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Original DBL 1 Port VoIP GSM Gateway

Model: GoIP_1

1 Port VoIP GSM Gateway GoIP-1 for call termination (VoIP to GSM) and origination (GSM to VoIP). It is SIP&H.323 based and compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster. It can enable to make 1/4/8/16 calls simultaneously from IP phones to GSM networks and GSM networks to IP phone.

Connection Graph:

GoIP_1 Photo show:

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 12 VDC 2A (110V-220V) Connect the 12V DC 2A Adapter provided to this power jack. 4 Reset Press this button to reset the GoIP1 GSM VoIP to factory defaults. 2. Features Key Features
  • 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
  • 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
  • DTMF: RFC 2833, In-band DTMF, SIP INFO
  • TCP/IP V4 (IP V6 auto adapt)
  • ITU-T H.323 V4 Standard
  • H.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
4. Software Specifications
  • LINUS 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
5. Hardware Specifications
  • Characteristics of the hardware and Parameters
  • Processor : ARM9E 133MHz
  • DSP :VPDSP101 95MHz
  • RAM: 8M
  • Flash : 4M
  • Power : 12 VDC 2A (110V-220V) (AC/DC adapter included)
  • GSM Module Type: 850MHz, 900MHz, 1800MHz, 1900MHz
  • Consumption: The Maximum 3 W
  • LEDs : RUN, GSM, LAN, PC
  • Network Ports: 2 100/10BASE-T
  • Weight : 105 Grams Without including the weight of DC Adapter
  • Working Temperature: 0-40
  • Working Humidity: 40-90 Not Congealed
  • Colour : Blue
  • GSM SIM Ports: 1
  • VoIP Channels : 1

Application case

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.