Designed for users looking to connect their analog devices to a VoIP network at home or in the office. The HT802 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 2 FXS ports to create a high-quality network solution.

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Description

The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

Features

  • Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port
  • TLS and SRTP security encryption technology to protect calls and accounts
  • Automated provisioning options include TR-069 and XML config files
  • Supports 3-way voice conferencing
  • Failover SIP server automatically switches to secondary server if main server loses connection
  • Supports T.38 Fax for creating Fax-over-IP
  • Supports a wide range of caller ID formats
  • Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
1 line
1 line
Caller ID
Caller ID
Encryption security
Encryption security
Auto provisioning
Auto provisioning
3 way conference
3 way conference
Failover
Failover
Fax support
Fax support

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Designed for users looking to connect their analog devices to a VoIP network at home or in the office. The HT802 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 2 FXS ports to create a high-quality network solution.

Features :

  • Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port
  • TLS and SRTP security encryption technology to protect calls and accounts
  • Automated provisioning options include TR-069 and XML config files
  • Supports 3-way voice conferencing
  • Failover SIP server automatically switches to secondary server if main server loses connection
  • Supports T.38 Fax for creating Fax-over-IP
  • Supports a wide range of caller ID formats
  • Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning

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