Original DBL GSM VoIP Router (32 SIMs)
Model: GoIP32
General Information:
The GoIP32 enables direct routing between IP and GSM network without the use of a FXO port or the PSTN network.It is the GSM network and the VoIP network connecting seamlessly new poducts. Mobile phone SIM cards will be installed in the device, and GoIP32 users can be achieved through the GSM network on the car aligh.It bulit-in SIP and H.323, configuration flexibility. SIP can be thoroughly used when electricity came display numbers. With this GoIP32, the usage of VoIP is greatly enhanced with significant savings on long distance and roaming charges.
1. Key Features
32 GSM channels, up to 32 SIM cards
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP & H.323 protocol, Communicates with other gateway or PC
Quad band, IMEI changeable, Remote Access
Support of SMB32 SIM Bank
Optional SMS termination
Allows your program send/receive SMS with AT command
Easy to install and administrate
Auto Balance and Recharge
Auto BTS changeable
Support one stage dialing
Support free mode-two stage dialing and assigned mode-one stage dialing
Call Back feature
All functions can be set on web
Provide CDR
2. Enhanced Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Dial in mode or dial out mode only
Call forward from GSM to VoIP and VoIP to GSM
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
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
3. 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 (STUN)- Simple Traversal of 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
Web-base Management
PPP over Ethernet (PPPoE)
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP Client
Hyper Text Transfer Protocol (HTTP)
Dynamic Host Configuration Protocol (DHCP)
User account authentication using MD5
4. Appearance
1) LAN
Connect this port to an Ethernet Switch/Router, the Ethernet of a DSL modem, or
other network access equipment.
2) PC
Connect a computer or other network device to this port.
3) POWER :DC12V/4.5A +-10%
4) Reset
Press this button to reset the GoIP32 GSM VoIP Gateway to factory defaults.
5. Protocol
-
TCP/IP V4 (IP V6 auto adapt)
-
ITU-T H.323 V4 Standard
-
.2250 V4 Standard
-
H.245 V7 Standard
-
H.235 StandardMD5,HMAC-SHA1
-
ITU-T G.711 alaw/ulaw, G.729A, G.729AB, and G.723.1 Voice Codec
-
RFC1889 Real Time Data Transmission
-
Proprietary Firewall-Pass-Through Technology
-
SIP V2.0 Standard
- Simple Traversal of UDP over NAT (STUN)
-
Web-base Management
-
PPP over Ethernet (PPPoE)
-
PPP Authentication Protocol (PAP)
-
Internet Control Message Protocol (ICMP)
-
TFTP Client
-
Hyper Text Transfer Protocol (HTTP)
-
Dynamic Host Configuration Protocol (DHCP)
-
Domain Name System (DNS)
-
User account authentication using MD5
- Out-band DTMF Relay: RFC 2833 and SIP Info
6. Software Specifications
-
LINUX OS
-
Built-in HTTP Web Server
-
PPPoE Dial-up
-
NAT Broadband Router Functions
-
DHCP Client
-
DHCP Server
-
Firmware On-line upgrade
-
PSTN Caller ID transmit
-
Multiple Language Support
-
Supported call divert
-
Supported PSTN auto call out to PSTN
- Supported Multi_devices Cooperate Mode(Group Mode)
- Supported SMS call out
7. Hardware Specifications
-
Characteristics of the hardware and Parameters
-
Processor: ARM11 700MHZ
-
DSP:VPDSP101-4 100MHz
-
RAM :128M
-
Flash :8M
-
Power: DC12V/4.5A +-10% Input AC100V to AC240V
-
GSM Module Type: Default 900M/1800M Optional 850M/1900M
-
Consumption: The Maximum 5 W
-
LEDs: RUN, GSM, LAN, PC,GSM
-
Network Ports: 2 RJ45; Supported NAT 100/10BASE-T
-
Weight :900 Grams Full Set
-
Working Temperature: 040
-
Working Humidity : 4090 Not Congealed
-
Colour : Grey
-
GSM SIM Ports: 32
- VoIP Channels : 32
8. Application cases:
A1: 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.
A2: IP PBX Call Origination and Termination
1.Instead of FXO gateways, GoIP are as a call termination and origination device for the IP PBX as shown in the diagram above.
2.VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports.
3.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.
4.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.
1.Instead of FXO gateways, GoIP are as a call termination and origination device for the IP PBX as shown in the diagram above.
2.VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports.
3.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.
4.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.
A3: 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.
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.
A4: 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.
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.
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.
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| Type | VoIP Adapter |
|---|---|
| Model Number | GoIP32 |
| Color | Grey |
| Support SIP | YES |
| Support Relay | YES |
| Support Encryption of VPN | YES |
| IMEI Changeable | YES |
| Support SIP and H.232 | YES |
| Warranty | 1 Year |
| Place of Origin | Guangdong, China (Mainland) |
| is_customized | Yes |
| 32 GSM SIM | Yes, support |
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