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.

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).

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.