Currency
USD
Need Help? (877) 271-9484 | [email protected]
Top Deals  Deal of the Day  Track Order
Navigation
Cart 0
Product added to cart
Menu
Need Help? (877) 271-9484
[email protected]
Account
Settings
main product photo
Internet Telephony Gateway

Planet VGW-820FS 8-Port SIP VoIP Gateway (8 FXS)

MPN: VGW-820FS
SKU: PLA-VGW-820FS
UPC:
Be the first to review this product
$303.60
Special Price : (Login for Your Price)
Show more Show less
Availability
Usually Ships 10-14 Business Days
Wish List Compare

High Quality yet Affordable for All Businesses

PLANET VGW-820FS enterprise-class 8-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

PLANET VGW-820FS 8-port FXS SIP VoIP Gateway is a fully IETF SIP RFC3261 standard compliant residential gateway that provides a total solution for integrating voice-data network, with built-in SIP trunk and TLS/SRTP security, up to 8 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. The VGW-820FS offers quality voice communications and real-time fax data over IP networks and it does not need human resources to deploy a VoIP network. With the optimized SIP architecture, PLANET VGW-820FS is the ideal choice for P2P/SIP proxy (IP PBX) voice chat, and ITSP cost-saving solution.

 

More Information
MPN: VGW-820FS
ManufacturerPlanet Technology
Show Group Special PriceYes
Hardware
WAN 1 x 10/100BASE-TX RJ45 port
LAN 3 x 10/100BASE-TX RJ45 port
Voice 8 x RJ11 connection (8 x Foreign eXchange Station)
Weight 1000g
Dimensions (W x D x H) 240 x 154 x 37 mm
Power Requirements 100-240V AC, 50-60 Hz@DC12V 2A
Power Consumption 18W
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 Conference
  • Message Waiting Indicator
Software Features
  • Hunting Group
  • Web ACL
  • Telnet ACL
  • Action URL
  • PPPoE/IPv4/IPv6
  • 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 ~ 45 degrees C
Storage Temperature -20 ~ 80 degrees C
Operating Humidity 10%~90% relative humidity, non-condensing
Emission CE, FCC

SIP Applications

  • IETF SIP RFC3261 based on UDP/TCP/TLS
  • 8-line FXS connects to analog phone set or PABX
  • 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 (RFC4733, RFC2833/SIP INFO)
  • Echo cancellation exceeding ITU-T G.168, up to 128ms tail length
  • Supports SIP Trunk and Caller ID: DTMF/FSK CLI Presentation

 

Internet Features

  • Supports SNMP v1/v2/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 

 

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

 

Write Your Own Review
Only registered users can write reviews. Please Sign in or create an account
To Top