HT80x - Administration Guide

  • Updated on August 3, 2022

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

PRODUCT OVERVIEW

The HT801 is a one-port analog telephone adapter (ATA) while the HT802 is a 2-port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection, and auto provisioning options enable users to take advantage of VoIP on analog phones and enable service providers to offer high-quality IP service. The HT801/HT802 are an ideal ATA for individual use and for large-scale commercial IP voice deployments.

Feature Highlights

The following table contains major features of HT801 and HT802:

C:\Users\User\Desktop\Documentation\HT8xx\HT801\HT801-image\front.png

HT801

HT802

  • 1 SIP profile through 1 FXS port on HT801, 2 SIP profiles through 2 FXS ports on HT802,
and a single 10/100Mbps port on both models. 3-way voice conferencing. Wide range of caller ID formats. Advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multi-language prompts, flexible dial plan,
  • and more.
  • T.38 Fax for creating Fax-over-IP and GR-909 Line Testing Functionalities.
  • TLS and SRTP security encryption technology to protect calls and accounts.
  • Automated provisioning options include TR-069 and XML config files.
  • Failover SIP server automatically switches to secondary server if main server loses connection.
  • Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning.
Table 1: HT801/HT802 Features at a Glance

HT801/HT802 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 and the HT802.

HT801

HT802

Interfaces

Telephone Interfaces

One (1) RJ11 FXS port

Two (2) RJ11 FXS ports

Network Interface

One (1) 10/100Mbps auto-sensing Ethernet port (RJ45)

LED Indicators

POWER, INTERNET, PHONE

POWER, INTERNET, PHONE1,
PHONE2

Factory Reset Button

Yes

Voice, Fax, Modem

Telephony Features

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

Voice Codecs

G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.723.1, G.729A/B, G.726, G.722, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation

Fax over IP

T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through.

Short/Long Haul Ring Load

5 REN: Up to 1km on 24 AWG

2 REN: Up to 1km on 24 AWG

Caller ID

Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID.

Disconnect Methods

Busy Tone, Polarity Reversal/Wink, Loop Current

Signalling

Network Protocols

TCP/IP/UDP, RTP/RTCP, FTP/FTPS, HTTP/HTTPS, ARP/RARP, ICMP, DNS, DDNS, DHCP, NTP, TFTP, SSH, Telnet, STUN, SIP (RFC3261), SIP over TCP/TLS, SRTP, TR-069.

QoS

Layer 3 (ToS, DiffServ, MPLS)

DTMF Methods

In-audio, RFC2833 and/or SIP INFO

Provisioning and Control

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

Security

Media

SRTP

Control

TLS/SIPS/HTTPS/SSH/Telnet

Management

Syslog support, SSH, Telnet remote management using web browser

Physical

Universal Power Supply

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

Environmental

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

Dimensions and Weight

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

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

Compliance

Compliance

FCC Part15B EN55032, EN55024, EN61000-3-2, EN61000-3-3, EN60950-1, AS/NZS CISPR22, AS/NZS60950.1, S003

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

Table 2: HT801/HT802 Technical Specifications

GETTING STARTED

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

Equipment Packaging

The HT801 ATA package contains:

Figure 1: HT801 Package Contents

The HT802 ATA package contains:

Figure 2: HT802 Package Contents

Note: Check the package before installation. If you find anything missing, contact your system administrator.

HT801/HT802 Ports Description

The following figure describes the different ports on the back panel of the HT801.

Figure 3: HT801 Back Panel

The following figure describes the different ports on the back panel of the HT802.

Figure 4: HT802 Back Panel

Phone for HT801

Phone 1 & 2 for HT802

Connects the analog phones/fax machines to the phone adapter using an RJ-11 telephone cable.

Internet port

Connects the phone adapter to your router or gateway using an Ethernet RJ45 network cable.

Micro-USB Power

Connects the phone adapter to PSU (5V – 1A).

Reset

Factory reset button. Press for 7 seconds to reset factory default settings.

Table 3: Definition of the HT801/HT802 Connectors

Connecting HT801/HT802

The HT801 and HT802 are designed for easy configuration and easy installation, to connect your HT801 or HT802, please follow the steps above:

  1. Insert a standard RJ11 telephone cable into the phone port and connect the other end of the telephone cable to a standard touch-tone analog telephone.
  2. Insert the Ethernet cable into the internet or LAN port of the HT801/ht802 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.)
  3. Insert the power adapter into the HT801/HT802 and connect it to a wall outlet.

Power, Ethernet, and Phone LEDs will be solidly lit when the HT801/HT802 is ready for use.

Figure 5: Connecting the HT801/HT802

HT801/HT802 LEDs Pattern

There are 3 LED buttons on HT801 and 4 LED buttons on HT802 that help you manage the status of your Handy Tone.

Figure 6 : HT801/HT802 LEDs Pattern

LED LightsStatus

Power LED

The Power LED lights up when the HT801/HT802 is powered on, and it flashes when the HT802 is booting up

Internet LED

The Ethernet LED lights up when the HT801/HT802 is connected to your network through the Ethernet port, and it flashes when there is data being sent or received.

Phone LED for HT801

Phone LED 1&2 for HT802

The phone 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 indicate
  • voicemail
Table 4: HT801/HT802 LEDs Pattern Description

CONFIGURATION GUIDE

The HT801/HT802 can be configured in one of two ways:

  • The IVR voice prompt menu.
  • The Web GUI is embedded on the HT801/HT802 using the PC’s web browser.

Obtain HT801/HT802 IP Address via Connected Analogue Phone

HT801/HT802 is by default configured to obtain the IP address from the DHCP server where the unit is located. To know which IP address is assigned to your HT801/HT802, you should access the “Interactive Voice Response Menu” of your adapter via the connected phone and check its IP address mode.

Please refer to the steps below to access the interactive voice response menu:

  1. Use a telephone connected to the phone ports of your HT801/HT802.
  2. Press *** (press the star key three times) to access the IVR menu and wait until you hear “Enter the menu option “.
  3. Press 02 and the current IP address will be announced.

Understanding HT801/HT802 Interactive Voice Prompt Response Menu

The HT801/HT802has a built-in voice prompt menu for simple device configuration which lists actions, commands, menu choices, and descriptions. The IVR menu works with any phone connected to the HT801/HT802. Pick up the handset and dial “***” to use the IVR menu.

MenuVoice PromptOptions

Main Menu

“Enter a Menu Option”

Press “*” for the next menu option

Press “#” to return to the main menu

Enter 01-05, 07,10, 13-17,47 or 99 menu options

01

“DHCP Mode”,

“Static IP Mode”

Press “9” to toggle the selection

If using “Static IP Mode”, configure the IP address information using menus 02 to 05.

If using “Dynamic IP Mode”, all IP address information comes from the DHCP server automatically after reboot.

02

“IP Address “+ IP address

The current WAN IP address is announced

If using “Static IP Mode”, enter a 12-digit new IP address. You need to reboot the HT801/HT802 for the new IP address to take Effect.

03

“Subnet “+ IP address

Same as menu 02

04

“Gateway “+ IP address

Same as menu 02

05

“DNS Server “+ IP address

Same as menu 02

07

Preferred Vocoder

Press “9” to move to the next selection in the list:

  • PCM U / PCM A
  • iLBC
  • G-726
  • G-723
  • G-729
  • OPUS
  • G722

10

“MAC Address”

Announces the Mac address of the unit.

13

Firmware Server IP Address

Announces current Firmware Server IP address. Enter a 12-digit new IP address.

14

Configuration Server IP Address

Announces current Config Server Path IP address. Enter a 12-digit new IP address.

15

Upgrade Protocol

Upgrade protocol for firmware and configuration update. Press “9” to toggle between TFTP / HTTP / HTTPS / FTP / FTPS. The default is HTTPS.

16

Firmware Version

Announces Firmware version information.

17

Firmware Upgrade

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

  • always check
  • check when pre/suffix changes
  • never upgrade

47

“Direct IP Calling”

Enter the target IP address to make a direct IP call, after dial tone. (See “Make a Direct IP Call”.)

86

Voice Mail

Access to your voice mails messages.

99

“RESET”

Press “9” to reboot the device

Enter the MAC address to restore the factory default setting

(See Restore Factory Default Setting section)

701-702

Phone calls between different ports of the same HT802

HT802 supports inter-port calling from the voice menu.

70X (X is the port number)

“Invalid Entry”

Automatically returns to the main menu

“Device not registered”

This prompt will be played immediately after off-hook If the device is not registered, and the option “Outgoing Call without Registration” is in NO

Table 5: Voice Prompt Menu

Five success tips when using the voice prompt

  • “*” shifts down to the next menu option and “#” returns to the main menu.
  • “9” functions as the ENTER key in many cases to confirm or toggle an option.
  • All entered digit sequences have known lengths – 2 digits for 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 key 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/HT802 embedded Web server responds to HTTP GET/POST requests. Embedded HTML pages allow a user to configure the HT801/HT802 through a web browser such as Google Chrome, Mozilla Firefox, and Microsoft’s IE.

Accessing the Web UI

  1. Connect the computer to the same network as your HT801/HT802.
  2. Make sure the HT801/HT802 is booted up.
  3. You may check your HT801/HT802 IP address using the IVR on the connected phone.

Please see Obtain HT801/HT802 IP Address via Connected Analogue Phone

  1. Open the Web browser on your computer.
  2. Enter the HT801/HT802’s IP address in the address bar of the browser.
  3. Enter the administrator’s password to access the Web Configuration Menu (the default password is admin).

Notes:

  • The computer must be connected to the same sub-network as the HT801/HT802. This can be easily done by connecting the computer to the same hub or switch as the HT801/HT802.
  • Recommended Web browsers:
    • Microsoft Internet Explorer: version 10 or higher.
    • Google Chrome: version 58.0.3 or higher.
    • Mozilla Firefox: version 53.0.2 or higher.
    • Safari: version 5.1.4 or higher.
    • Opera: version 44.0.2 or higher.

Web UI Access Level Management

There are two default passwords for the login page:

User LevelUsernamePasswordWeb Pages Allowed

End User Level

user

123

Only Status and Basic Settings

Administrator Level

admin

admin

All pages

Viewer Level

viewer

viewer

View all pages. Changes are not allowed.

Table 6: Web access level management

The password is case-sensitive with a maximum length of 25 characters. When changing any settings, always submit them by pressing the Update or Apply button at the bottom of the page. After submitting the changes in all the Web GUI pages, reboot HT801/HT802 to have the changes take effect if necessary; most of the options under the Advanced Settings and FXS Port (x) pages require a reboot.

Saving the Configuration Changes

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

Changing Admin Level Password

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

Changing User Level Password

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

Changing Viewer Password

  1. Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
  2. Enter your admin password (default: admin).
  3. Press Login to access your settings.
  4. Go to Basic Settings 🡪 New Viewer Password and enter the new viewer password.
  5. Confirm the new viewer password.
  6. Press Apply at the bottom of the page to save your new settings.
Figure 9: Viewer Level Password

Changing HTTP Web Port

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

Web Configuration Pages Definitions

This section describes the options in the HT801/HT802 Web UI. As mentioned, you can log in as an administrator or an end user.

  • Status: Displays the system info, network status, account status, and line options.
  • Basic Settings: Configures the end user level password, IP address modes, web access, time zone settings, and language.
  • Advanced Settings: Configures networks, upgrading and provisioning, TR-069, security settings, date and time, Syslog, audio settings, call settings and call progress tones.
  • FXS Port (1,2): Configures SIP settings, SIP registration, accounts settings, NAT settings, call features, and ring tones.

Status Page Definitions

Account Status

MAC Address

Shows device ID in hexadecimal format. This is needed by network administrators for troubleshooting. The MAC address will be used for provisioning and can be found on the label on the original box and on the label located on the bottom panel of the device.

IPv4 Address

Displays assigned IPv4 address.

IPv6 Address

Displays assigned IPv6 address.

VPN IPv4 Address

Displays assigned OpenVPN IPv4 address.

VPN IPv6 Address

Displays assigned OpenVPN IPv6 address.

Product Model

Displays product model info. Default is HT801/HT802.

Hardware Version

Displays the hardware revision information and the part number.

Software version

  • Program: Specifies Program version. This is the main firmware release number, which is always used for identifying the software system of the HT801/HT802.
  • Bootloader: Specifies Boot version.
  • Core: Specifies Core version.
  • Base: Specifies Base version.
  • CPE: Specifies CPE version. CPE version is displayed only when HT801/HT802 is connected to an ACS using TR-069 protocol.

Software Status

Indicates actual software status.

System Up Time

Indicates actual system time and uptime since the last reboot.

CPU Load

Indicates CPU load (%)

Network Cable Status

Indicates physical network cable status:
Status (Up/Down), Speed (Mbps), Operational Mode (Full/Half Duplex)

PPPoE Link Up

Indicates PPPoE connection status.

NAT

Indicates the type of NAT when it is configured.

Port Status

Displays relevant information regarding the FXS ports, their registration, current status, and their appropriate User ID.

Port Options

Displays DND and call forward information on FXS ports.

CDR File

Download, Preview. Or Delete call history records from the web GUI. Only the last 1000 records will be available.

SIP File

Download, Preview or, Delete locally stored SIP trace.

Note: Send SIP Log” must be enabled to be able to capture the trace.

Provision

Displays provisioning status.

Core Dump

Provides generated core dump file if unit malfunctions.

Clean will be displayed if no issues.

GR909

Click on “Test Page” to be redirected to the

lines tests page (Hazardous Potential, Foreign Electromotive Forces, Resistive Faults, Receiver Off-hook

and Ringer Equivalent Number)

Table 8: Status Page Definitions

Basic Settings Page Definitions

Basic Settings

New End User Password

Configures user level password. Case sensitive and max. Length of 25 characters.

Confirm End User Password

Re-enter the end user password to confirm change user password on web GUI to avoid typo or mistakes.

New Viewer Password

Configures viewer-level password. Case sensitive and max. length of 25 characters.

Confirm Viewer Password

Re-enter the viewer password to confirm change viewer password on web GUI to avoid typo or mistakes.

Web Session Timeout

Configure timer to logout web session during idle. Default is 10 min. Range is 2-60 min.

Web Access Attempt Limit

Configure attempt limit before lockout. Default is 5. Range is 1-10.

Lockout time interval

If login attempt failed 5 times, login would be locked out for the time length. (Default 15 mins. Range 1-15 min).

Web Access Mode

Allows users to choose the Web Access Mode between “HTTPS”, “HTTP” and “Disabled”. If “Disabled” is selected, web UI access will be disabled.

By default, “HTTP” is selected.

HTTP Web Port

Customizes HTTP port used to access the HT801/HT802 web UI. Default is 80.

HTTPS Web Port

Customizes HTTPS port used to access the HT801/HT802 web UI. Default is 443.

Disable SSH

Enables/disables the SSH access. Default is No (enabled).

SSH Port

Allows users to self-configure SSH Port number.

By default, the port number is 22.

SSH Idle Timeout

Configures SSH session timeout. [0 – 86400] seconds; Default is 0.

Disable Telnet

Enables/disables the Telnet access. Default is Yes (disabled).

Telnet Port

Allows users to self-configure Telnet Port number. By default, the port number is 23.

Telnet Idle Timeout

Configure Telnet session timeout. [0 – 86400] seconds; Default is 0.

Whitelist for WAN Side

Users can configure the whitelist for WAN Side to be used for remote management.

Multiple IPs are supported and need to be separated by “space”.

Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

Note: If both blacklist and whitelist are not empty, the blacklist is processed first, followed by the whitelist.

Blacklist for WAN Side

Users can configure the blacklist for WAN Side to ban WAN side web access.

Multiple IPs are supported and need to be separated by “space”.

Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

Note: If both blacklist and whitelist are not empty, the blacklist is processed first, followed by the whitelist.

Internet Protocol

Selects one of the following IP protocol modes:

  • IPv4 Only: Enforce IPv4 protocol only.

  • IPv6 Only: Enforce IPv6 protocol only.

  • Both, Prefer IPv4: Enable both IPv4 and IPv6 and prefer IPv4.

  • Both, prefer IPv6: Enable both IPv4 and IPv6 and prefer IPv6.

Note: Make sure to reboot the phone for the changes to take effect.

IPv4 Address

Allows users to configure the appropriate network settings on the HT80x to obtain IPv4 address. Users could select “DHCP”, “Static IP” or “PPPoE”. By default, it is set to “DHCP”.

Dynamically assigned via DHCP

All the field values for the static IP mode are not used (even though they are still saved in the flash memory.) The HT801/802 acquires its IP address from the first DHCP server it discovers from the LAN it is connected.

  • DHCP hostname: Specifies the name of the client. The name may or may not be qualified with the local domain name. This field is optional but may be required by ISP.

  • DHCP domain name: Specifies the domain name that client should use when resolving hostname via the Domain Name System.

  • DHCP vendor class ID: Exchanges vendor class ID by clients and servers to convey particular configuration or other identification information about a client. Default is HT8XX.

Use PPPoE

Set the PPPoE account settings. If selected, ATA attempt to establish a PPPoE session if any of the PPPoE fields is set.

  • PPPoE account ID: Defines the PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection.

  • PPPoE password: Specifies the PPPoE account password.

  • PPPoE Service Name: Defines PPPoE service name. If your ISP uses a service name for the PPPoE connection, enter the service name here. This field is optional. Default is blank.

Preferred DNS server

Specifies preferred DNS server to use when DHCP or PPPoE are set.

Statically configured as IP address

Configure IP address, subnet Mask, default router IP address, 1st preferred DNS server, 2nd preferred DNS server. These fields are set to zero by default.

IPv6 Address

Allows users to configure the appropriate network settings on the HT80x to obtain IPv6 address. Users could select “DHCP”, “Static IP”. By default, it is set to “DHCP”.

  • DHCP mode: all the field values for the static IP mode are not used (even though they are still saved in the flash memory.) The HT801/HT802 acquires its IP address from the first DHCP server it discovers from the LAN it is connected.

  • Static IP mode: 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).

Enable Management Interface

Allows administrator to setup a Virtual Network Interface on top of the physical interface for device management. Default is No.

Management Access

Chooses whether to access using “Management Interface Only” (Default) Or “Both Service and Management Interfaces”

Enable SNMP Through Management Interface

This feature allows users to route SNMP packets through management interface. Default setting is No.

Management Interface IPv4 Address

Configures Voice VLAN Type :

  • Default is dynamically assigned via DHCP

Or,

  • statically configured as:

  • IP Address : Default is 192.168.100.100

  • Subnet Mask : Default is 255.255.255.0

  • Default Router  : Default is 192.168.100.1

  • DNS Server 1 : Default is 0.0.0.0

  • DNS Server 2 : Default is 0.0.0.0

  • 802.1Q/VLAN Tag

vlan tagging : [0 – 4094]; Default is 0

  • 802.1p priority value :

priority : [0 – 7]; Default is 0

Time Zone

Selects time zone to define date/time on the device.

Self-Defined Time Zone

Allows users to define their own time zone.

Allow DHCP server to set Time Zone

Obtains time zone setting (offset) from a DHCP server using DHCP Option 2; it will override selected time zone. If set to “No”, the analogue adapter will use selected time zone even if provided by DHCP server.

The Default setting is Yes.

Language

Configures the languages of the voice prompt and web interface, except Spanish that it is only in IVR. Available languages: English, Chinese, Traditional Chinese, Russian, Spanish IVR.

Reset Type

Gives the administrator the option to restore default configuration on the HT801/HT802.

There are 3 types of factory reset:

  • ISP Data Reset: All VoIP related configuration (mainly everything located on FXS page).

  • VOIP Data Reset: All ISP (Internet Service Provider) configuration which may affect the IP address.

  • Full Reset: Both VoIP and ISP related configuration at the same time.

Note: After you choose the reset type, you must click the reset button to take effect.

Table 9: Basic Settings Page

Advanced Settings Page Definitions

New Admin Password

Defines the administrator level password to access the Advanced Web Configuration page. This field is case sensitive. Only the administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reasons after clicking update and saved. Max. length is 39 characters and minimum are 8, the password should contain at least 1 upper case and 1 lower case and 1 special character.

Confirm Admin Password

Confirms the new admin password.

Disable User Level Web Access

Disable or enable User Level Web Access. Default is No.

Disable Viewer Level Web Access

Disable or enable Viewer Level Web Access Default is No.

Layer 2 QoS

Sets values for 802.1Q/VLAN Tag. Default is 0. Valid range is 0-4094.

SIP 802.1p. Default is 0. Valid range is 0-7.

RTP 802.1p. Default is 0. Valid range is 0-7.

Blacklist for WAN Side Port

It could be either port range or single port separated by a “,”

Example: “5000-6000, 7000 “.

STUN Server is

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

Keep-alive interval

Sends periodically a blank UDP packet to SIP server to keep the “ping hole” on the NAT router open. Default is 20 seconds.

Use STUN to detect network connectivity

Uses STUN keep-alive to detect WAN side network problems. If keep-alive request does not yield any response for configured number of times (minimum 3), the device will restart the TCP/IP stack.

If the STUN server does not respond when the device boots up, the feature is disabled.

Default setting is No.

Use DNS to detect network connectivity

Uses DNS to detect WAN side network problems.

Default setting is “No”.

Use ARP to detect network connectivity

Uses ARP to check the network connectivity.

Default is “Yes”.

Verify host when using HTTPS

Enables / disables the host verification when using HTTPS.

Upgrade via

Selects firmware upgrade/provisioning method: TFTP, HTTP, HTTPS, FTP or FTPS. Default is HTTPS.

Firmware Server Path

Sets IP address or FQDN of firmware server. The URL of the server that hosts the firmware release. Default is fm.grandstream.com/gs.

Config Server Path

Sets IP address or FQDN of configuration server. The URL of the server that hosts the configuration file to provision HT8xx.

Default is fm.grandstream.com/gs.

XML Config File Password

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

HTTP/HTTPS FTP/FTPS Username

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 Prefix

Checks if configuration files are with matching prefix before downloading them. This field enables user to store different configuration files in one directory on the provisioning server.

Config File Postfix

Checks if configuration files are with matching postfix before downloading them. This field enables user to store different configuration files in one directory on the provisioning server.

Allow DHCP Option 66 to Override Server

Obtains configuration and upgrade server’s information using options 66 from DHCP server.

Note: If DHCP Option 66 is enabled, the HT8xx will attempt downloading the firmware file from the server URL provided by DHCP, even though Config Server Path is left blank.

The server URL provided by DHCP can include authentication credentials using following format: “username:password@Provisioning_Server_IP”.

3CX Auto Provision

Sends multicast “SUBSCRIBE” message for provisioning at booting stage, used for PnP (Plug-and-Play) configuration. Default is Yes.

Automatic Upgrade

Specifies when the firmware upgrade process will be initiated; there are 4 options:

No: The HT8xx 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).

Always Check for New Firmware at Boot up

Configures the HT8xx to always search for the new firmware at boot up. During the boot stage, the HT8xx will contact the firmware upgrade server to search for a new firmware, when available it will start the upgrade process, otherwise it will boot normally.

Check New Firmware only when F/W pre/suffix changes

Configure the HT8xx to search for the new firmware when the firmware prefix / suffix changes.

When this option is selected, the HT8xx will check for updates only when the pre/suffix has been changed.

Always Skip the Firmware Check

Configures the HT8xxto skip the firmware check, when this option is selected the HT8xx will always skip searching for a new firmware.

Configuration File Types Allowed

Allows users to configure provision configuration file type in xml file only or all file types.

Disable SIP NOTIFY Authentication

Disables the SIP NOTIFY Authentication on the phone adapter. If set to “Yes”, the phone adapter will not challenge NOTIFY with 401.

The default setting is “No”

Authenticate Conf File

Authenticates configuration before being accepted. This protects the configuration from unauthorized modifications. Default is No.

Validate Server Certificates

This feature allows users to validate server certificates with our trusted list of TLS connections. The device needs to reboot after changing the setting.

Default is enabled.

Trusted CA certificates A

Uses the certificate for Authentication if “Check Domain Certificates” is set to “Yes” under “Account”–> “SIP Settings”.

Trusted CA certificates B

Uses the certificate for Authentication if “Check Domain Certificates” is set to “Yes” under “Account”–> “SIP Settings”.

Load CA Certificates

This feature allows user to specify which certificate to trust when performing server authentication.

Build-in trusted : (Default) Build-in trusted certificates

Custom trusted certificate: Uploaded Certificates

All trusted Certificates: Both built-in and uploaded Certificates

SIP TLS Certificate

Specifies SSL certificate used for SIP over TLS is in X.509 format. The HT8xx has built-in private key and SSL certificate.

Maximum supported length is 4069.

SIP TLS Private Key

Specifies TLS private key used for SIP over TLS is in X.509 format.

Maximum supported length is 4069.

SIP TLS Private Key Password

Specifies SSL Private key password used for SIP Transport in TLS/TCP.

Custom Certificate

(Private Key + Certificate)

Allows users to update to the device their own certificate signed by custom CA certificate to manage client authentication.

Enable TR-069

Sets the phone adapter system to enable the “CPE WAN Management Protocol” (TR-069). Default setting is Yes.

ACS URL

Specifies URL of TR-069 Auto Configuration Servers (e.g., http://acs.mycompany.com), or IP address.

Default setting is: “https://acs.gdms.cloud”

ACS Username

Enters username to authenticate to ACS.

ACS Password

Enters password to authenticate to ACS.

Periodic Inform Enable

Sends periodic inform packets to ACS. Default is Yes.

Periodic Inform Interval

Sets frequency that the inform packets will be sent out to ACS.

Default is 86400 seconds.

Connection Request Username

Enters username for ACS to connect to the HT8xx.

Connection Request Password

Enters password for ACS to connect to the HT8xx.

Connection Request Port

Configures the TR-069 connection request port. The value range is 0 to 65535.Default is 7547

CPE SSL Certificate

Configures the Cert File for the phone adapter to connect to the ACS via SSL.

CPE SSL Private Key

Specifies the Cert Key for the phone adapter to connect to the ACS via SSL.

Enable SNMP

Default is No.

SNMP Version

Choose between (Version 1, Version 2c, or Version 3).

SNMP Port

Listening Port of SNMP daemon (Default 161).

SNMP Trap IP Address

IP address of trap destination. Up to 3 trap destinations are supported. Users should enter the IP addresses separated with comma (,).

SNMP Trap port

Port of Trap destination (Default 162)

SNMP Trap Version

Choose between (Version 1, Version 2c, or Version 3).

SNMP Trap Interval

Time interval between traps (Default is 5).

SNMPv1/v2c Community

Name of SNMPv1/v2c community.

SNMPv1/v2c Trap Community

Name of SNMPv1/v2c trap community.

SNMPv3 Username

Username for SNMPv3.

SNMPv3 Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

authUser: Users with security level authNoPriv and context name as auth.

privUser: Users with security level authPriv and context name as priv.

SNMPv3 Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA.”

SNMPv3 Privacy Protocol

Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.

SNMPv3 Authentication Key

Enter the Authentication Key.

SNMPv3 Privacy Key

Enter the Privacy Key.

SNMPv3 Trap Username

Username for SNMPv3 Trap.

SNMPv3 Trap Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

authUser: Users with security level authNoPriv and context name as auth.

privUser: Users with security level authPriv and context name as priv.

SNMPv3 Trap Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA”.

SNMPv3 Trap Privacy Protocol

Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.

SNMPv3 Trap Authentication Key

Enter the Trap Authentication Key.

SNMPv3 Trap Privacy Key

Enter the Trap Privacy Key.

Enable RADIUS Web Access Control

Default is No.

Action upon RADIUS Auth Server Error

Choose action upon RADIUS server error. Default is Authenticate Locally (Default Authenticate Locally)

RADIUS Auth Server Address

Address of RADIUS Auth server.

RADIUS Auth Server Port

Port of RADIUS Auth server.

RADIUS Shared Secret

Set RADIUS shared secret.

RADIUS VSA Vendor ID

Configure RADIUS VSA Vendor ID to match RADIUS server’s configuration. Default is 42397 for Grandstream Networks Inc.

RADIUS VSA Access Level Attribute

Configure RADIUS VSA Access Level Attribute to match RADIUS server’s configuration. Incorrect setting would cause Radius authenticate fail.

Enable DDNS

Allow users to use DDNS.

DDNS Server

Selects DDNS Server: dyndns, freedns.afraid.org, zoneedit.com, no-ip.com, oray.net.

Default is dyndns.

DDNS Username

64 characters as Max String Length.

DDNS Password

64 characters as Max String Length.

DDNS Hostname

64 characters as Max String Length.

DDNS Hash

64 characters as Max String Length.

Enable OpenVPN®

Allow user to enable OpenVPN®. Default is No.

OpenVPN® Server Address

Specify the IP address or FQDN for the OpenVPN® Server.

OpenVPN® Port

Specify the listening port of the OpenVPN® server. Default is 1194

OpenVPN® Interface type

Specify the Interface type of OpenVPN® whether TAP or TUN. Default is TUN.

OpenVPN® Transport

Specify the Transport Type of OpenVPN® whether UDP or TCP. The default is UDP.

Enable OpenVPN® LZO Compression

Enable OpenVPN® LZO Compression. Default is Yes.

OpenVPN® Encryption

Select the OpenVPN® Encryption. Default is BF-CBC 128 bit (default key).

OpenVPN® Digest

Select the OpenVPN® Digest. Default is SHA1.

OpenVPN® CA

Specifies the OpenVPN® CA. Maximum Character Number is 8192.

OpenVPN® Certificate

Specifies the OpenVPN® Certificate. Maximum Character Number is 8192.

OpenVPN® Client Key

Specifies the Client Key. Maximum Character Number is 8192.

OpenVPN® Client Key Password

Configures the OpenVPN® Client Key Password. Maximum Length is 64.

OpenVPN® username

Configure the OpenVPN® username.

OpenVPN® password

Configure the OpenVPN® password

System Ring Cadence

Configuration option is set ring cadence on FXS port for all incoming calls.

Syntax: c=on1/off1-on2/off2-on3/off3; (3 cadences maximum)

Default is set to c=2000/4000; (US standards)

Call Progress Tones:

Dial Tone

Ringback Tone

Busy Tone

Reorder Tone

Confirmation Tone

Call Waiting Tone

Prompt Tone

Conference Party Hangup Tone *

Special Proceed Indication Tone

Using these settings, users can configure tone frequencies and cadence according to their preference. By default, they are set to North American frequencies.

Configure these settings with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In order to set a continuous tone, OFF should be zero. Otherwise, it will ring ON ms and a pause of OFF ms and then repeat the pattern.

Example configuration for N.A.

Dial tone:

f1=350@-13, f2=440@-13, c=0/0;

Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; […]

(Note: freq: 0 – 4000Hz; vol: -30 – 0dBm)

* “Conference Party Hang-up Tone” will apply only if the “Special Feature” is set to “MTS”.

Special Proceed Indication Tone: This feature allows user to configure the tone played when user goes offhook and there is voicemail on the subscribed mailbox. Need to set ‘MWI Tone’ to ‘Special Proceed Indication Tone’ to use this feature.

Prompt Tone Access Code

Key pattern to get Prompt Tone. Maximum 20 digits.

Lock Keypad Update

Configuration update via keypad (analog phone connected to FXS port keypad using IVR menu) is disabled if set to Yes.

Disable Voice Prompt

Voice prompt is disabled if set to Yes.

Disable Direct IP Call

Direct IP call is disabled if set to Yes.

Play Busy Tone When Account is unregistered

When this feature is set to Yes, device will play busy tone when the FXS port account is not registered, and the attached analog phone is offhook.

Blacklist for Incoming Calls

Allow users to block incoming calls from specific list of numbers.

Maximum allow 10 SIP numbers and each number should be separated by a comma (‘,’) in web UI.

Other allowed characters are 0-9, comma (“,”), asterisk (‘*’), pound sign (‘#’) and plus sign (‘+’).

NTP Server

Defines the URL or IP address of the NTP server. The ATA may obtain the date and time from the server. The default setting is “pool.ntp.org.”

Allow DHCP Option 42 to override NTP server

Defines whether DHCP Option 42 should override NTP server or not. When enabled, DHCP Option 42 will override the NTP server if it is set up on the LAN. The default setting is Yes.

DHCP Option 17 Enterprise Number

Configure the DHCP option 17 number. Default is 3561

CDR File Option

By default, the device will split the allowed memory for CDR file into 2 parts. Device will create the first CDR file which is half of the allowed size, when it is full, device will create the second file.

When “CDR File Option” is set to Default “Keep”, device will keep the call records when both files are full, no more new record will be stored.

When this feature is set to “Overwrite”, device will clear the first CDR file and start storing again.

The CDR file output will be available at Status page: [CDR File]

SIP File Option

By default, the device will split the allowed memory for SIP file into 2 parts. Device will create the first SIP file which is half of the allowed size, when it is full, device will create the second file.

When “SIP File Option” is set to Default “Keep”, device will keep the call records when both files are full, no more new record will be stored.

When this feature is set to “Overwrite”, device will clear the first SIP file and start storing again.

The SIP file output will be available at Status page: [SIP File]

Note: “Send SIP Log” must be enabled to be able to capture the trace.

Disable Weak TLS Cipher Suites

Allows users to disable weak ciphers DES/3DES and RC4, Symmetric Encryption SEED, Symmetric Authentication MD5, Protocol Version SSLv2/SSLv3 or Disable All of the Above Weak TLS Ciphers Suites.

Default is No.

Minimum TLS Version

This feature allows customer to choose desired Minimum TLS Version.

Choices are:

Unlimited

TLS 1.0

TLS 1.1

TLS 1.2

Default is Unlimited.

Maximum TLS Version

This feature allows customer to choose desired Maximum TLS Version.

Choices are:

Unlimited

TLS 1.0

TLS 1.1

TLS 1.2

Default is Unlimited.

Syslog Protocol

If set to SSL/TLS, the syslog messages will be sent through secured TLS

Protocol to syslog server.

Default setting is UDP.

Note: The CA certificate is required to connect with the TLS server

A reboot is required to take effect.

Syslog Server

URL or IP address of syslog server.

Note: A reboot is required to take effect.

Syslog Level

Select the 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)

3. sent or received SIP message (DEBUG level)

4. SIP message summary (INFO level)

5. inbound and outbound calls (INFO level)

6. registration status change (INFO level)

7. negotiated codec (INFO level)

8. Ethernet link up (INFO level)

9. SLIC chip exception (WARNING and ERROR levels)

10. memory exception (ERROR level)

extra syslog style (EXTRA DEBUG level)

Note: A reboot is required to take effect.

Send SIP Log

Configures whether the SIP log will be included in the syslog messages.

The default setting is No.

Always Send HTTP Basic Authentication Information

If set to Yes, the device will send configured username and password within HTTP request without server sending authentication challenge.

Automatic Reboot

Default is No. When “Yes, reboot every day at hour” or “Yes, reboot every week at day” or “Yes, reboot every month at day” is checked, user can specify “Hour of the day (0-23)” or “Day of the week (0-6)” or “Day of the month (0-30)”. Default time is Monday 1AM.

Download Device Configuration

Press Download button to download device configuration file to local computer. The filename is “config.txt”. The file is plain text and not including password fields.

Download Device XML Configuration

Press Download to download device configuration file to local computer. The filename is “config.xml”. The file will not include password fields.

Upload Firmware

Press Upload from local directory button to load the firmware file to the device from your computer. The firmware filename should be “ht80xfw.bin” (ht802fw.bin for HT802 for instance), "ht81xfw.bin” for HT812/HT814 or "ht818fw.bin” for HT818.

Upload Configuration

Press Upload from local directory button to load configuration file to the device from your computer. The configuration file should be an XML file (for instance: “config.xml”). Note: The field

is not mandatory in the document but if available only device with specified MAC address will accept the configuration file.

Export Backup Configuration

Press Download button to export device backup configuration to computer.

The output is “cfg<mac>_enc.xml” (where <mac> is the MAC address of the device). The file is encrypted and can be used on same device only.

Restore From Backup Configuration

Press Upload button to restore device configuration from previously exported backup configuration.

E911/HELD Protocol:

Enable E911

Enable Enhanced 911 call. Default is disabled

HELD Protocol

Configure HELD transfer protocol. HTTP or HTTPS

HELD Synchronization Interval

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

Location Server

Configure the primary Location Information Server (LIS) address

Location Server Username

Configure the user name of the primary Location Information Server (LIS)

Location Server Password

Configure the password of the primary Location Information Server (LIS)

Secondary Location Server

Configure the seconary Location Information Server (LIS) address

Secondary Location Server Username

Configure the user name of the secondary Location Information Server (LIS)

Secondary Location Server Password

Configure the password of the secondary Location Information Server (LIS)

HELD Location Types

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

HELD NAI

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

HELD Identity 1-10

HELD Identity

HELD Identity 1-10 Value

HELD Identity value

E911 Emergency Numbers

A user can configure multiple emergency numbers separated with the delimiter symbol ";".

Geolocation-Routing Header

If "Yes", E.911 INVITE message includes the "Geolocation-Routing" header with the value "Yes"

Priority Header

If "Yes", E.911 INVITE message includes the "Priority" header with the value "emergency"

FXS Ports Pages Definitions

FXS Port (1,2)

Accounts Active

Activates/Deactivates the accounts. The FXS port configuration will not change if disabled, although the port will not be operational, in this state, there will be no dial tone when picking up the analog phone and making/receiving calls will not be possible.

Primary SIP Server

Configures SIP server IP address (Supports both IPv4 and IPv6 addresses) or domain name provided by VoIP service provider. (For example: sip.mycompany.com, IPv4: 192.168.5.170, or IPv6: 2001:1260:1:277::3). This is the primary SIP server used to send/receive SIP messages from/to HT8xx.

Failover SIP Server

Defines failover SIP server IP address (Supports both IPv4 and IPv6 addresses) or domain name provided by VoIP service provider. (For example: sip.mycompany.com, IPv4: 192.168.5.170, or IPv6: 2001:1260:1:277::3)This server will be used if primary SIP server becomes unavailable.

Prefer Primary SIP Server

Selects to prefer primary SIP server. The account will register to primary Server if registration with Failover server expires. Default is No.

Outbound Proxy

Specifies IP address (Supports both IPv4 and IPv6 addresses) or domain name of outbound Proxy, or media gateway, or session border controller. (For example: proxy.myprovider.com, or IP address, if any: IPv4: 192.168.5.170/ IPv6: 2001:1260:1:277::3). Used by HT801/HT802 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work, and only outbound proxy can correct the problem.

Backup Outbound Proxy

Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. (For example: proxy.myprovider.com, or IP address, if any: IPv4: 192.168.5.170/ IPv6: 2001:1260:1:277::3). By default, this field is left empty.

Prefer Primary Outbound Proxy

If the user configures this option to “Yes”, when registration expires, the device will re-register via primary outbound proxy.

By default, this option is disabled.

Allow DHCP Option 120 (override SIP Server)

Configures the HT802 to collect SIP server address from DHCP option 120.

Default is No.

SIP Transport

Selects transport protocol for SIP packets; UDP or TCP or TLS. Please make sure your SIP Server or network environment supports SIP over the selected transport method. Default is UDP.

SIP URI Scheme When Using TLS

Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.

Use Actual Ephemeral Port in Contact with TCP/TLS

Controls the port information in the Via header and Contact header. If set to “No”, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the connection. The default setting is “No”.

NAT Traversal

Indicates type of NAT for each account. This parameter configures whether the NAT traversal mechanism is activated. Users could select the mechanism from No, Keep-alive, STUN, UPnP.

Default setting is No.

SIP User ID

Defines user account information provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.

Authenticate ID

Determines account authenticate ID provided by VoIP service provider (ITSP). Can be identical to or different from “SIP user ID”.

Authenticate Password

Specifies account password provided by VoIP service provider (ITSP) to register to SIP servers.

Name

Chooses a name to be associated to user.

DNS Mode

Selects DNS mode to use for the client to look up server. One mode can be chosen. Default is A Record.

A Record: resolves IP Address of target according to the domain name.

SRV: DNS SRV resource records indicate how to find services for various protocols.

NAPTR/SRV: Naming Authority Pointer according to RFC 2915.

Use Configured IP: if SIP server is configured as domain name, device will not send DNS query, but will use “Primary IP” or “Backup IP” to send SIP message if at least one of them are not empty. It will try to use “Primary IP” first, after 3 tries without any response, it will switch to “Backup IP 1”, then “Backup IP 2”, and then it will switch back to “Primary IP” after 3 re-tries.

DNS SRV use Registered IP

When this option is set to “Yes”, when the HT is registered on second SRV and makes an outbound call, it will try the second SRV (registered IP) first.

By default, this option is disabled, and the DNS SRV will use first SRV instead of the registered IP.

DNS SRV Failover Mode

Configure the preferred IP mode for DNS SRV or NAPTR/SRV. If “default” is selected, first IP from query result will be applied; If “Saved one until DNS TTL” is selected, previous IP will be applied before reaches DNS timeout; If “Saved one until no response“ is selected, previous IP will be applied even after DNS time out until it is unreachable.

• Default: If the option is set with “default”, it will again try to send register messages to one IP at a time, and the process repeats.

• Saved one until DNS TTL: If the option is set to “Saved one until DNS TTL”, it will send register messages to the previously registered IP first. If no response, it will try to send one at a time for each IP. This behavior lasts if DNS TTL (time-to-live) is up.

• Saved one until no responses: If the option is set with “Saved one until no responses”, it will send register messages to the previously registered IP first, but this behavior will persist until the registered server does not respond.

Failback Timer

When the primary SBC is up, device will send SIP requests to the primary SBC. If at any point device fails over to the secondary SBC, the SIP requests will stay on the failover SBC for the duration of the failback timer. When the timer expires, device will send SIP requests to the primary SBC, (in minutes. Default is 60 minutes, max 45 days).

Register Before DNS SRV Failover

This feature is used to control whether the device need to initiate a new registration request (following existing DNS SRV fail-over mode) first and then direct the nonregistration SIP request (INVITE) to the new successfully registered server or not.

TEL URI

Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the HT801/HT802 has an assigned PSTN Number.

Disabled: Use “SIP User ID” information in the Request-Line and “From” header.

User=Phone: “User=Phone” parameter will be attached to the Request-Line and “From” header in the SIP request to indicate the E.164 number. If set to “Enable”.

Enabled: “Tel:” will be used instead of “sip:” in the SIP request.

Please consult your carrier before changing this parameter. Default is Disabled.

SIP Registration

Controls whether the HT801/HT802 needs to send REGISTER messages to the proxy server. Default setting is Yes.

Unregister on Reboot

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with “Expires=0” parameter to the SIP server. This will unregister the SIP account under the concerned FXS page. Default is No.

Outgoing Call Without Registration

Enables the ability to place outgoing calls even if the account is not registered (if allowed by ITSP); device will not be able to receive incoming calls. Default is No.

Register Expiration

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

Reregister Before Expiration

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

SIP Registration Failure Retry Wait Time

Sends re-register request after specific time (in seconds) when registration process fails. Maximum interval is 3600 seconds (1 hour). Default is 20 seconds.

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Sends re-register request after specific time (in seconds) when registration process fails with error 403 Forbidden. Maximum interval is 3600 seconds (1 hour). Default is 1200 seconds.

MWI Tone

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

Enable SIP OPTIONS/NOTIFY Keep Alive

Enables SIP OPTIONS or SIP NOTIFY to track account registration status so the phone adapter will send periodic OPTIONS/NOTIFY messages to server to track the connection status with the server. Default setting is No.

SIP OPTIONS/NOTIFY Keep Alive Interval

Configures the time interval when the phone adapter sends OPTIONS or NOTIFY messages to the SIP server. The default setting is 30 seconds, which means the phone adapter will send an OPTIONS/NOTIFY message to the server every 30 seconds. The default range is 1-64800.

SIP OPTIONS/NOTIFY Keep Alive Max Lost

Defines the Number of max lost packets for SIP OPTIONS or SIP NOTIFY Keep Alive before re-registration. Between 3-10, default is 3.

Layer 3 QoS

Defines Diff-Serv values for SIP and RTP.

SIP DSCP (Diff-Serv value in decimal, 0-63, default 26)

RTP DSCP (Diff-Serv value in decimal, 0-63, default 46)

Local SIP Port

Defines local port to use by the HT801/HT802 for listening and transmitting SIP packets. Default value for FXS 1 is 5060 and 5062 for FXS 2.

Local RTP Port

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

Use Random SIP Port

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

Use Random RTP Port

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

Enable RTCP

Allows users to enable RTCP. Default setting is Yes.

Hold Target Before Refer

Allows user to hold the phone call before referring it. If set to No, the call will not be hold before referred. Default is Yes.

Refer-To Use Target Contact

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

Transfer on Conference Hang-up

If set to “Yes”, when the phone hangs up as the conference initiator, the conference call will be transferred to the other parties so that other parties will remain in the conference call. Default setting is No.

Remove OBP from Route Header

Removes outbound proxy info in “Route” header when sending SIP packets.

Default is No.

Support SIP Instance ID

Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft.

Default is No.

Validate Incoming SIP Message

Validates incoming messages.

Default is No.

Check SIP User ID for Incoming INVITE

Checks SIP User ID in the Request URI of incoming INVITE; if it does not match the HT801/HT802 SIP User ID, the call will be rejected. Direct IP calling will also be disabled. Default is No.

Authenticate Incoming INVITE

Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.

Authenticate server certificate domain

Configures whether to validate the domain certificate when download the firmware/config file. If it is set to “Yes”, the phone will download the firmware/config file only from the legitimate server.

Default setting is “No“.

Authenticate server certificate chain

Configures whether to validate the server certificate when download the firmware/config file. If it is set to “Yes”, the phone will download the firmware/config file only from the legitimate server.

Default setting is “No“.

Allow Incoming SIP Messages from SIP Proxy Only

Checks SIP address of the Request URI in the incoming SIP message; if it does not match the SIP server address of the account, the call will be rejected.

Default is No.

Use Privacy Header

Determines if the “Privacy header” will be presented in the SIP INVITE message and if it includes the caller info in this header. If set to Default, it will add Privacy header unless special feature is Telkom SA or CBCOM. Default is Default.

Use P-Preferred-Identity Header

Specifies if the P-Preferred-Identity Header will be presented in the SIP INVITE message. If set to “default”, the P-Preferred-Identity Header will be omitted in SIP INVITE message when Telkom SA or CBCO is active. If set to “Yes”, the P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted.

Default setting is Default.

Use P-Access-Network-Info Header

With this feature enabled, device will populate the WAN access node with IEE-802.11a, IEE-802.11b in P-Access-Network-Info SIP header.

Use P-Emergency-Info Header

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

Use P-Asserted-Identity Header

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

SIP REGISTER Contact Header Uses

Specifies which address (LAN or WAN address) the device will detect to use it in SIP Register Contact Header. When set to LAN, Contact header will include local IP from ATA in REGISTER messages, while if set to WAN, host/port/contact will be updated from SIP 401/403/404/407 Via header “received”/”rport” parameters in REGISTER messages. Default is LAN Address.

Caller ID Fetch Order

Selects the Caller ID display order which need to be respected by the HT801/HT802. The available options are:

Auto: When set to “Auto”, the HT801/HT802 will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE.

Disabled: When set to “Disabled”, all incoming calls are displayed with “Unavailable”.

From Header: When set to “From Header”, the HT801/HT802 will use the FROM header to display the caller ID.

Allow SIP Factory Reset

Allows to reset the devices directly through SIP Notify. If “Allow SIP Factory Reset” is set to “YES” under FXS PORT, then the ATA receives the NOTIFY from the SIP server with Event: reset, the HT should perform a factory reset after the authentication.

The authentication in this case can be either with:

The admin password if no SIP account is configured on the HT.

With the credentials of the SIP account if configured on the ATA.

Maximum Number of SIP Request Retries

This feature allows user to configure the number of SIP retries before failover occurs. ( between 1 and 10, default is 2)

SIP T1 Timeout

Defines T1 timeout value.

It is an estimate of the round-trip time between the client and server transactions.

For example, the HT801/HT802 will attempt to send a request to a SIP server.

The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time defined by T2. Default is 0.5 seconds.

SIP T2 Interval

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling of T1 continues until it reaches T2 value. Default is 4 seconds.

SIP Timer D

Configure the SIP Timer D defined in RFC3261. 0 – 64 seconds.

Default is 0.

DTMF Payload Type

Defines payload type for DTMF using RFC2833. Default is 101.

Preferred DTMF method (in order)

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

Inband DTMF Duration

Allows to adjust the inband DTMF duration sent from ATA to IPPBX.

Default is 100 ms. Valid range: 40-2000 ms.

Inter-duration: 50 ms. Valid range: 40-2000 ms.

Disable DTMF Negotiation

Uses above DTMF order without negotiation.

Default is No.

Generate Continuous RFC2833 Events

When enabled, the RFC2833 events are generated until key is released. Default is No.

Send Hook Flash Event

If set to yes, flash will be sent as DTMF event.

Default is No.

Flash Digit Control

When it set to YES it allows the user to perform some call setting when both channels are used while pressing:

“Flash + 1” in order to hang up the current call and resume a call that was held.

“Flash + 2” in order to hold the current call and resume a call that was held.

“Flash + 3” in order to perform 3-way conference.

“Flash + 4” in order to perform attended transfer.

Note: Please refer to the user guide for detailed steps to perform above operations.

Enable Call Waiting alert-info In 180 Ringing Response

When set to Yes, Alert-Info header will be added in 180 Ringing for Call Waiting case

Callee Flash to 3WC

When this feature is set to Yes, device would be able to set up the 3-way conference call even when device is the callee in the second call. Default is No.

Off Hook Auto Dial

Configures a user ID or extension number that is automatically dialed when off-hook. Only the user part of a SIP address needs to be entered. The HT80x will automatically append the “@” and the host portion of the corresponding SIP address.

Off Hook Auto Dial Delay

Specifies the auto-dial delay in seconds after off hook.

Valid range is 0-60 seconds. Default is 0.

Off Hook Auto-Dial DTMF

When ‘Off Hook Auto-Dial’ is configured, this feature allows user to configure DTMF digits to send after ‘OffHook auto-dial’ call gets connected.

Proxy-Require

Determines a SIP Extension to notify the SIP server that the HT801/HT802 is behind a NAT/Firewall.

Use NAT IP

Defines NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP.

SIP User-Agent

This feature allows users to configure SIP User Agent. If not configured, device will use the default User Agent header.

SIP User-Agent Postfix

Configures the SIP User-Agent Postfix.

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 "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

RFC2543 Hold

Toggles between RFC2543 hold and RFC3261 hold. RFC2543 hold allows to disable the hold music sent to the other side, in this case IP address (0.0.0.0) it will be sent in SDP instead of the IP address of the unit RFC 3261 (a line) will play the hold music to the other side.

Disable Call Waiting

Disables receiving a second incoming call when the line is engaged.

Default is No.

Disable Call Waiting Caller ID

Disables displaying caller ID when receiving a second incoming call.

Default is No.

Disable Call Waiting Tone

Disables playing call waiting tone during active call when receiving a second incoming call. The CWCID will still be displayed. Default is No.

Disable Connected Line ID

Disables displaying the number of the person answering the phone. Default is No.

Disable Receiver Off Hook Tone

Enables / disables the warning to alert that the phone has been left off-hook for an extended period of time. Default is No.

Disable Reminder Ring for On-Hold Call

Enables playing the reminder ring. Default is No

Disable Reminder Ring for DND

This feature allows user to disable reminder ring when FXS port is on DND mode. Default is No

Disable Visual MWI

Disables use of visual message waiting indicator when there is an unread voicemail message. Default is No.

Visual MWI Type

Configures Visual WMI Type of signal sent to the analog phone to make it turn the lamp ON upon receiving a Voice mail.

Check the phone’s manual to find out what signal is supported, FSK (default) or NEON.

Note: Some phones (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.

Do Not Escape ‘#’ as %23 in SIP URI

Replaces # by %23 in some special situations. Default is No.

Disable Multiple m Line in SDP

Sends only one m line in SDP, regardless of how many m fields are in the incoming SDP. Default is No.

Ring Timeout

Stops ringing when incoming call if not answered within a specific period of time. Valid range is 0-300 seconds, 0 means no timeout.

Default is 60 seconds.

Delayed Call Forward Wait Time

Forwards incoming call if not answered within a specific period of time when delayed call forward is activated locally (using *92 code).

Default value is 20 seconds.

No Key Entry Timeout

Initiates the call within this time interval if no additional key entry during dialing stage. Valid range is 1-15 seconds. Default is 4 seconds.

Early Dial

Sends an early INVITE each time a key is pressed when a user dials a number. Otherwise, only one INVITE is sent after full number is dialed (user presses Dial Key or after “no key entry timeout” expires).

This option should be used only if there is a SIP proxy is configured and supporting 484 responses (Incomplete Address).

Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). Default is No.

This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

Adds specified prefix to dialed number.

Use # as Dial Key

Treats “#” as the “Send” (or “Dial”) key. If set to “No”, this “#” key can be included as part of the dialed number. Default is Yes.

Disable # as Redial Key

With this feature and feature ‘Use # as Dial Key’ set to Yes, the # key will act as dial key but not as redial key. Default No.

Dial Plan

Dial Plan Rules:

1. Accept Digits: 1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d

2. Grammar: x – any digit from 0-9;

a. xx+ – at least 2 digits number;

b. xx – exactly 2 digits number;

c. ^ – exclude;

d. . – wildcard, matches one or more characters

e. [3-5] – any digit of 3, 4, or 5;

f. [147] – any digit 1, 4, or 7;

g. <2=011> – replace digit 2 with 011 when dialing

h. <=1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed

i. | – or

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 feature. As an example { *x+ } will allow to dial * followed by any length of numbers.

SUBSCRIBE for MWI

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

Subscribe Retry Wait Time upon 403 Forbidden

This feature allows users to adjust the subscribe retry waiting time while it is rejected with 403 forbidden. Default is 0. Valid range: 0-10080 in minutes.

Send Anonymous

Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, blocking caller ID. Default is No.

Anonymous Call Rejection

Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Default is No.

Special Feature

Selects Soft switch vendors’ special requirements Example of vendors: BroadSoft, CBCOM, RNK, Huawei, China Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS, TELEFONICA SPAIN, IZZ, Oi_BR, ROSTELECOM. Default is Standard.

Enable Session Timer

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

Session Expiration

Enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE). When the session interval expires, if there is no refresh via an UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. Valid range is 90-64800 seconds. Default is 180 seconds.

Min-SE

Defines Minimum session expiration (in seconds).

Valid range is 90-64800 seconds. Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it.

Default is No.

Callee Request Timer

Uses session timer when receiving inbound calls with session timer request.

Default is No.

Force Timer

Uses session timer even if the remote party does not support this feature. Selecting “No” will enable session timer only when the remote party supports it. Default is No.

To turn off Session Timer, select “No” for Caller and Callee Request Timer, and Force Timer.

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls.

UAC: The handy tone acts as the refresher.

UAS: Callee or proxy server act as the refresher.

Default is Omit.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls:

UAS: The handy tone acts as the refresher.

UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE

Uses INVITE message to refresh the session timer.

Default is No.

Enable 100rel

Appends “100rel” attribute to the value of the required header of the initial signaling messages. Default is No.

Add Auth Header on Initial REGISTER

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request.

Default is No.

Conference URI

Allows users to manually configure the conference URL. The default is null.

Use First Matching Vocoder in 200OK SDP

Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same order received in INVITE.

Default is No.

Preferred Vocoder

Configures vocoders in a preference list (up to 8 preferred vocoders) that will be included with same order in SDP message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, G722, iLBC and OPUS.

Voice Frames per TX

Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for G711/G726/G723/other codecs respectively.

G723 Rate

Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is 6.3kbps.

iLBC Frame Size

Specifies iLBC packet frame size (20ms or 30ms).

Default is 20ms.

Disable OPUS Stereo in SDP

Disables OPUS stereo in SDP. If set to Yes, HT8XX will remove /2 from offer.

Default is No.

iLBC Payload type

Determines payload type for iLBC. Valid range is between 96 and 127.

Default is 97.

OPUS Payload Type

Determines payload type for OPUS. Valid range is between 96 and 127. Default is 123.

VAD

Allows detecting the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. Default is No.

Symmetric RTP

Changes the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device. Default is No.

Fax Mode

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

Re-IINVITE after Fax Tone Detection Mode

Permits the unit to send out the re-INVITE for T.38 or Fax Pass Through if a fax tone is detected.

Default is Enabled

Jitter Buffer Type

Selects jitter buffer type (Fixed or Adaptive) based on network conditions.

Default is Adaptive.

Jitter Buffer Length

High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.

Medium (initial 100ms, min 20ms, max 200ms).

Low (initial 50ms, min 10ms, max 100ms).

SRTP Mode

Selects SRTP mode to use (“Disabled”, “Enabled but not forced”, or “Enabled and forced”). Default is Disabled.

It uses SDP Security Description to exchange key. Please refer to

SDES: https://tools.ietf.org/html/rfc4568

SRTP: https://www.ietf.org/rfc/rfc3711.txt

Crypto Life Time

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

SLIC Setting

Depends on standard phone type (and location).

Caller ID Scheme

Selects the caller ID scheme. For example: Bellcore/Telcordia, ETSI-FSK …

DTMF Caller ID

Defines the start and stop tones (Default, A, B, C, D or #) to delimit CID.

Disable Unknown Caller ID

This feature allows users to disable the analog phone’s caller ID when the device receive a call with Anonymous, unavailable or unknown as FROM USER and without Display-INFO. Default No

Replace Beginning ‘+’ with 00 in Caller ID

When this feature is set to Yes, device will replace the “+” sign at the beginning of a number in the FROM header with “00”. Default is No.

Polarity Reversal

Reverses the polarity upon call establishment and termination.

Default is No.

Loop Current Disconnect

Allows the traditional PBX used with HT801/HT802 to apply this method for signaling call termination. Method initiates short voltage drop on the line when remote (VoIP) side disconnects an active call. Default is No.

Play busy/reorder tone before Loop Current Disconnect

Allows user to configure if it will play busy/reorder tone before loop current disconnect upon call fail. Default is No.

Loop Current Disconnect Duration

Configures the duration of voltage drop described in topic above. HT80X supports a duration range from 100 to 10000 ms. Default value is 200.

Enable Pulse Dialing

Allow users to enable Pulse Dialing option under FXS Port. Default is No.

Pulse Dialing Standard

Allows users to use Swedish pulse dialing standard and New Zealand Standard.

Default is General Standard.

Enable Hook Flash

Enables the FLASH button to be used for terminating calls. Default is Yes.

Hook Flash Timing

Defines the time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time value. HT801/HT802 supports a range from 40 to 2000 ms.

Default values are 300 minimum and 1100 maximum.

On Hook Timing

Specifies the on-hook time for an on-hook event to be validated. HT801/HT802 supports a range from 40 to 2000 ms. Default value is 400.

Gain

Adjusts the voice path volume.

• Rx is a gain level for signals transmitted by FXS

• Tx is a gain level for signals received by FXS.

Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.

User can adjust volume of call using the Rx gain level parameter and the Tx gain level parameter located on the FXS port configuration page. If call volume is too low when using the FXS port (i.e., the ATA is at user site), adjust volume using the Rx gain level parameter under the FXS port configuration page. If voice volume is too low at the other end, user may increase the far end volume using the Tx gain level parameter under the FXS port configuration page.

Disable Line Echo Canceller (LEC)

Disables the LEC will per call base. Recommended for FAX/Data calls.

Default is No.

Disable Network Echo Suppressor

Disables the NEC will per call base. Recommended for FAX/Data calls.

Default is No.

Outgoing Call Duration Limit

Defines the call duration limit for the outgoing calls.

Valid range is 0-180 minutes. Default is 0 (No limit).

Ring Frequency

Customizes ring frequency. Valid options: 20Hz – 25Hz. Default is 20Hz.

Enable High Ring Power

Configures a high ringing voltage output for the HT801/802.

OnHook DC Feed Current

Adjust DC feed current.

RFC2833 Events Count

This feature allows users to customize the count of RFC2833 events.

Valid range is 2 – 10. Default is 8.

RFC2833 End Events Count

This feature allows users to customize the count of RFC2833 end events. Valid range is 2 – 10. Default is 3.

Distinctive Ring Tone

Customizes the Ring Tone 1 to 10 with associate caller ID: when selected, if caller ID is configured, then the device will ONLY use this ring tone when the incoming call is from the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller ID is configured, the selected ring tone will be used for all incoming calls using the FXS port. Distinctive ring tones can be configured not only for matching a whole number, but also for matching prefixes. In this case symbol “x+” will be used.

For example: If configured as 617x+, Ring Tone 1 will be used in case of call arrived from the area code 617. Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page.

Note: If server supports Alert-Info header and standard ring tone set (Bellcore) or distinctive ring tone 1-10 is specified, then the ring tone in the Alert-Info header from server will be used. Bellcore rings and tones are independent from custom ring tones. The custom ring tones can also be specified by alert-info header, for example

Alert-Info: ;info=ring5

Ring tones

Configures the ring tone cadence preferences. User has 10 choices.

(Syntax: c=on1/off1-on2/off2-on3/off3;)

Note: The configuration, completed in Distinctive Ring Tones block in the same page, applies to ring tones cadences configured here.

Default is c=2000/4000;

Distinctive Call Waiting Tone

Customizes the Call Waiting Tone 1 to 10 with associate caller ID: when selected, if caller ID is configured, then the device will ONLY use this call waiting tone when the incoming call waiting is from the Caller ID. When selected but no Caller ID is configured, the selected call waiting tone will be used for all incoming waiting calls using the FXS port. Distinctive Call Waiting Tones can be configured not only for matching a whole number, but also for matching prefixes. In this case symbol “x+” will be used.

For example: If configured as 617x+, Call Waiting Tone 1 will be used in case of waiting call arrived from the area code 617. Any other incoming call waiting will be using cadence defined in parameter Call Waiting Tone located under Advanced Settings Configuration page.

Call Waiting Tones

This feature allows user to customize call waiting tone. User has 10 choices.

Syntax: f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];

(Frequencies are in (300, 3400) Hz and cadence on and off are in (0, 64000) ms)

Note: The configuration, completed in Distinctive Call Waiting Tones block in the same page, applies to call waiting cadences configured here.

Default is f1=440@-13,c=300/10000;

Call Features Settings

Enable Call Features

When enabled, Do Not Disturb, Call Forward and other call features can be used via the local feature codes on the phone. Otherwise, the ITSP feature codes will be used. Enable All will override all individual features enable setting. Default is Yes.

Reset Call Features

Allows users to reset all call features configuration. Default is No

SRTP Feature

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

– Enable SRTP: Default is 16

– Disable SRTP: Default is 17

SRTP per call Feature

– Enable SRTP per call: Default is 18

– Disable SRTP per call: Default is 19

CID Feature

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

– Enable CID: Default is 31

– Disable CID: Default is 30

CID per call Feature

– Enable CID per call: Default is 82

– Disable CID per call: Default is 67

Direct IP Calling Feature

Allow users to customize the Direct IP feature code. Default is Yes

– Direct IP Calling: Default is 47

CW Feature

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

– Enable CW: Default is 51

– Disable CW: Default is 50

CW per call Feature

– Enable CW per call: Default is 71

– Disable CW per call: Default is 70

Call Return Feature

Allow users to customize the Call Return feature code. Default is Yes

– Call return: Default is 69

Unconditional Forward Feature

Allow users to customize the Unconditional Forward feature codes.

Default is Yes

– Enable Unconditional Forward: Default is 72

– Disable Unconditional Forward: Default is 73

Busy Forward Feature

Allow users to customize the Busy Forward feature codes. Default is Yes

– Enable Busy Forward: Default is 90

– Disable Busy Forward: Default is 91

Delayed Forward Feature

Allow users to customize the Delayed Forward feature codes. Default is Yes

– Enable Delayed Forward: Default is 92

– Disable Delayed Forward: Default is 93

Paging Feature

Allow users to customize the Paging feature code. Default is Yes

– Paging: Default is 74

DND Feature

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

– Enable DND: Default is 78

– Disable DND: Default is 79

Blind Transfer Feature

Allow users to customize the Blind Transfer feature code. Default is Yes

– Enable Blind Transfer: Default is 87

Disable LEC per call Feature

Default is Yes

– Disable LEC per call: Default is 03

Disable Bellcore Style 3-Way Conference

Gives the users the possibility of making conference calls by pressing “Flash” key, when it is enabled by dialing *23 +second callee number. Default is No

Star Code 3WC Feature

Default is Yes

– Star Code 3WC: Default is 23

Forced Codec Feature

Allow users to customize the Forced Codec feature code. Default is Yes

– Forced Codec: Default is 02

PCMU Codec Feature

Default is Yes

– PCMU Codec: Default is 7110

PCMA Codec Feature

Default is Yes

– PCMA Codec: Default is 7111

G723 Codec Feature

Default is Yes

– G723 Codec: Default is 723

G729 Codec Feature

Default is Yes

– G729 Codec: Default is 729

iLBC Codec Feature

Default is Yes

– iLBC Codec: Default is 7201

G722 Codec Feature

Default is Yes

– G722 Codec: Default is 722

NAT Settings

If you plan to keep the handy tone within a private network behind a firewall, we recommend using STUN Server. The following three settings are useful in the STUN Server scenario:

  1. STUN Server (under advanced settings webpage) enter a STUN server IP (or FQDN) that you may have or look up a free public STUN server on the internet and enter it on this field. If using public IP, keep this field blank.
  2. Use random SIP/RTP ports (under 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 FXS web page): Set to “Yes” when the gateway is behind a firewall on a private network.

DTMF Methods

The HT801/HT802 support the following DTMF mode:

  • DTMF in-audio
  • DTMF via RTP (RFC2833)
  • DTMF via SIP INFO

Set priority of DTMF methods according to your preference. This setting should be based on your server’s DTMF setting.

Preferred Vocoder (Codec)

The HT801/HT802 support the following voice codecs. On FXS ports pages, choose the order of your favorite codecs:

  • PCMU/A (or G711µ/a)
  • G729 A/B
  • G723.1
  • G726
  • iLBC
  • OPUS
  • G722

Configuring HT801/HT802 Through Voice Prompts

As mentioned previously, The HT801/HT802 has a built-in voice prompt menu for simple device configuration.

Please refer to “Understanding HT801/HT802 Interactive Voice Prompt Response Menu” for more information about IVR and how to access its menu.

  • DHCP MODE

Select voice menu option 01 to enable HT801/HT802to use DHCP.

  • STATIC IP MODE

Select voice menu option 01 to enable HT801/HT802 to use STATIC IP mode, then use options 02, 03, 04, and 05 to set up IP address, Subnet Mask, Gateway, and DNS server respectively.

  • FIRMWARE SERVER IP ADDRESS

Select voice menu option 13 to configure the IP address of the firmware server.

  • CONFIGURATION SERVER IP ADDRESS

Select voice menu option 14 to configure the IP address of the configuration server.

  • UPGRADE PROTOCOL

Select the menu option 15 to choose firmware and configuration upgrade protocol between TFTP, HTTP, HTTPS, FTP, and FTPS. The default is HTTPS.

  • FIRMWARE UPGRADE MODE

Select voice menu option 17 to choose firmware upgrade mode among the following three options: 1) Always check, 2) check when pre/suffix changes, and 3) never upgrade.

Register a SIP Account

The HT801/HT802 supports 2 FXS ports which can be configured with 2 SIP accounts. Please refer to the following steps in order to register your accounts via the web user interface:

  1. Access your HT801/HT802 web UI by entering its IP address in your favorite browser.
  2. Enter your admin password (default: admin).
  3. Press Login to access your settings.
  4. Go to FXS Port (1 or 2) pages.
  5. In the FXS Port tab, 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. Prefer Primary SIP Server to No or Yes depending on your configuration. Set to No if no Failover SIP Server is defined. If “Yes”, the account will register to the Primary SIP Server when failover registration expires.
    5. Outbound Proxy: Set your Outbound Proxy IP Address or FQDN. Leave empty if not available.
    6. SIP User ID: User account information, provided by VoIP service provider (ITSP). Usually in the form of a digit similar to a phone number or actually a phone number.
    7. Authenticate ID: SIP service subscriber’s Authenticate ID is used for authentication. Can be identical to or different from SIP User ID.
    8. Authenticate Password: SIP service subscriber’s account password to register to the SIP server of ITSP. For security reasons, the password will field will be shown as empty.
    9. Name: Any name to identify this specific user.

For more information, related to the above options please refer to FXS Port Settings.

  1. Press Apply at the bottom of the page to save your configuration.
Figure 11: SIP Accounts Settings

After applying your configuration, your account will register to your SIP Server, you can verify if it has been correctly registered with your SIP server or from your HT801/HT802 web interface under 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).

Figure 12: Account Status
Note:

When all the FXS ports are registered (for HT802), the simultaneous ring will have one second delay between each ring on each phone.

Call Features

The HT801/HT802 support all the traditional and advanced telephony features.

KeyCall Features

*02

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

*03

Disable LEC (per call) Dial “*03” +” number”.

No dial tone is played in the middle.

*16

Enable SRTP.

*17

Disable SRTP.

*30

Block Caller ID (for all subsequent calls).

*31

Send Caller ID (for all subsequent calls).

*47

Direct IP Calling. Dial “*47” + “IP address”.

No dial tone is played in the middle.

*50

Disable Call Waiting (for all subsequent calls).

*51

Enable Call Waiting (for all subsequent calls).

*67

Block Caller ID (per call). Dial “*67” +” number”.

No dial tone is played in the middle.

*82

Send Caller ID (per call). Dial “*82” +” number”.

No dial tone is played in the middle.

*69

Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.

*70

Disable Call Waiting (per call). Dial “*70” +” number”.

No dial tone is played in the middle.

*71

Enable Call Waiting (per call). Dial “*71” +” number”.

No dial tone is played in the middle.

*72

Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for the dial tone and hang up. (Dial tone indicates successful forward)

*73

Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait for the dial tone, then hang up.

*74

Enable Paging Call: Dial “*74” and then the destination phone number you want to page.

*78

Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.

*79

Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.

*87

Blind Transfer.

*90

Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for the dial tone then hang up.

*91

Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for the dial tone, then hang up.

*92

Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for the dial tone then hang up.

*93

Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for the dial tone, then hang up.

Flash/
Hook

Toggles between an 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 the

pound sign will serve as a

Re-Dial key.

Table 11: HT801/HT802 Call Features

Rebooting HT801/HT802 from Remote

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

UPGRADING AND PROVISIONING

The HT801/HT802 can be upgraded via TFTP/HTTP/HTTPS/FTP/FTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS/FTP/FTPS server and selecting a download method. Configure a valid URL for TFTP or FTP/FTPS or HTTP/HTTPS (default is HTTPS); the server’s name can be FQDN or IP address.

Examples of valid URLs:

firmware.grandstream.com

fw.ipvideotalk.com/gs

Firmware Upgrade procedure

Please follow the below steps to upgrade the firmware version of your HT801/HT802:

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

Upgrading via Local Directory

  1. Download the firmware file from the Grandstream website;
  2. Unzip it and copy the file into a folder on your PC;
  3. From the HT801/HT802 web interface (Advanced Settings page) you can browse your hard drive and select the folder you previously saved the file (HT802fw.bin);
  4. Click “Upload Firmware” and wait a few minutes until the new program is loaded.

Always check the status page to see that the program version has changed.

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

For users that would like to use remote upgrading without a local TFTP/HTTP/HTTPS/FTP/FTPS server, Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their devices via this server. Please refer to the webpage:

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:

http://www.solarwinds.com/products/freetools/free_tftp_server.aspx

http://tftpd32.jounin.net/.

Instructions for local firmware upgrade via TFTP:

  1. Unzip the firmware files and put all of them in the root directory of the TFTP server.
  2. Connect the PC running the TFTP server and the phone to the same LAN segment.
  3. Launch the TFTP server and go to the File menu🡪Configure🡪Security to change the TFTP server’s default setting from “Receive Only” to “Transmit Only” for the firmware upgrade.
  4. Start the TFTP server and configure the TFTP server in the phone’s web configuration interface.
  5. Configure the Firmware Server Path to the IP address of the PC.
  6. Save and Apply the changes and reboot the HT801/HT802.

End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use the 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 of the firmware with different versions 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.

In addition, when the field “Check New Firmware only when F/W pre/suffix changes” is set to “Yes”, the device will only issue a firmware upgrade request if there are changes in the firmware Prefix or Postfix.

Managing Firmware and Configuration File Download

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

Configuration File Download

Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File (binary or XML) through TFTP, FTP/FTPS, or HTTP/HTTPS. The Config Server Path is the TFTP, FTP/FTPS, or HTTP/HTTPS server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The Config Server Path can be the same or different from the Firmware Server Path.

A configuration parameter is associated with each particular field on the web configuration page.

A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with the “New Password” in the Web GUI🡪Maintenance🡪Web/SSH Access page🡪Admin Password. For a detailed parameter list, please refer to the corresponding firmware release configuration template.

When the HT801/HT802 boots up or reboots, it will send a request to download a file named “cfgxxxxxxxxxxxx” followed by a configuration XML file named “cfgxxxxxxxxxxxx.xml”, where “xxxxxxxxxxxx” is the MAC address of the phone, i.e., “cfg000b820102ab” and “cfg000b820102ab.xml”. If the download of “cfgxxxxxxxxxxxx.xml” file is not successful, the provision program will download a generic cfg.xml file and then download cfg<Model>.xml. The configuration file name should be in lower case letters.

HT801/HT802 supports DHCP option 67 allowing to provide a custom name for the provisioning file. If DHCP option 67 is used, the following file download sequence will be applied:

Step 1: cfg<MAC>

Step 2: <option 67 bootfile> 🡪 cfg<MAC>.xml 🡪 cfg.xml 🡪 cfg<Model>.xml

Notes:

  1. Only XML or binary config file formats are accepted.
  2. The MAC header in the XML config file should be the device MAC or needs to be removed completely.
Figure 14: XML Config File – MAC Header

RESTORE FACTORY DEFAULT SETTINGS

Warning:

Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore them to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.

There are three (3) methods for resetting your unit:

Using the Reset Button

To reset default factory settings using the reset button please follow the steps above:

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

Using the IVR Command

Reset default factory settings using the IVR prompt:

  1. Dial “***” for the voice prompt.
  2. Enter “99” and wait for the “reset” voice prompt.
  3. Enter the encoded MAC address (Look below for how to encode the MAC address).
  4. Wait 15 seconds and the 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:

KeyMapping

0-9

0-9

A

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

B

222

C

2222

D

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

E

333

F

3333

Table 12: 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 HT801/HT802 UI by entering its IP address in your favorite browser.
  2. Enter your admin password (default: admin).
  3. Press Login to access your settings.
  4. Go to Basic Settings🡪Reset Type.
  5. Press the Reset button (after selecting the reset type).

Full Reset: This will make a full reset.

ISP Data: This will reset only the basic settings, like IP mode, PPPoE, and Web port.

VOIP Data: This will reset only the data related to a service provider like SIP server, sip user ID, provisioning and others.

  • Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.
  • If the HT801/HT802 was previously locked by your local service provider, pressing the RESET button will only restart the unit. The device will not return to factory default settings.

Reset using SIP NOTIFY

  1. Access your HT801/HT802 UI by entering its IP address in your favorite browser.
  2. Go to FXS Port.
  3. Set “Allow SIP Factory Reset” to “Yes”.
  4. Once a SIP NOTIFY with “event: reset” is received, the phone will perform a 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 administration guide for HT801/HT802. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.

Firmware version 1.0.41.2

Firmware version 1.0.39.4

Firmware version 1.0.37.1

Firmware version 1.0.35.4

  • Added support to allow HT8xx provision admin password without special characters.
  • Added support for Israel time zone with DST.

Firmware version 1.0.33.4

  • Added feature Special Proceed Indication Tone. [Special Proceed Indication Tone].
  • Added feature MWI Tone. [MWI Tone].

Firmware version 1.0.31.1

Firmware version 1.0.29.8

Firmware version 1.0.27.2

  • No Major Changes

Firmware version 1.0.25.5

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

Firmware version 1.0.23.5

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

Firmware version 1.0.21.4

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

Firmware version 1.0.19.11

  • Added “Disable” option for Web access mode. [Web Access Mode]

• Moved “Trusted CA certificates” from Profile1/Profie2 to Advanced Settings and renamed as Trusted

CA Certificates A and Trusted CA Certificates B. [Trusted CA certificates A] [Trusted CA certificates B]

  • Added Feature “Disable User Level Web Access” and “Disable Viewer Level Web Access”. [Disable User Level Web Access][Disable Viewer Level Web Access]
  • Added Feature “Use P-Asserted-Identity Header”. [ Use P-Asserted-Identity Header]
  • Added Feature” Load CA Certificates”. [Load CA Certificates]
  • Added Feature “Connection Request Port”. [Connection Request Port]
  • Added OI_BR to special feature. [Special Feature]
  • Added support to set Ring Timeout to 0 for unlimited ring timeout. [Ring Timeout]
  • Added New Zealand Standard for Pulse Dialing Standard. [Pulse Dialing Standard]
  • Added support for Russian Language. [Language]
  • Increased “SIP TLS Certificate” and “SIP TLS Private Key” supported maximum length from 2048 to 4096.[SIP TLS Certificate][SIP TLS Private Key]

Firmware Version 1.0.17.5

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

Firmware Version 1.0.15.4

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

Firmware Version 1.0.13.7

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

Firmware Version 1.0.11.6

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

Firmware Version 1.0.10.6

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

Firmware Version 1.0.9.3

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

Firmware Version 1.0.8.7

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

Firmware Version 1.0.5.11

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

Firmware Version 1.0.3.7

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

Firmware Version 1.0.3.2

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

Firmware Version 1.0.2.5

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

Firmware Version 1.0.2.3

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

Firmware Version 1.0.1.9

  • This is the initial version.

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