HT801 V2

HT802 V2

HT801 V2

HT802 V2

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, NET, PHONE

POWER, NET, PHONE1,
PHONE2

Factory Reset Button

Yes

Voice, Fax, Modem

Telephony Features

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

Voice Codecs

G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.722, G.723.1, G.729A/B, G.726-32, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation

Fax over IP

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

Short/Long Haul Ring Load

5 REN: Up to 1km on 24 AWG

2 REN: Up to 1km on 24 AWG

Caller ID

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

Dial Methods

DTMF, Pulse

Disconnect Methods

Busy Tone, Polarity Reversal/Wink, Loop Current

Signaling

Network Protocols

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

DTMF Methods

In-audio, RFC2833 and/or SIP INFO

Provisioning and Control

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

QoS

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

Security

Media

SRTP

Control

TLS/SIPS/HTTPS

Management

Syslog support, SSH, Telnet remote management using web browser

Physical

Universal Power Supply

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

Environmental

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

Dimensions and Weight

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

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

Compliance

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

K.21

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

K.21

Phone port for HT801 V2

Phone 1 & 2 ports for HT802 V2

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

NET port

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

Micro USB DC Power

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

Reset

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

LED Indication

 Status

Power LED

The Power LED lights up when the HT801 V2/HT802 V2 is powered on, and it flashes when the unit is booting up

NET LED

The NET LED lights up when the HT801 V2/HT802 V2 is connected to a network through the Ethernet port, and it flashes when there is data being sent or received.

Phone LED for HT801 V2

Phone LED 1&2 for HT802 V2

The phone LEDs 1 & 2 indicate the status of the respective FXS port-phone on the back panel

OFF – Unregistered

ON (Solid Blue) – Registered and Available

Blinking every second – Off-Hook / Busy

Slow blinking – FXS LEDs indicates voicemail

User Level

User

Password

Web Pages Allowed

End User Level

user

123

View all pages but can only modify basic settings

Administrator Level

admin

random password located at the back of the unit

Browse all pages and modify all settings

Viewer Level

viewer

viewer

View all pages but no changes allowed.

Product Model

Displays the Product model: HT801V2 or HT802V2.

Serial Number

Displays the device’s serial number.

Hardware Version

Hardware Version

Displays the hardware version.

Part Number

Displays the part number.

Software Version

Software Version

Vendor Default Config Version

Project

Displays the unique project ID of the custom configuration file.

Vendor Default Config Version

Displays the version of the custom configuration file.

Operation State

Software Status

Displays whether the software is running correctly on the device.

Mem

Displays the memory usage.

System Up Time

Displays the duration of the system up time.

System Current Time

Displays the HT8xx current time.

CPU Load

Displays the processor load.

Provision

Displays the last status of provisioning.

System Info

Downloads a diagnostic system information file from the device, containing technical details such as firmware version, uptime, system status, and other runtime information useful for troubleshooting.
Note: The device periodically saves the current system time to an internal time file. This is included in the System Info download package to assist with troubleshooting time-related issues.

Core Dump

Core Dump

Displays the core dump file when available and which could be downloaded for troubleshooting purpose.

MAC Address

Displays the device’s MAC address.

IPv4 Address

Displays the assigned IPv4 address.

IPv6 Address

Displays the assigned IPv6 address.

VPN IPv4 Address

Displays the connected VPN IPv4 address, if available.

VPN IPv6 Address

Displays the connected VPN IPv4 address, if available.

Network Cable Status

Displays the status and speed of the NET port, it is shown as UP if connected, and DOWN if disconnected.

PPPoE Link Up

Indicates active connectivity through PPPoE.

NAT

Displays the type of NAT used.

Certificate Type

Displays the certificate type.

Enable Voice Prompt

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

Enable Direct IP Call

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

Enabled by default.

Blocklist For Incoming Calls

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

Lock Keypad Update

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

Play Busy Tone When Account is unregistered

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

DHCP Option 17 Enterprise Number

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

ATA Settings

STUN Server

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

Keep-Alive Interval

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

Use STUN to detect network connectivity

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

Time Zone

NTP Server

This parameter sets the IP address of the NTP server. The device will obtain the date and time from the server.

Secondary NTP Server

 This feature allows users to configure the secondary NTP server, by default, no secondary NTP Server is configured.

Allow DHCP Option 42 to override NTP server

If DHCP Option 42 is enabled, the NTP server can be changed. Enabled by default.

Time Zone

Selects the time zone where the phone is located and controls the date/time display,
Note: you can set the timezone to be the self defined time.

Self-Defined Time Zone

Allows users to define their own time zone when time zone is set to “Using self-defined Time Zone”.

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

 
MTZ+6MDT+5

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

M4.1.0,M11.1.0

The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)

The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday…)

The 3rd number indicates weekday: 0,1,2,..,6 (for Sun, Mon, Tues, … ,Sat)

Therefore, this example is the DST which starts from the Second Sunday of March to the 1st Sunday of November.

Allow DHCP server to set Time Zone

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

Local Time

Language

IVR Language

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

System Ring Cadence

Cadence on+off value range (0, 16000) milliseconds.

Prompt Tone Access Code

The key pattern to get Prompt Tone. Maximum 20 digits. No default value provided

CPT Settings:

Using these settings, users can configure tone frequencies and cadence according to their preference. By default, they are set to North American frequencies.
Configure these settings with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In order to set a continuous tone, OFF should be zero. Otherwise, it will ring ON ms and a pause of OFF ms and then repeat the pattern.

Example configuration for N.A.
Dial tone:
fxsch1:f1=350@-17,f2=440@-17,c=0/0;fxoch1-4:f1=350@-17,f2=440@-17,c=0/0;

Syntax: fxsch1:f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];<br>fxoch x-y:f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]]
(Frequencies are in (10, 4000) Hz and cadence on and off are in (0, 64000) ms)

Security Settings → Web/SSH Access

Basic Settings

Web Access Mode

Selects the web page access protocol. The available options are: HTTP, HTTPS, Disabled.

The default settings is “HTTPS”.

HTTP Web Port

Defines the HTTP Web Port. 

The default setting is “80”. The valid range is 1-65533.

HTTPS Web Port

Defines the HTTPS Web Port. 
The default setting is “443”. The valid range is 1-65533.

Web Session Timeout

Defines the web session timeout in minutes.
The default setting is “10”, the valid range is 1-60.

Web Access Attempt Limit

Defines the number of failed web login attempts allowed before web access is locked.

The default setting is “5”, the valid range is 1-10.

Web Lockout Duration

Defines the duration for which web access is locked in minutes.
The default setting is “15”, the valid range is 1-60.

Enable User Level Web Access

Enables or disables user-level web access.
Disabled by default.

Enable Viewer Level Web Access

Enables or disables viewer-level web access.
Disabled by default.

SSH

Enable SSH

Enables or disables SSH access to the phone.
Enabled by default.

SSH Port

Defines the SSH port. This port cannot be the same as the Telnet Port.
The default setting is “22”. The valid range is  1-65533. 

SSH Idle Timeout

Configures SSH session Timeout in seconds. 
The default setting is “0”. The valid range is 0-86400. 

Enable Telnet

Enables or disables Telnet access.

Disabled by default.

Telnet Port

Defines the Telnet port. This port cannot be the same as the SSH Port.
The default setting is “23”. The valid range is  1-65535. 

Telnet Idle Timeout

Configures Telnet session Timeout in seconds.
The default setting is “0”. The valid range is 0-86400. 

Security Controls for SSH/Telent Access

Specifies security measures and controls that apply to the SSH/Telnet protocol to ensure secure and authorized access to the device, the following options are supported:

The Default Value is “Allow any user to set any PValue”.

Enable Web/SSH Access

Enables or Disables the Web and SSH access through the WAN port. The available options are the following:

The default setting is “Yes”.

Access Control

Allow List for WAN Side

If Enable Web/SSH Access is set to Yes or Auto. Users can configure the white List for WAN Side to be used for remote management.

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

Example:192.168.5.222 192.168.5.223 192.168.7.0/24

Note: If both blocklist and allowlist are not empty, the blocklist is processed first, followed by the allow list.

Block List for WAN Side

If Enable Web/SSH Access is set to Yes or Auto. Users can configure the black List for WAN Side to ban WAN side web access.

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

Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

Note: If both blocklist and allowlist are not empty, the blocklist is processed first, followed by the allow list.

Security Settings → User Info Management

Password Rules

Enable strict password rules

Enable or diables strict password rules.

Minimum password length

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

Required number of character classes

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

Allowed Character classes

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

Password

New Admin Password

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

Confirm Admin Password

Confirms the entered admin password.

New User Password

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

Confirm User Password

Confirms the entered user password.

New Viewer Password

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

Confirm Viewer Password

Confirms the entered viewer password.

Security Settings →  Client Certificate

SIP TLS

Enable/Disable Weak Cipher Suites

Controls whether the device allows the use of weakened or deprecated TLS cipher suites. You can either allow weak TLS ciphers or selectively disable specific weak encryption, authentication, or key-exchange algorithm. Disabling weak cipher suites is recommended to improve security and comply with modern TLS standards.
The available option are : 

By default, it is set to “Disable All Of The Above Weak TLS Ciphers Suites”

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. The maximum supported length is 8192.

SIP TLS Private Key

Specifies TLS private key used for SIP over TLS is in X.509 format. The maximum supported length is 4069.

SIP TLS Private Key Password

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

TLS Version

Minimum TLS Version

This feature allows customer to choose desired Minimum TLS Version.
The available options are:

The default setting is “TLS 1.2”.

Maximum TLS Version

This feature allows customer to choose desired Maximum TLS Version.

The available options are:

The default setting is “Unlimited”.

Custom Certificate

Custom Certificate

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

Security Settings →  Trusted CA Certificates

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

These trusted CA certificates will be used for the authentication server TLS certificate

Security Settings → SCEP

Basic Settings

Enable SCEP

Enables or disables the SCEP feature.

Disabled by default.

SCEP Server URL

Defines the SCEP server URL (CA or RA). The URL must include the protocol prefix. Currently, only HTTP is supported.

CA FingerPrint Infomation

Defines the fingerprint of the CA certificate.

Challenge Password

Defines the temporary challenge password included in the Certificate Signing Request (CSR) when requesting a new certificate.

Local configuration

Local Public and Private Key Length

Specifies the length of the local public and private keys.

The default setting is “3072 bit”.

The Name of Certificate Applicant

Defines the name of the certificate applicant. This field is required.

The Country of Certificate Applicant Located

Selects the country where the certificate applicant is located.

The State of Certificate Applicant Located

Defines the state or province where the certificate applicant is located.

The Locality of Certificate Applicant Located

Defines the locality of where the certificate applicant is located.

The Organization of Certificate Applicant Located

Defines the location of the organization to which the certificate applicant belongs.

The Organization Unit of Certificate Applicant

Defines the organization unit of the certificate applicant, such as a company, government agency, or nonprofit organization.

HTTP Authentication

CID

Configures the HTTP authentication user ID used to access the CA server.

Password

Configures the HTTP authentication password used to access the CA server.

Certificates and Updates

CA Certificates

Displays the CA certificate obtained through SCEP.

Note: Copy the displayed CA certificate and import it into the trusted CA certificate store on the CA server.

Client Certificate Alternative Name

Configures additional identifiers for the client certificate. These values are added as Subject Alternative Name (SAN) entries on the server and allow the device to be identified using multiple aliases. 

In the configuration file, different aliases are separated by semicolons (;).

The available options are:

The default setting is “$MAC.local”.

Update Interval before Certificate Expiration

Defines the number of days (1–30) before expiration to trigger certificate renewal. The system attempts renewal when the remaining validity is less than the defined interval.

Pressing Update Now starts the renewal manually.
The default setting is “20”.

Scope of Use

Configures where the CA and client certificates are applied. Certificates for selected modules are managed automatically via SCEP, without manual upload or renewal.

The available options are: Custom Certificate, 802.1X, OpenVPN, TR069.
The default setting is “Custom Certificate”.

TR-069

Enable TR-069

Enable/disable the “CPE WAN Management Protocol” (TR-069). The default setting is Yes.
Note: When configured, users can send a “false ring” signal through the TR-069 protocol for testing pruposes.

ACS URL

ACS Username

Enter username to authenticate to ACS.

ACS Password

Enter password to authenticate to ACS.

Periodic Inform Enable

If scheduled connection is enabled, the phone will send connection requests to the ACS server regularly. Default setting is “No”.
Note: FXS port status can be detected through TR-069.

Periodic Inform Interval

Connection Request Username

Configure the ACS username for connecting the phone.

Connection Request Password

Configure the ACS password for the connected phone.

Connection Request Port

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

CPE SSL Certificate

Configure the certificate file for the telephony adapter to connect to the ACS over SSL.

CPE SSL Private Key

Specify the certificate key for the phone adapter to connect to the ACS via SSL.

SNMP Settings

Enable SNMP

Enables the Simple Network Management Protocol.
Default is “No”

SNMP Version

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

SNMP Port

Listening Port of SNMP. Valid range is 161 or 1025-65535. (Default is 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. Valid range is 162 or 1025-65535. (Default 162).

SNMP Trap Version

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

SNMP Trap Interval

 Time interval in minutes between traps. (Default is 5).

SNMPv1/v2c Community

Name of SNMPv1/v2c community.

SNMPv1/v2c Trap Community

 Name of SNMPv1/v2c trap community.

SNMPv3 User Name

 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 User Name

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.

MIB

 Download Management Information Base (MIB) file.

Management Interface

Enable Management Interface

 Allows activation of a dedicated interface for remote management and configuration of the device. Disabled by Default.

Management Access

 Defines the allowed methods (protocols) and permissions for remote access and configuration of the device, two options can be selected:

Enable SNMP Through Management Interface

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

Enable TR069 Through Management Interface

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

Enable Syslog Through Management Interface

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

Enable Radius Through Management Interface

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

Management Interface IPv4 Address

802.1Q/VLAN Tag

 Allows the tagging of network packets with VLAN information to segment and prioritize network traffic.
The Valid Range is 0-4094, Default value is 0.

802.1p priority value

 Assigns a priority value to network packets for Quality of Service (QoS) and traffic prioritization purposes.
The Valid Range is 0-7, Default value is 0.

IPv4 address type

 Configures the address type of the management network port, the options are:

IP Address

Configure the IPv4 address of the management network port.

Subnet Mask

Defines the subnet mask.

Default Router

 Defines default gateway.

DNS Server 1

Defines DNS server 1 address.

DNS Server 2

Defines DNS server 2 address.

Enable RADIUS Web Access Control

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

Action upon Radius Auth Server Error

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

Enable Bypass Locked State

Once enabled, RADIUS accounts can bypass the webUI locked state caused by local accounts.
Disabled by default.

RADIUS Auth Protocol

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

RADIUS Auth Server Address

Configures RADIUS authentication server address.

RADIUS Auth Server Port

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

RADIUS Shared Secret

Configures the shared secret key.

RADIUS VSA Vendor ID

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

RADIUS VSA Access Level Attribute

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

E911

Enable E911

Enable Enhanced 911 call. Default is disabled.

E911 Emergency Numbers

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

Geolocation-Routing Header

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

Priority Header

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

Geolocation Header Format

Selects the format used for the SIP Geolocation header when sending emergency calls:

Set to “Legacy” by default.

HELD

HELD Protocol

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

HELD Synchronization Interval

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

HELD Location Types

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

HELD Use LLDP Information

If set to “Yes”, LLDP protocol will be used to gather information about the HT device including information about the ChassisID and PortID. Otherwise, the system defaults to using the MAC address of the gateway and phone.

Default setting is “No”.

HELD NAI

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

HELD Identity 1-10

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

HELD Identity 1-10 Value

Defines the HELD Identity value.

Location Server

Location Server

Configures the location server (LIS) address.

Location Server Username

Configures Location Server (LIS) username.

Location Server Password

Configures Location Server (LIS) password.

Secondary Location Server

Configures secondary location server (LIS) address.

Secondary Location Server Username

Configures secondary location server (LIS) username.

Secondary Location Server Password

Configures secondary location server (LIS) password.

Ethernet Settings

General Settings

Internet Protocol

Selects one of the following IP protocol modes:

Note: Make sure to reboot the HT801 V2/HT802 V2 unit for the changes to take effect.

IPv4

IPv4 address type

Allows users to configure the appropriate network settings on the HT801 V2/HT802 V2 to obtain IPv4 address. Users could select DHCP, Static IP or PPPoE. By default, it is set to DHCP.

Host Name (Option 12)

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

Allows user to configure DHCP domain name. This option specifies the domain name that the client should use when resolving hostnames via the Domain Name System. This field is optional.

Vendor Class ID (Option 60)

Exchanges vendor class ID by clients and servers to convey particular configuration or other identification information about a client. Default is HT8XXV2.

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.

 Preferred DNS server 1-4

Specifies preferred DNS server to use when DHCP or PPPoE are set.
You can set up yo 4 Preferred DNS Servers.

IPv6

IPv6 address type

Configures the IPv6 address of the phone:

Static address type

Static IPv6 Address

When using fully static type IPv6, enter a static IPv6 address.

IPv6 Prefix Length

Enter the static IPv6 address prefix length.

DNS Server 1

Enter DNS server 1 address.

DNS Server 2

Enter DNS server 2 addres.

Preferred DNS Server

Enter preferred DNS server address.

802.1x Settings

802.1x Mode

 Allow users to enable/disable 802.1X Mode. The modes available are:

Disabled by default.

802.1x Identity

Enter 802.1X identity information.

802.1x Password

Enter 802.1X MD5 password.

802.1x CA Certificate

 Paste down the 802.1X certificate .pem file.

802.1x Client Certificate

Paste down the Client.pem certificate file containing certificate and key. Maximum allowed length is 8192.

Enable DDNS

Enable DDNS function. Disabled by default

DDNS Server

Select DDNS server used, options are:

DDNS Username

Sets the username of the DDNS server.

DDNS Password

Sets the password of the DDNS server.

DDNS Hostname

Sets the domain name of the DDNS server.

DDNS Hash

Sets the hash value of the DDNS server.

OpenVPN® Settings

Enable OpenVPN®

Enables or disables the OpenVPN® feature.
Disbaled by default.

OpenVPN® Server Address

Specifies the IP address or FQDN of the primary OpenVPN® server.

OpenVPN® Port

Configures the port of the primary OpenVPN® server.
The default setting is “1194”.

OpenVPN® Server Secondary Address

Specifies the IP address or FQDN of the secondary OpenVPN® server. If the primary OpenVPN® server is unavailable, the device automatically switches to the secondary server to maintain a secure VPN connection.

OpenVPN® Secondary Port

Configures the port of the secondary OpenVPN® server.
The default setting is “1194”.

Randomly Select Server

Enables or disables random server selection. When enabled, OpenVPN randomly selects a configured server when initiating a connection. When disabled, servers are used in the order in which they are configured.
Disabled by default.

OpenVPN® Interface type

Specifies the Interface type of OpenVPN®. The available options are: TAP or TUN.
The default setting is “TUN”.

OpenVPN® Transport

Specifies the Transport Type of OpenVPN®. The available options are: UDP or TCP.
The default setting is “UDP”.

Enable OpenVPN® LZO Compression

Enable or disables OpenVPN® LZO Compression. 
Enabled by default.

OpenVPN® Encryption

 Select the OpenVPN® Encryption algorithm. The available options are: 

The default setting is “BF-CBC 128 bit default key”.

OpenVPN® Diges

Select the OpenVPN® Digest. Default is SHA1.

OpenVPN® Username

Configure the OpenVPN® username.

OpenVPN® Password

 Configure the OpenVPN® password.

OpenVPN® CA

Specifies the OpenVPN® CA. The maximum length is 8192.

OpenVPN® Certificate

Specifies the OpenVPN® Certificate. The maximum length is 8192.

OpenVPN® Client Key

Specifies the Client Key. The maximum length is 8192.

OpenVPN® Client Key Password

Configures the OpenVPN® Client Key Password. The maximum length is 64.

OpenVPN® TLS Key

Specifies the static TLS key used by TLS-Auth or TLS-Crypt to authenticate or encrypt the OpenVPN control channel.

OpenVPN® TLS Key Type

Selects how the static TLS key is used to protect the OpenVPN control channel. The available options are:

The default setting is “none”.

OpenVPN® Additional Options

Defines additional OpenVPN configuration parameters that are appended to the OpenVPN configuration file. Parameters must be separated by semicolons (;).
Use this option with caution. Ensure the specified parameters are supported by OpenVPN and do not unintentionally override other configured settings.

Example: connect-retry 5;tls-client;keepalive 10 60

Advanced Settings

Enable LLDP

Controls the LLDP (Link Layer Discovery Protocol) service.

The default setting is “Enabled”.

LLDP TX Interval

Configures LLDP TX Interval (in seconds). Valid range is 1 to 3600.

The Default value is 60.

Enable CDP

Enables/Disables CDP “Cisco Discovery Protocol”.

The default setting is “Enabled”.

Enable Manual VLAN Configuration

Allows the device to keep the VLAN configuration without applying it.

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets that can be used to identify and distinguish different VLANs.

Valid range is 0 to 4094. The default value is 0.

Layer 2 QoS SIP 802.1p

Prioritizes SIP traffic at Layer 2 using the 802.1p standard for Quality of Service (QoS).

The valid range 0-7. The default value is 0.

Layer 2 QoS RTP 802.1p

Prioritizes RTP (Real-time Transport Protocol) traffic at Layer 2 using the 802.1p standard for Quality of Service (QoS).

The valid range 0-7. The default value is 0.

Use DNS to detect network connectivity

Use DNS to detect WAN side network issues.

The default setting is “No”.

Use ARP to detect network connectivity

Use ARP to check network connection.

The default value is “Yes”.

Maximum Transmission Unit (MTU)

Configures the network Maximum Transmission Unit (MTU) in bytes, which defines the largest packet size that can be transmitted. The default is 1500 bytes, with a valid range of 576–1500 bytes.

Firewall settings

Black List for WAN Side Port

Ports on the blocklist will be disabled on the device.

Proxy

HTTP Proxy

Specifies the HTTP proxy URL that the phone uses to send HTTP traffic. The proxy server acts as an intermediary, routing packets from the phone to their destination.

HTTPS Proxy

Specifies the HTTPS proxy URL that the phone uses to send traffic. The proxy server acts as an intermediary, routing encrypted packets from the phone to their destination.

Bypass Proxy for

Specifies URL that should be accessed directly instead of through the configured proxy. Traffic matching this entry bypasses the proxy server.

Static DNS Cache

NAPTR

NAPTR DNS Cache Name

The domain name to which this resource record refers.

Maximum length is 512 characters.

NAPTR DNS Cache Time Interval (s)

(Time to Live) refers to the time interval that the record may be cached before the source of the information should be consulted again.

Valid range is from 300 to 65535. Default value is “300”.

NAPTR DNS Cache Order

Specifies the order in which the NAPTR record need to be processed, low numbers are processed before high numbers.

Valid range is from 0 to 65535. Default value is “0”.

NAPTR DNS Cache Preference

Refers to the order in which NAPTR records with the same “Order” values need to be processed. records are processed from lower preference numbers to higher preference numbers.

Valid range is from 0 to 65535. Default value is “0”.

NAPTR DNS Cache Replacement

This parameter contains the next FQDN to query for NAPTR, SRV, or address records.

Maximum length is 64 characters.

NAPTR DNS Cache Service

Specifies the services available in this domain. The replacement field is used to get to this service.

It can also specify the protocol used to communicate with the server that offers this service.

In SIP, three services are defined along with their resolution services (resolution services are defined after the “+” sign):

Maximum length is 64 characters and default value is “SIP+D2U”.

SRV

SRV DNS Cache Name

The domain name of the wanted service.

Maximum supported length is 512 characters.

SRV DNS Cache Time Interval (s)

(Time to Live) refers to the time interval that the record may be cached before the source of the information should be consulted again.

Valid range is from 300 to 65535. Default value is “300”.

SRV DNS Cache Priority

Enables to prioritize target hosts that support the given service. Lowest number will get the highest priority. Target hosts with the same priority will be tried in an order defined by the weight field.

Valid range is from 0 to 65535. Default value is “0”.

SRV DNS Cache Weight

Specifies the weight for entries with the same priority. Larger weight will get higher priority.

Valid range is from 0 to 65535. Default value is “0”.

SRV DNS Cache Target

The domain name of the target host.

Maximum supported length is 64 characters.

SRV DNS Cache Port

The port on the target host of the specified service.

The valid range is 0-65535. Default value is “0”.

A

A DNS Cache Name

The domain name to which the record refers.

Maximum supported length is 512 characters.

A DNS Cache Time Interval (s)

(Time to Live) refers to the time interval that the record may be cached before the source of the information should be consulted again.

Valid range is from 300 to 65535. Default value is “300”.

A DNS Cache IP Address

The IP address associated with the DNS record name.

Upgrade → Firmware

Upgrade via Manual Upload

Firmware

Uploads the .bin firmware file.

Upgrade via Network

Upgrade Via

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

Firmware Server Path

Sets IP address or FQDN of firmware server. The URL of the server that hosts the firmware release.

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

HTTP/HTTPS/FTP/FTPS User Name

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

HTTP/HTTPS/FTP/FTPS Password

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

Firmware File Prefix

Checks if firmware file is with matching prefix before downloading it. This field enables user to store different versions of firmware files in one directory on the firmware server.

Firmware File Postfix

Checks if firmware file is with matching postfix before downloading it.
This field enables user to store different versions of firmware files in one directory on the firmware server.

Upgrade → Config File

Manual Configuration

Upload Configuration

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

Restore From Backup Configuration

Restore From Backup Configuration.

Download Device Configuration

Downloads Device Configuration in .txt format.

Download Device XML Configuration

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

Export Backup Configuration

Export Backup Configuration in .xml format.

Configure via Network

Config Upgrade via

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

Config Server Path

Sets IP address or FQDN of configuration server. The URL of the server that hosts the configuration file to provision HT80xV2.

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

HTTP/HTTPS/FTP/FTPS User Name

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

HTTP/HTTPS/FTP/FTPS Password

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

XML Config File Password

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

Config File Prefix

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

Config File Postfix

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

Upgrade → Provision

Allow DHCP Option 66 or 160 to override server

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

Notes:

Auto Provision

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

Enable using tags in URL

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

Additional Override DHCP option

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

Automatic Upgrade

Automatic Upgrade

Specifies when the firmware upgrade process will be initiated; there are 4 options:
Note: The HT80xV2 will only do an upgrade once at boot up.
Check every X minutes: User needs to specify a period in minutes.
Check every day: User needs to specify the start hour and the end hour of the day (0-23).
Check every week: User needs to specify “Day of the week (0-6)”. (Day of week is starting from Sunday).
Default is No.

Randomized Automatic Upgrade

Randomized Automatic Upgrade within the range of hours of the day or postpone the upgrade every X minute(s) by random 1 to X minute(s).

Firmware Upgrade and Provisioning

Defines when the device checks the server for new firmware and configuration files:

Default value is set to “Always Skip the Firmware Check”.

Config Provision

Download and Process All Available Config Files

This feature allows users to download and process all available config files. By default, the device will provision the first available config in the order of cfgMAC > cfgMAC.xml > cfgMODEL.xml > and cfg.xml (corresponding to device-specific, model-specific, and global configs). 

If this option is enabled, the device will inverse the downloading process to cfg.xml > cfgMODEL.xml > cfgMAC.bin > cfgMAC.xml and add cfgMAC_override.xml. The following files will override the files that have already been loaded and processed.

The default value is “No”.

Config Provision Order

Administrators can define the preferred order in which the device checks for configuration files during startup. The system will sequentially attempt to load the listed files, and try to apply the first successfully retrieved configuration, the files available are:

Upgrade → Advanced Settings

Verify host when using HTTPS

 Enables / disables the host verification when using HTTPS. Disabled by default.

Configuration File Types Allowed

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

Enable SIP NOTIFY Authentication

The device will challenge NOTIFY with 401 when set to “Yes”.
Set to “Yes” by default.

Validate Server Certificates

Configures whether to validate the server certificate when downloading the firmware/config file. If set to “Yes”, the device will download the firmware/config file only after the server is validated.
Disabled by default
Note: This option requires a reboot to take effect.

Syslog

Syslog Server

URL or IP address of the syslog server.

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)

Syslog Protocol

Allow encrypted SSL/TLS transmission to the syslog server if SSL/TLS is selected, the default value is UDP.

Notes:

Syslog Header Format

Determines how the headers in syslog messages are formatted, the options are:

Send SIP Log

Sets whether to include SIP logs in the system log

Default value is “No”.

Debug

Ethernet Capture

 Enables the capturing and analysis of Ethernet network traffic for troubleshooting and monitoring purposes.

With secret key information

Allows users to make packet capture including the secret key to decrypt the captured TLS packets. Default value is Yes.

Port Record

Allows the recording of port-level call information, such as call duration, caller ID, and call status, for monitoring and logging purposes.

Core Dump

Provides generated core dump file if unit malfunctions. Clean will be displayed if no issues.

GR909

Test To Run

Selects the type of test that wll be performed for the specified line, the following tests can be performed:

FXS Line

Defines the FXS line under test.

Run Interval

Defines the run Interval in seconds, the default value is 120 seconds, the default range is 0-604800 seconds.

CDR Records

Displays a list of the last 1000 call records, the ionformation included in the records are the following:

Clicking the “Download” button will download the last 1000 call records.
Clicking the “Delete” button will delete the last 1000 call records.

SIP Messages

The SIP messages tab displays a list of SIP headers that are captured after each triggered call, the information displayed on the tab are similar to the information that can be captured using the wireshark tool, and filtering by SIP protocol.

Clicking the “Download” icon will download the whole log of the captured traces.

Clicking “Delete” will delete all the captured traces.

This tool can be useful for network administrators that want to treoubleshoot the call flows without having to use an external tool, for analyzing the network

Note: The SIP PUBLISH method (RFC 6035) enables the gateway to send status updates and event notifications to a SIP server or PBX. This allows for real-time monitoring, presence tracking, and improved call management in VoIP networks.

Reboot

Automatic restart

This option triggeres an automatic reboot of the unit, the available options are:

Restore Factory

Reset Type

Note: After choosing the reset type, you will have to click the reset button for it to take effect.

FXS PORT x → General Settings

Account Registration

Profile 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: fe80::20b:82ff:fe75:211d). This is the primary SIP server used to send/receive SIP messages from/to HT801 V2/HT802 V2.

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 fe80::20b:82ff:fe75:211d:). 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, IPv4: 192.168.5.170, or IPv6: fe80::20b:82ff:fe75:211d). It’s Used by HT801 V2/HT802 V2 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and only outbound proxy can correct the problem

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: fe80::20b:82ff:fe75:211d). By default, this field is left empty.

Prefer Primary Outbound Proxy

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

From Domain

Allows users to add the actual domain name, it will override the from header.This is an optional configuration.

Allow DHCP Option 120 (override SIP server)

Configures the HT801 V2/HT802 V2 to collect SIP server address from DHCP option 120. Default is No.

SIP User ID

Defines The user part of a SIP address

Authenticate ID

Defines the Authenticate ID of the User

Authenticate Password

Defines the Authenticate password

Name

Defines the name of the user

Tel URI

If the device has an assigned PSTN telephone number, this field should be set to “User=phone”. A “User=phone” parameter will be attached to the Request-Line and “To” header in the SIP request to indicate the E.164 number. If set to “Enabled”, “tel:” will be used instead of “sip:” in the SIP request.
Disabled by default.

Network Settings

Layer 3 QoS SIP DSCP

the Diff-Serv value for SIP packets in decimal, the range is 0-63, default value is 26

Layer 3 QoS RTP DSCP

Diff-Serv valuefor RTP packets in decimal, the range is 0-63, default value is 46

DNS Mode

Selects DNS mode to use for the client to look up server. One mode can be chosen.
A Record (Default): resolves IP Address of target according to domain name.
SRV: DNS SRV resource records indicate how to find services for various protocols.
NAPTR/SRV: Naming Authority Pointer according to RFC 2915.
Use Configured IP: If the SIP server is configured as domain name, device will not send DNS queries, but will use “Primary IP” or “Backup IP” to send SIP message if at least one of them is not empty. It will try to use “Primary IP” first, after 3 tries without any response, it will switch to “Backup IP 1”, then “Backup IP 2”, and then it will switch back to “Primary IP” after 3 retries.

DNS SRV Failover Mode

 Configure the preferred IP mode when DNS Mode is SRV or NAPTR/SRV.
• Default SIP request will always be sent to the address with the top priority based on the SRV query result, even if this address is different from the registered IP address.
• Saved one until DNS TTL SIP request will always be sent to the registered IP address until DNS TTL expires or registered IP address is unreachable.
• Saved on until no response SIP request will always be sent to the registered IP address only until registered IP address is unreachable.
• Failback follows failback expiration time: the primary server regains control only after the failback expiration time, allowing the secondary server to handle requests until then.

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).

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).

Register Before DNS SRV Failover

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

Primary IP

The main IP address used for communication and control.

Backup IP1

The secondary IP address used as a backup for communication in case the Primary IP fails.

Backup IP2

 The third IP address used as an additional backup for communication in case both the Primary IP and Backup IP1 fail.

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, Auto, VPN.

The default setting is “No”.

Use NAT IP

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

Proxy-Require

Determines a SIP Extension to notify the SIP server that the HT801 V2/HT802 V2 is behind a NAT/Firewall.

FXS PORT x → SIP Settings

SIP Basic Settings

SIP Registration

Controls whether the HT801 V2/HT802 V2 needs to send REGISTER messages to the proxy server. The default setting is Yes.

SIP Transport

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

Unregister On Reboot

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with “Expires=0” parameter to the SIP server. This will unregister the SIP account under the concerned FXS page.Unregister on reboot option can be set to “No”, “All” or “Instance”.

Default value is set to “No”

Outgoing Call without Registration

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

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).

Re-register before Expiration

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

SIP Registration Failure Retry Wait Time

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

Use Random SIP Registration Failure Retry Wait Time

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

Default setting is disabled

Random SIP Registration Failure Retry Wait Time Range

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

Valid values range from 60 to 600 seconds.

SIP Registration Failure Retry Wait Time upon 403 Forbidden

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

Port Voltage Off upon no SIP Registration or SIP Registration Failure

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

Delay Time of Port Voltage Off Timer Since Boot

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

Enable SIP OPTIONS/NOTIFY Keep Alive

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

SIP OPTIONS/NOTIFY Keep Alive Interval

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

SIP OPTIONS/NOTIFY Keep Alive Max Lost

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

SUBSCRIBE for MWI

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

Subscribe Retry Wait Time upon 403 Forbidden

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

Local SIP Port

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

Use Random SIP Port

 Controls whether to use configured or random SIP ports. This is usually necessary when multiple HT801 V2/HT802 V2 are behind the same NAT.
The default is Yes.

Hold Target Before Refer

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

Refer-To Use Target Contact

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

Remove OBP from Route Header

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

Support SIP Instance ID

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

Support outbound

Enables/disables outbound support.

Support GRUU

Enables/disables GRUU support.

SIP URI Scheme When Using TLS

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

Use Actual Ephemeral Port in Contact with TCP/TLS

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

Use Request Routing ID in SIP INVITE Header

This options enables the inclusion of a routing ID in the SIP INVITE header for directing the request to a specific endpoint or route.
Note: the HT supports multiple DIDs per FXS port, this allows single port to receive calls for multiple numbers based on the configured Routing ID or other call-matching settings.

Use Privacy Header

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

Use P-Preferred-Identity Header

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

Use P-Access-Network-Info Header

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

Use P-Emergency-Info Header

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

Use P-Asserted-Identity Header

Enables or disables sending the P-Asserted-Identity Header in the SIP INVITE messages.

Disabled by default.

Use P-Early-Media Header

Enables or disables sending the P-Early-Media Header in the SIP INVITE messages.

Disabled by default.

SIP REGISTER Contact Header Uses

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

SIP REGISTER Contact Header Uses

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

SIP T1 Timeout

Defines T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, HT801 V2/HT802 V2 will attempt to send a request to a SIP server. The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time is defined by T2. The default is 0.5 seconds.

SIP T2 Interval

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

SIP Timer B

Configures SIP Timer B defined in RFC3261.

0-128 sec. Default 0, which means using 64*T1.

When this SIP Timer is set to 0, its value is calculated as:

SIP Timer B = SIP T1 Timeout × 64

E.g., If the SIP T1 Timeout is set to its default value of 0.5 seconds, then:

SIP Timer B = 0.5 × 64 = 32 seconds

SIP Timer D

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

SIP Timer F

Configures SIP Timer F defined in RFC3261.

0-128 sec. Default 0, which means using 64*T1.

When this SIP Timer is set to 0, its value is calculated as:

SIP Timer F = SIP T1 Timeout × 64

E.g., If the SIP T1 Timeout is set to its default value of 0.5 seconds, then:

SIP Timer F = 0.5 × 64 = 32 seconds

Disable Multiple m line in SDP

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

Enable 100rel

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

Add Auth Header On Initial REGISTER

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request.
Default is No.

Enable Call Waiting Alert-Info in 180 Ringing Response

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

SIP User-Agent

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

SIP User-Agent Postfix

Configures the SIP User-Agent Postfix

Add MAC in User-Agent

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

Use MAC Header

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

CSeq Tracking Mode

 Defines how the device tracks SIP CSeq values.

Use per-dialog if the SIP platform expects continuous CSeq sequencing within a dialog (per RFC behavior); per-call is typically more tolerant with certain SIP servers and proxies.
It is set to “per-call” by default.

Session Timer

Enable Session Timer

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

Session Expiration

Enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or 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). Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it. Valid range is 90-64800 seconds. Default is No.

Callee Request Timer

 Uses session timer when receiving inbound calls with session timer request.
Default is No.

Force Timer

Uses session timer even if the remote party does not support this feature. Selecting “No” will enable session timer only when the remote party supports it. To turn off Session Timer, select “No” for Caller and Callee Request Timer, and Force Timer. Default is No

UAC Specify Refresher

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

UAS Specify Refresher

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

Force INVITE

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

When To Restart Session After Re-INVITE received

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

FXS PORT x → Codec Settings

DTMF Settings

Preferred DTMF

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

Force DTMF to be sent via SIP INFO simultaneously

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

Inband DTMF Duration

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

DTMF Payload Type

 Defines payload type for DTMF using RFC2833.

Enable Multiple Sampling Rates in SDP telephone-event

This option allows the device to advertise multiple supported sampling rates for DTMF events in the SDP (Session Description Protocol), which enhances compatibility with different VoIP systems by ensuring proper DTMF signal interpretation across varying codec sampling rates.
When the option is enabled, the HT includes multiple supported sampling rates for DTMF (telephone-event) in the SDP negotiation, which allows better interoperability with VoIP systems using different codec rates.
When disabled, only a single default sampling rate is advertised, which may lead to DTMF tone mismatches if the remote system expects a different rate.

Inband DTMF Tx Gain

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

DSP DTMF Detector Duration Threshold

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

DSP DTMF Detector Min Level

determines the minimum level of DTMF tones that the DSP DTMF detector can reliably detect, range is -45-0 dBm, default is -25

DSP DTMF Detector Snr

Sets the signal-to-noise ratio threshold for the DSP DTMF detector, determining the sensitivity to distinguish DTMF tones from background noise in the audio signal.
Range: 0-12, default is 1

DSP DTMF Detector Deviation

 Specifies the allowed frequency deviation from standard DTMF tones that the DSP DTMF detector can accommodate.
Range: 0-25, default is 25

DSP DTMF Detector Twist

Defines the maximum deviation from ideal tone frequencies allowed for reliable DTMF detection by the DSP.
Range: 0-12dB, default is 5

Enable DTMF Negotiation

Uses DTMF order without negotiation. Default is No.

RFC2833 Events Count

This feature allows users to customize the count of RFC2833 events. Default is 8.

RFC2833 End Events Count

This feature allows users to customize the count of RFC2833 end events. Default is 3.

Vocoder Settings

Vocoder Settings (in listed order)

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

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.

G.723 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.

Enable OPUS Stereo in SDP

Enabkes OPUS stereo in SDP. Default is Yes.

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.

Enable Audio RED with FEC

Enables the use of audio redundancy (RED) with forward error correction (FEC) to enhance audio quality and resilience against packet loss in real-time communication systems such as VoIP

Audio FEC Payload Type

Specifies the payload type used for transmitting audio forward error correction (FEC) packets in a VoIP system.

Audio RED Payload Type

Defines the payload type assigned to audio redundancy (RED) packets in a VoIP system,

Silence Suppression

If set to “Yes”, when the device detects that there is no voice flow in the call, the device will send out a small number of VAD packets (instead of voice packets)
Disabled by default.

Use First Matching Vocoder in 200OK SDP

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

Fax Mode

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

T.38 Max Bit Rate

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

Enable T.38 TEP

Enables or disables T.38 Talker Echo Protection (TEP).

When enabled, TEP is applied during T.38 fax calls to help reduce echo interference. However, it may affect transmission reliability on some connections. Enable this feature only if echo is impacting fax quality.

Disabled by default.

Re-INVITE After Fax Tone Detected

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

Re-INVITE Upon CNG Count

This feature enables users to initiate a re-invite request for fax transmission when the fax machine is the sender.

The feature is controlled by an adjustable setting, where a value of 0 disables the feature, and a value of 1 or greater allows the ATA to automatically initiate the re-invite request once the CNG (Calling Tone) count threshold is met.

The valid setting range for this feature is between 0 and 6. The default setting is 0.

Jitter Buffer Type

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

Jitter Buffer Length

High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.
Medium (initial 100ms, min 20ms, max 200ms).
Low (initial 50ms, min 10ms, max 100ms).

Crypto Life Time

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

RTP Setting

Local RTP Port

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

Use Random RTP Port

Controls whether to use configured or random RTP ports. This is usually necessary when multiple HT801 V2/HT802 V2 are behind the same NAT.
The default is Yes.

Random RTP Port Range

Set Random RTP Port Range：1024-65535, Min – Max.

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.

Enable RTCP

 Allows users to enable RTCP. The default setting is “Yes”.

RTP/RTCP Keep Alive On Hold

Enables or disables RTP/RTCP keep-alive packets during call hold to maintain connectivity and prevent session timeouts.
Disabled by default.

SRTP Mode

Selects SRTP mode to use:

Default is Disabled

SRTP uses SDP Security Description to exchange key. Please refer to the below links for more information

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

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

SRTP Key Length

 Allows users to select supported SRTP Key Length. the available values are :

VQ RTCP-XR Collector Name

Configures the host name of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Address

Configures the IP address of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Port

Configures the port of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.
The default setting is “5060”.

FXS PORT x → Analog Signal Line Configuration

Analog Signal Line Configuration

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

Configures start and stop tone for DTMF caller ID scheme.

Polarity Reversal

Reverses the polarity upon call establishment and termination.
Default is No.

Loop Current Disconnect

Allows the traditional PBX used with HT801 V2/HT802 V2 to apply this method for signaling call termination. The method initiates a short voltage drop on the line when the remote (VoIP) side disconnects an active call. The default is No.

Play Busy/Reorder Tone Before Loop Current Disconnect

Enables or disables playing the busy/reorder tone before loop current disconnect upon call fail.
Disabled by default

Busy/Reorder Tone Delay Timer After Loop Current Disconnect

Configures the Busy/reorder tone delay timer after the loop current disconnects.
The default setting is “25”. The valid range is 0-30.

Loop Current Disconnect Duration

 Configures the duration of voltage drop described in the topic above. HT801 V2/HT802 V2 support a duration range from 100 to 10000ms. The default value is 200.

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 or New Zealand pulse dialing 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 V2/HT802 V2 support 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 V2/HT802 V2 support 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 (ie. the ATA is at user site), adjust volume using the Rx gain level parameter under the FXS port configuration page.
If voice volume is too low at the other end, user may increase the far end volume using the Tx gain level parameter under the FXS port configuration PAGE.

Disable Line Echo Canceller (LEC)

Disables the LEC per call base. Recommended for Fax/Data calls.
Default is No.

Ring Frequency

Chooses an appropriate ringing frequency. Default is 20Hz.

Ring Power

Selects ringing power settings, the options are: 45Vrms, 50Vrms, and 55Vrms , the higher the value, the more powerful the ring is.
Default is 45Vrms.

OnHook DC Feed Current

This feature is used to adjust DC feed current.

FXS PORT x → Call Settings

Dial Setting

Off-hook Auto Dial

Configures the user ID, extension or IP address to dial automatically when offhook.

Offhook Auto-Dial Delay

Sets the delay time before automatic dialing begins after going off-hook on an FXO port.
Range is 0-60 seconds, default is 0

No Key Entry Timeout

Initiates the call within this time interval if no additional key entry during dialing stage. 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 Incomplete Address” responses. Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). Default is No.
This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

Adds specified prefix to dialed number.

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. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.

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.

Enable  # as Redial Key