GXV34x0 - Admin Guide

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN®

Network Interfaces

Dual switched 10/100/1000 Mbps ports with integrated PoE/PoE+

Graphic Display

8" 1280×800 capacitive touch screen (10 points) IPS LCD

Camera

Tiltable 2 megapixel CMOS camera with privacy shutter, 1080p 30fps

Bluetooth

Yes, integrated Bluetooth 5.0

Wi-Fi

Yes, dual-band (2.4GHz & 5GHz) with 802.11 a/b/g/n/ac/ax, 2T2R, Wi-Fi Display & AirPlay

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets), 3.5mm stereo headset with microphone, USB 3.0 port, Type-C, HDMI-out, HDMI-in

Feature Keys

2 function touch keys VOLUME +/-, 3 dedicated Android touch keys HOME, MENU, and BACK

Voice Codec

Wide-band Opus, wide-band G.722, G.711μ/a, G.729A/B, G.726-32, iLBC, in-band and out-ofband DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC, ANS, Noise Shield 2.0

Video Codec and Capabilities

H.264 BP/MP/HP, video resolution up to 1080p, frame rate up to 30 fps, bit rate up to 4Mbps, 3-way video conference (1080p@30fps), BFCP, people video(up to 1080p@30fps) + content video(up to 1080p@15fps), anti-flickering and auto exposure

Telephony Features

Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 12-way audio conference(including the host), shared-call-appearance (SCA) / bridged-line-appearance (BLA), virtual MPK, downloadable contacts (XML, LDAP, up to 2000 items), call record(local and server), call log (up to 2000 records), call waiting, auto answer, XML customization of screen, flexible dial plan, personalized music ringtones and music on hold, server redundancy & fail-over

Sample Applications

Android

Runs Android 11

Applications Deployment

Supports Android 11 compliant applications to be developed, downloaded and run on the device with provisioning control.

HD Audio

Yes, 2 omnidirectional microphones, HD handset and speakerphone with support for wideband audio

Base Stand

Integrated stand with multiple adjustable angles

QoS

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

Security

User and administrator level passwords, random default admin password, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access control, Kensington Security Slot (Kensington Lock) support, anti-hacking secure boot

Multi-language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese, and more

Upgrade/Provisioning

Firmware upgrade via TFTP / HTTP / HTTPS or local HTTP upload, mass provisioning using TR-069 or AES encrypted XML configuration file, GDMS.
Note: Users can remotely access the GXV3480's web interface through the GDMS device options menu.

Power & Green Energy Efficiency

Universal power adapter included: Input: 100-240VAC 50-60Hz; Output 12VDC 1.5A (18W)
Integrated PoE* 802.3af Class 3, PoE+ 802.3at, Class 4
*Must use PSU or PoE+ to power up the phone when using USB devices

Temperature and Humidity

Operation: 0°C to 40°C

Storage: -10°C to 60°C

Humidity: 10% to 90% Non-condensing

Package Content

GXV3480 phone, handset with cord, base stand, universal power supply, network cable, screen cleaning cloth, quick installation guide

FCC: CFR 47 Part 15 Subpart B Class B; CFR 47 Part 15 Subpart C; CFR 47 Part 15 Subpart E; Part 68 (HAC) 

IC: RSS-247, RSS-GEN, RSS-102, ICES-003, CS-03. 

CE: EN 55032 Class B;EN 55035; EN IEC 61000-3-2; EN 61000-3-3; EN IEC 62368-1; ETSI EN 300328; ETSI EN 301893; ETSI EN 301489-1; ETSI EN 301489-17; EN IEC 62311 

UKCA: BS EN 55032 Class B; BS EN 55035; BS EN IEC 61000-3-2; BS EN 61000-3-3;  BS EN IEC 62368-1; ETSI EN 300328; ETSI EN 301893; ETSI EN 301489-1; ETSI EN 301489-17; BS EN IEC 62311 

RCM: AS/ACIF S040 AS/CA S004; AS/NZS CISPR 32; AS/NZS 62368.1; AS/NZS 4268. 

HDMI 

RoHS 2.0

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6, OpenVPN®

Network Interfaces

Dual switched 10/100/1000 Mbps ports with integrated PoE/PoE+

Graphic Display

5.0 inch 1280×720 capacitive touch screen (5 points) HD IPS LCD

Camera

Tiltable 2 megapixel CMOS camera with privacy shutter, 1080p 30fps

Bluetooth

Yes, integrated Bluetooth 5.0

Wi-Fi

Yes, dual-band dual-stream Wi-Fi 5(2.4GHz & 5GHz) with 802.11 a/b/g/n/ac

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics headsets), USB 2.0 port, USB 3.0 port

Feature Keys

11 functions keys for CONFERENCE, TRANSFER, SEND/REDIAL, MUTE, EARPHONE, SPEAKERPHONE, VOLUME -/+. 3 dedicated Android keys for HOME, RECENT, and BACK

Voice Codec

Wide-band Opus, wide-band G.722, G.711μ/a, G. 729A/B, G.726-32, iLBC, in-band and out-ofband DTMF (In audio, RFC2833, SIP INFO), VAD, CNG, AEC, PLC, AJB, AGC, ANS, Noise Shield 2.0

Video Codec and Capabilities

H.264 BP/MP/HP, video resolution up to 720p, frame rate up to 30 fps, bit rate up to 2Mbps, 3-way video conference (720p@30fps), anti-flickering and auto exposure

Telephony Features

Hold, transfer, forward (unconditional/no-answer/busy), call park/pickup, 10-way audio conference(including the host), shared-call-appearance (SCA) / bridged-line-appearance (BLA), virtual
Programmable Key, downloadable contacts (XML, LDAP, up to 1000 items), call record(local and server), call log (up to 1000 records), call waiting, auto answer, XML customization of screen, flexible dial plan, personalized music ringtones and music on hold, server redundancy & failover

Extention Module

Yes, can power up to 4 GBX20 EXT modules which feature a 272x480 color LCD, 20 quick-dial/BLF keys with dual-color LED, 2 navigation keys, and less than 1.2W power consumption per unit.

Sample Applications

Android

Runs Android 11

Applications Deployment

Supports Android 11 compliant applications to be developed, downloaded and run on the device with provisioning control.

HD Audio

Yes, 2 omnidirectional microphones, HD handset and speakerphone with support for wideband audio

Base Stand

Yes, desktop stand with three adjustable levels

QoS

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

Security

User and administrator level passwords, random default admin password, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access control, Kensington Security Slot (Kensington Lock) support, anti-hacking secure boot

Multi-language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese, and more

Upgrade/Provisioning

Firmware upgrade via TFTP / HTTP / HTTPS or local HTTP upload, mass provisioning using TR-069 or AES encrypted XML configuration file, GDMS

Power & Green Energy Efficiency

Universal power adapter included: Input: 100-240VAC 50-60Hz; Output 12VDC 1.5A (18W)
Integrated PoE* 802.3af Class 3, PoE+ 802.3at, Class 4
*Must use PSU or PoE+ to power up the phone when using USB devices

Temperature and Humidity

Operation: 0°C to 40°C

Storage: -10°C to 60°C

Humidity: 10% to 90% Non-condensing

Package Content

GXV3450 phone, handset with cord, base stand, universal power supply, network cable, screen cleaning cloth, quick installation guide

Compliance

FCC: CFR 47 Part 15 Subpart B Class B; CFR 47 Part 15 Subpart C; CFR 47 Part 15 Subpart E; Part 68 (HAC) 

IC: RSS-247, RSS-GEN, RSS-102, ICES-003, CS-03

CE: EN 55032 Class B;EN 55035; EN IEC 61000-3-2; EN 61000-3-3; EN IEC 62368-1; ETSI EN 300328; ETSI EN 301893; ETSI EN 301489-1; ETSI EN 301489-17; EN IEC 62311 

UKCA: BS EN 55032 Class B; BS EN 55035; BS EN IEC 61000-3-2; BS EN 61000-3-3;  BS EN IEC 62368-1; ETSI EN 300328; ETSI EN 301893; ETSI EN 301489-1; ETSI EN 301489-17; BS EN IEC 62311 

RCM: AS/ACIF S040 AS/CA S004; AS/NZS CISPR 32; AS/NZS 62368.1; AS/NZS 4268

RoHS2.0

GXV3450

Item

Name

1

Camera Adjusting Wheel

Scroll up/down to adjust the camera angle.

2

Phone Stand Slot

Put the phone stand from left to right into the slots.

3

Handset Port

RJ9 handset connector port.

4

Headset Port

RJ9 headset connector port (supporting EHS with Plantronics headset).

Power Jack

12V DC Power connector port.

7

PC Port

10/100/1000Mbps RJ-45 port connecting to PC.

8

USB Port 2.0

USB devices can be connected via the USB port. For example, connect a USB flash drive to save captured pictures.

9

Side GBX20 Connection Slot

The slots for connecting GXV3450 and GBX20.

10

USB 3.0 

USB devices can be connected via the USB port. For example, connect a USB flash drive to save captured pictures.

Item

Name

1

Camera Adjusting Wheel

Scroll up/down to adjust the camera angle.

2

Phone Stand Slot

Put the phone stand from left to right into the slots.

3

Handset Port

RJ9 handset connector port.

4

Headset Port

RJ9 headset connector port (supporting EHS with Plantronics headset).

Power Jack

12V DC Power connector port.

7

PC Port

10/100/1000Mbps RJ-45 port connecting to PC.

8

USB Port 2.0

USB devices can be connected via the USB port. For example, connect a USB flash drive to save captured pictures.

9

Side GBX20 Connection Slot

The slots for connecting GXV3450 and GBX20.

10

USB 3.0 

USB devices can be connected via the USB port. For example, connect a USB flash drive to save captured pictures.

Status

Features

Advanced

Call Settings

Enable Video Call

Configures the video call function for this account. If set to “Default”, it will be configured according to global video call function.

Start Video Automatically

Permits the phone system to enable the video feature automatically when it makes an outbound call. If set to “Yes”, the video codec attributes will be included in the SIP INVITE message. Or the attributes will not be included.

The default setting is “Yes”.

Remote Video Request

Configures the preference to handle video request from the remote party during an audio call. The default is “Prompt”.

Video Layout

Defines whether to enter full screen when incoming video call is answered.

Note: “Fullscreen” is not supported on GXV3470. “Equal Split Screen” is not supported on GXV3450.

Auto Answer

Sets the phone system to allow to answer an incoming call automatically when idle. If it is set to “Yes”, the phone will automatically enable the speaker phone to answer all the incoming calls after a short reminding beep. If set to “Enable Intercom/Paging”, it will automatically answer the incoming calls whose SIP INVITE includes auto-answer tag in the info header.

The default setting is “No”.

Play Warning Tone for Auto Answer Intercom

When this option is enabled, the phone will play a warning tone When auto-answering intercom.

The default setting is “Yes”.

Intercom Barge

Configures whether to answer the incoming intercom call when there is already an active call on the phone. When” Intercom Barging” is enabled, and if the current active call is an intercom call, the incoming intercom call will be automatically rejected; otherwise if the current active call is not an intercom call, the current active call will be put on hold and the incoming intercom will be automatically answered. When “Intercom Barging” is disabled, a prompt will show up indicating the incoming intercom call without interrupting the current active call. Default setting is disabled.

Auto Preview

Configures whether to turn on video to preview the video of the caller. If set to "Yes", the user can view the video and hear the caller on the incoming page when there is an incoming call. If set to "Yes with Ringing", the caller can view the video of the caller and hear the ringtone on the incoming page but cannot hear the caller. The default setting is "No".

Note: If Auto Answer function has been enabled, this function does not take effect.

Send Anonymous

Sets the phone system to make an anonymous outgoing call. If set to “Yes”, the “From” header in the SIP INVITE messages will be set to anonymous, essentially blocking the Caller ID to be displayed. Default is “No”.

Intercept Anonymous Calls

If set to "Yes", anonymous calls will be automatically blocked.

Call Log

Categorizes the call logs saved for this account.

The default setting is “Log All”.

Enable Call Features

Configures the local start command feature. If it is set to “Yes”, the feature will be enabled to recognize the local star code command. Otherwise, it will be disabled. The default setting is “No”.

Enable Call Waiting

Configures the call waiting function for this account. If set to "Default", it will be configured according to global call waiting function.

Mute on Answer Intercom Call

When enabled, phone will mute the incoming intercom call based on Call-Info/Alert Info Headers. Default is disabled.

Transfer on 3-way Conference Hang up

Transfers conference from hosted party when hang up, thus other parties can continue the conference without interruption. Default is unchecked.

Use # as Dial Key

Treats “#” as the “Send” (or “Dial”) key when set to “Yes”. If set to “No”, this “#” key can be included as part of the dialed number or it will be used as redial key when the input area has no number (please make sure the dial plan is properly configured to allow dialing # out). Default is “Yes”.

Use # as Redial Key

Allows users to configure the "#" key as the "Redial" key. If set to "Yes", the "#" key will immediately redial the last call. In this case, this key is essentially equivalent to the "Redial" key. If set to "No", the "#" key is treated as part of the dialed string. Default is “Yes”.

DND Call Feature On

Configures the feature code to enable the DND (Do Not Disturb) feature for this account. If it is configured, the phone will dial the feature code automatically when the DND feature is enabled.

DND Call Feature Off

Configures the feature code to disable the DND (Do Not Disturb) feature for this account. If it is configured, the phone will dial the feature code automatically when the DND feature is disabled.

No Key Entry Timeout (s)

Determines the expiration timer (in seconds) for no key entry. The dialed digit will be sent out if no other digits entered within the set period. The default value is 4 seconds. The valid range is 1 - 15. This feature does not work if the dialer page is entered via the Account Widget on the phone.

Ring Timeout (s)

Defines the expiration timer (in seconds) for the rings with no answer,- If set to "0", ringing will never time out.

The default setting is 60. The valid range is 0, 10 - 300..

Refer-To Use Target Contact

Sets the phone system to use the target’s Contact header tag to the Refer-To header in the SIP REFER message during an attended transfer.

The default setting is “No”.

RFC2543 Hold

If yes, c=0.0.0.0 will be used in INVITE SDP for hold.

Call Forwarding

Sets the Call Forwarding feature for this account.

Dial Plan

Dial Plan Prefix

Configures the digits prepended to the dialed number.

Disable Dial Plan

Enables/disables the Dial plan mechanism for different cases. If the specific case is checked, the Dial plan mechanism will be disabled.

The default setting is unchecking all the cases.

Dial Plan

Configures the dial plan to establish 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;

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

Allow 311, 611, and 911 or any 10-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+}

Allow 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 allow any length of numbers with leading digit 2, replacing the 2 with 011 when dialed.

Example 4: {0,x+}

If user dials 0 second dial tone will be played, and users can continue entering digits. For instance, if pressing 01234, after pressing 0 second dial tone is played, full number is sent in INVITE.

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

Allow any number of digits, OR any number with a leading +, OR any number with a leading *, OR any number with a leading * followed by a 2 digits number and a *. To dial + from keypad, press on 0 until + appears on LCD.

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):

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.

Caller IDs

Caller ID Display

Specifies which header tag will be used from the SIP INVITE message for the Caller ID display.

Callee ID Display

Specifies which header tag will be used from the SIP INVITE message for the Callee ID display.

Ringtone

Account Ringtone

Configures the ringtone for the account. Users can set ringtones from the dropdown list. User can also import customized ringtone from LCD Setting menu.

The customized ringtone file name will also be showed up in the dropdown list that allows user to select.

Ignore Alert-Info header

Configures the default ringtone.

If set to “yes”, the incoming alert info header from the SIP server will be ignored and default configured ringtone will be played.

The default setting is “No”.

Match Incoming Caller ID

Specifies the rules for the incoming calls. If the incoming caller ID or Alert Info matches the number, pattern or Alert Info text rules, the phone will play the selected distinctive ringtone. The rule policy:

·  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:

· 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: <http://127.0.0.1>; info=priority

Distinctive Ring Tone

Selects the distinctive ring tone if the incoming caller ID matched the specified Match Incoming Caller ID rule. If so, the phone will play the selected ringtone.

Special Features

Configures phone’s settings to meet different vendors’ server requirements.

Users can choose from Standard, CBCOM, RNK, China Mobile, ZTE IMS, Mobotix, ZTE NGN, or Huawei IMS depending on the server type.

Call Settings

Feature Key Synchronization

This feature is used for BroadSoft / Metaswitch call feature synchronization. When it’s set to BroadSoft / Metaswitch, DND and Call Forward features can be synchronized with BroadSoft / Metaswitch server. The call forward function will take effect on the server side while the local call forward function is not effective.

The default setting is “Disable”.

Conference URI

Configures the network-based conference URI (the BroadSoft Standard). If it is configured, end user needs to tap the N-way key during the conference to transfer the host to the remote media server.

Broadsoft Call Center

When enabled, Feature Key Synchronization will be enabled regardless of web settings. Default is Disabled.

Hoteling Event

Call Center Status

When set to “Yes”, the phone will send SUBSCRIBE to the server to obtain call center status. Default is Disabled.

SCA

Enable SCA (Shared Call Appearance)

Note: The Enable SCA settings now supports provision via strings. The value “0” can be replaced with “line”or “private”; value “1” can be replaced with “shared” or “sharedline”.

Enable Barge-in

Enables/disables the Barge-In feature. If it is set to “Yes”, the user could tap the SCA account to barge into an active session with another shared line.

The default setting is “No”.

Line-seize Timeout (s)

Configures the interval (in seconds) when the line seize is considered timed out when Shared Line feature is used. Valid range is 15-60.

Call Park

BroadSoft Call Park

Configures whether to send SUBSRCIRBE message to BroadSoft server to obtain Call Park notifications.
Disabled by Default.

Call Park Feature Code

Configures the feature access code for parking current call to a parking lot or to another extension.

Auto-filling Call Park Retrieve Feature Code

If enabled, the Call Park Retrieve feature code will be automatically filled in the dialing interface when the phone is in call pickup.

Enabled by Default.

Call Park Retrieve Feature Code

Configures the feature access code for retrieving the parked call on accounts or BLF keys.

Enable Video Call

Enables the video call feature on the phone.

The default setting is “Yes”.

Enable Direct IP Call Mode

Configures enable/disable direct IP call mode of the phone. If set to "Yes", the feature of direct IP call will be enabled.

Default is “No”.

Enable Quick IP-call Mode

This feature allows users to make a direct IP call by dialing the last octet of the IP address (last 3 digits of IP address). Note: The phones must be on the same LAN/VPN. No SIP server is required so the field should be left blank. When set to "Yes", if XXX is dialed, where X is 0-9 and XXX <=255, the phone will make a direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the LAN/VPN IP address regardless of the subnet mask. XX or X is also valid so a leading 0 is not required. See User Guide for details.
This feature is available only when "Direct IP Call Mode" is enabled.
Disabled by Default.

Enable Paging Call Mode

Configures enable/disable paging call mode of the phone. If set to "Yes", the feature of paging call will be enabled.

Default is “No”.

Enable Call Waiting

Enables the call waiting feature. If it is not checked, the phone system will reject the second incoming call during an active session without user’s knowledge. But this missed call record will be saved to remind users.

The default setting is “Yes”.

Enables Call Waiting Tone

Sets the phone system to play the call waiting tone if there is another incoming call. If it is set to “No”, the phone will only display the indicator on the LCD screen for another incoming call. The default setting is “Yes”.

Enable DND Reminder Ring

Configures the phone system to play the DND reminder ringtone for the incoming call if the DND feature is enabled. If it set to “No”, the phone will keep mute instead of playing a ring splash to indicate an incoming call when DND is enabled. The default setting is “Yes”.

Enable Transfer

Enables the transfer feature. When set to “No”, the phone system will block the TRANSFER key on the LCD screen. The default setting is “Yes”.

Hold Call Before Completing Transfer

When set to “Yes” the phone holds the second call before completing the attended transfer (it sends the INVITE method to hold the call before sending the REFER method).

Default Transfer Mode

Sets the default transfer mode for the phone system. If the Blind Transfer or Attended Transfer mode is set, the phone system will use the specific mode to transfer an active call. The users still have privilege to switch the mode on the LCD screen when they tap the transfer key.

The default setting is “Blind Transfer”.

Enable transfer via non-Transfer Programmable Key

Programmable Key with the type speed dial, BLF and speed dial via active account will perform as transfer programmable keys under active call. The transfer mode during the call depends on the “Default Transfer Mode” mentioned above. MPK can also be selected as forward/transfer destination on the ringing screen when [Enable Function for Incoming Call] is set to “Call Transfer”.

Special Function for Incoming Call

Enables the preview feature for the incoming video calls. Defines the function for the incoming video call.

Note: By pressing the preview button, the phone will send the SIP 183 message to the caller’s camera, based on SIP RFC3261; the caller’s camera should start sending the stream to the phone upon receiving the SIP 183.

The default setting is “None”.

Enable Conference

Enables the conference. When set to “No”, the phone will block the conference application. The default setting is “Yes”.

Auto Conference

Allows the phone system to invite all call parties into a conference by pressing the Conf key. If it is disabled, the end user has to add each call party to conference manually. The default setting is “No”.

Hold Call before Adding conferee 

Configures whether to place the current call on hold before adding new member(s) to a conference. If enabled, the current call will be put on hold when the host presses Conference or Add key to invite new member(s). When an invited member answers the call and agrees to attend the conference, the host needs to manually resume the conference with the new member added. If disabled, the current call will not be put on hold and the invited member will join the meeting automatically after answering the call.

Auto Mute Mode 

Configures whether to mute the call on entry automatically.

Note: This function only take effect when the phone is from the idle status to call status. Users could click the Mute button on call interface to cancel the current mute status.

Always Ring Speaker

Determines if the speaker will play the ringtone if the speaker channel is not set as default channel. If set to “Yes”, the phone will force to play the ring speaker in speaker channel. The default setting is “No”.

End user might need this feature when the headset is connected.

Offhook Auto Dial

Configures the User ID/extension to dial automatically when the phone is off-hook. The phone will use the first account to dial the configured numbers out.

The default setting is “No”.

Offhook Auto Dial Delay (s) 

Defines the timer for warm line dialing. After the timer expires, the phone system will dial the configured number in Off-hook Auto Dial automatically. If it is not configured, the configured number will be dialed immediately.

Off-hook/On-hook Timeout (s)

If configured, the phone will exit the dial-up screen when timeout after Offhook or Onhook. default is 30s.

The valid range is 10-60s.

Handset Option 

Configures the Handset options and can be set to:

Default setting is “Auto”.

Auto Unhold When Pressing Line Key

Configures when there are multiple lines, whether to UnHold the line automatically when click the line being held and hold the line in the primary call.
Note: the hold situation which is set manually will not be put on hold automatically. The default setting is “No”.

Default Phone APP

Configures phone APP for the device. The default setting is Phone of Grandstream, but if you develop a new phone application using SDK, you can choose the new one to replace the original. The configuration of “Quickly Launch 3rd Party APP” is not available if the default phone application changes.

Quickly Launch Specified APP

Selects the specified application that launch quickly by offhook/clicking the Dial or Conference. Once selected, the package name and activity of specified application will be displayed below. When user offhooks / click the Dial or Conference application, it will automatically enter the configured application interface. The default setting is blank, which means using GS phone application.

Record Mode

Configures recording mode.

Enable Auto Record When Call Established

Configures whether to auto record when a call is established. If set to “Yes”, the call recording will start automatically when the call is established.

Rejected Call Notification

Enable/Disable rejected call notification. Once enabled, a missed call will prompt on LCD when reject the incoming call.
Default is “No”.

Return Code When Rejecting Incoming Calls

When rejecting an incoming call, the phone will send the selected type of SIP response for the call.

Return Code When DND Is Enabled 

When DND is enabled, the phone will send the selected type of SIP response

Group Listen with Speaker

Configures whether to enable speaker audio output when in a call with handset/headset. If enabled, the speaker audio output is enabled, but you cannot talk through the speaker.

One-Way Call Function Buttons

Users can configure and customize the call functions in the call interface during a normal call. The available call functions are: Mute, Hold, Video, Record, Transfer, Call Details, New Call, Keyboard, Media Channel, Conference.

Note: Users can choose only 3 of these functions to be shown on the call interface during a one-way call.

Conference Call Function Buttons

Users can configure and customize the call functions in the call interface during a conference. The available call functions are: Invite, Mute, Hold, Record, Call Details, New Call, Keyboard, Media Channel.

Note: Users can choose only 3 of these functions to be shown on the call interface during a conference.

Call Function Button Display Timeout (s)

Configures the timeout period for call function buttons display. If set to “0”, the buttons will be always displayed. The value range is 0-30 seconds.

Virtual Account Group Avaya Mode 

If set to “Yes”, when processing SIP Register 3XX Response, it will parse the address site in 3XX, modify the account server info “SIP Server: port” & “SIP Transaction” in virtual account group and initiate registration again. This feature is designed for the Avaya customers.

Virtual Account Group Concurrent Registration

Filter Characters

Sets the characters for filter when dial out numbers. Users could set up multiple characters. For example, if set to “[()-]”, when dial (0571)-8800-8888, the character “()-“ will be automatically filtered and dial 057188008888 directly.

Escape '#' as %23 in SIP URI

Determines which characters will be included in the SIP INVITE URI if end users input #. If it is set to “Yes”, the phone will replace the # by %23. Otherwise, it will include # in the SIP INVITE message.

Default is “Yes”.

Auto Config CPT by Region

Configures whether to choose Call Progress Tone automatically by region. If set to “Yes”, the phone will configure CPT (Call Progress Tone) according to different regions automatically. If set to “No”, you can manual configure CPT parameters.

The default setting is “No”.

Call Progress Tones:

Configures tone frequencies according to user preference. By default, the tones are set to North American frequencies. 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. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeats the pattern.

Please refer to the document below to determine your local call progress tones: https://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

Call-Waiting Tone Gain

Adjusts the call waiting tone volume. Users can select “Low”, “Medium” or “High”.

The default setting is “Low”.

Default Ring Cadence

Defines the ring cadence for the phone. The default setting is: c=2000/4000.

Video Frame Rate

Configures video frame rate for SIP video call from “5 frames/second”, “15 frames/second”, “25 frames/second” and “30 frames/second”. The default setting is 15 frames/second. The video frame rate is adjustable based on network conditions. Increasing the frame rate will significantly increase the amount of data transmitted, therefore consuming more bandwidth. The video quality will be affected due to packet loss if extra bandwidth is not allocated.

Configures the video display mode to "Original proportion", "Cut proportionally" or "Add black margin proportionally".

The default setting is "Equal proportional cutting".

Enable Frame Skipping in Video Decoder 

Enables the phone system for frame skipping in video decoder. If it is enabled, the video decoder will skip the P frame and start decoding from the next I frame. Enabling this option will help reduce flickering in the video when the bandwidth is limited in the network environment. 

The default setting is "Yes".

Enable OpenVPN®

This enables/disables OpenVPN® functionality and requires the user to have access to an OpenVPN® server. The default setting is No. NOTE: To use OpenVPN® functionalities, users must enable OpenVPN® and configure all the settings related to OpenVPN®, including server address, port, OpenVPN® CA, certificate and key. Additionally, the user must also set the SIP account to use “OpenVPN” for the “Nat Traversal” (under Account-> General Settings-> Network Settings).

OpenVPN® Mode

Simple mode only supports some basic or common parameters configuration; Professional mode suppports configuration file upload, which is totally customized by need, please refer to https://openvpn.net for more information.

Enable OpenVPN® Comp-lzo

Enables OpenVPN® LZO compression. When the LZO compression is enabled on the OpenVPN® server, you must turn on it at the same time. Otherwise, the network will fail to connect.

OpenVPN® Server Address

The URL/IP address for the OpenVPN® server.

OpenVPN® Port

The network port for the OpenVPN® server.

By default, it is set to 1194.

OpenVPN® Transport

Determines network protocol (UDP or TCP) used for OpenVPN®.

Default is UDP.

OpenVPN® CA

OpenVPN® CA file (ca.crt) required by the OpenVPN® server for authentication purposes. Press “Upload” to upload the corresponding file to the device. The file will not be uploaded if it is not in the correct format.

OpenVPN® Client Certificate

OpenVPN® Client certificate file (*.crt) required by the OpenVPN® server for authentication purposes. Press “Upload” to upload the corresponding file to the device. The file will not be uploaded if it is not in the correct format.

OpenVPN® Client Key

The OpenVPN® Client key (*.key) required by the OpenVPN® server for authentication purposes. Press “Upload” to upload the corresponding file to the device. The file will not be uploaded if it is not in the correct format.

OpenVPN® Cipher Method

The cipher method of OpenVPN®, must be the same cipher method used by the OpenVPN® server. Available options are: “Blowfish”, “AES-128” or “AES-256”. Default is “Blowfish”.

OpenVPN® User Name

OpenVPN® authentication username (optional).

OpenVPN® Password

OpenVPN® authentication username (optional).

NTP Server

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

NTP Update Interval (m)

This allows the user to configure interval for updating time from the NTP server. Valid value is between 5 and 1440 minutes. If set to 0, it will follow the default update rules. Default value is 0.

Allow DHCP Option 42 to Override NTP Server

Obtains NTP server address from a DHCP server using DHCP Option 42; it will override configured NTP Server. If set to “No”, the phone will use configured NTP server to synchronize time and date even if a NTP server is provided by DHCP server.

The default setting is “Yes”.

Allow DHCP Option 2 to Override Time Zone Setting

Obtains time zone setting (offset) from a DHCP server using DHCP Option 2; it will override selected time zone.

If set to “No”, the phone will use selected time zone even if provided by DHCP server. The default setting is Yes.

Enable Self-Defined Time Zone.

Enable/Disable Custom Time Zones.

Note: in order to be able to enable Custom Time zones, DHCP Option 42 and DHCP Option 2 needs to be disabled.
This option is disabled by Default.

Self-defined Time Zone 

This parameter allows the users to define their own time zone.

The syntax is: std offset dst [offset], start [/time], end [/time]

Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0

MTZ+6MDT+5

This indicates a time zone with 6 hours offset with 1 hour ahead (when

daylight saving) which is U.S central time. If it is positive (+) if the local time

zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is east.

M4.1.0,M11.1.0

The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)

The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday…)

The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, … ,Sat)

Therefore, this example is the DST which starts from the First Sunday of April to the 1st Sunday of November.

Time Display Format 

Specifies which format will be used to display the time.

It can be selected from 12-hour and 24-hour format.

Date Display Format 

Determines which format will be used to display the date.

It can be selected from the drop-down list.

Normal (M/D/YYYY): 1/31/2012
YYYY/MM/DD: 2012/01/31
MM/DD/YYYY: 01/31/2012
DD/MM/YYYY: 31/01/2012
The default setting is MM/DD/YYYY.

Language

Language

Sets the language to display on the phone’s LCD.

Select Language File

Press “Browse” to bring up a file selection menu to select the local .txt file to upload to the phone

LCD & LED Management

Enable Missed Call Backlight

If set to “Yes”, LCD backlight will be turned on when there is a missed call on the phone.

The default setting is “Yes”.

Note: Reboot is required for the setting to take effect

Enable Missed Call Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when there is missed call on the phone.

Default setting is “Yes”.

Enable MWI Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when there is new voicemail on the phone. If it set to “No”, the LED indicator will keep off if the phone system receives SIP NOTIFY message about unread voice mail.

The default setting is “Yes”.

Enable New Message Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when there is new message on the phone.

Default setting is “Yes”.

Enable Contact Full Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when the contact storage or message storage is full.

The default setting is “Yes”.

Enable Indicator When LCD is Off

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when the LCD screen is off. If it set to “No”, the LED indicator will keep off when the LCD screen is off.

The default setting is “Yes”.

Screen Timeout

Configures the timeout interval of the LCD backlight. If set to "never", the screen will always stay on.

Screensaver Timeout

Configures the screensaver timeout.

Peripherals Interface Management

Enable Touch Keys

Configures whether to enable touch keys. Once disabled, the five keys "Volume-, Volume +, Home, Menu and Back" will not take effect. 

Default is enabled.

Enable Local Camera

Configures whether the Local camera will be activated or not.
When this option is disabled, the camera cannot be turned on by any app for sending video. By default it’s enabled.

Audio Control

RJ9 Headset TX Gain (dB)

Configures the Transmission Gain in RJ9 headset channel.

It can be selected from the dropdown list. The default setting is 0dB:

-24
-18
-12
-6
0
+6
+12
+18
+24

Configures the transmission gain of the 3.5mm earphone. It controls the audio signal sent out from the phone.

Note: not available with GXV3450

Headset Type

Specifies which type of headset will be connected to the phone system. It can be selected from the dropdown list:

Normal Headset
Plantronics EHS
If a normal RJ11 headset is connected, it should set to “Normal Headset”. If a Plantronics EHS headset is used, it should set to “Plantronics EHS”

If set to "Yes", the headset can control the onhook and offhook.

Note: not available with GXV3450

Handset TX Gain (dB)

Configures the transmission gain of the handset. Default setting is “-6dB”.

-4
-2
0
+2
+4
+6

Enable Handset Noise Shield 2.0 

When Handset Noise Shield feature is enabled, the remote party will hear less environmental noise during a call. If set to "High Shielding", most of the environmental noise can be shielded. If set to "Soft Shielding", some environmental comfort noise will remain for the remote party.

Handfree TX Gain (dB)

This feature configures the transmission gain for handsfree mode. Default setting is “0dB”.

-16
-12
-8
-4
0
+4
+8
+12
+16

Media Volume

Configures the volume of media.

Alarm Volume

Configures the volume of alarm.

Ringtone Volume

Configures the volume of ringtone.

Configure Manually

Download Device Configuration 

Downloads the phone’s configuration file in text format. The config file includes all the P value parameters for phone’s current settings except password for security purpose. Users can use the Grandstream configuration file generator to generate binary config file from this text file.

Upload Device Configuration

Uploads configuration file to the phone.

Notes:

•The GXV34x0 supports only txt format for config file upload.

•Reboot is required to take effect.

Configure via Network

Use Grandstream GAPS

It is used to configure the download path and update mode for the configuration file server.

Config Update Mode

Selects provisioning method: TFTP, HTTP or HTTPS. Default setting is “HTTPS”.

Config Server Path

Sets IP address or domain name of configuration server. The server hosts a copy of the configuration file to be installed on the phone.

Users can also configure variables in the provisioning server URL; The function allows users to configure a unified provisioning server URL for different phone models. Currently, the following variables are supported in the provisioning server URL:

Variables $PN and $MAC can be embedded in server URL setting in Web UI and also in DHCP Option 66.

Default settings is “fm.grandstream.com/gs”.

HTTP/HTTPS Username

Configures the user name for the config HTTP/HTTPS server.

HTTP/HTTPS Password

Configures the password for the config HTTP/HTTPS server.

Config File Prefix

Checks if configuration files are with matching prefix before downloading them.

This field enables user to store different configuration files in one directory on the provisioning server.

Config File Postfix

Checks if configuration files are with matching postfix before downloading them.

This field enables user to store different configuration files in one directory on the provisioning server.

Authenticate Config 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.

The default setting is “No”.

XML Config File Password

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

Start provision

Start Provision

After setting up the configuration, Click to start provision.

Automatic upgrade

Automatic Upgrade 

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

Note: Day of week is starting from Sunday.

The default setting is “No”.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent.

Upgrade with Prompt

If set to “Yes”, the phone will pop up a prompt after downloading the firmware files to confirm whether start upgrading. Otherwise, the phone will automatically start upgrading process.

The default setting is “Yes”.

DHCP Option

Allow DHCP Option 150, 43, 160 and 66 Override Server

Obtains configuration and upgrade server’s information from DHCP server using options 43, 160 and 66.

If DHCP option 43, 160 and 66 is enabled on the LAN side, the device will reset the CPE, upgrade, network VLAN tag, and priority configuration according to option 43 sent by the server. At the same time, the update mode and server path of the configuration upgrade mode will be reset according to the option 160 and 66 sent by the server.

If set to "Prefer, fallback when failed", the device can fallback to use the configured provisioning server under its Firmware and Config server path in case the server from the DHCP Option fails.

Default setting is “Prefer, fallback when failed”.

Allow DHCP Option 120 to Override SIP Server

Configures the phone system to allow the DHCP offer message to override the Config Server Path via the Option 120 header.

Default setting is “Yes”.

Allow DHCP Option 242 (Avaya IP Phones) 

Enables DHCP Option 242. Once enabled, the phone will use the configuration info issued by the local DHCP in Option 242 to configure proxy, transport protocol and server path.

The default setting is “Yes”.

Config provision

Download And Process All Available Config Files

By default, the device will provision the first available config in the order of cfgMAC.xml,  cfgMODEL.xml, and cfg.xml,(corresponding to device specific, model specific, and global configs). If set to "Yes", the device will download and apply (override) all available configs in the order of , cfg.xml, cfgMODEL.xml, cfgMAC.xml

Default is “No”.

Config Provision

Device will download configuration files and provision by configured order.

PNP Feature

PNP (3CX) Auto Provision 

Sets the phone system to broadcast the SIP SUBSCRIBE message during booting up to allow itself to be discovered and be configured by the SIP platform.

The default setting is “Yes”.

Send HTTP Basic Authentication By Default

Determine whether to send basic HTTP authentication information to the server by default when using wget to download firmware or config file. If set to "Yes", send HTTP/HTTPS user name and password no matter the server needs authentication or not. If set to "No", only send HTTP/HTTPS user name and password when the server needs authentication.

Enable SIP NOTIFY Authentication

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

Enabled Authentication Server Validation

Configures whether to validate the server certificate when downloading the firmware/config file. If enabled, the phone will download the firmware/config file only after the server is validated.

CA Certificate

Select the CA certificate for server verification.

User Certificate

Select the user certificate to be used for mutual server authentication. If the private key is included, upload it with the user certificate.

Enable EEE Mode

Configures enable/disable EEE (Energy-Efficient Ethernet) mode. If set to “Yes”, the phone will turn on the EEE mode.
Note: Regardless of whether the EEE mode is turned on or off, the network will reconnect.
Default is “No”.

mDNS Override Server

Sets the phone system to broadcast the Multicast DNS (mDNS) message during booting up to allow itself to be discovered and be configured by the SIP platform. If it is set to “User Type A’, the phone system will broadcast the MDNS message “A_grandstream-cfg.local”; if it is set to “Use Type SRV”, the MDNS message will be “SRV_grandstream-cfg.local”.

The default setting is “Use Type A”.

Factory Reset

Resets the phone system to the default factory setting mode.