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GOIP 32 gateway gsm
Avaliação dos clientes: ( 736 comentários )

Preço: R$ 7.900,00


  • Descriçao do Produto
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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 
● 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





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



 Hardware  Parameters  Remark

 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  12V/DC  
 Weight  0.45kg  
 Size  25*14*6(cm)  
 Operating Temperature  0-45℃  
 Working Humidity  40%-90% non-condensation  
 Color  grey  
 SIM port  4  
 RJ45 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  Support  
 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, 
 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  


1. User Manuals
     GoIP Series User Manual V1.4

2. Inbound Call Centre for Customer Service  
1.Inbound call centre is operated by a company to administer incoming product support or information inquiries from consumers.
2.This application requires customers to call in via a single phone number. Each GSM channel in a GoIP has its own phone number. In order to achieve calling in via a single number, GoIPs must be configured in a phone group mode with one GoIP as a host and the others as clients. 
3.An incoming call to the host GoIP is automatically forwarded to an idle GSM channel in the phone group. The call is then routed to a call attendant via a VoIP connection.
4.Key advantages: quick setup, scalable, portable, low system cost, low operating cost especially used in cellular network no charge on incoming calls. 

3. Outbound Call Centre for Telemarketing / Solicitation  
1.Outbound call centers are normally operated for telemarketing, solicitation of charitable or political donations, and debt collection etc.
2.In this application, only outgoing calls are made. GoIPs could be configured as SIP clients to registering to the SIP server / IP PBX or as a SIP Trunk to accept calls routed from the SIP Server / IP PBX.
3.Key advantages: quick setup, portable, scalable, low system cost, low operating cost. 

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