Yeastar NeoGate TG1600 VoIP to GSM Gateway
  • Yeastar NeoGate TG1600 VoIP to GSM Gateway
  • Yeastar NeoGate TG1600 VoIP to GSM Gateway

Yeastar NeoGate TG1600 VoIP to GSM Gateway

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  • - Brend: Yeastar
  • - Kod Produkta: Yeastar NeoGate TG1600 VoIP to GSM Gateway
  • - Dostupnost: Na Stanju
238,190.00 RSD

Qty

Yeastar NeoGate TG1600 GSM Gatway

REDUKCIJA TROŠKOVA !
Yeastar TG1600 je VoIP GSM/CDMA prenosnik sa 16 kanala i povezivost sa GSM/CDMA mrežom za "softswitches", i IP-PBX uređaje.

SPECIFIKACIJA
    Broj GSM/CDMA kanala (Max): 16
    Tip mreže: 850/900/1800/1900MHz
    CDMA Frekvencija: 800MHz
    Antena spliter (4 u 1): podržano
    Protokol: SIP, IAX2
    Prenosni Protokol: UDP, TCP, TLS, SRTP
    Codec: G.711 (alaw/ulaw), G.722, G.726, G.729A, GSM, ADPCM, Speex
    Echo Cancellation: ITU-T G.168 LEC
    DTMF: RFC2833, SIP INFO, In-band
    Tip poziva: Terminacija (VoIP to GSM/CDMA), Orginacija (GSM/CDMA to VoIP)
    LAN: 1 (10/100Mbps)
    Konzol port: 1
    Network protokol: FTP, TFTP, HTTP, SSH
    NAT Traversal: Static NAT, STUN
    Network: DHCP, DDNS, Firewall, OpenVPN, Static IP, QoS, Static Route, VLAN
    Dimenzije: 440 x 280 x 44
    Napajanje: AC 100-240V
    Radna temperatura: 0° do 40°C, (32° do 104° F)
    Temperatura skladištenja: -20° do 65°C, (4° do 149° F)
    Vlažnost: 10-90% bez kondenzacije

KORISTI
    Bogatstvo funkcija: Veliki set funkcija koje mogu da ispune vaše različite potrebe i smanje troškove
    Jednostavno upravljanje: Jednostavno i intuitivno Web bazirano konfigurisanje štedi vaše vreme
    Pouzdanost: Stabilnost sa naprednom hardware-skom i software-skom arhitekturom
    Odlučna interoperabilnost: Interoperablnost sa širokom listom "softswitch" PBX i IP-PBX (Elastix, Lync Server i dr.)

KARAKTERISTIKE
1 Stage/2 Stage Dial  
Balance Alarm
Call Back
Call Detail Record (CDR)
Call Duration Limitation
Call Progress Tone Generation
Call Status Display
Call Transfer
Call Waiting
Caller ID/CLIR
Carrier Selection: Auto/Manual
Configure backup/restore
Firmware upgrade by HTTP/TFTP
Gain Adjustment
GSM/CDMA/UMTS Ports Group Manage
Hotline
Incoming /Outgoing Routing rules
IP Blacklist
Network Attack Alert
NTP
Open API for SMS and USSD
Packet Capture
PIN Modify
Real Open API Protocol (Based on Asterisk)
Send Bulk SMS
Session Timer
SIP Peer Mode: Support
SIP Response Code Switch
SIP server for IP phones: Support
SIP Trunk: Support
SMS Center  
SMS Sending and Receiving
System Logs
USSD
VoIP Trunk Group  Web based configuration
White List and Black List


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