Description
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 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 worldwide.
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.
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, multilanguage prompts, flexible dial plan and more
| HT801 Specifications | |
| Telephone Interfaces | One (1) FXS port |
| Network Interfaces | One (1) 10/100Mbps auto-sensing ethernet port (RJ45) |
| LED Indicators | POWER, INTERNET, PHONE |
| Factory Reset Button | Yes |
| Voice, Fax, Modem | |
| Telephony Features | Caller ID display or block, call waiting, flash, blind or attended transfer, forward, hold, do not disturb, 3-way conference |
| Voice Codecs | G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation |
| Fax Over IP | T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through |
| Short/Long Haul Ring Load | 5 REN: Up to 1km on 24 AWG |
| Caller ID | Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID |
| Disconnect Methods | Busy Tone, Polarity Reversal/Wink, Loop Current |
| Signaling | |
| Network Protocols | TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SSH, STUN, SIP (RFC3261), SIP over TCP/TLS, SRTP, TR-069 |
| QoS | Layer 2 (802.1Q VLAN, SIP/RTP 802.1p) and Layer 3 (ToS, DiffServ, MPLS) |
| DTMF Method | In-audio, RFC2833 and/or SIP INFO |
| Provisioning and Control | HTTP, HTTPS, SSH, TFTP, TR-069, secure and automated provisioning using AES encryption, syslog |
| Security | |
| Media | SRTP |
| Control |
TLS/SIPS/HTTPS |
| Management | Syslog support, SSH, remote management using web browser |
| Physical | |
| Universal Power Supply | Input: 100-240VAC, 50-60Hz Output: 5.0VDC/1.0A |
| Environmental |
Operational: 32º – 104ºF or 0º – 40ºC Storage: 14º – 140ºF or -10º – 60ºC Humidity: 10 – 90% Non-condensing |
| Dimension and Weight | Dimensions: 100mm x 100mm x 29.5mm Weight: 102 g |
| Compliance | FCC: Part15B CE: EN55032, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1 RCM: AS/NZS CISPR22, AS/NZS60950.1, S003 K.21 |