The HT801 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 HT801 comes with 1 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.
Designed for users looking to connect their analog devices to a VoIP network, in either a home or office. The HT801 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 1 FXS ports to create a high-quality network solution.
Description
Features
- Supports 1 SIP profile through a single FXS port 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
- Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more
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Designed for users looking to connect their analog devices to a VoIP network, in either a home or office. The HT801 is a powerful analog telephone adapter that is easily deployable and manageable. It comes equipped with 1 FXS ports to create a high-quality network solution.
Features
- Supports 1 SIP profile through a single FXS port 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
- Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more
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