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SIP VoIP Gateway

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SIP VoIP Gateway
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Cost-effective, High-performance VoIP Communication
To build high-performance VoIP communications at a low cost, PLANET now introduces the latest member of its gateway family, the VGW-804 enterprise-class 8-port SIP VoIP Gateway. The VGW-804 provides added flexibility during migration to Unified Communications by supporting the traditional analog devices, which include analog phones, fax machines, modems, voicemail systems, and speakerphones. It helps the enterprises to save money on long-distance calls; 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. The VGW-804 also allows call to be transferred to anyone at any location within the voice system, which enables the enterprises to communicate more effectively and is helpful to streamline business processes.

SIP Standard Compliance
The VGW-804 supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VGW-804 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-804 is a full-featured enhanced business SIP Gateway that addresses the communication needs of the enterprises. It provides 4-line FXO plus 4-line FXS interface which allows connection with 4 analog PSTN telephone lines and with 4-line analog telephone set to make or receive VoIP call over Internet or VPN network. This device is suitable for office PABX to enable to have VoIP call without changing cabling, dial plan and extension number.

The VGW-804 supports all kinds of SIP-based gateway features and multiple contact filter functions, such as 24 SIP trunk accounts, both IPv6 and IPv4 protocols, flexible dial plan and route plan features, and switch of analog and VoIP signal to help both protocols to communicate efficiently.

Secure, High-Quality VoIP Communication 
The VGW-804 can effortlessly deliver secure 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. 

Supporting Caller ID
Both the FXS and FXO ports of the VGW-804 support caller ID function, helping users identify calling number and verify number easily. 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).




Artikelnummer 8658-0559-01
Administrator PLANET
Lieferzeit 28 Tage




  • Supports SIP 2.0 (RFC 3261)
  • Supports IPv6 and IPv4 simultaneously
  • Up to 24 SIP service domains and Caller ID
  • Supports auto HTTP provision and fax feature
  • Flexible routes plan, dial plan and SIP trunk
  • Life-line for emergency calls

Internet Features

  • IPv4 (RFC 791) and IPv6
  • IPv6 auto configuration (RFC 4862)
  • MAC clone setting
  • Vendor Class ID
  • DDNS (Planet DDNS, Easy DDNS, DynDNS)
  • DNS client
  • Firewall
  • URL / IP / MAC / Port Filter
  • Port forwarding (TCP, UDP or both)
  • Bandwidth control (download and upload), maximum bandwidth priority setting

SIP Applications

  • SIP Session Timer (RFC 8048)
  • SIP Session Refresher: UAC or UAS
  • SIP Encryption
  • Supports Outbound Proxy / STUN NAT Traversal
  • Supports Primary and Backup SIP Server

Call Features

  • Supports peer to peer dialing
  • 4-line FXO connects to PSTN line
  • 4-line FXS connects to analog phone set or PABX
  • Caller ID recognition DTMF (before/after 1st ring) and FSK (before 1st ring), ETSI and Bellcore
  • DTMF Caller ID start and stop bit configurable
  • T.38 fax volume configuration

FXO/FXS Line Configuration

  • Line ID / Line phone number
  • Polarity reversal detection or generation for call establishment and billing
  • VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing
  • Outgoing SIP Caller ID selection

Routing Plan

  • Prefix match and length
  • Priority / Cyclic / Simultaneous ring
  • Programmable hunting cycle

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