Planet VGW-420FO 4-Port SIP VoIP Gateway (4*FXO): IETF SIP 2.0, T.38 FAX, Mul
Planet VGW-420FO 4-Port SIP VoIP Gateway (4*FXO): IETF SIP 2.0, T.38 FAX, Mul
Planet VGW-420FO 4-Port SIP VoIP Gateway (4*FXO): IETF SIP 2.0, T.38 FAX, Mul

Planet

Planet VGW-420FO 4-Port SIP VoIP Gateway (4*FXO): IETF SIP 2.0, T.38 FAX, Mul

Sale price$272.00
SKU: VGW-420FO

Usually Ships in 10 - 12 days

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High-performance VoIP Gateway

To build high-performance VoIP communications at a low cost, PLANET has launched the latest VoIP gateway model, the VGW-420FO enterprise-class 4-port SIP VoIP Gateway. The VGW-420FO gateway provides added flexibility during migration to Unified Communications by supporting the traditional analog devices. For example, the remote workers can dial in through a Unified VoIP Communication System just like an extension call but no long-distance call charge would occur. It also allows call to be transferred to anyone at any location within the voice system, which enables the enterprise to communicate more effectively and is helpful to streamline business processes.

 

VoIP Protocol Standard Compliance

The VGW-420FO supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VGW-420FO is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better multi-media exchange services.

Enhanced, Full-Featured Business Gateway

The VGW-420FO is a full-featured enhanced business SIP Gateway that addresses the communication needs of the enterprises. It provides the 4-line FXO gateway with SIP protocol IP device which allows connection with 4 analog PSTN telephone lines set to make or receive VoIP call over Internet or VPN network. This VGW-420FO is suitable for office PABX to enable to have VoIP call without changing cabling, dial plan and extension number, The VGW-420FO supports all kinds of SIP-based gateway feature and multiple contact filter functions, such as 4 SIP trunk accounts, both IPv6 and IPv4 protocols, flexible dial plan and route plan features, and switch analog and VoIP signal, to help both protocols to communicate.   

 

Secure, High-quality VoIP Communication 

The VGW-420FO effortlessly delivers secured toll voice quality by utilizing cutting-edge 802.1p QoS (Quality of Service), 802.1Q VLAN tagging, and IP TOS (Type of Service) technology. Using voice and data, VLAN can easily separate the data and voice, thus maintaining the best quality. 

Caller ID Function

Both the FXO and FXS ports of the VGW-420 series (VGW-420FO/VGW-420FS) support caller ID function, help user identify calling number easily and verify number. It also helps to block anonymous call by filtering strange calls. The FXS port transmits Caller ID, while the FXO port receives Caller ID. The Caller ID interoperates with analog phones, public switched telephone networks (PSTN) and private branch exchanges (PBXs).

Highlights

  • Supports SIP 2.0 (RFC3261) 
  • Supports IPv6 and IPv4 simultaneously
  • Up to 32 SIP service domains
  • Caller ID Support
  • Supports auto HTTP provision and fax feature
  • Flexible Routes Plan, Dial Plan and SIP Trunk

Internet Features

  • Supports SNMP v1/v2c/v3 
  • Supports VLAN 802.1P and 802.1Q
  • Supports Layer 3 QoS and DiffServ
  • Supports STUN (RFC 3489) and Outbound Proxy
  • Supports TR069 and Auto Provisioning 
  • Supports TLS/SRTP Security

SIP Applications

  • IETF SIP RFC3261 based on UDP/TCP/TLS
  • 4-line FXO connects to PSTN line
  • Fax over T.38 and Pass-through
  • ITU-T G.711 A-law, G.711 μ-law, G.723.1 and G.729 voice coding
  • In-band/out of band DTMF (RFC2833/SIP INFO)
  • Echo cancellation exceeding ITU-T G.168, up to 128ms tail length
  • SIP Trunk support
  • Caller ID: DTMF/FSK support

Call Features

  • Call waiting/transfer (blind transfer, attend transfer)
  • Call hold/quick pick
  • Call forwarding unconditional
  • Call forwarding on no reply
  • Hotline/speed dial/direct IP call
  • Do Not Disturb (DND)/three-way conferencing

FXO Line Configuration

  • Line ID/line phone number
  • Reverse polarity detection or generation for call establish and billing
  • VoIP dial to FXO/PSTN line
  • Outgoing SIP caller ID selection

Routing Plan

  • Prefix match and length
  • Priority/Cycle/Simultaneous Ring

 

 

Hardware
WAN 1 x 10/100BASE-TX RJ45 port
LAN 2 x 10/100BASE-TX RJ45 port
Voice 4 x RJ11 connection (4 x FXO)
Weight 853g
Dimensions (W x D x H) 242 x 152 x 35 mm
Power Requirements 100-240VAC, 50-60 Hz@DC12V 2A
Power Consumption 12W
Protocols and Standard
FXO Dial Mode: DTMF/Pulse Dialing
Caller ID: FSK, DTMF
Reverse Polarity
Answer Delay
Busy Tone Detection
No Current Detection
Voice & Fax G.711A/U law, G.723.1, G.729A/B, G.726
Silence Suppression
Comfort Noise Generation (CNG)
Voice Activity Detection (VAD)
Echo Cancellation(G.168), with up to 128ms
Adaptive (Dynamic) Jitter Buffer
Programmable Gain Control
T.38/Pass-through
Modem/POS
DTMF mode: Signal/RFC2833/INBAND
VLAN 802.1P/802.1Q
Layer3 QoS and DiffServ
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
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
OpenVPN
Software Features Port Group
Web ACL
Telnet ACL
Action URL
Digitmap
Routing Rules based Prefixes
Caller/Called Number Manipulation
Speed Dial
Management SNMP v1/v2c/v3
TR069, TR181
Auto Provisioning
Web/Telnet
Configuration Backup/Restore
Firmware Upgrade via Web
CDR
Syslog
Network Capture
NTP/Daylight Saving Time
IVR Local Maintenance
Environments
Operating Temperature 0 ~ 50 degrees C
Storage Temperature -20 ~ 80 degrees C
Operating Humidity 10%~90% relative humidity, non-condensing
Emission CE, FCC