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Grandstream HT812 Analog Telephone Adapter 2 FXS Port

Grandstream HT812 Analog Telephone Adapter 2 FXS Port

Features

  • Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
  • Includes a built-in NAT router which can handle routing speeds up to 100MBps
  • 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

KSh5,300.00

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Availability: In Stock

Grandstream HT812 Analog Telephone Adapter 2 FXS Port

Grandstream HT812 Analog Telephone Adapter 2 FXS Port

The HT812 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 using 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 HT812 comes with 2 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

Designed for users looking to connect their analog devices to a VoIP network at home or in the office. The HT812 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 2 FXS ports and an integrated Gigabit NAT router.

Features

  • Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
  • Includes a built-in NAT router which can handle routing speeds up to 100MBps
  • 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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