GrandStream HT801 VoIP SIP ATA Telephone Line Adapter - 1x FXS
GrandStream HT801 Analog to VoIP Telephone Adapter
An easy-to-use single (1) port ATA for converting a regular telephone/cordless phone to VoIP line service.
The HT801 is a single port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones.
It also allows service providers to offer high quality IP service to their market. The HT801 is an ideal ATA for individual use as well as commercial IP voice deployments.
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 which can be easily deployed and managed. It comes equipped with 1 FXS ports to create a high-quality VoIP / SIP network solution for single telephone line user needs.
GrandStream HT801 Specifications
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.
GrandStream HT801 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, multi-language prompts, flexible dial plan and more