GXP16xx Series - Administration Guide

Status 🡪 Account Status

Account x

Account index.

SIP User ID

Displays the configured SIP User ID.

SIP Server

Displays the configured SIP Server address.

SIP Registration

Displays SIP registration status YES/NO.

Status 🡪 Network Status

MAC Address

Global unique ID of device, in HEX format. The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device.

IP Setting

DHCP, Static IP or PPPoE.

IPv4 Address

The IPv4 address obtained on the phone.

IPv6 Address

The IPv6 address obtained on the phone.

OpenVPN IP

The IP address of the phone obtained from the OpenVPN server

Subnet Mask

The subnet mask obtained on the phone.

Gateway

The gateway address obtained on the phone.

DNS Server 1

The DNS server address 1.

DNS Server 2

The DNS server address 2.

PPPoE Link Up

PPPoE connection status.

NAT Type

NAT Type

NAT Traversal

Display the NAT traversal status of Account 1 and Account 2.

Status 🡪 System Info

Product Model

Product model of the phone.

Part Number

Product part number.

Software Version

Core Version: core version number.

Program Version.This is the main firmware release number, which is always used for identifying the software system of the phone.

Locale Version. This is the version of localization language package.

Recovery Version.

IP Geographic Information

City: Displays the city name.

Language:Displays current language been used.

Time Zone:Displays current time zone.

Special Feature

OpenVPN Support:Display the status of the OpenVPN feature.

System Time

System up time: Displays system up time since the last reboot.

System time:Displays current system time on the phone system.

Service Status

GUI and Phone service status.

System Information

Download System Information.

Core Dump

Core dump file that could be downloaded for troubleshooting purpose.

Status 🡪 GNU GPL License

GNU GPL License

Shows the GNU GPL License information.

Account x 🡪 General Settings

Account Active

This field indicates whether the account is active. The default setting is “Yes”.

Account Name

The name associated with each account to be displayed on the LCD.

SIP Server

The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (ITSP).

Secondary SIP Server

The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails.

Outbound Proxy

IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session Border Controller. It is used by the phone for Firewall or NAT penetration in different network environments. If a symmetric NAT is detected, STUN will not work and ONLY an Outbound Proxy can provide a solution.

Backup Outbound Proxy

Secondary outbound proxy which will be used when the primary proxy cannot be connected.

BLF Server

The URL or IP address, and port of the BLF Server. When “BLF Server” is configured, the SUBSCRIBE request will be sent to this address for BLF.

SIP User ID

User account information provided by your VoIP service provider (ITSP). It is usually in the form of digits similar to phone number or actually a phone number.

Users can also register an account with a SIP user ID that carries “@”. For example, you can put “111@test.com” in your SIP User ID field. When registering, the phone will register your account as “111@test.com” instead of 111.

Note: The server domain will not be included in the SIP from header.

Authentication ID

SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from the SIP User ID. This field supports up to 8192 characters.

Authentication Password

The account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered. After it is saved, this will appear as hidden for security purpose.

Name

The SIP server subscriber’s name (optional) that will be used for Caller ID display.

Voice Mail Access Number

This parameter allows you to access voice messages by pressing the MESSAGE button on the phone.

This ID is usually the VM portal access number. For example, in Asterisk server, 8500 could be used.

Account Display

This option allows you to configure how your SIP account label will be displayed on the phone’s screen. If set to “Username”, LCD account label will display the Account Name configured for this SIP account. If set to “User ID”, it will then display the SIP User ID configured for this SIP account.

Account x 🡪 Network Settings

DNS Mode

This parameter controls how the Search Appliance looks up IP addresses for hostnames. There are four modes: A Record, SRV, NATPTR/SRV, and Use Configured IP. The default setting is “A Record”. If the user wishes to locate the server by DNS SRV, the user may select “SRV” or “NATPTR/SRV”. If “Use Configured IP” is selected, please fill in the three fields below:

Primary IP: Primary IP address where the phone sends DNS query to;

Backup IP 1;

Backup IP 2.

If SIP server is configured as domain name, phone will not send DNS query, but use “Primary IP” or “Backup IP x” to send SIP message if at least one of them are not empty. Phone will try to use “Primary IP” first. After 3 tries without any response, it will switch to “Backup IP x”, and then it will switch back to “Primary IP” after 3 re-tries. If SIP server is already an IP address, phone will use it directly even “User Configured IP” is selected.

NAT Traversal

This parameter configures whether the NAT traversal mechanism is activated. Users could select the mechanism from No, STUN, Keep-Alive, UPnP, Auto or VPN. If set to “STUN” and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages. The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be “Keep-Alive”. Configure this to be “No” if an outbound proxy is used. “STUN” cannot be used if the detected NAT is symmetric NAT.

Support Rport (RFC 3581)

This option enables the utilization of symmetric response routing. When enabled, the "rport" field gets appended to the Via header in the SIP Request. The necessary data is then extracted from the SIP 200OK Response during a SIP Register operation, allowing the rewriting of SIP Contact information and its application in subsequent SIP Requests.
Enabled by Default.

Proxy-Require

A SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall. Do not configure this parameter unless this feature is supported on the SIP server.

Use SBC

Indicate whether an SBC server is used.

Account x 🡪 SIP Settings 🡪 Basic Settings

TEL URI

If the phone has an assigned PSTN telephone number, this field should be set to “User=Phone”. Then a “User=Phone” parameter will be attached to the Request-Line and “TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in the SIP request. The default setting is “Disable”.

SIP Registration

Selects whether the phone will send SIP Register messages to the proxy/server. The default setting is “Yes”.

Unregister On Reboot

Allows the SIP user’s registration information to be cleared when the phone reboots. The SIP REGISTER message will contain “Expires: 0” to unbind the connection. Three options are available:

If set to “All“, the SIP user’s registration information will be cleared when the phone reboots. The SIP Contact header will contain “*” to notify the server to unbind the connection.

If set to “Instance“, the SIP user will be unregistered on current phone only.

If set to “No”, the phone will not unregister the SIP account when rebooting.

The default setting is “No”.

Register Expiration

Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar. The default value is 60 minutes. The maximum value is 64800 minutes (about 45 days).

Subscribe Expiration

Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified registrar. The default is 60, and maximum value is 64800 (about 45 days).

Reregister Before Expiration

Specifies the time frequency (in seconds) that the phone sends re-registration request before the Register Expiration. The default value is 0.

Enable OPTIONS Keep Alive

Select whether the phone will keep sending a message to check the connection with the server.

OPTIONS Keep Alive Interval

Specifies the frequency (in second) in which the phone will send the Keep Alive message to the server.

OPTIONS Keep Alive Max Lost

Specifies the maximum number of allowed lost packet before the phone will refresh its registration.

Local SIP Port

Defines the local SIP port used to listen and transmit. The default value is 5060 for Account 1, 5062 for Account 2, and 5064 for Account 3.

SIP Registration Failure Retry Wait Time

Specifies the interval to retry registration if the process is failed. The default value is 20 seconds.

SIP T1 Timeout

SIP T1 Timeout is an estimate of the round-trip time of transactions between a client and server. If no response is received the timeout is increased and request re-transmit retries would continue until a maximum amount of time define by T2. The default setting is 0.5 seconds.

SIP T2 Timeout

SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. The default setting is 4 seconds.

SIP Transport

Determines the network protocol used for the SIP transport. Users can choose from TCP, UDP and TLS.

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

This option is used to control 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.

Outbound Proxy Mode

Configures the outbound proxy mode, to place in route header in sending SIP messages, or always sent to outbound proxy. Default setting is “in route”

Support SIP Instance ID

Defines whether SIP Instance ID is supported or not.

Default setting is “Yes”.

SUBSCRIBE for MWI

When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically. The phone supports synchronized and non-synchronized MWI.

The default setting is “No”.

SUBSCRIBE for Registration

When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically. The default setting is “No”.

Enable 100rel

The use of the PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is very important in order to support PSTN internetworking. To invoke a reliable provisional response, the 100rel tag is appended to the value of the required header of the initial signaling messages.

Callee ID Display

The phone supports Callee ID display as follows:

When set to “Auto” (Default), the phone will update the callee ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and To Header in the 180 Ringing.

When set to “Disabled“, callee id will be displayed as “Unavailable”.

When set to “To Header“, caller id will not be updated and displayed as To Header.

Caller ID Display

The phone supports Caller ID display as follows:

When set to “Auto” (Default), the phone 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.

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

When set to “From Header”, the phone will display the caller ID based on the From Header in the incoming SIP INVITE.

Note: During call, if the phone receives NOTIFY message with type header:”application/cid-info+xml” and event header:”cid-info-update”, the phone will take the message and update CID information and CID image.

Ignore Alert-Info Header

This option is used to configure default ringtone. If set to “Yes”, configured default ringtone will be played even if Alert-info header is present.

The default setting is “No”.

Add Auth Header On Initial REGISTER

When enabled, Device will send Authentication Header on the first Register instead of second Register.

Default is No.

Account x 🡪 SIP Settings 🡪 Custom SIP Headers

Use Privacy Header

Controls whether the Privacy header will present in the SIP INVITE message or not, whether the header contains the caller info. When set to “Default”. If set to “Yes”, the Privacy Header will always show in INVITE. If set to “No”, the Privacy Header will not show in INVITE. Default setting is “Default”.

Use P-Preferred-Identity Header

Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message or not. The default setting is “default”. If set to “Yes”, the P-Preferred-Identity Header will always show in INVITE. If set to “No”, the P-Preferred-Identity Header will not show in INVITE.

Use P-Access Network-Info Header

Enables/disables the use of P-Access-Network-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.

Use P-Emergency-Info Header

Enables/disables the use of P-Emergency-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.

Use MAC Header

If Yes except REGISTER, the sip message for register or unregister will contains MAC address in the header, and all the outgoing SIP messages except REGISTER message will attach the MAC address to the User-Agent header; If Yes to ALL, the sip message for register or unregister will contains MAC address in the header, and all the outgoing SIP message including REGISTER will attach the MAC address to the User-Agent header; If No, neither will the MAC header be included in the register or unregister message nor the MAC address be attached to the User-Agent header for any outgoing SIP message.

Account x 🡪 SIP Settings 🡪 Advanced Features

Line-seize Timeout

For Shared Call Appearance, phone must send a SUBSCRIBE-request for the line-seize event package whenever a user attempts to take the shared line off hook. “Line Seize Timeout” is the line-seize event expiration timer. The default value is 15 seconds. The valid range is from 15 to 60.

Eventlist BLF URI

Configures the Eventlist BLF URI on the phone to monitor the extensions in the list with Multi-Purpose Key. If the server supports this feature, users need to configure an Eventlist BLF URI on the service side first (i.e., BLF1006@myserver.com) with a list of extensions included. On the phone, in this “Eventlist BLF URI” field, fill in the URI without the domain (i.e., BLF1006). To monitor the extensions in the list, under Web GUI 🡪Settings🡪Programmable Keys page, please select “Eventlist BLF” in the key mode, choose account, enter the value of each extension in the list. (Supported only on GXP1628/GXP1630)

Auto Provision Eventlist BLFs

When enabled, empty multi-purpose keys will be automatically provisioned to the monitored extensions in the Eventlist BLF. Default setting is “Disabled”. (Supported only on GXP1628/GXP1630)

Conference URI

Configures the conference URI when using Broadsoft N-way calling feature.

Music On Hold URI

Music On Hold URI to call when a call is on hold if server supports it.

BLF Call-pickup Prefix

Configures the prefix prepended to the BLF extension when the phone picks up a call with BLF key. (Supported only on GXP1628/GXP1630)

Call Pickup Barge-in Code

Set feature code of call pickup barge-in feature.

(Supported only on GXP1628/GXP1630)

PUBLISH for Presence

Enables Presence feature on the phone.

(Supported only on GXP1628/GXP1630)

Enable User Presence Subscription

If the user presence subscription is enabled, the phone will subscribe and notify about user presence to the sip server.

Note: To enable this feature, register Genesys account on your phone’s SIP account first. The SIP server should support this feature.

Hide Login Soft Key On Auto User Presence Subscription

This option allows you to hide the soft key to login or logout on idle screen if auto user ID and password are configured.

Auto User Presence ID

With this ID, the phone will subscribe user presence automatically once the account is registered.

Auto User Presence password

With this password, the phone will subscribe user presence automatically once the account is registered.

Special Feature

Different soft switch vendors have special requirements. Therefore, users may need select special features to meet these requirements. Users can choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, PhonePower and iPageOn depending on the server type. Default is “Standard”.

Broadsoft

Broadsoft Call Center

Default setting is “Disabled”. When set to “Enabled”, a soft key “BSCCenter” is displayed on LCD. User can access different Broadsoft Call Center agent features via this softkey. Please note that “Feature Key Synchronization” will be enabled regardless of this setting.

Hoteling Event

With “Hoteling Event” enabled, user can access the Hoteling feature option by pressing the “BSCCenter” softkey. Default setting is “No”.

Call Center Status

When set to “Yes”, the phone will send SUBSCRIBE to the server to obtain call center status. The default setting is “No”.

Feature Key Synchronization

This feature is used for Broadsoft call feature synchronization. When it is enabled, DND, Call Forward features and Call Center Agent status can be synchronized between Broadsoft server and phone. Default is “Disabled”.

Broadsoft Call Park

When enabled, phone will send the SUBSCRIBE to Broadsoft server to obtain Broadsoft Call Park notification.

Notes: The phone shows parked call notification on its LCD.

If parked member is a monitored extension on BLF MPK key, the MPK will be flashing red.

Account x 🡪 SIP Settings 🡪 Session Timer

Enable Session Timer

This option is used to enable or disable session timer on the phone side when server side can provide both session timer UPDATE or session audit UPDATE. The default setting is “Yes”.

Session Expiration

The SIP Session Timer extension (in seconds) that enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE).

If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires.

Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds.

The valid range is from 90 to 64800.

Caller Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it makes outbound calls.

Callee Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it receives inbound calls.

Force Timer

If Force Timer is set to “Yes”, the phone will use the session timer even if the remote party does not support this feature.

If Force Timer is set to “No”, the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select “No”.

UAC Specify Refresher

As a Caller, select UAC to use the phone as the refresher; or select UAS to use the Callee or proxy server as the refresher.

UAS Specify Refresher

As a Callee, select UAC to use caller or proxy server as the refresher; or select UAS to use the phone as the refresher.

Force INVITE

The Session Timer can be refreshed using the INVITE method or the UPDATE method. Select “Yes” to use the INVITE method to refresh the session timer. The default setting is “No”.

Account x 🡪 SIP Settings 🡪 Security Settings 🡪 Security

Check Domain Certificates

Choose whether the domain certificates will be checked or not when TLS/TCP is used for SIP Transport. The default setting is “No”.

Validate Certification Chain

Choose whether certification chain will be validated when TLS/TCP mode is used.

Validate Incoming Messages

Choose whether the incoming messages will be validated or not. The default setting is “No”.

Check SIP User ID for incoming INVITE

If set to “Yes”, SIP User ID will be checked in the Request URI of the incoming INVITE. If it does not match the phone’s SIP User ID, the call will be rejected. The default setting is “No”.

Accept Incoming SIP from Proxy Only

When set to “Yes”, the SIP address of the Request URL in the incoming SIP message will be checked. If it does not match the SIP server address of the account, the call will be rejected. The default setting is “No”.

Authenticate Incoming INVITE

If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response. Default setting is “No”.

Account x 🡪 Audio Settings

Send DTMF

This parameter specifies the mechanism to transmit DTMF digits.

There are 3 supported modes:

In audio, which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs).

RFC2833, which means to specify DTMF with RTP packet. Users could know the packet is DTMF in the RTP header as well as the type of DTMF.

SIP INFO, which use SIP info to carry DTMF. The defect of this mode is that it is easily to cause desynchronized of DTMF and media packet for the reason the SIP and RTP are transmitted respectively. Default setting is “RFC2833”

DTMF Payload Type

Configures the payload type for DTMF using RFC2833. Default value is 101.

Preferred Vocoder

7 different vocoder types are supported on the phone, including G.711 U-law (PCMU), G.711 A-law (PCMA), G.723.1, G.729A/B, G.722 (wide band), iLBC, and G726-32. Users can configure vocoders in a preference list that is included with the same preference order in SDP message.

Use First Matching Vocoder in 200OK SDP

When set to “Yes”, the device will use the first matching vocoder in the received 200OK SDP as the codec.

The default setting is “No”.

Codec Negotiation Priority

This option allows user to choose which Codec sequence to use.

If set to “Caller”, the phone will negotiate the codecs based on the SDP from the received SIP INVITE. If set to “Callee”, the phone will negotiate the audio codecs based on the codec priority on the phone.

Disable Multiple m line in SDP

If enabled, the phone always responses 1 m line in SDP regardless multiple m lines are offered.

SRTP Mode

Enables the SRTP mode based on your selection.

Default setting is “Disabled”.

Symmetric RTP

Defines whether symmetric RTP is supported or not.

Default setting is “No”.

Silence Suppression

Controls the silence suppression/VAD feature of the audio codecs except for

G.723 (pending) and G.729. If set to “Yes”, a small quantity of RTP packets containing comfort noise will be sent during the periods of silence. If set to “No”, this feature is disabled. Default setting is “No”

Voice Frames Per TX

Configures the number of voice frames transmitted per packet.

When configuring this, it should be noted that the “ptime” value for the SDP will change with different configurations here. This value is related to the codec used and the actual frames transmitted in payload during the call. For end users, it is recommended to use the default setting, as incorrect settings may influence the audio quality. The default setting is 2.

G723 Rate

This option determines the encoding rate for G723 codec. Users can choose from 6.3kbps encoding rate and 5.3kbps encoding rate. The default setting is “5.3kbps encoding rate”.

G.726-32 Packing Mode

Selects “ITU” or “IETF” for G.726-32 packing mode.

The default setting is “IETF”.

iLBC Frame Size

This option determines the iLBC packet frame size. Users can choose from 20ms and 30ms. The default setting is “30ms”.

iLBC payload Type

This option is used to specify iLBC payload type. Valid type is 96-127.

Jitter Buffer Type

Selects either Fixed or Adaptive based on network conditions. The default setting is “Adaptive”.

Jitter Buffer Length

Defines jitter buffer length based on network conditions.

Hide Vocoder

Hides vocoder information on call screen if set to “yes”. Default settings is “No”.

Account x 🡪 Call Settings

Early Dial

Selects whether to enable early dial. If it is set to “Yes”, the SIP proxy must support 484 responses. The default setting is “No”.

Dial Plan Prefix

Sets the prefix added to each dialed number.

Dial Plan

A dial plan establishes the expected number and pattern of digits for a telephone number. This parameter configures the allowed dial plan for the phone.

Dial Plan Rules:

1. Accepted 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;

3. Grammar: X – any character from 0-9, a-z, A-Z.

4. xx+ – at least 2-digit numbers

5. xx – only 2-digit numbers

6. XX – two characters ( AA, Ab, 1C, f5, 68,…)

7. test : only string “test” will pass the dial plan check

8. ^ – exclude

9. [3-5] – any digit of 3, 4, or 5

10. [147] – any digit of 1, 4, or 7

11. <2=011> – replace digit 2 with 011 when dialing

12. | – the OR operand

Example 1: {[369]11 | 1617xxxxxxx}

Allow 311, 611, and 911 or any 11-digit numbers with leading digits 1617;

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

Block any number of leading digits 1900 or add prefix 1617 for any dialed 7-digit numbers;

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

Allows any number with leading digit 1 followed by a 3-digit number, followed by any number between 2 and 9, followed by any 7-digit number OR Allows any length of numbers with leading digit 2, replacing the 2 with 011 when dialed.

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 dialing7 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 – allows dialing special and emergency numbers 311, 411, 611 and 911.

Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.

Max length of dial plan is up to 1024 characters.

Bypass Dial Plan

Bypass the dial plan when dialing from one of the available items:

Contacts ; Call History Incoming Call ; Call History Outgoing Call ; Dialing Page ; MPK ; API

The default settings bypass the dial plan for the MPK only.

Call Log

Configures Call Log setting on the phone. You can log all calls, only log incoming/outgoing calls or disable call log. The default setting is “Log All Calls”.

Send Anonymous

If set to “Yes”, the “From” header in outgoing INVITE messages will be set to anonymous, blocking the Caller ID to be displayed.

Anonymous Call Rejection

If set to “Yes”, anonymous calls will be rejected. The default setting is “No”.

Auto Answer

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep.

Auto Answer Numbers

The function allows users to have the phone configured with a pre-defined list of numbers that will perform auto answer.

For “Auto Answer Numbers”, it accepts:

Digits :1,2,3,4,5,6,7,8,9; x – any digit from 0-9; xx – any two digits from 0-9;

[1-5] – any digit from 1 to 5; +: it matches the previous character as many times as needed like regular expression.

Auto Answer Numbers can be split with “;”, for example: 1x;2xxx;3x+

Refer-To Use Target Contact

If set to “Yes”, the “Refer-To” header uses the transferred target’s Contact header information for attended transfer. The default setting is “No”.

Transfer on Conference Hangup

Defines whether the call is transferred to the other party if the initiator of the conference hangs up.

The default setting is “No”.

Disable Recovery on Blind Transfer

Defines whether to enable/disable recovery to the call to the transferee on failing blind transfer to the target.

Blind Transfer Wait Timeout

Defines the timeout (in seconds) for waiting SIP frag response in blind transfer. Valid range is 30 to 300.

No Key Entry Timeout

Defines the timeout (in seconds) for no key entry.

If no key is pressed after the timeout, the digits will be sent out. The default value is 4 seconds.

Key As Send

Allows users to configure the “#” key as the “Send” key. If set to “Yes”, the “#” key will immediately dial out the input digits.

In this case, this key is equivalent to the “Send” key. If set to “No”, the “#” key is included as part of the dialing string.

On Hold Reminder Tone

If enabled, the user will hear 3 beeps as hold reminder, the first on hold reminder occurs at 1 min and the subsequent on hold reminder occurs every 30 sec. Default is “Disabled”.

Call Recording On

Configures the DTMF sequence sent when pressing the Record key during a call on this account.

Call Recording Off

Configures the DTMF sequence sent when pressing the Record key during a call on this account when turning recording off.

Hide Dialing Password

Allows user to hide the password when the dialing number matches the configured prefix.

Remote Info Display

Choose which caller information to display once receiving a call:

Name and Number: Display the caller ID name and number.

Number: Display only the caller number.

Name: Display only the caller ID name.

Hide Remote User ID

Settings to allow user to hide and display name of remote user ID that has a length more than specific in the incoming and/or outgoing calls

Ring Tone

Account Ring Tone

Configures ring tone for the account.

Match Incoming Caller ID

Specifies matching rules with number, pattern or Alert Info text. (Up to 10 rules). When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone. Matching rules:

Specific caller ID number. For example, 8321123;

A defined pattern with certain length using x and + to specify, where x could be any digit from 0 to 9. Samples:

xx+ : at least 2-digit number;

xx : only 2-digit number;

[345]xx: 3-digit number with the leading digit of 3, 4 or 5;

[6-9]xx: 3-digit number with the leading digit from 6 to 9.

Alert Info text

Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info:

Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring.

Ring Timeout

Defines the timeout (in seconds) for the rings on no answer.

Default is 60.

Ringback Tone at No Early Media

Account x 🡪 Intercom Settings

Allow Auto Answer by Call-Info/Alert-Info

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP Call-Info/Alert-Info header sent from the server/proxy.

Mute on answer Intercom call

When enabled, phone will mute the incoming intercom call by Call-Info/Alert-Info

Play warning tone for Auto Answer Intercom

When enabled, phone will play warning tone when auto answer Intercom

Custom Alert-Info for Auto Answer

Used exclusively to match the contents of the Alert-Info header for auto answer.

The default auto answer headers will not be matched if this is defined.

Account x 🡪 Feature Codes

Enable Local Call Features

When enabled, Do Not Disturb, Call Forwarding and other call features can be used via the local feature codes on the phone. Otherwise, the provisioned feature codes from the server will be used. User configured feature codes will be used only if server provisioned feature codes are not provided.

Do Not Disturb (DND)—On

Configures DND feature code to turn on DND.

Do Not Disturb

(DND)—Off

Configures DND feature code to turn off DND.

Call Forward

Unconditionally(All)—On

Configures Call Forward All feature code to activate unconditional call forwarding.

Call Forward

Unconditionally(All)—Off

Configures Call Forward All feature code to deactivate unconditional call

Forwarding.

Target

Configures the extension that the call will be forwarded to.

Call Forward Busy— ON

Configures Call Forward Busy feature code to activate busy call forwarding.

Call Forward Busy— Off

Configures Call Forward Busy feature code to deactivate busy call forwarding.

Target

Configures the extension that the call will be forwarded to.

Call Forward Delayed

(No Answer)—On

Configures Call Forward Delayed feature code to activate no answer call forwarding.

Call Forward Delayed

(No Answer)—Off

Configures Call Forward Delayed feature code to deactivate no answer call forwarding.

Target

Configures the extension that the call will be forwarded to.

Delayed Call Forward Wait Time

Defines the timeout (in seconds) before the call is forwarded on no answer.

Valid range is 1 to 120. Default is 20

Network 🡪 Basic Settings

Internet Protocol

Selects one of the following IP protocol modes:

IPv4 Only: Enforce IPv4 protocol only.

IPv6 Only: Enforce IPv6 protocol only.

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

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

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

IPv4 Address

Allows users to configure the appropriate network settings on the phone to obtain IPv4 address. Users could select “DHCP”, “Static IP” or “PPPoE”.

By default, it is set to “DHCP”.

Host name (Option 12)

Specifies the name of the client, added for the DHCP INFORM using Option 12. This field is optional but may be required by some Internet Service Providers.

Vendor Class ID (Option 60)

Used by clients and servers to exchange vendor class ID.

PPPoE Account ID

Enter the PPPoE account ID.

PPPoE Password

Enter the PPPoE Password.

PPPoE Service Name

Enter the PPPoE Service Name.

IPv4 Address

Enter the IP address when static IP is used.

Subnet Mask

Enter the Subnet Mask when static IP is used for IPv4.

Gateway

Enter the Default Gateway when static IP is used for IPv4.

DNS Server 1

Enter the DNS Server 1 when static IP is used for IPv4.

DNS Server 2

Enter the DNS Server 2 when static IP is used for IPv4.

Preferred DNS Server

Enter the Preferred DNS Server for IPv4.

IPv6 Address

Allows users to configure the appropriate network settings on the phone to obtain IPv6 address.

Users could select “Auto-configured” or “Statically configured” for the IPv6 address type.

Static IPv6 Address

Enter the static IPv6 address when Full Static is used in “Statically configured” IPv6 address type.

IPv6 Prefix Length

Enter the IPv6 prefix length when Full Static is used in “Statically configured” IPv6 address type.

IPv6 Prefix (64 bits)

Enter the IPv6 Prefix (64 bits) when Prefix Static is used in “Statically configured” IPv6 address type.

DNS Server 1

Enter the DNS Server 1 for IPv6.

DNS Server 2

Enter the DNS Server 2 for IPv6.

Preferred DNS server

Enter the Preferred DNS Server for IPv6.

Network 🡪 Advanced Settings

802.1X mode

Allows the user to enable/disable 802.1X mode on the phone. The default value is disabled. It can be set to EAP-MD5, EAP-TLS or EAP-PEAPv0/MSCHAPv2.

Note: GXP16xx has the ability to send an EAPOL-Logoff message when a connected PC is disconnected from the network.

EAPOL-Logoff stands for “Extensible Authentication Protocol over LAN Logoff.” It’s a network protocol message used to signal the termination or logoff of a device from a network that uses the Extensible Authentication Protocol (EAP) for authentication and security purposes.

802.1X Identity

Enter the Identity for the 802.1X mode.

MD5 Password

Enter the MD5 Password for the 802.1X mode.

Note: the use of special characters is supported.

802.1X CA Certificate

Upload 802.1X CA certificate to the phone; or delete existed 802.1X CA certificate from the phone.

802.1X Client Certificate

Upload 802.1X Client certificate to the phone; or delete existed 802.1X Client certificate from the phone.

HTTP Proxy

Specifies the HTTP proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.

HTTPS Proxy

Specifies the HTTPS proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.

Layer 3 QoS for SIP

Defines the Layer 3 QoS parameter for SIP. This value is used for IP Precedence, Diff-Serv or MPLS. The default value is 26.

Layer 3 QoS for RTP

Defines the Layer 3 QoS parameter for RTP. This value is used for IP Precedence, Diff-Serv or MPLS. The default value is 46.

Enable DHCP VLAN

Enables VLAN settings auto-configuration using DHCP options 132 & 133 tunneled through DHCP Option 43.

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.

Layer 2 QoS 802.1p Priority Value

Assigns the priority value of the Layer2 QoS packets. The default value is 0.

PC Port Mode

Configures the PC port mode. When set to “Mirrored”, the traffic in the LAN port will go through PC port as well and packets can be captured by connecting a PC to the PC port. The default setting is “Enable”.

PC Port VLAN Tag

Assigns the VLAN Tag of the PC port.

PC Port Priority Value

Assigns the priority value of the PC port.

Enable CDP

Allows users to enable/disable CDP feature which will allow the phone to receive CDP packets from the uplink. Default is Enabled.

Note: When CDP feature is enabled, make sure VLAN is also configured accordingly. It will need reboot to take effect for both enabling/disabling.

Enable LLDP

Users can Enable/Disable the LLDP (Link Layer Discovery Protocol) service to advertise information about the phone to other devices on the network. Default is Enabled.

Maximum Transmission Unit (MTU)

Defines the Maximum Transmission Unit value (MTU) in bytes.

Default is 1500.

Network 🡪 Remote Control

CSTA Control

Indicates whether CSTA Control feature is enabled. Change of this configuration will need the system reboot to make it take effect.

Network 🡪 Open VPN® Settings

OpenVPN® Enable

Enables/Disables the OpenVPN® feature. Default settings is No.

OpenVPN® Server Address

Configures the address of the OpenVPN® server.

OpenVPN® Port

Defines the port of the OpenVPN® server. Default is 1194.

OpenVPN® Transport

Determines network protocol UDP or TCP used for OpenVPN®.

Default is UDP.

OpenVPN® CA

Uploads the OpenVPN® CA.

OpenVPN® Certificate

Uploads the OpenVPN® Certificate.

OpenVPN® Client Key

Uploads the OpenVPN® Client Key.

OpenVPN® Client Key Password

Sets the password for the client.key file.

OpenVPN® TLS Auth Key

The OpenVPN® TLS Auth Key is used to add an additional layer of security to the connection by requiring a shared secret between the client and the server. This key is used during the TLS handshake phase to authenticate the parties involved and establish a secure communication channel.  you can upload the tls-auth.key file by clicking "upload".

OpenVPN® Cipher Method

Must be the same cipher method used by the OpenVPN® server

OpenVPN® Username

OpenVPN® authentication username (optional)

OpenVPN® Password

OpenVPN® authentication password (optional)

Network 🡪 SNMP Settings

Enable SNMP

Enable/Disable SNMP service. Default is No.

Version

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

Port

Listening Port of SNMP daemon (Default 161).

SNMP Trap IP

IP address of trap destination.

SNMP Trap port

Port of Trap destination (Default 162)

SNMP Trap Interval

Time interval between traps (Default is 5).

SNMP Trap Version

Choose between (Version 1, Version 2).

Community

Name of SNMPv1/v2c community.

SNMP Trap Community

Name of SNMPv1/v2c trap community.

SNMP Username

Username for SNMPv3.

Security Level

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

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

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

Authentication Protocol

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

Privacy Protocol

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

Authentication Key

Enter the Authentication Key.

Privacy Key

Enter the Privacy Key.

SNMP Trap Username

Username for SNMPv3 Trap.

Trap Security Level

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

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

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

Trap Authentication Protocol

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

Trap Privacy Protocol

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

Trap Authentication Key

Enter the Trap Authentication Key.

Trap Privacy Key

Enter the Trap Privacy Key.

Maintenance 🡪 Web Access

Enable User Web Access

Administrator can disable or enable user web access. This option is disabled by default.

New Password

Allows the administrator to set the password for user-level web GUI access. This field is case sensitive with a maximum length of 30 characters.

Confirm Password

Confirms the end user password field to be the same as above.

Current Password

Determines the current password which is required to set a new admin password.

New Password

Allows users to change the admin password. The password field is purposely hidden for security purpose. This field is case sensitive with a maximum length of 30 characters.

Confirm Password

Confirms the admin password field to be the same as above.

Maintenance 🡪 Upgrade and Provisioning

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, Always Skip the Firmware Check.

Always Authenticate Before Challenge

Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before being challenged by the server.

Validate Hostname in Certificate

To validate the hostname in the SSL certificate.

Allow DHCP Option 43 and Option 66 to Override Server

Default setting is “Yes”. DHCP option 66 originally was only designed for TFTP server. And then was extended to support an HTTP URL. GXP phones support both TFTP and HTTP server via option 66. Users can also use DHCP option 43 vendor specific option to do this. DHCP option 43 approach has priorities.

Note:

Users should include http:// on the IP address or the URL of DHCP option 66 on the server if they want to use an http server. If the URL/IP is without a scheme (protocol) the phone adds tftp:// by default. If the URL/IP includes tftp://, the phone will not add tftp:// and use the URL/IP as it is.

Additional Override DHCP option

Additional DHCP Option that will be used as a firmware server instead of the setting one or name server from option 43 and 66. However, this option will be effective only when option ‘Allow DHCP Option 43 and Option 66 to Override Server’ is enabled, the options are 150 and 160.

Allow DHCP Option 120 to override SIP Server

Enables DHCP Option 120 from local server to override the SIP Server on the phone. The default setting is “No”.

3CX Auto Provision

Enables automatic provisioning using PnP. If enabled, the phone will start sending a SIP SUBSCRIBE message to a multicast IP address 224.0.1.75 at boot up.

The phone should receive response message 200 or 500 from the server, followed by a SIP NOTIFY including URL from where to download configuration file to get provisioned. The default setting is “Yes” (Enabled).

Automatic Upgrade

Enables automatic upgrade and provisioning.

The default setting is “No”.

Hour of the Day (0-23)

Defines the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes. The default value is 1.

Day of the Week (0-6)

Defines the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes. The default value is 1.

Disable SIP NOTIFY Authentication

Device will not challenge NOTIFY with 401 when set to Yes.

Firmware Upgrade Confirmation

Allow users to automatically upgrade to the version that the server has provided without need to confirm the upgrade to avoid end users manually select no when doing mass upgrade. When set to “Yes”, the phone will ask the user to upgrade. Default setting is “Yes”.

Provision

Click Start to trigger the Provision request.

Config Upgrade Via

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

Config Server Path

Defines the server path for provisioning. It could be different from the firmware server for upgrading.

Support added to configure variables in the provisioning server URL. Currently, the following variables are supported in the provisioning server URL:

• $PN: it is used to identify the directory name of the provisioning server directory where the corresponding boot files and configuration files are located.

• $MAC: it is used to identify the MAC address of the IP phone.

Variables $PN and $MAC can be embedded in server URL setting in Web UI and also in DHCP Option 66. Example (Web UI): /192.168.0.2/$PN_$MAC Example (DHCP Option 66): tftp://192.168.0.2/$PN_$MAC $PN will be replaced with phone model, e.g., GXP1625 $MAC will be replaced with phone’s MAC address, e.g., 000b829a8ffe

Config HTTP/HTTPS Username

The username for the HTTP/HTTPS server (configuration file).

Config HTTP/HTTPS Password

The password for the HTTP/HTTPS server (configuration file).

Config File Prefix

This field enables user to store different configuration files in one single directory on the configuration server. If configured, only the configuration file with the matching prefix will be downloaded.

Config File Postfix

This field enables user to store different configuration files in one single directory on the configuration server. If configured, only the configuration file with the matching postfix will be downloaded.

XML Config File Password

The password for encrypt/decrypt the XML configuration file using OpenSSL.

Authentication Conf File

Sets the phone system to authenticate configuration file before applying it. When set to “Yes”, the configuration file must include value P1 with phone system’s administration password. If it is missed or does not match the password, the phone system will not apply it. Default setting is “No”.

Download Device Configuration

User can press the “Download” button to export the device’s configuration file. The configuration file is named as “config.txt”.

Upload Device Configuration

User can press the “Upload” button to upload configuration files to the phone.

Firmware Upgrade Via

Allows users to choose the firmware upgrade method: TFTP, HTTP, FTP/FTPS or HTTPS for firmware file download.

Firmware Server Path

Defines the server path for the firmware server. It could be different from the configuration server for provisioning.

Support added to configure variables in the provisioning server URL. Currently, the following variables are supported in the provisioning server URL:

• $PN: it is used to identify the directory name of the provisioning server directory where the corresponding boot files and configuration files are located.

• $MAC: it is used to identify the MAC address of the IP phone.

Variables $PN and $MAC can be embedded in server URL setting in Web UI and also in DHCP Option 66. Example (Web UI): /192.168.0.2/$PN_$MAC Example (DHCP Option 66): tftp://192.168.0.2/$PN_$MAC $PN will be replaced with phone model, e.g., GXP1625 $MAC will be replaced with phone’s MAC address, e.g., 000b829a8ffe

Firmware HTTP/HTTPS Username

The username for the HTTP/HTTPS server (Firmware file).

Firmware HTTP/HTTPS Password

The password for the HTTP/HTTPS server (Firmware file).

Firmware File Prefix

This field enables user to store different firmware files in one single directory on the firmware server. If configured, only the firmware file with the matching prefix will be downloaded.

Firmware File Postfix

This field enables user to store different firmware files in one single directory on the firmware server.

If configured, only the firmware file with the matching postfix will be downloaded.

Maintenance 🡪 Syslog

Syslog Protocol

Choose whether to send syslog messages over UDP or secured SSL/TLS connection.

Default is UDP.

Notes:

Syslog Server

The URL or IP address of the syslog server for the phone to send syslog to.

Note: By adding port number to the Syslog server field (i.e 172.18.1.1:1000), the phone will send syslog to the corresponding port of that IP.

Syslog Level

Selects the level of logging for syslog.

The default setting is “None”.

There are 4 levels: DEBUG, INFO, WARNING and ERROR.

Syslog messages are sent based on the following events:

Note: Syslog messages also includes MAC address and firmware version.

Syslog Keyword Filtering

Configures syslog message to be filtered based on keywords provided.

If multiple keywords are filled, it should be separated by “,”.

Note: No spaces allowed.

Send SIP Log

Configures whether the SIP log will be included in the syslog messages or not. The default setting is “No”.

Maintenance 🡪 Language

Display Language

Selects display language on the phone.

Language File Postfix

Specifies the language file postfix for downloaded language.

Maintenance 🡪 Action URL

Setup Completed

Specifies action URL for signal to be sending out when phone finished setting up processes.

Registered

Specifies action URL for signal to be sending out when phone successfully registered.

Unregistered

Specifies action URL for signal to be sending out when phone successfully unregistered SIP account.

Register Failed

Specifies action URL for signal to be sending out when the IP phone fails to register a SIP account.

Off Hook

Specifies action URL for signal to be sending out when phone is in off-hook state.

On Hook

Specifies action URL for signal to be sending out when phone is in on-hook state.

Incoming Call

Specifies action URL for signal to be sending out when phone received the incoming call.

Outgoing Call

Specifies action URL for signal to be sending out when phone make an outbound call.

Missed Call

Specifies action URL for signal to be sending out when phone missed a call.

Answered Call

Specifies action URL for signal to be sending out when phone answered a call.

Rejected Call

Specifies action URL for signal to be sending out when phone rejected a call.

Forwarded Call

Specifies action URL for signal to be sending out when phone forwarded a call.

Established Call

Specifies action URL for signal to be sending out when phone established a call.

Terminated Call

Specifies action URL for signal to be sending out when phone terminated a call.

Idle to Busy

Specifies action URL for signal to be sending out when phone changes from idle to busy.

Busy to Idle

Specifies action URL for signal to be sending out when phone changes from busy to idle.

Open DND

Specifies action URL for signal to be sending out when Do Not Disturb mode is enabled.

Close DND

Specifies action URL for signal to be sending out when Do Not Disturb mode is disabled.

Open Forward

Specifies action URL for signal to be sending out when forwarding is enabled.

Close Forward

Specifies action URL for signal to be sending out when forwarding is disabled.

Open Unconditional Forward

Specifies action URL for signal to be sending out when phone disables call unconditional forward option.

Close Unconditional Forward

Specifies action URL for signal to be sending out when phone enables call unconditional forward option.

Open Busy Forward

Specifies action URL for signal to be sending out when phone enables call busy forward option.

Close Busy Forward

Specifies action URL for signal to be sending out when phone disables call busy forward option.

Open No Answer Forward

Specifies action URL for signal to be sending out when phone enables call no answer forward option.

Close No Answer Forward

Specifies action URL for signal to be sending out when phone disables call no answer forward option.

Blind Transfer

Specifies action URL for signal to be sending out when phone performed a blind transfer.

Attended Transfer

Specifies action URL for signal to be sending out when phone performed an attended transfer.

Transfer Finished

Specifies action URL for signal to be sending out when phone successfully transferred a call.

Transfer Failed

Specifies action URL for signal to be sending out when phone fails to transfer a call.

Hold Call

Specifies action URL for signal to be sending out when phone placed a call on hold.

UnHold call

Specifies action URL for signal to be sending out when phone placed retrieved a held call.

Mute call

Specifies action URL for signal to be sending out when mutes a call.

UnMute call

Specifies action URL for signal to be sending out when un-mutes a call.

Open Syslog

Specifies action URL for signal to be sending out when syslog is enabled on the phone.

Close Syslog

Specifies action URL for signal to be sending out when syslog is disabled on the phone.

IP Change

Specifies action URL for signal to be sending out when IP address change.

Auto-Provisioning Finish

Specifies action URL for signal to be sending out when phone completed auto provisioning.

Maintenance 🡪 TR-069

ACS URL

URL for TR-069 Auto Configuration Servers (ACS).

TR-069 Username

ACS username for TR-069.

TR-069 Password

ACS password for TR-069.

Periodic Inform Enable

Enables periodic inform. If set to “Yes”, device will send inform packets to the ACS. The default setting is “No”.

Periodic Inform Interval

Sets up the periodic inform interval to send the inform packets to the ACS.

Connection Request Username

The username for the ACS to connect to the phone.

Connection Request Password

The password for the ACS to connect to the phone.

Connection Request Port

The port for the ACS to connect to the phone.

CPE SSL Certificate

The Cert File for the phone to connect to the ACS via SSL.

CPE SSL Private Key

The Cert Key for the phone to connect to the ACS via SSL.

Randomized TR069 Startup

When enabled, the phone will send out first TR-069 INFORM message to server on randomized timing between 1 to 3600 seconds after phone boots up.

Maintenance 🡪 Security Settings 🡪 Security

Configuration via Keypad Menu

Configures the access control for the users to configure from keypad Menu.

Warning: If the admin password is lost while constraint mode is enabled, your device may become permanently unusable. Remember to be careful when using constraint mode to avoid irreversible damage.

Enable STAR key Keypad locking

If set to “Yes”, the keypad can be locked by pressing and holding the STAR * key for about 4 seconds. A lock icon will show indicating the keypad is locked. The default setting is “Yes”.

Note: When the keypad is locked, users would need press and hold the STAR * key for about 4 seconds and then enter the password to unlock it.

If the Star Key Lock is enabled without specifying password, user can press and hold the STAR * key for 4 seconds and press OK to unlock the phone.

Keypad Lock Type

Users can set Keypad Functional Lock to their phone instead of completely locking the phone.

Password to lock/unlock

Configures the password to lock/unlock the keypad. The password field allows number with up to 32 characters.

Emergency numbers

Emergency numbers, separated by comma (,).

Keypad Lock Timer

Defines the timeout (in seconds) of idle screen for locking keypad. Valid range is 0 to 3600.

Validate Server Certificates

Validate server certificates with our trusted list for TLS connections. If set to “No”, device will bypass certificate validation (not recommended).

SIP TLS Certificate

SSL Certificate used for SIP TLS Transport. Managed through “Upload” and “Delete” buttons.

SIP TLS Private Key

SSL Private key used for SIP TLS Transport. Managed through “Upload” and “Delete” buttons.

SIP TLS Private Key Password

SSL Private key password used for SIP TLS Transport.

Custom Certificate

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

Web Access Mode

Allows users to enable web access for the phone by specifying which protocol (HTTP/HTTPS) will be used for web interface access or disable the web access by selecting “Disabled”. Default setting is HTTP.

HTTP Web Port

Configures the HTTP port under the HTTP web access mode.

HTTPS Web Port

Configures the HTTPS port under the HTTPS web access mode.

Disable SSH

Disable SSH access. SSH is enabled by default.

SSH Public Key

Allows to use authentication keys for SSH access. The public key should be loaded to phone’s web UI while the private key should be used in the SSH tool side. Note: This will allow upcoming SSH access without password.

Web/Keypad/Restrict mode Lockout Duration

Specifies the time in minutes that the web or LCD login interface will be locked out to user after five login failures. This logout time is used for web login, STAR keypad unlock, and LCD restrict mode admin login. Range is 0-60 minutes.

Web Session Timeout

Configures how long the WEB GUI can remain idle before logout, the accepted range of values is 2-60 minutes. The default setting is 10.

Web Access Attempt Limit

Configures the number of logins attempts before lockout. The accepted range is 1-10. The default value is 5

Minimum TLS Version

The function allows users to choose minimum TLS version for HTTPS provisioning and SIP transport. This setting requires reboot to take effect on HTTPS web access. Provisioning and sip transport do not need reboot.

Maximum TLS Version

The function allows users to choose maximum TLS version for HTTPS provisioning and SIP transport. This setting requires reboot to take effect on HTTPS web access. Provisioning and sip transport do not need reboot.

Enable/Disable Weak Ciphers

Allows users to enable Weak TLS Ciphers Suite or disable weak ciphers DES/3DES and RC4, Symmetric Encryption SEED, Symmetric Authentication MD5, Protocol Version SSLv2/SSLv3 or Disable All of The Above Weak TLS Ciphers Suites. This feature could force the TLS version/Cipher suites for HTTPS provisioning and the TLS version for sip transport (TLS/TCP) and HTTPS web access.

Maintenance 🡪 Security Settings 🡪 Trusted CA certificates

Trusted CA certificates