Menu

Voice Prompt

Options

Main Menu

“Enter a Menu Option”

Press “*” for the next menu option

Press “#” to return to the main menu

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

01

“DHCP Mode”,

“Static IP Mode”

“PPPoE Mode“

Press “9” to toggle the selection

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

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

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

02

“IP Address “ + IP address

The current WAN IP address is announced

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

03

“Subnet “ + IP address

Same as menu 02

04

“Gateway “ + IP address

Same as menu 02

05

“DNS Server “ + IP address

Same as menu 02

07

Preferred Vocoder

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

PCM U / PCM A

iLBC

G-726

G-723

G-729

OPUS

10

“MAC Address”

Announces the MAC address of the unit.

Note: The device has two MAC addresses. One for the WAN port and one for the LAN port. The device MAC address announced is the address of LAN port.

12

WAN Port Web Access

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

Default is Auto.

13

Firmware Server IP Address

Announces current Firmware Server IP address.

Enter 12-digit new IP address.

14

Configuration Server IP Address

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

15

Upgrade Protocol

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

Default is HTTPS.

16

Firmware Version

Announces Firmware version information.

17

Firmware Upgrade

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

Always check

Check when pre/suffix changes

Never upgrade

20

“Certification Type”

Annouces the Certification Information of the HT813.

47

“Direct IP Calling”

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

86

Voice Mail

Access to your voice mails messages.

99

“RESET”

Press “9” to reboot the device

Enter MAC address to restore factory default setting

(See RESTORE FACTORY DEFAULT SETTINGS section)

“Invalid Entry”

Automatically returns to main menu

“Device not registered”

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

Status

MAC Address

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 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 Address

Displays assigned IPv4 address.

WAN IPv6 Address

Displays assigned IPv6 address.

Product Model

Displays product model info. Default is HT813.

Hardware Version

Displays the hardware revision information and the part number.

Software Version

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.

Note: The device periodically saves the current time to a timefile on every reboot to help preserve time information between reboots.

PPPoE Link Up

Indicates PPPoE connection status.

NAT

Indicates type of NAT when it is configured.

Individual Certificate Generation

Indicates the current individual Certificate Generation.

Port Status

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

New End User Password

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

Confirm End User Password

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

New Viewer Password

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

Confirm Viewer Password

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

Web/SSH Access

Web Session Timeout

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

Web Access Attempt Limit

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

Lockout Time Interval

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

Web Access Mode

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

HTTP Web Port

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

HTTPS Web Port

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

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 Telnet

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

Telnet Port

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

WAN Side Web/SSH Access

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

Default setting is Auto.

White List for WAN Side

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

Black List for WAN Side

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

Internet Protocol

Selects one of the following IP protocol modes:

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

IPv4 Address

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

Dynamically assigned via DHCP

All the field values for the static IP mode are not used (even though they are still saved in the flash memory.) The 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 optional but may be required by ISP.
DHCP domain name: allows user to configure DHCP domain name. This option specifies the domain name that the client should use when resolving hostnames via the Domain Name System. This field is optional.
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.

Use PPPoE

Set the PPPoE account settings. If selected, ATA attempt to establish a PPPoE session if any of the PPPoE fields is set.
PPPoE account ID:Defines the PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection.
PPPoE password:Specifies the PPPoE account password.
PPPoE Service Name:Defines PPPoE service name. If your ISP uses a service name for the PPPoE connection, enter the service name here. This field is optional. Default is blank.

Preferred DNS server

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

Statically configured as IP address

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

IPv6 Address

Allows users to configure the appropriate network settings on the 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 acquires its IP address from the first DHCP server it discovers from the LAN it is connected.
Static IP mode: configure IP address, 1st and 2nd DNS server, preferred DNS server. These fields are set to zero by default.
Full Static: When enabling the option full static, users need to specify the Static IPv6 and the IPv6 Prefix length.
Prefix Static: When enabling the option prefix static, users need to specify the IPv6 Prefix (64 bits).

Time Zone

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

Self-Defined Time Zone

Allows users to define their own time zone.

Allow DHCP server to set Time Zone

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

Local Time

Allows users to synchronize the device’s date and time with their computer’s local time and date through the Web UI.

By clicking the “Sync PC” button, the HT813 will automatically update its internal clock to match the current time on the user’s PC.

Language

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

NAT/DHCP Server Information & Configuration

Device Mode

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 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 Maximum Ports

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

NAT TCP Timeout

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

NAT UDP Timeout

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

Uplink Bandwidth

Specifies the maximum uplink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set 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 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.

Downlink Bandwidth

Specifies the maximum downlink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be 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 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 Support

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

Reply to ICMP on WAN Port

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

Cloned WAN MAC Address

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

LAN Port VLAN Feature Under Bridge Mode

This feature allows users to configure a different VLAN tag and priority value for the second network port when HT is configured in bridge mode.

The priority value range is 0-7, The VLAN tag range is 0-4094.
The default VLAN Tag and Priority value are 0.

Enable LAN DHCP

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

LAN DHCP Base IP

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 IP, the LAN base IP address will be changed based on the network mask the effective subnet will be increased by 1.
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 Start IP

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

LAN DHCP End IP

Default value is 199. This parameter allows a user to limit the number of local network devices connected to the internal router.

LAN Subnet Mask

Sets the LAN subnet mask.
Default value is 255.255.255.0

DHCP IP Lease Time

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.

DMZ IP

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 forwarding.

Port Forwarding

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

Reset Type

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

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

PSTN Access Code

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

PIN for VoIP-to-PSTN Calls

Maximum 8 digits to authorize calling PSTN numbers from VoIP.

PIN for PSTN-to-VoIP Calls

Maximum 8 digits to authorize calling VoIP terminals from PSTN.

Unconditional Call Forward to PSTN

VoIP calls will be forwarded to the specified PSTN number.
Specify PSTN number.

Unconditional Call Forward to VoIP

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

Advanced Settings

New Admin Password

Defines the administrator level password to access the Advanced Web Configuration page. This field is case sensitive. Only the administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reasons after clicking update and saved.
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.

Layer 2 QoS

Sets values for:

Black List for WAN Side Port

It could be either port range or single port separated by a “,”
Example: “5000-6000, 7000 “.

STUN Server

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

Keep-alive Interval

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

Use STUN to detect network connectivity

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

Use DNS to detect network connectivity

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

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

Firmware Server Path

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

Config Server Path

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.

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.

XML Config File Password

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

HTTP/HTTPS/FTP/FTPS User Name

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

HTTP/HTTPS/FTP/FTPS Password

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

Firmware File Prefix

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

Firmware File Postfix

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

Config File Prefix

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

Config File Postfix

Checks if configuration files are with matching postfix before downloading them. It allows user to store 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”.

Always send HTTP Basic Authentication Information

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 and password no matter whether the server needs authentication or not. If set to “No”, only send HTTP/HTTPS user name and password when the server needs authentication.
Set to “No” by Default.

Allow DHCP Option 66 or 160 to Override the Server

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

Additional Override DHCP Option 

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

3CX Auto Provision

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

Automatic Upgrade

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

Default is No.

Randomized Automatic Upgrade

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

Always Check for New Firmware at Boot up

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

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

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

Always Skip the Firmware Check

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

Configuration File Types Allowed

allows users to configure provision configuration file type in xml file only or all file types.
Default value is “All”

Download and Process All Available Config Files

By default, device will provision the first available config in the order of cfgMAC.xml→cfgMAC→cfgMODEL.xml→cfg.xml (corresponding to device specific, model specific and global configs).

If this option is enabled, the HT813 will change the downloading process to cfg.xml→cfgMODEL.xml→cfgMAC.xml→cfgMAC. The following files will override the files that has already been loaded and processed. The default value for this parameter is “No”.

Note: Support for the new override config file option as “cfgMAC_override.xml” file has been added to the HT813 Model, which can be used to apply final changes.

Disable SIP NOTIFY Authentication

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

Authenticate Conf File

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

Validate Server Certificates

This feature allows users to validate server certificates with our trusted list of TLS connections. Default is 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.

Load CA Certificates

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.
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.

SIP TLS Certificate

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

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

Enable TR-069

Sets the ATA adapter system to enable the “CPE WAN Management Protocol” (TR-069). Default setting is No.

Notes:

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.

Periodic Inform Enable

Sends periodic “inform” packets to ACS.
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.

Connection Request Port

Configures the TR-069 Connection Request Port.
The value range is 0 to 65535. Default is 7547

CPE SSL Certificate

Configures the Cert File for the 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.

Enable SNMP

Enables the SNMP Service.
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

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 Protocol

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

SNMPv3 Trap Privacy Protocol

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

SNMPv3 Trap Authentication Key

Enter the Trap Authentication Key.

SNMPv3 Trap Privacy Key

Enter the Trap Privacy Key.

Enable RADIUS Web Access Control

Default is No.

Action upon RADIUS Auth Server Error

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

RADIUS Auth Protocol

Configure RADIUS authentication protocol, the available options are:

Default value is “PAP”.

RADIUS Auth Server Address

Address of RADIUS Auth server.

RADIUS Auth Server Port

Port of RADIUS Auth server.

RADIUS Shared Secret

Set RADIUS shared secret.

RADIUS VSA Vendor ID

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

RADIUS VSA Access Level Attribute

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

Enable DDNS

Allow users to use DDNS.

DDNS Server

Selects DDNS Server: dyndns.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.

System Ring Cadence

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

Call Progress Tones:

Dial Tone

Ringback Tone

Busy Tone

Reorder Tone

Confirmation Tone

Call Waiting Tone

Prompt Tone

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

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

Example configuration for N.A.

Dialtone:

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

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

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

Prompt Tone Access Code

Allows users to define a specific key pattern (up to 20 digits) that, when dialed, triggers the device to play the prompt tone.

Lock Keypad Update

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

Disable Voice Prompt

Voice prompt is disabled if set to Yes.

Disable Direct IP Call

Direct IP call is disabled if set to Yes.

Life Line Mode

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

Blacklist for Incoming Calls

Allow users to block incoming calls from a specific list of numbers.
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 (‘+’).

NTP Server

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

Secondary NTP Server

Allows users to configure the secondary NTP server, by default, no secondary NTP Server is configured.

Allow DHCP Option 42 to override NTP server

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

DHCP Option 17 Enterprise Number

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 provided to the device.
Default is 3561.

Disable Weak TLS Cipher Suites

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

Default is Enable Weak TLS Ciphers Suites.

Minimum TLS Version

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

Maximum TLS Version

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

Default is Unlimited.

Syslog Protocol

This feature allows users to customize the Syslog Protocol. The Syslog protocol can be either UDP or 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:

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…), 

Send SIP Log

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

Information Capture

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

Automatic Reboot

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

Download Device Configuration

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

Download Device XML Configuration

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

Upload Firmware

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

Upload Configuration

Allows to upload configuration file to the device (supports both TXT and XML format).

Export Backup Configuration

Download backup file to local computer.
The backup file is XML and encrypted.

Restore From Backup Configuration

Uploads the backup file to the ATA to restore your saved configuration

Account Active

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

Primary SIP Server

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

Failover SIP Server

Specifies failover SIP server IP address or domain name provided by VoIP service provider.

This server will be used if the primary SIP server becomes unavailable.

Prefer Primary SIP Server

Selects to prefer primary SIP server. The account will register to primary Server if registration with Failover server expires.

Default is No.

Outbound Proxy

Specifies IP address or domain name of outbound Proxy, or media gateway, or session border controller. Used by 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 Proxy

Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. By default, this field is left empty.

Prefer Primary Outbound Proxy

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

By default, this option is disabled.

Allow DHCP Option 120 (override SIP Server)

Configures the HT813 to collect SIP server address from DHCP option 120.

Default is No.

SIP Transport

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

SIP URI Scheme When Using TLS

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

Use Actual Ephemeral Port in Contact with TCP/TLS

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

NAT Traversal

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

SIP User ID

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

Authenticate ID

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

Authenticate Password

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

Name

Chooses a name to be associated to user.

DNS Mode

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

Default is A Record.

DNS SRV Failover Mode

Allows users to choose how SIP requests are routed based on DNS SRV query results and registration status.

The default setting is “Default”.

Failback Timer

This feature allows the user to set how long (in minutes) the device should continue using the failover server (IP address) for SIP requests like REGISTER or INVITE after a failover has occurred. Once this timer expires, the device will attempt to switch back to the primary server, even if the failover server is still reachable.

Default value 60 minutes and max is 45 days.

Register Before DNS SRV Failover

This feature allows users to control whether the device should send a REGISTER request when switching to a new server due to DNS SRV failover. The options are:

The default settings is “No”.

Primary IP

Specifies the main IP address the device will use to send SIP requests when DNS Mode is set to “Use Configured IP”. This IP will be tried first before any backup options.

Backup IP1

Defines the first fallback IP address to be used if the Primary IP does not respond after 3 attempts. Acts as the first level of redundancy.

Backup IP2

Defines the second fallback IP address to be used if both the Primary IP and Backup IP1 fail to respond after 3 attempts each. Serves as the final failover option before cycling back to the Primary IP.

Tel URI

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 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.

Unregister on Reboot

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

Outgoing Call Without Registration

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

Register Expiration

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

Reregister before Expiration

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

SIP Registration Failure Retry Wait Time

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

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Allows users to set how long the device should wait before retrying a SUBSCRIBE request after receiving a 403 Forbidden response from the server. A value of 0 means the device will stop retrying immediately after a 403 response.

Default is 1200 seconds. The valid range is between 0 and 3600 seconds.

Enable SIP OPTIONS Keep Alive

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

SIP OPTIONS Keep Alive Interval

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

SIP OPTIONS Keep Alive Max Lost

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

Layer 3 QoS

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

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.

Local RTP Port

Defines the local RTP-RTCP port pair the HT813 will listen and transmit. It is the HT813 RTP port for channel 0. The default value for FXS port is 5004.

Use Random SIP Port

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

Use Random RTP Port

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

Enable RTCP

Allows users to enable RTCP. Default setting is Yes.

Hold Target Before Refer

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

Refer-To Use Target Contact

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

Transfer on Conference Hang-up

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

Disable Bellcore Style 3-Way Conference

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

Remove OBP from Route Header

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

Default is No.

Support SIP Instance ID

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

Validate Incoming SIP Messages

Validates incoming messages. Default is No.

Check SIP User ID for Incoming INVITE

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

Authenticate Incoming INVITE

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

Authenticate server certificate domain

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

Trusted domain name list

Specifies trusted domain names for TLS certificate validation. If a certificate’s domain name matches any entry in this list, it will be considered trusted.

Users can enter specific domain names or use wildcards (e.g., *.grandstream.com). Separate multiple entries with commas. (Max length is 256 characters)

Notes:

Authenticate server certificate chain

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

Trusted CA Certificates

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

Allow Incoming SIP Messages from SIP Proxy Only

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

Use Privacy Header

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

Use P-Preferred-Identity Header

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

Default setting is: Default.

Use P-Access-Network-Info Header

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

Use P-Emergency-Info Header

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

SIP REGISTER Contact Header Uses

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

Caller ID Fetch Order

Selects the Caller ID display order which need to be respected by the 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.

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 Reset

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

Maximum Number of SIP Request Retries

This setting allows the user to define how many times the device will retry sending a SIP request if no response is received from the server. Once the maximum number of retries is reached without a reply, the device will consider the request failed and trigger failover

Valid range is 1 to 10 and the default value is 2.

SIP T1 Timeout

Defines T1 timeout value.

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

For example, the 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.

SIP T2 Interval

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

SIP Timer D

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

DTMF Payload Type

Defines payload type for DTMF using RFC2833.

Preferred DTMF method (in order)

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

You can configure up to three priorities.

Force DTMF to be sent via SIP INFO simultaneously

When enabled, the device will send DTMF tones using the SIP INFO method in addition to the configured preffered DTMF method (e.g., RFC2833 or in-audio). This can improve compatibility with certain SIP servers or endpoints that require DTMF via SIP INFO.

When disabled (default), DTMF is sent only using the selected primary method.

Inband DTMF Duration

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

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

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

DSP DTMF Detector Duration Threshold

Allows users to configure the DSP DTMF Detector Duration and Inter-Duration Threshold.

The DSP DTMF Detector duration threshold varies from 20-200ms with 30ms as a Default Value.

The DSP DTMF Detector inter-duration threshold varies from 20-200ms with 30ms as a Default Value.

Disable DTMF Negotiation

Uses above DTMF order without negotiation.

Default is No.

Generate Continuous RFC2833 Events

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

Send Hook Flash Event

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

Flash Digit Control

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

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

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

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

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

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

Enable Call Features

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.

Default is Yes.

Off Hook Auto Dial

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

Off Hook Auto Dial Delay

Specifies the auto-dial delay after off hook.

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 Header

Configures the SIP User-Agent Header.

SIP User-Agent

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

Disable Call Waiting

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

Default is No.

Disable Call Waiting Caller ID

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

Disable Call Waiting Tone

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

Disable Connected Line ID

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

Disable Receiver Off Hook Tone

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

Disable Reminder Ring for On-Hold Call

Enables playing the reminder ring. Default is No

Disable Visual MWI

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

Do Not Escape ‘#’ as %23 in SIP URI

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

Disable Multiple m Line in SDP

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

Ring Timeout

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

Delayed Call Forward Wait Timeout

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

No Key Entry Timeout

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

Early Dial

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

This option should be used only if there is a SIP proxy is configured and supporting 484 responses (Incomplete Address). Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). Default is No.

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

Dial Plan Prefix

Adds specified prefix to dialed number.

Use # as Dial Key

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

Default is Yes.

Dial Plan

Dial Plan Rules:

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

2. Grammar: x – any digit from 0-9;

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. – replace digit 2 with 011 when dialing

8. – 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+ | xxxxxxx} –

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

Example 3: {1xxx[2-9]xxxxxx | 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. | [2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }

Dial Plan

Explanation of example rule (reading from left to right):

^1900x. – prevents dialing any number started with 1900

[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

011[2-9]x. – allows international calls starting with 011

[3469]11 – allow dialing special and emergency numbers 311, 411, 611 and 911

Note: In some cases, user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature. As an example { *x+ } will allow to dial * followed by any length of numbers.

SUBSCRIBE for MWI

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

Send Anonymous

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

Anonymous Call Rejection

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

Special Feature

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

Enable Session Timer

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

Session Expiration

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

Min-SE

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

Caller Request Timer

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

Default is No.

Callee Request Timer

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

Default is No.

Force Timer

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

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

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls.

UAC: The handy tone acts as the refresher.

UAS: Callee or proxy server act as the refresher.

Default is Omit.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls:

UAS: The handy tone acts as the refresher.

UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE

Uses INVITE message to refresh the session timer.

Default is No.

When to Restart Session After Re-INVITE received

Allows users to support to delay posting Media Change Event with this new feature,it can be set to “Immediately” or to “After replying 200OK”

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.

Add Auth Header on Initial REGISTER

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

Default is No.

Conference URI

Allows users to manually configure the conference URL.

Use First Matching Vocoder in 200OK SDP

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

Default is No.

Preferred Vocoder

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

Voice Frames per TX

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

G723 Rate

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

iLBC Frame Size

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

Default is 20ms.

Disable OPUS Stereo in SDP

Disables OPUS stereo in SDP.

Default is No.

iLBC Payload type

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

Default is 97.

OPUS Payload Type

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

VAD

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

Symmetric RTP

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

Fax Mode

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

Re-INVITE after Fax Tone Detected

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

Jitter Buffer Type

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

Jitter Buffer Length

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

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

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

SRTP Mode

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

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

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

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

SRTP Key Length

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

1. AES 128&256 bit

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.

SLIC Setting

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

EUROPEAN CTR21

GERMANY

INDIA/NEW ZEALAND

JAPAN CO

JAPAN PBX

STANDARD 900 ohms

UK

Caller ID Scheme

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

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

DTMF Caller ID

Defines the start and stop tones as delimiters for the 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 Disconnect

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

Play busy/reorder tone before Loop Current Disconnect

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

Loop Current Disconnect Duration

Configures the duration of voltage drop described in topic above. HT813 supports a duration range from 100 to 10000 ms. Default value is 200.

Enable Pulse Dialing

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

Pulse Dialing Standard

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

Enable Hook Flash

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

Hook Flash Timing

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

Default values are 300 minimum and 1100 maximum.

On Hook Timing

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

Gain

Adjusts the voice path volume.

• Rx is a gain level for signals transmitted by FXS

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

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

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

Disable Line Echo Canceller (LEC)

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

Default is No.

Disable Network Echo Suppressor

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

Default is No.

Outgoing Call Duration Limit

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

Incoming Call Duration Limit

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

Ring Frequency

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

Default is 20Hz.

Enable High Ring Power

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

RFC2833 Events Count

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

RFC2833 End Events Count

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

RFC2833 Convert Mode

Allows users to configure how DTMF tones (keypad tones from analog phones) are detected and transmitted over VoIP using RFC2833. This setting determines how the ATA processes in-band analog DTMF tones and converts them into digital RFC2833 signals during SIP calls.

The available options are:

Distinctive Ring Tone

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.

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

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

Alert-Info:; info=ring5

Ring tones

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

Account Active

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

Primary SIP Server

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

Failover SIP Server

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

Prefer Primary SIP Server

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

Outbound Proxy

Specifies IP address or domain name of outbound Proxy, or media gateway, or session border controller. Used by 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 Proxy

Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. By default, this field is left empty.

Prefer Primary Outbound Proxy

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

SIP Transport

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

SIP URI Scheme When Using TLS

When TLS is enabled on the FXO HT813 device, the SIP URI Scheme When Using TLS option allows users to specify the type of SIP URI scheme that will be used during the communication. The available options typically include:

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.

NAT Traversal

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

SIP User ID

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

Authenticate ID

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

Authenticate Password

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

Name

Chooses a name to be associated to user.

DNS Mode

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

Default is A Record.

DNS SRV Failover Mode

Allows users to choose how SIP requests are routed based on DNS SRV query results and registration status.

The default setting is “Default”.

Failback Timer

This feature allows the user to set how long (in minutes) the device should continue using the failover server (IP address) for SIP requests like REGISTER or INVITE after a failover has occurred. Once this timer expires, the device will attempt to switch back to the primary server, even if the failover server is still reachable.

Default value 60 minutes and max is 45 days.

Register Before DNS SRV Failover

This feature allows users to control whether the device should send a REGISTER request when switching to a new server due to DNS SRV failover. The options are:

The default settings is “No”.

Primary IP

Specifies the main IP address the device will use to send SIP requests when DNS Mode is set to “Use Configured IP”. This IP will be tried first before any backup options.

Backup IP1

Defines the first fallback IP address to be used if the Primary IP does not respond after 3 attempts. Acts as the first level of redundancy.

Backup IP2

Defines the second fallback IP address to be used if both the Primary IP and Backup IP1 fail to respond after 3 attempts each. Serves as the final failover option before cycling back to the Primary IP.

Tel URI

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 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.

Unregister on Reboot

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with 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 Without Registration

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

Register Expiration

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

Reregister Before Expiration

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

SIP Registration Failure Retry Wait Time

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

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Allows users to set how long the device should wait before retrying a SUBSCRIBE request after receiving a 403 Forbidden response from the server. A value of 0 means the device will stop retrying immediately after a 403 response.

Default is 1200 seconds. The valid range is between 0 and 3600 seconds.

Enable SIP OPTIONS Keep Alive

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

SIP OPTIONS Keep Alive Interval

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

SIP OPTIONS Keep Alive Max Lost

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

Layer 3 QoS

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

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.

Local RTP Port

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

Use Random SIP Port

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

Use Random RTP Port

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

Enable RTCP

Allows users to enable RTCP. Default setting is Yes.

Remove OBP from Route Header

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

Default is No.

Support SIP Instance ID

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

Validate Incoming SIP Message

Validates incoming messages. Default is No.

Check SIP User ID for Incoming INVITE

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

Note: the HT813 also accepts incoming INVITEs if the SIP User ID matches either the To: header or the Request URI. If neither field matches, the call will still be rejected with 403 Forbidden.

Authenticate Incoming INVITE

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

Authenticate server certificate domain

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

Trusted domain name list

Specifies trusted domain names for TLS certificate validation. If a certificate’s domain name matches any entry in this list, it will be considered trusted.

Users can enter specific domain names or use wildcards (e.g., *.grandstream.com). Separate multiple entries with commas. (Max length is 256 characters)

Notes:

Authenticate server certificate chain

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

Trusted CA Certificates

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

Allow Incoming SIP Messages from SIP Proxy Only

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

Use Privacy Header

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

Use P-Preferred-Identity Header

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

Default setting is: Default.

Use P-Access-Network-Info Header

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

Use P-Emergency-Info Header

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

SIP REGISTER Contact Header Uses

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

Allow SIP Factory Reset

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

Maximum Number of SIP Request Retries

This setting allows the user to define how many times the device will retry sending a SIP request if no response is received from the server. Once the maximum number of retries is reached without a reply, the device will consider the request failed and trigger failover

Valid range is 1 to 10 and the default value is 2.

SIP T1 Timeout

Defines T1 timeout value.

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

For example, the 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.

SIP T2 Interval

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

SIP Timer D

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

DTMF Payload Type

Defines payload type for DTMF using RFC2833.

Preferred DTMF method (in order)

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

You can configure up to 3 Priorities.

Force DTMF to be sent via SIP INFO simultaneously

When enabled, the device will send DTMF tones using the SIP INFO method in addition to the configured preffered DTMF method (e.g., RFC2833 or in-audio). This can improve compatibility with certain SIP servers or endpoints that require DTMF via SIP INFO.

When disabled (default), DTMF is sent only using the selected primary method.

Inband DTMF Duration

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

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

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

DSP DTMF Detector Duration Threshold

Allows users to configure the DSP DTMF Detector Duration and Inter-Duration Threshold.

The DSP DTMF Detector duration threshold varies from 20-200ms with 30ms as a Default Value.

The DSP DTMF Detector inter-duration threshold varies from 20-200ms with 30ms as a Default Value.

Disable DTMF Negotiation

Uses above DTMF order without negotiation.

Default is No.

Generate Continuous RFC2833 Events

When enabled the RFC2833 events are generated until key is released.

Default is No.

Flash Digit Control

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

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

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

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

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

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

Auto Send DTMF Upon Call Start

Configures DTMF digits that the ATA will automatically send to the TDM (Time Division Multiplexing) side as soon as a VoIP call is connected.

TDM in this context refers to the legacy telephony interface (analog FXO) that uses time-slot-based signaling. This feature is useful for automatically sending access codes, IVR inputs, or post-dial digits.

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 Header

Configures the SIP User-Agent Header.

SIP User-Agent

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

Do Not Escape ‘#’ as %23 in SIP URI

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

Disable Multiple m Line in SDP

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

Ring Timeout

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

Early Dial

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

This option should be used only if there is a SIP proxy is configured and supporting 484 responses (Incomplete Address). Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).

Default is No.

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

Dial Plan Prefix

Adds specified prefix to dialed number.

Use # as Dial Key

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

Default is Yes.

Dial Plan

Dial Plan Rules:

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

2. Grammar: x – any digit from 0-9;

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. – replace digit 2 with 011 when dialing

8. – 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+ | xxxxxxx} –

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

Example 3: {1xxx[2-9]xxxxxx | 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. | [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

[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically

1[2-9]xx[2-9]xxxxxx – allows dialing to any US/Canada Number with 11 digits length

011[2-9]x. – allows international calls starting with 011

[3469]11 – allow dialing special and emergency numbers 311, 411, 611 and 911

Note: In some cases, user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature. As an example { *x+ } will allow to dial * followed by any length of numbers.

SUBSCRIBE for MWI

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

Anonymous Call Rejection

Rejects incoming calls with anonymous caller ID with “486 Busy here” message.

Default is No.

Special Feature

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

Session Expiration

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

Min-SE

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

Caller Request Timer

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

Default is No.

Callee Request Timer

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

Default is No.

Force Timer

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

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

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls.

UAC: The handy tone acts as the refresher.

UAS: Callee or proxy server act as the refresher.

Default is Omit.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls:

UAS: The handy tone acts as the refresher.

UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE

Always refresh with INVITE instead of UPDATE Default is No.

When to Restart Session After Re-INVITE received

Allows users to support to delay posting Media Change Event with this new feature,it can be set to “Immediately” or to “After replying 200OK”

The default value is “Immediately”.

INVITE Ring-No-Answer Timeout (sec)

Between 5-300 seconds. Default 40 seconds.

Enable 100rel

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

Add Auth Header on Initial REGISTER

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

Use First Matching Vocoder in 200OK SDP

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

Preferred Vocoder

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

Voice Frames per TX

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

G723 Rate

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

iLBC Frame Size

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

Default is 20ms.

Disable OPUS Stereo in SDP

Disables OPUS stereo in SDP. Default is No.

iLBC Payload type

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

OPUS Payload Type

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

VAD

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

Symmetric RTP

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

Fax Mode

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

Re-Invite after Fax Tone Detected

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

Jitter Buffer Type

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

Jitter Buffer Length

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

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

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

SRTP Mode

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

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

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

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

SRTP Key Length

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

1. AES 128&256 bit

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.

Caller ID Scheme

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

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

DTMF Caller ID

Defines the start and stop tones as delimiters for the caller ID.

Start Tone and Stop Tone can be set to “Default”, “A”, “B”, “C”, “D” or “#”

FSK Caller ID Minimum RX Level (dB)

An adjustable value for the Caller ID signal to help this device to recognize Caller ID from different networks. Range: -96 to -0dB. Default -40dB.

FSK Caller ID Seizure Bits

Range is from 0 to 800 bits. Default 70.

FSK Caller ID Mark Bits

Range is from 0 to 800 bits. Default 40.

Caller ID Transport Type

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

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

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

Hook Flash Duration (ms)

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

Gain

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.

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.