ATA from China manufacturer - Shenzhen Skyline International Communication Co.,Ltd

ATA

Price:	130.0~150.0 USD
Payment Terms:	T/T,WU;Money Gram,Paypal
Place of Origin:	Guangdong, China (Mainland)


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Product Detail

Model No.: HT-882

Production Capacity: 1000 pcs/month

Delivery Date: 3 days

port: 8

model number: HT-882

Means of Transport: Land,Air

Packing: 31cm x 17cm x 8cm

Brand Name: Skyline

weight: 0.64kg

support: sip&H.323

1.Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
2.Peer-to-Peer IP Calls
3.VLAN and QoS support
4.NAT Transversal and Router functions

8 port ATA gateway,FXS VoIP gateway

The HT-882 VoIP gateway is designed as a compact, high performance, and low cost VoIP Analog Terminal Adaptor (FXS Gateway).It comes with 8 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications.The HT-882 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-882 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

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

Eight 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 2833RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842A Message Summary and Message Waiting Indicator
RFC 3489Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891SIP "Replaces" Header
RFC 3892SIP 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