GoIP 4 Asterisk GSM Gateway , TFTP / ICMP VoIP Gateway LEDs for Power
Quick detail:
- Brand name: DBL
- Model Number: GoIP 4
- Type: VoIP GSM Gateway
- SIM Cards : 4 sim card
- Size: 25*14*6 cm
- Support: SMS Server , VPN
Product Detail Description:
1,Single or Multiple Server Registrations
2,LEDs for Power, Ready, Status, WAN, PC, FXS
3,Two 10/100 Ethernet for WAN / LAN connections
4,Call forward from GSM to VoIP and VoIP to GSM
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.
Key Features
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Open Standard VoIP Protocols (IETF SIP V2) |
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Single or Multiple Server Registrations |
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Two 10/100 Ethernet for WAN / LAN connections |
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Peer-to-Peer IP Calls |
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Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer |
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Line Echo Cancellation |
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VLAN and QoS support |
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NAT Transversal and Router functions |
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Voice prompts, HTTP Web, Auto Provision support for configuration and updates |
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Highly stable embedded Linux operating system in high performance ARM 9 Processor |
Basic Function
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LEDs for Power, Ready, Status, WAN, PC, FXS |
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Dial in mode or dial out mode only |
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Call forward from GSM to VoIP and VoIP to GSM |
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Dial Plan |
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Retransmit GSM Caller ID to VoIP terminal |
Applications:
1. 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.
2)Call Back
1,Cal
