Easy Integration. With , you avoid the hassle and expense of replacing your existing routers, WAN connections or phone system required by other VOIP solutions. simply plugs into your Ethernet network. Neither your phone service or network is placed at risk. Minimum requirements: Ethernet network, WAN connection, IP addresses.

Save Thousands of Dollars Each Month. can save your company substantial amounts in long distance charges. Even if your company uses one of the most inexpensive calling plans, a network can quickly return your investment and begin paying you back.

Award-winning Voice Quality. With , you'll experience consistent toll-quality voice connections. Using the Perceptual Speech Quality Measurement (PSQM), Internet Telephony magazine found that 's MultiVOIP gateway delivered exceptional voice quality. In fact, it outranked the competition.

Interoperability. utilizes the H.323 and SIP protocols to provide complete interoperability with other Internet telephony solutions. The inbound IP call protocol is automatically detected and the voice channel is dynamically configured to match. The outbound IP call protocol is configured with the phone number allowing you the flexibility to call H.323 or SIP devices from the same port. In addition, also supports T.38 real-time fax relay for interoperability among other VOIP equipment.

PSTN Fail-over. PSTN fail-over allows to automatically route calls over the PSTN network when the IP network is congested or completely down. This feature heightens reliability and augments QoS when conditions threaten to undermine voice quality. Utilizing user definable controls, continually checks if the LAN/WAN is threatened by packet loss or latency, or to see if the network is completely down. If it detects a problem, switches to “survivability mode” transparently routing all calls over PSTN lines connected to the gateway. continues to monitor the connection and automatically switches back to the LAN/WAN once the conditions improve.

Advanced Speech Technologies. supports the Differentiated Services (DiffServ) Quality of Service (QoS) protocol which sets priorities for voice and fax traffic and allows transparent delivery. DiffServ helps move time-sensitive voice traffic across even low-bandwidth WAN connections, like 56K and ISDN, with the priority and quality required by voice. Other features such as adaptive echo cancellation, forward error correction, bad frame interpolation, tunable latency and dynamic jitter buffers, further enhance voice quality.

Complete Support for Multiple Telephony Interfaces. For maximum investment protection, the 's two, four and eight-port models accommodate changing communication needs by providing a programmable FXS/FXO and an E&M interface for each port. This allows to connect directly to a phone, fax machine, key phone system or PBX. It automatically detects whether the incoming call is a voice or fax call. The single port supports FXS and FXO interfaces, while the digital connects directly to a PBX or PSTN line via T1/E1 or PRI.

Bandwidth Management. Bandwidth is used only when someone is speaking. The silence suppression/Voice Activity Detection (VAD) feature is an option that frees unused call bandwidth for data traffic. This is significant, since callers are usually silent for 60 percent of the call. When using silence suppression, also offers Comfort Noise Generation (CNG) at the receiving end so the user knows the line has not dropped. In addition, supports voice compression standards like G.729 (8:1) and G.723 (10:1). These standards help minimize the bandwidth required for voice. G.723, for instance, is the maximum compression rate and requires only 5.3K bps (plus an added 7-8K bps for IP overhead). Even at maximum compression, your VOIP solution will still provide toll-quality voice.

Management. is easily managed locally using a windows-based software application or remotely by the central office with a web browser or SNMP. also includes its own SNMP management software which provides central site configuration, management and call monitoring for all gateways on the network. It utilizes a Windows interface that makes it easy to view events like usage tracking, live use reporting, call history, and voice quality statistics. In addition, it eases administration by automatically e-mailing call logs based on volume or time.

No User Training. provides single stage dialing by utilizing a Uniform Dialing Plan that is consistent with the E.164 (PSTN) standard numbering plan. This includes automatic appending and stripping of digits to dialed numbers to ensure that users will not require additional training to make VOIP calls. In fact, placing calls with is like using your existing phone system.

Supplementary Services. supports H.450 supplementary services to provide for call transfer, call forwarding, call hold, call waiting, and name identification. It also supports Q.SIG, an inter-PBX signaling protocol, for networking PBX supplementary services in a multi- or uni-vendor environment. In addition, supports SIP extensions providing call forward and call transfer capabilities.

Avaya™ Small Office Media Gateway Solution. can also provide an affordable small office media gateway solution that delivers the features of Avaya™'s Communication Manager software to the branch offices of large corporations. The gateway, with integrated gatekeeper, cost-effectively extends the call features and networking benefits of a centralized Avaya™ Media Server to small branch offices, utilizing traditional analog devices, over an IP infrastructure. also renders local office survivability, in the case of a LAN or WAN failure, by providing local, reliable PSTN trunking.