GHP63X/W – Administration Guide

GHP63X/W

Protocols/Standards

SIP: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, Simple, TLS, SRTP.

Graphic Display

3.5” High-Resolution (480x320) Color LCD

Network Interface

One auto-negotiation 10/100 Mbps ethernet port integrated PoE Class 2

Wi-Fi

GHP630W/GHP631W integrated dual-band 2.4GHz & 5GHz 802.11 a/b/g/n/ac/ax

Keypad

26 keys, including:
-6 Context-Sensitive Soft keys;
-12 Standard Phone Digits keys (0-9: speed dial keys configuration on web page, * , #);
-5 Function keys (Flash, Redial, Hands-free, Voice Mail, Hold);
-3 Volume Control keys, Up/Down/Mute (with red color LED)

HD Audio

One HD handset support for wide band audio

Voice Conference

2 SIP accounts and lines, up to 3-way conference

Voice Codecs

Support for G.729A/B, G.711μ/a-law, G.726, G.722 (wide-band), G.723, iLBC, full band Opus, Inband and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB

Voice Capabilities

Dial, Answer, Redial, Flash ,Hands-free, Voice Mail(with the server), Transfer, Conference, Hold/Unhold, Mute /Unmute, flexible dial plan, Speed dial server redundancy & fail-over Call out

Telephony Features

Dial, Answer, Redial, Flash, Hands-free, Voice mail (with the server) Hold/Unhold, Mute/Unmute, flexible dial plan, Speed dial server redundancy, fail-over  Call out. 

Advanced Features

Support for multicast paging and E911 service.
Support GDS Door opening.
Support for advanced jitter-resilience algorithm.

USB

USB Type-C port for charging external devices (500mA MAX)

QoS

Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

Security

Secure boot, random default password, unique security certificate per device, administrator level passwords, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control

Multi-language

English, Spanish, French, Chinese (pending)

Upgrade/Provisioning

Firmware upgrade via FTP / TFTP / HTTP / HTTPS, fast boot in 15 seconds, mass provisioning using GDMS/TR-069 or AES encrypted XML configuration file

Power & Green Energy Efficiency

Universal power adapter (GHP630W/GHP631W):
Input: 100~240Vac 50~60Hz
Output: 12V/0.5A(6W)

Temperature and Humidity

Operation: 0°C to 40°C
Storage: -10°C to 60°C
Humidity: 10% to 90% (Non-condensing)

Package Content

GHP phone, handset with cord, Base Stand, universal power supply (GHP630W, GHP631W), faceplate, Quick Installation Guide

Color

White (GHP630,GHP630W), Black (GHP631,GHP631W)

Physical

Unit size:210mm(L)*147mm(W)*89.8mm(H) (with handset and base stand)
Unit weight: 601g (GHP630/GHP631), 658g (GHP630W/GHP631W US), 668g (GHP630W/GHP631W World)
Package weight: 824g (GHP630/GHP631), 881g (GHP630W/GHP631W US),891g (GHP630W/GHP631W World)

GHP630 & GHP631 Compliance

FCC: Part 15 Subpart B, Class B; Part 68. 316/317.
CE: EN 55032; EN 55035; EN IEC 61000-3-2; EN 61000-3-3; EN IEC 62368-1.
UKCA: BS EN 55032; BS EN 55035; BS EN IEC 61000-3-2; BS EN 61000-3-3; BS EN 62368-1.
RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004.
IC: ICES-003; CS-03 Part V.

GHP630W & GHP631W Compliance

FCC: Part 15 Subpart B, Class B; Part 15 Subpart C,15.247; Part 15 Subpart E,15.407; Part 68.
316/317.
CE: EN 55032; EN 55035; EN IEC 61000-3-2; EN 61000-3-3; EN IEC 62368-1; ETSI EN 301489-1-
17; ETSI EN 300 328; ETSI EN 301 893; EN IEC 62311.
UKCA: BS EN 55032; BS EN 55035; BS EN IEC 61000-3-2; BS EN 61000-3-3; BS EN 62368-1; ETSI
EN 301489-1-17; ETSI EN 300 328; ETSI EN 301 893; BS EN IEC 62311.
RCM: AS/NZS CISPR32; AS/NZS 4268; AS/NZS 2772.2; AS/NZS 62368.1; AS/CA S004.
IC: RSS-247; RSS-Gen; RSS-102; ICES-003; CS-03 Part V.

Status → Account Status

Account Status

This interface indicates the information of the accounts which are configured on the IP phone.

Status → Network Status

Ethernet

MAC Address

This field indicates the MAC address of the device.

IP Settings

This field indicated whether the obtained IP address was configured by DHCP or set statically.

IPv4 Address

This field displays the IPv4 address

IPv6 Address

This field displays the IPv6 address, if available

Subnet Mask

This field displays the subnet mask

Gateway

This field indicates the default gateway allocated.

DNS Server 1

This field displays the primary DNS server

DNS Server 2

This field displays the secondary DNS server

NAT Type

This field displays the NAT type used

Wi-Fi (Only for GHP63xW)

WLAN MAC Address

This field indicates the WLAN MAC Address.

SSID

This field indicates the SSID that the device is connected to.

Country Code

This field indicates the Country Code.

IPv4 Address Type

This field indicates theIPv4 Address Type, dynamic or static.

IPv4 Address

This field indicates the IPv4 Address.

Gateway

This field indicates the Gateway IP Address.

IPv4 NAT Type

This field indicates the IPv4 NAT Type.

IPv6 Address NAT Type

This field indicates the IPv6 NAT Type.

DNS & NAT

DNS Mode

Displays the DNS Mode used for both accounts.

NAT Traversal

Displays whether NAT Traversal is enabled or not, for both accounts.

Status → System Info

Information

This field indicates the product model.

Part number

This field indicates the part number.

Serial Number

Displays the GHP6xx's Serial Number.

Certificate Type

Displays the certificate type.

Software version

This field indicates the version of the firmware installed. Including the following types

Language

This field indicates the language used for displaying the web user interface.

Recommended Time Zone

This field indicates the recommended time zone.

System Up Time

This field indicates how long the devices has been turned on. 

System Time

This field indicates the time and date set on the system.

System Time Zone

This field indicates the system time zone configured on the device.

Download System Information

This option allows downloading all the the system information.

Capture Screen

Captures a screenshot from the phone LCD, by clicking the start button, once completed, the exact date and time of the screenshot will be displayed along the button that alows the screenshot download

Clear Screenshots

Deletes the latest screenshot taken

Service Status

Displays information about the service status, by displaying parameters such as the gui memory usage, the phone memory usage, and the cpe memory usage

Core Dump

Refers to the process of saving the contents of the phone's  memory (RAM) to a storage medium, when a program or the  encounters a critical error or crash. Core dumps are useful for developers and system administrators because they provide a snapshot of the program's state at the time of the crash.

Basic Settings

SIP Registration

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

UNREGISTER on Reboot

Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar.

The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

SUBSCRIBE Expiration

Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified registrar.

The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

Re-Register before Expiration

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

Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600. The default value is 20 seconds.

If enabled, the phone will add Authorization header in initial REGISTER request.

Default is “Disabled”.

Enable OPTIONS Keep Alive

Configures whether to enable SIP OPTIONS to track account registration status. If enabled, the phone will send periodic OPTIONS messages to server to track the connection status with the server.

Default is “Disabled”.

OPTIONS Keep Alive Interval

Configures the time interval the phone sends OPTIONS message to the server. If set to 30 seconds, it means the phone will send an OPTIONS message to the server every 30 seconds.

OPTIONS Keep Alive Max Tries

Configures the maximum number of times the phone will try to send OPTIONS message consistently to server without receiving a response. If set to “3”, the phone will send OPTIONS message 3 times. If no response from the server, the phone will re-register.

SUBSCRIBE for MWI

When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically.

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”.

Use Privacy Header

Configures whether the “Privacy Header” is present in the SIP INVITE message.

The default setting is “default”.

Use P-Preferred- Identity Header

Configures whether the “P-Preferred-Identity Header” is present in the SIP INVITE message.

Use X-Grandstream-PBX Header

Configures to use X-Grandstream-PBX header in SIP request.

Default setting is “Yes”.

Use P-Access-Network-Info Header

Configures to use P-Access-Network-Info header in SIP request.

Default setting is “Yes”.

Use P-Emergency-Info Header

Configures to use P-Emergency-Info header in SIP request.

Default setting is “Yes”.

Use MAC Header

The default setting is “No”.

Add MAC in User-Agent

The default setting is “No”.

SIP Transport

Selects the network protocol used for the SIP transport.

The default setting is “UDP”.

Enable TCP Keep-alive

Configures whether to enable TCP Keep-alive for the TCP connection between the terminal and the SIP server.

SIP Listening Mode

Configures whether or not to listen to multiple SIP protocols.

The default setting is “Transport Only”.

Local SIP Port

Configures the local SIP port used to listen and transmit.

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

Configures whether the actual ephemeral port in contact with TCP/TLS will be used when TLS/TCP is selected for SIP Transport.

The default setting is “No”.

Support SIP Instance ID

Configures whether SIP Instance ID is supported or not.

The default setting is “Yes”.

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. Default is 4 seconds.

Outbound Proxy Mode

Configures whether to put the Outbound Proxy in the Route header, or if SIP messages should always be sent to Outbound Proxy.

Default is “in route”.

Enable 100rel

When enabled, the 100rel tag is appended to the value of the Supported header of the initial signaling messages.

The default setting is “No”.

Use Route Set in Notify (Follow RFC 6665)

Configures whether to use route set in NOTIFY (follow RFC 6665).

Session Timer

Enable Session Timer

Configures whether to enable session timer function. It 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. If set to “Yes”, the phone will use the related parameters when sending session timer according to “Session Expiration”. If set to “No”, session timer will be disabled.

The default setting is “No”.

Session Expiration

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.

The default setting is “No”.

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.

The default setting is “No”.

Force Timer

If set to “Yes”, the phone will use the Session Timer even if the remote party does not support this feature. Otherwise, Session Timer is enabled only when the remote party supports it.

The default setting is “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. When set to “Omit”, the refresh object is not specified.

The default setting is “UAC”.

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.

The default setting is “UAC”.

Force INVITE

Select “Yes” to force using the INVITE method to refresh the session timer.

The default setting is “No”.

Audio

Preferred Vocoder

(Choice 1 – 8)

Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference. Users can configure vocoders in a preference list that is included with the same preference order in SDP message.

Codec Negotiation Priority

Configures the phone to use which codec sequence to negotiate as the callee. When set to “Caller”, the phone negotiates by SDP codec sequence from received SIP Invite. When set to “Callee”, the phone negotiates by audio codec sequence on the phone. The default setting is “Callee”.

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”.

iLBC Frame Size

Selects iLBC packet frame size. Users can choose from 20ms and 30ms. The default setting is “30ms”.

iLBC Payload Type

Specifies iLBC payload type. Valid range is 96 to 127. Cannot be the same as Opus or DTMF payload type. Valid range is 96 to 127. The default setting is “97”.

G.726-32 Packing Mode

Selects “ITU” or “IETF” for G726-32 packing mode.

The default setting is “ITU”.

G.726-32 Dynamic Payload Type

Specifies G.726-32 payload type. Valid range is 96 to 127. Default is 127.

Opus Payload Type

Specifies Opus payload type. Valid range is 96 to 127. It cannot be the same as iLBC or DTMF Payload Type. Default value is 123.

Send DTMF

Specifies the mechanism to transmit DTMF digits. There are 3 supported modes:

Default setting is “RFC2833”.

DTMF Payload Type

Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS payload type.

Silence Suppression

If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. For codec G.723 and G.729 only.

Default setting is “No”.

Jitter Buffer Type

Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting is “Adaptive”.

Jitter Buffer Length

Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default setting is “300ms”.

Voice Frames Per TX

Configures the number of voice frames transmitted per packet. It is recommended that the IS limit value of Ethernet packet is 1500 bytes or 120 kbps. 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 in the codec table or negotiate the payload type during the actual call. For example, if set to 2 and the first code is G.729, G.711 or G.726, the “ptime” value in the SDP datagram of the INVITE request is 20 ms. If the “Voice Frame/TX” setting exceeds the maximum allowed value, the phone will use and save the maximum allowed value for the selected first codec. It is recommended to use the default setting provided, and incorrect setting may affect voice quality.

The default setting is 2.

G.723 Rate

Selects encoding rate for G723 codec.

RTP Settings

SRTP Mode

Enable SRTP mode based on your selection from the drop-down menu.

The default setting is “No”.

SRTP Key Length

Allows users to specify the length of the SRTP calls. Available options are:

Default setting is AES 128&256 bit

Crypto Life Time

Enable or disable the crypto lifetime when using SRTP. If users set to disable this option, phone does not add the crypto lifetime to SRTP header. The default setting is “Yes”.

RTCP Mode

Configure RTCP port negotiation rules.

RTCP Keep-Alive Method

Configures the RTCP channel keep-alive packet type.

RTP Keep-Alive Method

Configures the RTP channel keep-alive packet type.

Symmetric RTP

Configures whether Symmetric RTP is used or not. Symmetric RTP means that the UA uses the same socket/port for sending and receiving the RTP stream. The default setting is “No”.

RTP IP Filter

General

Key as Send

Pressing the selected key will immediately dial out.

No Key Entry Timeout

Configures the timeout (in seconds) for no key entry.
If no key has been pressed after the timeout, the collected digits will be sent out.

Send Anonymous

If set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous.

Anonymous Call Rejection

If set to "Yes", anonymous calls will be rejected.

Enable Call Waiting

Enables the call waiting feature. If set to "No", new incoming calls will be rejected after the call is established.

RFC2543 Hold

If set to "Yes", c=0.0.0.0 will be used in INVITE SDP for hold.

Ring Timeout

Configures the timeoput (in seconds) for the phone to ring when an incoming call is not answered. Valid range is 30 to 3600.

Call Log

Auto Answer

Auto Answer

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep. Default setting is “No”.

Auto Answer Numbers

Allows the user to configure specific numbers to auto answer. If not set, all numbers will be auto answered If auto answer is enabled. Up to 10 numbers can be configured.

Intercom

Play warning tone for Auto-Answer Intercom

If enabled, phone will play warning tone when auto answering 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.

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.

Default is “Yes”.

Allow Barging by Call-Info/Alert-Info

When enabled, the phone will automatically put the current call on hold and answer the incoming call based on the SIP Call-Info/Alert-Info header sent from the server/proxy. However, if the current call was answered based on the SIP Call-Info/Alert-Info header, then all other incoming calls with SIP Call-Info/Alert-Info headers will be rejected automatically. Default setting is “No”.

Mute on Intercom Answer

If enabled, the phone will mute the microphone after answering an intercom call via Call-Info/Alert-Info.

Transfer

Transfer on Conference Hangup

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

Enable Local Call Features

When enabled, the phone will use the local call function, dial *87 plus number to use local blind forwarding; When disable, the server-side call function code will be used.
Enabled by Default.

Enable Recovery on Blind Transfer

Enable Recovery to the call to the transferee on failing blind transfer to the target.

Blind Transfer Wait Timeout

Configures the timeout (in seconds) when waiting for sipfrag response in blind transfer. Valid range is 30 to 300

Refer-To Use Target Contact

If enabled, the "Refer-to" hear uses the transferred target's Contact header information for attended transfer.

Dial plan

Dial Plan Prefix

Configures a prefix added to all numbers when making outbound calls.

Bypass Dial Plan

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

Dial Plan

Configures the dial plan rule. For syntax and examples, please refer to user manual for more details.

Dial Plan Rules:

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

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

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.

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

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.
 

Ringtone

Account Ring Tone

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu.

Ignore Alert-Info header

Configures to play default ringtone by ignoring Alert-Info header.

The default setting is “No”.

Match Incoming Caller ID

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

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.

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: <http://127.0.0.1>; info=priority

Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info matches one of the 10 rules, the phone will ring with the associated ringtone.

Security Settings

Check Domain Certificates

Configures whether the domain certificates will be checked when TLS/TCP is used for SIP Transport. The default setting is “No”.

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured.

The default setting is “No”.

Validate Incoming SIP Messages

Specifies if the phone will check the incoming SIP messages Caller ID and CSeq headers. If the message does not include the headers, it will be rejected. The default setting is “No”.

Allow Unsolicited REFER

Configures whether to dial the number carried by Refer-to header after receiving out-of-dialog SIP REFER request actively.

If set to “Disabled“, the phone will send error warning and stop dialing.

If set to “Enabled/Force Auth“, the phone will dial the number after sending authentication. If the authentication fails, it will stop dialing.

If set to “Enabled“, the phone will dial all numbers carried by SIP REFER.

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”.

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”.

Allow SIP Reset

Allow SIP Notification message to perform factory reset.

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.

The default setting is “No”.

MOH ( Music on Hold ) 

On Hold Reminder Tone

Configures to play reminder tone when the call is on hold.

Music On Hold URI

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

Special Feature

Specifies the server type for special requirements.

Swap Account Settings

Local RTP Port

Configures the local RTP port used to listen and transmit. The valid range is 1024 to 65400 and it must be even.

Local RTP Port Range

Configures the range of local RTP port. Valid value is from 24 to 10000.

Use Random Port

If set to "Yes", the parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. This parameter must be set to "No" for incoming direct IP calls (outgoing IP calls are not affected).

Keep-Alive Interval

Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep "ping hole" on the NAT router to open.

STUN Server

This option sets IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.

Use NAT IP

Configures the NAT IP address used in SIP/SDP messages. It should ONLY be used if required by your ITSP.

Delay Registration

Configures the specific time that the account will be registered after booting up.

Enable Outbound Notification

Configures whether to enable outbound notifications such as Action URL.

General

Preferred Default Account 

Select the preferred default account for on-hook or off-hook dialing. when the selected account is unavailable, system will use the first available account to dial out.

Mute Key Functions While Idle

If set to 'Idle Mute", clicking the MUTE key while idle will cause the phone to be muted automatically when answering incoming calls.

Do Not Escape '#' as %23 in SIP URI

Replaces # by %23 for some special situations.

User-Agent Prefix

Configures the prefix in the User-Agent header

Outgoing

Enable Direct IP Call

Enables Direct IP Call feature.

Onhook Dial Barging

When the option is set to "Disabled", onhook dialing won't be interrupted by an incoming call.

Off-hook Auto Dial

Configures the digits to be dialed via the first account when the phone is off-hook.

Off-hook Auto Dial Delay

Off-hook/On-hook Timeout (s)

If configured, when the phone is in the off-hook or on-hook, dialing state, it will go idle after the timeout (in seconds). Valid range is 10 to 60.

Incoming

Allow Incoming Call before Ringing

This allows incoming calls after dialed but before ringing. This can be used under custom user configuration based on need. 

Enable Call Waiting

Enables the call waiting feature. If set to "No", new incoming calls will be rejected after the call is established.

Enable Call Waiting Tone

Enables Call Waiting alert tone when another incmong call is received while a call is in progress. 

Auto Answer Delay

Configures the delay for automatically answering the incoming call. Valid range is 0 to 10 (seconds).

In Call

Enable Sending DTMF via specific MPKs

Allows certain MPKs to send DTMF in-call. This option does not affect Dial DTMF

Enable Auto Unmute

If the option is enabled, automatically unmute phone when a user unholds the call or establishes a new call.

Enable Busy Tone on Remote Disconnect

Playing busy tone when call is disconnected remotely.

Enable Mute Key In Call

When set to "No", the mute key will not work while on call.

Transfer

Enable Transfer

Enables Call Transfer feature.

Hold Call Before Completing Transfer

When set to "No", the phone will not hold the current call or the transfer target for an Attended Transfer.

Conference

Enable Conference

Enables the Conference feature.

Call Progress Tones

Configures ring or tone frequencies based on parameters from local telecom. The default value is North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.


Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence.
To set a continuous ring, OFF should be zero. Otherwise, it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.
Call Waiting tone gain can be set to either: Low, Medium, or High, it is set to Low by Default

System Ringtone 

Dial Tone

Second Dial Tone

Message Waiting

Ring Back Tone 

Call Waiting Tone 

Call Waiting Tone Gain

Busy Tone 

Multicast Paging Function

Enable or disable multicast paging.

Paging Barging

During an active call if incoming multicast page has higher priority (1 being the highest) than this value, the call will be held and multicast page will be played.

Paging Priority Active

If enabled, during a multicast page if another multicast is received with higher priority (1 being the highest) that one will played instead.

Multicast Listening

Secifies the label and multicast listening address, it supports up to 10 multicats addresses

Advanced Network Settings

DNS Refresh Timer (m) 

Configures the refresh time (in minutes) for DNS query. If set to "0", the phone will use the DNS query TTL from DNS server response.

DNS Failure Cache Duration (m)

Configures the duration (in minutes) of previous DNS query fails. If set to "0", the feature will be disabled. 
Note: Only valid for SIP registration.

Enable LLDP

Controls the LLDP (Link Layer Discovery Protocol) service.

LLDP TX Interval

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

Enable CDP

If enabled, the device will use the Cisco Discovery Protocol feature.

Layer 3 QoS for SIP

Configures the layer 3 QoS parameter for SIP. This value is used for IP Precendence, DiffServ, or MPLS.

Layer 3 QoS for RTP

Configures the layer 3 QoS parameter for RTP. This value is used for IP Precendence, DiffServ, or MPLS.

Enable DHCP VLAN

Configures to enable or disable auto-configure for VLAN settings via DHCP.

Enable Manual VLAN Configuration

Allow phone to keep the VLAN configuration without applying them.

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets.

Layer 2 QoS 802.1p Priority Value

Assigns the priority value of the Layer 2 QoS packets. Valid range is 0 to 7

Maximum Transmission Unit (MTU)

Configures the MTU in bytes

Remote Control

Action URI Support

Configures whether to enable phone to handle action URI request.

Action URI Allowed IP List

List of allowed IP addresses from which the phone receives the action URI.

CSTA Control

Configures whether to enable the CSTA Control feature.

Wi-Fi Function

This parameter enables/disables the Wi-Fi function.

Wi-Fi Band

Set the type of Wi-Fi band.

Country Code

Configures Wi-Fi Country Code

Available Wi-Fi

Available Wi-Fi 

Transfer Mode via MPK

Parameter

Description

Test Password Strength

Checks password strength to ensure better security

Admin Password

Current Password

The current admin password is required to set a new admin password.

New Password

Set new password for web GUI access as Admin. This field is case sensitive.

Confirm Password

Enter the new Admin password again to confirm.

IVR

Keypad Password

Configure the keypad password used in the IVR.

Enable IVR

If enabled, IVR is available. If disabled, enable configuration item "Enable Basic Settings in IVR" cannot take effect.
Enabled By Default.

Enable Basic settings in IVR

If this option is enabled, IVR can modify the IP, gateway, subnet mask, DNS, profile download protocol and firmware download protocol of the phone.
If this option is disabled, the IVR refuses to execute the above instructions.
Enabled by Default.

Parameter

Description

Minimum TLS Version

Configures the minimum TLS version supported by the phone. Minimum TLS version must be less than or equal to maximum TLS version. 

Maximum TLS Version

Configures the maximum TLS version supported by the phone. Maximum TLS version must be greater than or equal to minimum TLS version.

Enable/Disable Weak Cipher Suites

Defines the function for weak cipher suites. If set "Enable Weak TLS Cipher Suites", allow users to encrypt data by weak TLS cipher suites.

SIP TLS Certificate

The Cert File for the phone to connect to SIP Server via TLS.

SIP TLS Private Key

The Cert Key for the phone to connect to SIP Server via TLS.

SIP TLS Private Key Password

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

Custom Certificate

The uploaded custom certificate will be used for SSL/TLS communication instead of the phone default certificate.

Load CA Certificates

Phone will verify the server certificate based on the built-in, custom or both trusted certificates list.

Display Control

LCD

Backlight Brightness: Active

Configures the LCD brightness when the phone is active. Valid range is 10 to 100 where 100 is the brightest.

Backlight Brightness: Idle

Configures the LCD brightness when the phone is idle. Valid range is 0 to 100 where 0 is off and 100 is the brightest.

Active Backlight Timeout

Configures active backlight timeout (in minutes). The valid range is 0 to 90.
Default value is 1 minute.

Enable Missed Call Backlight

f set to "Yes", LCD backlight will be turned on when there is a missed call on the phone.

Blank Screen Timeout

Configures how long it waits to turn off LCD automatically during non-office hour under Standard Mode. The valid range is 0 to 90 minutes. 0 means disable Blank Screen feature

Office Hour

Configures office hour for each day of the week.

LED

New Message LED Indicator

Configure the behavior of the LED indicator when there is a new voice mail on the phone. If it is set to "off", the LED will not change in any state when there is a message.

Wallpaper

Wallpaper Source

Configures the location where wallpapers are stored.

Wallpaper Server Path

Configures the directory or file path of wallpaper.

Upload Wallpaper

The uploaded wallpaper must be in JPG or PNG format. 500 KB or smaller.

Color Background

Enter a color to use in HEX format.

e.g. #000000

Reference: http://htmlcolorcodes.com

Screensaver

Screensaver

Configures to turn on/off the screensaver feature.

Screensaver Source

Configures location where the screensaver is loaded from. If from USB, please name the folder "screensavers" and put screensaver pictures there.

Show Date and Time

Configures whether to display date and time on screensaver.
Enabled by default.

Screensaver Timeout

Configures the minutes of idle time before the screensaver activates. Valid range is 3 to 60.

Screensaver Server Path

Configures server path that contains screensaver definition XML.

Screensaver XML Download Interval

Configures the screensaver XML download interval (in minutes). If set to 0, automatic download will be disabled. Valid range is 5 to 720.

Audio Control

Speaker Ring Volume

Configures speaker ring volume, the valid range is 1 to 8.

Lock Speaker Volume

Lock volume adjustment when the option is enabled.

Enable Warning Tone

Configures whether to enable the warning tone of the phone. If disabled, the network disconnection and voicemail will not generate any tone.

Handset TX Gain (dB)

Configures the transmission gain of the handset.

Enable HAC

If enabled, the phone will be compatible with nearby hearing aids.

Parameter

Description

Enable TR-069

Enable or disable TR-069

ACS URL

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

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 TR-069 Auto Configuration Server.

Periodic Inform Interval

Configures periodic inform interval to send the inform packets to TR-069 Auto Configuration Server.

Connection Request Username

The username for the TR-069 Auto Configuration Server to connect the phone.

Connection Request Password

The password for the TR-069 Auto Configuration Sever to connect to the phone.

Connection Request Port

The port for the TR-069 Auto Configuration Server to connect to the phone.

CPE SSL Certificate

The Cert File for the phone to connect to the TR-069 Auto Configuration Sever via SSL.

CPE SSL Private Key

The Cert Key for the phone to connect to the TR-069 Auto Configuration Sever via SSL 

Start TR-069 at Random Time

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

Parameter

Description

Upgrade via Manual Upload

Upload Firmware File to Update

Configures to upload and upgrade firmware.

Upgrade via Network

Firmware Uprgade via

Configures firmware upgrade method as TFTP, FTP, FTPS, HTTP, HTTPS.

Firmware Server Path

Configures the server path for firmware download.

Firmware Server Username

The username for the firmware server.

Firmware Server Password

The password for the firmware server.

Firmware File Prefix

If configured, only the firmware with the matching prefix will be downloaded and flashed into the phone.

Firmware File Postfix

If configured, only the firmware with the matching postfix will be downloaded and flashed into the phone.

Upgrade Detection

Upgrade

Start the upgrade.

Parameter

Description

Configure Manually

Download Device Configuration

Click to download the device configuration file in .txt format.

Download Device Configuration (XML)

Click to download the device configuration file in .XML format.

Download User Configuration

Click to download the user's configuration file.

Upload Device Configuration

Upload configuration file to the phone.

Export Backup Package

Export a backup package which contains device configuration and personal data.

Restore from Backup Package

Restore from the uploaded backup package.

Configure via Network

Configure Uprgade via 

Choose the protocol that will be used for upgrading.

Configure Server Path

Defines the server path for provisioning.

Configure Sever Username

The username for the config server.

Configure Sever Password

The password for the config server.

Always Indicate before Challenge

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

Config File Prefix

If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone.

Config File Postfix

If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.

Authenticate Conf File

If enabled, the phone will authenticate configuration file before acceptance.

XML Config File Password

Configures the password for encrypting the XML configuration file using OpenSSL. This is required for the phone to decrypt the encrypted XML configuration file.

Parameter

Description

Auto Upgrade

Automatic Upgrade

Enables automatic upgrade and provisioning.

Start Upgrading at Random Time

Configures whether the phone will upgrade automatically at a random time within the configured time interval.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent.

DHCP Option

Allow DHCP Option 43 and Option 66 to Override Server

If set to "Yes" on the LAN side, the phone will reset the CPE, upgrade, network VLAN tag and priority configuration according to option 43 sent by the server. At the same time, the upgrade mode and server path of the configuration mode will be reset according to option 66 sent by the server. If set to "prefer, fallback when failed", the phone can fallback to use the configured provisioning server under its Firmware and Config server path in case the server from DHCP Option fails.

Allow DHCP Option 120 to Override SIP Server

Specifies the name of the client. This field is optional but may be required by Internet Service Provider.

Additional Override DHCP Option

Configures additional DHCP Option to be used for firmware server instead of the configured firmware server or the server from DHCP Option 33 and 66.

This option will be effective only when "Allow DHCP Option 43 and Option 66 to Override Server" is enabled.

Config Provision

Download and Process All Available Config Files

By default, the device will follow cfgMAC.xml, cfgMAC, cfgMODEL.xml, cfg.xml, devMAC.cfg sequence searches for the first available configuration file and updates the configuration file, and the subsequent configuration files will be ignored.If this configuration item is enabled, the device will follow cfg.xml, cfgMODEL.xml, cfgMAC, cfgMAC.xml, devMAC.cfg downloads configuration files one by one and loads updates. If the same configuration items exist between files, the configuration items that are later in the download order will overwrite the previous contents.

User Protection

When user protection is on, P-values that the user sets will not be changed by provision or provider.

3CX Auto Provision

If enabled, the phone will send SUBSCRIBE requests to the multicast address in LAN during bootup for the automatic provisioning. This feature requires 3CX server support.

Parameter

Description

Validate Hostname in Certificate

Configures tp validate the hostname in the SSL certificate.

Enable SIP NOTIFY Authentication

Device will challenge NOTIFY with 401 when set to "Yes".

Factory Reset

Initiate factory reset.
All the configuration on the device will be erased.

Parameter

Description

Syslog Protocol

Allows sending syslog through secured TLS protocol to the syslog server.
Note: CA verification is required.

Syslog Server

The URL/IP address for the syslog server.

Syslog Level

Selects the level logging of syslog.

Syslog Keyword Filter

Syslog will be filtered based on the configured keywords. Multiple keywords can be separated by ",".

Send SIP Log

Configures whether the SIP log will be included in the syslog messages. 

Parameter

Description

Phone Status 

Setup Completed

Configures the Action URL to send when phone finishes setup process.

Registered

Configures the Action URL to send when phone successfully registers as a SIP account.

Unregistered

Configures the Action URL to send when phone unregisters a SIP account.

Register Failed

Configures the Action URL to send when phone fails to register a SIP account.

Idle to Busy

Configures the Action URL to send when the phone's state changes form idle to busy.

Busy to Idle

Configures the Action URL to send when phone's state changes from busy to idle.

Auto Provision Completed

Configures the Action URL to send when phone's auto provision process is completed.

IP Change

Configures the Action URL to send when the IP address changes.

Call Operation

Off-hook

Configures the Action URL to send when phone is in off-hook state.

On-hook

Configures the Action URL to send when the phone is on-hook state.

Incoming Call

Configures the Action URL to send when the phone receives an incoming call.

Outgoing Call

Configures the Action URL to send when phone places a call.

Missed Call

Configures the Action URL to send when phone has a missed call.

Established a Call 

Configures the Action URL to send when phone establishes a call.

Terminated Call

Configures the Action URL to send when the phone terminates a call.

Answered Call

Configures the Action URL to send when phone answers an incoming call.

Blind Transfer 

Configures the Action URL to send when the phone performs blind transfer.

Attended Transfer

Configures the Action URL to send when the phone performs attended transfer.

Transfer Completed

Configures the Action URL to send when the phone successfully transfers a call.

Transfer Failed

Configures the Action URL to send when the phone fails to transfer a call.

Hold Call

Configures the Action URL to send when the phone places a call on hold.

Unhold Call

Configures the Action URL to send when the phone resumes the call on hold.

Mute Call

Configures the Action URL when the phone mutes a call.

Unmute Call

Configures the Action URL when the phone unmutes a call.

Parameter

Description

Hotel Name

Configures the hotel name.

Hotel Address

Configures the hotel address.

Hotel Phone Number

Configures the hotel phone number.

Hotel Fax Number

Configures the hotel fax number.

Hotel Room Number

Configures the hotel's room number. 

Parameter

Description

Enable E911

Enable or disable E911.

HELD Protocol

Configures HELD transfer protocol.

HELD Synchronization Interval

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

Location Server

Configures the primary Location Information Server (LIS) adderss.

Location Server Username

Configures the username of the primary Location Information Server (LIS).

Location Sever Password

Configures the password of the primary Location Information Server (LIS).

Secondary Location Server

Configures the secondary Location Information Server.

Secondary Location Server Username

Configure the username of the secondary Local Information Server.

Secondary Location Server Password

Configure the password of the secondary Location Information Server.

HELD Location Types

Configure "locationType" element in the location request.

HELD Use LLDP Information

If set to "Yes", the information from LLDP-support switch is used to generate ChassisID and portID; otherwise

HELD NAI

If set to "Yes", Network Access Identifier (NAI) is included as a device identity in the location request sent to the Location Information 

E911 Emergency Numbers

A user can configure multiple emergency numbers separated with the delimiter symbol ";".

Geolocation-Routing Header

If set to "Yes", E.911 INVITE message includes the "Geolocation-Routing" header with the value "Yes".

Priority Header

If set to "Yes", E.911 INVITE messages includes the "Priority" header with the value "emergency".

Parameter

Description

Use Auto Location Service

Configures to enable or disable auto location services on the phone.

GDS

Connect to a GDS37XX and send OpenDoor request.

Notes:

The “Access Password” on GHP6XX should be matching “Remote PIN to Open the door” on GDS37XX.

General Settings

Enable PNP

Configures whether to enable the PNP function. After enabling it, the automatic configuration is supported, and up to 100 devices can be configured.
Disabled by Default.

PNP URL Mode

Select the PNP URL mode, including local and custom, custom does not support filling in the local machine-related address.

PNP URL

The configuration terminal can take the server address of the configuration file. The server address can be the IP address of the terminal that provides PNP services. It cannot be configured as a locally related parameter.

Template Management

Profile templates can be managed in this module. The current model only supports the management of one profile template. It supports the generation of batch configuration CSV files based on the profile template. You can fill in the relevant parameter values in this file, and start the PNP batch configuration process after the application.

Upload Profile Template

Users can upload a .xml template file downloaded by respecting the following guidelines: 

Batch Configure CSV

Users can upload a .csv template file download while respecting the following guidelines: 

Effects View

Displays the data inserted from the .csv template to the .xml configuration file.

Progress Data

When connected to a network, the phone can automatically identify and configure itself with the appropriate settings for that network. This eliminates the need for manual configuration, making it faster and easier to set up and manage IP phone deployments.
the connected networks will be displayed in the list.

MAC Discovery 

During the plug-and-play process, the network infrastructure uses the MAC Discovery feature to detect the IP phone's MAC address and automatically configure it with the appropriate settings, such as IP address, subnet mask, default gateway, and other network-related parameters. This eliminates the need for manual configuration and speeds up the deployment of IP phones in large-scale environments.
A list of MAC Addresses with their respective IP address, Product models, and operations is displayed, and the actions that can be performed on those IP phones are the following : 

Basic Settings

Enable Account Sharing

Select whether to enable Account Sharing.

Role in Account Sharing

Select the role that the current device will play in the network, the guest device role does not need to register an account on IP PBX, and can make calls in and out of the network through the account set by the host device role.

Group Name

Set the group name, in the host-guest mode, devices with the same group name can discover each other.

Note: This item is mandatory if using Account Sharing. The verification format is domain type

Group Password

In the host-guest mode, after setting the group password, the guest device with the same group password as the host device can successfully register an account on the host device.

Note:This item is mandatory if using Account Sharing.

Account Settings

Account

For the host device role, this setting determines which host device account will be used as the guest device outgoing and incoming account for calls outside the Account Sharing. For the guest device role, this setting determines which account the guest device will use to register on the host device.

Account Name

This setting specifies the account name corresponding to the account used by the guest
device.

Sync Ringing In Group

Set whether to enable synchronization of all successfully registered guest device ringtones within the group.

Discovered Host Device list

Shows registered devices on the local network for monitoring, it displays the following information about the discovered devices:

Contacts

Add Contact

The user can add a contact to the phonebook by clicking the "Add Contact" icon, the following parameters can be defined for each contact:

Group Management

Add Group

The user can create a contacts group by clicking the "Add group" icon, the following parameters can be defined

Phonebook Management

Enable Phonebook XML Download

Enables Phonebook XML download via HTTP, HTTPS, FTP or TFTP.

HTTP/HTTPS Username

Defines the username for the HTTP/HTTPS server.

HTTP/HTTPS Password

Defines the password for the HTTP/HTTPS server

Phonebook XML Server Path

Configures the server path to download XML phonebook file. This field could be IP address or URL, with up to 256 characters.

Phonebook Download Interval

Configures the phonebook download interval (in minutes). If set to 0, automatic download will be disabled. Valid range is 5 to 720.

Remove Manually-edited Entries on Download

If set to "Yes", when XML phonebook is downloaded, the entries added manually will be automatically removed.

Import Group Method

When set to "Replace", the existing groups will be completely replaced by imported one. When set to "Append", the imported groups will be appended to the current one

Sort Phonebook by

Configures to sort phonebook based on the selection of first name or last name. If you select "Last name", the contact's last name will be displayed first, and the phone book will be sorted by last name; if you select "First name", the contact's first name will be displayed first, and the phone book will be sorted by first name.

Download XML Phonebook

Downloads the XML Phonebook file.

Upload XML Phonebook

Uploads XML Phonebook file to the phone in xml format

Phonebook Key Function

Configures the behavior of the Phonebook key.

Default Search Mode

Configures the default phonebook search mode.

Replace Duplicate Items

Replace duplicate items by name or number