This Product was added to our catalogue on Monday, 20 January 2014.
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238.36 $
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Описание товара
VoIP GSM Gateway GoIP-1   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 1 call simultaneously from IP phones to GSM networks and GSM networks to IP phone.
Key Features
1 GSM channels
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
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 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
Free Software
SMS Server
SIM Server ( Sim Bank Scheduler Server )
Relay Server
Remote Access
Technical Specifications
Protocols: SIP/H.323
DTMF sending: REC2833
No. of voice channels: 4
            DHCP/DNSClient,IEEE802.1P/ Q.ToS / DiffServ,
            NAT Traversal, STUN, uPnP,
            IP Assignment, Static IP, DHCP, PPPoE
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)
Voice Quality, VAD, CNG, AEC, LEC, Packet loss
Network bands: GSM850/900/1800/1900MHz
Configuration way: WEB interface
Hardware Specifications
Processor: ARM9E 133MHz
DSP: VPDSP101-4 100MHz
Memory: RAM 16MB/ Flash 4MB
GSM Module: 850MHz, 900MHz, 1800MHz, 1900MHz
Power: 12 VDC 2A (110V-220V) (AC/DC adapter included)
Power consumption: 15W maximum
Operating temperature: 10°C to 40°C (32°F to 104°F)
Storage temperature: 0°C to 50°C (32°F to 122°F)
Size: 195mm (W) x 340mm (L) x 60mm (H)
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