Planet VGW-420FS

4-Port FXS SIP VoIP Gateway

 

High Quality yet Affordable for All Businesses

PLANET VGW-420FS enterprise-class 4-port SIP VoIP Gateway provides added flexibility during migration to Unified Communications by supporting the traditional analog devices. These devices include analog phones, fax machines, modems, voicemail systems and speakerphones.

Enhanced, Full-Featured Business Gateway

The VGW-420FS 4-port FXS SIP VoIP Gateway, compliant with IETF SIP RFC3261 standard, provides a total solution for integrating voice-data networks, with built-in SIP trunk and TLS/SRTP security, and up to 4 concurrent connections. Voice communications can be established from anywhere around the world, and it not only provides quality voice communications, but also offers secure, reliable Internet sharing capabilities for daily voice and Internet communications.

Distributed VoIP Network Infrastructure

PLANET VGW series is easy to use for all types of businesses as it offers quality voice communications and real-time fax data over IP networks. Moreover, it does not need human resources to deploy a VoIP network. With the optimized SIP architecture, PLANET VGW-420FS is the ideal choice for P2P/SIP proxy (IP PBX) voice chat, and ITSP cost-saving solution.

 

Specifications :

Hardware
WAN	1 x 10/100/1000BASE-T RJ45 port
LAN	1 x 10/100/1000BASE-T RJ45 port
Voice	4 x RJ11 connection (4 x Foreign eXchange Station)
Weight	300g
Dimensions (W x D x H)	194 x 110 x 28 mm
Power Requirements	100-240VAC, 50-60 Hz@DC12V 1A
Power Consumption	5W
Protocols and Standard
FXS	Dial Mode: DTMF and Pulse Pulse: 10 and 20 PPS Caller ID: DTMF/FSK CLI Presentation Max. Cable Length: 3KM Reversed Polarity Programmable Call Progress Tone
Voice & Fax	G.711A/U law, G.723.1, G.729A/B,G.726 and iLBC 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 Modem/POS DTMF mode: Signal/RFC2833/INBAND VLAN 802.1P and 802.1Q Layer3 QoS and DiffServ
VoIP	Protocol: SIP v2.0 (UDP/TCP), RFC3261, SDP, RTP (RFC2833), RFC3262, 3263, 3264, 3265, 3515, 2976, 3311 RTP/RTCP, RFC2198, 1889 RFC4028 Session Timer RFC3266 IPv6 in SDP RFC2806 TEL URI RFC3581 NAT and rport Primary/Backup SIP Server Outbound Proxy DNS SRV/ A Query/NATPR Query SIP Trunk Early Media/Early Answer NAT: STUN, Static/Dynamic NAT
Supplementary Service	Call Waiting Blind Transfer Attend Transfer Call Forward on Busy Call Forward on No Reply Unconditional Call Forward Warm/Immediately Hotline Call Hold Do-not-disturb 3-way Conferencing Message Waiting Indicator
Software Features	Hunting Group Web ACL Telnet ACL Action URL PPPoE/IPv4 Digitmap Bandwidth Optimization Routing Rules based Prefixes Caller/Called Number Manipulation
Management	SNMP v1/v2/v3 TR069 Auto provisioning Web/telnet Configuration backup/restore Firmware upgrade via Web CDR Syslog Ping and tracert test Network capture Outward test (GR909) NTP and daylight saving time IVR local maintenance
Environments
Operating Temperature	0 ~ 40 degrees C
Storage Temperature	-20 ~ 80 degrees C
Operating Humidity	10%~90% relative humidity, non-condensing
Emission	CE, FCC