China GoIP 8 8 Channel VoIP GSM Gateway 8 SIM Card Wireless VoIP Terminal Support SMS Server

GoIP 8 8 Channel VoIP GSM Gateway/ 8 SIM Card Wireless VoIP Terminal Support SMS Server
GoIP GSM Gateway Specification:
1. 8GSM channels
2. Bulk SMS/USSD
3. Good ACD,ASR
4. Auto Balance and Recharge
5. VoIP SIP&H323
6. Remote Access
7. Optional SMS termination
8. Support IMEI changeable
9.Quad band, IMEI changeable
Overview

A VoIP GSM Gateway (GoIP) bridges voice communications between the GSM and the IP networks seamlessly.In principle, it consists of a GSM module and a VoIP module. With a SIM card installed, the GSM module registers to the GSM network for voice and SMS services.

GoIP comes with various models of 1, 4, 8, 16 and 32GSM channels. When integrating in a VoIP system, GoIP is highly scalable in meeting customer's requirement on the number of channels (lines). In addition with its power functions, high voice quality, and low price, GoIP is an inevitable choice of many system integrators, call termination operatiors, small companies, and individuals.

Key Functions

portant;line-height:18px;color:rgb(0,0,0);margin:1em 0px;font-family:Arial, Helvetica, sans-serif;">GSM Carriers and base Stations OptionalSending SMS / USSD via built-in webpage or SMS ServerRemote SIMAuto balance check and recharge (prepaid SIM)Call time restrictionsRemote Control ManagementRelay Server support for NAT and encryption of IP packetsPPTP VPN and Relay EncryptionBuilt-in SIP Proxy for peer-to-peer operationGSM Quad Band: 850 / 900/1800/1900 MHzCall forward from GSM to VoIP and VoIP to GSMCall Back

Key Advantages:

portant;line-height:18px;color:rgb(0,0,0);margin:1em 0px;font-family:Arial, Helvetica, sans-serif;">Lightweight and PortableEasy to install and administrateIMEI ChangeableGSM base Station OptionalSupport SIM Bank/ SIM SeverManual/ Automatic Selection OperatorsSending and Receiving SMS and USSD (Web Interface)Flexible Dial plan fo number alternation, call screen.Flexible configuration modes.All functions can be set on web.Provide CDR report

Free Software/Server

Remote Control Server----Access Interface Remotely.

Relay Server----Relay Encryption( Make Terminals Traversal the NAT without STUN and Outbound Proxy)

SMS Server----Remote SIM cards; Send Bulk SMS; Provide CDR and ASR; Auto Balance and Recharge.

SIM Server----Remote SIM Cards on Duty; Set GSM Group( Assign several SIMs per GSM Port); Set talk time per SIM, Set Day of week, time range; Monitor CDR, ASR, ACD.

Autocfg Server---- Maintence all DBL termial device anywhere; embeded remote control server; Remote sim auto configuration.

Protocols: SIP/H.323

TCP/IP:IP/TCP/UDP/RTP/RTCP/,CMP/ARP/RARP/SNTP, DHCP/DNSClient,IEEE802.1P/Q.ToS/DiffServ. NAT Traversal, STUN, UPNP.

IP Assignment, Static IP, DHCP, PPPoE

Package info

Key FeaturesOpen Standard VoIP Protocols (SIP& H. 323)Single or Multiple Server RegistrationsTwo 10/100 Ethernet for WAN / LAN connectionsPeer-to-Peer IP CallsSpeech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter bufferLine Echo CancellationVLAN and QoS supportNAT Transversal and Router functionsVoice prompts, HTTP Web, Auto Provision support for configuration and updatesHighly stable embedded Linux operating system in high performance ARM 9 Processor


Basic FeaturesLEDs for Power, Ready, Status, WAN, PC, FXSDial in mode or dial out mode onlyCall forward from GSM to VoIP and VoIP to GSMDial PlanRetransmit GSM Caller ID to VoIP terminal





Enhanced FeaturesDynamic selection of codecAdvanced jitter bufferAutomatic traversal of NAT and firewallVLAN / QosRouterEcho cancellation for SpeakerphoneComfort noise generation (CNG)Voice activity detection (VAD)Auto provisioning (requires auto provisioning server)On line firmware upgradeMulti-language support: English and Chinese








Supported StandardsITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450RFC 1889 - RTP/RTCPRFC 2327 -SDPRFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony SignalsRFC 2976 - SIP INFO MethodRFC 3261 - SIPRFC 3264 - Offer/Answer model with SDPRFC 3515 - SIP REFER MethodRFC 3842 - A Message Summary and Message Waiting IndicatorRFC 3489 - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)RFC 3891 - SIP " Replaces" HeaderRFC 3892 - SIP Referred-By Mechanismdraft-ietf-sipping-cc-transfer-04 - Session Initiation Protocol Call ControlTransferCodec: G. 711 (A/µ law), G. 729A/B, G. 723.1DTMF: RFC 2833, In-band DTMF, SIP INFOOperating temperature: 10° C to 40° C (50° F to 104° F)

Physical and Environmental
Storage temperature: 0° C to 50° C (32° F to 122° F)Power: 12 VDC 4.5A (110V-220V) (AC/DC adapter included)Warranty: one year

Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal