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Grandstream Networks THE HT801 COMES WITH 1 EASY-TO-USE FXS PORTS, STATE-OF-THE-ART ENCRYPTION WITH A

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$88.88
$49.38
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SKU: HT801
Availability: In Stock
Weight: 1.00 LBS
Shipping: Calculated at Checkout
Supported network protocols: TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, TELNET, STUN, SIP (RFC3261), SIP TCP/TLS, SRTP, TR-069
Ethernet LAN data rates: 10,100 Mbit/s
Security algorithms: SRTP, SSL/TLS
Foreign eXchange Station (FXS) ports quantity: 1
Ethernet LAN (RJ-45) ports: 1
Web-based management: Yes
Syslog: Yes
Reset button: Yes
Quality of Service (QoS) support: Yes
Output voltage: 5 V
Output current: 0.0416666666666667
Input voltage: 100 - 240 V
Input frequency: 50 - 60 Hz
Dimensions (WxDxH): 3.94 x 3.94 x 1.16 (100 x 100 x 29.5 mm)"
Operating temperature (T-T): 0 - 40 °C
Storage temperature (T-T): -10 - 60 °C
Operating relative humidity (H-H): 10 - 90%
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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. 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. - 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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