GSM VoIP Gateway Ports: 4,8,16,32
GSM VoIP Gateway Ports: 4,8,16,32
General Information:
The GoIP GSM 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 products. Mobile phone SIM cards will be installed in the device, and GoIP 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 GoIP, the usage of VoIP is greatly enhanced with significant savings on long distance and roaming charges.
Basic Features :
- For call termination (VoIP to GSM) and origination (GSM to VoIP) }
- Standard SIP & H.323 protocol
- GSM: quad-band 850/900/1800/1900MHz
- GSM SIM Ports: 4 / 8/ 16 / 32
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Connection diagram:
- LAN : Connect this port to an Ethernet Switch/Router, the Ethernet of a DSL modem, or other network access equipment.
- PC : Connect a computer or other network device to this port.
- POWER (DC12V/2000mA) Connect the 12V/2000mA Adapter provided to this power jack.
- Reset : Press this button to reset the GoIP to factory defaults.
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
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
Hardware Specifications :
- Characteristics of the hardware and Parameters
- Processor: ARM9E 133MHz
- DSP:VPDSP101-4 100MHz
- RAM :16M
- Flash :4M
- Power: DC12V/2000mA +-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: 0-40
- Working Humidity : 40-90 Not Congealed
- Colour : Grey
- GSM SIM Ports: 4 / 8/ 16 / 32
- VoIP Channels : 4 / 8 / 16 / 32
Application case:
A1: Sending Bulk SMS Service
- Sending bulk sms text messages is a common technique for telemarketing to reach the target customers.
A bulk SMS system can - be implemented quickly and easily using GoIPs and our proprietary SMS server. Telemarketers are no have full control on how and when they want to send text messages.
- 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.
- 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.
A2: Call back
- 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.
- GoIP could be used to achieve this function alone or as an terminal that is integrated in an existing call back server / platform
- .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.
- 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.
Major Advantages :
- Lightweight and Portable
- IMEI Changeable
- GSM Base Station Optional
- Support SIM Bank/ SIM Sever
- Manual/ Automatic Selection Operators
- Sending and Receiving SMS and USSD (Web Interface)
Free Software/ Sever :
- Remote Control Server: Access Interface Remotely
- Relay Server: Relay Encryption (Make Terminals Traversal the NAT without STUN and Outbound Proxy )
SMS Server:
- Send Bulk SMS
- Provide CDR and ASR
- Auto Balance and Recharge
SIM Server:
- Rotate SIM Cards on Duty
- Set GSM Group (Assign several SIMs Per GSM Port)
- Set Talk Time per SIM, Set Day of week, Set Time Range
- Monitor CDR, ASR, ACD
Key Features :
- Provide 1, 4, 8, 16 , 32 cellular channels for IP-PBX
- Open Standard VoIP Protocols (SIP&H.323)
- Single or Multiple Server Registrations
- Two 10/100 Ethernet for WAN / LAN connections
- Peer-to-Peer IP Calls
- Quad band GSM module: 850MHz, 900 MHz, 1800 MHz, 1900MHz
- 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
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
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 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
- 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