such development is what is being called the so-called "Internet in a suitcase" (above), which has been funded by $2 million in State Department money via a grant to members of the New America Foundation.

One suitcase in and of itself doesn't create an Internet. Instead, a series of such devices, relying on "mesh network" technology creates an "invisible wireless web without a centralized hub." Using these devices, the transmission, whether it's voice, image, e-mail, or any data, can hop from "suitcase to suitcase," bypassing the official network.

If it sounds like sort of a series of cell phone towers, it could indeed be thought of in those terms. Aaron Kaplan, an Austrian cybersecurity expert says that he has helped set up a functioning mesh network in Vienna. He added that similar related systems have already been used in Indonesia, Venezuela, and elsewhere in the world. His work will be used as part of the suitcase project.

Source

 

Samsung announced two new models in their Galaxy Tab line, the Galaxy Tab 10.1 and the Galaxy Tab 8.9. The Tab 10.1 has been redesigned since we saw it at MWC and both tablets can now boast that they are the world's thinnest tablets at 8.6 mm.

Both will feature an updated Live View interface on top of Honeycomb, specifically designed for the platform. They each have dual core 1GHz processors, will support HSPA+ as well as have WiFi-only versions. 

Samsung also announced pricing and availability today, something that they don't often do. 

Here is what they announced:

• Galaxy Tab 10.1 WiFi-only 16GB: $499, available June 8th
• Galaxy Tab 10.1 WiFi-only 32GB: $599, available June 8th
• Galaxy Tab 8.9 WiFi-only 16GB: $469, available "early summer"
• Galaxy Tab 8.9 WiFi-only 32GB: $569, available "early summer"

Samsung announced some pretty impressive additions to their Galaxy Tab lineup and is stepping up their game in terms of pricing.

Source

"We designed WiFi Direct to unleash a wide variety of applications which require device connections, but do not need the Internet or even a traditional network," said Edgar Figueroa, CEO of the WiFi Alliance, in a statement. "WiFi Direct empowers users to connect devices — when, where and how they want to, and our certification program delivers products that work well together, regardless of the brand."  

The five WiFi Certified WiFi Direct products, the Alliance announced, are:  

• Atheros XSPAN Dual-band 802.11n PCIe mini card;  

• Broadcom Dual-Band 802.11n 2x2 MiMO PCIe half mini card;  

• Intel Centrino Advanced-N 6200 WiFi card;  

• Ralink MIMObility 802.11 2x2 PCIe half mini card; and  

• Realtek HM92C00 PCIe mini card. 

WiFi Direct technology is intended for both consumer and enterprise devices, as it offers the latest WPA2 security protections. It also works with older WiFi Certified devices, offers a “push-button” setup and—unlike the limited distances of Bluetooth technology—works at a typical WiFi range.

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According to Grandstream, the new models were designed to “meet the need for crystal clear voice communication.” The new GXP models come packed with business productivity features and applications, HD audio quality, and broad SIP interoperability with third-party SIP platforms and devices.

The introduction of the GXP2120 and GXP2100 telephones add to Grandstream’s HD voice solutions, joining the recently introduced GXP2110. “Multiple desktop model options give SMBs the ability to select a HD telephone that best fits their size and need,” states a Grandstream Networks press release submitted to EH Publishing.

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The result of the MegaPath-Speakeasy-Covad merger and one of the nation’s largest end-to-end communications networks, announced its limited-time free Cisco SPA 303-3 IP phone promotion. New business customers who purchase unlimited or global Hosted Voice calling plans and have a minimum of five lines will receive IP phones featuring a high-resolution graphic display for each employee. This program is part of the second phase of MegaPath’s post-merger integration, offering common products, pricing and promotions across all three company websites.

Forecasts from ABI Research1 show that the VoIP market is set to double in size in the next five years, exceeding $20 billion by the year 2015. By offering free phones, MegaPath is removing the last barrier to VoIP adoption – upfront costs. Speakeasy successfully implemented a similar promotional program earlier this year, which resulted in an increase in new customer inquiries by more than 40 percent.

MegaPath has a long-standing relationship with Cisco, as the company has been providing Cisco-powered MPLS nationwide network-based solutions for more than 10 years. MegaPath is a Cisco Master Managed Service Provider with voice and VPN certifications and small business specialists.

In addition to offering free phones to new customers, MegaPath is enhancing the functionality of its Hosted Voice calling plans with new productivity-enhancing features for all voice customers. This includes features such as:

• Push to Talk: Point-to-point intercom functionality between two phones in the same group, which can be used with Instant Group Call to simulate paging functionality.
• N-Way Calling: Enables a user to add up to 15 total call participants.
• Hoteling: Allows a user to log in as a guest on a host phone and use it as they would their normal desk phone. All incoming calls are routed to the host phone, while all outbound calls appear to come from the user’s desk phone.
• Answer Confirmation: This enhancement to Simultaneous Ring and Sequential Ring allows a user to prevent callers from leaving voice messages on personal cell phones when using Find Me/Follow Me features.

These features are available at no cost. To take advantage of these new features, current customers can simply reboot their phones.

MegaPath is also adding Instant Group Call, available for an additional fee, which allows users to instantly set-up a conference bridge for up to 20 telephone numbers. This feature is useful as an emergency line or for recurring team meetings. Instant Group Call can also be used in conjunction with Push to Talk to simulate paging functionality.

Grandstream Networks has extended the portfolio of its popular GXP series enterprise SIP telephones with the introduction of the new GXP2110. Based on Grandstream’s broadly interoperable SIP stack, the GXP2110 SIP telephone delivers superior HD audio quality for crystal clear voice communications, packed IP telephony features and integrated Web applications, as well as support for highly flexible XML customization and strong security protection.

Extending Grandstream’s commitment to quality, feature, and performance, the new GXP2110 SIP phone comes standard with HD handset and high performance full duplex speakerphone, a broad range of voice codecs, dual network ports with integrated PoE, 4 line keys, 3 soft keys, 18 programmable BLF keys, 5-way conferencing, large 240x120 backlit graphical LCD with 8-level grayscale, multiple languages, large phone book and call log (2,000 records), automated provisioning using TR-069 and encrypted XML file, extension module expandability, and a number of integrated Web applications such as real-time local weather, stock, currency, RSS news, etc. In the near future, more advanced Web applications will continue to be integrated via FREE firmware upgrade and an open Web service API will also be provided for advanced custom enterprise/Web application development. Grandstream plans to continue to expand the GXP phone series with new models during the rest of this year.

Pricing and Availability

The GXP2110 is commercially available for purchase now through Grandstream’s worldwide distribution channels at a MSRP of US$139.

REDCOM unveils the new SLICE IP. The SLICE IP is an advanced Class 4/5 softswitch core powered by TRANSip, REDCOM’s VoIP technology suite that provides an innovative IP communications solution. TRANSip integrates key IP Multimedia Subsystem elements in one single platform, bringing call management functionality and directory services for IP subscribers.

The SLICE IP offers features and interoperability highlighted as follows:

• Class 4/5 Tandem / End Office applications
• SIP-based VoIP capabilities – up to 2,000 SIP subscribers (per unit)
• SIP trunks – to optimize your operating cost
• Built-In Compression – to manage bandwidth usage
• SNMP monitoring
• Call Detail Recording – on a per call basis
• External Database Interface – enhancing administration support
• Host Control Interface – for additional call management capabilities
• IP Centrex business line groups – for additional revenue stream
• Modem over IP (V.150.1) – for secure end instrument interoperability
• Fax over IP (T.38) – for end instrument interoperability
• Compatibility with industry-standard Unified Messaging platforms

The SLICE IP is just 1U high (4.4 cm / 1.75 inches) and features a simple, integrated design that makes it easy to deploy, reducing installation and operating costs while delivering a host of new revenue generating features. Flexible compression methods are built-in, helping service providers manage low-revenue subscribers by reducing bandwidth usage, while retaining high-end subscribers with premium services.

In addition, the SLICE IP’s extensive SIP-based VoIP and trunking capabilities allow service provides to take advantage of their broadband deployment while adding new revenue. The SLICE IP’s capabilities ensure that the service provider’s network resources fit their business model for current and future growth, allowing them to quickly adapt to new communications requirements.

The SLICE IP is scalable and flexible for future growth. Its scalability allows the SLICE IP to be stacked with another SLICE IP, increasing the subscriber base up to 4,000 per node. Its interoperable flexibility also allows the SLICE IP to be stacked with REDCOM’s flagship SLICE® 2100™ system when access to TDM or legacy protocols is needed. The SLICE IP can be custom-configured for multiple applications, including Corporate Networks, Central Offices, Emergency Response, Call Centers and Command & Control.

The MTX-3 offers seamless interaction with AV and environmental systems, providing true feedback of all settings, and displaying metadata information for all digital media.  Crestron’s infiNET EX wireless technology provides reliable two-way communications throughout a residence or commercial structure utilising a 2.4 GHz mesh network.

A complete infiNET EX network uses the lighting dimmers and other devices throughout the structure as wireless relay stations, each receiving and passing on wireless commands to the central gateway.

Every device that is added to the network effectively increases the range, strength, and reliability of the entire network by providing multiple redundant signal paths, ensuring that every button press is executed instantly and consistently.

The MTX-3 can also communicate directly with the gateway if no other infiNET EX devices are installed. Up to six MTX-3s can be assigned to a single gateway.

Source

“With new components such as broadband data communications, enhanced voice over IP services, and advanced functions designed specifically for ground forces, the system provides significant levels of increased performance and operational capabilities,” as the company claims.

EB Tough VoIP can be used as a stand-alone system or the solution units EB Tough VoIP Terminal and the EB Tough VoIP Network Extender can be integrated to a customers' existing communications infrastructure. According to the company, the products can be leveraged in many ways, including:

• Rugged and secure VoIP clients for military communications
• Radio over IP clients
• Tactical data radio environment
• Demanding environments where limitations in communications can greatly affect operational capabilities

The EB Tough VoIP Terminal can be used with application specific headsets and handsets that are commonly used with tactical radio or digital field telephony equipment.

The EB Network Extender is an integrated unit that provides the functionalities of a VoIP terminal, Ethernet switch and multi-channel SHDSL (Single-Pair High-speed Digital Subscriber Line) modem and it can be used to provide broadband wired connectivity, and VoIP services to the field, in LAN or WAN configuration.

"EB Tough VoIP was designed to fulfill the communications needs of military units, and we continue to strive towards achieving that goal," said Mikko Viitaniemi, senior manager, Defense Products, Wireless Solutions at EB

Source:  VoIP Biz News

"As desktop phones continue to migrate from being voice-centric devices to acting as complete dashboards for unified communications, the speed of application execution is becoming more and more critical," said Dr. Michael Knieling, Executive Vice President of Marketing and Sales for snom technology AG. "The addition of a gigabit switch came from customer requests, and we are very pleased that we were able to add innovation while keeping the same price point."

The snom 821 offers a large high-resolution TFT color display for brilliant depiction of call lists, phone directories, and caller information via the integrated XML browser. In addition, the phone's wideband technology captures more than double the frequency spectrum that a standard handset does, providing crystal clear sound quality.

As a business communications endpoint, the snom 821 features snom's industry-renowned firmware, allowing for robust security supported through secure integrated open VPN, TLS and SRTP, and high sound quality using the handset or the speaker function. The snom 821 also supports all of the leading SIP-based IP PBX and unified communications environments, including Microsoft Office Communications Server 2007 R2.

As with all of snom's desktop phones, the snom 821 also offers the necessary functionality for any business user, including features such as 5-way conferencing, multiple ringtones, and up to twelve different SIP identities.

The snom 821 and 870, as well as associated accessories such as the PA1 public address system, the MeetingPoint conference module, and the snom Vision sidecar, are designed to take the desktop business phone beyond its traditional position as a person-to-person voice communications device and make it a truly interactive and intelligent business communication device. Armed with full-color and touchscreen interfaces, web programmable ports, and a wide array of interoperable unified communications elements, the entire suite of next generation snom devices represents the future of desktop interaction.

 Source:  Snom Technologies AG

Virtual PBX has expanded its Open VoIP Peering service to include support of any SIP-compliant phones. Users of any SIP-compliant softphone or desktop phone can now tap into the benefits of the Virtual PBX Open VoIP Peering service. This service allows small businesses to take advantage of the robustness and features of traditional hosted PBX services and leverage the cost-effectiveness of VoIP technology. Today’s announcement allows customers more freedom to mix and match SIP-compliant phones with Virtual PBX’s hosted PBX services.

Breaking Down Proprietary Barriers for Users

Hosted IP-PBX providers that support native VoIP traffic usually have closed proprietary implementations. Customers have usually been locked into VoIP, softphone and even telephone handset offerings from the same company that provides the PBX service. Virtual PBX is giving its users more choice by supporting open standards. Rather than being limited to the features and capabilities of just a single vendor, clients can choose the offering that best fits their needs. Virtual PBX believes this will ultimately provide richer services for clients, since each player can bring unique content and features to the table. At the same time, total costs can be cut by sending more traffic through less expensive technologies. There is no extra cost from Virtual PBX for accessing softphones as opposed to other players who charge extra for proprietary digital-enabled phone extensions.

Up until this announcement, Virtual PBX’s VoIP peering implementation worked only with VoIP service that used the North America Numbering Plan to address phones. Now, customers can choose any phone that complies with established SIP standards, including products from top vendors like Truphone, Callcentric, Ekiga and Gizmo5, giving full freedom of choice in price, features and vendors.

Reducing Costs

Hosted IP-PBX services provide businesses with phone systems without the high costs of installing and maintaining PBX equipment. Typical prices range from $25 to $60 per user per month, averaging about $40 per user. Virtual PBX’s VoIP Peering product, the iVPBX plan, reduces that price to just $10 per user. The iVPBX product includes all standard Virtual PBX features, such as a virtual receptionist, ACD queuing, voicemail, conferencing and follow-me forwarding.

WorldGate unveils the Ojo Vision digital video phone, the next generation in voice and video over broadband communications services. Ojo Vision will be previewed this week in Booth 2161 on the floor of The Cable Show 2010 in Los Angeles along with the rest of its lineup of award-winning family of Ojo digital video phones, telephony services and turn-key services platform.

WorldGate is an all-in-one provider of innovative, IP multimedia telephony services combining world-class engineering and hardware manufacturing with a complete, turn-key digital voice and video phone service. The new Ojo Vision features plug-and-play installation, a sleek design, brilliant high-resolution seven-inch screen with high-fidelity audio and stunning “Ojo Quality” video performance. WorldGate’s newest offering also features a robust suite of VoIP telephony features and rich multimedia applications along with advanced networking technology that delivers an unprecedented user experience—all at an affordable cost.

Ojo Vision Digital Video Phone features include:

Highest Quality Video:

• Seven-inch digital color 16:9 TFT LCD (800 x 480 high-resolution), fully backlit and adjustable
• CMOS sensor camera with auto focus, auto exposure and high/low light sensitivity
• TI DaVinci digital video dual-core chipset with advanced video/audio codec implementation
• Real-time adaptive bandwidth with configurable bit rates from 80kbps to 1Mbps

Innovative Bonus Features:

• Screen Saver/Digital Photo Frame allows users to display favorite photos and images when the phone is not in use
• Connect to a large screen LCD/TV/projector for expanding viewing possibilities
• Future enhancements will include powerful three-way video conferencing capabilities

Advanced Telephony Capabilities:

• Complete feature suite including three-way calling, caller ID, call waiting, call forwarding, auto attendant, speed dial and more
• FXS Telephone Port provides digital integration for existing household phones
• NAT transversal technology for zero configuration and plug-and-play installation
• Intuitive multi-language graphical user interface

Partnership Opportunities:

• WorldGate’s patented platform allows service providers, agents, and customer acquisition companies to offer turnkey digital voice and video phone services with minimal risk and investment while providing speed-to-market capability
• Retail, agent, wholesale, white label, and private label programs are available.

Availability:

Ojo Vision is scheduled for general availability in the near future through WorldGate and WorldGate Authorized Channel Partners. Monthly calling plans, including features and rates will be announced concurrent with the commercial availability of the Ojo Vision.

Mirial announces that Iris2Iris joined the Certified for Mirial Softphone program with its eye contact equipment IRIS EQ5-WIN-AE.

The first to provide Full-HD support, embedded MCU functionalities and advanced media encryption, Mirial Softphone is a professional software-only client for desktop video conferencing. It is fully compliant with all major visual communication standards, it seamlessly integrates into every H.323 and SIP visual communication network and is available both for Mac and PC.

IRIS is a hardware device that creates online eye contact for desktop video conferencing. IRIS is provided with a 15 inch monitor and integrated microphone, video and speakers. Based on plug and play, IRIS is as easy to be installed as a webcam.

Both teams at Mirial and Iris2Iris reported a fulfilling user experience with clear audio and high quality image, enriched by face to face feeling thanks to eye contact. Moreover, the IRIS can be placed on your desktop as a second monitor to be used for video conferencing whenever is needed.

A 30-day try&buy version of Mirial Softphone can be downloaded from Mirial website.