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Product Details

4 Ports VoIP GSM Gateway/GoIP

   
Packing:  20PCS/CTN
Model NO.:  GOIP 400
Standard:  252*160*30mm
Productivity:  10, 000PCS/Month
Unit Price:  Negotiable
Shipment Terms:  FOB
Payment Terms:  T/T,Western Union,Paypal
Minimum Order:  1 Pieces
Price Valid Time:  From Oct 25,2011 To Oct 25,2012
Trademark:  Etross
Origin:  China
Usage:  Laptop
Type:  VoIP Products
Style:  Wireless
Back-up Battery:  Available
CE:  Available
Port Number:  4 Ports
Working Frequency:  850/900/1800/1900MHz
Export Markets:  North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe
Product Description

Frequency: GSM 850/900/1800/1900MHz
Description:
GoIP_4 SIM Card Broadband Phone Gateway is a new product that connect the
GSM and the VoIP seamlessly. To GoIP_4 what is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP Softswitch. SIP and H. 323 agreement are built in the GoIP_4 and configured flexible. Caller ID can be seen by using SIP. Flexible routing can meet the need of all kinds of call forwarding; Even more special is that GoIP_4 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group.

Key Features:
Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2) Single or Multiple Server Registrations
Two 10/100 Ethernet circuits connect to the LAN and an additional device GSM module for making GSM calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
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
LEDs for Power, Ready, Status, WAN, PC, GSM Call forward from GSM to VoIP and VoIP to GSM Dial in mode or dial out mode only Dial Plan Password protection for both GSM dial in or dial out Retransmit GSM Caller ID to VoIP terminal Enhanced Features Dynamic selection of codex
Advanced jitter buffer Automatic traversal of NAT and firewall VLAN / Qos Router Echo cancellation for Speaker phone 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

Physical and Environmental

Operating temperature: 10° C to 40° C (50° F to 104° F)
Storage temperature: 0° C to 50° C (32° F to 122° F)
Weight: 500 g (1 lb) (Including AC/DC Adapter)
Size: 160 mm (W) x 252 mm (L) x 30 mm (H)
Power: 12 Vdc 500 mA (AC/DC adapter included)

 
 

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