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PPPOE Asterisk GSM Gateway ARM 9 Processor For Termination Box

Quick detail:


# Model Number: GoIP 1I
# type: VoIP GSM Gateway
# SIM Cards : 1 sim card
# Protocol: SIP & H.323
# Support: SMS Server , VPN







Description:



1,NAT Transversal and Router functions

2,Call forward from GSM to VoIP and VoIP to GSM

3,Two 10/100 Ethernet for WAN / LAN connections

4,VLAN and QoS support



Asterisk GSM Gateway GoIP1I an IP-based device that enables inbound and outbound VoIP and GSM cellular calls. It is a great alternative to PSTN, especially in area where GSM service is available for VoIP call termination.



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



- - - -
Open Standard VoIP Protocols (IETF SIP V2) |
Single or Multiple Server Registrations |
Two 10/100 Ethernet for WAN / LAN connections |
Peer-to-Peer IP Calls |
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer |
Line Echo Cancellation |
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 Function



- - - -
LEDs for Power, Ready, Status, WAN, PC, FXS |
Dial in mode or dial out mode only |
Call forward from GSM to VoIP and VoIP to GSM |
Dial Plan |
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,Call Back is referring to the telecommunications event that occurs when the originator of a call is immediately called back in a second call as a response.

2,GoIP could be used to achieve this function alone or as an terminal that is integrated in an existing call back server / platform.

3,For standalone operation, GoIP receives a call with caller ID information and then rejects the call immediately without answering the call. GoIP then calls back the caller so that he can dial a phone number to make a call. In this case, GoIP must register to a VoIP Service Provider who can offer terminate the call.

4,In a call back system, GoIP acts as a device to initiate the call back function. Typically, this is done in two ways. The first method is to send an SMS with the callees phone number to the GoIP. The GoIP then sends both the callers and callees phone numbers to the call back server to complete the call back function. The second method is to call the GoIP and the hang up (with the call being answered). GoIP sends the callers phone number to the call back server and the call back server calls the caller directly so that the caller can then dial a phone number to make a call.





- - - -
Hardware | Parameters | Remark |
Model | GoIP 1 | |
Processor | ARM9 133MHZ | |
DSP | VP-101-1 196MHz | |
RAM | 16M | |
FLASH | 4M | |
Network Card | 100/10BASE-T 2 | |
LED | operating and circuit light | |
Power consumption | 12VDC | |
Net Weight | 0.1kg | |
Operating Temperature | 045 | |
Working Humidity | 4090 non-condensation | |
Color | grey | |
RJ11 port | 1 | |
| | |
Speech Characteristics | Service Condition | Remark |
G.168 Echo cancellation | support | 16mS |
g.723 | support | |
g.711A/u | support | |
g.729A/B | support | |
GSM | Nonsupport | |
PLC | support | |
CNG | support | |
VAD | support | |
Jitter Buffer | support | |
T.38 | Nonsupport | |
|
Network Characteristic | Parameters | Remark |
LAN Port |
DHCP | support | |
PPPoE | support | |
Static IP | support | |
NAT Transversal | Relay or Port forwarding | Relay need coordinate with DBL Relay Server |
Network time | NTP / SNTP | |
10/100BASE-T | 10/100 auto adaptation | |
PC Port |
Static IP | support | |
DHCP | Max support 200 terminals | |
10/100BASE-T | 10/100 auto adaptation | |
Switch mode | support | |
| | |
Protocols | ITU H.323 V4 and IETF SIP V2, SIP (RFC3261) TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP,
ICMP, DNS, DHCP, NTP, TFTP, TELNET, PPPoE | support |
| | |
User Interface | Parameters | Remark |
Web page Configuration | HTTP | |
Multiple password | 3 password | customized service |
Keyboard setup | support | |
Online update | http | |
Broadcast IP | support | Chinese , English |
Billing | support | coordinate with DBL Billing Software |
Language | Chinese, English | |
Multi-regional warning tone | China,HK,USA,UK,German etc. | |
Warranty | one year | |
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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
VoIP GSM Gateway , GoIP GSM Gateway , Asterisk GSM Gateway
Contact Information:
Country:
China
Address:
Room 5C,Aozhihao Integrated Building, Xinzhou 4th Street, Futian District, Shenzhen, Guangdong ,China
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