• Dual 10/100Mbps Ethernet ports
• Support Caller ID display or block, call waiting, hold, mute transfer & more
• 3-way conferencing
• Integrated Power-over-Ethernet (802.3af) and web interface
• Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
• Large phonebook (up to 2,000 contacts) and call history (up to 2,000 records)

• 7″ resistive touch screen color LCD (800×400), 1.3M pixel CMOS camera with privacy shutter
• Dual switched 10M/100M Ethernet ports with integrated PoE, integrated WiFi (802.11b/g/n)
• Dual USB ports, SD/MMC/SDHC, headset, stereo audio output, video output
• High fidelity full-duplex speakerphone with high performance acoustic echo canceller
• H.264/H.263/H.263+ video compression at bit rate of 32kbps-2Mbps, frame rate of up to 30fps, video resolution of VGA/WQVGA/QVGA (H.264) and CIF/QCIF (H.263/H.263+)
• G.711 (a/u-law), G.722 (wideband), G.723.1, G.729A/B, GSM-FR, G.726-32 voice codec and AAC, MP3, WMA, Real, Ogg-Vorbis audio codec
• Multi-language, full Web browser, weather, news, stocks, currency, world clock, calendar, games, Google Voice, Internet radio, YouTube, movie trailer, Last.fm, digital photo frame, photo album integration with Yahoo Flickr/Photobucket/Phanfare, Yahoo/MSN/Google IM (pending), Facebook and Twitter (pending), virtual BLF keys, SDK/API, etc.
• Smart NAT traversal technology to allow zero-configuration plug-and-play
• Strong security protection based on TLS/SRTP/HTTPS and AES

Simplicity and ease-of-use.
The VVX 310 phone is high-quality, brings cost effective solutions to any SOHO through advanced UC features. The graphic user interface of the VVX 310 makes usability and navigation easy, and requires little or no training.

Best-in-class deployment and administration
The VVX 310 phone is easy to deploy and simple to manage. Using an enterprise-grade, web-based, configuration method that allows administrators to easily provide and maintain even a small number of phones throughout the entire organization.

• Backlit grayscale graphical LCD (208 x 104)
• 6 line or speed dial keys
• HD Voice up to 7KHz on all audio paths (Speaker, Handset, Headset)
• 2 x Ethernet 10/100 or GigE (VVX 310)
• Asian character support
• Hard Keys: 12-key dial pad, home, speaker, mute, headset, volume, messages, hold, transfer
• 4-way navigation cluster with center “select” key
• Supports VVX Expansion Module and VVX Color Expansion Module (Expandability up to 3 modules)

The GXW4104 include 4 ports of FXO. Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments. Manageability, a simple configuration, superb voice and video quality and feature rich functionality.

Features:

• 4 FXO port media gateways
• Video surveillance port
• External power supply
• Two RJ-45 ports 10/100 Mbps (switched or routed)
• TFTP and HTTP firmware upgrade support
• Multiple SIP accounts (choice of 3 profiles per account)
• Programmable PSTN line settings for different countries/regions
• 1 or 2 stage dialing
• Caller ID
• G.168 – echo cancellation
• Flexible DTMF transmission: In Audio, RFC2833, SIP Info or any combination of the 3
• Selectable, multiple LBR coders per channel
• T.38 Fax compliant

The GXW4216 is a next generation high performance high-density analog VoIP gateway that is fully compliant with SIP standard and interoperable with various VoIP systems, analog PBX and phones on the market. It features multiple FXS analog telephone ports, superb voice quality, rich telephony functionalities, easy provisioning, flexible dialing plans, advanced security protection, and strong performance in handling high volume voice calls. The GXW42XX series gateway offers small and medium businesses a cost-effective hybrid IP and analog telephone system that allows them to enjoy the benefits of VoIP communications while preserving investment on existing analog phones, Fax machines and legacy PBX systems.

• 16 telephone FXS ports with both RJ11 and 50-pin Telco connectors, as well as per port LED
• Gigabit Ethernet and a backlit 128×32 graphic display with support for multiple languages
• 2/3/4 SIP Server profiles per system and independent SIP account per port
• Supported voice/Fax codecs include G.711, G.723.1, G.726A/B, iLBC, T.38 Fax
• Advanced carrier grade 128ms-tail-length G.168 line echo cancellation
• Over-voltage and over-current protection(ITU-T, K.21)
• Support VMWI, NEON LED and stutter tone for message waiting indication
• Automated provisioning using TR-069 or AES encrypted XML configuration file; TLS/SRTP/HTTPS for advanced security and privacy protection

With FXO interfaces, it is the ideal solution to deploy private or hosted toll bypass networks. It provides a simple, transparent and cost-effective way of maintaining a connection to the PSTN. It enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. The C7 has the additional benefit of supporting high compression codecs simultaneously on each voice ports, thus saving valuable bandwidth.

The Mediatrix C7 Series gateways make use of existing broadband access equipment to connect to any standards-based VoIP network. It offers security features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling and media transmission aspects.

The Mediatrix C7 series is targeting:

• Enterprises with branch offices and remote offices, including SOHOs.
• Enterprises seeking access to the benefits of Voice over IP without discarding existing legacy equipment.
• Enterprises wishing to implement Voice over IP through a gradual, transparent, and cost-effective migration.
• Multi-tenant / multi-dwelling units and premises wishing to implement Voice over IP while maintaining existing wiring and handsets.
• Enterprises looking to implement secure RTP (SRTP) and Secure SIP (SIP over TLS).