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1-Sim Card,Goip 1,Asterisk Linux Gsm Voip Gate-way Box GoIP-1
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.
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
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
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
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
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
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
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.
IP PBX Call Origination and Termination
1.Instead of FXO gateways, GoIP are as a call termination and origination device for the IP PBX as shown in the diagram above.
2.VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports.
3.Outside callers can then call in via the GoIP GSM ports to reach any of the VoIP endpoints that are registered to the IP PBX.
4.GoIP can be configured in a group mode such that all GSM ports can be used by just dialing only one GSM number. Please refer to the Call Center Application for more information.
1). A GoIP gateway
2). 12V/2A DC(For GoIP_4) or 12V/3A DC (For GoIP_8) transformer
3). An Ethernet cable (2 m)