Welcome to
Shenzhen Skyline Technology Co., Ltd

Free Member
Shenzhen Skyline Technology Co.,  Ltd
Shenzhen Skyline Technology Co., Ltd
China
  • Sell

    GOIP 4( sip and H.323) ,  4 channel voip gsm gateway/ goip voip gateway call termination for PBX,  Asterisk

    GOIP 4( sip and H.323) , 4 channel voip gsm gateway/ goip voip gateway call termination for PBX, Asterisk

    Price:
    $473
    Quantity Order:
    Origin:
    China
    Payment Method:
    Telegraphic Transfer (T/T)
    Pack. & Delivery:
    10 pieces / carton 43*38*34cm/carton
    Overview

    GOIP_4 is a 4 SIM Card Broadband Phone Gateway that had been developed by Skyline Co. 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 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


You have [3] new inquiries.
Go to Member Menu

Home - Trade List - Product List - Inquiry List - Cooperation List - Company List
Copyright © 2024 Indomonster.com. All Rights Reserved.