Grandstream HT801, HT802 Administration Guide

Grandstream Networks, Inc.
HT801/HT802 Analog Telephone Adaptors
Administration Guide
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HT801/HT802 Administration Guide
Version 1.0.25.5
COPYRIGHT
©2021 Grandstream Networks, Inc. http://www.grandstream.com
All rights reserved. Information in this document is subject to change without notice. Reproduction or
transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without
the express written permission of Grandstream Networks, Inc. is not permitted. The latest electronic version of this user manual is available for download here:
http://www.grandstream.com/support
Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc.
in the United States, Europe and other countries.
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with your devices as it may cause damage to the products and void the manufacturer warranty.
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HT801/HT802 Administration Guide
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GNU GPL INFORMATION
HT801/HT802 firmware contains third-party software licensed under the GNU General Public License
(GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General
Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download
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HT801/HT802 Administration Guide
Version 1.0.25.5
Table of Content
DOCUMENT PURPOSE ................................................................................................. 7
CHANGE LOG ................................................................................................................ 8
Firmware version 1.0.25.5 ..................................................................................................................... 8
Firmware version 1.0.23.5 ..................................................................................................................... 8
Firmware version 1.0.21.4 ..................................................................................................................... 8
Firmware version 1.0.19.11.................................................................................................................... 8
Firmware Version 1.0.17.5 ..................................................................................................................... 9
Firmware Version 1.0.15.4 ..................................................................................................................... 9
Firmware Version 1.0.13.7 ..................................................................................................................... 9
Firmware Version 1.0.11.6 ................................................................................................................... 10
Firmware Version 1.0.10.6 ................................................................................................................... 10
Firmware Version 1.0.9.3 ..................................................................................................................... 10
Firmware Version 1.0.8.7 ..................................................................................................................... 10
Firmware Version 1.0.5.11 ................................................................................................................... 11
Firmware Version 1.0.3.7 ..................................................................................................................... 11
Firmware Version 1.0.3.2 ..................................................................................................................... 11
Firmware Version 1.0.2.5 ..................................................................................................................... 12
Firmware Version 1.0.2.3 ..................................................................................................................... 12
Firmware Version 1.0.1.9 ..................................................................................................................... 12
GUI INTERFACE EXAMPLES ...................................................................................... 13
WELCOME ................................................................................................................... 14
PRODUCT OVERVIEW ................................................................................................ 15
Feature Highlights ................................................................................................................................ 15
HT801/HT802 Technical Specifications ............................................................................................... 16
GETTING STARTED ..................................................................................................... 18
Equipment Packaging .......................................................................................................................... 18
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HT801/HT802 Ports Description .......................................................................................................... 19
Connecting HT801/HT802 ................................................................................................................... 20
HT801/HT802 LEDs Pattern ................................................................................................................ 20
CONFIGURATION GUIDE ............................................................................................ 22
Obtain HT801/HT802 IP Address via Connected Analogue Phone..................................................... 22
Understanding HT801/HT802 Interactive Voice Prompt Response Menu .......................................... 22
Configuration via Web Browser ........................................................................................................... 24
Accessing the Web UI .................................................................................................................. 24
Web UI Access Level Management .............................................................................................. 25
Saving the Configuration Changes ............................................................................................... 25
Changing Admin Level Password ................................................................................................. 25
Changing User Level Password ................................................................................................... 25
Changing Viewer Password.......................................................................................................... 26
Changing HTTP Web Port ............................................................................................................ 26
Web Configuration Pages Definitions .................................................................................................. 27
Advanced Settings Page Definitions ............................................................................................ 27
Status Page Definitions ................................................................................................................ 38
Basic Settings Page Definitions .................................................................................................... 40
FXS Ports Pages Definitions ........................................................................................................ 44
NAT Settings ........................................................................................................................................ 63
DTMF Methods .................................................................................................................................... 63
Preferred Vocoder (Codec) .................................................................................................................. 63
Configuring HT801/HT802 Through Voice Prompts ............................................................................ 63
Register a SIP Account ........................................................................................................................ 64
Call Features ........................................................................................................................................ 66
Rebooting HT801/HT802 from Remote ............................................................................................... 68
UPGRADING AND PROVISIONING ............................................................................ 69
Firmware Upgrade procedure .............................................................................................................. 69
Upgrading via Local Directory .............................................................................................................. 70
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Upgrading via Local TFTP/HTTP/HTTPS/FTP/FTPS Servers ............................................................ 70
Firmware and Configuration File Prefix and Postfix ............................................................................ 71
Managing Firmware and Configuration File Download ........................................................................ 71
Configuration File Download ................................................................................................................ 71
RESTORE FACTORY DEFAULT SETTINGS ............................................................... 73
Using the Reset Button ........................................................................................................................ 73
Using the IVR Command ..................................................................................................................... 73
Reset from Web Interface (Reset Type) .............................................................................................. 74
Reset using SIP NOTIFY ..................................................................................................................... 74
EXPERIENCING HT801/HT802 .................................................................................... 75
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Table of Tables
Table 1: HT801/HT802 Features at a Glance ............................................................................................. 15
Table 2: HT801/HT802 Technical Specifications ......................................................................................... 16
Table 3: Definition of the HT801/HT802 Connectors .................................................................................. 19
Table 4: HT801/HT802 LEDs Pattern Description ...................................................................................... 21
Table 5: Voice Prompt Menu ....................................................................................................................... 22
Table 6: Web access level management .................................................................................................... 25
Table 7: Advanced Settings ......................................................................................................................... 27
Table 8: Status Page Definitions ................................................................................................................. 38
Table 9: Basic Settings Page ...................................................................................................................... 40
Table 10: FXS Ports .................................................................................................................................... 44
Table 11: HT801/HT802 Call Features ........................................................................................................ 66
Table 12: MAC Address Key Mapping ........................................................................................................ 73
Table of Figures
Figure 1: HT801 Package Contents ............................................................................................................ 18
Figure 2: HT802 Package Contents ............................................................................................................ 18
Figure 3: HT801 Back Panel ....................................................................................................................... 19
Figure 4: HT802 Back Panel ....................................................................................................................... 19
Figure 5: Connecting the HT801/HT802 ..................................................................................................... 20
Figure 6: HT801/HT802 LEDs Pattern ........................................................................................................ 20
Figure 7: Admin Level Password ................................................................................................................ 25
Figure 8: User Level Password ................................................................................................................... 26
Figure 9: Viewer Level Password ................................................................................................................ 26
Figure 10: Web HTTP Port .......................................................................................................................... 27
Figure 11: SIP Accounts Settings ................................................................................................................ 65
Figure 12: Account Status ........................................................................................................................... 66
Figure 13: Firmware Upgrade Page ............................................................................................................ 70
Figure 14: XML Config File - MAC Header ................................................................................................. 72
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HT801/HT802 Administration Guide
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DOCUMENT PURPOSE

This document describes the basic concept and tasks necessary to use and configure your
HT801/HT802. In addition, it covers the topic of connecting and configuring the HT801/HT802, making
basic operations and the call features. Please visit http://www.grandstream.com/support to download the
latest “HT801/HT802 User Guide”.
This guide covers following topics:
Product overview Getting started Configuration guide Upgrade and provisioning Restore factory default settings
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HT801/HT802 Administration Guide
Version 1.0.25.5

CHANGE LOG

This section documents significant changes from previous versions of administration guide for
HT801/HT802. Only major new features or major document updates are listed here. Minor updates for
corrections or editing are not documented here.

Firmware version 1.0.25.5

Added support for “OpenVPN”. [OpenVPN] Added support of “Maximum Number of SIP Request Retries”. [Maximum Number of SIP Request
Retries]
Added support for “Failback Timer”. [Failback Timer]

Firmware version 1.0.23.5

Added support for “DNS SRV Failover Mode”. [DNS SRV Failover Mode] Added support of “Register Before DNS SRV Failover”. [Register Before DNS SRV Failover] Added support for “Offhook Auto-Dial DTMF”. [Off Hook Auto-Dial DTMF] Added Special Feature IZZI to support N-Way conference hosted on Nokia IMS. [Special Feature]

Firmware version 1.0.21.4

Added support for IPv6 address without square brackets. [Primary SIP Server] [Failover SIP Server]
[Outbound Proxy] [Backup Outbound Proxy]
Added support for DHCP Domain Name configuration. [DHCP domain name] Added support of “Use Configured IP” for “DNS Mode”. [Use Configured IP] Added support for “Play Busy Tone When Account is unregistered”. [Play Busy Tone When Account is
unregistered]

Firmware version 1.0.19.11

Added “Disable" option for Web access mode. [Web Access Mode] Moved “Trusted CA certificates” from Profile1/Profie2 to Advanced Settings and renamed as Trusted
CA Certificates A and Trusted CA Certificates B. [Trusted CA certificates A] [Trusted CA certificates B] Added Feature “Disable User Level Web Access” and “Disable Viewer Level Web Access”. [Disable
User Level Web Access][Disable Viewer Level Web Access]
Added Feature “Use P-Asserted-Identity Header”. [ Use P-Asserted-Identity Header] Added Feature” Load CA Certificates”. [Load CA Certificates] Added Feature “Connection Request Port”. [Connection Request Port] Added OI_BR to special feature. [Special Feature] Added support to set Ring Timeout to 0 for unlimited ring timeout. [Ring Timeout] Added New Zealand Standard for Pulse Dialing Standard. [Pulse Dialing Standard]
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HT801/HT802 Administration Guide
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Added support for Russian Language. [Language] Increased “SIP TLS Certificate” and “SIP TLS Private Key” supported maximum length from 2048 to
4096.[SIP TLS Certificate][SIP TLS Private Key]

Firmware Version 1.0.17.5

Added support for Minimum TLS Version. [Minimum TLS Version] Added support for Maximum TLS Version. [Maximum TLS Version] Added support for Traditional Chinese language.[Language]

Firmware Version 1.0.15.4

Added more choices to feature “Disable Weak TLS Ciphers”. [Disable Weak TLS Cipher Suites] Added feature “Syslog Protocol”. [Syslog Protocol] Added support for “Distinctive Call Waiting Tone”. [Distinctive Call Waiting Tone] Added support for “Call Waiting Tones”. [Call Waiting Tones] Added support for DHCP option 67. [Configuration File Download] Added support for GDMS platform. [ACS URL]

Firmware Version 1.0.13.7

Added ability to support provisioning server path containing the server authentication credentials for
the DHCP option 66. [Allow DHCP Option 66 to Override Server]
Added support to send SNMP trap to 3 different servers. [SNMP Trap IP Address] Added feature “Call Features Settings”. [Call Features Settings] Added feature “Use SIP User Agent”. [SIP User-Agent] Updated “Use SIP User Agent Header” to “SIP User Agent Postfix”. [SIP User Agent Postfix]. Added feature “Disable Reminder Ring for DND”. [Disable Reminder Ring for DND] Added feature “CDR File Option”. [CDR File Option] Added feature “SIP File Option”. [SIP File Option] Added feature “Disable Weak TLS Cipher Suites”. [Disable Weak TLS Cipher Suites] Added feature “Pulse Dialing Standard”. [Pulse Dialing Standard] Added feature “Callee Flash to 3WC”. [Callee Flash to 3WC] Added feature “RFC2833 Count”. [RFC2833 Events Count] [RFC2833 End Events Count] Added feature “Replace Beginning ‘+’ with 00 in Caller ID”. [Replace Beginning '+' with 00 in Caller ID] Added feature “Disable Unknown Caller ID”. [Disable Unknown Caller ID] Added feature “Disable # as Redial Key”. [Disable # as Redial Key] Added feature “Reset Call Features”. [Reset Call Features] Added support to view, download, and delete the call history through device Web UI. [CDR File] Added support to store SIP file locally. [SIP File]
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HT801/HT802 Administration Guide
Version 1.0.25.5

Firmware Version 1.0.11.6

Added support to display CPU load on the device web STATUS page. [CPU Load] Added support for SIP keep-alive to use SIP NOTIFY. [Enable SIP OPTIONS/NOTIFY Keep Alive] Added feature Network Cable Status on Web UI status page. [Network Cable Status] Added support for Management Interface. [Enable Management Interface] Added feature SSH Idle Timeout. [SSH Idle Timeout] Added feature Telnet Idle Timeout. [Telnet Idle Timeout] Added feature Use ARP to detect network connectivity. [Use ARP to detect network connectivity] Added feature Call Record. [CDR File]

Firmware Version 1.0.10.6

Added feature “Inband DTMF Duration”. [Inband DTMF Duration] Added feature “RFC2543 Hold”. [RFC2543 Hold] Added feature “Visual MWI Type”. [Visual MWI Type] Added feature “Ring Frequency”. [Ring Frequency] Added feature “Allow SIP Factory Reset”. [Allow SIP Factory Reset] [Reset using SIP NOTIFY] Added support for G722 Codec. [HT801/HT802 Technical Specifications] Added support for allow user to choose preference codec from PCMU and PCMA for FAX pass-through
codec. [Fax Mode]

Firmware Version 1.0.9.3

Added feature “Custom Certificate”. [Custom Certificate] Added feature “Use P-Access-Network-Info Header”. [Use P-Access-Network-Info Header] Added feature “Use P-Emergency-Info Header”. [Use P-Emergency-Info Header] Added feature “Conference Party Hang up Tone” when “Special Feature” is set to MTS. [Call Progress
Tones]
Add support for HTTPS based on TLS v1.2

Firmware Version 1.0.8.7

Added [CenturyLink, MTS] to Special Feature. Added support for upgrade device via [FTP/FTPS] server. [UPGRADING AND PROVISIONING] Added support to have the call waiting tone through SIP INFO. Added feature “Validate Server Certificates”. [Validate Server Certificates] Added support for [DDNS] Added feature Blacklist for Incoming Calls. [Blacklist for Incoming Calls] Added support for [Telnet] Added feature [Play busy/reorder tone before Loop Current Disconnect]
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Firmware Version 1.0.5.11

Changed default "Upgrade Via" from HTTP to HTTPS. [Upgrade via] [UPGRADE PROTOCOL] Added the ability to schedule [Automatic Reboot] Added support for [SNMP] Added support for 3 level access through RADIUS authorization (Admin, User and [viewer]) Added option to customize number of failed [Web Access Attempt Limit] to access web GUI Added option to customize idle time to logout the web access with [Web Session Timeout] Added option to disable WAN side ports [Black List for WAN Side Port] Added feature “Caller ID Fetch Order” option under FXS port settings. [Caller ID Fetch Order] Added feature “Enable High Ring Power” option under FXS port settings. [Enable High Ring Power] Added feature “Internet Protocol” to choose from “IPv4 Only”, “IPv6 Only”, “Both, prefer IPv4”, “Both,
prefer IPv6”. [Internet Protocol]
Added feature “IPv6 Address” to configure IPv6 Address. [IPv6 Address]

Firmware Version 1.0.3.7

Added option “Use Actual Ephemeral Port in Contact with TCP/TLS” to force device to use actual
ephemeral port. [Use Actual Ephemeral Port in Contact with TCP/TLS]
Added option “SIP URI Scheme When Using TLS” to choose between ‘SIP’ and ‘SIPS’. [SIP URI
Scheme When Using TLS]
Added Option “Backup Outbound Proxy” to use backup Outbound Proxy if Outbound Proxy registration
expires. [Backup Outbound Proxy]
Added option “Prefer Primary Outbound Proxy” to enable registration through primary outbound proxy
if registration expires. [Prefer Primary Outbound Proxy]
Added option “Enable RTCP” to enable RTCP function through Web UI. [Enable RTCP] Added option “Hold Target Before Refer” to enable device to hold before being referred. [Hold Target
Before Refer]
Added Option “Enable Session Timer” to disable session timer. [Enable Session Timer] Added feature “Conference URI” to support Conference URI. [Conference URI] Added feature “White List for WAN Side” for remote management. [White List for WAN Side] Added feature “Black List for WAN Side” for remote management. [Black List for WAN Side] Added option “Web Access Mode” to choose between “HTTPS” and “HTTP” to access device Web UI. Added feature “HTTPS Web Port” to set HTTPS web port instead of using default HTTPS port. [HTTPS
Web Port]
Added feature “SSH Port” to self-configure SSH port. [SSH Port] Added SNMP related features.

Firmware Version 1.0.3.2

This is the initial version for HT801. Added option “DNS SRV use Registered IP” [DNS SRV use Registered IP].
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HT801/HT802 Administration Guide
Version 1.0.25.5
Changed default NTP server from us.pool.ntp.org to pool.ntp.org.

Firmware Version 1.0.2.5

Changed OPUS Payload Type default value to 123 to match other GS products. [OPUS Payload Type].

Firmware Version 1.0.2.3

Added a re-enter box to confirm change user and admin password on web GUI to avoid typo or mistakes
[Confirm Admin Password][Confirm End User Password].
Add new iLBC libraries to improve iLBC audio quality.

Firmware Version 1.0.1.9

This is the initial version.
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HT801/HT802 Administration Guide
Version 1.0.25.5

GUI INTERFACE EXAMPLES

http://www.grandstream.com/sites/default/files/Resources/ht80x_web_gui.zip
1. Screenshot of Login Page.
2. Screenshots of Status Page.
3. Screenshots of Basic Settings Page.
4. Screenshots of Advanced Settings Page.
5. Screenshots of FXS Port 1 Page.
6. Screenshots of FXS Port 2 Page.
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HT801/HT802 Administration Guide
Version 1.0.25.5

WELCOME

The HT801/HT802 analog telephone adaptors provide transparent connectivity for analog phones and
faxes to the world of internet voice. Connecting to any analog phone, fax or PBX, the HT801/HT802 are an
effective and flexible solution for accessing internet-based telephone services and corporate intranet
systems across established LAN and internet connections. The Grandstream handy tones HT801/HT802
are new additions to the popular handy tone ATA product family. This manual will help you learn how to
operate and manage your HT801/HT802 analog telephone adaptor and make the best use of its many
upgraded features including simple and quick installation, 3-way conferencing, direct IP-IP Calling, and new
provisioning support among other features. The HT801/HT802 are very easy to manage and configure and
are specifically designed to be an easy to use and affordable VoIP solution for both the residential user and
the teleworker.
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HT801/HT802 Administration Guide
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PRODUCT OVERVIEW

The HT801 is a one-port analog telephone adapter (ATA) while the HT802 is a 2-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/HT802 are an ideal ATA for individual
use and for large scale commercial IP voice deployments.

Feature Highlights

The following table contains major features of HT801 and HT802:
Table 1: HT801/HT802 Features at a Glance
HT801
HT802
1 SIP profile through 1 FXS port on HT801, 2 SIP profiles through 2 FXS
ports on HT802 and single 10/100Mbps port on both models.
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.
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HT801/HT802 Administration Guide
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HT801/HT802 Technical Specifications

Following tables resumes all the technical specifications including the protocols/standards supported,
voice codecs, telephony features, languages and upgrade/provisioning settings for the HT801 and the
HT802.
Table 2: HT801/HT802 Technical Specifications
Interfaces
HT801
HT802
Telephone Interfaces
One (1) RJ11 FXS port
Two (2) RJ11 FXS ports.
Network Interface
One (1) 10/100Mbps auto-sensing Ethernet port (RJ45).
LED Indicators
POWER, INTERNET, PHONE
POWER, INTERNET, PHONE1,
PHONE2.
Factory Reset Button
Yes.
Voice, Fax, Modem HT801
HT802
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.723.1, G.729A/B, G.726,
G.722, 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.
2 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
HT801
HT802
Network Protocols
TCP/IP/UDP, RTP/RTCP, FTP/FTPS, HTTP/HTTPS, ARP/RARP, ICMP, DNS,
DDNS, DHCP, NTP, TFTP, SSH, Telnet, STUN, SIP (RFC3261), SIP over
TCP/TLS, SRTP, TR-069.
QoS
Layer 3 (ToS, DiffServ, MPLS).
DTMF Methods
In-audio, RFC2833 and/or SIP INFO.
Provisioning and
Control
HTTP, HTTPS, FTP, FTPS, SSH, Telnet, TFTP, TR-069, secure and automated
provisioning using AES encryption, syslog.
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Security
HT801
HT802
Media
SRTP.
Control
TLS/SIPS/HTTPS/SSH/Telnet.
Management
Syslog support, SSH, Telnet remote management using web browser.
Physical
HT801
HT802
Universal Power
Supply
Input: 100-240VAC, 50-60Hz. Output: 5.0VDC/1.0A.
Environmental
Operational: 32o – 104
o
F or 0º – 40ºC.
Storage: 14o – 140
o
F or -10º – 60ºC.
Humidity: 10 – 90% Non-condensing.
Dimensions and
Weight
100mm x 100mm x 29.5mm 102g
100mm x 100mm x 29.5mm 114g (without package).
Compliance HT801
HT802
Compliance
FCC Part15B EN55032, EN55024,
EN61000-3-2, EN61000-3-3,
EN60950-1, AS/NZS CISPR22,
AS/NZS60950.1, S003
FCC 15B, AS/NZS CISPR22,
AS/NZS60950, EN55022, EN55024,
EN60950, EN61000-3-2, EN61000-3-
3, UL (Power supply).
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HT801/HT802 Administration Guide
Version 1.0.25.5

GETTING STARTED

This chapter provides basic installation instructions including the list of the packaging contents and
information for obtaining the best performance with the HT801/HT802.

Equipment Packaging

The HT801 ATA package contains:
The HT802 ATA package contains:
Figure 2: HT802 Package Contents
Note: Check the package before installation. If you find anything missing, contact your system administrator.
1x HT802
1x 5V Power Adapter
1x Ethernet Cable
1x Quick Installation Guide
1x GPL Statement
Figure 1: HT801 Package Contents
1x HT801
1x 5V Power Adapter
1x Ethernet Cable
1x Quick Installation Guide
1x GPL Statement
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HT801/HT802 Administration Guide
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HT801/HT802 Ports Description

The following figure describes the different ports on the back panel of the HT801.
The following figure describes the different ports on the back panel of the HT802.
Figure 4: HT802 Back Panel
Table 3: Definition of the HT801/HT802 Connectors
Phone for HT801
Phone 1 & 2 for HT802
Connects the analog phones / fax machines to the phone adapter using an RJ-11 telephone cable.
Internet port
Connects the phone adapter to your router or gateway using an Ethernet
RJ45 network cable.
Micro-USB Power
Connects the phone adapter to PSU (5V – 1A).
Reset
Factory reset button. Press for 7 seconds to reset factory default settings.
Phone 1 & 2
(RJ-11 FXS Ports)
Micro USB
Power DC
Internet
Port
Phone
(RJ-11 FXS Port)
Micro USB
Power DC
Internet
Port
Figure 3: HT801 Back Panel
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HT801/HT802 Administration Guide
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Connecting HT801/HT802

The HT801 and HT 802 a re designed for easy configuration and easy installation, to connect your HT801
or HT802, please follow the steps above:
1. Insert a standard RJ11 telephone cable into the phone port and connect the other end of the telephone cable to a standard touch-tone analog telephone.
2. Insert the Ethernet cable into the internet or LAN port of the HT801/ht802 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.)
3. Insert the power adapter into the HT801/HT802 and connect it to a wall outlet.
Power, Ethernet and Phone LEDs will be solidly lit when the HT801/HT802 is ready for use.
Figure 5: Connecting the HT801/HT802

HT801/HT802 LEDs Pattern

There are 3 LED buttons on HT801 and 4 LED buttons on HT802 that help you manage the status of your
Handy Tone.
Figure 6: HT801/HT802 LEDs Pattern
HT801
or
HT802
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Table 4: HT801/HT802 LEDs Pattern Description
LED Lights
Status
Power LED
The Power LED lights up when the HT801/HT802 is powered on and it flashes when the HT802 is booting up
Internet LED
The Ethernet LED lights up when the HT801/HT802 is connected to your
network through the Ethernet port and it flashes when there is data being sent or received.
Phone LED for HT801
Phone LED 1&2 for
HT802
The phone LED 1 & 2 indicate status of the respective FXS port-phone on
the back panel
OFF - UnregisteredON (Solid Blue) - Registered and AvailableBlinking every second - Off-Hook / Busy Slow blinking - FXS LEDs indicates voicemail
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HT801/HT802 Administration Guide
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CONFIGURATION GUIDE

The HT801/HT802 can be configured via one of two ways:
The IVR voice prompt menu. The Web GUI embedded on the HT801/HT802 using PC's web browser.

Obtain HT801/HT802 IP Address via Connected Analogue Phone

HT801/HT802 is by default configured to obtain the IP address from DHCP server where the unit is located.
To know which IP address is assigned to your HT801/HT802, you should access to the “Interactive Voice
Response Menu” of your adapter via the connected phone and check its IP address mode.
Please refer to the steps below to access the interactive voice response menu:
1. Use a telephone connected to phone ports of your HT801/HT802.
2. Press *** (press the star key three times) to access the IVR menu and wait until you hear “Enter the menu option “.
3. Press 02 and the current IP address will be announced.

Understanding HT801/HT802 Interactive Voice Prompt Response Menu

The HT801/HT802has a built-in voice prompt menu for simple device configuration which lists actions,
commands, menu choices, and descriptions. The IVR menu works with any phone connected to the
HT801/HT802. Pick up the handset and dial “***” to use the IVR menu.
Table 5: Voice Prompt Menu
Menu
Voice Prompt
Options
Main
Menu
“Enter a Menu Option”
Press “*” for the next menu option Press “#” to return to the main menu Enter 01-05, 07,10, 13-17,47 or 99 menu options
01
“DHCP Mode”, “Static IP Mode”
Press “9” to toggle the selection
If using “Static IP Mode”, configure the IP address
information using menus 02 to 05.
If using “Dynamic IP Mode”, all IP address information
comes from the DHCP server automatically after
reboot.
02
“IP Address “+ IP address
The current WAN IP address is announced
If using “Static IP Mode”, enter 12-digit new IP
address. You need to reboot the HT801/HT802 for
the new IP address to take Effect.
03
“Subnet “+ IP address
Same as menu 02
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04
“Gateway “+ IP address
Same as menu 02
05
“DNS Server “+ IP
address
Same as menu 02
07
Preferred Vocoder
Press “9” to move to the next selection in the list:
PCM U / PCM A iLBC G-726 G-723 G-729 OPUS G722
10
“MAC Address”
Announces the Mac address of the unit.
13
Firmware Server IP
Address
Announces current Firmware Server IP address.
Enter 12-digit new IP address.
14
Configuration Server IP
Address
Announces current Config Server Path IP address.
Enter 12-digit new IP address.
15
Upgrade Protocol
Upgrade protocol for firmware and configuration
update. Press “9” to toggle between TFTP / HTTP /
HTTPS / FTP / FTPS. Default is HTTPS.
16
Firmware Version
Announces Firmware version information.
17
Firmware Upgrade
Firmware upgrade mode. Press “9” to toggle among
the following three options:
always check check when pre/suffix changes never upgrade
47
“Direct IP Calling”
Enter the target IP address to make a direct IP call,
after dial tone. (See “Make a Direct IP Call”.)
86
Voice Mail
Access to your voice mails messages.
99
“RESET”
Press “9” to reboot the device Enter MAC address to restore factory default setting (See Restore Factory Default Setting section)
701-702
Phone calls between
different ports of the same
HT802
HT802 supports inter-port calling from voice menu. 70X (X is the port number) “Invalid Entry”
Automatically returns to main menu
“Device not registered”
This prompt will be played immediately after off hook
If the device is not registered and the option “Outgoing
Call without Registration” is in NO
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Five success tips when using the voice prompt
“*” shifts down to the next menu option and “#” returns to the main menu. “9” functions as the ENTER key in many cases to confirm or toggle an option. All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP
address. For IP address, add 0 before the digits if the digits are less than 3 (i.e. - 192.168.0.26
should be key in like 192168000026. No decimal is needed).
Key entry cannot be deleted but the phone may prompt error once it is detected.

Configuration via Web Browser

The HT801/HT802 embedded Web server responds to HTTP GET/POST requests. Embedded HTML
pages allow a user to configure the HT801/HT802 through a web browser such as Google Chrome, Mozilla
Firefox and Microsoft’s IE.

Accessing the Web UI

1. Connect the computer to the same network as your HT801/HT802.
2. Make sure the HT801/HT802 is booted up.
3. You may check your HT801/HT802 IP address using the IVR on the connected phone.
Please see Obtain HT801/HT802 IP Address via Connected Analogue Phone
4. Open Web browser on your computer.
5. Enter the HT801/HT802’s IP address in the address bar of the browser.
6. Enter the administrator’s password to access the Web Configuration Menu (default password is: admin).
Notes:
The computer must be connected to the same sub-network as the HT801/HT802. This can be
easily done by connecting the computer to the same hub or switch as the HT801/HT802.
Recommended Web browsers:
o Microsoft Internet Explorer: version 10 or higher. o Google Chrome: version 58.0.3 or higher. o Mozilla Firefox: version 53.0.2 or higher. o Safari: version 5.1.4 or higher. o Opera: version 44.0.2 or higher.
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Web UI Access Level Management

There are two default passwords for the login page:
Table 6: Web access level management
User Level
Username
Password
Web Pages Allowed
End User Level
user
123
Only Status and Basic Settings
Administrator Level
admin
admin
All pages
Viewer Level
viewer
viewer
View all pages. Changes not
allowed.
The password is case sensitive with maximum length of 25 characters. When changing any settings, always
submit them by pressing Update or Apply button on the bottom of the page. After submitting the changes
in all the Web GUI pages, reboot HT801/HT802 to have the changes take effect if necessary; most of the
options under the Advanced Settings and FXS Port (x) pages require reboot.

Saving the Configuration Changes

After users makes changes to the configuration, pressing Update button will save but not apply the changes until Apply button is clicked. Users can instead directly press Apply button. We recommend rebooting or powering cycle the phone after applying all the changes.

Changing Admin Level Password

1. Access your HT801/HT802 web UI by entering its IP address in your favorite browser (screenshots below are from HT802 but the same applies to HT801).
2. Enter your admin password (default: admin).
3. Press Login to access your settings and navigate to Advanced SettingsAdmin Password.
4. Enter the new admin password.
5. Confirm the new admin password.
6. Press Apply at the bottom of the page to save your new settings.
Figure 7: Admin Level Password

Changing User Level Password

1. Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
2. Enter your admin password (default: admin).
3. Press Login to access your settings and navigate to Basic SettingsEnd User Password.
4. Enter the new end-user password.
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5. Confirm the new end-user password.
6. Press Apply at the bottom of the page to save your new settings.
Figure 8: User Level Password

Changing Viewer Password

1. Access your HT812/HT814 web UI by entering its IP address in your favorite browser.
2. Enter your admin password (default: admin).
3. Press Login to access your settings.
4. Go to Basic Settings New Viewer Password and enter the new viewer password.
5. Confirm the new viewer password.
6. Press Apply at the bottom of the page to save your new settings.
Figure 9: Viewer Level Password

Changing HTTP Web Port

1. Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
2. Enter your admin password (default: admin).
3. Press Login to access your settings and navigate to Basic SettingsWeb Port.
4. Change the current port to your desired/new HTTP port. Ports accepted are in range [1-65535].
5. Press Apply at the bottom of the page to save your new settings.
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Figure 10: Web HTTP Port

Web Configuration Pages Definitions

This section describes the options in the HT801/HT802 Web UI. As mentioned, you can log in as an
administrator or an end user.
Status: Displays the system info, network status, account status, and line options.  Basic Settings: Configures the end user level password, IP address modes, web access, time zone
settings and language.
Advanced Settings: Configures networks, upgrading and provisioning, TR-069, security settings,
date and time, syslog, audio settings, call settings and call progress tones.
FXS Port (1,2): Configures SIP settings, SIP registration, accounts settings, NAT settings, call features,
ring tones.

Advanced Settings Page Definitions

Table 7: Advanced Settings
Advanced Settings
New Admin Password
Defines the administrator level password to access the Advanced Web
Configuration page. This field is case sensitive. Only the administrator can
configure the “Advanced Settings” page. Password field is purposely left
blank for security reasons after clicking update and saved. Max. length is 25
characters.
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Confirm Admin Password
Confirms the new admin password.
Disable User Level Web
Access
Disable or enable User Level Web Access. Default is No.
Disable Viewer Level Web
Access
Disable or enable Viewer Level Web Access Default is No.
Layer 2 QoS
Sets values for 802.1Q/VLAN Tag. Default is 0. Valid range is 0-4094. SIP 802.1p. Default is 0. Valid range is 0-7. RTP 802.1p. Default is 0. Valid range is 0-7.
Black List for WAN Side Port
It could be either port range or single port separated by a “,” Example: “5000-6000, 7000 “.
STUN Server is
Configures IP address or domain name of STUN server. Only non-
symmetric NAT routers work with STUN.
Keep-alive interval
Sends periodically a blank UDP packet to SIP server to keep the "ping hole"
on the NAT router open. Default is 20 seconds.
Use STUN to detect network
connectivity
Uses STUN keep-alive to detect WAN side network problems. If keep-alive
request does not yield any response for configured number of times
(minimum 3), the device will restart the TCP/IP stack.
If the STUN server does not respond when the device boots up, the feature
is disabled. Default setting is No.
Use DNS to detect network
connectivity
Uses DNS to detect WAN side network problems. Default setting is “No”.
Use ARP to detect network
connectivity
Uses ARP to check the network connectivity. Default is Yes.
Verify host when using
HTTPS
Enables / disables the host verification when using HTTPS.
Upgrade via
Selects firmware upgrade/provisioning method: TFTP, HTTP, HTTPS, FTP
or FTPS. Default is HTTPS.
Firmware Server Path
Sets IP address or FQDN of firmware server. The URL of the server that
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hosts the firmware release. Default is fm.grandstream.com/gs.
Config Server Path
Sets IP address or FQDN of configuration server. The URL of the server that
hosts the configuration file to provision HT801/HT802. Default is fm.grandstream.com/gs.
XML Config File Password
Decrypts XML configuration file when encrypted. The password used for
encrypting the XML configuration file using OpenSSL.
HTTP/HTTPS FTP/FTPS User
Name
Enters user name to authenticate with HTTP/HTTPS FTP/FTPS server.
HTTP/HTTPS FTP/FTPS
Password
Enters password to authenticate with HTTP/HTTPS FTP/FTPS server.
Firmware File Prefix
Checks if firmware file is with matching prefix before downloading it. This
field enables user to store different versions of firmware files in one directory
on the firmware server.
Firmware File Postfix
Checks if firmware file is with matching postfix before downloading it.
This field enables user to store different versions of firmware files in one
directory on the firmware server.
Config File Prefix
Checks if configuration files are with matching prefix before downloading
them. This field enables user to store different configuration files in one
directory on the provisioning server.
Config File Postfix
Checks if configuration files are with matching postfix before downloading
them. This field enables user to store different configuration files in one
directory on the provisioning server.
Allow DHCP Option 66 to
Override Server
Obtains configuration and upgrade server’s information using options 66 from
DHCP server.
Note: If DHCP Option 66 is enabled, the HT801/HT802 will attempt
downloading the firmware file from the server URL provided by DHCP, even
though Config Server Path is left blank.
The server URL provided by DHCP can include authentication credentials
using following format: “username:password@Provisioning_Server_IP”.
3CX Auto Provision
Sends multicast “SUBSCRIBE” message for provisioning at booting stage,
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used for PnP (Plug-and-Play) configuration. Default is Yes.
Automatic Upgrade
Specifies when the firmware upgrade process will be initiated; there are 4
options:
No: The HT801/HT802 will only do upgrade once at boot up. Check every X minutes: User needs to specify a period in minutes.Check every day: User needs to specify the start hour and the end
hour of the day (0-23).
Check every week: User needs to specify “Day of the week (0-6)”.
(Day of week is starting from Sunday).
Default is No.
Randomized Automatic
Upgrade
Randomized Automatic Upgrade within the range of hours of the day or postpone the upgrade every X minute(s) by random 1 to X minute(s).
Always Check for New
Firmware at Boot up
Configures the HT801/HT802 to always search for the new firmware at boot
up. During the boot stage, the HT801/HT802 will contact the firmware
upgrade server to search for a new firmware, when available it will start the
upgrade process, otherwise it will boot normally.
Check New Firmware only
when F/W pre/suffix changes
Configure the HT801/HT802 to search for the new firmware when the
firmware prefix / suffix changes.
When this option is selected, the HT801/HT802 will check for updates only
when the pre/suffix has been changed.
Always Skip the Firmware
Check
Configures the HT801/HT802 to skip the firmware check, when this option
is selected the HT801/HT802 will always skip searching for a new firmware.
Disable SIP NOTIFY
Authentication
Disables the SIP NOTIFY Authentication on the phone adapter. If set to
“Yes”, the phone adapter will not challenge NOTIFY with 401. The default setting is “No
Authenticate Conf File
Authenticates configuration before being accepted. This protects the
configuration from unauthorized modifications. Default is No.
Validate Server Certificates
This feature allows users to validate server certificates with our trusted list
of TLS connections. The device needs to reboot after changing the setting. Default is enabled. DDNS
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Trusted CA certificates A
Uses the certificate for Authentication if “Check Domain Certificates” is set to “Yes” under “Account”--> “SIP Settings”.
Trusted CA certificates B
Uses the certificate for Authentication if “Check Domain Certificates” is set to “Yes” under “Account”--> “SIP Settings”.
Load CA Certificates
This feature allows user to specify which certificate to trust when performing
server authentication.
Build-in trusted : (Default) Build-in trusted certificates Custom trusted certificate: Uploaded Certificates All trusted Certificates: Both built-in and uploaded Certificates
SIP TLS Certificate
Specifies SSL certificate used for SIP over TLS is in X.509 format. The
HT801/HT802 has built-in private key and SSL certificate. Maximum supported length is 4069.
SIP TLS Private Key
Specifies TLS private key used for SIP over TLS is in X.509 format. Maximum supported length is 4069.
SIP TLS Private Key
Password
Specifies SSL Private key password used for SIP Transport in TLS/TCP.
Custom Certificate (Private Key + Certificate)
Allows users to update to the device their own certificate signed by custom
CA certificate to manage client authentication.
Enable TR-069
Sets the phone adapter system to enable the “CPE WAN Management
Protocol” (TR-069). Default setting is Yes.
ACS URL
Specifies URL of TR-069 Auto Configuration Servers (e.g.,
http://acs.mycompany.com), or IP address. Default setting is: “https://acs.gdms.cloud”
ACS Username
Enters username to authenticate to ACS.
ACS Password
Enters password to authenticate to ACS.
Periodic Inform Enable
Sends periodic inform packets to ACS. Default is Yes.
Periodic Inform Interval
Sets frequency that the inform packets will be sent out to ACS. Default is 86400 seconds.
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Connection Request
Username
Enters username for ACS to connect to the HT801/HT802.
Connection Request
Password
Enters password for ACS to connect to the HT801/HT802.
Connection Request Port
Configures the TR-069 connection request port. The value range is 0 to
65535.Default is 7547
CPE SSL Certificate
Configures the Cert File for the phone adapter to connect to the ACS via
SSL.
CPE SSL Private Key
Specifies the Cert Key for the phone adapter to connect to the ACS via SSL.
Enable SNMP
Default is No.
SNMP Version
Choose between (Version 1, Version 2c, or Version 3).
SNMP Port
Listening Port of SNMP daemon (Default 161).
SNMP Trap IP Address
IP address of trap destination. Up to 3 trap destinations are supported. Users
should enter the IP addresses separated with comma (,).
SNMP Trap port
Port of Trap destination (Default 162)
SNMP Trap Version
Choose between (Version 1, Version 2c, or Version 3).
SNMP Trap Interval
Time interval between traps (Default is 5).
SNMPv1/v2c Community
Name of SNMPv1/v2c community.
SNMPv1/v2c Trap Community
Name of SNMPv1/v2c trap community.
SNMPv3 User Name
User name for SNMPv3.
SNMPv3 Security Level
noAuthUser: Users with security level noAuthnoPriv and context name as
noAuth.
authUser: Users with security level authNoPriv and context name as auth. privUser: Users with security level authPriv and context name as priv.
SNMPv3 Authentication
Protocol
Select the Authentication Protocol: “None” or “MD5” or “SHA.”
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SNMPv3 Privacy Protocol
Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
SNMPv3 Authentication Key
Enter the Authentication Key.
SNMPv3 Privacy Key
Enter the Privacy Key.
SNMPv3 Trap User Name
User name for SNMPv3 Trap.
SNMPv3 Trap Security Level
noAuthUser: Users with security level noAuthnoPriv and context name as
noAuth.
authUser: Users with security level authNoPriv and context name as auth. privUser: Users with security level authPriv and context name as priv.
SNMPv3 Trap Authentication
Protocol
Select the Authentication Protocol: “None” or “MD5” or “SHA”.
SNMPv3 Trap Privacy
Protocol
Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
SNMPv3 Trap Authentication
Key
Enter the Trap Authentication Key.
SNMPv3 Trap Privacy Key
Enter the Trap Privacy Key.
Enable RADIUS Web Access
Control
Default is No.
Action upon RADIUS Auth
Server Error
Choose action upon RADIUS server error. Default is Authenticate Locally
(Default Authenticate Locally)
RADIUS Auth Server Address
Address of RADIUS Auth server.
RADIUS Auth Server Port
Port of RADIUS Auth server.
RADIUS Shared Secret
Set RADIUS shared secret.
RADIUS VSA Vendor ID
Configure RADIUS VSA Vendor ID to match RADIUS server's configuration.
Default is 42397 for Grandstream Networks Inc.
RADIUS VSA Access Level
Attribute
Configure RADIUS VSA Access Level Attribute to match RADIUS server's
configuration. Incorrect setting would cause Radius authenticate fail.
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Enable DDNS
Allow users to use DDNS.
DDNS Server
Selects DDNS Server: dyndns, freedns.afraid.org, zoneedit.com, no-ip.com,
oray.net. Default is dyndns.
DDNS Username
64 characters as Max String Length.
DDNS Password
64 characters as Max String Length.
DDNS Hostname
64 characters as Max String Length.
DDNS Hash
64 characters as Max String Length.
Enable OpenVPN
Allow user to enable OpenVPN. Default is No.
OpenVPN Server Address
Specify the IP address or FQDN for the OpenVPN Server.
OpenVPN Port
Specify the listening port of the OpenVPN server. Default is 1194
OpenVPN Interface type
Specify the Interface type of OpenVPN whether TAP or TUN. Default is TUN.
OpenVPN Transport
Specify the Transport Type of OpenVPN whether UDP or TCP. Default is UDP.
Enable OpenVPN LZO
Compression
Enable OpenVPN LZO Compression. Default is Yes.
OpenVPN Encryption
Select the OpenVPN Encryption. Default is BF-CBC 128 bit (default key).
OpenVPN Digest
Select the OpenVPN Digest. Default is SHA1.
OpenVPN CA
Specifies the OpenVPN CA. Maximum Character Number is 8192.
OpenVPN Certificate
Specifies the OpenVPN Certificate. Maximum Character Number is 8192.
OpenVPN Client Key
Specifies the Client Key. Maximum Character Number is 8192.
OpenVPN Client Key
Password
Configures the OpenVPN Client Key Password. Maximum Length is 64.
System Ring Cadence
Configuration option is set ring cadence on FXS port for all incoming calls. Syntax: c=on1/off1-on2/off2-on3/off3; (3 cadences maximum)
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Default is set to c=2000/4000; (US standards)
Call Progress Tones:
Dial Tone Ringback Tone Busy Tone Reorder Tone Confirmation Tone
Call Waiting Tone Prompt Tone Conference Party
Hangup Tone *
Using these settings, users can configure tone frequencies and cadence
according to their preference. By default, they are set to North American
frequencies.
Configure these settings with known values to avoid uncomfortable high pitch
sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period
of silence. In order to set a continuous tone, OFF should be zero. Otherwise
it will ring ON ms and a pause of OFF ms and then repeat the pattern. Example configuration for N.A. Dial tone: f1=350@-13, f2=440@-13, c=0/0; Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [...] (Note: freq: 0 - 4000Hz; vol: -30 - 0dBm) * “Conference Party Hang-up Tone” will apply only if the “Special Feature” is set to “MTS”.
Prompt Tone Access Code
Key pattern to get Prompt Tone. Maximum 20 digits.
Lock Keypad Update
Configuration update via keypad (analog phone connected to FXS port
keypad using IVR menu) is disabled if set to Yes.
Disable Voice Prompt
Voice prompt is disabled if set to Yes.
Disable Direct IP Call
Direct IP call is disabled if set to Yes.
Play Busy Tone When
Account is unregistered
When this feature is set to Yes, device will play busy tone when the FXS port
account is not registered and the attached analog phone is offhook.
Blacklist for Incoming Calls
Allow users to block incoming calls from specific list of numbers.
Maximum allow 10 SIP numbers and each number should be separated by a
comma (‘,’) in web UI.
Other allowed characters are 0-9, comma (“,”), asterisk (‘*’), pound sign (‘#’)
and plus sign (‘+’).
NTP Server
Defines the URL or IP address of the NTP server. The ATA may obtain the
date and time from the server. The default setting is “pool.ntp.org”.
Allow DHCP Option 42 to
override NTP server
Defines whether DHCP Option 42 should override NTP server or not. When
enabled, DHCP Option 42 will override the NTP server if it’s set up on the
LAN. The default setting is Yes.
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DHCP Option 17 Enterprise
Number
Configure the DHCP option 17 number. Default is 3561
CDR File Option
By default, the device will split the allowed memory for CDR file into 2 parts.
Device will create the first CDR file which is half of the allowed size, when it
is full, device will create the second file.
When “CDR File Option” is set to Default “Keep”, device will keep the
call records when both files are full, no more new record will be stored.
When this feature is set to “Override”, device will clear the first CDR
file and start storing again.
The CDR file output will be available at Status page: [CDR File]
SIP File Option
By default, the device will split the allowed memory for SIP file into 2 parts.
Device will create the first SIP file which is half of the allowed size, when it is
full, device will create the second file.
When “SIP File Option” is set to Default “Keep”, device will keep the
call records when both files are full, no more new record will be stored.
When this feature is set to “Override”, device will clear the first SIP
file and start storing again.
The SIP file output will be available at Status page: [SIP File]
Note:Send SIP Log” must be enabled to be able to capture the trace.
Disable Weak TLS Cipher
Suites
Allows users to disable weak ciphers DES/3DES and RC4, Symmetric
Encryption SEED, Symmetric Authentication MD5, Protocol Version
SSLv2/SSLv3 or Disable All of the Above Weak TLS Ciphers Suites. Default is No.
Minimum TLS Version
This feature allows customer to choose desired Minimum TLS Version. Choices are:
- Unlimited
- TLS 1.0
- TLS 1.1
- TLS 1.2
Default is Unlimited.
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Maximum TLS Version
This feature allows customer to choose desired Maximum TLS Version. Choices are:
- Unlimited
- TLS 1.0
- TLS 1.1
- TLS 1.2
Default is Unlimited.
Syslog Protocol
If set to SSL/TLS, the syslog messages will be sent through secured TLS Protocol to syslog server. Default setting is UDP. Note: The CA certificate is required to connect with the TLS server A reboot is required to take effect.
Syslog Server
URL or IP address of syslog server.
Note: A reboot is required to take effect.
Syslog Level
Select the HT813 to report the log level. Default is NONE. The level is one of
EXTRA DEBUG, DEBUG, INFO, WARNING or ERROR. Syslog messages
are sent based on the following events:
1. product model/version on boot up (INFO level)
2. NAT related info (INFO level)
3. sent or received SIP message (DEBUG level)
4. SIP message summary (INFO level)
5. inbound and outbound calls (INFO level)
6. registration status change (INFO level)
7. negotiated codec (INFO level)
8. Ethernet link up (INFO level)
9. SLIC chip exception (WARNING and ERROR levels)
10. memory exception (ERROR level)
extra syslog style (EXTRA DEBUG level)
Note: A reboot is required to take effect.
Send SIP Log
Configures whether the SIP log will be included in the syslog messages. The default setting is No.
Automatic Reboot
Default is No. When “Yes, reboot every day at hour” or “Yes, reboot every
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week at day” or “Yes, reboot every month at day” is checked, user can specify
“Hour of the day (0-23)” or “Day of the week (0-6)” or “Day of the month (0-
30)”. Default time is Monday 1AM.
Download Device
Configuration
Press Download button to download device configuration file to local
computer. The filename is “config.txt”. The file is plain text and not including
password fields.
Download Device XML
Configuration
Press Download to download device configuration file to local computer. The
filename is “config.xml”. The file will not include password fields.
Upload Firmware
Press Upload from local directory button to load the firmware file to the
device from your computer. The firmware filename should be “ht80xfw.bin”
(ht802fw.bin for HT802 for instance).
Upload Configuration
Press Upload from local directory button to load configuration file to the
device from your computer. The configuration file should be an XML file (for
instance: “config.xml”). Note: The field <mac> is not mandatory in the
document but if available only device with specified MAC address will accept
the configuration file.
Export Backup Configuration
Press Download button to export device backup configuration to computer.
The output is “cfg<mac>_enc.xml” (where <mac> is the MAC address of the
device). The file is encrypted and can be used on same device only.
Restore From Backup
Configuration
Press Upload button to restore device configuration from previously exported
backup configuration.

Status Page Definitions

Table 8: Status Page Definitions
Account Status
MAC Address
Shows device ID in hexadecimal format. This is needed by network
administrators for troubleshooting. The MAC address will be used for
provisioning and can be found on the label on original box and on the label
located on the bottom panel of the device.
IPv4 Address
Displays assigned IPv4 address.
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IPv6 Address
Displays assigned IPv6 address.
VPN IPv4 Address
Displays assigned OpenVPN IPv4 address.
VPN IPv6 Address
Displays assigned OpenVPN IPv6 address.
Product Model
Displays product model info. Default is HT801/HT802.
Hardware Version
Displays the hardware revision information and the part number.
Software version
Program: Specifies Program version. This is the main firmware release
number, which is always used for identifying the software system of the
HT801/HT802.
Bootloader: Specifies Boot version. Core: Specifies Core version. Base: Specifies Base version. CPE: Specifies CPE version. CPE version is displayed only when
HT801/HT802 is connected to an ACS using TR-069 protocol.
Software Status
Indicates actual software status.
System Up Time
Indicates actual system time and uptime since last reboot.
CPU Load
Indicates CPU load (%)
Network Cable Status
Indicates physical network cable status:
Status (Up/Down), Speed (Mbps), Operational Mode (Full/Half Duplex)
PPPoE Link Up
Indicates PPPoE connection status.
NAT
Indicates type of NAT when it’s configured.
Port Status
Displays relevant information regarding the FXS ports, their registration,
current status and their appropriate User ID.
Port Options
Displays DND and call forward information on FXS ports.
CDR File
Download, Preview. Or, Delete call history records from the web GUI. Only
the last 1000 records will be available.
SIP File
Download, Preview or, Delete locally stored SIP trace.
Note: Send SIP Log” must be enabled to be able to capture the trace.
Provision
Displays provisioning status.
Core Dump
Provides generated core dump file if unit malfunctions. Clean will be displayed if no issues.
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GR909
Click on “Test Page” to be redirected to lines tests page (Hazardous
Potential, Foreign Electromotive Forces, Resistive Faults, Receiver Off hook
and Ringer Equivalent Number)

Basic Settings Page Definitions

Table 9: Basic Settings Page
Basic Settings
New End User Password
Configures user level password. Case sensitive and max. Length of 25
characters.
Confirm End User Password
Re-enter the end user password to confirm change user password on web
GUI to avoid typo or mistakes.
New Viewer Password
Configures viewer level password. Case sensitive and max. Length of 25 characters.
Confirm Viewer Password
Re-enter the viewer password to confirm change viewer password on web
GUI to avoid typo or mistakes.
Web Session Timeout
Configure timer to logout web session during idle. Default is 10 min. Range is 2-60 min.
Web Access Attempt Limit
Configure attempt limit before lockout. Default is 5. Range is 1-10.
Lockout time interval
If login attempt failed 5 times, login would be locked out for the time length.
(Default 15 mins. Range 1-15 min).
Web Access Mode
Allows users to choose the Web Access Mode between “HTTPS”, “HTTP”
and “Disabled”. If “Disabled” is selected, web UI access will be disabled. By default, “HTTP” is selected.
HTTP Web Port
Customizes HTTP port used to access the HT813 web UI. Default is 80.
HTTPS Web Port
Customizes HTTPS port used to access the HT812/HT814 web UI. Default
is 443.
Disable SSH
Enables/disables the SSH access. Default is No (enabled).
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SSH Port
Allows users to self-configure SSH Port number. By default, the port number is 22.
SSH Idle Timeout
Configures SSH session timeout. [0 – 86400] seconds; Default is 0.
Disable Telnet
Enables/disables the Telnet access. Default is Yes (disabled).
Telnet Port
Allows users to self-configure Telnet Port number. By default, the port number is 23.
Telnet Idle Timeout
Configure Telnet session timeout. [0 – 86400] seconds; Default is 0.
White List for WAN Side
Users can configure the white List for WAN Side to be used for remote management.
Multiple IPs are supported and need to be separated by “space”.
Example: 192.168.5.222 192.168.5.223 192.168.7.0/24 Note: If both blacklist and whitelist are not empty, the blacklist is
processed first, followed by the whitelist.
Black List for WAN Side
Users can configure the black List for WAN Side to ban WAN side web access.
Multiple IPs are supported and need to be separated by “space”.
Example: 192.168.5.222 192.168.5.223 192.168.7.0/24 Note: If both blacklist and whitelist are not empty, the blacklist is
processed first, followed by the whitelist.
Internet Protocol
Selects one of the following IP protocol modes:
IPv4 Only: Enforce IPv4 protocol only. IPv6 Only: Enforce IPv6 protocol only. Both, Prefer IPv4: Enable both IPv4 and IPv6 and prefer IPv4. Both, prefer IPv6: Enable both IPv4 and IPv6 and prefer IPv6.
Note: Make sure to reboot the phone for the changes to take effect.
IPv4 Address
Allows users to configure the appropriate network settings on the HT80x to
obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE".
By default, it is set to "DHCP".
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Dynamically assigned via
DHCP
All the field values for the static IP mode are not used (even though they
are still saved in the flash memory.) The HT801/802 acquires its IP address
from the first DHCP server it discovers from the LAN it is connected.
DHCP hostname: Specifies the name of the client. The name may
or may not be qualified with the local domain name. This field is
optional but may be required by ISP.
DHCP domain name: Specifies the domain name that client
should use when resolving hostname via the Domain Name
System.
DHCP vendor class ID: Exchanges vendor class ID by clients and
servers to convey particular configuration or other identification
information about a client. Default is HT8XX.
Use PPPoE
Set the PPPoE account settings. If selected, ATA attempt to establish a
PPPoE session if any of the PPPoE fields is set.
PPPoE account ID: Defines the PPPoE username. Necessary if
ISP requires you to use a PPPoE (Point to Point Protocol over
Ethernet) connection.
PPPoE password: Specifies the PPPoE account password. PPPoE Service Name: Defines PPPoE service name. If your ISP
uses a service name for the PPPoE connection, enter the service
name here. This field is optional. Default is blank.
Preferred DNS server
Specifies preferred DNS server to use when DHCP or PPPoE are set.
Statically configured as IP
address
Configure IP address, subnet Mask, default router IP address, 1st preferred
DNS server, 2nd preferred DNS server. These fields are set to zero by
default.
IPv6 Address
Allows users to configure the appropriate network settings on the HT80x to
obtain IPv6 address. Users could select "DHCP", "Static IP”. By default, it
is set to "DHCP".
DHCP mode: all the field values for the static IP mode are not used
(even though they are still saved in the flash memory.) The
HT801/HT802 acquires its IP address from the first DHCP server it
discovers from the LAN it is connected.
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Static IP mode: configure IP address, 1
st
and 2nd DNS server,
preferred DNS server. These fields are set to zero by default.
- Full Static: When enabling the option full static, users need to
specify the Static IPv6 and the IPv6 Prefix length.
- Prefix Static: When enabling the option prefix static, users need
to specify the IPv6 Prefix (64 bits).
Enable Management Interface
Allows administrator to setup a Virtual Network Interface on top of the physical interface for device management. Default is No.
Management Access
Chooses whether to access using Management Interface Only (Default) Or Both Service and Management Interfaces
Management Interface IPv4
Address
Configures Voice VLAN Type :
Default is dynamically assigned via DHCP
Or,
statically configured as:
o IP Address : Default is 192.168.100.100 o Subnet Mask : Default is 255.255.255.0 o Default Router : Default is 192.168.100.1 o DNS Server 1 : Default is 0.0.0.0 o DNS Server 2 : Default is 0.0.0.0
802.1Q/VLAN Tag
vlan tagging : [0 – 4094]; Default is 0
802.1p priority value :
priority : [0 – 7]; Default is 0
Time Zone
Selects time zone to define date/time on the device.
Self-Defined Time Zone
Allows users to define their own time zone.
Allow DHCP server to set Time
Zone
Obtains time zone setting (offset) from a DHCP server using DHCP Option
2; it will override selected time zone. If set to “No”, the analogue adapter
will use selected time zone even if provided by DHCP server. The Default setting is Yes.
Language
Configures the languages of the voice prompt and web interface, except
Spanish that it is only in IVR. Available languages: English, Chinese,
Traditional Chinese, Russian, Spanish IVR.
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Reset Type
Gives the administrator the option to restore default configuration on the
HT801/HT802. There are 3 types of factory reset:
ISP Data Reset: All VoIP related configuration (mainly everything
located on FXS page).
VOIP Data Reset: All ISP (Internet Service Provider) configuration
which may affect the IP address.
Full Reset: Both VoIP and ISP related configuration at the same time.
Note: After you choose the reset type, you must click the reset button to
take effect.

FXS Ports Pages Definitions

Table 10: FXS Ports
FXS Port (1,2)
Accounts Active
Activates/Deactivates the accounts. The FXS port configuration will not
change if disabled, although the port will not be operational, in this state,
there will be no dial tone when picking up the analog phone and
making/receiving calls will not be possible.
Primary SIP Server
Configures SIP server IP address (Supports both IPv4 and IPv6 addresses)
or domain name provided by VoIP service provider. (For example:
sip.mycompany.com, IPv4: 192.168.5.170, or IPv6: 2001:1260:1:277::3).
This is the primary SIP server used to send/receive SIP messages from/to
HT8xx.
Failover SIP Server
Defines failover SIP server IP address (Supports both IPv4 and IPv6
addresses) or domain name provided by VoIP service provider. (For
example: sip.mycompany.com, IPv4: 192.168.5.170, or IPv6:
2001:1260:1:277::3)This server will be used if primary SIP server becomes
unavailable.
Prefer Primary SIP Server
Selects to prefer primary SIP server. The account will register to primary
Server if registration with Failover server expires. Default is No.
Outbound Proxy
Specifies IP address (Supports both IPv4 and IPv6 addresses) or domain
name of outbound Proxy, or media gateway, or session border controller.
(For example: proxy.myprovider.com, or IP address, if any: IPv4:
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192.168.5.170/ IPv6: 2001:1260:1:277::3). Used by HT801/HT802 for
firewall or NAT penetration in different network environments. If symmetric
NAT is detected, STUN will not work, and only outbound proxy can correct
the problem.
Backup Outbound Proxy
Configures the backup outbound proxy to be used when the “Outbound
Proxy” registration fails. (For example: proxy.myprovider.com, or IP
address, if any: IPv4: 192.168.5.170/ IPv6: 2001:1260:1:277::3). By
default, this field is left empty.
Prefer Primary Outbound
Proxy
If the user configures this option to “Yes”, when registration expires, the
device will re-register via primary outbound proxy. By default, this option is disabled.
Allow DHCP Option 120
(override SIP Server)
Configures the HT802 to collect SIP server address from DHCP option 120. Default is No.
SIP Transport
Selects transport protocol for SIP packets; UDP or TCP or TLS. Please make
sure your SIP Server or network environment supports SIP over the selected
transport method. Default is UDP.
SIP URI Scheme When Using
TLS
Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.
Use Actual Ephemeral Port in
Contact with TCP/TLS
Controls the port information in the Via header and Contact header. If set to “No”, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the connection. The default setting is “No”.
NAT Traversal
Indicates type of NAT for each account. This parameter configures whether
the NAT traversal mechanism is activated. Users could select the
mechanism from No, Keep-alive, STUN, UPnP. Default setting is No.
SIP User ID
Defines user account information provided by VoIP service provider (ITSP).
Usually in the form of digit similar to phone number or actually a phone
number.
Authenticate ID
Determines account authenticate ID provided by VoIP service provider
(ITSP). Can be identical to or different from “SIP user ID”.
Authenticate Password
Specifies account password provided by VoIP service provider (ITSP) to
register to SIP servers.
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Name
Chooses a name to be associated to user.
DNS Mode
Selects DNS mode to use for the client to look up server. One mode can be
chosen. Default is A Record.
A Record: resolves IP Address of target according to domain name. SRV: DNS SRV resource records indicate how to find services for
various protocols.
NAPTR/SRV: Naming Authority Pointer according to RFC 2915. Use Configured IP: if SIP server is configured as domain name,
device will not send DNS query, but will use "Primary IP" or "Backup
IP" to send SIP message if at least one of them are not empty. It will try
to use "Primary IP" first, after 3 tries without any response, it will switch
to "Backup IP 1", then "Backup IP 2", and then it will switch back to
"Primary IP" after 3 re-tries.
DNS SRV use Registered IP
When this option is set to “Yes”, when the HT is registered on second SRV
and makes an outbound call, it will try the second SRV (registered IP) first.
By default, this option is disabled, and the DNS SRV will use first SRV
instead of the registered IP.
DNS SRV Failover Mode
Configure the preferred IP mode for DNS SRV or NAPTR/SRV. If "default"
is selected, first IP from query result will be applied; If "Saved one until DNS
TTL" is selected, previous IP will be applied before reaches DNS timeout; If
"Saved one until no response“ is selected, previous IP will be applied even
after DNS time out until it’s unreachable.
Default: If the option is set with “default”, it will again try to send register
messages to one IP at a time, and the process repeats.
Saved one until DNS TTL: If the option is set to “Saved one until DNS
TTL”, it will send register messages to the previously registered IP first. If no
response, it will try to send one at a time for each IP. This behavior lasts if
DNS TTL (time-to-live) is up.
Saved one until no responses: If the option is set with “Saved one until
no responses”, it will send register messages to the previously registered IP
first, but this behavior will persist until the registered server does not
respond.
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Failback Timer
When the primary SBC is up, device will send SIP requests to the primary
SBC. If at any point device fails over to the secondary SBC, the SIP requests
will stay on the failover SBC for the duration of the failback timer. When the
timer expires, device will send SIP requests to the primary SBC, (in minutes.
Default is 60 minutes, max 45 days).
Register Before DNS SRV
Failover
This feature is used to control whether the device need to initiate a new
registration request (following existing DNS SRV fail-over mode) first and
then direct the nonregistration SIP request (INVITE) to the new successfully
registered server or not.
TEL URI
Indicates E.164 number in “From” header by adding “User=Phone”
parameter or using “Tel:” in SIP packets, if the HT801/HT802 has an
assigned PSTN Number.
Disabled: Use “SIP User ID” information in the Request-Line and
“From” header.
User=Phone: “User=Phone” parameter will be attached to the
Request-Line and “From” header in the SIP request to indicate the
E.164 number. If set to "Enable".
Enabled: "Tel:" will be used instead of "sip:" in the SIP request.
Please consult your carrier before changing this parameter. Default is
Disabled.
SIP Registration
Controls whether the HT801/HT802 needs to send REGISTER messages
to the proxy server. Default setting is Yes.
Unregister on Reboot
Controls whether to clear SIP user’s information by sending un-register
request to the proxy server. The un-registration is performed by sending a
REGISTER message with “Expires=0” parameter to the SIP server. This will
unregister the SIP account under the concerned FXS page. Default is No.
Outgoing Call Without
Registration
Enables the ability to place outgoing calls even if the account is not
registered (if allowed by ITSP); device will not be able to receive incoming
calls. Default is No.
Register Expiration
Refreshes registration periodically with specified SIP proxy (in minutes).
Maximum interval is 65535 minutes (about 45 days). Default is 60 minutes
(or 1 hour).
Reregister Before Expiration
Sends re-register request after specific time (in seconds) to renew
registration before the previous registration expires.
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SIP Registration Failure Retry
Wait Time
Sends re-register request after specific time (in seconds) when registration
process fails. Maximum interval is 3600 seconds (1 hour). Default is 20
seconds.
SIP Registration Failure Retry
Wait Time upon 403
Forbidden
Sends re-register request after specific time (in seconds) when registration
process fails with error 403 Forbidden. Maximum interval is 3600 seconds
(1 hour). Default is 1200 seconds.
Enable SIP OPTIONS/NOTIFY
Keep Alive
Enables SIP OPTIONS or SIP NOTIFY to track account registration status
so the phone adapter will send periodic OPTIONS/NOTIFY messages to
server to track the connection status with the server. Default setting is No.
SIP OPTIONS/NOTIFY Keep
Alive Interval
Configures the time interval when the phone adapter sends OPTIONS or
NOTIFY messages to the SIP server. The default setting is 30 seconds,
which means the phone adapter will send an OPTIONS/NOTIFY message
to the server every 30 seconds. The default range is 1-64800.
SIP OPTIONS/NOTIFY Keep
Alive Max Lost
Defines the Number of max lost packets for SIP OPTIONS or SIP NOTIFY
Keep Alive before re-registration. Between 3-10, default is 3.
Layer 3 QoS
Defines Diff-Serv values for SIP and RTP. SIP DSCP (Diff-Serv value in decimal, 0-63, default 26) RTP DSCP (Diff-Serv value in decimal, 0-63, default 46)
Local SIP Port
Defines local port to use by the HT801/HT802 for listening and transmitting
SIP packets. Default value for FXS 1 is 5060 and 5062 for FXS 2.
Local RTP Port
Defines the local RTP-RTCP port pair the HT801/HT802 will listen and
transmit. It is the HT801/HT802 RTP port for channel 0. The default value for FXS port is 5004.
Use Random SIP Port
Controls whether to use configured or random SIP ports. This is usually
necessary when multiple HT801/HT802 are behind the same NAT. Default
is No.
Use Random RTP Port
Controls whether to use configured or random RTP ports. This is usually
necessary when multiple HT801/HT802 are behind the same NAT. Default
is No.
Enable RTCP
Allows users to enable RTCP. Default setting is Yes.
Hold Target Before Refer
Allows user to hold the phone call before referring it. If set to No, the call will
not be hold before referred. Default is Yes.
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Refer-To Use Target Contact
Includes target’s “Contact” header information in “Refer-To” header when
using attended transfer. Default is No.
Transfer on Conference
Hang-up
If set to "Yes", when the phone hangs up as the conference initiator, the
conference call will be transferred to the other parties so that other parties
will remain in the conference call. Default setting is No.
Remove OBP from Route
Header
Removes outbound proxy info in “Route” header when sending SIP packets. Default is No.
Support SIP Instance ID
Includes “SIP Instance ID” attribute to “Contact” header in REGISTER
request as defined in IETF SIP outbound draft. Default is No.
Validate Incoming SIP
Message
Validates incoming messages. Default is No.
Check SIP User ID for
Incoming INVITE
Checks SIP User ID in the Request URI of incoming INVITE; if it doesn't
match the HT801/HT802 SIP User ID, the call will be rejected. Direct IP
calling will also be disabled. Default is No.
Authenticate Incoming
INVITE
Challenges the incoming INVITE for authentication with SIP 401
Unauthorized message. Default is No.
Authenticate server
certificate domain
Configures whether to validate the domain certificate when download the
firmware/config file. If it is set to "Yes", the phone will download the
firmware/config file only from the legitimate server. Default setting is "No".
Authenticate server
certificate chain
Configures whether to validate the server certificate when download the
firmware/config file. If it is set to "Yes", the phone will download the
firmware/config file only from the legitimate server. Default setting is "No".
Allow Incoming SIP
Messages from SIP Proxy
Only
Checks SIP address of the Request URI in the incoming SIP message; if it
doesn't match the SIP server address of the account, the call will be rejected. Default is No.
Use Privacy Header
Determines if the “Privacy header” will be presented in the SIP INVITE
message and if it includes the caller info in this header. If set to Default, it
will add Privacy header unless special feature is Telkom SA or CBCOM.
Default is Default.
Use P-Preferred-Identity
Header
Specifies if the P-Preferred-Identity Header will be presented in the SIP
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INVITE message. If set to “default", the P-Preferred-Identity Header will be
omitted in SIP INVITE message when Telkom SA or CBCO is active. If set
to "Yes", the P-Preferred-Identity Header will always be presented. If set to
"No", it will be omitted. Default setting is: Default.
Use P-Access-Network-Info
Header
With this feature enabled, device will populate the WAN access node with
IEE-802.11a, IEE-802.11b in P-Access-Network-Info SIP header.
Use P-Emergency-Info
Header
This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for
emergency calls.
Use P-Asserted-Identity
Header
With this feature set to Yes, device will send’ P-Asserted-Identity Header’ on
the SIP Invite. Default is No
SIP REGISTER Contact
Header Uses
Specifies which address (LAN or WAN address) the device will detect to use
it in SIP Register Contact Header. When set to LAN, Contact header will
include local IP from ATA in REGISTER messages, while if set to WAN,
host/port/contact will be updated from SIP 401/403/404/407 Via header
"received"/"rport" parameters in REGISTER messages. Default is LAN
Address.
Caller ID Fetch Order
Selects the Caller ID display order which need to be respected by the
HT801/HT802. The available options are:
Auto: When set to “Auto”, the HT801/HT802 will look for the caller ID
in the order of P-Asserted Identity Header, Remote-Party-ID Header
and From Header in the incoming SIP INVITE.
Disabled: When set to "Disabled", all incoming calls are displayed with "Unavailable".
From Header: When set to “From Header”, the HT801/HT802 will use the FROM header to display the caller ID.
Allow SIP Factory Reset
Allows to reset the devices directly through SIP Notify. If “Allow SIP Factory
Reset” is set to “YES” under FXS PORT, then the ATA receives the NOTIFY
from the SIP server with Event: reset, the HT should perform a factory reset
after the authentication. The authentication in this case can be either with:
The admin password if no SIP account is configured on the HT. With the credentials of the SIP account if configured on the ATA.
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Maximum Number of SIP
Request Retries
This feature allows user to configure the number of SIP retries before
failover occurs. ( between 1 and 10, default is 4)
SIP T1 Timeout
Defines T1 timeout value.
It is an estimate of the round-trip time between the client and server
transactions.
For example, the HT801/HT802 will attempt to send a request to a SIP
server.
The time it takes between sending out the request to the point of getting a
response is the SIP T1 timer. If no response is received the timeout is
increased to (2*T1) and then (4*T1). Request re-transmit retries would
continue until a maximum amount of time defined by T2. Default is 0.5
seconds.
SIP T2 Interval
Identifies maximum retransmission interval for non-INVITE requests and
INVITE responses. Retransmitting and doubling of T1 continues until it
reaches T2 value. Default is 4 seconds.
SIP Timer D
Configure the SIP Timer D defined in RFC3261. 0 - 64 seconds. Default is 0.
DTMF Payload Type
Defines payload type for DTMF using RFC2833. Default is 101.
Preferred DTMF method (in
order)
Sorts DTMF methods (in-audio, via RTP (RFC2833) or via SIP INFO) by
priority.
Inband DTMF Duration
Allows to adjust the inband DTMF duration sent from ATA to IPPBX. Default is 100 ms. Valid range: 40-2000 ms. Inter-duration: 50 ms. Valid range: 40-2000 ms.
Disable DTMF Negotiation
Uses above DTMF order without negotiation. Default is No.
Generate Continuous
RFC2833 Events
When enabled, the RFC2833 events are generated until key is released.
Default is No.
Send Hook Flash Event
If set to yes, flash will be sent as DTMF event. Default is No.
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Flash Digit Control
When it set to YES it allows the user to perform some call setting when
both channels are used while pressing:
"Flash + 1" in order to hang up the current call and resume a call that
was held.
"Flash + 2" in order to hold the current call and resume a call that was
held.
"Flash + 3" in order to perform 3-way conference. "Flash + 4" in order to perform attended transfer.
Note: Please refer to the user guide for detailed steps to perform above
operations.
Callee Flash to 3WC
When this feature is set to Yes, device would be able to set up the 3
way conference call even when device is the callee in the second call.
Default is No.
Off Hook Auto Dial
Configures a user ID or extension number that is automatically dialed when
off-hook. Only the user part of a SIP address needs to be entered. The
HT80x will automatically append the “@” and the host portion of the corresponding SIP address.
Off Hook Auto Dial Delay
Specifies the auto-dial delay in seconds after off hook. Valid range is 0-60 seconds. Default is 0.
Off Hook Auto-Dial DTMF
When ‘Off Hook Auto-Dial’ is configured, this feature allows user to
configure DTMF digits to send after ‘OffHook auto-dial’ call gets connected.
Proxy-Require
Determines a SIP Extension to notify the SIP server that the HT801/HT802
is behind a NAT/Firewall.
Use NAT IP
Defines NAT IP address used in SIP/SDP messages. It should only be used
if required by ITSP.
SIP User-Agent
This feature allows users to configure SIP User Agent. If not configured,
device will use the default User Agent header.
SIP User-Agent Postfix
Configures the SIP User-Agent Postfix
RFC2543 Hold
Toggles between RFC2543 hold and RFC3261 hold. RFC2543 hold allows
to disable the hold music sent to the other side, in this case IP address
(0.0.0.0) it will be sent in SDP instead of the IP address of the unit RFC 3261
(a line) will play the hold music to the other side.
Disable Call Waiting
Disables receiving a second incoming call when the line is engaged.
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Default is No.
Disable Call Waiting Caller ID
Disables displaying caller ID when receiving a second incoming call. Default is No.
Disable Call Waiting Tone
Disables playing call waiting tone during active call when receiving a second
incoming call. The CWCID will still be displayed. Default is No.
Disable Connected Line ID
Disables displaying the number of the person answering the phone. Default
is No.
Disable Receiver Off Hook
Tone
Enables / disables the warning to alert that the phone has been left off-hook
for an extended period of time. Default is No.
Disable Reminder Ring for
On-Hold Call
Enables playing the reminder ring. Default is No
Disable Reminder Ring for
DND
This feature allows user to disable reminder ring when FXS port is on DND
mode. Default is No
Disable Visual MWI
Disables use of visual message waiting indicator when there is an unread
voicemail message. Default is No.
Visual MWI Type
Configures Visual WMI Type of signal sent to the analog phone to make it
turn the lamp ON upon receiving a Voice mail.
Check the phone’s manual to find out what signal is supported, FSK (default)
or NEON.
Note: Some phones (depends on the model of the analog phone) when this
feature is set to NEON it might auto ring (short beeps) when there is a voice
mail available for that FXS port where it is connected.
Do Not Escape '#' as %23 in
SIP URI
Replaces # by %23 in some special situations. Default is No.
Disable Multiple m Line in
SDP
Sends only one m line in SDP, regardless of how many m fields are in the
incoming SDP. Default is No.
Ring Timeout
Stops ringing when incoming call if not answered within a specific period of
time. Valid range is 0-300 seconds, 0 means no timeout. Default is 60 seconds.
Delayed Call Forward Wait
Time
Forwards incoming call if not answered within a specific period of time when
delayed call forward is activated locally (using *92 code). Default value is 20 seconds.
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No Key Entry Timeout
Initiates the call within this time interval if no additional key entry during
dialing stage. Valid range is 1-15 seconds. Default is 4 seconds.
Early Dial
Sends an early INVITE each time a key is pressed when a user dials a
number. Otherwise, only one INVITE is sent after full number is dialed (user
presses Dial Key or after “no key entry timeout” expires).
This option should be used only if there is a SIP proxy is configured and
supporting 484 responses (Incomplete Address).
Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found
error). Default is No.
This feature is NOT designed to work with and should NOT be enabled for
direct IP-to-IP calling.
Dial Plan Prefix
Adds specified prefix to dialed number.
Use # as Dial Key
Treats “#” as the “Send” (or “Dial”) key. If set to “No”, this “#” key can be
included as part of the dialed number. Default is Yes.
Disable # as Redial Key
With this feature and feature ‘Use # as Dial Key’ set to Yes, the # key will act
as dial key but not as redial key. Default No.
Dial Plan
Dial Plan Rules:
1. Accept Digits: 1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
2. Grammar: x - any digit from 0-9; a. xx+ - at least 2 digits number; b. xx – exactly 2 digits number; c. ^ - exclude; d. . – wildcard, matches one or more characters e. [3-5] - any digit of 3, 4, or 5; f. [147] - any digit 1, 4, or 7; g. <2=011> - replace digit 2 with 011 when dialing h. <=1> - add a leading 1 to all numbers dialed, vice versa will remove
a 1 from the number dialed
i. | - or
Example 1: {[369]11 | 1617xxxxxxx} –
Allow 311, 611, 911, and any 10-digit numbers of leading digits
1617
Example 2: {^1900x+ | <=1617>xxxxxxx} –
Block any number with leading digits 1900 and add prefix 1617
for any dialed 7-digit numbers
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Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –
Allow any length of number with leading digit 2 and 10 digit-
numbers of leading digit 1 and leading exchange number
between 2 and 9; If leading digit is 2, replace leading digit 2 with
011 before dialing.
3. Default: Outgoing - { x+ | \+x+ | *x+ | *xx*x+ }
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. |
[3469]11 }
Explanation of example rule (reading from left to right):
^1900x. - prevents dialing any number started with 1900
<=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers
by dialing 7 numbers and 1617 area code will be added automatically
1[2-9]xx[2-9]xxxxxx - allows dialing to any US/Canada Number with 11 digits length
011[2-9]x. - allows international calls starting with 011
[3469]11 - allow dialing special and emergency numbers 311, 411, 611
and 911
Note: In some cases, user wishes to dial strings such as *123 to activate
voice mail or other application provided by service provider. In this case *
should be predefined inside dial plan feature. As an example { *x+ } will allow
to dial * followed by any length of numbers.
SUBSCRIBE for MWI
Sends SUBSCRIBE periodically (depends on “Register Expiration”
parameter) for message waiting indication. Default is No.
Send Anonymous
Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing
INVITE message to “anonymous”, blocking caller ID. Default is No.
Anonymous Call Rejection
Rejects incoming calls with anonymous caller ID with “486 Busy here”
message. Default is No.
Special Feature
Selects Soft switch vendors’ special requirements Example of vendors:
BroadSoft, CBCOM, RNK, Huawei, China Mobile, ZTE IMS, PhonePower,
TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS, TELEFONICA
SPAIN, IZZ, Oi_BR, ROSTELECOM. Default is Standard.
Enable Session Timer
Disable the session timer when this option is set to “No”. By default, this
option is enabled.
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Session Expiration
Enables SIP sessions to be periodically “refreshed” via a SIP request
(UPDATE, or re-INVITE). When the session interval expires, if there is no
refresh via an UPDATE or re-INVITE message, the session will be
terminated. Session Expiration is the time (in seconds) at which the session
is considered timed out, if no successful session refresh transaction occurs
beforehand. Valid range is 90-64800 seconds. Default is 180 seconds.
Min-SE
Defines Minimum session expiration (in seconds). Valid range is 90-64800 seconds. Default is 90 seconds.
Caller Request Timer
Uses session timer when making outbound calls if remote party supports it. Default is No.
Callee Request Timer
Uses session timer when receiving inbound calls with session timer request. Default is No.
Force Timer
Uses session timer even if the remote party does not support this feature.
Selecting “No” will enable session timer only when the remote party supports
it. Default is No.
To turn off Session Timer, select “No” for Caller and Callee Request Timer,
and Force Timer.
UAC Specify Refresher
Specifies which end will act as refresher for outgoing calls.
UAC: The handy tone acts as the refresher. UAS: Callee or proxy server act as the refresher.
Default is Omit.
UAS Specify Refresher
Specifies which end will act as refresher for incoming calls:
UAS: The handy tone acts as the refresher. UAC: Callee or proxy server act as the refresher.
Default is Omit.
Force INVITE
Uses INVITE message to refresh the session timer. Default is No.
Enable 100rel
Appends “100rel” attribute to the value of the required header of the initial
signaling messages. Default is No.
Add Auth Header on Initial
REGISTER
Adds “Authentication” header with blank “nonce” attribute in the initial SIP
REGISTER request. Default is No.
Conference URI
Allows users to manually configure the conference URL. The default is null.
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Use First Matching Vocoder
in 200OK SDP
Includes only the first matching vocoder in its 200OK response, otherwise it
will include all matching vocoders in same order received in INVITE. Default is No.
Preferred Vocoder
Configures vocoders in a preference list (up to 8 preferred vocoders) that
will be included with same order in SDP message. Vocoder types are G.711
A-/U-law, G.726-32, G.723, G.729, G722, iLBC and OPUS.
Voice Frames per TX
Transmits a specific number of voice frames per packet. Default is 2;
increases to 10/20/32/64 for G711/G726/G723/other codecs respectively.
G723 Rate
Operates at specified encoding rate for G.723 vocoder. Available encoding
rates are 6.3kbps or 5.3kbps. Default is 6.3kbps.
iLBC Frame Size
Specifies iLBC packet frame size (20ms or 30ms). Default is 20ms.
Disable OPUS Stereo in SDP
Disables OPUS stereo in SDP. If set to Yes, HT8XX will remove /2 from
offer. Default is No.
iLBC Payload type
Determines payload type for iLBC. Valid range is between 96 and 127. Default is 97.
OPUS Payload Type
Determines payload type for OPUS. Valid range is between 96 and 127.
Default is 123.
VAD
Allows detecting the absence of audio and conserves bandwidth by
preventing the transmission of "silent packets" over the network. Default is
No.
Symmetric RTP
Changes the destination to send RTP packets to the source IP address and
port of the inbound RTP packet last received by the device. Default is No.
Fax Mode
Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through
If using Pass-through mode, select preference codec as PCMU or PCMA.
Re-IINVITE after Fax Tone
Detection Mode
Permits the unit to send out the re-INVITE for T.38 or Fax Pass Through if a
fax tone is detected. Default is Enabled
Jitter Buffer Type
Selects jitter buffer type (Fixed or Adaptive) based on network conditions. Default is Adaptive.
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Jitter Buffer Length
High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders
can meet the high requirement.
Medium (initial 100ms, min 20ms, max 200ms). Low (initial 50ms, min 10ms, max 100ms).
SRTP Mode
Selects SRTP mode to use (“Disabled”, “Enabled but not forced”, or
“Enabled and forced”). Default is Disabled. It uses SDP Security Description to exchange key. Please refer to SDES: https://tools.ietf.org/html/rfc4568 SRTP: https://www.ietf.org/rfc/rfc3711.txt
Crypto Life Time
Adds crypto life time header to SRTP packets. Default is Yes.
SLIC Setting
Depends on standard phone type (and location).
Caller ID Scheme
Selects the caller ID scheme. For example: Bellcore/Telcordia, ETSI-FSK …
DTMF Caller ID
Defines the start and stop tones (Default, A, B, C, D or #) to delimit CID.
Disable Unknown Caller ID
This feature allows users to disable the analog phone’s caller ID when the
device receive a call with Anonymous, unavailable or unknown as FROM
USER and without Display-INFO. Default No
Replace Beginning '+' with 00
in Caller ID
When this feature is set to Yes, device will replace the “+” sign at the
beginning of a number in the FROM header with “00”. Default is No.
Polarity Reversal
Reverses the polarity upon call establishment and termination. Default is No.
Loop Current Disconnect
Allows the traditional PBX used with HT801/HT802 to apply this method for
signaling call termination. Method initiates short voltage drop on the line
when remote (VoIP) side disconnects an active call. Default is No.
Play busy/reorder tone before
Loop Current Disconnect
Allows user to configure if it will play busy/reorder tone before loop current
disconnect upon call fail. Default is No.
Loop Current Disconnect
Duration
Configures the duration of voltage drop described in topic above. HT80X
supports a duration range from 100 to 10000 ms. Default value is 200.
Enable Pulse Dialing
Allow users to enable Pulse Dialing option under FXS Port. Default is No.
Pulse Dialing Standard
Allows users to use Swedish pulse dialing standard and New Zealand Standard.
Default is General Standard.
Enable Hook Flash
Enables the FLASH button to be used for terminating calls. Default is Yes.
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Hook Flash Timing
Defines the time period when the cradle is pressed (Hook Flash) to simulate
FLASH. To prevent unwanted activation of the Flash/Hold and automatic
phone ring-back, adjust this time value. HT801/HT802 supports a range
from 40 to 2000 ms. Default values are 300 minimum and 1100 maximum.
On Hook Timing
Specifies the on-hook time for an on-hook event to be validated.
HT801/HT802 supports a range from 40 to 2000 ms. Default value is 400.
Gain
Adjusts the voice path volume.
Rx is a gain level for signals transmitted by FXS
Tx is a gain level for signals received by FXS.
Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume:
-6dB.
User can adjust volume of call using the Rx gain level parameter and the Tx
gain level parameter located on the FXS port configuration page. If call
volume is too low when using the FXS port (i.e. the ATA is at user site), adjust
volume using the Rx gain level parameter under the FXS port configuration
page. If voice volume is too low at the other end, user may increase the far
end volume using the Tx gain level parameter under the FXS port
configuration page.
Disable Line Echo Canceller
(LEC)
Disables the LEC will per call base. Recommended for FAX/Data calls. Default is No.
Disable Network Echo
Suppressor
Disables the NEC will per call base. Recommended for FAX/Data calls. Default is No.
Outgoing Call Duration Limit
Defines the call duration limit for the outgoing calls. Valid range is 0-180 minutes. Default is 0 (No limit).
Ring Frequency
Customizes ring frequency. Valid options: 20Hz – 25Hz. Default is 20Hz.
Enable High Ring Power
Configures a high ringing voltage output for the HT801/802.
RFC2833 Events Count
This feature allows users to customize the count of RFC2833 events. Valid range is 2 – 10. Default is 8.
RFC2833 End Events Count
This feature allows users to customize the count of RFC2833 end events.
Valid range is 2 – 10. Default is 3.
Distinctive Ring Tone
Customizes the Ring Tone 1 to 10 with associate caller ID: when selected,
if caller ID is configured, then the device will ONLY use this ring tone when
the incoming call is from the Caller ID. System Ring Tone is used for all other
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calls. When selected but no Caller ID is configured, the selected ring tone
will be used for all incoming calls using the FXS port. Distinctive ring tones
can be configured not only for matching a whole number, but also for
matching prefixes. In this case symbol “x+” will be used.
For example: If configured as 617x+, Ring Tone 1 will be used in case of
call arrived from the area code 617. Any other incoming call will ring using
cadence defined in parameter System Ring Cadence located under
Advanced Settings Configuration page.
Note: If server supports Alert-Info header and standard ring tone set
(Bellcore) or distinctive ring tone 1-10 is specified, then the ring tone in the
Alert-Info header from server will be used. Bellcore rings and tones are
independent from custom ring tones. The custom ring tones can also be
specified by alert-info header, for example
Alert-Info: ;info=ring5
Ring tones
Configures the ring tone cadence preferences. User has 10 choices.
(Syntax: c=on1/off1-on2/off2-on3/off3;)
Note: The configuration, completed in Distinctive Ring Tones block in the
same page, applies to ring tones cadences configured here. Default is c=2000/4000;
Distinctive Call Waiting Tone
Customizes the Call Waiting Tone 1 to 10 with associate caller ID: when
selected, if caller ID is configured, then the device will ONLY use this call
waiting tone when the incoming call waiting is from the Caller ID. When
selected but no Caller ID is configured, the selected call waiting tone will be
used for all incoming waiting calls using the FXS port. Distinctive Call
Waiting Tones can be configured not only for matching a whole number, but
also for matching prefixes. In this case symbol “x+” will be used.
For example: If configured as 617x+, Call Waiting Tone 1 will be used in
case of waiting call arrived from the area code 617. Any other incoming call
waiting will be using cadence defined in parameter Call Waiting Tone located
under Advanced Settings Configuration page.
Call Waiting Tones
This feature allows user to customize call waiting tone. User has 10 choices.
Syntax: f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];
(Frequencies are in (300, 3400) Hz and cadence on and off are in (0, 64000)
ms)
Note: The configuration, completed in Distinctive Call Waiting Tones block
in the same page, applies to call waiting cadences configured here. Default is f1=440@-13,c=300/10000;
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Call Features Settings
Enable Call Features
When enabled, Do Not Disturb, Call Forward and other call features can be
used via the local feature codes on the phone. Otherwise, the ITSP feature
codes will be used. Enable All will override all individual features enable
setting. Default is Yes.
Reset Call Features
Allows users to reset all call features configuration. Default is No
SRTP Feature
Allow users to customize the SRTP feature codes. Default is Yes
- Enable SRTP: Default is 16
- Disable SRTP: Default is 17
SRTP per call Feature
- Enable SRTP per call: Default is 18
- Disable SRTP per call: Default is 19
CID Feature
Allow users to customize the CID feature codes. Default is Yes
- Enable CID: Default is 31
- Disable CID: Default is 30
CID per call Feature
- Enable CID per call: Default is 82
- Disable CID per call: Default is 67
Direct IP Calling Feature
Allow users to customize the Direct IP feature code. Default is Yes
- Direct IP Calling: Default is 47
CW Feature
Allow users to customize the CW feature codes. Default is Yes
- Enable CW: Default is 51
- Disable CW: Default is 50
CW per call Feature
- Enable CW per call: Default is 71
- Disable CW per call: Default is 70
Call Return Feature
Allow users to customize the Call Return feature code. Default is Yes
- Call return: Default is 69
Unconditional Forward
Feature
Allow users to customize the Unconditional Forward feature codes. Default is Yes
- Enable Unconditional Forward: Default is 72
- Disable Unconditional Forward: Default is 73
Busy Forward Feature
Allow users to customize the Busy Forward feature codes. Default is Yes
- Enable Busy Forward: Default is 90
- Disable Busy Forward: Default is 91
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Delayed Forward Feature
Allow users to customize the Delayed Forward feature codes. Default is Yes
- Enable Delayed Forward: Default is 92
- Disable Delayed Forward: Default is 93
Paging Feature
Allow users to customize the Paging feature code. Default is Yes
- Paging: Default is 74
DND Feature
Allow users to customize the CW feature codes. Default is Yes
- Enable DND: Default is 78
- Disable DND: Default is 79
Blind Transfer Feature
Allow users to customize the Blind Transfer feature code. Default is Yes
- Enable Blind Transfer: Default is 87
Disable LEC per call Feature
Default is Yes
- Disable LEC per call: Default is 03
Disable Bellcore Style 3-Way
Conference
Gives the users the possibility of making conference calls by pressing
“Flash” key, when it’s enabled by dialing *23 +second callee number. Default
is No
Star Code 3WC Feature
Default is Yes
- Star Code 3WC: Default is 23
Forced Codec Feature
Allow users to customize the Forced Codec feature code. Default is Yes
- Forced Codec: Default is 02
PCMU Codec Feature
Default is Yes
- PCMU Codec: Default is 7110
PCMA Codec Feature
Default is Yes
- PCMA Codec: Default is 7111
G723 Codec Feature
Default is Yes
- G723 Codec: Default is 723
G729 Codec Feature
Default is Yes
- G729 Codec: Default is 729
iLBC Codec Feature
Default is Yes
- iLBC Codec: Default is 7201
G722 Codec Feature
Default is Yes
- G722 Codec: Default is 722
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NAT Settings

If you plan to keep the handy tone within a private network behind a firewall, we recommend using STUN
Server. The following three settings are useful in the STUN Server scenario:
1. STUN Server (under advanced settings webpage) enter a STUN server IP (or FQDN) that you may
have or look up a free public STUN server on the internet and enter it on this field. If using public IP,
keep this field blank.
2. Use random SIP/RTP ports (under advanced settings webpage), this setting depends on your network settings. Generally, if you have multiple IP devices under the same network, it should be set to Yes. If using a public IP address, set this parameter to No.
3. NAT traversal (under FXS web page): Set to “Yes” when gateway is behind firewall on a private network.

DTMF Methods

The HT801/HT802 support the following DTMF mode:
DTMF in-audio DTMF via RTP (RFC2833) DTMF via SIP INFO
Set priority of DTMF methods according to your preference. This setting should be based on your server
DTMF setting.

Preferred Vocoder (Codec)

The HT801/HT802 support following voice codecs. On FXS ports pages, choose the order of your favorite
codecs:
PCMU/A (or G711µ/a) G729 A/B G723.1 G726 iLBC OPUS G722

Configuring HT801/HT802 Through Voice Prompts

As mentioned previously, The HT801/HT802 has a built-in voice prompt menu for simple device
configuration.
Please refer to “Understanding HT801/HT802 Interactive Voice Prompt Response Menu” for more
information about IVR and how to access its menu.
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DHCP MODE
Select voice menu option 01 to enable HT801/HT802to use DHCP.
STATIC IP MODE
Select voice menu option 01 to enable HT801/HT802 to use STATIC IP mode, then use option 02,
03, 04, 05 to set up IP address, Subnet Mask, Gateway and DNS server respectively.
FIRMWARE SERVER IP ADDRESS
Select voice menu option 13 to configure the IP address of the firmware server.
CONFIGURATION SERVER IP ADDRESS
Select voice menu option 14 to configure the IP address of the configuration server.
UPGRADE PROTOCOL
Select the menu option 15 to choose firmware and configuration upgrade protocol between TFTP,
HTTP, HTTPS, FTP and FTPS. Default is HTTPS.
FIRMWARE UPGRADE MODE
Select voice menu option 17 to choose firmware upgrade mode among the following three options:
1) Always check, 2) check when pre/suffix changes, and 3) never upgrade.

Register a SIP Account

The HT801/HT802 supports 2 FXS ports which can be configured with 2 SIP accounts. Please refer to the
following steps in order to register your accounts via web user interface:
1. Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
2. Enter your admin password (default: admin).
3. Press Login to access your settings.
4. Go to FXS Port (1 or 2) pages.
5. In FXS Port tab, set the following:
a. Account Active to Yes. b. Primary SIP Server field with your SIP server IP address or FQDN. c. Failover SIP Server with your Failover SIP Server IP address or FQDN. Leave empty if not
available.
d. Prefer Primary SIP Server to No or Yes depending on your configuration. Set to No if no Failover
SIP Server is defined. If “Yes”, account will register to Primary SIP Server when failover registration
expires.
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e. Outbound Proxy: Set your Outbound Proxy IP Address or FQDN. Leave empty if not available. f. SIP User ID: User account information, provided by VoIP service provider (ITSP). Usually in the
form of digit similar to phone number or actually a phone number.
g. Authenticate ID: SIP service subscriber’s Authenticate ID used for authentication. Can be identical
to or different from SIP User ID.
h. Authenticate Password: SIP service subscriber’s account password to register to SIP server of
ITSP. For security reasons, the password will field will be shown as empty.
i. Name: Any name to identify this specific user.
For more information, related to above options please refer to FXS Port Settings.
6. Press Apply at the bottom of the page to save your configuration.
Figure 11: SIP Accounts Settings
After applying your configuration, your account will register to your SIP Server, you can verify if it has been
correctly registered with your SIP server or from your HT801/HT802 web interface under StatusPort
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StatusRegistration (If it displays Registered, it means that your account is fully registered, otherwise it will display Not Registered so in this case you must double check the settings or contact your provider).
Figure 12: Account Status
Note: When all the FXS ports are registered (for HT802), the simultaneous ring will have one second delay
between each ring on each phone.

Call Features

The HT801/HT802 support all the traditional and advanced telephony features.
Table 11: HT801/HT802 Call Features
Key
Call features
*02
Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729
(G729), *027201 (iLBC). *02722 (G722).
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*03
Disable LEC (per call) Dial “*03” +” number”. No dial tone is played in the middle.
*16
Enable SRTP .
*17
Disable SRTP.
*30
Block Caller ID (for all subsequent calls).
*31
Send Caller ID (for all subsequent calls).
*47
Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle.
*50
Disable Call Waiting (for all subsequent calls).
*51
Enable Call Waiting (for all subsequent calls).
*67
Block Caller ID (per call). Dial “*67” +” number”. No dial tone is played in the middle.
*82
Send Caller ID (per call). Dial “*82” +” number”. No dial tone is played in the middle.
*69
Call Return Service: Dial *69 and the phone will dial the last incoming phone number
received.
*70
Disable Call Waiting (per call). Dial “*70” +” number”. No dial tone is played in the middle.
*71
Enable Call Waiting (per call). Dial “*71” +” number”. No dial tone is played in the middle.
*72
Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait
for dial tone and hang up. (Dial tone indicates successful forward)
*73
Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait
for dial tone, then hang up.
*74
Enable Paging Call: Dial “*74” and then the destination phone number you want to page.
*78
Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.
*79
Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.
*87
Blind Transfer.
*90
Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial
tone then hang up.
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*91
Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then
hang up.
*92
Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for
dial tone then hang up.
*93
Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone,
then hang up.
Flash/
Hook
Toggles between active call and incoming call (call waiting tone). If not in conversation,
flash/hook will switch to a new channel for a new call.
#
Pressing pound sign will serve as Re-Dial key.

Rebooting HT801/HT802 from Remote

Press the “Reboot” button at the bottom of the configuration menu to reboot the ATA remotely. The web
browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in
again.
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UPGRADING AND PROVISIONING

The HT801/HT802 can be upgraded via TFTP/HTTP/HTTPS/FTP/FTPS by configuring the URL/IP Address
for the TFTP/HTTP/HTTPS/FTP/FTPS server and selecting a download method. Configure a valid URL for
TFTP or FTP/FTPS or HTTP/HTTPS (default is HTTPS); the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com fw.ipvideotalk.com/gs

Firmware Upgrade procedure

Please follow below steps to upgrade the firmware version of your HT801/HT802:
1. Access your HT801/HT802 UI by entering its IP address in your favorite browser.
2. Enter your admin password (default: admin).
3. Press Login to access your settings.
4. Go to Advanced SettingsFirmware Upgrade and Provisioning page and enter the IP address
or the FQDN for the upgrade server in "Firmware Server Path" field and choose to upgrade via
TFTP, FTP/FTPS or HTTP/HTTPS.
5. Make sure to check “Always Check for New Firmware”.
6. Update the change by clicking the " Apply" button at the bottom of the page. Then "Reboot" or
power cycle the HT802 to update the new firmware.
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Figure 13: Firmware Upgrade Page

Upgrading via Local Directory

1. Download the firmware file from Grandstream web site;
2. Unzip it and copy the file in to a folder in your PC;
3. From the HT801/HT802 web interface (Advanced Settings page) you can browse your hard drive and select the folder you previously saved the file (HT802fw.bin);
4. Click “Upload Firmware” and wait few minutes until the new program is loaded.
Note: Always check the status page to see that the program version has changed.

Upgrading via Local TFTP/HTTP/HTTPS/FTP/FTPS Servers

For users that would like to use remote upgrading without a local TFTP/HTTP/HTTPS/FTP/FTPS server,
Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software
upgrades for their devices via this server. Please refer to the webpage:
http://www.grandstream.com/support/firmware
Alternatively, users can download, for example, a free TFTP or HTTP server and conduct a local firmware upgrade. A free window version TFTP server is available for download from:
http://www.solarwinds.com/products/freetools/free_tftp_server.aspx http://tftpd32.jounin.net/.
Instructions for local firmware upgrade via TFTP:
1. Unzip the firmware files and put all of them in the root directory of the TFTP server.
2. Connect the PC running the TFTP server and the phone to the same LAN segment.
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3. Launch the TFTP server and go to the File menuConfigureSecurity to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade.
4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface.
5. Configure the Firmware Server Path to the IP address of the PC.
6. Save and Apply the changes and reboot the HT801/HT802.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.

Firmware and Configuration File Prefix and Postfix

Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and
Postfix. This makes it the possible to store all of the firmware with different version in one single directory.
Similarly, Config File Prefix and Postfix allows device to download the configuration file with the matching
Prefix and Postfix. Thus, multiple configuration files for the same device can be stored in one directory.
In addition, when the field “Check New Firmware only when F/W pre/suffix changes” is set to “Yes”, the
device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.

Managing Firmware and Configuration File Download

When “Automatic Upgrade” is set “Yes, every” the auto check will be done in the minute specified in this
field. If set to “daily at hour (0-23)”, Service Provider can use P193 (Auto Check Interval) to have the
devices do a daily check at the hour set in this field with either Firmware Server or Config Server. If set to
weekly on day (0-6)” the auto check will be done on the day specified in this field. This allows the device
periodically to check if there are any new changes need to be taken on a scheduled time. By defining
different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration
File download in minutes to reduce the Firmware or Provisioning Server load at any given time.

Configuration File Download

Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File (binary or XML) through TFTP, FTP/FTPS or HTTP/HTTPS. The Config Server Path is the TFTP, FTP/FTPS or HTTP/HTTPS server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The Config Server Path can be the same or different from the Firmware Server Path. A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with the "New Password" in the Web GUIMaintenanceWeb/SSH Access pageAdmin Password. For a detailed parameter list, please refer to the corresponding firmware release configuration template.
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When the HT801/HT802 boots up or reboots, it will send a request to download a file named
"cfgxxxxxxxxxxxx” followed by a configuration XML file named "cfgxxxxxxxxxxxx.xml", where
"xxxxxxxxxxxx" is the MAC address of the phone, i.e., "cfg000b820102ab" and "cfg000b820102ab.xml". If the download of "cfgxxxxxxxxxxxx.xml" file is not successful, the provision program will download a generic cfg.xml file and then download cfg<Model>.xml. The configuration file name should be in lower case letters.
HT801/HT802 supports DHCP option 67 allowing to provide custom name for the provisioning file. If DHCP option 67 is used, the following file download sequence will be applied: Step 1: cfg<MAC> Step 2: <option 67 bootfile> cfg<MAC>.xml cfg.xml cfg<Model>.xml
Notes:
1. Only XML or binary config file formats are accepted.
2. The MAC header in XML config file should be the device MAC or needs to be removed
completely.
Figure 14: XML Config File - MAC Header
For more details on XML provisioning, please refer to:
http://www.grandstream.com/sites/default/files/Resources/gs_provisioning_guide.pdf
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RESTORE FACTORY DEFAULT SETTINGS

Warning:
Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup
or print all the settings before you restore to the factory default settings. Grandstream is not responsible for
restoring lost parameters and cannot connect your device to your VoIP service provider.
There are three (3) methods for resetting your unit:

Using the Reset Button

To reset default factory settings using the reset button please follow the steps above:
1. Unplug the Ethernet cable.
2. Locate the reset hole on the back panel of your HT801/HT802.
3. Insert a pin in this hole, and press for about 7 seconds.
4. Take out the pin. All unit settings are restored to factory settings.

Using the IVR Command

Reset default factory settings using the IVR prompt:
1. Dial “***” for voice prompt.
2. Enter “99” and wait for “reset” voice prompt.
3. Enter the encoded MAC address (Look below on how to encode MAC address).
4. Wait 15 seconds and device will automatically reboot and restore factory settings.
Encode the MAC Address
1. Locate the MAC address of the device. It is the 12-digit HEX number on the bottom of the unit.
2. Key in the MAC address. Use the following mapping:
Table 12: MAC Address Key Mapping
Key
Mapping
0-9
0-9 A 22 (press the “2” key twice, “A” will show on the LCD)
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B
222 C 2222
D
33 (press the “3” key twice, “D” will show on the LCD)
E
333
F
3333
For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”.

Reset from Web Interface (Reset Type)

1. Access your HT801/HT802 UI by entering its IP address in your favorite browser.
2. Enter your admin password (default: admin).
3. Press Login to access your settings.
4. Go to Basic SettingsReset Type.
5. Press Reset button (after selecting the reset type).
Full Reset: This will make a full reset. ISP Data: This will reset only the basic settings, like IP mode, PPPoE and Web port. VOIP Data: This will reset only the data related with a service provider like SIP server, sip user ID,
provisioning and others.
Note:
- Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.
- If the HT801/HT802 was previously locked by your local service provider, pressing the RESET
button will only restart the unit. The device will not return to factory default settings.

Reset using SIP NOTIFY

1. Access your HT801/HT802 UI by entering its IP address in your favorite browser.
2. Go to FXS Port.
3. Set “Allow SIP Factory Reset” to “Yes”.
4. Once a SIP NOTIFY with “event: reset” is received, the phone will perform factory reset.
Note: Received SIP NOTIFY will be first challenged for authentication purpose before taking factory
reset action. The authentication can be done either using admin credentials (if no SIP account is
configured) or using SIP account credentials.
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EXPERIENCING HT801/HT802

Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware
releases, additional features, FAQs, documentation and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream analogue telephone adapter, it will be sure to bring
convenience to both your business and personal life.
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