GoIP VOIP Gateway GSM Converter SIP IP Phone Adapter IP PBX GOIP-1 SMS gateway IMEI change voip telephones

Description:

The GoIP series gateway is a broadband relay gateway newly developed by DBL Technology. It is a new product for seamless connection between the GSM network and VoIP network. 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.

GoIP is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks. GoIP is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GSM gateway also provides significant savings in usage, infrastructure and maintenance cost compared to conventional PSTN.

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.

Basic Function

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

Key Features:

Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)

Single or Multiple Server Registrations

Two 10/100 Ethernet for WAN / LAN connections

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

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

Competitive Advantage:

Remote SIM operation for SIM Card management

SMPP support for 3rd party development of SMS Applications.

Free server utilities for remote access and SMS management.

Telnet mode for sending AT commands to GSM module.

Compact and light weight design

Direct customer support

Call Forward

1.Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using VoIP.

2.Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network.

3.As shown in the network topology diagram, a VoIP Service Provider is using GoIPs as call origination and termination devices.

- A call dialed to a GoIP (right hand side) via GSM is first routed via VoIP and then terminated via a VoIP end point or VoIP Service Provider.

- A VoIP call originated from the left hand side is routed to a GoIP on the right hand side and then is dialed out as a GSM call.

1). A GoIP gateway GOIP-1

2). 12V/2A DC(For GoIP_4) or 12V/3A DC (For GoIP_8) transformer

3). An Ethernet cable (2 m)