User's Manual
Table Of Contents
- DOCUMENT PURPOSE
- CHANGE LOG
- GUI INTERFACE EXAMPLES
- WELCOME
- PRODUCT OVERVIEW
- GETTING STARTED
- CONFIGURATION GUIDE
- Obtain HT801 IP Address via Connected Analogue Phone
- Understanding HT801 Interactive Voice Prompt Response Menu
- Configuration via Web Browser
- Web Configuration Pages Definitions
- NAT Settings
- DTMF Methods
- Preferred Vocoder (Codec)
- Configuring HT801 Through Voice Prompts
- Register a SIP Account
- Call Features
- Rebooting HT801 from Remote
- UPGRADING AND PROVISIONING
- RESTORE FACTORY DEFAULT SETTINGS
- EXPERIENCING HT801

P a g e | 10
HT801 Administration Guide
PRODUCT OVERVIEW
The HT801 is a 1 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 and enables service providers to offer high quality IP service. The
HT801 is an ideal ATA for individual use and for large scale commercial IP voice deployments.
Feature Highlights
The following table contains the major features of the HT801:
Table 1: HT801 Features at a Glance
HT801 Technical Specifications
The following table resumes all the technical specifications including the protocols / standards supported,
voice codecs, telephony features, languages and Upgrade/ Provisioning settings for the HT801.
Table 2: HT801 Technical Specifications
Interfaces
Telephone Interfaces
One (1) RJ11 FXS port
Network Interface
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,
HT801
1 SIP profile through 1 FXS port and single 10/100Mbps port
3-way voice conferencing
Wide range of caller ID formats
Advanced telephony features, including call transfer, call forward, call-
waiting, do not disturb, message waiting indication, multi-language
prompts, flexible dial plan and more
T.38 Fax for creating Fax-over-IP and GR-909 Line Testing Functionalities
TLS and SRTP security encryption technology to protect calls and
accounts
Automated provisioning options include TR-069 and XML config files
Failover SIP server automatically switches to secondary server if main
server loses connection
Use with Grandstream’s UCM series of IP PBXs for Zero Configuration
provisioning
GR-909 Line Testing Functionalities










