GRANDSTREAM HT801 V2 Analog Telephone Adapters User Guide

HT801 V2	HT802 V2
1 FXS port	2 FXS ports
Single 10/100Mbps port	Single 10/100Mbps port
Support 3-way voice conferencing	Support 3-way voice conferencing
Wide range of caller ID formats	Wide range of caller ID formats

Features

Caller ID display or block
Call waiting

Flash
Blind or attended transfer

Call forward
Call hold

Do not disturb
3-way conference

Technical Specifications

The HT801 V2 and HT802 V2 support various protocols/standards,
voice codecs, telephony features, languages, and
upgrade/provisioning settings.

Interfaces:

Telephone Interfaces: RJ11 FXS port(s)
Network Interface: One (HT801) or Two (HT802)
RJ11 FXS ports

Voice, Fax, Modem:

Telephony Features: Caller ID display or
block, call waiting, 3-way conference, and more

Voice Codecs: G.711, G.722, G.729A/B, and
more
Fax over IP: T.38 compliant Group 3 Fax
Relay

Signaling:

Network Protocols: TCP/IP, UDP, SIP, SRTP, and
more

DTMF Methods: In-audio, RFC2833, SIP INFO

Provisioning and Control:

Syslog: Support for management
QoS: Layer 2 and Layer 3 support

Product Usage Instructions

Installation

Connect the RJ11 FXS port(s) to your analog telephone(s).
Connect the Ethernet port to your network.

Power on the device.

Configuration

You can configure the device using HTTP, HTTPS, SSH, TFTP, or
TR-069 protocols.

Usage

Utilize the various telephony features such as caller ID
display, call waiting, call transfer, and conferencing according to
your needs.

FAQs

Q: How can I reset the device to factory settings?

A: You can reset the device by pressing the Factory Reset
Button.

Q: What network protocols are supported by the HT801 V2 and
HT802 V2?

A: The devices support TCP/IP, UDP, SIP, SRTP, and more.

Q: Can I use the HT801 V2/HT802 V2 with Grandstream’s UCM
series of IP PBXs?

A: Yes, you can use them with Grandstream’s UCM series for Zero
Configuration provisioning.

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Grandstream Networks, Inc.
HT80x V2 – Administration Guide

INTRODUCTION
The HT801 V2 and HT802 V2 are analog telephone adapters (ATAs) that allow users to connect analog phones and faxes with a VoIP (Voice over IP) system to create high-quality and manageable IP telephony solutions for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection, and auto provisioning options enable users to take advantage of VoIP on analog phones. It also allows service providers to offer high-quality IP service to their market. The HT80x V2 series are ideal ATAs for individual use as well as commercial IP voice deployments worldwide.
This manual will help you learn how to operate and manage your HT801 V2/HT802 V2 analog telephone adapters and leverage their many features including but not limited to simple and quick installation, 3-way conferencing, direct IP Calling, and automated provisioning.
PRODUCT OVERVIEW
Feature Highlights
The following table contains major features of HT801 V2 and HT802 V2:

HT801 V2 HT802 V2

1 FXS port on HT801 V2 and 2 FXS ports on HT802 V2. Single 10/100Mbps port on both models. Support 3-way voice conferencing. Wide range of caller ID formats. Advanced telephony features, including Caller ID display
or block, blind and attended call transfer, call forward, call-waiting, call hold, do not disturb, flash and more. T.38 Fax for creating Fax-over-IP. 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.

HT801 V2/HT802 V2 Features at a Glance

HT801 V2/HT802 V2 Technical Specifications
The following tables summarize all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for the HT801 V2 and the HT802 V2.

Interfaces
Telephone Interfaces
Network Interface

HT801 V2
One (1) RJ11 FXS port One (1) 10/100Mbps auto-sensing Ethernet port (RJ45)

HT802 V2 Two (2) RJ11 FXS ports

LED Indicators

POWER, NET, PHONE

POWER, NET, PHONE1, PHONE2

Factory Reset Button

Yes

Voice, Fax, Modem

Telephony Features

Caller ID display or block, call waiting, flash, blind or attended transfer, forward, hold, do not disturb, 3-way conference

Voice Codecs

G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726-32, 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

Dial Methods DTMF, Pulse

Disconnect Methods

Busy Tone, Polarity Reversal/Wink, Loop Current

Signaling

Network Protocols

TCP/IP/UDP, RTP/RTCP (RFC1889, 1890), HTTP/HTTPS, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SSH, Telnet, STUN (RFC3489, 5389), SIP (RFC3261), SIP over TCP/TLS, SRTP, SNMP, TR-069, IMS/3GPP, IPoE

DTMF Methods

In-audio, RFC2833 and/or SIP INFO

Provisioning and Control

HTTP, HTTPS, SSH, TFTP, TR-069, secure and automated provisioning using AES encryption, syslog

QoS

Layer 2 (802.1Q VLAN, SIP/RTP 802.1p) and Layer 3 (ToS, DiffServ, MPLS)

Security

Media

SRTP

Control

TLS/SIPS/HTTPS

Management Syslog support, SSH, Telnet remote management using web browser

Physical

Universal Power Supply

Input: 100-240VAC, 50-60Hz. Output: 5.0VDC/1.0A

Environment al

Operational: 32° 104°F or 0º 40ºC. Storage: 14° 140°F or -10º 60ºC. Humidity: 10 90% Non-condensing

Dimensions and Weight

· 100mm x 100mm x 29.5mm · 102g (without package)

· 100mm x 100mm x 29.5mm · 114g (without package).

Compliance

FCC: Part15B CE: EN55032, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, RCM: AS/NZS CISPR22, AS/NZS60950.1, S003 K.21

FCC 15B, AS/NZS CISPR22, AS/NZS60950, EN55022, EN55024, EN60950, EN61000-3-2, EN61000-3-3, UL (Power supply) K.21

HT801 V2/HT802 V2 Technical Specifications

GETTING STARTED
This section provides basic installation instructions including the list of the packaging contents and information to connect the HT801 V2/HT802 V2.

Equipment Packaging
The HT801 V2 ATA package contains:

The HT802 V2 ATA package contains:

HT801 V2 Package Contents

HT802 V2 Package Contents Note: Check the package before installation. If you find anything missing, contact your system administrator.
HT801 V2/HT802 V2 Ports Description
The following figure describes the different ports on the back panel of the HT801 V2.
HT801 V2 Ports The following figure describes the different ports on the back panel of the HT802 V2.

HT802 V2 Ports

Phone port for HT801 V2 Phone 1 & 2 ports for HT802 V2
NET port Micro USB DC Power
Reset

Connects the analog phones/fax machines to the phone adapter using an RJ-11 telephone cable. Connects the phone adapter to your router or gateway using an Ethernet RJ45 network cable. Connects the phone adapter to PSU (5V 1A). Factory reset button. Press for 7 seconds to reset factory default settings.

HT801 V2/HT802 V2 Ports Definition

Connecting HT801 V2/HT802 V2
In order to connect your HT801 V2 or HT802 V2, please follow the steps mentioned below:
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 NET port of the HT801 V2/HT802 V2 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 V2/HT802 V2 and connect it to a wall outlet.
Power, NET, and Phone LEDs will be solidly lit once the HT801 V2/HT802 V2 is ready for use.
HT801 V2/HT802 V2 LEDs Pattern
There are 3 LED indications on HT801 V2 and 4 on HT802 V2 as shown in the below figure:

HT801 V2/HT802 V2 LEDs Indications

LED Indication Power LED NET LED
Phone LED for HT801 V2 Phone LED 1&2 for HT802 V2

Status
The Power LED lights up when the HT801 V2/HT802 V2 is powered on, and it flashes when the unit is booting up
The NET LED lights up when the HT801 V2/HT802 V2 is connected to a network through the Ethernet port, and it flashes when there is data being sent or received.
The phone LEDs 1 & 2 indicate the status of the respective FXS port-phone on the back panel OFF – Unregistered ON (Solid Blue) – Registered and Available Blinking every second – Off-Hook / Busy Slow blinking – FXS LEDs indicates voicemail

HT801 V2/HT802 V2 LEDs Pattern Description

CONFIGURATION GUIDE
There are two ways to configure HT801 V2/HT802 V2 :
The IVR voice prompt menu. The embedded Web GUI on the HT801 V2/HT802 V2 using the PC’s web browser.
Obtain HT80x V2 IP Address via Connected Analog Phone
HT801 V2/HT802 V2 is by default configured to obtain the IP address from the DHCP server where the unit is located. To know which IP address is assigned to your HT801 V2/HT802 V2, you should access the IVR (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 the phone for the HT801 V2 or phone 1 or phone 2 ports of your HT802 V2. 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 HT80x V2 Interactive Voice Prompt Response Menu
The HT801 V2/HT802 V2 has 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 V2/HT802 V2.
To access the IVR Menu, users need to dial “***”.

Menu Main Menu
01
02 03 04 05
07
10 13 14

Voice Prompt “Enter a Menu Option”
“DHCP Mode”, “Static IP Mode” “PPPoE Mode”
“IP Address ” + IP address “Subnet ” + IP address “Gateway ” + IP address “DNS Server ” + IP address
Preferred Vocoder
“MAC Address” Firmware Server IP Address Configuration Server IP Address

Options
Press “*” for the next menu option Press “#” to return to the main menu Enter 01-05, 07,10, 12-17, 20, 47 or 99 menu options
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. If using “PPPoE Mode”, configure PPPoE Username and Password from web GUI to get IP from your ISP.
The current WAN IP address is announced If using “Static IP Mode”, enter 12-digit new IP address. You need to reboot your HT801 V2/HT802 V2 for the new IP address to take Effect.
Same as menu 02
Same as menu 02
Same as menu 02
Press “9” to move to the next selection in the list: PCM U / PCM A iLBC G-726 G-723 G-729 OPUS G722
Announces the MAC address of the unit. Note: The device has two MAC addresses. One for the WAN port and one for the LAN port. The device MAC address announced is the address of LAN port.
Announces current Firmware Server IP address. Enter 12-digit new IP address.
Announces current Config Server Path IP address. Enter 12-digit new IP address.

Upgrade protocol for firmware and

Upgrade Protocol

configuration update. Press “9” to toggle

between TFTP/HTTP/HTTP /FTP/FTPS

Firmware Version

Announces Firmware version information.

Firmware upgrade mode. Press “9” to toggle among the following three options:

Firmware Upgrade

Always check

Check when pre/suffix changes

Never upgrade

Certificate Type

Annouces certificate information.

“Direct IP Calling”

Enter the target IP address to make a direct IP call, after dial tone.

Voice Mail

Access to your voice mails messages.

Press “9” to reboot the device

Enter MAC address to restore factory

“RESET”

default setting

(See Restore Factory Default Setting section)

“Invalid Entry”

Automatically returns to main menu

“Device not registered” Five success tips when using the voice prompt

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

“*” 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 the menu option and 12 digits for the 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 keyed in like 192168000026. No decimal is needed).
Key entry cannot be deleted but the phone may prompt an error once it is detected.
Dial *98 to announce the extension number of the port.

Configuration via Web Browser
The HT801 V2/HT802 V2 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
Please follow the steps mentioned below to access the HT801 V2/HT802 V2 Web UI:
1. Connect the computer to the same network as your HT801 V2/HT802 V2. 2. Make sure the HT801 V2/HT802 V2 is booted up. 3. You may check your HT801 V2/HT802 V2 IP address using the IVR on the connected phone. (To do this, please refer to
this section [Obtain HT80x V2 IP Address via Connected Analog Phone])

4. Open the Web browser on your computer. 5. Enter the HT801 V2/HT802 V2’s IP address in the address bar of the browser. 6. Enter “admin” under the username as well as the administrator’s password to access the Web Configuration Menu (the
default password can be found on the sticker located at the back of the unit).
Note: The computer must be connected to the same sub-network as the HT801 V2/HT802 V2. This can be easily done by connecting the computer to the same hub or switch as the HT801 V2/HT802 V2.

Web UI Access Level Management
There are three access levels for the login page:

User Level End User Level Administrator Level Viewer Level

User user admin viewer

Password

Web Pages Allowed

123

View all pages but can only modify basic settings

random password located at the back of the unit

Browse all pages and modify all settings

viewer

View all pages but no changes allowed.

Notes:
The password is case-sensitive and must contain 8-20 characters, at least one number, one uppercase, and one lowercase letter. When changing any settings, always submit them by pressing the Update or Apply button at the bottom of the page. Some changes require a reboot of the HT80x V2 unit such as FXS Port settings.
By default, user and viewer access levels are disabled. In order to enable them please access System Settings Security Settings Web Access.
After initial login using the default admin/user password, the user will be prompted to change the password immediately.

Saving the Configuration Changes
After modifying any configuration parameters, users can save the changes by clicking on the Save button. Once the configuration is saved, an Apply button will appear at the top of the page to allow users to apply the changes. Users can also directly click on the Save and Apply button for the configuration changes to be saved and applied.
Note: We recommend rebooting or powering cycling the phone after applying all the changes.
Changing Admin/User/Viewer Level Password
To change the password for any of the access levels, please refer to the following steps: 1. Access your HT801 V2/HT802 V2 Web UI by entering its IP address on your browser.

2. Enter the access level (admin, user, or viewer) under your username and your default password. (please refer to this section [Web UI Access Level Management] for the default password)
3. Press Login to access your configuration settings and navigate to System Settings Security Settings User Info Management Password.
4. Enter the new admin/user/viewer password and confirm it (The new Password must contain 8-20 characters, at least one number, one uppercase, and a lowercase letter).
5. Save and apply the new admin/user/viewer password.
Changing the HTTP/HTTPS Web Port
1. Access your HT801 V2/HT802 V2 web UI by entering its IP address on your browser. 2. Enter “admin” under the username, then type your admin password.
3. Press Login to access your settings and navigate to System Settings Security Settings Web/SSH Access
Web Access. 4. Choose the desired Web Access Mode and the new HTTP/HTTPS port. The valid range is [1-65535]. 5. Save and apply the new port number.

Web Configuration Pages Definitions
The following sections describe the configuration options on the HT801 V2/HT802 V2 Web UI.

Status Page Definitions System Info

Product Model Serial Number Hardware Version Part Number

Displays the Product model: HT801V2 or HT802V2. Displays the device’s serial number. Displays the hardware version. Displays the part number.

Software Version
Software Status Mem System Up Time System Current Time CPU Load Provision

Program: Displays the current software version running on the device. Bootloader: Displays the bootloader version. Core: Displays the Core version. Base: Displays the Base version. CPE: Displays the Customer Premises Equipment version.
Displays whether the software is running correctly on the device.
Displays the memory usage.
Displays the duration of the system up time.
Displays the HT8xx current time.
Displays the processor load.
Displays the last status of provisioning.

System Info
Network Status
MAC Address IPv4 Address IPv6 Address VPN IPv4 Address VPN IPv6 Address Network Cable Status PPPoE Link Up NAT Certificate Type
Port Status
Port Status Port Hook SIP User ID Registration Sip Port Port Options Port DND Forward Busy Forward Delayed Forward CID

Downloads the system info.
Displays the device’s MAC address. Displays the assigned IPv4 address. Displays the assigned IPv6 address. Displays the connected VPN IPv4 address, if available. Displays the connected VPN IPv4 address, if available. Displays the status and speed of the NET port, it is shown as UP if connected, and DOWN if disconnected. Indicates active connectivity through PPPoE. Displays the type of NAT used. Displays the certificate type.
Displays the analog port : FXS1, FXS2 (for HT802 V2). Shows the status of the port, On Hook, Off Hook, Not connected… Displays the SIP user ID registred on the port. Displays wether the port is registered or not. Displays the SIP port used by the analog interface.
Displays the FXS port. Displays whether the connected analog phone is on DND mode or not. Indicates the forwarding number. Indicates the busy forwarding number. Indicates the delayed forwarding number. Indicates the Caller ID.

Call Waiting SRTP

Indicates if Call waiting is enabled. Indicates if Secured RTP is enabled.

System Settings Page Definitions

Basic Settings

Enable Voice Prompt

Enable/disables Voice Prompt, it is disabled if set to No. Enabled by default.

Enable Direct IP Call

Enable/diasbles direct IP calling, it is disabled if set to No. Enabled by default.

Blocklist For Incoming Calls

Blocks calls from specific numbers. Use “,” to separate numbers.

Lock Keypad Update

The configuration update via keypad is disabled if set to Yes. Default value is “No”.

Play Busy Tone When Account is unregistered

If set to Yes, busy tone will be played when user goes offhook from an unregistered account. Default value is “No”.

DHCP Option 17 Enterprise Number

Fill in the DHCP Option 17 enterprise number, the default is 3561. Valid range is 0-65535.

ATA Settings

STUN Server

Defines the IP address or domain name of the STUN server. Only non-symmetric NAT routers work with STUN.

Keep-Alive Interval

Specifies how often the device sends a blank UDP packet to the SIP server in order to keep the “ping hole” on the NAT router to open. Default is 20 seconds. Range is 10-160 seconds.

Use STUN to detect network connectivity

Enable/Disable Using STUN keep-alive to detect WAN side network problems Disable by default.

Time and Language

Time Zone NTP Server
Secondary NTP Server Allow DHCP Option 42 to override NTP server Time Zone Self-Defined Time Zone

This parameter sets the IP address of the NTP server. The device will obtain the date and time from the server.
This feature allows users to configure the secondary NTP server, by default, no secondary NTP Server is configured.
If DHCP Option 42 is enabled, the NTP server can be changed. Enabled by default.
Selects the time zone where the phone is located and controls the date/time display, Note: you can set the timezone to be the self defined time.
Allows users to define their own time zone when time zone is set to “Using self-defined Time Zone”.

The syntax is: std offset dst [offset], start [/time], end [/time] Default is set to: MTZ+6MDT+5,M3.2.0,M11.1.0

MTZ+6MDT+5

This indicates a time zone with 6 hours offset with 1 hour ahead (when daylight saving) which is U.S central time. If it is positive (+) if the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is east.

M4.1.0,M11.1.0 The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec) The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday…) The 3rd number indicates weekday: 0,1,2,..,6 (for Sun, Mon, Tues, … ,Sat) Therefore, this example is the DST which starts from the Second Sunday of March to the 1st Sunday of November.

Allow DHCP server to set Time Zone

Allows the local server’s DHCP option 2 to override the phone’s time zone setting.

Local Time

When Sync PC is pressed, the HT8xx will synchronise the date and time from the connected PC.

Language

IVR Language

Selects IVR voice prompt language type, the supported languages are: English, Chinese, Russian, Spanish.

Ringtone

System Ring Cadence
Prompt Tone Access Code
CPT Settings:
Dial Tone Ringback Tone Busy Tone Reorder Tone Confirmation Tone Call Waiting Tone Wait for Dial-Tone Conference Party Hangup Tone Special Proceed Indication Tone Special Condition Tone

Cadence on+off value range (0, 16000) milliseconds.
The key pattern to get Prompt Tone. Maximum 20 digits. No default value provided
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: fxsch1:f1=350@-17,f2=440@-17,c=0/0;fxoch1-4:f1=350@-17,f2=440@-17,c=0/0;
Syntax: fxsch1:f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];<br>fxoch xy:f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]] (Frequencies are in (10, 4000) Hz and cadence on and off are in (0, 64000) ms)

Security Settings
Web/SSH Access Web Access

Web Access Mode HTTP Web Port HTTPS Web Port Web Session Timeout Web Access Attempt Limit Web Lockout Duration Enable User Level Web Access Enable Viewer Level Web Access
Validate Server Certificates
SSH Enable SSH SSH Port SSH Idle Timeout EnableTelnet Telnet Port Telnet Idle Timeout
Security Controls for SSH/Telent Access

Sets web page access protocol, the options are: HTTP, HTTPS, Disabled.
Defines the HTTP Web Port, defaut is 80.
Defines the HTTPS Web Port, defaut is 443.
Defines the web session timeout, default value is 10 Minutes, the valid range is 1-60 minutes.
Defines the Web Access Attempt Limit before locking the web access, default is 5, valid range is 1-10.
Defines for how long the web access will be locked, default value is 15 Minutes, valid range is 0-60 Minutes.
Disables or enables user-level web access. Default value is “No” : the user level web access is disabled.
Disables or enables viewer-level web access. Default value is “No” : the viewer level web access is disabled.
Configures whether to validate the server certificate when downloading the firmware/config file. If set to “Yes”, the device will download the firmware/config file only after the server is validated. This setting requires a reboot to take effect.
If set to “Yes”, the phone will allow SSH access, Set to “Yes” by default.
Defines the SSH port, default is 22. Cannot be the same as Telnet Port. Valid range 1-65535.
Configures SSH session timeout (0-86400) seconds. default is “0” meaning no session timeout.
Enables/disables Telnet access. The default is “No”.
Defines the telnet port, the default is 23. Cannot be the same as SSH Port. Valid range 1-65535.
Configures Telnet session timeout (0-86400) seconds. default is “0” meaning no session timeout.
Specifies security measures and controls that apply to the SSH/Telnet protocol to ensure secure and authorized access to the device, the following options are supported: Only Allow SSH private IP users to set system Pvalue: This option permits only users
with SSH access from private IP addresses to modify specific system-level configuration parameters (system Pvalues). Allow all SSH users to set system Pvalue: With this setting, all users having SSH access can modify specific system-level configuration parameters (system Pvalues). Allow all SSH users to set any Pvalue: This allows all users with SSH access to modify any configuration parameter (P-value). Allow any user to set any Pvalue: Any user, regardless of their privileges, can modify any configuration parameter (Pvalue) through SSH or Telnet access. Prohibit setting Pvalue: This option blocks all users from modifying any configuration parameter (Pvalue) via SSH or Telnet access. The Default Value is “Allow any user to set any PValue”.

Access Control Allow List for WAN Side
Block List for WAN Side

If WAN Side Web/SSH Access is set to Yes or Auto. 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 blocklist and allowlist are not empty, the blocklist is processed first, followed by the allow list.
If WAN Side Web/SSH Access is set to Yes or Auto. 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 blocklist and allowlist are not empty, the blocklist is processed first, followed by the allow list.

User Info Management

Password Rules

Enable strict password rules

Enable strict password rules.

Minimum password length

Defines the Minimum password length, Range: 4-30, default is 8.

Required number of character classes

Sets the minimum number of character classes that a password should contain, composed of allowed combinations of different character classes;Range: 0-4, default is 3.

Allowed Character classes

Defines the Allowed Character classes, which are: Lower case, Upper case, Numbers, Special characters.

Password

New Admin Password

Defines the new admin password, Must contain 4-30 characters, When strict password rules are enabled, the password needs to comply with the settings in the password rules; Do not display password for security reasons.

Confirm Admin Password

Confirms the entered admin password.

New User Password

Defines the new user password, Must contain 4-30 characters, When strict password rules are enabled, the password needs to comply with the settings in the password rules; Do not display password for security reasons.

Confirm User Password

Confirms the entered user password.

New Viewer Password

Defines the new viewer password, Must contain 4-30 characters, When strict password rules are enabled, the password needs to comply with the settings in the password rules; Do not display password for security reasons.

Confirm Viewer Password

Confirms the entered viewer password.

Client Certificate

SIP TLS
Disable Weak TLS Cipher Suites
SIP TLS Certificate SIP TLS Private Key SIP TLS Private Key Password TLS Version
Minimum TLS Version
Maximum TLS Version
Custom Certificate Custom Certificate Trusted CA Certificates
Load CA Certificates
Trusted CA Certificates A Trusted CA Certificates B Trusted CA Certificates C Trusted CA Certificates D
Management Settings
TR-069

Allows users to disable weak ciphers DES/3DES and RC4, Symmetric Encryption SEED, Symmetric Encryption Encryption IDEA/eNULL, Symmetric Authentication MD5, Protocol Version SSLv2/SSLv3, Server Authentication aNULL/aECDH, Key Exchange Algorithm kRSA, or Disable All of the Above Weak Symmetric Encryption/Symmetric Authentication/Protocol Version/Server Authentication/Key Exchange/TLS Ciphers Suites. Default is “Enable Weak TLS Ciphers Suites”.
Fill in the SSL certificate file required when the ATA uses TLS as the SIP transmission method.
Fill in the SSL certificate file required when the ATA uses TLS as the SIP transmission method.
Fill in the SSL private key password required when the phone uses TLS as the SIP transmission method.
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.
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.
Allows users to update to the device their own certificate signed by a custom CA certificate to manage client authentication.
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
These trusted CA certificates will be used for the authentication server TLS certificate

Enable TR-069
ACS URL
ACS Username ACS Password
Periodic Inform Enable
Periodic Inform Interval Connection Request Username Connection Request Password Connection Request Port CPE SSL Certificate CPE SSL Private Key SNMP Settings Enable SNMP SNMP Version SNMP Port SNMP Trap IP Address SNMP Trap Port SNMP Trap Version SNMP Trap Interval SNMPv1/v2c Community SNMPv1/v2c Trap Community SNMPv3 User Name
SNMPv3 Security Level

Enable/disable the “CPE WAN Management Protocol” (TR-069). The default setting is Yes. Note: When configured, users can send a “false ring” signal through the TR-069 protocol for testing pruposes.
Specifies URL of TR-069 Auto Configuration Servers (e.g., http://acs.mycompany.com), or IP address. Default setting is: “https://acs.gdms.cloud”
Enter username to authenticate to ACS.
Enter password to authenticate to ACS.
If scheduled connection is enabled, the phone will send connection requests to the ACS server regularly. Default setting is “No”. Note: FXS port status can be detected through TR-069.
Sets frequency that the inform packets will be sent out to ACS. Default is 300 seconds.
Configure the ACS username for connecting the phone.
Configure the ACS password for the connected phone.
Configures the TR-069 connection request port. The value range is 0 to 65535.Default is 7547
Configure the certificate file for the telephony adapter to connect to the ACS over SSL.
Specify the certificate key for the phone adapter to connect to the ACS via SSL.
Enables the Simple Network Management Protocol. Default is “No”
Choose between (Version 1, Version 2c, or Version 3).
Listening Port of SNMP. Valid range is 161 or 1025-65535. (Default is 161).
IP address of trap destination. Up to 3 trap destinations are supported. Users should enter the IP addresses separated with comma (,).
Port of Trap destination. Valid range is 162 or 1025-65535. (Default 162).
Choose between (Version 1, Version 2c, or Version 3).
Time interval in minutes between traps. (Default is 5).
Name of SNMPv1/v2c community.
Name of SNMPv1/v2c trap community.
Username for SNMPv3.
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 SNMPv3 Privacy Protocol SNMPv3 Authentication Key SNMPv3 Privacy Key SNMPv3 Trap User Name
SNMPv3 Trap Security Level
SNMPv3 Trap Authentication Protocol SNMPv3 Trap Privacy Protocol SNMPv3 Trap Authentication Key SNMPv3 Trap Privacy Key MIB Management Interface Enable Management Interface
Management Access

Select the Authentication Protocol: “None” or “MD5” or “SHA.”
Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
Enter the Authentication Key.
Enter the Privacy Key.
Username for SNMPv3 Trap.
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.
Select the Authentication Protocol: “None” or “MD5” or “SHA”.
Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
Enter the Trap Authentication Key.
Enter the Trap Privacy Key.
Download Management Information Base (MIB) file.
Allows activation of a dedicated interface for remote management and configuration of the device. Disabled by Default.
Defines the allowed methods (protocols) and permissions for remote access and configuration of the device, two options can be selected: Management Interface Only: allows remote access and configuration solely through the
dedicated management interface. Both Service and Management Interfaces: permits remote access and configuration
through both the management interface and the service interface.

Enable SNMP Through Management Interface

Allows the activation of SNMP (Simple Network Management Protocol) access exclusively through the dedicated management interface for monitoring and managing the device remotely. Disabled by Default.

Enable TR069 Through Management Interface

Allows the activation of TR-069 protocol access exclusively through the dedicated management interface for remote device management and provisioning. Disabled by Default.

Enable Syslog Through Management Interface

Allows the activation of syslog communication exclusively through the dedicated management interface for remote logging and monitoring of device events. Disabled by Default.

Enable Radius Through Management Interface

Configure routing so that Radius functionality is accessible through management. Disabled by default.

Management Interface IPv4 Address

802.1Q/VLAN Tag

Allows the tagging of network packets with VLAN information to segment and prioritize network traffic.

802.1p priority value
IPv4 address type
IP Address Subnet Mask Default Router DNS Server 1 DNS Server 2
RADIUS Settings

The Valid Range is 0-4094, Default value is 0.
Assigns a priority value to network packets for Quality of Service (QoS) and traffic prioritization purposes. The Valid Range is 0-7, Default value is 0.
Configures the address type of the management network port, the options are: Dynamically configured by DHCP. Statically configured as.
Configure the IPv4 address of the management network port.
Defines the subnet mask.
Defines default gateway.
Defines DNS server 1 address.
Defines DNS server 2 address.

Enable RADIUS Web Access Control

Selects whether to enable RADIUS web access control. Default is “No”.

Action upon Radius Auth Server Error

Select the RADIUS authentication server error handling method,the user can choose between: Reject Access, or Authentificate locally, the default is “Authenticate Locally”.

RADIUS Auth Protocol

Configures RADIUS authentication protocol. Users can choose the following option: PAP, CHAP, MsCHAPv1, MsCHAPv2. Default is “PAP”.

RADIUS Auth Server Address

Configures RADIUS authentication server address.

RADIUS Auth Server Port

Configure RADIUS authentication server port. Valid range is 0-65535. Default port number is 1812.

RADIUS Shared Secret

Configures the shared secret key.

RADIUS VSA Vendor ID

Configure the RADIUS VSA provider ID to match the RADIUS server’s configuration. The default value for Grandstream Networks Inc is 42397

RADIUS VSA Access Level Attribute

Configure the RADIUS VSA access level attributes to match the configuration of the RADIUS server. Incorrect settings will cause Radius authentication to fail.

E911/HELD
E911 Enable E911 E911 Emergency Numbers

Enable Enhanced 911 call. Default is disabled.
A user can configure multiple emergency numbers separated with the delimiter symbol “;”. The default number is 911.

Geolocation-Routing Header

If “Yes”, E.911 INVITE message includes the “Geolocation-Routing” header with the value “Yes”. Default is disabled.

Priority Header

If “Yes”, E.911 INVITE message includes the “Priority” header with the value “emergency”. Default is disabled.

HELD

HELD Protocol

Configure HELD transfer protocol. HTTP or HTTPS. Default setting is HTTP.

HELD Synchronization Interval

The valid synchronization interval is between 30 to 1440 minutes. The synchronization is off when the interval is 0. The default value is 0.

HELD Location Types

Configure “locationType” element in the location request. “geodetic”, “civic” and “location URI”.

HELD Use LLDP Information

If set to “Yes”, LLDP protocol will be used to gather information about the HT device including information about the ChassisID and PortID. Otherwise, the system defaults to using the MAC address of the gateway and phone.
Default setting is “No”.

HELD NAI

If ‘Yes’ is selected, the Network Access Identifier (NAI) is included as the device identity in location requests sent to the Location Information Server (LIS). Default setting is “No”.

HELD Identity 1-10

The HELD Identity refers to Hierarchical Mobile IPv6 (HMIPv6) Enhanced Location-based Discovery (HELD) protocol. This protocol is utilized to determine the location of a mobile device when making an emergency call, such as a 911 call, over an IP-based network. You can configure up to 10 HELD identities.

HELD Identity 1-10 Value

Defines the HELD Identity value.

Location Server

Configures the location server (LIS) address.

Location Server Username

Configures Location Server (LIS) username.

Location Server Password

Configures Location Server (LIS) password.

Secondary Location Server

Configures secondary location server (LIS) address.

Secondary Location Server Username Configures secondary location server (LIS) username.

Secondary Location Server Password

Configures secondary location server (LIS) password.

Network Settings Page Definitions

Ethernet Settings

General Settings Internet Protocol

Selects one of the following IP protocol modes: IPv4 Only: Enforce IPv4 protocol only.

IPv4 IPv4 address type Host Name (Option 12) DHCP Domain Vendor Class ID (Option 60) PPPoE account ID PPPoE password PPPoE Service Name Preferred DNS server 1-4 IPv6
IPv6 address type
Static address type
Static IPv6 Address IPv6 Prefix Length DNS Server 1 DNS Server 2 Preferred DNS Server

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 HT801 V2/HT802 V2 unit for the changes to take effect.
Allows users to configure the appropriate network settings on the HT801 V2/HT802 V2 to obtain IPv4 address. Users could select DHCP, Static IP or PPPoE. By default, it is set to DHCP.
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.
Allows user to configure DHCP domain name. This option specifies the domain name that the client should use when resolving hostnames via the Domain Name System. This field is optional.
Exchanges vendor class ID by clients and servers to convey particular configuration or other identification information about a client. Default is HT8XXV2.
Defines the PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection.
Specifies the PPPoE account password.
Defines PPPoE service name. If your ISP uses a service name for the PPPoE connection, enter the service name here. This field is optional.
Specifies preferred DNS server to use when DHCP or PPPoE are set. You can set up yo 4 Preferred DNS Servers.
Configures the IPv6 address of the phone: 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 FXO Gateway acquires its IP address from the first DHCP server it discovers from the LAN it is connected. Statically configured as: configure IP address, 1st 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).
When using fully static type IPv6, enter a static IPv6 address.
Enter the static IPv6 address prefix length.
Enter DNS server 1 address.
Enter DNS server 2 addres.
Enter preferred DNS server address.

802.1x Settings
802.1x Mode
802.1x Identity 802.1x Password 802.1x CA Certificate 802.1x Client Certificate
DDNS Settings
Enable DDNS
DDNS Server
DDNS Username DDNS Password DDNS Hostname DDNS Hash
OpenVPN® Settings
Enable OpenVPN® OpenVPN® Server Address OpenVPN® Port OpenVPN® Interface type OpenVPN® Transport Enable OpenVPN® LZO Compression

Allow users to enable/disable 802.1X Mode. The modes available are: EAP_MD5 EAP_TLS EAP-PEAPv0/MSCHAPv2 Disabled by default. Enter 802.1X identity information. Enter 802.1X MD5 password. Paste down the 802.1X certificate .pem file. Paste down the Client.pem certificate file containing certificate and key. Maximum allowed length is 8192.
Enable DDNS function. Disabled by default Select DDNS server used, options are: dyndns.org freedns.afraid.org zoneedit.com no-ip.com oray.net Sets the username of the DDNS server. Sets the password of the DDNS server. Sets the domain name of the DDNS server. Sets the hash value of the DDNS server.
Allow users to enable/disable OpenVPN®. Default is No. Specify the IP address or FQDN for the OpenVPN® Server. Specify the listening port of the OpenVPN® server. Valid range is 1-65535. Default is 1194. Specify the Interface type of OpenVPN® whether TAP or TUN. Default is TUN. Specify the Transport Type of OpenVPN® whether UDP or TCP. The default is UDP.
Enable OpenVPN® LZO Compression. Default is Yes.

OpenVPN® Encryption OpenVPN® Diges OpenVPN® Username OpenVPN® Password OpenVPN® CA OpenVPN® Certificate OpenVPN® Client Key OpenVPN® Client Key Password
OpenVPN® Additional Options
Advanced Settings

Select the OpenVPN® Encryption. Default is BF-CBC 128 bit (default key).
Select the OpenVPN® Digest Algorithm. Default is SHA1.
Configure the OpenVPN® username.
Configure the OpenVPN® password.
Specifies the OpenVPN® CA. Maximum Character Number is 8192.
Specifies the OpenVPN® Certificate. Maximum Character Number is 8192.
Specifies the Client Key. Maximum Character Number is 8192.
Configures the OpenVPN® Client Key Password. Maximum Length is 64.
Additional options to be appended to the OpenVPN® config file. Note: Additional options are seperated by semicolon. For example: connect-retry 5;tls-client;keepalive 10 60. Please use with caution. Make sure that the options are supported by OpenVPN® and do not unnecessarily override other configurations above.

Advanced Settings

Enable LLDP

Controls the LLDP (Link Layer Discovery Protocol) service. The default setting is “Enabled”.

LLDP TX Interval

Configures LLDP TX Interval (in seconds). Valid range is 1 to 3600. The Default value is 60.

Enable CDP

Enables/Disables CDP “Cisco Discovery Protocol”. The default setting is “Enabled”.

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets that can be used to identify and distinguish different VLANs.
Valid range is 0 to 4094. The default value is 0.

Layer 2 QoS SIP 802.1p

Prioritizes SIP traffic at Layer 2 using the 802.1p standard for Quality of Service (QoS). The valid range 0-7. The default value is 0.

Layer 2 QoS RTP 802.1p

Prioritizes RTP (Real-time Transport Protocol) traffic at Layer 2 using the 802.1p standard for Quality of Service (QoS).
The valid range 0-7. The default value is 0.

Use DNS to detect network connectivity

Use DNS to detect WAN side network issues. The default setting is “No”.

Use ARP to detect network connectivity

Use ARP to check network connection. The default value is “Yes”.

Firewall settings

Black List for WAN Side Port

Ports on the blocklist will be disabled on the device.

Static DNS Cache

NAPTR

NAPTR DNS Cache Name

The domain name to which this resource record refers. Maximum length is 512 characters.

NAPTR DNS Cache Time Interval (s)

(Time to Live) refers to the time interval that the record may be cached before the source of the information should be consulted again.
Valid range is from 300 to 65535. Default value is “300”.

NAPTR DNS Cache Order

Specifies the order in which the NAPTR record need to be processed, low numbers are processed before high numbers.
Valid range is from 0 to 65535. Default value is “0”.

NAPTR DNS Cache Preference

Refers to the order in which NAPTR records with the same “Order” values need to be processed. records are processed from lower preference numbers to higher preference numbers.
Valid range is from 0 to 65535. Default value is “0”.

NAPTR DNS Cache Replacement

This parameter contains the next FQDN to query for NAPTR, SRV, or address records. Maximum length is 64 characters.

NAPTR DNS Cache Service

Specifies the services available in this domain. The replacement field is used to get to this service.
It can also specify the protocol used to communicate with the server that offers this service.
In SIP, three services are defined along with their resolution services (resolution services are defined after the “+” sign):
“SIPS+D2T”: Secure SIP, TLS over TCP. “SIP+D2T”: SIP over TCP. “SIPS+D2S”: Secure SIP, TLS over SCTP. “SIP+D2S”: SIP over SCTP. “SIP+D2U”:SIP over UDP. Maximum length is 64 characters and default value is “SIP+D2U”.

SRV

SRV DNS Cache Name

The domain name of the wanted service. Maximum supported length is 512 characters.

SRV DNS Cache Time Interval (s)

(Time to Live) refers to the time interval that the record may be cached before the source of the information should be consulted again.
Valid range is from 300 to 65535. Default value is “300”.

SRV DNS Cache Priority

Enables to prioritize target hosts that support the given service. Lowest number will get the highest priority. Target hosts with the same priority will be tried in an order defined by the weight field.
Valid range is from 0 to 65535. Default value is “0”.

SRV DNS Cache Weight

Specifies the weight for entries with the same priority. Larger weight will get higher priority. Valid range is from 0 to 65535. Default value is “0”.

SRV DNS Cache Target

The domain name of the target host. Maximum supported length is 64 characters.

SRV DNS Cache Port

The port on the target host of the specified service. The valid range is 0-65535. Default value is “0”.

A DNS Cache Name

The domain name to which the record refers. Maximum supported length is 512 characters.

A DNS Cache Time Interval (s)

(Time to Live) refers to the time interval that the record may be cached before the source of the information should be consulted again.
Valid range is from 300 to 65535. Default value is “300”.

A DNS Cache IP Address

The IP address associated with the DNS record name.

Maintenance Page Definitions

Upgrade

Firmware

Upgrade via Manual Upload

Firmware

Uploads the .bin firmware file.

Upgrade via Network

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 hosts the firmware release. Note: You can specify the protocol used in the Firmware Server Path. (example: https://192.168.5.120), this will bypass the “Upgrade Via” method.

HTTP/HTTPS/FTP/FTPS User Name Enters username 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

Manual Configuration

Upload Configuration

Uploads the configuration file in .txt or .xml formats.

Restore From Backup Configuration

Restore From Backup Configuration.

Download Device Configuration

Downloads Device Configuration in .txt format.

Download Device XML Configuration

Downloads Device Configuration in .xml format (The P-values will be sorted in ascending order).

Export Backup Configuration

Export Backup Configuration in .xml format.

Configure via Network

Config Upgrade via

Selects Config file upload method: TFTP, HTTP, HTTPS, FTP or FTPS. Default is HTTPS.

Config Server Path

Sets IP address or FQDN of configuration server. The URL of the server that hosts the configuration file to provision HT80xV2.
Note: You can specify the protocol used in the Config Server Path. (example: https://192.168.5.120), this will bypass the “Upgrade Via” method.

HTTP/HTTPS/FTP/FTPS User Name Enters username to authenticate with HTTP/HTTPS FTP/FTPS server.

HTTP/HTTPS/FTP/FTPS Password

Enters password to authenticate with HTTP/HTTPS FTP/FTPS server.

XML Config File Password

Decrypts XML configuration file when encrypted. The password used for encrypting the XML configuration file using OpenSSL.

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.

Provision

Allow DHCP Option 66 or 160 to override server

Obtains configuration and upgrade server’s information using options 66 from DHCP server. Note: If DHCP Option 66 is enabled, the HT80xV2 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”.
This option is enabled by default.

Auto Provision

When enabled, the device sends SUBSCRIBE in multicast mode. If 3CX, UCM and other IPPBX servers are used as SIP servers, the device can be automatically configured. Enabled by default.

Enable using tags in URL

Allows users to configure variables on the configuration server path to differentiate the directories on the server. Disabled by default.

Additional Override DHCP option

Configures additional DHCP Option to be used for firmware/config server instead of the configured firmware/config server or the server from DHCP Option 66 and 160. This option requires a reboot to take effect.

Automatic Upgrade

Specifies when the firmware upgrade process will be initiated; there are 4 options: Note: The HT80xV2 will only do an 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).

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent. The options are:
Always Check for New Firmware at Boot up Check New Firmware only when F/W pre/suffix changes

Download and Process All Available Config Files
Advanced Settings Verify host when using HTTPS Configuration File Types Allowed Authenticate Conf File

This feature allows users to download and process all available config files. By default, the device will provision the first available config in the order of cfgMAC > cfgMAC.xml > cfgMODEL.xml > and cfg.xml (corresponding to device-specific, model-specific, and global configs). If this option is enabled, the device will inverse the downloading process to cfg.xml > cfgMODEL.xml > cfgMAC.bin > cfgMAC.xml and add cfgMAC_override.xml. The following files will override the files that have already been loaded and processed. The default value is “No”.
Enables / disables the host verification when using HTTPS. Disabled by default.
Allows users to configure provision configuration file type in xml file only or all file types.
Authenticates configuration before being accepted. This protects the configuration from unauthorized modifications. Default is No.

System Diagnosis

Syslog Syslog Server Syslog Level

URL or IP address of the syslog server.
Select the HT8xx 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)

Syslog Protocol
Syslog Header Format Send SIP Log Debug Ethernet Capture With secret key information Port Record Core Dump GR909
Test To Run
FXS Line Run Interval
File Management
CDR Records

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)
Allow encrypted SSL/TLS transmission to the syslog server if SSL/TLS is selected, the default value is UDP. Notes: The validity of the server CA certificate will be verified.
Determines how the headers in syslog messages are formatted, the options are: Default: Uses the software’s or system’s standard header format RFC3164 Compliant: Adheres to the RFC 3164 standard for syslog headers,
Sets whether to include SIP logs in the system log Default value is “No”.
Enables the capturing and analysis of Ethernet network traffic for troubleshooting and monitoring purposes.
Allows users to make packet capture including the secret key to decrypt the captured TLS packets. Default value is Yes.
Allows the recording of port-level call information, such as call duration, caller ID, and call status, for monitoring and logging purposes.
Provides generated core dump file if unit malfunctions. Clean will be displayed if no issues.
Selects the type of test that wll be performed for the specified line, the following tests can be performed: Hazardous Potential Test Foreign Electromotive Forces Test Resistive Faults Test Receiver Offhook Test Ringer Equivalent Number Test
Defines the FXS line under test.
Defines the run Interval in seconds, the default value is 120 seconds, the default range is 0604800 seconds.
Displays a list of the last 1000 call records, the ionformation included in the records are the following: UserID ToNumber FromNumber

SIP Messages
Device Manager
Reboot Automatic restart Restore Factory
Reset Type

StartTime StartTalkTime EndTime Duration State Direction Clicking the “Download” button will download the last 1000 call records. Clicking the “Delete” button will delete the last 1000 call records.
The SIP messages tab displays a list of SIP headers that are captured after each triggered call, the information displayed on the tab are similar to the information that can be captured using the wireshark tool, and filtering by SIP protocol.
Clicking the “Download” icon will download the whole log of the captured traces. Clicking “Delete” will delete all the captured traces. This tool can be useful for network administrators that want to treoubleshoot the call flows without having to use an external tool, for analyzing the network
This option triggeres an automatic reboot of the unit, the available options are:
No Reboot everday: you can specify the reboot hour Reboot everyweek: you can specify the day of the week, and the hour of the day, for the
reboot to take place Reboot every month: you can specify the day of the month, and the hour of the day, for the
reboot to take place
Gives the administrator the option to restore the default configuration on the HT801 V2/HT802 V2. There are 3 types of factory reset:
ISP Data Reset: All ISP (Internet Service Provider) configurations that may affect the IP address will be reset (including WAN static IP).
VoIP Data Reset: All VoIP-related configurations (mainly everything located on FXS page).
Full Reset: Both VoIP and ISP-related configuration at the same time. Note: After choosing the reset type, you will have to click the reset button for it to take effect.

Port Settings Page Definitions

General Settings Account Registration Profile Active Primary SIP Server

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.
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

Failover SIP Server
Prefer Primary SIP Server
Outbound Proxy
Backup Outbound Proxy Prefer Primary Outbound Proxy From Domain Allow DHCP Option 120 (override SIP server) SIP User ID Authenticate ID Authenticate Password Name
Tel URI
Network Settings Layer 3 QoS SIP DSCP Layer 3 QoS RTP DSCP
DNS Mode

IPv6: fe80::20b:82ff:fe75:211d). This is the primary SIP server used to send/receive SIP messages from/to HT801 V2/HT802 V2.
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 fe80::20b:82ff:fe75:211d:). This server will be used if primary SIP server becomes unavailable.
Selects to prefer primary SIP server. The account will register to primary Server if registration with Failover server expires. Default is No.
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, IPv4: 192.168.5.170, or IPv6: fe80::20b:82ff:fe75:211d). It’s Used by HT801 V2/HT802 V2 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
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: fe80::20b:82ff:fe75:211d). By default, this field is left empty.
If the user configures this option to “Yes”, when the registration expires, the device will reregister via primary outbound proxy. By default, this option is disabled.
Allows users to add the actual domain name, it will override the from header.This is an optional configuration.
Configures the HT801 V2/HT802 V2 to collect SIP server address from DHCP option 120. Default is No.
Defines The user part of a SIP address
Defines the Authenticate ID of the User
Defines the Authenticate password
Defines the name of the user
If the device has an assigned PSTN telephone number, this field should be set to “User=phone”. A “User=phone” parameter will be attached to the Request-Line and “To” header in the SIP request to indicate the E.164 number. If set to “Enabled”, “tel:” will be used instead of “sip:” in the SIP request. Disabled by default.
the Diff-Serv value for SIP packets in decimal, the range is 0-63, default value is 26
Diff-Serv valuefor RTP packets in decimal, the range is 0-63, default value is 46
Selects DNS mode to use for the client to look up server. One mode can be chosen. A Record (Default): 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 the SIP server is configured as domain name, device will not send DNS queries, but will use “Primary IP” or “Backup IP” to send SIP message if at least one of them is 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 retries.

DNS SRV Failover Mode
Failback Timer Maximum Number of SIP Request Retries Register Before DNS SRV Failover Primary IP Backup IP1 Backup IP2

Configure the preferred IP mode when DNS Mode is SRV or NAPTR/SRV. · Default SIP request will always be sent to the address with the top priority based on the SRV query result, even if this address is different from the registered IP address. · Saved one until DNS TTL SIP request will always be sent to the registered IP address until DNS TTL expires or registered IP address is unreachable. · Saved on until no response SIP request will always be sent to the registered IP address only until registered IP address is unreachable. · Failback follows failback expiration time: the primary server regains control only after the failback expiration time, allowing the secondary server to handle requests until then.
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).
This feature allows user to configure the number of SIP retries before failover occurs. (between 1 and 10, default is 2).
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 non-registration SIP request (INVITE) to the new successfully registered server or not.
The main IP address used for communication and control.
The secondary IP address used as a backup for communication in case the Primary IP fails.
The third IP address used as an additional backup for communication in case both the Primary IP and Backup IP1 fail.
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, Auto, VPN.

NAT Traversal
Use NAT IP Proxy-Require SIP Settings SIP Basic Settings SIP Registration

No: NAT traversal is disabled; the phone doesn’t attempt to handle NAT issues. Keep-alive: Sends periodic packets to keep the NAT mapping open for incoming calls. STUN: Uses a STUN server to discover the public IP address and port, helping traverse NAT. UPnP: Uses Universal Plug and Play to automatically configure the router to allow SIP traffic through NAT. Auto: Automatically selects the best NAT traversal method based on the network environment. VPN: Routes all SIP traffic through a secure Virtual Private Network, bypassing NAT issues. Defeault is “No”. Defines NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP. Determines a SIP Extension to notify the SIP server that the HT801 V2/HT802 V2 is behind a NAT/Firewall.
Controls whether the HT801 V2/HT802 V2 needs to send REGISTER messages to the proxy server. The default setting is Yes.

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.

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.Unregister on reboot option can be set to “No”, “All” or “Instance”.
1. Set to “No”: If the “Unregister on reboot” option is set to “No”, it means that the SIP user’s information will not be cleared when the device reboots. In other words, the SIP account will remain registered with the server even after the device is rebooted.
2. Set to “All”: If the “Unregister on reboot” option is set to “All”, it means that all SIP accounts associated with the device will be unregistered when the device reboots. This option clears the SIP user’s information for all the FXS ports on the device.
3. Set to “Instance”: If the “Unregister on reboot” option is set to “Instance”, it means that only the SIP account associated with the concerned FXS port will be unregistered when the device reboots. This option clears the SIP user’s information for only the specific FXS port that is affected by the reboot.
Default value is set to “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.

Refreshes registration periodically with specified SIP proxy (in minutes). Maximum interval is 65535 minutes (about 45 days). Default is 60 minutes (or 1 hour).

Re-register before Expiration

Sends re-register request after specific time (in seconds) to renew registration before the previous registration expires.

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.

Use Random SIP Registration Failure Retry Wait Time

When enabled, the waiting time to resend a SIP REGISTER request to the SIP server in case of registration failure will be a random number in the following range.
Default setting is disabled

Random SIP Registration Failure Retry Wait Time Range

Allows users to set a wait time (in seconds) before attempting to send a SIP registration after each failure. The wait time will be chosen randomly within this range.
Valid values range from 60 to 600 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.

Port Voltage Off upon no SIP Registration or SIP Registration Failure

Cuts off voltage to the port if there is no SIP registration or if SIP registration fails, preventing unwanted calls. (in minutes. Between 0-60, default is 0. 0 means port voltage is never turned off)

Delay Time of Port Voltage Off Timer Since Boot

Sets the delay time after boot before the FXO port voltage turns off. It controls the timing for powering down FXO ports on the gateway. The value is in minutes. Between 0-60, default is 0

Enable SIP OPTIONS/NOTIFY Keep Alive

Enables SIP OPTIONS or SIP NOTIFY to track account registration status so the ATA will send periodic OPTIONS/NOTIFY message 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 ATA send OPTIONS or NOTIFY message to SIP server. The default setting is 30 seconds, which means the ATA 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 Keep Alive before re-registration. Between 3-10, default is 3.

SUBSCRIBE for MWI

Sends SUBSCRIBE periodically (depends on “Register Expiration” parameter) for message waiting indication. Default is No.

Subscribe Retry Wait Time upon 403 Forbidden

Configures (in minutes) how long it takes to send a SUBSCRIBE request after a “403 Forbiden” response. Valid range is 0-10080. Default is 0, max 7 days.

Local SIP Port

Defines local port to use by the HT801 V2/HT802 V2 for listening and transmitting SIP packets. Default value is 5060

Use Random SIP Port

Controls whether to use configured or random SIP ports. This is usually necessary when multiple HT801 V2/HT802 V2 are behind the same NAT. The default is No.

Hold Target Before Refer

Allows user to hold or not hold the phone call before referring. The default setting is Yes.

Refer-To Use Target Contact

Includes target’s “Contact” header information in “Refer-To” header when using attended transfer. Default is No.

Remove OBP from Route Header

Removes outbound proxy info in “Route” header when sending SIP packets. Default setting is No.

Support SIP Instance ID

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

Support outbound

Enables/disables outbound support.

Support GRUU

Enables/disables GRUU support.

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. Default is No.

Use Request Routing ID in SIP INVITE Header

This options enables the inclusion of a routing ID in the SIP INVITE header for directing the request to a specific endpoint or route.

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 INVITE message. If set to “default”, the P-Preferred-Identity Header will be omitted in SIP INVITE message when Telkom SA or CBCOM 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, IEE802.11b in P-Access-Network-Info SIP header. Default is Yes.

Use P-Emergency-Info Header

This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for emergency calls. Default is Yes.

Use P-Asserted-Identity Header

When this feature is set to Yes, device will send P-Asserted-Identity Header on the SIP Invite. Default setting is No.

SIP REGISTER Contact Header Uses

Configures whether the SIP REGISTER Contact Header Uses LAN address or WAN address. Default setting is “LAN Address”.

SIP REGISTER Contact Header Uses

SIP REGISTER Contact Header Uses LAN address or WAN address By default, it uses LAN Address

SIP T1 Timeout

Defines T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, HT801 V2/HT802 V2 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 is defined by T2. The 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 the T2 value. The default is 4 seconds.

SIP Timer D

Configures SIP Timer D defined in RFC3261. 0 64 seconds. Default is 0.

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.

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.

Enable Call Waiting Alert-Info in 180 Ringing Response

Activates the Call Waiting feature by including Call Waiting Alert-Info in the 180 Ringing response sent to the caller during an incoming call. Default value is no.

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

Add MAC in User-Agent

This feature allows users to configure “User-Agent” in the SIP header field, when this feature is set to”No”, “User-Agent” does not carry a MAC, when this feature is set “Yes except REGISTER”, “User-Agent” in REGISTERSIP header field does not carry MAC but other SIP packet header fields carries MAC, when this feature is set to “Yes to all SIP”, “User-Agent” in the SIP packet header field will carries the MAC.

Use MAC Header

This feature allows users to configure MAC Header in the SIP packet header field, when this feature is set to “No” ,the MAC header field is not carried in the SIP packet header field, when this feature is set to “REGISTER Only”, the register packet header field carries the MAC header field but the remaining SIP packets do not carry MAC header fields, when this feature is set to “Yes to all SIP”, the MAC header field is carried in the SIP data packet header field.

Session Timer

Enable Session Timer

Disable the session timer when this option is set to “No”. By default, this option is enabled.

Session Expiration

Enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or reINVITE). When the session interval expires, if there is no refresh via an UPDATE or reINVITE 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). Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it. Valid range is 9064800 seconds. 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. To turn off Session Timer, select “No” for Caller and Callee Request Timer, and Force Timer. Default is No

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls. Default is Omit. UAC: The device acts as the refresher. UAS: Callee or proxy server act as the refresher.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls. Default is UAC: UAS: The device acts as the refresher. UAC: Callee or proxy server act as the refresher.

Force INVITE

Uses INVITE message to refresh the session timer. Default is No.

When To Restart Session After ReINVITE received

Allows users to delay posting Media Change Event, it can be set to “Immediately” or to “After replying 200OK” The default value is “Immediately”.

Codec Settings

DTMF Settings

Preferred DTMF

Sorts DTMF methods (in-audio, via RTP (RFC2833) or via SIP INFO) by priority.

Force DTMF to be sent via SIP INFO simultaneously

Configures whether or not to force DTMF to be sent via SIP INFO simultaneously. Default setting is No.

Inband DTMF Duration

Allows users to config the Inband DTMF Duration and inter-duration. The default Duration is 100 ms. Valid range: 40-2000 ms. The default Inter-duration is 50 ms. Valid range: 40-2000 ms.

DTMF Payload Type

Defines payload type for DTMF using RFC2833.

Inband DTMF Tx Gain

Adjusts the transmit gain for inband Dual-Tone Multi-Frequency (DTMF) tones during call signaling. The Valid range is -12-12 db, default is 0

DSP DTMF Detector Duration Threshold

Allows users to config the DSP DTMF Detector Duration Threshold. The default Duration is 30 ms. Valid range: 20-2000 ms. The default Inter-duration is 30 ms. Valid range: 20-2000 ms.

DSP DTMF Detector Min Level DSP DTMF Detector Snr
DSP DTMF Detector Deviation
DSP DTMF Detector Twist Enable DTMF Negotiation RFC2833 Events Count RFC2833 End Events Count Vocoder Settings Vocoder Settings (in listed order)
Voice Frames per TX G.723 Rate iLBC Frame Size Enable OPUS Stereo in SDP iLBC Payload Type Opus Payload Type Enable Audio RED with FEC Audio FEC Payload Type Audio RED Payload Type Silence Suppression

determines the minimum level of DTMF tones that the DSP DTMF detector can reliably detect, range is -45-0 dBm, default is -25
Sets the signal-to-noise ratio threshold for the DSP DTMF detector, determining the sensitivity to distinguish DTMF tones from background noise in the audio signal. Range: 0-12, default is 1
Specifies the allowed frequency deviation from standard DTMF tones that the DSP DTMF detector can accommodate. Range: 0-25, default is 25
Defines the maximum deviation from ideal tone frequencies allowed for reliable DTMF detection by the DSP. Range: 0-12dB, default is 5
Uses DTMF order without negotiation. Default is No.
This feature allows users to customize the count of RFC2833 events. Default is 8.
This feature allows users to customize the count of RFC2833 end events. Default is 3.
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, iLBC and OPUS
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.
Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is 6.3kbps.
Specifies iLBC packet frame size (20ms or 30ms). Default is 20ms.
Enabkes OPUS stereo in SDP. Default is Yes.
Determines payload type for iLBC. Valid range is between 96 and 127. Default is 97.
Determines payload type for OPUS. Valid range is between 96 and 127. Default is 123.
Enables the use of audio redundancy (RED) with forward error correction (FEC) to enhance audio quality and resilience against packet loss in real-time communication systems such as VoIP
Specifies the payload type used for transmitting audio forward error correction (FEC) packets in a VoIP system.
Defines the payload type assigned to audio redundancy (RED) packets in a VoIP system,
If set to “Yes”, when the device detects that there is no voice flow in the call, the device will send out a small number of VAD packets (instead of voice packets) Disabled by default.

Use First Matching Vocoder in 200OK SDP

When enabled, the gateway will use the first matching code in 200OK SDP to make calls Disabled by default.

Fax Mode

Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through. If using Passthrough mode, select preference codec as PCMU or PCMA.

T.38 Max Bit Rate

Selects the maximum T.38 bit rate. Lowering the maximum fax bit rate may help improve the fax success rate. Only effective when the fax machine is fax receipt. Available options are: 4800bps, 9600bps, 14400bps. Default value is 9600bps.

Re-INVITE After Fax Tone Detected

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.

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).

Crypto Life Time RTP Setting

Adds crypto life time header to SRTP packets. Default is Yes.

Local RTP Port

Defines the local RTP-RTCP port pair the HT801 V2/HT802 V2 will listen and transmit. It is the HT801 V2/HT802 V2 RTP port for channel 0. The default value for FXS port is 5004

Use Random RTP Port Random RTP Port Range Symmetric RTP Enable RTCP RTP/RTCP Keep Alive On Hold
SRTP Mode
SRTP Key Length

Controls whether to use configured or random RTP ports. This is usually necessary when multiple HT801 V2/HT802 V2 are behind the same NAT. The default is No.
Set Random RTP Port Range1024-65535, Min – Max.
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.
Allows users to enable RTCP. The default setting is “Yes”.
Enables or disables RTP/RTCP keep-alive packets during call hold to maintain connectivity and prevent session timeouts. Disabled by default.
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
Allows users to select supported SRTP Key Length. the available values are : AES 128&256 bit AES 128 bit AES 256 bit

Analog signal line configuration SLIC Setting Caller ID Scheme DTMF Caller ID Polarity Reversal
Loop Current Disconnect
Play busy/reorder tone before Loop Current Disconnect Loop Current Disconnect Duration Enable Pulse Dialing Pulse Dialing Standard Enable Hook Flash
Hook Flash Timing
On Hook Timing

Depends on standard phone type (and location).
Selects the caller id scheme, for example: Bellcore/Telcordia, ETSI-FSK …
Configures start and stop tone for DTMF caller ID scheme.
Reverses the polarity upon call establishment and termination. Default is No.
Allows the traditional PBX used with HT801 V2/HT802 V2 to apply this method for signaling call termination. The method initiates a short voltage drop on the line when the remote (VoIP) side disconnects an active call. The default is No.
Allow user to configure if it will play busy/reorder tone before loop current disconnect upon call fail. Default is No.
Configures the duration of voltage drop described in the topic above. HT801 V2/HT802 V2 support a duration range from 100 to 10000ms. The default value is 200.
Allow users to enable Pulse Dialing option under FXS Port. Default is No.
Allows users to use Swedish pulse dialing standard or New Zealand pulse dialing standard. Default is General Standard.
Enables the FLASH button to be used for terminating calls. Default is Yes.
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 V2/HT802 V2 support a range from 40 to 2000 ms. Default values are 300 minimum and 1100 maximum.
Specifies the on-hook time for an on-hook event to be validated. HT801 V2/HT802 V2 support a range from 40 to 2000 ms. Default value is 400.
Adjusts the voice path volume.

Gain
Disable Line Echo Canceller (LEC) Ring Frequency Ring Power

· 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 (ie. 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.
Disables the LEC per call base. Recommended for Fax/Data calls. Default is No.
Chooses an appropriate ringing frequency. Default is 20Hz.
Selects ringing power settings, the options are: 45Vrms, 50Vrms, and 55Vrms , the higher the value, the more powerful the ring is.

OnHook DC Feed Current Call Settings Offhook Auto-Dial Offhook Auto-Dial Delay No Key Entry Timeout
Early Dial
Dial Plan Prefix Dial Plan

Default is 45Vrms.
This feature is used to adjust DC feed current.
Configures the user ID/extension to dial automatically when offhook.
Sets the delay time before automatic dialing begins after going off-hook on an FXO port. Range is 0-60 seconds, default is 0
Initiates the call within this time interval if no additional key entry during dialing stage. Default is 4 seconds.
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 Incomplete Address” responses. 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.
Adds specified prefix to dialed number.
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 j. Flag T when adding a “T” at the end of the dial plan, the phone will wait for 3 seconds before dialing out. This gives users more flexibility on their dial plan setup. E.g. with dial plan 1XXT, phone will wait for 3 seconds to let user dial more than just 3 digits if needed. Originally the phone will dial out immediately after dialing the third digit. 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 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. 1. 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

Use # as Dial Key Enable # as Redial Key RFC2543 Hold Disable Call-Waiting Disable Call-Waiting Caller ID Disable Call-Waiting Tone
Send Anonymous
Disable Connected Line ID Enable Receiver Offhook Tone Enable Reminder Ring for On-Hold Call Enable Reminder Ring for DND Enable Visual MWI
Visual MWI Type
MWI Tone Transfer on Conference Hangup Ringing Transfer

feature. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.
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.
Disables # to act as Redial key. If set to “Yes” and feature “Use # as Dial Key” set to Yes, the # key will act as dial key but not as redial key. Default is Yes .
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 . RFC3261 (a line) will play the hold music to the other side.
Disables receiving a second incoming call when the line is engaged. Default is No.
Disables displaying caller ID when receiving a second incoming call. Default is No.
Disables playing call waiting tone during active call when receiving a second incoming call. The CWCID will still be displayed. Default is No.
Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, blocking caller ID. Default is No. Note: When the caller is marked as anonymous, the current phone Caller ID will be set as “private caller”, and when is marked as unavailable, the current phone caller ID is marked as “unknown caller”
Disables displaying the number of the person answering the phone. Default is No.
Enables / disables the warning to alert that the phone has been left off-hook for an extended period of time. Default is No.
Enables playing the reminder ring. Default is No.
This feature allows user to disable reminder ring when FXS port is on DND mode. Default is Yes.
Disables use of visual message waiting indicator when there is an unread voicemail message. Default is No.
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 (depending 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.
When set to Default, device will play Stutter Dial Tone when there is voicemail, if set to Special Proceed Indication Tone, device will play the configured special proceed indication tone upon user offhook when there is voicemail
Determines whether a call is automatically transferred to a specified destination after a conference call ends. Disabled by Default
Allows users to trigger the transfer once the user hangs up the call, Default setting is “No”.

No Answer Timeout Send Hook Flash Event
Flash Digit Control
Callee Flash to 3WC
Ring Timeout
Caller ID Display
Enable Unknown Caller ID Replace Beginning ‘+’ with 00 in Caller ID Number of Beginning Digits to Strip from Caller ID Escape ‘#’ as %23 in SIP URI Enable Connected Line ID Advanced Settings Conference URI Special Feature Allow SIP Reset Validate Incoming SIP Message Check SIP User ID for Incoming INVITE
Authenticate Incoming INVITE
Allow Incoming SIP Messages from SIP Proxy Only

Defines the timeout (in seconds) before the call is forwarded on no answer. Valid range is 1 to 120.
Default is No. If set to yes, flash will be sent as DTMF event.
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.
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.
Stops ringing when incoming call if not answered within a specific period of time. When set to 0 there will be no ringing timeout. Default is 60 seconds.
When set to “Auto”, the device will look for the Caller ID in the order of P-Asserted Identity, Remote-Party-ID and From header field in the incoming SIP INVITE. When set to “Disabled”, all incoming calls are displayed with “Unavailable”. When set to “From Header”, the device will look for the Caller ID in the From header field of the incoming SIP INVITE.
Enables the Unknown Caller ID information Enabled by default
Replace Beginning ‘+’ with 00 in Caller ID Disbaled by default.
Range is between 0 and 10, default is 0
Replaces # by %23 for some special situations.
If set to “No”, Connected Line ID will not be displayed even received
Configures the conference URI when using BroadSoft N-way calling feature.
Specifies the server type for special requirements.
Allows ressting the unit using the SIP requests.
Checks the authenticity and integrity of incoming SIP messages for enhanced security.
Checks the SIP User ID in the Request URI of the incoming INVITE; if it doesn’t match the HT801 V2/HT802 V2 SIP User ID, the call will be rejected. Direct IP calling will also be disabled. The default is No.
Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.
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.

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. The default setting is “No”.

Trusted Domain Name List

Supports setting specific domain names and wildcard domain names, wildcard domain names such as “*.grandstream.com”, use “,” to separate domain names

Authenticate Server Certificate chain

Verifies certificate when communication method is TCP/TLS

Call Features Settings

Enable Call Features

When enabled, Do No 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 per call

Enable SRTP: Default is 16

Disable SRTP per call

Disable SRTP: Default is 17

SRTP per call Feature

Enable SRTP per call: Default is 18 Disable SRTP per call: Default is 19

Enable SRTP per call

Set the function code to enable SRTP (the default value is 18), which is only valid for the current call

Disable SRTP per call

Set the function code to disable SRTP (the default value is 18), which is only valid for the current call

CID Feature

Allow users to customize the CID feature codes. Default is Yes

Enable CID

Enable CID: Default is 31

Disable CID

Disable CID: Default is 30

CID per call Feature

Whether to display the user ID, it is only valid for the current call

Enable CID per call

Enable CID per call: Default is 82

Disable CID per call

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

Direct IP Calling: Default is 47

CW Feature

Allow users to customize the call waiting feature codes. Default is Yes

Enable CW

Enable CW: Default is 51

Disable CW

Disable CW: Default is 50

CW per call Feature Enable CW per call Disable CW per call Call Return Feature Call Return Unconditional Forward Feature Enable Unconditional Forward Disable Unconditional Forward Busy Forward Feature Enable Busy Forward Disable Busy Forward Delayed Forward Feature Enable Delayed Forward Disable Delayed Forward Paging Feature Paging DND Feature Enable DND Disable DND Blind Transfer Feature Enable Blind Transfer Disable LEC per call Feature Disable LEC per call Set Offhook Auto-Dial Feature Enable set Offhook Auto-Dial Play registration id Feature

Enable or cancel the call waiting function, only the current call is valid Enable CW per call: Default is 71 Disable CW per call: Default is 70 Allow users to customize the Call Return feature code. Default is Yes Call return: Default is 69 Allow users to customize the Unconditional Forward feature codes. Default is Yes Set the function code to enable unconditional call forwarding (default value is 72) Set the function code to cancel unconditional call forwarding (default value is 73) Whether to enable the call forwarding function when busy Set the function code to enable call forwarding when busy (default value is 90) Set the function code to cancel call forwarding when busy (default value is 91) Whether to enable the function of transferring calls without answering Set the function code to enable call transfer without answering (default value is 92) Set the function code for canceling unanswered call transfer (default value is 93) Allow users to customize the Paging feature code. Default is Yes Paging: Default is 74 Allow users to customize the CW feature codes. Default is Yes Enable DND: Default is 78 Disable DND: Default is 79 Allow users to customize the Blind Transfer feature code. Default is Yes Enable Blind Transfer: Default is 87 Whether to disable the LEC (Line Echo Cancellation) function of the current call (only the current call is prohibited) Set the function code to disable the LEC call function for the current call (default value is 03) Configures whether to Set Offhook Auto-Dial Feature. Enable to set Offhook Auto-Dial (default value is 77). Whether it is displayed as a registration ID.

Enable playing registration id Disable Bellcore Style 3-Way Conference Star Code 3WC Feature Star Code 3WC Forced Codec Feature Forced Codec PCMU Codec Feature PCMU Codec PCMA Codec Feature PCMA Codec G723 Codec Feature G723 Codec G729 Codec Feature G729 Codec iLBC Codec Feature iLBC Codec G722 Codec Feature G722 Codec Ringtone Custom Ring Tone 1 Custom Ring Tone 1 will be used when the caller is Custom Ring Tone 2 Custom Ring Tone 2 will be used when the caller is Custom Ring Tone 3 Custom Ring Tone 3 will be used when the caller is

Set to enable the function code to display the registration ID (default value is 98). Whether to set the function code for the three-party conference will only take effect after closing the Bellcore three-party conference. Allows users to initiate a three-way conference call by dialing a predefined star code. Set the function code of the three-party conference (default value is 23). Whether to enable the function of forcing the speech encoding type Sets the function code to enable forced voice coding type calls (default value is 02) Whether to force PCMU speech encoding Sets the function code to enable forced PCMU voice encoding calls (default value is 7110) enables the PCMU codec for audio encoding and decoding in VoIP communications, Set the function code to enable forced PCMA voice encoding (default value is 7111) Whether to force G723 speech encoding Set the function code to enable forced G723 speech encoding (default value is 723) Whether to force G729 speech encoding Set the function code to enable forced G729 speech encoding (default value is 729) Whether to force iLBC voice encoding Set the function code to enable forced iLBC voice encoding (default value is 7201) Whether to force G722 speech encoding Set the function code to enable forced G722 codec voice encoding (The default value is 722)
Set the caller ID using custom ringtone 1. If the caller number matches, a custom ringtone 1 will be used. Set the caller ID using custom ringtone 2. If the caller number matches, a custom ringtone 2 will be used. Set the caller ID using custom ringtone 3. If the caller number matches, a custom ringtone 3 will be used.

Ring Tone 1-10

Define the Ringtone(s) frequencies, Syntax: c=on1/ off1[- on2/ off2[- on3/ off3]]; (melody on/off time must be a multiple of 1ms) Default value: c=2000/4000;

Call Waiting Tone 1-3 Call Waiting Tone 1-3 used if incoming caller ID is
Call Waiting Tone 1-10

If the caller number matches, a custom call waiting tone will be used
Set the incoming call number using custom call waiting tone
Defines the call waiting tone cadence Syntax: f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]]; (f is in Hz, on and off are calculated in 1ms. on is the ringing time , off is the silent time) Default value: f1=440@-13,c=300/10000;

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 a public IP, keep this field blank.
2. Use random SIP/RTP ports (under the 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 the FXS web page), set this to Yes when the gateway is behind a firewall on a private network.

DTMF Methods
The HT80x V2 support the following DTMF mode: DTMF in-audio DTMF via RTP (RFC2833) DTMF via SIP INFO
Set the priority of DTMF methods according to your preference. This setting should be based on your server DTMF setting.

Preferred Vocoder (Codec)
The HT80x V2 supports the following voice codecs. On Profile pages, choose the order of your favorite codecs:
PCMU/A (or G711µ/a) G729 A/B G723.1 G726 iLBC OPUS G722

Configuring HT80x V2 through Voice Prompts
As mentioned previously, The HT80x V2 has a built-in voice prompt menu for simple device configuration. Please refer to “Understanding HT80x V2 Interactive Voice Prompt Response Menu” for more information about IVR and how to access its menu.

DHCP MODE
Select voice menu option 01 to enable HT80x V2 to use DHCP.
STATIC IP MODE
Select voice menu option 01 to enable HT80x V2 to use STATIC IP mode, then use options 02, 03, 04, and 05 to set up IP address, Subnet Mask, Gateway, and DNS server respectively.
PPPOE MODE
Select voice menu option 01 to allow the HT80x V2 to enable the PPPoE mode. PPPoE Username and Password should be configured from web GUI.
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 menu option 15 to choose firmware and configuration upgrade protocol between TFTP, HTTP, HTTPS, FTP, and FTPS.
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.
WAN PORT WEB ACCESS
Select voice menu option 12 to enable/disable web access from the WAN port. Press 9 in this menu to toggle between enable / disable. The default is disabled.
Configuration through a Central Server
The HT80x V2 can be automatically configured from a central provisioning system. When HT80x V2 boots up, it will send TFTP, HTTP/HTTPS or FTP/FTPS requests to download configuration files, “cfgec74d7xxxxxx” and “cfgec74d7xxxxxx.xml”, where “ec74d7xxxxxx” is the LAN MAC address of the HT80x V2. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue a request a generic configuration file “cfg<Model>.xml” followed by a request to cfg.xml. Configuration file name should be in lower case letters. The configuration data can be downloaded via TFTP, HTTP/HTTPS or FTP/FTPS from the central server. A service provider or an enterprise with large deployment of HT80x V2 can easily manage the configuration and service provisioning of individual devices remotely from a central server.
Grandstream provides a central provisioning system GAPS (Grandstream Automated Provisioning System) to support the automated configuration of Grandstream devices. GAPS uses HTTPS and other communication protocols to communicate with each individual Grandstream device for firmware upgrades, remote reboot, etc. Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or with certain special provisioning settings. At boot-up, Grandstream devices by default point to Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so that they will be redirected to customer’s TFTP, HTTP/HTTPS or FTP/FTPS server for further provisioning. Grandstream also provides configuration tools (Windows and Linux/Unix version) to facilitate the task of generating device configuration files.
The Grandstream configuration tools are free to end users. The configuration tools and configuration templates are available for download from https://www.grandstream.com/support/tools
Register a SIP Account
The HT80x V2 support Profiles that can configure two SIP servers,

Please refer to the following steps in order to register your accounts via web user interface. Access your HT80x V2 web UI by entering its IP address in your favorite browser. Enter your admin password (default: found on the sticker on the back of the unit). Press Login to access your settings. Go to FXS PORT web pages and set the following: 1. Account Active to Yes. 2. Primary SIP Server field with your SIP server IP address or FQDN. 3. Failover SIP Server with your Failover SIP Server IP address or FQDN. Leave empty if not available. 4. Outbound Proxy: Set your Outbound Proxy IP Address or FQDN. Leave empty if not available. 5. SIP User ID: Enter the 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. 6. Authenticate ID: SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User ID. 7. 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. 8. Name: Any name to identify this specific user. Press Apply at the bottom of the page to save your configuration.
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 from your HT80x V2 web interface under Status Port Status Registration (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).

Call Features
The HT80x V2 support all the traditional and advanced telephony features.

Key

Call Features

*02

Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *027201 (iLBC), *02722 (G722).

Disable LEC (per call) Dial “*03″ +” number”. *03
No dial tone is played in the middle.

*16

Enable SRTP

*17

Disable SRTP

*18

Enable SRTP per call. Meaning it’s only valid for the current call.

*19

Disable SRTP per call. Meaning it’s only valid for the current call.

*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 *77 *78 *79 *87 *90
*91
*92
*93 *98 Flash/ Hook #

Enable Paging Call: Dial “*74” and then the destination phone number you want to page.
Enable set Offhook Auto-Dial.
Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.
Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.
Blind Transfer
Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then hang up.
Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone, then hang up
Enable to play the registration ID.
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.

Ethernet Capture
Ethernet Capture involves intercepting a secret key information file during the TLS handshake between the HT8xx and a SIP server to extract the secret key information file during the TLS handshake, the file downloaded is a .txt file including the client’s secret key.
The process includes initiating capture, re-registering the HT8xx with TLS to a SIP server, waiting for a SIP Register request, and then stopping the capture. The goal is to extract the secret key information file, to do that please follow the below steps described below:
1. Start capturing the communication between the HT8xx and the server. to do that go to Maintenenace System Diagnosis Debug, and make sure “With Secret Key information” is set to Yes

1. Re-register the HT8xx account or port to the SIP server using the TLS protocol.

1. Allow the system to wait until the ATA sends a SIP Register request to the server. Once the SIP Register request is sent, stop capturing the communication.
1. After completing the above steps, the expectation is that the captured data will include the secret key information file in a .txt file under the name “sslkeylogfile”
Important Please use Information capture only when authorized since extracting secret key information without proper authorization is considered unethical.
Rebooting HT80x V2 Remotely
Press “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.
UPGRADING AND PROVISIONING
The HT80x V2 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, HTTP/HTTPS or FTP/FTPS; 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 HT80x V2: 1. Access your HT80x V2 UI by entering its IP address in your favorite browser. 2. Enter your admin password (default: found on the sticker on the back of the unit). 3. Press Login to access your settings.
4. Go to Maintenance Upgrade Firmware page and enter the IP address or the FQDN for the upgrade server in
“Firmware Server Path” field and choose to upgrade via TFTP, HTTP/HTTPS or FTP/FTPS. 5. Update the change by clicking the “Save and Apply” button at the bottom of the page. Then “Reboot” or power cycle the
HT80x V2 to update the new firmware.
Firmware Upgrade Page
Upgrading via Local Directory
1. Download the firmware file from the Grandstream web site 2. Unzip it and copy the file in to a folder in your PC
3. From the HT80x V2 web interface Maintenance Upgrade Firmware you can browse your hard drive and select the
folder you previously saved the file. 4. Click “Upload” and wait few minutes until the new program is loaded.
Always check the status page to see that the program version has changed.
the filename in URL for the firmware upgrade has been disabled.
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:

https://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: https://www.solarwinds.com/products/freetools/free_tftp_server.aspx https://pjo2.github.io/tftpd64/ 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. 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 HT80x V2. End users can also choose to download a free HTTP server from https://httpd.apache.org/ or use a Microsoft IIS web server.
Firmware and Configuration File Prefix and Postfix
Firmware Prefix and Postfix allow the device to download the firmware name with the matching Prefix and Postfix. This makes it possible to store all the firmware with different version in one single directory. Similarly, Config File Prefix and Postfix allow the 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.
Firmware File Prefix and Postfix

Managing Automatic Upgrade
When “Automatic Upgrade” is set “Check” the auto check will be done in the minute specified in this field. If set to “Every day in 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
Automatic Upgrade
Managing Device using GDMS Cloud
The GDMS Device Management System is a cloud solution from Grandstream that facilitates the provisioning and management of Grandstream devices, including the HT8xx ATAs. The platform enables users to centralize the control and management of multiple HT8xx devices, deployed across various locations, on a single interface. It provides functionalities to upgrade, add, configure, and monitor HT8xx devices from one centralized dashboard. For more details on the supported HT8xx models and their deployment to the GDMS platform, refer to the following guide: GDMS Unified Communications User Guide. If you don’t have a GDMS account yet, visit the following website to create one: GDMS.Cloud.
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, HTTP/HTTPS or FTP/FTPS. The Config Server Path is the TFTP, HTTP/HTTPS or FTP/FTPS 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 1 to 5 (Could be extended in the future) digit numeric numbers. i.e., P30 is associated with the “NTP
Server” in the Web GUI System Settings Time and Language Time Zone NTP Server. For a detailed parameter list, please refer to the corresponding firmware release configuration template.
When the HT80x V2 boots up or reboots, it will send a request to download the xml file named “cfgxxxxxxxxxxxx.xml” followed by the binary file named “cfgxxxxxxxxxxxx”, 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<Model>.xml file and then download cfg.xml. The configuration file name should be in lower case letters.
HT818 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: <option 67 bootfile>
Step 2: cfg<MAC>.xml cfg<MAC>cfg<Model>.xml cfg.xml
Notes:
1. The Only acceptable Config file formats to be used when provisioning are XML or binary.
2. Make sure the MAC header on the Config file is the provisioned device’s MAC address or you can remove the header completely.
3. When the <option 67 bootfile> is downloaded from the server, the cfg<MAC>.xml is not requested.
For more details on XML provisioning, please refer to:

SIP Device Provisioning Guide

The filename in URL for config provision has been disabled.
Configuration and Firmware Upgrade through Resync SIP NOTIFY
HT80x V2 supports triggering firmware and configuration upgrade using Resync SIP NOTIFY. The event:resync NOTIFY allows the device to re-synchronize its configuration by checking the provisioning server to download any updates.
Below is an example of a resync SIP NOTIFY:
NOTIFY sip:device@domain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK-d4f2-9a75 Max-Forwards: 70 From: “Provisioning Server” <sip:provisioning@domain.com>;tag=prov-server To: <sip:device@domain.com>;tag=device-987654 Call-ID: 987654xyz321 CSeq: 12345 NOTIFY Event: resync Content-Type: application/simple-message-summary Content-Length: 45
CA Bundle Manifest
CA (Certification Authority) Bundle Manifest is a collection of trusted security certificates used to verify the authenticity of websites or services. It helps ensure secure and encrypted connections by confirming the identity of the remote server, and protecting against unauthorized access and data interception.
you can access the CA Bundle Manifest by clicking on the bottom corner of the web page as displayed in the screenshot below:

This will redirect you to a web page displaying a list of certification names and details, with their corresponding expiration dates:
RESTORE FACTORY DEFAULT SETTINGS
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 HT80x V2. 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:

Key

Mapping

0-9

22 (press the “2” key twice, “A” will show on the LCD)

222

2222

33 (press the “3” key twice, “D” will show on the LCD)

333

3333

MAC Address Key Mapping
For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”
Reset from Web Interface (Reset Type)
1. Access your HT80x V2 UI by entering its IP address in your favorite browser. 2. Enter your admin password (default: found on the sticker on the back of the unit). 3. Press Login to access your settings.
4. Go to Maintenance Restore Factory Reset 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 Reset: This will reset only the data related with a service provider like SIP server, sip user ID, provisioning and others.

Factory Reset from the phone will be disabled if the “Lock keypad update” is set to “Yes”.
If the HT80x V2 were 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 HT80x V2 UI by entering its IP address in your favorite browser. 2. Go to Port Settings Profile # page. 3. Set “Allow SIP Factory Reset” to “Yes”. (Default is No) 4. Once a SIP NOTIFY with “event: reset” is received, the ATA will perform factory reset.
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.
CHANGE LOG
This section documents significant changes from previous versions of the admin guide for HT80x V2. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.
Firmware Version 1.0.3.5
Added support for 55Vrms Ring Voltage (For New Zealand/Australia) SLIC. (P4234/4235 Ring Power. 0 45Vrms default, 3 50Vrms, 4 55Vrms). [Ring Power] Added “HELD Use LLDP Information”. [HELD Use LLDP Information] Added support to Enable LLDP. [Enable LLDP] Added “LLDP TX Interval”. [LLDP TX Interval] Added support to Enable CDP. [Enable CDP] Added “Use Random SIP Registration Failure Retry Wait Time”. [Use Random SIP Registration Failure Retry Wait Time] Added ability to configure minimum and maximum values for Registration retry timer. [Random SIP Registration Failure Retry Wait Time Range] Added support to configure a static DNS SRV record. [Static DNS Cache] Added support for firmware upgrade via resync SIP Notify. [Configuration and Firmware Upgrade through Resync SIP NOTIFY] Firmware Version 1.0.1.16
This is the initial Firmware.

Documents / Resources

GRANDSTREAM HT801 V2 Analog Telephone Adapters [pdf] User Guide
HT801 V2 Analog Telephone Adapters, HT801 V2, Analog Telephone Adapters, Telephone Adapters, Adapters

References

User Manual

HT801 V2 Analog Telephone Adapters

Product Information

Specifications

Features

Technical Specifications

Product Usage Instructions

Installation

Configuration

Usage

FAQs

Q: How can I reset the device to factory settings?

Q: What network protocols are supported by the HT801 V2 and
HT802 V2?

Q: Can I use the HT801 V2/HT802 V2 with Grandstream’s UCM
series of IP PBXs?

Documents / Resources

References

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