GXV3480

GXV3470

GXV3450

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

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 14

Applications Deployment

Supports Android 14 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

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. 

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, 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

7″ 1280×800 capacitive touch screen (5 points) HD IPS LCD

Camera

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

Bluetooth

Yes, integrated Bluetooth

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 headset port, USB 2.0 port, USB 3.0 port, HDMI-out

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 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 & fail-over

Sample Applications

Android

Runs Android 14

Applications Deployment

Supports Android 14 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, built-in 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 GXV3470’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

GXV3470 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 

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

Wi-Fi

Yes, dual-band dual-stream Wi-Fi (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 272×480 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 14

Applications Deployment

Supports Android 14 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
Note: Users can remotely access the GXV3450’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

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

GXV3480

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

Product Model

Identifies the exact IP phone model.

Hardware Version

Indicates the device’s hardware revision.

Part Number

Manufacturer-assigned product identifier.

Serial Number

Unique hardware identifier for the device.

System Version

Firmware version ID. This is the main software release version, which is used to identify the software system of the phone.

Boot Version

Version of the device bootloader code.

Kernel Version

Version of the system’s OS kernel.

CPE Version

Firmware version related to CPE functions.
Note: Supports configuration tasks via GDMS (Grandstream Device Management System), this enables remote provisioning and centralized device management.

Device Individual Certificate

Built-in unique certificate for secure provisioning.

System Up Time

Elapsed time since the last device reboot.

Account Registration

Account Active

Indicates whether the account is active. 
Disabled by default.

Account Name

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

SIP Server

Specifies the URL or IP address, and port of the SIP server. This should be provided by VoIP service provider (ITSP).

SIP User ID

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

SIP Authentication ID

Configures the SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from the SIP User ID.

SIP Authentication Password

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

Display Name

Specifies the SIP server subscriber’s name (optional) that will be used for Caller ID display. The configured content will be included in the From, Contact and P-Preferred-Identity headers of SIP INVITE message.

Tel URI

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

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

Voice Mail Access Number

Sets if the phone system allows users to access the voice messages by pressing the MESSAGE key on the phone. This ID is usually the VM portal access number. For example, in UCM6xxx IPPBX, *97 could be used.

Network Settings

Outbound Proxy

Configures the IP address or the domain name of the primary outbound proxy, media gateway or session border controller. It’s used by the phone for firewall or NAT penetration in different network environments. If a symmetric NAT is detected, STUN will not work and only an outbound proxy can provide a solution

Secondary Outbound Proxy

Sets IP address or domain name of the secondary outbound proxy, media gateway or session border controller. The phone system will try to connect the Secondary outbound proxy only if the primary outbound proxy fails.

DNS Mode

Defines which DNS service will be used to lookup IP address for SIP server’s hostname. There are three modes: 

To locate the server by DNS SRV set this option to “SRV” or “NATPTR/SRV”. Default setting is “A Record”.

Maximum Number Of SIP Request Retries

This setting configures the maximum number of retries for the device to send requests to the server. In DNS SRV configuration, if the destination address does not respond, all request messages are resent to the same address according to the configured retry times.
Valid range: 1-10.
The default setting is “2”.

DNS SRV Failover Mode

Configures the preferred method for DNS SRV failover. There are Three DNS SRV Failover mode: 

Failback Expiration(m)

This option is configured when DNS SRV failover mode “Failback follows failback expiration timer” is chosen. It specifies the duration (in minutes) since failover to the current SIP server or Outbound Proxy before making failback attempts to the primary SIP server or Outbound Proxy.
Valid range: 1 – 64800
The default setting is “60”.

Register Before DNS SRV Fail-over

To set register before DNS SRV Fail-over.

Disabled by default.

NAT Traversal

Specifies which NAT traversal mechanism will be enabled on the phone system. 

The default setting is “Auto”.

Media NAT Traversal

Configures the media NAT traversal method.

The default setting is “Auto”.

Candidate Type

Configures the ICE candidate type when Media NAT Traversal is set to “ICE”.

The default setting is “All”.

STUN/TURN Server

Configures the STUN/TURN server settings when Media NAT Traversal is enabled.

Proxy-Require

Adds the Proxy-Required header in the SIP message. It is used to indicate proxy-sensitive features that must be supported by the proxy. Do not configure this parameter unless this feature is supported on the SIP server.

Parameter

Description

SIP Basic Settings

SIP Registration

Allows the phone system to send SIP REGISTER messages to the proxy/server.

The default setting is “Yes”.

Unregister before New Registration

Controls whether to clear SIP user’s information by sending un-register request to the proxy server.

The default setting is “Instance”.

Register Expiration (m)

Configures the time period (in minutes) in which the phone refreshes its registration with the specified registrar. The default setting is 60.

The maximum value is 64800 (about 45 days).

Subscribe Expiration (m)

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

Default value is 60.

Re-register before Expiration (s)

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

The range is from 0 to 64,800.

Registration Retry Wait Time (s)

Configures the time period (in seconds) in which the phone will retry the registration process in the event that is failed. The default setting is 20.

The maximum value is 3600 (1 hour).

Add Auth Header on Initial REGISTER

Configure if the SIP account needs to add Auth header in RE-REGISTER.

Enable SIP OPTIONS Keep Alive

Enables SIP OPTIONS to track account registration status so the phone will send periodic OPTIONS message to server to track the connection status with the server.

The default setting is “No”.

SIP OPTIONS Keep Alive Interval (s)

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

The default range is 1-64800.

SIP OPTIONS Keep Alive Maximum Tries

Configures the maximum times of sending OPTIONS message consistently from the phone to server. Phone will keep sending OPTIONS messages until it receives response from SIP server. The default setting is “3”, which means when the phone sends OPTIONS message for 3 times, and SIP server does not respond this message, the phone will send RE-REGISTER message to register again.

The valid range is 3-10.

Subscribe for MWI

Configures the phone system to subscribe voice message service. If it is set to “Yes”, the phone system will periodically send SIP SUBSCRIBE message for Message Waiting Indication service. GXV3480 phone system supports both synchronized and non-synchronized MWI.

The default setting is “No”.

Use Privacy Header

Determines if the Privacy header will be presented in the SIP INVITE message and if it includes the caller info in this header. If it is set to “Default”, the Privacy Header will be omitted in INVITE when “Huawei IMS” special feature is active. If set to “Yes”, it will always be presented. If set to “No”, it will always be omitted.

The default setting is “Default”.

Use P-Preferred-Identity Header

Specifies if the P-Preferred-Identity Header will be presented in the SIP INVITE message. If set to “default”, the P-Preferred-Identity Header will be omitted in SIP INVITE message when “Huawei IMS” special feature is active. If set to “Yes”, the P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted.

The default setting is “Default”.

Use P-Access-Network-Info Header

Enables/disables the use of P-Access-Network-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header.

Default setting is “No”.

Use P-Emergency-Info Header

Enables/disables the use of P-Emergency-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header.

Default setting is “No”.

Use P-Asserted-Identity Header

Includes P-Asserted-Identity header in SIP request.

Use Zoom E911 X-Switch-Info SIP Header

Configure whether the “Zoom E911 X-Switch-Info SIP Header” is present in the SIP register message.

Use MAC Header

Configures whether to use MAC header. If set to “No”, all SIP REGISTER messages will not contain MAC header. If set to “Only for REGISTER”, MAC header will only be included in SIP REGISTER messages. If set to “Yes to All SIP”, all outgoing SIP messages will contain MAC header.

Add MAC in User-Agent

Configures whether to add MAC address in User-agent header. If set to “No”, all outgoing SIP messages will not attach MAC address to the User-agent header. If set to “Yes except REGISTER”, all outgoing SIP messages except REGISTER message will attach the MAC address to the User-agent header. If set to “Yes to All SIP”, all outgoing SIP messages including REGISTER message will attach MAC address to the User-agent header.

SIP Transport

Sets the network protocol used for the SIP transport. Users can choose from UDP/TCP/TLS.

Local SIP Port

Configures the local SIP port used for transmit and receive SIP messages. If set to 0, the phone will use a random port to initiate registration.

SIP URI Scheme When Using TLS

Specifies whether “sip:” or “sips:” will be used when TLS/TCP is selected for SIP Transport.

Use Actual Ephemeral Port in Contact with TCP/TLS

Configures whether to use the actual ephemeral port when TCP/TLS is selected for SIP Transport.

Support SIP Instance ID

Configures whether to support SIP Instance ID.

SIP T1 Timeout

Configures SIP T1 timeout. T1 is a transaction round-trip delay (RTT) between the server and the client. If the network latency is high, choose a larger value to ensure stable use.

SIP T2 Interval

Configures the SIP T2 timeout. The SIP T2 timeout is the time interval between the INVITE response and the non-INVITE request.

SIP Timer D Interval

Configures the wait time for response retransmissions when the INVITE receives a 3xx to 6xx response. Valid range is 0-64 seconds. If set to 0, the parameter will not take effect, and the true time interval is equal to T1*64.

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.

Enable 100rel

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

Use Route Set in NOTIFY (Follow RFC 6665)

Configures whether to use route set in NOTIFY (follow RFC 6665). If enabled, the Request URI of the refresh/cancel subscription will use the URI in the received NOTIFY Contact (RFC 6665); otherwise, the URI in the previously subscribed 200 OK Contact will be used.

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 enabled, the phone will use the related parameters when sending session timer according to “Session Expiration”. If disabled, session timer will be not be used.

Session Expiration (s)

Configures the phone system’s SIP session expiration. Session Expiration is the time (in seconds) when the session is considered timed out, provided that no successful session refresh transaction occurs beforehand. Valid range is 90-64800.

Min-SE (s)

The minimum session expiration (in seconds).

UAC Specify Refresher

As a Caller, if set to “UAC”, the phone will be the refresher. If set to “UAS”, the callee or proxy server will be the refresher. If set to “Omit”, the refresher will not be specified.

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.

Caller Request Timer

If enabled and the remote party supports session timer, the phone will enable the session timer feature when it makes outbound calls.

Callee Request Timer

If enabled and both parties support session timer, the phone will use session timer feature when it receives inbound calls.

Force Timer

If enabled, the phone will use the session timer even if the remote party does not support this feature. Otherwise, the session timer will be enabled only when the remote party supports this feature.

Force INVITE

The session timer can be refreshed using the INVITE method or the UPDATE method. If enabled, the INVITE method will be used to refresh the session.

Preferred Audio Codec

Preferred Audio Codec

Lists the available and enabled Audio codecs for this account. Users can enable the specific audio codecs by moving them to the selected box and set them with a priority order from top to bottom.
This configuration will be included with the same preference order in the SIP SDP message.
The supported audio codecs are:

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

Configures the phone to use the first matching codec in the 200OK message.
The default value is 0.

ILBC Frame Size

Sets the ILBC (Internet Low Bitrate Codec) frame size if ILBC is used. Users can select it from 20ms or 30ms.
The default setting is 30ms.

G726-32 ITU Payload

Configures G726-32 payload type for ITU packing mode. Payload 2 is static and payload dynamic is dynamic.
The default setting is “2”.

G726-32 Dynamic PT

Specifies the G726-32 payload type, and the valid range is 96 to 126.

The default setting is “126”.

Opus Payload Type

Defines the desired value (96-126) for the payload type of the Opus codec.
The default value is 123.

DTMF

Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency) signals.
There are 3 supported modes: in audio, RFC2833, or SIP INFO.

The default setting is “RFC2833”.

DTMF Payload Type

Configures the RTP payload type that indicates the transmitted packet contains DTMF digits. Valid range is from 96 to 126.
Default value is 101.

Enable Audio RED with FEC

If set to “Yes”, FEC will be enabled for audio call.
The default setting is “No”.

Audio FEC Payload Type

Configures audio FEC payload type. The valid range is from 96 to 126.
The default value is 121.

Audio RED Payload Type

Configures audio RED payload type. The valid range is from 96 to 126.
The default value is 124.

Silence Suppression

Enables the silence suppression/VAD feature. If it is 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.
If set to “No”, this feature is disabled. The default setting is “No”.

Voice Frames Per TX

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

Preferred Video Codec

Preferred Video Codec

Lists the available and enabled Video codecs for this account. Users can enable the specific video codecs by moving them to the selected box and set them with a priority order from top to bottom. This configuration will be included with the same preference order in the SIP SDP message.
The supported video codecs are:

Enable Video FEC

When enabled, the video sender will temporarily allocate part of the bandwidth to one data channel to send FEC data to system, thus to improve the video quality the receiver gets. Enabling this function will take up part of bandwidth and reduce call rate.
The default setting is “Yes”.

Enable RFC5168 Support

Enables/disables RFC5168 mechanism for video calls. RFC5168 allows SIP party to request the sender to refresh its video frame in H.264, or refresh the full picture in VP8.
The default setting is “Yes”.

Video FEC Mode

If set to 0, FEC is not sent by separate port. If set 1, FEC is sent by separate port.
Default setting is 0.

FEC Payload Type

Configures FEC payload type. The range is 96-126. Default setting is 120.

Packetization Mode

Set video packetization mode. If set to “Single NAL Unit Mode”, the packetization mode will be negotiated as single NAL unit mode when dial video calls, if the other party does not support the negotiation, then single NAL unit mode will be used for video encoding by default. If set to “Non-Interleaved Mode”, the packetization mode will be negotiated as Non-interleaved mode when dial video calls, If the other party does not support negotiation, then the Non-interleaved mode will be used for for video encoding by default; If set to “Prefer Non-Interleaved Mode”, the packetization mode will be negotiated as prefer Non-interleaved mode when dial video calls, if the other party does not support the negotiation, then prefer Non-interleaved mode will be used for video encoding by default.

H.264 Image Size

Sets the H.264 image size. It can be selected from the dropdown list.

Note: For some network environment, the default setting “720P” might be too high that causes no video or video quality issue during video call. In this case, please change “H.264 Image Size” to “VGA” or “CIF” and change “Video Bit Rate” to “384kbps” or lower. The default setting is 720P.

Use H.264 Constrained Profiles

Configures that whether to set H.264 constrained profiles.
The default setting is “No”.

H.264 Profile Type

Selects the H.264 profile type from the dropdown list.

Note: Lower levels are easier to decode, but higher levels offer better compression. Usually, for the best compression quality, choose “High Profile”; for playback on low-CPU machines or mobile devices, choose “Baseline Profile”. If “BP/MP/HP” is selected, all three profiles “Baseline Profile” “Main Profile” and “High Profile” will be used for negotiation during video decoding to achieve the best result. This is usually used in video conference when there is higher requirement on the video.

Video Bit Rate

Configures the bit rate for video call. It can be selected from the dropdown list. The default setting is 2048 kbps. The valid range is from 32 – 2048 kbps.
Note: The video bit rate can be adjusted based on the network environment. Increasing the video bit rate may improve video quality if the bandwidth is permitted. If the bandwidth is not permitted, the video quality will decrease due to packet loss. For some network environment, the default setting “720P” might be too high that causes no video or video quality issue during video call. In this case, please change “H.264 Image Size” to “VGA” or “CIF” and change “Video Bit Rate” to “384kbps” or lower.

SDP Bandwidth Attribute

Sets the SDP bandwidth attribute. It can be selected from the drop-down list. The default setting is “Media Level”.

Note: Please do not modify this setting without knowing the session format supported by the server. Otherwise, it might cause video decoding failure.

H.264 Payload Type

Specifies the H.264 codec message payload type format. The default setting is 99.
The valid range is from 96 to 127.

Presentation Settings

Packet Retransmission

If set to “NACK”, the signaling will carry NACK info. After negotiation, the media will use NACK to retransmit lost packets. If set to “NACK+RTX (SSRC-GROUP)”, the signaling will carry both NACK and RTX info. After negotiation, the media will use NACK+RTX (SSRC-GROUP) to achieve packet loss retransmission. If set to “Disabled”, packet loss retransmission cannot be used.
The default setting is “NACK+RTX (SSRC-GROUP)”.

Enable BFCP

If set to “Yes”, the device will be able to receive the presentation stream in video calls and video meetings.

Initial INVITE with Media Info

Initial INVITE SDP contains presentation media.

Presentation H.264 Image Size

Selects the H.264 image size. Users can select 1080P or 720P.

Presentation H.264 Profile Type

Select the Presentation H.264 Profile Type from “Baseline Profile”, “Main Profile”, “High Profile” and “BP&MP&HP”. The default setting is “BP&MP&HP”.
The lower the profile type is, the easier the packet can be decoded. However, higher level has high compression ratio. For device with low CPU, select “Baseline Profile” to play record; “Baseline Profile” is more likely to be used in a video conference that has high demanding for the video quality. Select among the three types to achieve best video effect.

Presentation Video Bit Rate

Configures the bit rate of the video. The video bit rate can be adjusted based on the network environment. Increasing the video bit rate may improve video quality if the bandwidth is permitted. If the bandwidth is not permitted, the video quality will decrease due to packet loss. Video Bit Rate can be set to integer value from 512kbps to 2048kbps.

Presentation Video Frame Rate

Configure the video frame rate for presentation.

BFCP Transport Protocol

Defines the transport protocol used for BFCP. Users can choose from Auto/UDP/TCP. The default setting is “UDP” first, if not supported, then choose “TCP”. If choose “Auto”, automatically switches between “UDP” and “TCP”.

RTP Settings

SRTP Mode

Sets if the phone system will enable the SRTP (Secured RTP) mode. It can be selected from dropdown list:

SRTP uses encryption and authentication to minimize the risk of denial of service. (DoS). If the server allows to use both RTP and SRTP, it should be configured as “Enabled but not forced”.
Selecting “Optional” allows SRTP to be used but does not require it. The phone negotiates with connected devices to enable SRTP when possible; if agreed, the communication is encrypted and authenticated, otherwise it proceeds without SRTP.
The default setting is “Disable”.

SRTP Key Length

Configures all the AES (Advanced Encryption Standard) key size within SRTP. It can be selected from dropdown list:

If it is set to “AES 128 & 256 bit”, the phone system will provide both AES 128 and 256 cipher suite for SRTP. If set to “AES 128 bit”, it only provides 128-bit cipher suite; if set to “AES 256 bit”, it only provides 256-bit cipher suite. The default setting is “AES128&256 bit”.

Enable SRTP Key Lifetime

Defines the SRTP key lifetime. When this option is set to be enabled, during the SRTP call, the SRTP key will be valid within 231 SIP packets, and phone will renew the SRTP key after this limitation. Default is “Yes”.

RTCP Destination

Configures a remote server URI where the RTCP messages will be sent to during an active call.

Symmetric RTP

Configures if the phone system enables the symmetric RTP mechanism.
If it is set to “Yes”, the phone system will use the same socket/port for sending and receiving the RTP messages. The default setting is “No”.

RTP IP Filter

Receives the RTP packets from the specified IP address and Port by communication protocol. If it is set to “IP Only”, the phone only receives the RTP packets from the specified IP address based on the communication protocol; If it is set to “IP and Port”, the phone will receive the RTP packets from the specified IP address with the specified port based on the communication protocol. The default setting is “Disable”.

RTP Timeout Timer (s)

Disconnects the call automatically when there is no RTP stream for a specific timeout. Default is 30 seconds.

VQ RTCP-XR Collector Name

Configures the host name of the RTCP server that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Address

Configures IP address of the RTCP server that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Port

Configures the port of the RTCP server that accepts voice quality reports contained in SIP PUBLISH messages.

Call Settings

Video Call Mode

Configures the video call behavior for this account.

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.

Transfer to VM Feature Code

During a call, press the transfer to voicemail softkey, enter the extension number, and then transfer the call to the extension voicemail. At this point, the code will be used as an extension prefix and called into the server to complete the transfer to voicemail event.
Note: The default feature code is “*”.

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

Call Forward Type

Sets the Call Forwarding feature for this account. Available options are:

Forward All To 

Specifies the number to forward calls to when Call Forward Type is set to “Unconditional”.

Schedule

Configures the time period during which calls are forwarded when Call Forward Type is set to “Time-based”. The time is in 24-hour format (HH:mm), for example, 10:00.

In Schedule Forward To

Specifies the number to forward calls to during the configured Schedule when Call Forward Type is set to “Time-based”.

Outside Schedule Forward To

Specifies the number to forward calls to outside the configured Schedule when Call Forward Type is set to “Time-based”.

Enable Busy Forward

Enables call forwarding when the phone is busy and Call Forward Type is set to “Other”.

Forward Busy To

Specifies the number to forward calls to when the phone is busy and Call Forward Type is set to “Other”.

Enable No Answer Forward 

Enables call forwarding when a call is not answered and Call Forward Type is set to “Other”.

Forward No Answer To

Specifies the number to forward calls to when a call is not answered and Call Forward Type is set to “Other”.

No Answer Timeout (s)

Configures the timeout, in seconds, before an unanswered call is forwarded when Call Forward Type is set to “Other”.

Enable DND Forward 

Enables call forwarding when Do Not Disturb (DND) is enabled and Call Forward Type is set to “Other”.

Forward DND To

Specifies the number to forward calls to when Do Not Disturb (DND) is enabled and Call Forward Type is set to “Other”.

Dial Plan

Dial Plan Prefix

Configures the digits prepended to the dialed number.

Disable Dial Plan

Enables or 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.

Security Settings

Check Domain Certificates

Sets the phone system to check the domain certificates if TLS/TCP is used for SIP Transport.

The default setting is “No”.

Trusted Domain Name List 

Configures a list of up to 10 trusted domain names used for TLS certificate verification via Subject Alternative Name (SAN). If a domain matches an entry in the trusted list, the certificate is considered trusted. The list supports alphanumeric characters and symbols such as “-”, “.”, and “”. It allows wildcard entries “.grandstream.com” and subdomains “.grandstream.com”. By default, the remote proxy domain name and SIP server domain name are included as trusted.

Validate Certification Chain

Configures whether to validate certification chain, when TLS/TCP is configured for SIP Transport. If this is set to “Yes”, phone will validate server against the new certificate list.

The default setting is “No”.

SIP CA Certificate

Select the CA certificate for server verification.

SIP User Certificate

Select the user certificate to access SIP TLS authentication content required by some specific servers. If the private key is included, upload it with the user certificate.

Validate Incoming SIP Messages

Specifies if the phone system 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

It is used to configure whether to dial the number carried by Refer-to after receiving SIP REFER request actively. The default is “Disabled”.

Accept Incoming SIP from Proxy Only

If enabled, the SIP address of the request URL in the incoming SIP message will be checked. If it doesn’t match the SIP server address of the account, the request will be ignored.

Check SIP User ID for Incoming INVITE

Configures the phone system to check the SIP User ID in the Request URI of the SIP INVITE message from the remote party. If it doesn’t match the phone’s SIP User ID, the call will be rejected.

The default setting is “No”.

Allow SIP Reset

It is used to configure whether to allow SIP Notification message to perform factory reset on the phone.
The default setting is “No”.

Authenticate Incoming INVITE

Configures the phone system to authenticate the SIP INVITE message from the remote party. If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response.

The default setting is “No”.

SIP Realm used for Challenging INVITE & NOTIFY

Configure this item to validate incoming INVITE, but you must enable authenticate incoming INVITE first to make it take effect. You can verify the NOTIFY information for the provision, including check- sync, resync and reboot, but only when SIP NOTIFY authentication enabled first to make it take effect.

MOH

Enable MOH

Enables/Disables the MOH funtion.

MOH Mode

Configures MOH mode. If set to “Local MOH”, a local MOH audio file needs to be uploaded for this mode to work.

Upload Local MOH Audio File

Loads the MOH (Music on Hold) file to the phone. Click on “Browse” button to upload the music file from local PC. The MOH audio file has to be in .wav or .mp3 format.

Note: Please be patient while the audio file is being uploaded. It could take more than 3 minutes to finish the uploading especially the file size is large. The button will show as “Processing” during the uploading. Once done, it will show as “Browse” again. Click on “Save” on the bottom of the web page and “Apply” on the top of the web page to save the change.

Advanced Features

Virtual Account Group

It is used to set to categorize accounts in server mode groups, the accounts in the same group will be combined as one and the account widget will display the Caller ID in the account with lowest ID. The phone can answer any incoming calls to each account in groups.If user makes an outbound call, the phone system will use the lowest ID account by default. If the account fails or SIP INVITE message is timeout, the phone system will failover to the next account in the group with higher account ID.If all the accounts are not available in the group, the phone system will traverse all the accounts in the group and notify the end users the session is failed.

SIP RGISTER message will use the next SIP server in the same virtual group after the fifth trial fails and cannot be configured or changed.

Allow Sync Phonebook via SIP Notify

Allows users to synchronize XML phonebook upon receiving SIP NOTIFY message with header Event: sync-contacts

Note: Received SIP NOTIFY will be first challenged for authentication purpose before contacting configured server to download XML phonebook. The parameters used are the ones configured at [Download Contacts]. The authentication can be done either using admin credentials (if no SIP account is configured) or using SIP account credentials. Default is “Yes”.

Special Features

Configures special features to meet vendor server requirements. Users can select a profile depending on the system in use, including Standard, BroadSoft, China Mobile, ZTE IMS, Mobotix, ZTE NGN, Huawei IMS, NEC, WorldStone, MetaSwitch, Telefonica Spain, and Cisco.

Additional modes include SDP Lite Mode for simplified SDP handling and improved media negotiation, and Zoom for integration with Zoom services and platform-specific features.

The default setting is “Standard.”

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.
Disabled by Default.

Reboot on Check-sync Event without “reboot=”

If option is enabled, the phone will do reboot when receiving SIP NOTIFY check-sync event without “reboot=” header. If the option is disabled, the phone will do provision and not reboot when receiving SIP NOTIFY check-sync event without “reboot=” header.
Enabled by default.

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.
Disabled by Default.

Hoteling Event

Enables BroadSoft Hoteling Event feature.
Disabled by Default.

Call Center Status

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

Enable Call Monitor Function

Enable the call monitor function.

Note: This option only appears when Special Feature parameter is to “Zoom”.

Support Call Monitor Privacy Mode

The configuration supports the privacy mode for call monitor, which can be manually switched on or off on the LCD after it is enabled.

Note: This option only appears when Special Feature parameter is to “Zoom”.

Barge-in Feature Code

Configure the call monitor barge-in feature code. The accept code can consist of at least 1 digit and 16 digits at most.

Note: This option only appears when Special Feature parameter is to “Zoom”.

Listen Feature Code

Configure the call monitor listen feature code. The accept code can consist of at least 1 digit and 16 digits at most.

Note: This option only appears when Special Feature parameter is to “Zoom”.

Whisper Feature Code

Configure the call monitor whisper feature code. The accept code can consist of at least 1 digit and 16 digits at most.

Note: This option only appears when Special Feature parameter is to “Zoom”.

Takeover Feature Code

Configure the call monitor takeover feature code. The accept code can consist of at least 1 digit and 16 digits at most.

Note: This option only appears when Special Feature parameter is to “Zoom”.

SCA

Enable SCA (Shared Call Appearance)

Enables/disables the Shared Call Appearance (the Broadsoft Standard) feature for this account. If it is set to “Yes”, the phone system can update and share account status with another device.
The default setting is “No”.
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.
The 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.

Basic Settings

Local RTP Port

Defines the local RTP-RTCP port pair used to listen and transmit.

The following rule is applied: N>=0

Audio RTP port: Port_Value+10*N
Audio RTCP port: Port_Value+10*N+1
Video RTP port: Port_Value+10*N+2
Video RTCP port: Port_Value+10*N+3
BFCP Protocol port: Port_Value+10*N+6
BFCP RTP port: Port_Value+10*N+8
BFCP RTCP port: Port_Value+10*N+9.

The default value is 50040. The valid range is from 10000 to 65000.

Use Random Port

Forces the phone system to use random ports for both SIP and RTP messages. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is “No”.

Note: This parameter must be set to “No” for Direct IP Calling to work.

Hide User Info for Video Call

Configures whether to display user information in a video call. If set to “Yes”, user information will not be displayed in the upper left corner of video area during a video call.

Enable in-call DTMF display

Enables/disables the phone system to omit the DTMF digits displaying from the LCD screen.

The default setting is “Yes”.

Enable LDAP Timeout Auto Search 

Configures whether to display the matched content automatically in search of the LDAP contacts when timeout. If set to “No”, users need to click the “Search” button to search the matched contacts mentioned above.

The default setting is “Yes”.

Keep-alive Interval (s)

Specifies how the phone system will send a Binding Request packet to the SIP server in order to keep the “ping hole” on the NAT router to open. The default setting is 20 seconds.

The valid range is from 10 to 160.

STUN Server

Configures the URI of STUN (Simple Traversal of UDP for NAT) server. The phone system will send STUN Binding Request packet to the STUN server to learn the public IP address of its network. Only non-symmetric NAT routers work with STUN.

Use NAT IP

Configures the IP address for the Contact header and Connection Information in the SIP/SDP message. It should ONLY be used if it’s required by your ITSP.

The default setting is keep the box blank.

Delay Registration

Configure the specific time for account registration after booting up to avoid centralized registration. Set the range from 0 to 90s

default value is 0.

 Guest Features

Guest Login

Enables a temporary SIP user login function; requires defining a SIP domain. After reboot, the phone prompts for a guest SIP username and password on the LCD.
The option is disabled by default, and requires a reboot to take effect.

Guest Login Timeout

Sets the inactivity period after which a guest user is automatically logged out. Helps ensure session security and prevents unauthorized use.

Guest Login PIN Code

When timeout is disabled (set to “Never”), this PIN secures access by requiring it before guest login is permitted, adding a layer of authentication.

SIP Domain

Server Alias

A user-friendly label for identifying the SIP server (e.g., “Office SIP” instead of an IP or domain). Simplifies multi-server configuration management.

Server List

Lists up to 5 SIP domains tied to a server alias, ordered by registration priority. The phone attempts to register top-down if multiple domains are available.

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”.

Enable 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”.

Return Code for DND 

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

Enable Local In-call DTMF Tone in Speaker Mode 

Configures whether to play local DTMF tone during call when using speaker.

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 Target in 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).

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 and attended 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”.

Auto Unhold When Pressing Line Key 

When there are multiple calls and line key is pressed, this configures whether to resume the line automatically when pressing the line on hold and put the current line on hold. Note: If this is a manual hold operation rather than switching lines, the call will not be resumed automatically.

Preferred Default Account 

Configures 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. Is set to “Auto”, the phone will select the last used account to dial.

Call/Pickup

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.

At any time, the GXV3480 user can press the “Answer” button. If done, then the phone will send the SIP 200OK and call will be fully established.

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.

Rejected Call Notification

Configures whether to enable rejected call notification. Once enabled, a missed call will prompt on LCD when it is rejected.
Disabled by default.

Return Code for Call Rejection 

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

Enable SMS Reply

Enables the ability to send predefined SMS replies during an incoming call (while ringing). Useful for quick, non-verbal responses. Enabled by default.

Enable No Ringing

Allows the phone to suppress audible ringing on incoming calls, useful for silent environments or Do Not Disturb-like behavior. Enabled by default.

BLF

Enable Screen On with BLF/SCA Status Update

Enables turning on the LCD screen when BLF/SCA status is updated. Enabled by default.

Enable BLF Pickup Sound 

Enables playing a sound when a monitored BLF line is ringing. Disabled by default.

Record

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.

Preferences

Default Phone APP

Configures phone application for the device. The default setting is the Grandstream phone application. If users develop a new phone application using SDK, it can be chosen to replace the original one. Please note the configuration “Quickly Launch 3rd Party App” is not available if the default phone application changes.

Quickly Launch 3rd Party App

Selects an application to launch when users take handset off-hook or press the Dial/Conference button. Once selected, the package name and activity of the application will be displayed below. When users take handset off-hook or press the Dial/Conference button, the phone will automatically enter the configured application interface. Default setting is blank, which means using the Grandstream phone application.

Handset Option 

Configures the Handset options and can be set to:

Default setting is “Auto”.

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 button will always remain displayed.. The value range is 0-30 seconds.

DTMF Keypad Display Timeout (s)

Configures the timeout duration fot the DTMF kepad display during a call. If set to “0”, the keypad will always remain displayed. The value range is 0-30 seconds.

Preferred Contact Sources

Configures the priority of the contact ID source for incoming/outgoing calls displayed on the phone. Select one contact source and click the Up/Down arrow on the right to adjust the order. Note: If the “Caller ID Display” under the account is configured as “Disabled”, the caller number cannot be obtained, the phone will only display “Unavaliable”.

Advanced Settings

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

Configures the amount of concurrent accounts in virtual account group. If the total amount of virtual group accounts is “N”, the number of accounts the user sets is “n”, then the phone will register the first n accounts; If registration for all of these accounts failed, then register the last N-n accounts. The default value is 2

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”.

CPT Settings

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.

Call Tone Settings

Enable Meeting Tone

Activates periodic audio alerts during active meetings; plays a join tone when a member connects and a heartbeat tone every 20 seconds. Useful for ensuring participants are aware the call is ongoing.
Disabled by default.

Member Initiation Tone

Sets the dual-tone multi-frequency (DTMF) pattern that signals when a new participant joins. Helps other attendees identify join events audibly.

Active Meeting Tone

Defines the frequency and cadence of the in-call periodic alert (every 20 seconds) indicating the meeting is still active. Used for monitoring long-duration calls.

MOH Tone

Specifies the frequency set used when playing Music On Hold (MOH); can be replaced with a simple tone pattern for minimal setups. Useful when full MOH is unavailable.

Conference Tone Interval

Sets the repeat interval (e.g., every 20 seconds) at which the active meeting tone is played. Helps in call supervision and confirming call continuity without speech.

Multicast Paging

Paging Barge

Sets the threshold of paging calls. If the paging call’s priority is higher than the threshold, the existing call will be hold and the paging call will be answered. Otherwise, the existing call does not be affected. If it is set to Disable, any paging call will not be answered.

Default setting is “Disable”.

Activate Paging Priority

Determines if a new paging call whose priority is higher than the existing paging call will be answered. If it is checked, this feature will be enabled.

The default setting is disabled.

Multicast Paging Codec

Selects the codec type for the multicast paging call. This list includes PCMU, PCMA, G726-32, G722, and G729A/B, iLBC, Opus.

Enable Multicast Paging Video

Enables the video feature to establish a multicast paging call.

The default setting is disabled.

Multicast Paging Video Codec

Sets the video codec for the multicast paging call.

The default setting is “H.264”.

Multicast Paging Image Size

Sets the video image size for the multicast paging call. This list includes 1080P, 720P, 4CIF, VGA, CIF, QVGA, and QCIF.

Default setting is “VGA”.

Multicast Paging Video Bit Rate

Determines the video bit rate for the multicast paging call.

The default setting is “256 kbps”.

Multicast Paging Video Frame Rate

Configures the video frame rate for the multicast paging call. This list includes “15 frames/second”, “25 frames/second”, “30 frames/second”, and “Variable frames rate”.

Multicast Paging H.264 Profile Type

Specifies the H.264 codec profile type for the multicast paging call. This list includes “Baseline Profile”, “Main Profile”, and “High Profile”.

Note: Lower profile is easier to decode, while higher profile has higher compress rate. Usually, use Baseline profile for low CPU Performance device, and choose high profile for video conference.

Multicast Paging H.264 Payload Type

Determines the H.264 codec payload type for the multicast paging call.

The default setting is “99”.

Multicast Listening

Priority

Configures the IP address and port number for monitoring multicast paging call. When the initiator initiates a call, answer the call and display listening address and tag of the monitoring target. This feature supports Video Multicast, when the initiator initiates Video Multicast, it will automatically add the number 2 on the port of the listening address. Reboot the phone to make changes take effect.

Listening Address

Label

The valid IP address range is from 224.0.0.0 to 239.255.255.255. Users may also fill the label for each listening address corresponding to priority.

Main Account

Main Account

Configures the main account into account disaster recovery group. Each account can only be selected once.
Set to none by default.

SIP Server 

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

SIP User ID

Defines the user account information, provided by user’s VoIP service provider (ITSP). It’s usually in the form of digits similar to a phone number.

SIP Authentication ID 

SIP service subscriber’s ID used for authentication. It can be identical to or different from the SIP user ID.

SIP Authentication Password 

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

Outbound Proxy

The IP address or domain name of the primary outbound proxy, media gateway, or session border controller. It’s used for firewall or NAT penetration in certain network environment. For example, in a symmetric NAT environment STUN will not work and only an outbound proxy can provide the solution.

Backup Account

Backup Account 

If the outgoing/incoming call fails when using main account, the phone will try to use other backup accounts in the same group.Each account can only be selected once.

SIP Server

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

Outbound Proxy