Specifications

8 Port fox FXS Gateway usb fxo fxs

Change Imei Phone / Cheap Voip Phones/FXS Gateway (HT-882)

Overview
The HT-882 is designed as a compact, high performance, and low cost VoIP Analog Terminal Adapter (FXS Gateway). It comes with eight FXS port to interface with a traditional analog phone set or a PBX trunk line for VoIP communications. It bundles with lots of features to meet the demand in various network environment. It is an ideal low cost VoIP solution for travelers and SOHO users.

Key Features
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
Support T.38
Line Echo Cancellation
PPTP VPN ,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

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

Product picture

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