Grandstream Networks, Inc.

HT813
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HT813 - Administration Guide

Thank you for purchasing Grandstream’s HT813, It is the first ATA in the HandyTone 81x series offering functions as a true 3-in-1 gateway for PSTN
network, analog telephone FXS interface and IP network. It enables remote call origination and termination from/to PSTN. The HT813 is specifically
designed to be an easy to use and affordable VoIP solution for both the residential user and the remote user.

This administrator guide will help you learn how to operate and manage your HT813 Analog Telephone Adaptor and make the best use of its many
upgraded features including simple and quick installation.

PRODUCT OVERVIEW
The HT813 is an analog telephone adapter (ATA) featuring 1 analog telephone FXS port and 1 PSTN line FXO port. The integration of FXO and FXS
ports enables remote call origination and termination to and from the PSTN line. The 1 FXS port allows for extension of a VoIP service to 1 analog
phone. HT813’s ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take
advantage of VoIP on analog phones and enables service providers to offer high quality IP service.

Feature Highlights

The following table contains the major features of the HT813:

2 SIP profiles through 1 FXS port and 1 FXO port

Dual 100 Mbps LAN and WAN ports

3-way voice conferencing

Wide range of caller ID formats

HT813 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

TLS and SRTP security encryption technology to protect calls and accounts

Automated provisioning options include TR-069 and XML config files

Strong AES encryption with security certificate per unit

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

Lifeline support (FXS port will be hard-relayed to FXO port) in case of power outage.

Table 1: HT813 Features at a Glance

HT813 Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features,
languages, and upgrade/provisioning settings for the HT813.

Interfaces

Telephone Interfaces One (1) RJ11 FXS port, One (1) RJ11 FXO PSTN line port with lifeline support

Network Interface Two (2) 10/100 Mbps ports (RJ45) with integrated NAT router

LED Indicators POWER, FXO, FXS, WAN, LAN.

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

G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.723.1, G.729A/B, G.726, iLBC, OPUS, dynamic jitter buffer, advanced
Voice Codecs
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
3 REN: Up to 1km on 24 AWG
Ring Load

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

Disconnect Methods Busy Tone, Polarity Reversal/Wink, Loop Current

Signaling

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

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

DTMF Methods In-audio, RFC2833 and/or SIP INFO

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

Security

Media SRTP

Control TLS/SIPS/HTTPS

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

Physical

Input: 100-240VAC, 50/60Hz

Universal Power
Supply
Output: 12V/0.5A

Operational: 32° – 104°F or 0º – 40ºC

Environmental Storage: 14° – 140°F or -10º – 60ºC

Humidity: 10 – 90% Non-condensing

130.5 x 90.5 x 29 mm (L x W x D)
Dimensions and
Weight
Weight: 0.142Kg

Compliance
 Compliance FCC/CE/C-TICK/ITU-K.21

Table 2: HT813 Technical Specifications

GETTING STARTED
This chapter provides basic installation instructions including the list of the packaging contents and also information for obtaining the best
performance with the HT813.

Equipment Packaging

The HT813 ATA package contains:

Figure 1: HT813 Package Contents

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

HT813 Ports Description

The following figure describes the different ports on the back panel of the HT813.

Figure 2: HT813 Back Panel

Connects the analog phone / fax machine to the ATA using an

FXS
RJ-11 telephone cable.

FXO FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port.

Connects the ATA to your router, switch or modem using an Ethernet RJ45 network cable.

Connects the ATA to your PC or switch using an Ethernet RJ45 network cable.

DC 12V Connects the ATA to PSU (Output: 12V/0.5A )

 Factory reset button. Press for 7 seconds to reset factory default.
Reset
settings. Quick press will only reboot the unit.

Table 3: HT813 Connectors Definitions

Connecting HT813

The HT813 is designed for easy configuration and easy installation. To connect your HT813, please follow the steps below:

Scenario 1: Connecting the HT813 using WAN Port

When connecting HT813 using the WAN port, it will act as simple DHCP Client.

1. Insert a standard RJ11 telephone cable into the FXS port and connect the other end of the telephone cable to a standard touch-tone analog
telephone.

2. Connect the WAN port of the HT813 to a router, switch or modem using an Ethernet cable.

3. Insert the power adapter into the HT813 and connect it to a wall outlet and make sure to respect the technical specifications of the power
adapter used.

4. Power, WAN and FXS LED will be solidly lit when the HT813 is ready for use.

Scenario 2: Connecting the HT813 using LAN Port

When connecting the HT813 using the LAN port, it will act as a router and DHCP serving addresses, the devices connected with HT813 LAN will pull
DHCP addresses from your HT813.

1. Insert a standard RJ11 telephone cable into FXS port and connect the other end of the telephone cable to a standard touch-tone analog
telephone.

2. Connect a computer or switch to the LAN port of the HT813 using an Ethernet Cable.

3. Insert the power adapter into the HT813 and connect it to a wall outlet and make sure to respect the technical specifications of the power
adapter used.

4. Power, LAN and FXS LED will be solidly lit when the HT813 is ready for use.

Figure 3: Connecting the HT813

HT813 LEDs Pattern

There are four (4) LED types that help you manage the status of your HT813.

Figure 4: HT813 LEDs Pattern

LED Lights Status

Power LED The Power LED lights up when the HT813 is powered on and it flashes when the HT813 is booting up.

WAN LED The WAN LED lights up when the HT813 is connected to your network through the WAN port.

LAN LED The LAN LED lights up when the HT813 is connected to your network through the LAN port.

The FXS LEDs indicate status of the respective FXS/FXO port-phone on the back panel

OFF – Unregistered
FXS/FXO LED ON (Solid Blue) – Registered and Available

Blinking every 500 ms – Off-Hook / Busy

Slow blinking – FXS LEDs indicates voicemail

Table 4: HT813 LEDs Pattern Description

CONFIGURATION GUIDE
The HT813 can be configured via one of two ways:

The IVR voice prompt menu.

The Web GUI embedded on the HT813 using PC’s web browser.

Obtain HT813 IP Address via Connected Analogue Phone

HT813 is by default configured to obtain the IP address from DHCP server where the unit is located. To know which IP address is assigned to your
HT813, you should access to the “Interactive Voice Response Menu” of your adapter via the connected phone and check its IP address mode.
Please refer to the steps below to access the interactive voice response menu:

1. Use a telephone connected to FXS port of your HT813.

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 HT813 Interactive Voice Prompt Response Menu

The HT813 has a built-in voice prompt menu for simple device configuration which lists actions, commands, menu choices, and descriptions.
Connect analog phone to FXS port. Pick up the handset and dial “***” to use the IVR menu.

Menu Voice Prompt Options

Press “*” for the next menu option

Main
“Enter a Menu Option” Press “#” to return to the main menu
Menu

Enter 01-05, 07,10, 12-17,47 or 99 menu options

Press “9” to toggle the selection

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

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

If using “PPPoE Mode”, configure PPPoE Username and Password from web GUI to get IP from your ISP.

The current WAN IP address is announced

“IP Address “ + IP
02
address If using “Static IP Mode”, enter 12-digit new IP address. You need to reboot your HT813 for the new IP
address to take Effect.

03 “Subnet “ + IP address Same as menu 02

“Gateway “ + IP
04 Same as menu 02
address

“DNS Server “ + IP
05 Same as menu 02
address

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

PCM U / PCM A

iLBC
07 Preferred Vocoder G-726

G-723

G-729

OPUS

Announces the MAC address of the unit.

10 “MAC Address”
Note: The device has two MAC addresses. One for the WAN port and one for the LAN port. The device
MAC address announced is the address of LAN port.
 Menu Voice Prompt Options

Press “9” to toggle between Auto / Enabled / Disabled.

12 WAN Port Web Access
Default is Auto.

Announces current Firmware Server IP address.

Firmware Server IP
13
Address
Enter 12-digit new IP address.

Configuration Server IP
14 Announces current Config Server Path IP address. Enter 12-digit new IP address.
Address

Upgrade protocol for firmware and configuration update. Press “9” to toggle between TFTP / HTTP / FTP
/ FTPS or HTTPS.
15 Upgrade Protocol

Default is HTTPS.

16 Firmware Version Announces Firmware version information.

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

Always check
17 Firmware Upgrade
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.

Press “9” to reboot the device

99 “RESET” Enter MAC address to restore factory default setting

(See RESTORE FACTORY DEFAULT SETTINGS section)

“Invalid Entry” Automatically returns to main menu

This prompt will be played immediately after off hook If the device is not registered and the option
“Device not registered”
“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 menu option and 12 digits for IP address. For IP address, add 0 before the digits if
the digits are less than 3 (i.e. – 192.168.0.26 should be key in like 192168000026. No decimal is needed).

Key entry cannot be deleted but the phone may prompt error once it is detected.

Please make sure to reboot the device after changing network settings (IP Address, Gateway, Subnet…) to apply the new configuration.
Configuration via Web Browser

The HT813 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the HT813
through a web browser such as Google Chrome, Mozilla Firefox, and Microsoft’s IE.

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.

Accessing the Web UI

Via WAN port

1. You may check your HT813 IP address using the IVR on the connected phone.

Please see Obtain the HT813 IP address via the connected analogue phone

2. Open the web browser on your computer.

3. Enter the HT813’s IP address in the address bar of the browser.

4. Enter the administrator’s password to access the Web Configuration Menu.

Note: The computer must be connected to the same sub-network as the HT813. This can be easily done by connecting the computer to the same
hub or switch as the HT813.

Via LAN port

1. Power your HT813 using PSU with the right specifications.

2. Connect your computer or switch directly to your HT813 LAN port.

3. Open the web browser on your computer.

4. Enter the default LAN IP address (192.168.2.1) in the address bar of the browser.

5. Enter the administrator’s password to access the Web Configuration Menu.

6. Make sure to reboot your device after changing your settings to apply the new configuration.

Please make sure that your computer has a valid IP address on the range 192.168.2.x so you can access the web GUI of your HT813.

Web UI Access Level Management

There are three default passwords for the login page:

User Level Password Web Pages Allowed

End User Level 123 Only Status and Basic Settings

Administrator Level admin All pages

Viewer Level viewer Only checking. Not allowed to modify content.

Note
 The password is case sensitive with maximum length of 25 characters. When changing any settings, always submit them by pressing Update or
Apply button on the bottom of the page. After submitting the changes in all the Web GUI pages, if a reboot is required, the web page will prompt
the user to reboot by offering a reboot button on the web page. .

The user and viewer level access is disabled by default, to enable it,go under Advanced Settings.

The user must change his Admin/User/Viewer Account password after first time login attempt.

Saving the Configuration Changes

After users make changes to the configuration, pressing Update button will save but not apply the changes until Apply button is clicked. Users can
instead directly press Apply button. When a reboot is required to apply changes, the web page will prompt the user to reboot by offering a reboot
button on the web page.

Changing Admin Level Password

1. Access your HT813 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 Advanced Settings 🡪 New Admin Password and enter the new admin password. ( Must be 1 to 30 characters in length)

5. Confirm the new admin password.

6. Press Apply at the bottom of the page to save your new settings.

Figure 5: Admin Level Password

Changing User Level Password

1. Access your HT813 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 End User Password and 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 6: User Level Password

Changing Viewer Password

1. Access your HT813 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 7: Viewer Level Password

Changing HTTP/HTTPS Web Port

1. Access your HT813 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 🡪 HTTP(S) Web Port.

5. Make sure that the Web Access Mode is set to HTTP(S).

6. Change the current port to your new HTTP(S) port. Ports accepted are in range [1-65535].

7. Press Apply at the bottom of the page to save your new settings

Figure 8: Web HTTP(S) Port

Web Configuration Pages Definitions

This section describes the options in the HT813 Web UI.

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, SNMP, syslog, audio
settings, call settings and call progress tones.

FXS PORT: Configures SIP accounts settings.

FXO PORT: Configures SIP accounts settings.

Status Page Definitions

 Status

Shows devise 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.
Note: The device has two MAC addresses, one for the WAN port and one for the LAN port. The MAC address located on the
MAC Address
bottom panel of the device is the MAC address of the LAN port. The MAC address of the WAN port is the MAC address of LAN
port +1.
Example: MAC Address: WAN - “00:0B:82:25:AF:32”, LAN - “00:0B:82:25:AF:31”.

WAN IPv4
Displays assigned IPv4 address.
Address

WAN IPv6
Displays assigned IPv6 address.
Address

Product Model Displays product model info. Default is HT813.

Hardware
Displays the hardware revision information and the part number.
Version

● Bootloader: Specifies Boot version.

● Core: Specifies Core version.
Software ● Base: Specifies Base version.
Version ● Prog: Specifies Program version. This is the main firmware release number, which is always used for identifying the software
system of the HT813.
● CPE: Specifies CPE version. The CPE version is displayed only when HT813 is connected to an ACS using the TR-069 protocol.

Software Status Indicates the current software status of the HT (Running or Stopped).

System Up Time Indicates actual system time and uptime since last reboot.

PPPoE Link Up Indicates PPPoE connection status.

NAT Indicates type of NAT when it is configured.

Individual
Certificate Indicates the current individual Certificate Generation.
Generation

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

Port Options Displays relevant information regarding the FXS and FXO ports about their DND and call forward features.

Provision Displays provisioning status.

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

Basic Settings Page Definitions

Basic Settings

New End Configures user level password.

User Case sensitive and max. Length of 25 characters.
Password

Confirm End
User Re-enter the end user password to confirm the change of user password on the web GUI to avoid typos or mistakes.
Password

New Viewer Configures viewer level password.

Password Case sensitive and max. Length of 25 characters

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

Web/SSH Access

Web Session Configure timer to log out web session during idle.
Timeout The default is 10 min. The range is 2-60 min.

Web Access
Configure attempt limit before lockout.
Attempt
Default is 5. Range is 1-10.
Limit

Lockout If the login attempt failed 5 times, the login would be locked out for the time length.
Time Interval Default 15 mins. Range 1-60 min.

Web Access Allows users to choose the Web Access Mode between HTTPS and HTTP. If HTTPS is selected, web UI will be accessed using HTTPS.
Mode Default is HTTP.

HTTP Web
Customizes HTTP port used to access the HT813 web UI. Default is80.
Port

HTTPS Web
Customizes HTTPS port used to access the HT813 web UI. Default is443.
Port

Disable SSH Enables/disables SSH access. The default is No (disabled).

SSH Port Allows users to self-configure SSH Port number. By default, the port number is 22.

Disable�Tel
Enables/disables the Telnet access. The default is Yes (disabled).
net

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

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

WAN Side 1. No: No access to the web or SSH from any IP address on the WAN side.
Web/SSH 2. Yes: Access for the Web GUI and SSH is enabled on the WAN side.
Access 3. Auto:Only private IP could access the web or SSH on the WAN side.

Default setting is Auto.

If WAN Side Web/SSH Access is set to Yes or Auto. Users can configure the white List for WAN Side to be used for remote
White List management.
for WAN Multiple IPs are supported and need to be separated by space.
Side Example:192.168.5.222 192.168.5.223 192.168.7.0/24
Note: If both blacklist and whitelist are not empty, the blacklist is processed first, followed by the whitelist.

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

Selects one of the following IP protocol modes:

1. IPv4 Only:Enforce IPv4 protocol only.

Internet 2. IPv6 Only:Enforce IPv6 protocol only.
Protocol 3. Both, Prefer IPv4:Enable both IPv4 and IPv6 and prefer IPv4.
4. Both, prefer IPv6:Enable both IPv4 and IPv6 and prefer IPv6.

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

IPv4 Allows users to configure the appropriate network settings on the HT813 to obtain IPv4 address. Users could select DHCP, Static IP or
Address PPPoE. By default, it is set to DHCP.

All the field values for the static IP mode are not used (even though they are still saved in the flash memory.) The ATA 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
Dynamically
optional but may be required by ISP.
assigned via
DHCP domain name: allows user to configure DHCP domain name. This option specifies the domain name that the client should use
DHCP
when resolving hostnames via the Domain Name System. This field is optional.
DHCP vendor class ID: Exchanges vendor class ID by clients and servers to convey particular configuration or other identification
information about a client. Default isHT8XX.

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

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

Allows users to configure the appropriate network settings on the HT813 to obtain an IPv6 address. Users could select DHCP, or 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 ATA
IPv6
acquires its IP address from the first DHCP server it discovers from the LAN it is connected.
Address
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).

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

Self-Defined
Allows users to define their own time zone.
Time Zone

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

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

NAT/DHCP Server Information & Configuration

 Controls whether the device is working inNAT router, Bridge, or WAN Only mode.
NAT Router: In this mode, the WAN port acts as a DHCP client. LAN port is used as DHCP Base IP; devices connected behind the LAN
port will be assigned an IP from HT813 DHCP Server.
Bridge: In this mode, the WAN port acts as a DHCP client and pass-through to the LAN port; devices connected behind the LAN port
Device Mode
will get an IP from your network DHCP server (same as the WAN port).
WAN Only: In this mode, only the WAN port is active. LAN port is not used.
The default mode is NAT Router.
Save the setting and reboot prior to configuring the HT813.

NAT
Defines the number of ports that can be managed while in NAT router mode.
Maximum
Range: 0 4096, default is 1024. Typically, one port per connection
Ports

NAT TCP
NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 3600
Timeout

NAT UDP NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 3600, default is
Timeout 300

Specifies the maximum uplink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set
Uplink as: 128K, 256K, 512K, 1M, 2M, 3M, 4M, 5M, 10M or 15M. The primary function of this setting is to limit the uplink bandwidth for the
Bandwidth device internal system, signaling and NATed traffic. Example: When 512k is configured, there will be at least 512kbps limited for
internal system, signaling and NATed traffic. Voice or RTP stream will never be limited.

Specifies the maximum downlink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be
Downlink set as: 128K, 256K, 512K, 1M, 2M, 3M, 4M, 5M, 10M or 15M. The primary function of this setting is to limit the download bandwidth for
Bandwidth the device internal system, signaling and NATed traffic. Example: if 128 is configured, there will be at least 128kbps limited for internal
system, signaling and NATed traffic. Voice or RTP stream will never be limited.

Enable UPnP When set to Yes, the HT813 acts as a UPnP gateway for your UPnP-enabled applications.UPnP = Universal Plug and Play. The default
Support is No.

Reply to
When set to Yes, the HT813 responds to the PING command from other computers but is also made vulnerable to DOS attacks. The
ICMP on
default is No.
WAN Port

Cloned WAN
MAC This allows the user to change/set a specific MAC address on the WAN interface. Note: Set in Hex format.
Address

LAN Port
This feature allows users to configure a different VLAN tag and priority value for the second network port when HT is configured in
VLAN
bridge mode.
Feature
The priority value range is 0-7, The VLAN tag range is 0-4094.
Under Bridge
The default VLAN Tag and Priority value are 0.
Mode

Enable LAN When set to Yes, the device will function as a simple router and the LAN port will provide IP addresses to the internal network.
DHCP Connect the WAN port to ADSL/Cable modem or any other equipment that provides access to the public Internet

Base IP Address for a LAN port. The default factory setting is 192.168.2.1. Note: When the device detects WAN IP is conflicting with
LAN DHCP
LAN IP, the LAN base IP address will be changed based on the network mask the effective subnet will be increased by 1.
Base IP
For example; 192.168.2.1 will be changed to 192.168.3.1 if the net mask is 255.255.255.0. Then the device will reboot

LAN DHCP The default value is 100. The last segment of IP address is assigned to the HT813 in the LAN Network.
Start IP Default configuration assigns IP address (to local network devices) starting from 192.168.2.100.

LAN DHCP
Default value is 199. This parameter allows a user to limit the number of local network devices connected to the internal router.
End IP
 LAN Subnet Sets the LAN subnet mask.
Mask Default value is 255.255.255.0

DHCP IP
Default value is 120 hrs. (5 days). The length of time the IP address is assigned to the LAN clients. Value is set in units of hours.
Lease Time

This function forwards all WAN IP traffic to a specific IP address if no matching port is used by HT813 or in the defined port
DMZ IP
forwarding.

Port
Forwards a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port. Up to 8 rules are available.
Forwarding

Gives the administrator the option to restore the default configuration on the HT813.
There are 3 types of factory reset:

1. ISP Data Reset: All ISP (Internet Service Provider) configuration which may affect the IP address will be reseted (including WAN
Reset Type static IP).
2. VoIP Data Reset: All VoIP related configuration (mainly everything located on FXS page).
3. Full Reset: Both VoIP and ISP-related configuration at the same time.

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

PSTN
Key pattern to use PSTN line. Maximum 5 digits. The default is *00
Access Code

PIN for VoIP-

to-PSTN Maximum 8 digits to authorize calling PSTN numbers from VoIP.
Calls

PIN for
PSTN-to- Maximum 8 digits to authorize calling VoIP terminals from PSTN.
VoIP Calls

Unconditiona
l Call VoIP calls will be forwarded to the specified PSTN number.
Forward to Specify PSTN number.
PSTN

Unconditiona
l Call Incoming PSTN calls will be forwarded to the VoIP number.
Forward to Specify User ID, SIP Server and SIP Destination Port
VoIP

Advanced Settings Page Definitions

Advanced Settings

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
New Admin Password
blank for security reasons after clicking update and saved.
password length is 1 to 30 characters.

Confirm Admin Password Re-enter the admin password to confirm change admin password on web GUI to avoid typo or mistakes.

Disable User Level Web Access Disables User Level Web Acces, this option is enabled by default.

Disable Viewer Level Web Access Disables Viewer Level Web Access, this option is enabled by default.
 Sets values for:

Layer 2 QoS ● 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.

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

Black List for WAN Side Port
Example: “5000-6000, 7000 “.

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

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

Uses STUN keep-alive to detect WAN side network problems. If the keep-alive request does not yield any
Use STUN to detect network response for the configured number of times (minimum 3), the device will restart the TCP/IP stack. If the
connectivity STUN server does not respond when the device boots up, the feature is disabled.
The default setting is No.

Use DNS to detect network

Uses DNS to detect WAN side network problems. Default setting is No.
connectivity

Verify host when using HTTPS Enables / disables the host verification when using HTTPS.

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

Upgrade via

Sets IP address or domain name of firmware server. The URL of the server that hosts the firmware release.
Firmware Server Path
The default is fm.grandstream.com/gs.

Sets the IP address or domain name of the configuration server. The server hosts a copy of the configuration
file to be installed on the HT813.
Config Server Path Note: Starting from firmware 1.0.17.2 , you can specify the protocol used in the web request. (example:
https://192.168.5.120)
The Default is fm.grandstream.com/gs.

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

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

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

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

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

Checks if configuration files are with matching prefix before downloading them. It allows user to store
Config File Prefix
different configuration files in one directory on the provisioning server.

Checks if configuration files are with matching postfix before downloading them. It allows user to store
Config File Postfix
different configuration files in one directory on the provisioning server.

Enable Using tags in URL Allow users to configure variables on the configuration server path to differentiate the
directories on the server.
 Example: When provisioning, a user can define the mac address and IP address when sending the HTTP Send
request link in the following form "192.168.5.96:8060/?mac=[MAC]&lan_ip=[IP]", the link will look like this
example : http://192.168.5.99/mac=000b89a9064&lan_ip=192.168.5.99/cfg.xml
Default Value is "No".

Determines whether to send basic HTTP authentication information to the server by default when using a
"Wget" request to download firmware or configuration files. If set to "Yes", it will send HTTP/HTTPS user name
Always send HTTP Basic
and password no matter whether the server needs authentication or not. If set to "No", only send HTTP/HTTPS
Authentication Information
user name and password when the server needs authentication.
Set to "No" by Default.

Obtains configuration and upgrade server’s information using options 66 from DHCP server.
Allow DHCP Option 66 or 160 to
Note: If DHCP Option 66 is enabled, the HT813 will attempt downloading the firmware file from the server URL
Override the Server
provided by DHCP, even though Config Server Path is left blank

Allows users to enable the Additional Override DHCP Option in Option 150.
Additional Override DHCP Option
The default value is "None"

Sends multicast “SUBSCRIBE” message for provisioning at booting stage, used for PnP (Plug-and-Play)
3CX Auto Provision
configuration. Default is Yes.

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

● No: The HT813 will only do upgrade once at boot up.

Automatic Upgrade ● Check every X minutes: User needs to specify a period in minutes.
● Check every week: User needs to specify “Day of the week (0-6)”. (Day of week is starting from Sunday).
● Check every day: User needs to specify the start hour and the end hour of the day (0-23).

Default is No.

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

Configures the HT813 to always search for the new firmware at boot up. During the boot stage, the HT813 will
Always Check for New Firmware at
contact the firmware upgrade server to search for a new firmware, when available it will start the upgrade
Boot up
process, otherwise it will boot normally.

Check New Firmware only when F/W Configure the HT813 to search for the new firmware when the firmware prefix / suffix changes. When this
pre/suffix changes option is selected, the HT813 will check for updates only when the pre/suffix has been changed.

Configures the HT813 to skip the firmware check when this option is selected the HT813 will always skip
Always Skip the Firmware Check
searching for a new firmware.

allows users to configure provision configuration file type in xml file only or all file types.
Configuration File Types Allowed
Default value is "All"

By default, 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
HT813 will inverse the downloading process to cfg.xml > cfgGSC3570.xml > cfgMAC.bin > cfgMAC.xml. The
Download and Process All Available
following files will override the files that has already been loaded and processed.
Config Files
The default value is "No"
Note: Support for the new override config file option as “cfgMAC_override.xml” file has been added to the
HT813 Model.

Disables the SIP NOTIFY Authentication on the ATA adapter. If set to “Yes”, the ATA adapter will not challenge
Disable SIP NOTIFY Authentication
NOTIFY with 401. Default is No

Authenticates configuration before being accepted. This protects the configuration from unauthorized
Authenticate Conf File
modifications. Default is No.
 This feature allows users to validate server certificates with our trusted list of TLS connections. Default is
Validate Server Certificates enabled.
The device needs to reboot after changing the setting.

Trusted CA Certificates A Configures the entry of the first Trusted CA certificate

Trusted CA Certificates B Configures the entry of the second Trusted CA certificate.

This feature allows users to specify which CA certificate to trust when performing server authentication.
Available settings: Built-in trusted certificates, Custom trusted certificates and All trusted certificates.
Load CA Certificates The default is Built-in trusted certificates.
Note: “Let’s encrypt” root CA certificate has been updated on the firmware release 1.0.15.7
Note: Sectigo CA and Charter CA are some examples of Trusted CA Certficates.

Specifies SSL certificate used for SIP over TLS is in X.509 format. The HT813 has built-in private key and SSL
SIP TLS Certificate
certificate.

SIP TLS Private Key Specifies TLS private key used for SIP over TLS is in X.509 format. Maximum supported length 4096.

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

Custom Certificate Allows users to update to the device their own certificate signed by custom CA certificate to manage client
(Private Key + Certificate) authentication.

Sets the ATA adapter system to enable the “CPE WAN Management Protocol” (TR-069). Default setting is No.
Enable TR-069 Note: Starting from firmware version 1.0.17.2, some TR data models including
“Device.DeviceInfo.SupportedDataModel”were added.

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

ACS Username Enters username to authenticate to ACS.

ACS Password Enters password to authenticate to ACS.

Sends periodic "inform" packets to ACS.

Periodic Inform Enable
Default is No

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

Connection Request Username Enters username for ACS to connect to the HT813.

Connection Request Password Enters password for ACS to connect to the HT813.

Configures the TR-069 Connection Request Port.

Connection Request Port
The value range is 0 to 65535. Default is 7547

CPE SSL Certificate Configures the Cert File for the ATA to connect to the ACS via SSL.

CPE SSL Private Key Specifies the Cert Key for the ATA to connect to the ACS via SSL.

Enables the SNMP Service.

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.

Port of Trap port Port of Trap destination (Default 162)

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

SNMP Trap Interval Time interval between traps (Default is 5).

SNMPv1/v2c Community Name of SNMPv1/v2c community.

SNMPv1/v2c Trap Community Name of SNMPv1/v2c trap community.

SNMPv3 User Name User name for SNMPv3.

SNMPv3 Security Level ● noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.
● privUser : Users with security level authPriv and context name as priv.List Item 2

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 User name for SNMPv3 Trap.

SNMPv3 Trap Security Level noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

SNMPv3 Trap Authentication

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

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

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

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.

Configure RADIUS VSA Vendor ID to match RADIUS server’s configuration. Default is 42397 for Grandstream
RADIUS VSA Vendor ID
Networks Inc.
 Configure RADIUS VSA Access Level Attribute to match RADIUS server’s configuration. Incorrect setting would
RADIUS VSA Access Level Attribute
cause Radius authenticate fail.

Enable DDNS Allow users to use DDNS.

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

DDNS Username Enter DDNS Username. 64 characters as Max String Length.

DDNS Password Enter DDNS Password. 64 characters as Max String Length.

DDNS Hostname Enter DDNS Hostname. 64 characters as Max String Length.

DDNS Hash Enter DDNS Hash. 64 characters as Max String Length.

The configuration option is to set the ring cadence on the FXS port for all incoming calls.
System Ring Cadence Syntax: c=on1/off1-on2/off2-on3/off3; (3 cadences maximum)
Default is set to c=2000/4000; (US standards)

Using these settings, users can configure tone frequencies and cadence according to their preference. By
Call Progress Tones:
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
Dial Tone 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.

Ringback Tone Example configuration for N.A.

Busy Tone Dialtone:

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

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

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

Prompt Tone

Prompt Tone Access Code Key pattern to get Prompt Tone. Maximum 20 digits.

Configuration update via keypad (analog phone connected to FXS port keypad using IVR menu) is disabled if
Lock Keypad Update
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.

The lifeline feature ensures users can place/receive a PSTN call in an emergency situation. Three modes are
supported:

● Auto: In case of power loss or loss of SIP registration, the PSTN line will be seamlessly connected to
Life Line Mode analog phone connected to FXS port.
● Always Connected: PSTN line will be always connected to the phone connected to FXS port. VoIP calls will
not be allowed in this configuration.
● Always Disconnected: User can only make/receive VoIP calls. PSTN calls will not be possible. Default
setting is Auto.
 Allow users to block incoming calls from a specific list of numbers.
Blacklist for Incoming Calls Maximum allowed 10 SIP numbers and each number should be separated by a comma (‘,’) in the web UI.
Other allowed characters are 0-9, comma (“,”), asterisk (‘*’), pound sign (‘#’) and plus sign (‘+’).

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

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

This option contains vendor-specific option data, much like DHCPv4 option 43. There is an extra difference in
that in DHCPv6, this option carries a vendor ID as well, which allows for data from multiple vendors to be
DHCP Option 17 Enterprise Number
provided to the device.
Default is 3561.

This feature allows users to disable weak ciphers. The given choices are:

1. Enable Weak TLS Ciphers Suites.

2. Disable Symmetric Encryption RC4/DES/3DES.
3. Disable Symmetric Encryption SEED.
Disable Weak TLS Cipher Suites 4. Disable All Of The Above Weak Symmetric Encryption.
5. Disable Symmetric Authentication MD5.
6. Disable All Of The Above Weak Symmetric Authentication.
7. Disable Protocol Version SSLv2/SSLv3.

Default is Enable Weak TLS Ciphers Suites.

The Feature allows users to choose the Minimum TLS Version. Choices are:

1. Unlimited.
Minimum TLS Version 2. TLS 1.0
3. TLS 1.1
4. TLS 1.2

The Feature allows users to choose the Maximum TLS Version. Choices are:

1. Unlimited.
Maximum TLS Version 2. TLS 1.0
3. TLS 1.1
4. TLS 1.2

Default is Unlimited.

This feature allows users to customize the Syslog Protocol. The Syslog protocol can be either UDP or
Syslog Protocol
SSL/TLS. The default is UDP.

Syslog Server URL or IP address of syslog server.

Syslog Level Select HT813 to report the log level. The 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)
11. Extra syslog style (EXTRA DEBUG level)
 Example: In the process of configuring a Syslog server, the steps are to define Syslog protocol and set it to
UDP or SSL/TLS, enter the IP Address of your host machine define the Syslog level based on the information
required (DEBUG, INFO, WARNING...),

Configures whether the SIP log will be included in the Syslog messages.
Send SIP Log
The default setting is No.

Allows the device to make a packet capture, by clicking the capture button, when that is set, the user can
define the following:

1. With Secret Key information: Allows users to make packet capture including the secret key to decrypt the
captured TLS packets., set to "No" By Default
2. Status: Set to "Idle" when the packet capture is not started and to "Running" when packet capture is
Information Capture enabled.
3. Capture file: stores the registered Captured file and make it ready for download.

1. Status: Set to "Idle" when the packet capture is not started and to "Running" when packet capture is
enabled.
2. Capture file: stores the registered Captured file and make it ready for download.

Default is No. When “Yes, reboot every day at hour” or “Yes, reboot every week at day” or “Yes, reboot every
Automatic Reboot 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.

Allows user to download and save a text file containing all the P values of each setting as configured at that
Download Device Configuration
point on the unit. For Security Reasons, Passwords will not be Downloaded.

Allows user to download and save an XML file containing all the P values of each setting as configured at that
Download Device XML Configuration
point on the unit. For Security Reasons, Passwords will not be Downloaded.

Allows the user to upgrade the firmware with a single firmware file by browsing and loading the file from your
Upload Firmware
computer (local directory).

Upload Configuration Allows to upload configuration file to the device.

Download backup file to local computer.

Export Backup Configuration
The backup file is XML and encrypted.

Restore From Backup Configuration Uploads the backup file to the ATA to restore your saved configuration

FXS Port Page Definitions

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

Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used to
Primary SIP Server
send/receive SIP messages from/to HT813.

Specifies failover SIP server IP address or domain name provided by VoIP service provider.
Failover SIP Server
This server will be used if the primary SIP server becomes unavailable.

Selects to prefer primary SIP server. The account will register to primary Server if registration with Failover server expires.
Prefer Primary SIP
Server
Default is No.
 Specifies IP address or domain name of outbound Proxy, or media gateway, or session border controller. Used by HT813
Outbound Proxy 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 Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. By default, this field is
Proxy left empty.

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

Allow DHCP Option Configures the HT813 to collect SIP server address from DHCP option 120.
120 (override SIP
Server) Default is No.

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

SIP URI Scheme

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

Use Actual
Controls the port information in the Via header and Contact header. If set to “No”, these port numbers will use the
Ephemeral Port in
permanent listening port on the phone. Otherwise, it will use the ephemeral port for the connection. The default setting
Contact with
is “No”.
TCP/TLS

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

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

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

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

Name Chooses a name to be associated to user.

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

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

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

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

When this option is set to “Yes”, when the HT is registered on second SRV and makes an outbound call, it will try the
DNS SRV use
second SRV (registered IP) first. By default, this option is disabled and the DNS SRV will use first SRV instead of the
Registered IP
registered IP.
 Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the HT813
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
Tel URI 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 HT813 needs to send REGISTER messages to the proxy server. Default setting is Yes.

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration is
Unregister on
performed by sending a REGISTER message with Contact set to * and Expires=0 parameters to the SIP server. This will
Reboot
unregister the SIP account under the concerned FXS page. Default is No.

Outgoing Call
Enables the ability to place outgoing calls even if the account is not registered (if allowed by ITSP); device will not be able
Without
to receive incoming calls. Default is No.
Registration

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

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

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

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

Enable SIP OPTIONS Enables SIP OPTIONS to track account registration status so the phone adapter will send periodic OPTIONS message to
Keep Alive server to track the connection status with the server. Default setting is No.

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

SIP OPTIONS Keep

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

Defines Diff-Serv values for SIP and RTP. Defaults are:

Layer 3 QoS SIP DSCP: 26

RTP DSCP: 46

Local SIP Port Defines local port to use by the HT813 for listening and transmitting SIP packets. Default value for FXS is 5060.

Defines the local RTP-RTCP port pair the HT813 will listen and transmit. It is the HT813 RTP port for channel 0. The default
Local RTP Port
value for FXS port is 5004.
Use Random SIP Controls whether to use configured or random SIP ports. This is usually necessary when multiple HT813 are behind the
Port same NAT. Default is No.

Use Random RTP Controls whether to use configured or random RTP ports. This is usually necessary when multiple HT813 are behind the
Port same NAT. Default is No.

Enable RTCP Allows users to enable RTCP. Default setting is Yes.

Hold Target Before

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

Refer-To Use Target

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

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

Disable Bellcore
Gives the users the possibility of making conference calls by pressing “Flash” key, when it is enabled by dialing *23
Style 3-Way
+second callee number. Default is No
Conference

Remove OBP from

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

Support SIP Instance Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft.
ID Default is No.

Validate Incoming
Validates incoming messages. Default is No.
SIP Messages

Check SIP User ID Checks SIP User ID in the Request URI of incoming INVITE; if it does not match the HT813 SIP User ID, the call will be
for Incoming INVITE rejected. Direct IP calling will also be disabled. Default is No.

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

Authenticate server Configures whether to validate the domain certificate when download the firmware/config file. If it is set to “Yes”, the
certificate domain phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

Authenticate server Configures whether to validate the server certificate when download the firmware/config file. If it is set to “Yes”, the
certificate chain phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

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

Allow Incoming SIP

Checks SIP address of the Request URI in the incoming SIP message; if it does not match the SIP server address of the
Messages from SIP
account, the call will be rejected. Default is No.
Proxy Only

Determines if the “Privacy header” will be presented in the SIP INVITE message and if it includes the caller info in this
Use Privacy Header
header. If set to Default, it will add Privacy header unless special feature is Telkom SA or CBCOM. Default is Default.
 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
Use P-Preferred-
P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted.
Identity Header

Default setting is: Default.

Use P-Access-
With this feature enabled, device will populate the WAN access node with IEEE802.11a, IEEE-802.11b in P-Access-
Network-Info
Network-Info SIP header.
Header

Use P-Emergency-
This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for emergency calls.
Info Header

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

Selects the Caller ID display order which need to be respected by the HT813. The available options are:

Auto: When set to “Auto”, the HT813 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.
Caller ID Fetch Order
Disabled: When set to “Disabled”, all incoming calls are displayed

with “Unavailable”.

From Header: When set to “From Header”, the HT813 will use the FROM header to display the caller ID.

Allow SIP Factory

This feature allows user to reset the devices directly through SIP Notify. Default is No.
Reset

Defines T1 timeout value.

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

SIP T1 Timeout For example, the HT813 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.

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling
SIP T2 Interval
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 0

DTMF Payload Type Defines payload type for DTMF using RFC2833.

Preferred DTMF Sorts DTMF methods (in-audio, via RTP (RFC2833) or via SIP INFO) by priority.
method (in order)
You can configure up to three priorities.

Allows users to config the Inband DTMF Duration and Inter-Duration.

Inband DTMF
The inband DTMF Duration range varies from 40-2000ms with 100ms as A Default value.
Duration
The inband DTMF inter-Duration varies from 40-2000ms with 50ms as A Default value.

Allows users to configure the DSP DTMF Detector Duration and Inter-Duration Threshold.
DSP DTMF Detector
The DSP DTMF Detector duration threshold varies from 20-200ms with 30ms as a Default Value.
Duration Threshold
The DSP DTMF Detector inter-duration threshold varies from 20-200ms with 30ms as a Default Value.
 Uses above DTMF order without negotiation.
Disable DTMF
Negotiation
Default is No.

Generate
Continuous RFC2833 When enabled the RFC2833 events are generated until key is released. Default is No.
Events

Send Hook Flash

Default is No. If set to yes, flash will be sent as DTMF event.
Event

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

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

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

Enables do not disturb, call forward and other call features via the local feature codes on the base. Otherwise, ITSP
feature codes can be used.
Enable Call Features

Default is Yes.

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

Off Hook Auto Dial

Specifies the auto-dial delay after off hook.
Delay

Proxy-Require Determines a SIP Extension to notify the SIP server that the HT813 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.

Use SIP User Agent

Configures the SIP User-Agent Header.
Header

Configures SIP User-Agent. If not configured, device will use the default User Agent Header. The value range is 1024 to
SIP User-Agent
Maximum String Length. Default value is Null.

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

Disable Call Waiting
Default is No.

Disable Call Waiting

Disables displaying caller ID when receiving a second incoming call. Default is No.
Caller ID

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

Disable Connected
Disables displaying the number of the person answering the phone. Default is No.
Line ID

Disable Receiver Off Enables / disables the warning to alert that the phone has been left off-hook for an extended period of time. Default is
Hook Tone No.
Disable Reminder
Ring for On-Hold Enables playing the reminder ring. Default is No
Call

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

Do Not Escape ‘#’ as

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

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

Stops ringing when incoming call if not answered within a specific period of time. When set to 0, There will be no ringing
Ring Timeout
timeout. Default is 60 seconds.

Delayed Call
Forwards incoming call if not answered within a specific period of time when delayed call forward is activated locally
Forward Wait
(using *92 code). Default value is 20 seconds.
Timeout

No Key Entry
Initiates the call within this time interval if no additional key entry during dialing stage. Default is 4 seconds.
Timeout

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).

Early Dial 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.

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

1. xx+ – at least 2 digits number;

2. xx – exactly 2 digits number;

3. ^ – exclude;

4. . – wildcard, matches one or more characters

5. [3-5] – any digit of 3, 4, or 5;

6. [147] – any digit 1, 4, or 7;

7. <2=011> – replace digit 2 with 011 when dialing

8. < =1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed

9. | – or
Dial Plan
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+}

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-digit length

Dial Plan
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.

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

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

Anonymous Call
Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Default is No.
Rejection
 Selects Soft switch vendors’ special requirements Example of vendors: Standard, Broadsoft, CBCOM, RNK, Huawei, China
Special Feature
Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS. Default is Standard.

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

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
Session Expiration
the time (in seconds) at which the session is considered timed out if no successful session refresh transaction occurs
beforehand. Default is 180 seconds.

Min-SE Defines Minimum session expiration (in seconds). Default is 90 seconds.

Uses session timer when making outbound calls if remote party supports it.
Caller Request Timer
Default is No.

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

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.
Force Timer

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

Specifies which end will act as refresher for outgoing calls.

UAC Specify UAC: The handy tone acts as the refresher.

Refresher UAS: Callee or proxy server act as the refresher.

Default is Omit.

Specifies which end will act as refresher for incoming calls:

UAS Specify UAS: The handy tone acts as the refresher.

Refresher UAC: Callee or proxy server act as the refresher.

Default is Omit.

Uses INVITE message to refresh the session timer.

Force INVITE
Default is No.

When to Restart Allows users to support to delay posting Media Change Event with this new feature,it can be set to “Immediately” or to
Session After Re- “After replying 200OK”
INVITE received The default value is “Immediately”.

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

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request.
Add Auth Header on
Initial REGISTER
Default is No.

Conference URI Allows users to manually configure the conference URL.

 Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same
Use First Matching
order received in INVITE.
Vocoder in 200OK
SDP
Default is No.

Configures vocoders in a preference list (up to 7 preferred vocoders) that will be included with same order in SDP
Preferred Vocoder
message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, iLBC and OPUS.

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

Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is
G723 Rate
6.3kbps.

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

iLBC Frame Size
Default is 20ms.

Disables OPUS stereo in SDP.

Disable OPUS Stereo
in SDP
Default is No.

Determines payload type for iLBC. Valid range is between 96 and 127.
iLBC Payload type
Default is 97.

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

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

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

Fax Mode Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

Re-INVITE after Fax

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

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

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

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

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

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

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

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

Configures the SRTP Key Length , it can be set to :

1. AES 128&256 bit
SRTP Key Length 2. AES 128 bit
3. AES 256 bit
Default value is AES 128&256 bit
Crypto Life Time Adds crypto life time header to SRTP packets. Default is Yes.

Depends on standard phone type (and location). Available options:

USA 1 (BELLCORE 600 ohms)

USA 2 (BELLCORE 600 ohms + 2.16uF)

AUSTRALIA

CHINA CO

CHINA PBX
SLIC Setting EUROPEAN CTR21

GERMANY

INDIA/NEW ZEALAND

JAPAN CO

JAPAN PBX

STANDARD 900 ohms

UK

Selects the caller ID scheme. Available options:

Bellcore/Telcordia

ETSI-FSK during ringing

ETSI-FSK prior to ringing with DTAS

ETSI-FSK prior to ringing with LR+DTAS

ETSI-FSK prior to ringing with RP

ETSI-DTMF during ringing

ETSI-DTMF prior to ringing with DTAS

Caller ID Scheme ETSI-DTMF prior to ringing with LR+DTAS

ETSI-DTMF prior to ringing with RP

SIN 227 – BT

NTT JAPAN

DTMF Denmark prior to ringing with no DTAS no LR

DTMF Denmark prior to ringing with LR

DTMF Sweden/Finland prior to ringing with LR

DTMF Brazil

DTMF-FSK Brazil

Defines the start and stop tones as delimiters for the caller ID.
DTMF Caller ID
Start Tone and Stop Tone can be set to “Default”, “A”, “B”, “C”, “D” or “#”

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

Loop Current Allows the traditional PBX used with HT813 to apply this method for signaling call termination. Method initiates short
Disconnect voltage drop on the line when remote (VoIP) side disconnects an active call. Default is No.

Play busy/reorder
tone before Loop Allow user to configure if it will play busy/reorder tone before loop current disconnect upon call fail. Default is No.
Current Disconnect
Loop Current Configures the duration of voltage drop described in topic above. HT813 supports a duration range from 100 to 10000
Disconnect Duration ms. Default value is 200.

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

Pulse Dialing This feature allows users to use Swedish pulse dialing standard or New Zealand pulse dialing standard. Default is General
Standard Standard.

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

Defines the time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of
the Flash/Hold and automatic phone ring-back, adjust this time value. HT813 supports a range from 40 to 2000 ms.
Hook Flash Timing

Default values are 300 minimum and 1100 maximum.

Specifies the on-hook time for an on-hook event to be validated. HT813 supports a range from 40 to 2000 ms. Default
On Hook Timing
value is 400.

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.

Gain
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.

Disables the LEC will per call base. Recommended for FAX/Data calls.
Disable Line Echo
Canceller (LEC)
Default is No.

Disables the NEC will per call base. Recommended for FAX/Data calls.
Disable Network
Echo Suppressor
Default is No.

Outgoing Call
Defines the call duration limit for the outgoing calls, Default is 0 (No limit).
Duration Limit

Incoming Call
This feature allows users to configure the call duration limit for the incoming calls, default is 0 (No limit).
Duration Limit

Configures the Ring frequency in herts, it can be set to 20Hz or 25Hz,

Ring Frequency
Default is 20Hz.

Enable High Ring

Configures a high ringing voltage output for the FXS port of HT813.
Power

RFC2833 Events
This feature allows users to customize the count of RFC2833 events. Supported range is 2-10. Default is 8.
Count

RFC2833 End Events

This feature allows users to customize the count of RFC2833 end events. Supported range is 2-10. Default is 3.
Count
 Customizes the Ring Tone 1 to 3 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.

Distinctive Ring
For example: If configured as 617x+, Ring Tone 1 will be used in case of call arrived from the area code 617. Any other
Tone
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

Configures the ring tone cadence preferences. User has 10 choices. The configuration completed in Distinctive Ring
Ring tones
Tones block in the same page, applies to ring tones cadences configured here.

Table 9: FXS Page

FXO Port Page Definitions

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

Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used
Primary SIP Server
to send/receive SIP messages from/to HT813.

Specifies failover SIP server IP address or domain name provided by VoIP service provider. This server will be used if the
Failover SIP Server
primary SIP server becomes unavailable.

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

Specifies IP address or domain name of outbound Proxy, or media gateway, or session border controller. Used by
Outbound Proxy HT813 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 Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. By default, this field is
Proxy left empty.

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

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

When TLS is enabled on the FXO HT813 device, the SIP URI Scheme When Using TLS option allows users to specify the
SIP URI Scheme When type of SIP URI scheme that will be used during the communication. The available options typically include:
Using TLS sip: This is the standard SIP URI scheme that is used for non-secure communication.
sips: This is the secure version of the SIP URI scheme and is used for communication over a TLS encrypted connection.

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

Defines user account information provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone
SIP User ID
number or actually a phone number.
 Determines account authenticate ID provided by VoIP service provider (ITSP). Can be identical to or different from “SIP
Authenticate ID
user ID”.

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

Name Chooses a name to be associated to user.

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

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

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

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

Default is A Record.

When this option is set to “Yes”, when the HT is registered on second SRV and makes an outbound call, it will try the
DNS SRV use second SRV (registered IP) first.
Registered IP
By default, this option is disabled and the DNS SRV will use first SRV instead of the registered IP.

Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the HT813
has an assigned PSTN Number.

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

Tel URI 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 HT813 needs to send REGISTER messages to the proxy server. Default setting is Yes.

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration
Unregister on Reboot is performed by sending a REGISTER message with Contact set to * and Expires=0 parameters to the SIP server. This will
unregister the SIP account under the concerned FXO page. Default is No.

Outgoing Call Enables the ability to place outgoing calls even if the account is not registered (if allowed by ITSP); device will not be
Without Registration able to receive incoming calls. Default is No.

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

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

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

SIP Registration Sends re-register request after specific time (in seconds) when registration process fails with error 403 Forbidden.
Failure Retry Wait Maximum interval is 3600 seconds (1 hour).
Time upon 403
Forbidden Default is 1220 seconds.
Enable SIP OPTIONS Enables SIP OPTIONS to track account registration status so the phone adapter will send periodic OPTIONS message to
Keep Alive server to track the connection status with the server. Default setting is No.

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

SIP OPTIONS Keep

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

Defines Diff-Serv values for SIP and RTP. Defaults are:

Layer 3 QoS SIP DSCP: 26

RTP DSCP: 46

Local SIP Port Defines local port to use by the HT813 for listening and transmitting SIP packets. Default value for FXO port is 5062.

Defines the local RTP-RTCP port pair the HT813 will listen and transmit. It is the HT813 RTP port for channel 0. The
Local RTP Port
default value for FXS port is 5012.

Controls whether to use configured or random SIP ports. This is usually necessary when multiple HT813 are behind the
Use Random SIP Port
same NAT. Default is No.

Controls whether to use configured or random RTP ports. This is usually necessary when multiple HT813 are behind the
Use Random RTP Port
same NAT. Default is No.

Enable RTCP Allows users to enable RTCP. Default setting is Yes.

Removes outbound proxy info in “Route” header when sending SIP packets.
Remove OBP from
Route Header
Default is No.

Support SIP Instance Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft.
ID Default is No.

Validate Incoming SIP

Validates incoming messages. Default is No.
Message

Check SIP User ID for Checks SIP User ID in the Request URI of incoming INVITE; if it does not match the HT813 SIP User ID, the call will be
Incoming INVITE rejected. Direct IP calling will also be disabled. Default is No.

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

Authenticate server Configures whether to validate the domain certificate when download the firmware/config file. If it is set to “Yes”, the
certificate domain phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

Authenticate server Configures whether to validate the server certificate when download the firmware/config file. If it is set to “Yes”, the
certificate chain phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

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

Allow Incoming SIP

Checks SIP address of the Request URI in the incoming SIP message; if it does not match the SIP server address of the
Messages from SIP
account, the call will be rejected. Default is No.
Proxy Only
 Determines if the “Privacy header” will be presented in the SIP INVITE message and if it includes the caller info in this
Use Privacy Header
header. If set to Default, it will add Privacy header unless special feature is Telkom SA or CBCOM. Default is Default.

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
Use P-Preferred-
P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted.
Identity Header

Default setting is: Default.

Use P-Access- With this feature enabled, device will populate the WAN access node with IEEE802.11a, IEEE-802.11b in P-Access-
Network-Info Header Network-Info SIP header.

Use P-Emergency-Info
This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for emergency calls.
Header

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

Allow SIP Factory

This feature allows user to reset the devices directly through SIP Notify. Default is No.
Reset

Defines T1 timeout value.

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

SIP T1 Timeout For example, the HT813 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.

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling
SIP T2 Interval
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 0

DTMF Payload Type Defines payload type for DTMF using RFC2833.

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

You can configure up to 3 Priorities.

Allows users to configure the Inband DTMF Duration and Inter-Duration.

Inband DTMF
The inband DTMF Duration range varies from 40-2000ms with 100ms as A Default value.
Duration
The inband DTMF inter-Duration varies from 40-2000ms with 50ms as A Default value.

Allows users to configure the DSP DTMF Detector Duration and Inter-Duration Threshold.
DSP DTMF Detector
The DSP DTMF Detector duration threshold varies from 20-200ms with 30ms as a Default Value.
Duration Threshold
The DSP DTMF Detector inter-duration threshold varies from 20-200ms with 30ms as a Default Value.

Uses above DTMF order without negotiation.

Disable DTMF
Negotiation
Default is No.

When enabled the RFC2833 events are generated until key is released.
Generate Continuous
RFC2833 Events
Default is No.
 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 Digit Control
“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.

Proxy-Require Determines a SIP Extension to notify the SIP server that the HT813 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.

Use SIP User-Agent

Configures the SIP User-Agent Header.
Header

Configures SIP User-Agent. If not configured, device will use the default User Agent Header. The value range is 1024 to
SIP User-Agent
Maximum String Length. Default value is Null.

Do Not Escape ‘#’ as

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

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

Stops ringing when incoming call if not answered within a specific period of time. Default is 60 seconds. When
Ring Timeout
configure the Ring Timeout to 0, will have no ring timeout.

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
Early Dial 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.

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

1. xx+ – at least 2 digits number;

2. xx – exactly 2 digits number;

3. ^ – exclude;

4. . – wildcard, matches one or more characters

5. [3-5] – any digit of 3, 4, or 5;

6. [147] – any digit 1, 4, or 7;

7. <2=011> – replace digit 2 with 011 when dialing

8. < =1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed

9. | – or

Example 1: {[369]11 | 1617xxxxxxx} –

Allow 311, 611, 911, and any 10-digit numbers of leading digits 1617

Example 2: {^1900x+ | <=1617>xxxxxxx} –

Block any number with leading digits 1900 and add prefix 1617 for any dialed 7-digit numbers
Dial Plan
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+}

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.

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

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

Selects Soft switch vendors’ special requirements Example of vendors: Standard, Broadsoft, CBCOM, RNK, Huawei,
Special Feature
China Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS. Default is Standard.
 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
Session Expiration
the time (in seconds) at which the session is considered timed out if no successful session refresh transaction occurs
beforehand. Default is 180 seconds.

Min-SE Defines Minimum session expiration (in seconds). Default is 90 seconds.

Uses session timer when making outbound calls if remote party supports it.
Caller Request Timer
Default is No.

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

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.
Force Timer

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

Specifies which end will act as refresher for outgoing calls.

UAC: The handy tone acts as the refresher.

UAC Specify Refresher
UAS: Callee or proxy server act as the refresher.

Default is Omit.

Specifies which end will act as refresher for incoming calls:

UAS: The handy tone acts as the refresher.

UAS Specify Refresher
UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE Always refresh with INVITE instead of UPDATE Default is No.

When to Restart Allows users to support to delay posting Media Change Event with this new feature,it can be set to “Immediately” or to
Session After Re- “After replying 200OK”
INVITE received The default value is “Immediately”.

INVITE Ring-No-
Between 5-300 seconds. Default 40 seconds.
Answer Timeout (sec)

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

Add Auth Header on

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

Use First Matching

Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same
Vocoder in 200OK
order received in INVITE. Default is No.
SDP

Configures vocoders in a preference list (up to 7 preferred vocoders) that will be included with same order in SDP
Preferred Vocoder
message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, iLBC and OPUS.

Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for
Voice Frames per TX
G711/G726/G723/other codecs, respectively.
 Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is
G723 Rate
6.3kbps.

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

iLBC Frame Size
Default is 20ms.

Disable OPUS Stereo

Disables OPUS stereo in SDP. Default is No.
in SDP

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.

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

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

Fax Mode Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

Re-Invite after Fax

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

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

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

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

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

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

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

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

Configures the SRTP Key Length , it can be set to :

1. AES 128&256 bit
SRTP Key Length 2. AES 128 bit
3. AES 256 bit
Default value is AES 128&256 bit

Crypto Life Time Adds crypto life time header to SRTP packets. Default is Yes.
 Selects the caller ID scheme. Available options:

Bellcore/Telcordia

ETSI-FSK during ringing

ETSI-FSK prior to ringing with DTAS

ETSI-FSK prior to ringing with LR+DTAS

ETSI-FSK prior to ringing with RP

ETSI-DTMF during ringing

ETSI-DTMF prior to ringing with DTAS

Caller ID Scheme ETSI-DTMF prior to ringing with LR+DTAS

ETSI-DTMF prior to ringing with RP

SIN 227 – BT

NTT JAPAN

DTMF Denmark prior to ringing with no DTAS no LR

DTMF Denmark prior to ringing with LR

DTMF Sweden/Finland prior to ringing with LR

DTMF Brazil

DTMF-FSK Brazil

Defines the start and stop tones as delimiters for the caller ID.
DTMF Caller ID
Start Tone and Stop Tone can be set to “Default”, “A”, “B”, “C”, “D” or “#”

FSK Caller ID
An adjustable value for the Caller ID signal to help this device to recognize Caller ID from different networks. Range:
Minimum RX Level
-96 to -0dB. Default –40dB.
(dB)

FSK Caller ID Seizure

Range is from 0 to 800 bits. Default 70.
Bits

FSK Caller ID Mark

Range is from 0 to 800 bits. Default 40.
Bits

According to customer’s choice CID information will be transferred from PSTN network to VoIP network using following
rules:

Relay via SIP From – PSTN CID is in the SIP From field

Caller ID Transport Relay via P-Asserted-Identity – SIP From field uses the pre-configured account user Id. PSTN CID is in the P-
Type Asserted-Identity field

Relay via P-Preferred-Identity – PSTN CID is in the P-Preferred-Identity field

Send anonymous – SIP From field uses “anonymous”. PSTN CID is put in the P-Asserted-Identity field

Disable – PSTN CID will not be sent. SIP From field uses the pre-configured account user ID

Send Hook Flash to

If Yes, hook flash will be sent to PSTN upon receiving flash event from RFC2833 or SIP INFO.
PSTN

Hook Flash Duration The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust this time value to prevent
(ms) unwanted activation of the Flash/Hold and automatic phone ring-back.
 Voice path volume adjustment.

• RX is a gain level for signals transmitted by FXO (FXO-To-VoIP volume)

• TX is a gain level for signals received by FXO (FXO-To-PSTN volume).

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

Gain User can adjust volume of call on either end using the Rx Gain Level parameter and the TX Gain Level parameter
located on the FXO Port Configuration page. These parameters affect call volume ONLY for calls placed to/from PSTN
and VoIP networks.

If call volume is too low when using VoIP extension, adjust volume using the Rx Gain Level parameter under the FXO
Port Configuration page.

If voice volume is too low at the other end (PSTN side), user may increase the far end volume using the TX Gain Level
parameter under the FXO Port Configuration page.

Disable Line Echo

Disables the LEC will per call base. Recommended for FAX/Data calls. Default is No.
Canceller (LEC)

Disable Network Echo

Disables the NEC will per call base. Recommended for FAX/Data calls. Default is No.
Suppressor

Outgoing Call
Defines the call duration limit for the outgoing calls, Default is 0 (No limit).
Duration Limit

Incoming Call
This feature allows users to configure the call duration limit for the incoming calls, default is 0 (No limit).
Duration Limit

FXO Termination

Enable Current This value should be used in case the PSTN provider uses line power drop to indicate call completion to the end point.
Disconnect In this case the HT813 will search for a power drop.

This is a preconfigured value of duration for a line power drop used by specific service providers. For example, for a
Current Disconnect configured value of 500ms the device will ignore any random voltage drops on the line if duration of such drop is less
Threshold (ms) than 500ms and the call will NOT be considered as terminated. This is useful to prevent unnecessary call drops in some
low quality PSTN lines. Default is 100 ms. Range from 50 to 800 ms.

Enable PSTN
Disconnect Tone If set to Yes, arrived Busy Tone is used as the disconnect signal.
Detection

In certain countries, the central office will send a special busy tone to indicate when a call is disconnected from the
remote side. User can pre-configure this tone on the ATA. The user should know the frequency values and cadences of
these tones.

Here is an example for the syntax for a busy tone in the U.S.A:

PSTN Disconnect Tone Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;

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

Default: Busy Tone – f1=480@-32, f2=620@-32, c=500/500;

Note: Maximum supported cadences are 3

 This should be set to Yes only if the FXO lines are subscribed to PR service from PSTN Service provider. It is merely a PR
detect feature. Default is No.
Enable Polarity
Reversal Note: If there is no PR service from provider on the FXO line, and this setting is

configured to Yes, calls will not be successful.

You can select the AC termination by Country or by Impedance.

AC Termination Model
Default is Country-based.

15 Countries are selectable in this version of the F/W.

USA

AUSTRIA

AUSTRALIA/NEW ZEALAND

BELGIUM

CHINA

FINLAND

FRANCE

Country-Based GERMANY

GREECE

ITALY

JAPAN

NORWAY

SPAIN

SWEDEN

UK

Default is USA

Select the Impedance used by the PSTN service provider.

600R – 600 ohms

600C – 600 ohms + 2.16uF

900R – 900 ohms

900C – 900 ohms + 2.16uF

COMPLEX1 – 220 ohms + (820 ohms || 115nF)

COMPLEX2 – 270 ohms + (750 ohms || 150nF)

COMPLEX3 – 370 ohms + (620 ohms || 310nF)

Impedance-Based
COMPLEX4 – 600R, 270 ohms + (750 ohms || 150nF)

COMPLEX5 – 320 ohms + (1050 ohms || 230nF)

COMPLEX6 – 350 ohms + (1000 ohms || 210nF)

COMPLEX7 – 200 ohms + (680 ohms || 100nF)

COMPLEX8 – 370 ohms + (820 ohms || 110nF)

COMPLEX9 – 275 ohms + (780 ohms || 115nF)

COMPLEX10 – 120 ohms + (820 ohms || 110nF)

Default is 600R – 600 ohms

 The FXO port will ring the number of times configured in this field before sending the call to the VoIP side. The
Number of Rings
supported range is 1-50. Default is 4.

If Yes, the phone connected to the FXS port will ring a configured number of times (see above). If not, the phone
PSTN Ring Thru FXS
connected to the FXS port will not ring.

PSTN Ring Thru Delay If the PSTN Ring Thru Delay is set to Yes, all incoming PSTN calls through FXO will ring the phone connected to the FXS
(sec) port, after this delay or after caller id is detected (whichever comes first).

PSTN Ring Timeout

Range is 2-10 seconds. Default is 6 seconds. Option is used to detect PSTN hang up when FXO port is not answered.
(sec)

PSTN Idle Wait

Timeout between Used to customize timeout value between PSTN outgoing calls. Range is 0-10 seconds. Default is 4 seconds.
Outgoing Calls

Channel Dialing

Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out digits for VoIP to PSTN 1 stage
calls, and unconditional call forward to PSTN, and route to PSTN. Digit Length is the play time for each digit.
DTMF Digit Length
(ms)
Note: In order to receive the caller ID information, the delay should be set to a value larger than the delay required to
complete the PSTN caller ID delivery.

DTMF Dial Pause (ms) Dial pause is the time between 2 digits for the same scenario as explained above.

First Digit Timeout Used for PSTN to VoIP calls. PSTN users need to enter the FIRST digit within the first digit timeout period. Otherwise
(sec) the call will be dropped.

Inter-Digit Timeout
When dialing from the PSTN to VoIP, subsequent digits must be input within the period of inter-digit timeout.
Otherwise the dial plan thinks it is the end of the digit input.
(sec)

Wait for Dial tone is used for one stage VoIP to PSTN calls. If set to Yes, the device will first obtain a PSTN line and a dial
Wait for Dial Tone
tone from a central office. After obtaining the dial tone, the digits dialed will be sent to the central office.

Stage Method (1/2) This configuration is applicable for VoIP to PSTN calls and indicates one or two stage dialing methods.

The time to wait before HT813 initiates the call via PSTN line.
Min Delay Before Dial
PSTN Number
Default 500ms, range is from 50 to 65000ms.

Table 10: FXO Page

Important Settings

NAT Settings

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

1. STUN Server (under advanced settings webpage) enter a STUN server IP (or FQDN) that you may have or look up a free public STUN server on
the internet and enter it on this field. If using public IP, keep this field blank.

2. Use random SIP/RTP ports (under advanced settings webpage), this setting depends on your network settings. Generally, if you have multiple IP
devices under the same network, it should be set to Yes. If using a public IP address, set this parameter to No.

3. NAT traversal (under the FXS and FXO web page) Set this to Yes when gateway is behind firewall on a private network.
DTMF Methods

The HT813 supports the following DTMF mode:

• DTMF in-audio

• DTMF via RTP (RFC2833)

• DTMF via SIP INFO

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

Preferred Vocoder (Codec)

The HT813 supports following voice codecs. On FXS/FXO page, choose the order of your favorite codecs:

• PCMU/A (or G711µ/a)

• G729 A/B

• G723.1

• G726

• iLBC

• OPUS

Configuring HT813 Through Voice Prompts

As mentioned previously, The HT813 has a built-in voice prompt menu for simple device configuration. Please refer to “Understanding HT813
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 HT813 to use DHCP.

STATIC IP MODE

Select voice menu option 01 to enable HT813 to use STATIC IP mode, then use option 02, 03, 04, 05 to set up IP address, Subnet Mask, Gateway, and
DNS server, respectively.

PPPOE MODE

Select voice menu option 01 to allow the HT813 to enable the PPPoE mode. PPPoE Username and Password should be configured from web GUI.

FIRMWARE SERVER IP ADDRESS

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

CONFIGURATION SERVER IP ADDRESS

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

UPGRADE PROTOCOL

Select the menu option 15 to choose firmware and configuration upgrade protocol between TFTP, FTP, FTPS, HTTP and HTTPS. 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.
 WAN PORT WEB ACCESS

Select voice menu option 12 to enable/disable web access from WAN port. Press 9 in this menu to toggle between enable / disable. Default is
disabled.

Configuration through a Central Server

The HT813 can be automatically configured from a central provisioning system.

When HT813 boots up, it will send TFTP, FTP/FTPS or HTTP/HTTPS requests to download configuration files, “cfg000b82xxxxxx” and
“cfg00082xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the HT813. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the
provision program will issue request a generic configuration file “cfg.xml”. Configuration file name should be in lower case letters. The configuration
data can be downloaded via TFTP, FTP/FTPS or HTTP/HTTPS from the central server. A service provider or an enterprise with large deployment of
HT813 can easily manage the configuration and service provisioning of individual devices remotely from a central server.

Grandstream provides a central provisioning system GAPS (Grandstream Automated Provisioning System) to support automated configuration of
Grandstream devices. GAPS use enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate
with each individual Grandstream device for firmware upgrade, remote reboot, etc. Grandstream provides GAPS service to VoIP service providers.
Use GAPS for either simple redirection or with certain special provisioning settings. At boot-up, Grandstream devices by default point to
Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so
that it will be redirected to customer’s TFTP or HTTP/HTTPS server for further provisioning. Grandstream also provides configuration tools (Windows
and Linux/Unix version) to facilitate the task of generating device configuration files.

The Grandstream configuration tools are free to end users. The configuration tools and configuration templates are available for download from
https://www.grandstream.com/support/tools

Register a SIP Account

The HT813 supports 2 SIP accounts. Please refer to the following steps in order to register your accounts via web user interface

1. Access your HT813 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 (same steps for FXO) web pages and set the following:
1. Account Active to Yes.

2. Primary SIP Server field with your SIP server IP address or FQDN.

3. Failover SIP Server with your Failover SIP Server IP address or FQDN. Leave empty if not available.

4. Prefer Primary SIP Server to No or Yes depending on your configuration. Set to No if no Failover SIP Server is defined. If “Yes”, account
will register to Primary SIP Server when failover registration expires.

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 digit similar to phone number or
actually a phone number.

7. Authenticate ID: SIP service subscriber’s Authenticate ID 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 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.

5. Press Apply at the bottom of the page to save your configuration.

 Figure 9: FXS Port 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
from your HT813 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 10: Accounts Status

Rebooting HT813 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.
Call Features

The HT813 supports all the traditional and advanced telephony features.

Key Call Features

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

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

*03
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

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

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

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

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

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

*87 Blind Transfer

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

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

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

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

Flash
Toggles between active call and incoming call (call waiting tone). If not in conversation, flash/hook will switch to a new channel for a new
/
call.
Hook

# Pressing pound sign will serve as Re-Dial key.

Table 11: HT813 Call Features

Information Capture

Information Capture involves intercepting a secret key information file during the TLS handshake between the HT8xx and a SIP server for the
purpose of extracting the secret key information file during the TLS handshake, the file downloaded is a .txt file including the client secret key.

The process includes initiating capture, re-registering the HT8xx with TLS to a SIP server, waiting for a SIP Register request, and then stopping the
capture. The goal is to extract the secret key information file, to do that please follow the below steps described below:

Start capturing the communication between the HT8xx and the server. to do that go to Advanced settings => Information capture, make sure
“With Secret Key information” is set to Yes

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

Allow the system to wait until the ATA sends a SIP Register request to the server.

Once the SIP Register request is sent, stop capturing the communication.
 You have the possibility to now download the secret key.

After completing the above steps, the expectation is that the captured data will include the secret key information file in a .txt file under the
name “sslkeylogfile”

The extracted key gives you the possibility to decrypt the secured communication between the HT8xx and the SIP server.

Important

Please use Information capture only when authorized since extracting secret key information without proper authorization is considered unethical.

UPGRADING AND PROVISIONING

The HT813 can be upgraded via TFTP/FTP/FTPS/HTTP/HTTPS by configuring the URL/IP Address for the TFTP/FTP/FTPS/HTTP/HTTPS server and
selecting a download method. Configure a valid URL for TFTP or FTP/FTPS or HTTP/HTTPS (default is HTTPS); the server name can be FQDN or IP
address.

Examples of valid URLs:

firmware.grandstream.com or fw.ipvideotalk.com/gs

Firmware Upgrade procedure

Please follow below steps in order to upgrade the firmware version of your HT813:

1. Access your HT813 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
“Firmware Server Path” field and choose to upgrade via TFTP or HTTP/HTTPS or FTP/FTPS.

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 HT813 to update the new
firmware.
 Figure 11: Firmware Upgrade Page

Upgrading via Local Directory

1. Download the firmware file from Grandstream web site

2. Unzip it and copy the file in to a folder in your PC

3. From the HT813 web interface (Advanced Settings page) you can browse your hard drive and select the folder you previously saved the file
(HT8xfw.bin)

4. Click “Upload Firmware” and wait 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 Servers

For users that would like to use remote upgrading without a local TFTP/FTP/HTTP 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 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 HT813.

End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use

Microsoft IIS web server.

Firmware and Configuration File Prefix and Postfix

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

Managing Firmware and Configuration File Download

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

Configuration File Download

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

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

When the HT813 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. The configuration file name should be in
lower case letters.

Only XML or binary config file formats are accepted.

The MAC header in XML config file should be the device MAC or needs to be removed completely.

Figure 12: XML Config File – MAC Header

RESTORE FACTORY DEFAULT SETTINGS

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

Using the Reset Button

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

1. Unplug the Ethernet cable.

2. Locate the reset hole on the back panel of your HT813.

3. Insert a pin in this hole, and press for about 7 seconds.

4. Take out the pin. All unit settings are restored to factory settings

Using the IVR Command

Reset default factory settings using the IVR prompt:

1. Dial “***” for voice prompt.

2. Enter “99” and wait for “reset” voice prompt.

3. Enter the encoded MAC address (Look below on how to encode MAC address).

4. Wait 15 seconds and device will automatically reboot and restore factory settings.

Encode the MAC Address

1. Locate the MAC address of the device. It is the 12-digit HEX number on the bottom of the unit.

2. Key in the MAC address. Use the following mapping:

Key Mapping

Key Mapping

0-9 0-9

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

B 222

C 2222

D 33 (press the “3” key twice, “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 HT813 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 Reset button (after selecting the reset type).
Full Reset: This will make a full reset

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

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

Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.

If the HT813 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.

CHANGE LOG
This section documents significant changes from previous versions of the admin guide for HT813. Only major new features or major document
updates are listed here. Minor updates for corrections or editing are not documented here.

Firmware Version 1.0.17.3

Added support for “Disable User Level Web Access”. [Disable User Level Web Access]

Added support for “Disable Viewer Level Web Access”. [Disable Viewer Level Web Access]

Force user to change the password upon first login using the default password to the Admin/User/Viewer Account. [Web UI Access Level
Management]

Firmware Version 1.0.17.2

Added support “SIP URI Scheme When Using TLS” option on the FXO page. [SIP URI Scheme When Using TLS]

Added the “Incoming Call Duration Limit” in PSTN. [FXS] [FXO]

Added ability to have a second VLAN under switch mode. [LAN Port VLAN Feature Under Bridge Mode]

Added support for “https://” in Config Server Path field. [Config Server Path]

Added “Device.DeviceInfo.SupportedDataModel” in TR data model. [Enable TR-069]

Added some missing parameters to the TR069 data model. [TR069 data model]

Firmware Version 1.0.15.7

Added support for Always send HTTP Basic Authentication Information. [Always send HTTP Basic Authentication Information]

Added Sectigo CA and Charter CA to the Trusted CA Certificate List. [Load CA Certificates]

Added support for Ring frequency 25Hz on FXS port. [Ring Frequency]

Updated the “Let’s encrypt” root CA certificate. [Load CA Certificate]

Added support for SRTP Key Length. [SRTP Key Length]

Updated Syslog and make it more user-friendly. [Syslog level]

Reorganized the order of display of the software version on the status page. [Software version]

Added support for variable on Provisioning link. [Enable using tags in the URL]

Added feature Configuration File Types Allowed. [Configuration File Types Allowed]

Added support of DHCP Option 150. [Additional Override DHCP Option]

Added SSL Key Log File. [With Secret Key Information]

Added support for Inband DTMF Duration. [Inband DTMF Duration]

Added support for DSP DTMF Detector Duration Threshold. [DSP DTMF Detector Duration Threshold]

Added support for downloading and Processing ALL Available Config Files. [Download and Process ALL Available Config Files]

Added support for new override config file option in “cfgMAC_override.xml”. [cfgMAC_override.xml]

Added support for When to Restart Session After Re-INVITE received. [When to Restart Session After Re-INVITE received]
 Updated the Web Lockout duration range to a maximum of 60mins instead of 15mins. [Web Lockout Duration]

Added Individual Certificate Generation in status page. [Individual Certificate Generation]

Added support for Trusted CA certifcate A and B. [Trusted CA certifcate A] [Trusted CA certifcate B]

Firmware Version 1.0.13.3

No major changes.

Firmware Version 1.0.11.2

No major changes.

Firmware Version 1.0.9.1

Added feature “DHCP domain name”.[DHCP domain name]

Firmware Version 1.0.7.1

Added feature “Load CA Certificates”. [Load CA Certificates]

Increased “SIP TLS Certificate” and “SIP TLS Private Key” supported maximum length from 2048 to 4096. [SIP TLS Private Key]

Added New Zealand Standard for Pulse Dialing Standard. [Pulse Dialing Standard]

Firmware Version 1.0.5.2

Added support for unlimited ring without timeout. [Ring Timeout]

Added feature “Connection Request Port”. [Connection Request Port]

Added feature “SIP User-Agent” for FXS port settings. [SIP User-Agent]

Added feature “SIP User-Agent” for FXO port settings. [SIP User-Agent]

Firmware Version 1.0.3.12

Added support for T.38 Fax mode under FXO Port. [Fax Mode]

Updated “São Paulo” time zone to UTC-3. [Time Zone]

Added feature “Allow SIP Factory Reset” for FXS port Settings. [Allow SIP Factory Reset]

Added feature “Allow SIP Factory Reset” for FXO port Settings. [Allow SIP Factory Reset]

Added feature “Pulse Dialing Standard”. [Pulse Dialing Standard]

Added support for RFC2833 event Count. [RFC2833 Events Count]

Added support for RFC2833 end event Count. [RFC2833 End Events Count]

Added feature “Disable Weak TLS Cipher Suites”. [Disable Weak TLS Cipher Suites]

Added feature “Minimum TLS Version”. [Minimum TLS Version]

Added feature “Maximum TLS Version”. [Maximum TLS Version]

Added feature “Syslog Protocol”. [Syslog Protocol]

Firmware Version 1.0.1.2

No major changes.

Firmware Version 1.0.0.8

This is the initial version for HT813.

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