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Planet 32-Port SIP VoIP Gateway (32 FXS) VGW-3220FS

Planet 32-Port SIP VoIP Gateway (32 FXS) VGW-3220FS
Planet 32-Port SIP VoIP Gateway (32 FXS) VGW-3220FS
  • Stock: 15-20 Ημέρες
  • Model: VGW-3220FS
  • EAN: 4711605285575

High Quality yet Affordable for All Businesses

PLANET VGW-3220FS enterprise-class 32-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-3220FS 32-port FXS SIP VoIP Gateway is a fully IETF SIP RFC 3261 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 32 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-3220FS 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-3220FS is the ideal choice for P2P/SIP proxy (IP PBX) voice chat, and ITSP cost-saving solution.

 


The VGW-3220FS provides the essential features you need for business-class voice communications in an easy-to-manage solution. Designed for businesses with branch offices, it helps the enterprises to save money on long-distance calls.


SIP Applications

  • IETF SIP RFC3261 based on UDP/TCP/TLS
  • 32-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 Layer3 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), 3-way conferencing