The GSM Fixed Wireless Terminal  16 ports Gateway GoIP-16 is a 16 SIM Card  Broadband Phone Gateway

 that had been developed by CHINASKYLINE Ltd. GoIP_16SIM Card Broadband Phone Gateway is a new

product that connect the GSM and the VOIP seamlessly. To GoIP_16 what is installed on the Mobile

SIM Card, you can register the GSM telephone on the VoIP soft switch. SIP and H.323 agreement are built in

 the GoIP_16 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_16 support multi-device group, it can be easily

combined into arbitrary number of channels of Large Gateway Group

When the mobile phone SIM card is installed in the GoIP, users can register the GSM phone to the VoIP softswitch system.

Through the GoIP, users can realize the uplink and downlink calls between the GSM network and the VoIP network.

In addition, the GoIP supports the transparent transmission of the caller number from the PSTN to the VoIP. The GoIP features

 embedded SIP and H.323 protocols with flexible setting. The bi-directional password authentication (call authorization) and

trust list authentication greatly minimize the risk of charge losses and the flexible routing function can meet special requirements

 of various call forwarding. In particular, the GoIP gateway supports multi device groups, with flexible setting of large GSM gateway

 groups with different channel numbers. With its low price, excellent voice quality, and powerful features, the GoIP series gateway

 is the first choice for system integrators, traffic operators, and softswitch manufacturers

The GoIP series gateway includes:

GoIP-GSM, GoIP_4 and GoIP_8. GoIP _16.

Key Features

--Multiple GoIP16 grouping mode
--Provide 16 cellular channels for IP-PBX
--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

Hardware Specifications
--Processor: ARM9E 133MHz
--DSP: VPDSP101 196MHz
--Memory: RAM 16MB/ Flash 4MB
--GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz
--Power: Input AC100V ~ 240V, output DC12V/2A +-10%
--Power consumption: 32W maximum
--Network card: 100/10Base-T x2
--LED: Operation and lines light
--GSM Passway:16

--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: 1203 g (1 lb) (Including AC/DC Adapter)
--Warranty: one year

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

Price and support

we are real manufacturer and can give you factory price. we can give you professional free support services with the help of our technology support.

 NOTES:

>We provide strong tech support about VOIP server and A-Z termination. (6 days*12 hours)

>We can help you rent a VoIP server in USA.

>We can provide copy version VOS.

>We can install Relay server, SMS server,SMB server for you.

(for free)

>We can offer voip terminal guide.

>We may need your termina routes.

>We offer full tech support untill your gateway runs.