Brand New 16 port GOIP GSM gateway
GoIP VOIP Gateway GSM Converter
SIP IP Phone Adapter
Quad Band 16xGSM SIM Gateway:
Trunk to Asterisk iP PBX
Description:
The GoIP series gateway is a broadband relay gateway 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
Summary:
GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cell phone roaming via the VoIP network.
GoIP GSM Gateway 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 GSM Gateway is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GoIP GSM gateway also provides significant savings in usage (long distance or international), infrastructure and maintenance cost compared to conventional PSTN.
Specification:
1. 16 GSM channels,up to 16 sim cards
2. Quad band ,IMEI changeable
3. Support of SMB32 SIM Bank
4. VoIP SIP,H323,Remote Access
5. Optional SMS termination
6. Easy to install and administrate
7. Auto Balance and Recharge
Benefits of GoIP:
- Extensive product compatibility with industry leading vendor
- Cost-Savings on phone calls between mobiles or to PSTN
- Easy to install – IP device with Web based management interface
- Can be managed and monitored remotely over Internet - a great service to offer to customers by system integrators
- GoIPs can be grouped together to establish GSM gateway cluster
- Termination between GSM/VoIP
- Schedule or on-demand SMS Broadcast messages to users
- (Additional SMS server is required)
Key Features:
- Multiple GoIP grouping mode
- Provide one cellular channels for IP-PBX
- Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
- Single or Multiple Server Registrations
- Two 10/100 Ethernet circuits connect to the LAN and an additional device
- 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
Basic Features:
- 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
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
- Multi-language support: English and Chinese
Supported Standards:
- ITU: H.323 V4, H.225, H.235, H.245, H.450
- RFC 1889 - RTP/RTCP
- RFC 2327 SDP
- RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
- RFC 2976 SIP INFO Method
- RFC 3261 SIP
- RFC 3264 Offer/Answer model with SDP
- RFC 3515 SIP REFER Method
- RFC 3842 A Message Summary and Message Waiting Indicator
- RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
- RFC 3891 SIP Replaces Header
- RFC 3892 SIP Referred-By Mechanism
- draft-ietf-sipping-cc-transfer-04 Session Initiation Protocol Call Control - Transfer
- Codec: G.711 (A/µ law), G.729A/B, G.723.1
- DTMF: RFC 2833, In-band DTMF, SIP INFO
Hardware Specifications:
Hardware |
Parameters |
Remark |
Model |
GoIP-16 |
|
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 |
1.0Kg |
|
Size |
29*14.5*10.5(cm) |
|
Operating Temperature |
0-45℃ |
|
Working Humidity |
40%-90% non-condensation |
|
Color |
grey |
|
SIM port |
16 |
|
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,
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 |
English , Chinese |
|
Multi-regional warning tone |
China,HK,USA,UK,German etc. |
|
Warranty |
One year |
|
Free roaming:
In order to promote our product ,Now we have added some new funcitions in GoIP.By adding the new funcitions,you can build the call without using the softwhich or platform,just depending on the internet.it makes our calls more convenient and easier.what’s more, it can save a lot of call fee.Here is the following example:
Example:
peer to peer:it is a new function that you can use our voip without platform,what is more,it is free roaming for international call.you just pay the local call
MODEL 1 |
MODEL 2 |
MODEL 3 |
MODEL 4 |
Applications:
1. Call Forward
a. Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using VoIP.
b. Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network.
c. 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. IP PBX Call Origination and Termination
a. Instead of FXO gateways, GoIP are as a call termination and origination device for the IP PBX as shown in the diagram above.
b. VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports.
c. 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.
d. 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.
3.Sending Bulk SMS Service
1.Sending bulk sms text messages is a common technique for telemarketing to reach the target customers.
2.A bulk SMS system can be implemented quickly and easily using GoIPs and our proprietary SMS server. Telemarketers are now have full control on how and when they want to send text messages.
3.In addition, SMS text messages are now used widely in many companies, organizations, schools, clubs as a mean for broadcasting information. They can now build their own SMS system without paying expensive charges to their GSM server provider.
4.This system can also take the advantage of using the same GSM service provider to send sms to the phone subscribers in the same service provider.
4. 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.
Package Content:
1 * 16 Channel GSM VOIP Gateway GOIP 16
1 * Power Adapter
1 * Ethernet Cable