Gateway VoIP 4 fxs + 4 fxo Dinstar DAG1000 4S4O
Gateway VoIP con 4 porte fxs + 4 porte fxo Dinstar, 1 porta Wan, 3 porte LAN, plug RJ11 standard
DAG1000-4S4O è un gateway analogico per connettere terminali e linee analogiche da e per sistemi telefonici IP "on premise" e in cloud. E’ dotato di connettori plug RJ11 standard.
Dispone inoltre di 3 porte LAN ed 1 porta WAN.
Un gateway ideale ad esempio per trasportare linee analogiche verso un centralino full IP o su un centralino virtuale in cloud e riportare in azienda 4 terminali analogici su cui collegare telefoni tradizionali, interfacce citofoniche analogiche, cordless, ripetitori di chiamata, fax etc...
Interoperabile con i più disffusi centralini VoIP, virtuali e on premise come ad esempio Yeastar, Freepbx, Freeswitch, Asterisk, Broadsoft, 3CX, Panasonic, Unify, Avaya, Wildix, che utilizzano lo standard SIP.
Caratteristiche principali del gateway VoIP 4fxs + 4 fxo
- 4 FXS + FXO hybrid
- Flexible port group
- IP trunk
- Flexible routing and manipulation
- Data/ Voice/ Management VLAN
- Voice and FAX all in one
- Power and network failure life line
- Firmware update automatically /configure update
- TLS
Physical Interfaces
- Phone Interfaces: 4FX and 4FXO, RJ-11
- Ethernet Interfaces:1WAN 3LAN 10/100Mbps
Voice & FAX
- G.711A/U law, G.723.1, G.729 A/B
- Silence Suppression
- Comfort Noise Generation(CNG)
- Voice Activity Detection(VAD)
- Echo Cancellation(G.168), with up to 128ms
- Adaptive (Dynamic) Jitter Buffer
- Hook Flash
- Programmable Gain Control
- T.38/Pass-through
- High speed fax up to 14.4kbps
- Modem/POS
- DTMF mode: Signal/RFC2833/INBAND
- VLAN 802.1P/802.1Q (data/voice/management VLANs)
- Layer3 QoS and DiffServ
FXO
- Connector: RJ11
- Dial Mode: DTMF/Pulse Dialing
- Caller ID: FSK, DTMF
- Polarity Reversal
- Answer Delay
- Busy Tone Detection
- No Current Detection
FXS
- Connector: RJ11
- Dial Mode: DTMF and Pulse
- Pulse: 10 and 20 PPS
- Caller ID: DTMF/FSK CLI Presentation
- Max Cable Length: 3 km
- Reversed Polarity
- Programmable Call Progress Tone
VoIP
- Protocol:SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262,3263,3264,3265,3515,2976,3311
- SIP TLS/SRTP
- RTP/RTCP, RFC2198, 1889
- RFC4028 Session Timer
- RFC3266 IPv6 in SDP
- RFC2806 TEL URI
- RFC3581 NAT,rport
- Outbound Proxy
- DNS SRV/ A Query/NATPR Query
- SIP Trunk
- Early Media/Early Answer
- NAT:STUN, Static/Dynamic NAT
Software Features
- Port Group
- Web ACL
- Telnet ACL
- Action URL
- Digitmap
- Routing Rules based Prefixes
- Caller/Called Number
Maintenance
- SNMP v1/v2/v3
- TR069
- Auto Provisioning
- Web/Telnet
- Configuration Backup/Restore
- Firmware Upgrade via Web
- CDR
- Syslog
- Network Capture
- Outward Test(GR909)
- NTP/Daylight Saving Time
- IVR local Maintenance
- Cloud-based Management
Network
- Static/Dynamic IP,
- PPPoE
- DHCP Client
- IPv4/IPv6
- TCP, UDP,TFTP, FTP, ARP,RARP,
- Ping, NTP, SNTP, HTTP/HTTPS, DNS
- Ping / Tracert
- DHCP Option 66,120,121
- Software Features
- Port Group
- Web ACL
- Telnet ACL
- Action URL
- Digitmap
- Routing Rules based Prefixes
- Caller/Called Number
- Manipulation
Environmental
- Power supply:Input 100~240VAC Output 12VDC, 2A
- Power consumption(W): 20W
- Operator Temperature: 0 ℃ ~ 45 ℃
- Storage Temperature: -20 ℃ ~80 ℃
- Humidity: 10%-90% no condensation
- Compliance: CE, FCC, RoHs
Garanzia 12 mesi
