Fixed Wireless Terminal 16 Port Voip Sip Gateway Ram 32m Imei Change
Fixed Wireless Terminal 16 Port Voip Sip Gateway Ram 32m Imei Change
Fixed Wireless Terminal 16 Port Voip Sip Gateway Ram 32m Imei Change
Fixed Wireless Terminal 16 Port Voip Sip Gateway Ram 32m Imei Change

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Fixed Wireless Terminal 16 Port Voip Sip Gateway Ram 32m Imei Change

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Product description

Product Name: VoIP SIP Gateway Type: VoIP GSM Gateway
Protocol: SIP & H.323 Support: VPN, Relay Encryption, VoS, Asterisk
Applications: Call Forward / Call Back Processor: ARM9 133MHZ

Fixed Wireless Terminal 16 Port VoIP SIP Gateway RAM 32M IMEI Change

 

 

Quick details

 

  • Brand name: DBL
  • Model Number:GoIP-16
  • type: VoIP GSM Gateway
  • Protocol: SIP & H.323
  • Support:VPN, Relay Encryption, VoS, Asterisk

 

 

Competitive Advantage:

 

1.Support SIP and H.323 protocols

2.Sending bulk sms message

3.Quad band

4.IMEI Change

5.Relay Server

6.PPTP VPN .VOS

 

 

Description:

 

GSM SIP Gateway 16 Port GoIP VoIP Gateway GoIP-16 is a 16 SIM Card Broadband Phone Gateway that had been developed by DBL Co. GoIP-16 SIM Card Broadband Phone Gateway is a new product that connect the GSM and the VoIP seamlessly. To GoIP-16 what is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP Softswitch.SIP and H.323 agreement are built in the GoIP-16 and configured flexible. Caller ID can be seen by using SIP. Flexible routing can meet the need of all kinds of call forwarding; even more special is that GoIP-16 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group.

 

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 callee’s phone number to the GoIP. The GoIP then sends both the caller’s and callee’s 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 caller’s 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 16

 

Processor

ARM9 133MHZ

 

DSP

VP-101-1 196MHz

 

RAM

32M

 

FLASH

4M

 

Network Card

100/10BASE-T ×2

 

LED

operating and circuit light

 

Power consumption

12VDC

 

Net Weight

1.6kg

 

Operating Temperature

0-45℃

 

Working Humidity

40%-90% non-condensation

 

Color

grey

 

RJ11 port

16

 

 

 

 

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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