HT80x - Administration Guide

HT801

HT802

HT801

HT802

Interfaces

Telephone Interfaces

One (1) RJ11 FXS port

Two (2) RJ11 FXS ports

Network Interface

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

LED Indicators

POWER, INTERNET, PHONE

POWER, INTERNET, PHONE1,
PHONE2

Factory Reset Button

Yes

Voice, Fax, Modem

Telephony Features

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

Voice Codecs

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

Fax over IP

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

Short/Long Haul Ring Load

5 REN: Up to 1km on 24 AWG

2 REN: Up to 1km on 24 AWG

Caller ID

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

Dial Methods

DTMF, Pulse

Disconnect Methods

Busy Tone, Polarity Reversal/Wink, Loop Current

Signalling

Network Protocols

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

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 Control

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

Security

Media

SRTP

Control

TLS/SIPS/HTTPS/SSH/Telnet

Management

Syslog support, SSH, Telnet remote management using web browser

Physical

Universal Power Supply

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

Environmental

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

Dimensions and Weight

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

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

Compliance

Compliance

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

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

Phone for HT801

Phone 1 & 2 for HT802

Internet port

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

DC Power

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

Reset

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

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 HT812/HT814 for the new IP address to take Effect.

03

“Subnet “ + IP address

Same as menu 02

04

“Gateway “ + IP address

Same as menu 02

05

“DNS Server “ + IP address

Same as menu 02

07

Preferred Vocoder

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

PCM U / PCM A

iLBC

G-726

G-723

G-729

OPUS

G722

10

“MAC Address”

Announces the MAC address of the unit.

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 enable / disable.

Default is disabled.

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/HTTP /FTP/FTPS

16

Firmware Version

Announces Firmware version information.

17

Firmware Upgrade

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

Always check

Check when pre/suffix changes

Never upgrade

47

“Direct IP Calling”

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

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 Setting 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

Account Status

MAC Address

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

IPv4 Address

Displays assigned IPv4 address.

IPv6 Address

Displays assigned IPv6 address.

VPN IPv4 Address

Displays assigned OpenVPN IPv4 address.

VPN IPv6 Address

Displays assigned OpenVPN IPv6 address.

Product Model

Displays product model info. Default is HT801/HT802.

Hardware Version

Displays the hardware revision information and the part number.

Software version

Program: Specifies Program version. This is the main firmware release number, which is always used for identifying the software system of the HT801/HT802.

Bootloader: Specifies Boot version.

Core: Specifies Core version.

Base: Specifies Base version.

CPE: Specifies CPE version. CPE version is displayed only when HT801/HT802 is connected to an ACS using TR-069 protocol.

Software Status

Indicates actual software status.

System Up Time

Indicates actual system time and uptime since the last reboot.

CPU Load

Indicates CPU load (%)

Network Cable Status

Indicates physical network cable status:

Status (Up/Down), Speed (Mbps), Operational Mode (Full/Half Duplex)

PPPoE Link Up

Indicates PPPoE connection status.

NAT

Indicates the type of NAT when it is configured.

Port Status

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

Port Options

Displays DND and call forward information on FXS ports.

CDR File

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

SIP File

Download, Preview or, Delete locally stored SIP trace.

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

Provision

Displays provisioning status.

Core Dump

Provides a generated core dump file if the unit malfunctions.

Clean will be displayed if no issues are found.

GR909

Click on “Test Page” to be redirected to the lines tests page (Hazardous Potential, Foreign Electromotive Forces, Resistive Faults, Receiver Off-hook and Ringer Equivalent Number)

Basic Settings

New End User Password

Configures user-level password. Case sensitive and max. Length of 25 characters.
Note: The new Password must contain 8-30 characters, Purposely not displayed for security reasons.

Confirm End User Password

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

New Viewer Password

Configures viewer-level password. Case sensitive and max. length of 25 characters.
Note: The new Password must contain 8-30 characters,

Purposely not displayed for security reasons.

Confirm Viewer Password

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

Web Session Timeout

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

Web Access Attempt Limit

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

Lockout time interval

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

Web Access Mode

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

By default, “HTTP” is selected.

HTTP Web Port

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

HTTPS Web Port

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

Disable SSH

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

SSH Port

Allows users to self-configure SSH Port number.

By default, the port number is 22.

SSH Idle Timeout

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

Disable Telnet

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

Telnet Port

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

Telnet Idle Timeout

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

Whitelist for WAN Side

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

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

Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

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

Blacklist for WAN Side

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

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

Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

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

Internet Protocol

Selects one of the following IP protocol modes:

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

Dynamically assigned via DHCP

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

Use PPPoE

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

Preferred DNS server

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

Statically configured as IP address

IPv6 Address

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

Enable Management Interface

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

Management Access

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

Enable SNMP Through Management Interface

This feature allows users to route SNMP packets through the management interface. The default setting is "No".

Enable TR-069 Through Management Interface

This feature allows users to route TR069 packets through the management interface. The default setting is "No".

Enable Syslog Through Management Interface

This feature allows users to route Syslog packets through the management interface. The default setting is "No".

Configures Voice VLAN Type :

Or,

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

priority : [0 – 7]; Default is 0

Time Zone

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

Self-Defined Time Zone

Allows users to define their own time zone.

Allow DHCP server to set Time Zone

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

The Default setting is Yes.

Language

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

Reset Type

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

There are 3 types of factory reset:

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

FXS Port (1,2)

Accounts Active

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

Primary SIP Server

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

Failover SIP Server

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

Prefer Primary SIP Server

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

 
Will register to Primary Server if Failover registration expires: If failover server registration ends, the client reverts to the main server.

 
Will register to Primary Server if Primary Server responds, need to enable SIP OPTIONS/NOTIFY Keep-Alive: Client registers with the main server if it's responsive, requiring keep-alive methods.

 
No: The client doesn't prioritize the main SIP server.

 
The default is No.

Outbound Proxy

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

Backup Outbound Proxy

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

Prefer Primary Outbound Proxy

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

By default, this option is disabled.

From Domain

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

Allow DHCP Option 120 (override SIP Server)

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

Default is No.

SIP Transport

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

SIP URI Scheme When Using TLS

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

Use Actual Ephemeral Port in Contact with TCP/TLS

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

NAT Traversal

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

Default setting is No.

SIP User ID

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

Authenticate ID

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

Authenticate Password

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

Name

Chooses a name to be associated to user.

DNS Mode

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

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

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

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

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

DNS SRV use Registered IP

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

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

DNS SRV Failover Mode

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

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

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

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

Failback Timer

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

Register Before DNS SRV Failover

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

TEL URI

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

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

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

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

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

SIP Registration

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

Unregister on Reboot

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

Default value is set to "No"

Outgoing Call Without Registration

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

Register Expiration

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

Reregister Before Expiration

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

SIP Registration Failure Retry Wait Time

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

SIP Registration Failure Retry Wait Time upon 403 Forbidden

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

Port Voltage Off upon no SIP Registration or SIP Registration Failure

Allows users to configure the timer of Port Voltage Off.
For Port Voltage Off upon no SIP, the Valid Range is 0 to 60 with the Default value being 0

( 0 means the port voltage is never turned off )

Delay Time of Port Voltage Off Timer Since Boot

Allows users to set a delay time of FXS line voltage when there is no SIP registration.
The valid Range is 0 to 60, and the Default value is 0.

Enable SIP OPTIONS/NOTIFY Keep Alive

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

SIP OPTIONS/NOTIFY Keep Alive Interval

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

SIP OPTIONS/NOTIFY Keep Alive Max Lost

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

Layer 3 QoS

Defines Diff-Serv values for SIP and RTP.

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

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

Local SIP Port

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

Local RTP Port

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

Use Random SIP Port

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

Use Random RTP Port

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

Random RTP Port Range

This feature allows users to specify the minimum and maximum RTP port ranges that can be used with the HT device.
The Valid range is between 1024-65535

Enable RTCP

Allows users to enable RTCP. Default setting is Yes.

Hold Target Before Refer

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

Refer-To Use Target Contact

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

Transfer on Conference Hang-up

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

Remove OBP from Route Header

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

Default is No.

Support SIP Instance ID

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

Default is No.

Support Outbound

This feature allows users to set RFC5626 outbound. Enabled “Support SIP Instance ID” and “Support outbound” which could support RFC 5626
Disabled by Default.

Support GRUU

This feature allows users to set RFC5627 GRUU. Enabled “Support SIP Instance ID” and “Support GRUU” which could support RFC 5627
Disabled by Default.

Validate Incoming SIP Message

Validates incoming messages.

Default is No.

Check SIP User ID for Incoming INVITE

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

Authenticate Incoming INVITE

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

Authenticate server certificate domain

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

Default setting is “No“.

Authenticate server certificate chain

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

Default setting is “No“.

Allow Incoming SIP Messages from SIP Proxy Only

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

Default is No.

Use Privacy Header

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

Use P-Preferred-Identity Header

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

Default setting is Default.

Use P-Access-Network-Info Header

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

Use P-Emergency-Info Header

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

Use P-Asserted-Identity Header

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

Use P-Early-Media Header

Allows users to set up the P-Early-Media Header. Once activated via "Ringback Tone at No Early Media," the phone will emit a ringback tone if early media RTP packets face obstruction from the intermediary server. Furthermore, the phone will acquire a "P-Early-Media" header within the 180/183 response, indicating "inactive," "recvonly," or "gated" values to signify the absence of incoming RTP packets.
Disabled by Default.

SIP REGISTER Contact Header Uses

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

Caller ID Fetch Order

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

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

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

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

Allow SIP Factory Reset

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

The authentication in this case can be either with:

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

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

Maximum Number of SIP Request Retries

SIP T1 Timeout

Defines T1 timeout value.

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

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

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

SIP T2 Interval

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

SIP Timer D

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

Default is 0.

DTMF Payload Type

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

Preferred DTMF method (in order)

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

Force DTMF to be sent via SIP INFO simultaneously

This option forces the DTMF signals to be sent using the SIP INFO method for better compatibility with VoIP systems.
Disabled by Default.

Inband DTMF Duration

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

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

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

Inband DTMF Tx Gain

Allows users to configure the Inband DTMF Tx Gain.
The Valid rand is -12 to 12 db.

The Default value is 0db.

DSP DTMF Detector Duration Threshold

Allows users to config the DSP DTMF Detector Duration Threshold.

The default Duration is 30 ms. Valid range: 20-2000 ms.
The default Inter-duration is 30 ms. Valid range: 20-2000 ms.

Disable DTMF Negotiation

Uses above DTMF order without negotiation.

Default is No.

Generate Continuous RFC2833 Events

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

Send Hook Flash Event

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

Default is No.

Flash Digit Control

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

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

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

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

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

Note: Please refer to the user guide for detailed steps to perform above operations.
Note: Additional Flash Digit events were added on the  firmware upgarde 1.0.43.11

Enable Call Waiting alert-info In 180 Ringing Response

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

Callee Flash to 3WC

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

Off Hook Auto Dial

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

Off Hook Auto Dial Delay

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

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

Off Hook Auto-Dial DTMF

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

Proxy-Require

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

Use NAT IP

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

SIP User-Agent

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

SIP User-Agent Postfix

Configures the SIP User-Agent Postfix.

Add MAC in User-Agent

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

Use MAC Header

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

RFC2543 Hold

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

Disable Call Waiting

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

Default is No.

Disable Call Waiting Caller ID

Disables displaying caller ID when receiving a second incoming call.

Default is No.

Disable Call Waiting Tone

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

Disable Connected Line ID

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

Disable Receiver Off Hook Tone

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

Disable Reminder Ring for On-Hold Call

Enables playing the reminder ring. Default is No

Disable Reminder Ring for DND

This feature allows users to disable the reminder rings when the FXS port is in DND mode. The default value is Yes.

Disable Visual MWI

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

Visual MWI Type

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

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

Note: Some phones (depending on the model of the analog phone) when this feature is set to NEON it might auto ring (short beeps) when there is a voice mail available for that FXS port where it is connected.

MWI Tone

When set to Default, device will play Stutter Dial Tone when there is voicemail, if set to Special Proceed Indication Tone, device will play the configured special proceed indication tone upon user offhook when there is voicemail

Ringback Tone at No Early Media

Allows users to customize the "Ringback Tone at No Early Media" setting. Upon activation with "Use P-Early-Media Header," the phone will emit a ringback tone if early media RTP packets encounter obstruction from the intermediate server. The phone also obtains a "P-Early-Media" header in the 180/183 response, indicating values like "inactive," "recvonly," or "gated," signifying the absence of incoming RTP packets.
Disabled by Default.

Do Not Escape ‘#’ as %23 in SIP URI

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

Disable Multiple m Line in SDP

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

Ring Timeout

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

Default is 60 seconds.

Delayed Call Forward Wait Time

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

Default value is 20 seconds.

No Key Entry Timeout

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

Early Dial

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

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

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

Dial Plan Prefix

Adds specified prefix to dialed number.

Use # as Dial Key

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

Disable # as Redial Key

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

Dial Plan

Dial Plan Rules:

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

2. Grammar: x – any digit from 0-9;

a. xx+ – at least 2 digits number;

b. xx – exactly 2 digits number;

c. ^ – exclude;

d. . – wildcard, matches one or more characters

e. [3-5] – any digit of 3, 4, or 5;

f. [147] – any digit 1, 4, or 7;

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

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

i. | – or

j. Flag T when adding a “T” at the end of the dial plan, the phone will wait for 3 seconds before dialing out. This gives users more flexibility on their dial plan setup. E.g., with dial plan 1XXT, phone will wait for 3 seconds to let user dial more than just 3 digits if needed. Originally the phone will dial out immediately after dialing the third digit.

Example 1: {[369]11 | 1617xxxxxxx} –

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

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

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

Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –

Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing.

1. Default: Outgoing – { x+ | +x+ | *x+ | *xx*x+ }

Example of a simple dial plan used in a Home/Office in the US:

{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }

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

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

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

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

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

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

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

SUBSCRIBE for MWI

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

Subscribe Retry Wait Time upon 403 Forbidden

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

Send Anonymous

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

Anonymous Call Rejection

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

Special Feature

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

Enable Session Timer

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

Session Expiration

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

Min-SE

Defines Minimum session expiration (in seconds).

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

Caller Request Timer

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

Default is No.

Callee Request Timer

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

Default is No.

Force Timer

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

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

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls.

UAC: The handy tone acts as the refresher.

UAS: Callee or proxy server act as the refresher.

Default is Omit.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls:

UAS: The handy tone acts as the refresher.

UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE

Uses INVITE message to refresh the session timer.

Default is No.

When To Restart Session After Re-INVITE received

Allows users to delay posting Media Change Event.

It can be set to “Immediately” or to “After replying 200OK”
The default value is “Immediately”.

Enable 100rel

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

Add Auth Header on Initial REGISTER

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

Default is No.

Conference URI

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

Use First Matching Vocoder in 200OK SDP

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

Default is No.

Preferred Vocoder

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

Voice Frames per TX

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

G723 Rate

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

iLBC Frame Size

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

Default is 20ms.

Disable OPUS Stereo in SDP

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

Default is No.

iLBC Payload type

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

Default is 97.

OPUS Payload Type

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

VAD

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

Symmetric RTP

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

Fax Mode

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

Re-IINVITE after Fax Tone Detection Mode

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

Default is Enabled

Jitter Buffer Type

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

Default is Adaptive.

Jitter Buffer Length

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

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

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

SRTP Mode

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

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

Set to  AES 128&256 bit By Default.

Crypto Life Time

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

SLIC Setting

Depends on standard phone type (and location).

Caller ID Scheme

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

DTMF Caller ID

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

Disable Unknown Caller ID

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

Replace Beginning ‘+’  in Caller ID with

Allows users to replace the + sign with a defined string instead of the predefined 00.

Number of Beginning Digits to Strip from Caller ID

Allows users to remove specified digits from incoming caller IDs, ensuring displayed numbers are concise and recognizable on connected analog phones. This helps eliminate unnecessary information and enhance caller identification.

Polarity Reversal

Reverses the polarity upon call establishment and termination.

Default is No.

Loop Current Disconnect

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

Play busy/reorder tone before Loop Current Disconnect

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

Loop Current Disconnect Duration

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

Enable Pulse Dialing

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

Pulse Dialing Standard

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

Default is General Standard.

Enable Hook Flash

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

Hook Flash Timing

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

Default values are 300 minimum and 1100 maximum.

On Hook Timing

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

Gain

Adjusts the voice path volume.

• Rx is a gain level for signals transmitted by FXS

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

Disable Line Echo Canceller (LEC)

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

Default is No.

Disable Network Echo Suppressor

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

Default is No.

Outgoing Call Duration Limit

Defines the call duration limit for the outgoing calls.

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

Incoming Call Duration Limit

Defines the call duration limit for the incoming calls.
Valid range is 0-180 minutes. Default is 0 (No limit).

Ring Frequency

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

Enable High Ring Power

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

OnHook DC Feed Current

Adjust DC feed current.

RFC2833 Events Count

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

Valid range is 2 – 10. Default is 8.

RFC2833 End Events Count

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

RFC2833 Convert Mode

Allows users set the way DTMF tones (tones from phone keypresses) are transmitted in VoIP calls using the RFC2833 Convert Mode. The available options include a default mode, a mode for analog telephone adapters, and modes involving digital signal processors. These options determine how DTMF tones are encoded, sent, and processed during calls.

The default value is "DSP Normal Mode"

Distinctive Ring Tone

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

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

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

Alert-Info: ;info=ring5

Ring tones

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

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

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

Default is c=2000/4000;

Distinctive Call Waiting Tone

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

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

Call Waiting Tones

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

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

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

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

Default is f1=440@-13,c=300/10000;

Call Features Settings

Enable Call Features

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

Reset Call Features

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

SRTP Feature

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

– Enable SRTP: Default is 16

– Disable SRTP: Default is 17

SRTP per call Feature

– Enable SRTP per call: Default is 18

– Disable SRTP per call: Default is 19

CID Feature

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

– Disable CID: Default is 30

CID per call Feature

– Enable CID per call: Default is 82

– Disable CID per call: Default is 67

Direct IP Calling Feature

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

– Direct IP Calling: Default is 47

CW Feature

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

– Disable CW: Default is 50

CW per call Feature

– Enable CW per call: Default is 71

– Disable CW per call: Default is 70

Call Return Feature

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

– Call return: Default is 69

Unconditional Forward Feature

Allow users to customize the Unconditional Forward feature codes.

– Disable Unconditional Forward: Default is 73

Busy Forward Feature

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

– Enable Busy Forward: Default is 90

– Disable Busy Forward: Default is 91

Delayed Forward Feature

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

– Enable Delayed Forward: Default is 92

– Disable Delayed Forward: Default is 93

Paging Feature

– Paging: Default is 74

DND Feature

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

– Enable DND: Default is 78

– Disable DND: Default is 79

Blind Transfer Feature

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

– Enable Blind Transfer: Default is 87

Disable LEC per call Feature

Default is Yes

– Disable LEC per call: Default is 03

Disable Bellcore Style 3-Way Conference

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

Star Code 3WC Feature

Default is Yes

– Star Code 3WC: Default is 23

Play registration id Feature

Allows users to hear their device's account or registration number.
– Enable playing registration id : Default is 98

Set Offhook Auto-Dial Feature

Allows users to configure Enable/Disable the Set Off hook auto dial Feature Code.
– Enable Set Offhook Auto-Dial : Default is 77

Forced Codec Feature

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

– Forced Codec: Default is 02

PCMU Codec Feature

Default is Yes

– PCMU Codec: Default is 7110

PCMA Codec Feature

Default is Yes

– PCMA Codec: Default is 7111

G723 Codec Feature

Default is Yes

– G723 Codec: Default is 723

G729 Codec Feature

Default is Yes

– G729 Codec: Default is 729

iLBC Codec Feature

Default is Yes

– iLBC Codec: Default is 7201

G722 Codec Feature

Default is Yes

– G722 Codec: Default is 722

Key

Call Features

*02

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

*03

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

No dial tone is played in the middle.

*16

Enable SRTP

*17

Disable SRTP

*30

Block Caller ID (for all subsequent calls)

*31

Send Caller ID (for all subsequent calls)

*47

Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle.

*50

Disable Call Waiting (for all subsequent calls)

*51

Enable Call Waiting (for all subsequent calls)

*67

Block Caller ID (per call). Dial “*67” +” number”. No dial tone is played in the middle.

*82

Send Caller ID (per call). Dial “*82” +” number”. No dial tone is played in the middle.

*69

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

*70

Disable Call Waiting (per call). Dial “*70” +” number”. No dial tone is played in the middle.

*71

Enable Call Waiting (per call). Dial “*71” +” number”. No dial tone is played in the middle

*72

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

*73

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

*74

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.

*93

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

Flash/Hook

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

#

Pressing pound sign will serve as Re-Dial key.

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