2FXS /voip adapter with router/voip China/voip call

FXS:1 support ITU H323V4 and IETF SIPV2
2 ensure high quality speech
3 support VLAN and QoS
4 NAT Transversal
5 Router
2 Channel FXS sip gateway free roaming
Product Introduction:
The

two channels FXS VoIP gateway 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 softswitches 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
LINUX OS
Built-in HTTP Web Server
PPPOE Dial-up
NAT Broadband Router Functions
DHCP Client
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
One 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

Enhanced Features
DHCP Server
Firmware On-line upgrade
PSTN Caller ID transmit
Multiple Language Support
Supported call divert
Supported PSTN auto call out to PSTN
Supported Multi-devices Cooperate
Mode (Group Mode)
Supported SMS call out
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

Hardware Specifications
Processor: ARM9E 133MHz
DSP: VPDSP101 95MHz
Memory: RAM 16MB/ Flash 4MB
Power: Input AC100V ~ 240V, output 24VDC / 300mA
Power consumption: 4W maximum
Network card: 100/10Base-T x2
LED: Operation and lines light
RJ11:one 24V feed 48V ring
Operating temperature: 10°C to 40°C (32°F to 104°F)
Storage temperature: 0°C to 50°C (32°F to 122°F)
Working Humidity: 40% ~ 90% Not congealed
Weight: 95 g (1 lb) (Including AC/DC Adapter)
Warranty: one year
FREE ROAMING
Example:
peer to peer: it is a new function that you can use our VoIP without VoIP provider; what is more, it is free roaming for international call. You just need pay the local call.
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