Описание товара
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VoIP GSM Gateway GoIP-4 HyberTone’s GoIP is a VoIP GSM Gateway 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 4 calls simultaneously from IP phones to GSM networks and GSM networks to IP phone.
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Key Features
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4 GSM channels, up to 4 SIM cards
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For call termination (VoIP to GSM) and origination (GSM to VoIP)
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Standard SIP & H.323 protocol, Communicates with other gateway or PC
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Quad band, IMEI changeable, Remote Access
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Support of SMB32 SIM Bank
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Optional SMS termination
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Allows your program send/receive SMS with AT command
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Easy to install and administrate
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Auto Balance and Recharge
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Auto BTS changeable
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Support one stage dialing
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Support free mode-two stage dialing and assigned mode-one stage dialing
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Call Back feature
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All functions can be set on web
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Provide CDR
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Enhanced Features
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LEDs for Power, Ready, Status, WAN, PC, GSM
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Dial in mode or dial out mode only
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Call forward from GSM to VoIP and VoIP to GSM
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Dial Plan
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Password protection for both GSM dial in or dial out
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Retransmit GSM Caller ID to VoIP terminal
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Dynamic selection of codec
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Advanced jitter buffer
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Automatic traversal of NAT and firewall
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VLAN / Qos
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Echo cancellation for Speakerphone
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Comfort noise generation (CNG)
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Voice activity detection (VAD)
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Auto provisioning (requires auto provisioning server)
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On line firmware upgrade
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Multi-language support: English and Chinese
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Supported Standards
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ITU: H.323 V4, H.225, H.235, H.245, H.450
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RFC 1889 - RTP/RTCP
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RFC 2327 –SDP
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RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
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RFC 2976 - SIP INFO Method
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RFC 3261 – SIP
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RFC 3264 - Offer/Answer model with SDP
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RFC 3515 - SIP REFER Method
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RFC 3842 - A Message Summary and Message Waiting Indicator
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RFC 3489 (STUN)- Simple Traversal of UDP Through Network Address Translators (NATs)
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RFC 3891 - SIP “Replaces” Header
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RFC 3892 - SIP Referred-By Mechanism
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draft-ietf-sipping-cc-transfer-04 - Session Initiation Protocol Call Control Transfer
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Codec: G.711 (A/µ law), G.729A/B, G.723.1
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DTMF: RFC 2833, In-band DTMF, SIP INFO
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Web-base Management
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PPP over Ethernet (PPPoE)
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PPP Authentication Protocol (PAP)
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Internet Control Message Protocol (ICMP)
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TFTP Client
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Hyper Text Transfer Protocol (HTTP)
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Dynamic Host Configuration Protocol (DHCP)
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User account authentication using MD5
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Free Software
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SMS Server
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SIM Server ( Sim Bank Scheduler Server )
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Relay Server
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Remote Access
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Technical Specifications
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Protocols: SIP/H.323
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DTMF sending: REC2833
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No. of voice channels: 4
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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 |
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VoIP codec: G.711 PCM A-law/u-law (selectable)
G.729 Annex CS-ACELP at 8 kb/s G.723.1 MP-MLQ/ACELP at 6,3/5,3 kb/s (optional) |
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Voice Quality, VAD, CNG, AEC, LEC, Packet loss
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Network bands: GSM850/900/1800/1900MHz
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Configuration way: WEB interface
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Hardware Specifications
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Processor: ARM9E 133MHz
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DSP: VPDSP101-4 100MHz
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Memory: RAM 16MB/ Flash 4MB
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GSM Module: 850MHz, 900MHz, 1800MHz, 1900MHz
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Power: 12 VDC 2A (110V-220V) (AC/DC adapter included)
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Power consumption: 15W maximum
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Operating temperature: 10°C to 40°C (32°F to 104°F)
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Storage temperature: 0°C to 50°C (32°F to 122°F)
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Size: 195mm (W) x 340mm (L) x 60mm (H)
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