Product Details

2 ports Fxs gateway support H.323/SIP for

low calling cost

The two FXS Ports VoIP Gateway (HT-922) is designed as a compact, high performance, and low cost Analog Terminal Adaptor (FXS Gateway). It comes with 2 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications. The HT-922 is a full featured FXS gateway and is designed for easy installation and configuration. It supports the two most widely used Open VoIP Standards (SIP and H.323). This allows the HT-922 to interoperate seamlessly with soft switches or IP PBXs made by various vendors. Its high performance offers toll quality voice, flexible networking, and feature-rich call functions. It is an ideal low cost solution for SME environment where multiple lines are required.

Key Features

Static IP support
Switch mold support
support DHCP,PPPOE
Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
Two 10/100 Ethernet for WAN / LAN connections
Peer-to-Peer IP Calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Line Echo Cancellation
VLAN and QoS support
NAT Transversal and Router functions
Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features

Two RJ-11 FXS port for traditional phone set or PBX's trunk line
LEDs for Power, Ready, Status, WAN, PC, FXS
Call Forward, Call Hold, Call Transfer
Dial Plan
Caller ID
3 times re-password
Billing support
Keyboard setting
IP voice enrollment

Enhanced Features

Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese

Supported Standards

ITU: H.323 V4, H.225, H.235, H.245, H.450
RFC 1889 RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP “Replaces” Header
RFC 3892 SIP Referred-By Mechanism
draft-ietf-sipping-cc-transfer-04 Session Initiation Protocol Call Control - Transfer
Codec: G.711 (A/µ law), G.729A/B, G.723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO