GXV3275 WEB GUI SETTINGS. Grandstream Networks GXV3275

GXV3275 WEB GUI SETTINGS

The GXV3275 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the application phone through a Web browser such as Microsoft

’s IE,

Mozilla Firefox, Google Chrome and etc.

Figure 12: GXV3275 Web GUI - Login

ACCESS GXV3275 WEB GUI

The GXV3275 Web GUI URL is http://Phone-IP-Address,where the Phone-IP Address is the IP address displayed on the GXV3275 desktop account widget.

Firmware Version 1.0.1.20

GXV3275 Administration Guide Page 34 of 77

Figure 13: GXV3275 Main Screen - IP Address

To access the GXV3275 Web GUI:

1. Connect the computer to the same network as the phone.

2. Make sure the phone is turned on and shows its IP address on the account widget of the desktop.

3. Open a Web browser on your computer.

4.

Enter the phone’s IP address in the address bar of the browser.

5.

Enter the administrator’s login and password to access the Web Configuration Menu. The default login name and password for the administrator is "admin" and "admin". The default login name and password for the end-user is "user" and "123".

Note:



The computer has to be connected to the same sub-network as the phone. This can be easily done by connecting the computer to the same hub or switch as the phone connected to. In absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the back of the phone.



If the phone is properly connected to a working Internet connection, the phone will display its IP address. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255.

Users will need this number to access the Web GUI. For example, if the phone shows 192.168.40.154, please enter "http://192.168.40.154" in the address bar of the browser.



SAVING CHANGES

When changing any settings on the web UI, always SUBMIT them by pressing the Save button on the bottom of the page, and then clicking the Apply button on the top of the page to apply the configuration changes. For those options with next to it in the Web page, users must reboot the phone for the changes to take effect.

DEFINITIONS

This section describes the options in the GXV3275 Web GUI. As mentioned, you can log in as an administrator or an end user.



Status: Displays the Account Status, Network Status, and System Info of the phone.



Account (1-6): To configure each of the SIP accounts.



Advanced Settings: To configure General Settings, Call Features, Video Settings, Tone Generator and MPK Settings for MPK App.



Maintenance: To configure Network Settings, Wi-Fi Settings, Time Settings, Web/SSH Access,

Upgrade, Syslog, Logcat, Debug, Language, TR-069, Contacts and Device Manager.

The following table shows the web pages accessible by end user and administrator, respectively.

Table 4: GXV3275 Web Access

Username Default Password Accessible Web Pages User Type

Administrator admin admin

End User user 123

All pages



Status: Account Status, Network Status,

System Info.



Advanced Settings: Tone Generator, MPK

General Settings, MPK LCD Settings, MPK

EXT Settings.



Maintenance: Network Settings, Wi-Fi Settings,

Time Settings, Web/SSH Access, Logcat,

Debug, Language, Contacts, Device Manager.



TOOLBAR

The web UI tool bar is on the upper right corner of the web UI page.

Figure 14: GXV3275 Web UI Tool Bar



Theme: Select the web UI style to display the web page.



Reboot: Reboot the phone.



Exit: Log out from the web UI.



English: Select the display language for the web UI.

STATUS PAGE DEFINITIONS

STATUS/ACCOUNT STATUS

Account

Number

SIP Server

Status

6 SIP accounts on the phone. By default the 6th SIP account is the

IPVideoTalk account built-in with the phone.

SIP User ID for the account.

URL or IP address, and port of the SIP server.

Registration status for the SIP account.

STATUS/NETWORK STATUS

MAC Address

Address Type

IP Address

Subnet Mask

Default Gateway

DNS Server 1

DNS Server 2

NAT Type

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

The configured address type: DHCP, Static IP or PPPoE.

IP address of the phone.

Subnet mask of the phone.

Default gateway of the phone.

DNS Server 1 of the phone.

DNS Server 2 of the phone.

The type of NAT connection used by the phone.



STATUS/SYSTEM INFO

Product Model

Hardware Revision

Part Number

System Version

Recovery Version

Boot Version

Kernel Version

Android

TM

Version

System Up Time

Product model of the phone.

Hardware version number.

Product part number.

Firmware version. This is the main software release version.

Recovery image version.

Booting code version.

Kernel version.

Android

TM

OS version: 4.2.2.

System up time since the last reboot.

ACCOUNT PAGE DEFINITIONS

The GXV3275 has six lines that can be configured to accommodate 6 independent SIP accounts. Each

SIP account has an individual configuration page.

ACCOUNT/GENERAL SETTINGS

Account Active

Account Name

SIP Server

SIP User ID

SIP Authentication ID

SIP Authentication Password

Voice Mail UserID

This field indicates whether the account is active. The default value for the IPVideoTalk account (Account 6) is "Yes" and the default value for the first five accounts is "No".

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

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

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

The account password required for the phone to authenticate with the

ITSP (SIP) server before the account can be registered. After it is saved, this will appear as hidden for security purpose.

This parameter allows you to access voice messages by pressing the

MESSAGE button on the phone. This ID is usually the VM portal access number. For example, in UCM6100 IPPBX, *97 could be used.



Name

Tel URI

The SIP server subscriber's name (optional) that will be used for Caller

ID display.

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

ACCOUNT/NETWORK SETTINGS

Outbound Proxy

Secondary Outbound Proxy

DNS Mode

NAT Traversal

Proxy-Require

IP address or 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.

IP address or Domain name of the Secondary Outbound Proxy, Media

Gateway, or Session Border Controller. Secondary outbound proxy will be used when the primary outbound proxy fails.

This parameter controls how the Search Appliance looks up IP addresses for hostnames. There are three modes: A Record, SRV,

NATPTR/SRV. The default setting is "A Record". If the user wishes to locate the server by DNS SRV, the user may select "SRV" or

"NATPTR/SRV".

This parameter configures whether the NAT traversal mechanism is activated. Users could select the mechanism from NAT NO, STUN,

Keep-alive, UPnP, Auto or VPN.

If set to "STUN" and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages. The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be "Keep-alive". Configure this to be "NAT NO" if an outbound proxy is used. "STUN" cannot be used if the detected

NAT is symmetric NAT. Set this to "VPN" if OpenVPN is used.

The default setting is "Keep-alive".

A SIP Extension to notify the SIP server that the phone is behind a

NAT/Firewall. Do not configure this parameter unless this feature is



supported on the SIP server.

ACCOUNT/SIP SETTINGS

SIP Registration

Unregister Before New

Registration

Register Expiration (m)

Wait Time Retry Registration

(s)

Local SIP Port

SUBSCRIBE for MWI

Session Expiration (s)

Min-SE (s)

UAC Specify Refresher

UAS Specify Refresher

Selects whether or not the phone will send SIP Register messages to the proxy/server. The default setting is "Yes".

If set to "Yes", the SIP user's registration information will be cleared when the phone reboots. The SIP Contact header will contain "*" to notify the server to unbind the connection. The default setting is "No".

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

The amount of time (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 seconds.

Defines the local SIP port used to listen and transmit. The default setting is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066 for

Account 4, 5068 for Account 5, and 5070 for Account 6. The valid range is from 5 to 65535.

When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be sent periodically. The phone supports synchronized and non-synchronized MWI. The default setting is "No".

The SIP Session Timer extension (in seconds) that enables SIP sessions to be periodically "refreshed" via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session

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

64800.

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

The valid range is from 90 to 64800.

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

"Omit".

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

"UAC".



Force INVITE

Caller Request Timer

Callee Request Timer

Force Timer

Enable 100rel

Use Privacy Header

Use P-Preferred-Identity

Header

SIP Transport

SIP URI Scheme When Using

TLS

Use Actual Ephemeral Port in

Contact with TCP/TLS

The Session Timer can be refreshed using the INVITE method or the

UPDATE method. Select "Yes" to use the INVITE method to refresh the session timer. The default setting is "No".

If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. The default setting is "No".

If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it receives inbound calls. The default setting is "No".

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

"No", the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select "No". The default setting is "No".

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

"No".

Controls whether the Privacy Header will present in the SIP INVITE message or not. The default setting is "default": the Privacy Header will show in INVITE unless "Huawei IMS" special feature is on.

If set to "Yes", the Privacy Header will always show in INVITE. If set to

"No", the Privacy Header will not show in INVITE.

Controls whether the P-Preferred-Identity Header will present in the SIP

INVITE message or not. The default setting is "default": the

P-Preferred-Identity Header will show in INVITE unless "Huawei IMS" special feature is on.

If set to "Yes", the P-Preferred-Identity Header will always show in

INVITE. If set to "No", the P-Preferred-Identity Header will not show in

INVITE.

Determines the network protocol used for the SIP transport. Users can choose from TCP/UDP/TLS. The default setting is "UDP".

Specifies if "sip" or "sips" will be used when TLS/TCP is selected for SIP

Transport. The default setting is "sip".

This option is used to control the port information in the Via header and

Contact header. If set to No, these port numbers will use the permanent



Symmetric RTP

Support SIP Instance ID

Validate Incoming SIP

Messages

Authenticate Incoming INVITE

SIP T1 Timeout

SIP T2 interval

Remove OBP from route

Check Domain Certificate

Enable SCA (Shared Call

Appearance)

Enable BargeIn

Auto-filling Pickup Feature

Code

Pickup Feature Code

Line-seize Timeout listening port on the phone. Otherwise, they will use the ephemeral port for the particular connection. The default setting is "No".

Defines whether symmetric RTP is supported or not. The default setting is "No".

Defines whether SIP Instance ID is supported or not. The default setting is "Yes".

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

If set to "Yes", the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response. The default setting is "No".

SIP T1 Timeout (in seconds). The default setting is 0.5 sec.

SIP T2 Interval (in seconds). The default setting is 4 sec.

Configures to remove outbound proxy from route. This is used for the

SIP Extension to notify the SIP server that the device is behind a

NAT/Firewall. The default setting is "No".

Choose whether the domain certificates will be checked or not when

TLS/TCP is used for SIP Transport. The default setting is "No".

If set to "Yes", the Shared Call Appearance (Broadsoft Standard) will be used for the registered account. The default setting is "No".

If set to "Yes", the user could barge into an active call on a shared line.

The default setting is "No".

If set to "Yes", the "Pickup Feature Code" configured in web UI will be automatically filled in on the phone's dial pad when picking up the parked call. The default setting is "Yes". This option is used when

"Special Mode" is set to "Broadsoft" and "Enable SCA" is set to "Yes".

Configure the pickup feature code for call park. If "Auto-filling Pickup

Feature Code" is set to "Yes", this pickup feature code will be automatically filled in on the phone's dial pad when picking up the parked call. This option is used when "Special Mode" is set to

"Broadsoft" and "Enable SCA" is set to "Yes".

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

15 to 60.



ACCOUNT/CODEC SETTINGS

DTMF

DTMF Payload Type

Preferred Vocoder

ILBC Frame Size

Enable RFC5168 Support

H.264 Image Size

H.264 Profile Type

Video Bit Rate


This parameter specifies the mechanism to transmit DTMF digits. There are 3 supported modes: in audio which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs), via RTP

(RFC2833), or via SIP INFO. The default setting is "RFC2833".

This parameter sets the payload type for DTMF using RFC2833. Default is 101. The valid range is from 96 to 127.

5 different vocoder types are supported on the phone, including G.711

A-law (PCMA), G.711 U-law (PCMU), G.722, G.726-32 and iLBC. Users can configure vocoders in a preference list that is included with the same preference order in SDP message. Arrange your preferred vocoder orders using the Up and Down button.

Select the iLBC frame size when iLBC is selected. Users can select

20ms or 30ms. The default setting is 30ms.

If set to "Yes", RFC5168 support will be enabled for video call. The default setting is "No".

Select the H.264 image size from "720P" "4CIF" "VGA" "CIF" "QVGA" and "QCIF". The default setting is 720P.

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.

Select the H.264 profile type from "Baseline Profile" "Main Profile" "High

Profile" or "BP/MP/HP". The default setting is "Baseline Profile". 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.

Configures bit rate for video call. The default setting is 2048 kbps. The valid range is from 32 - 2048 kbps. 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.

Note: For some network environment, the default setting "720P" might


SDP Bandwidth Attribute

H.264 Payload Type

Silence Suppression

Voice Frames Per TX 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.

Select the SDP bandwidth attribute from "Standard" "Media Level"

"Session Level" and "No" to modify the session format. The default setting is "Media Level".



Standard: use AS format in session level; use TIAS format in media level



Media Level: use AS format in media level



Session Level: use AS format in session level



No: no modifications in the session format

Note:

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

Configures H.264 codec payload type. The default setting is 99. The valid range is from 96 to 127.

Controls the silence suppression/VAD feature. If set to "Yes", when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to "No", this feature is disabled. The default setting is "No".

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.

ACCOUNT/CALL SETTINGS

Start Video Automatically

Remote Video Request


If set to "Yes", the video will be started automatically when answering

SIP video call. The default setting is "Yes".

Configures the preference to handle video request from the other party during an audio call. Users could select "prompt", "accept" or "deny".

The default setting is "prompt".



"prompt": A message will be prompted if a video request is received.

Users can select "Yes" to establish video or "No" to reject the request.


Dial Plan Prefix

Dial Plan




"accept": Video request will be accepted automatically and video will be established.



"deny": Video request will be rejected automatically.

Configures the prefix to be added to each dialed number.

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

Dial Plan Rules:

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

2. Grammar: x - any digit from 0-9;

a) xx+ - at least 2 digit numbers

b) xx. - only 2 digit numbers

c) ^ - exclude

d) [3-5] - any digit of 3, 4, or 5

e) [147] - any digit of 1, 4, or 7

f) <2=011> - replace digit 2 with 011 when dialing

g) | - the OR operand

h) \+ - add + to the dialing number



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.

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 }


Refer-To Use Target Contact

Auto Answer

Send Anonymous

Anonymous Call Rejection

Call Log

Special Feature

Feature Key Synchronization

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



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



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



1[2-9]xx[2-9]xxxxxx |- allow dialing to any US/Canada Number with

11 digits length



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



[3469]11 - allow dialing special and emergency numbers 311, 411,

611 and 911

NOTE: In some cases where the user wishes to dial strings such as

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

If set to "Yes", the "Refer-To" header uses the transferred target's

Contact header information for attended transfer. The default setting is

"No".

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

Intercom/Paging", it will answer the call based on the SIP info header sent from the server/proxy. The default setting is "No".

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


If set to "Yes", anonymous calls will be rejected. The default setting is

"No".

Configures Call Log setting on the phone. You can log all calls, log incoming/outgoing calls (missed calls will not be logged), or disable call log. The default setting is "Log All".

Different soft switch vendors have special requirements. Therefore users may need select special features to meet these requirements. Users can choose from Standard, Broadsoft, CBCOM, RNK, China Mobile, ZTE

IMS, Mobotix, ZTE NGN, or Huawei IMS depending on the server type.

The default setting is "Standard".

This feature is used for Broadsoft call feature synchronization. When it's enabled, DND, Call Forward features and Call Center Agent status can



Enable Call Features

No Key Entry Timeout (s)

Ring Timeout (s)

Transfer on 3-way Conference

Hang up

Use # as Dial Key

Conference URI

Broadsoft Call Center

Hoteling Event

Call Center Status

Account Ring Tone

Call Forward Type be synchronized between Broadsoft server and phone. The default setting is "Disabled".

If set to "Yes", the call features using star codes on the phone will be supported locally. Otherwise, the feature code will need to be provided from the server side. The default setting is "No".

Defines the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the digits will be sent out. The default value is 4 seconds. The valid range is from 1 to 15.

Defines the timeout (in seconds) for the rings on no answer. The default setting is 60. The valid range is from 10 to 300.

If set to "Yes", when the phone hangs up as the conference initiator, the conference call will be transferred to the other parties so that other parties will remain in the conference call.

Allows users to configure the "#" key as the "Send" key. If set to "Yes", the "#" key will immediately dial out the input digits. In this case, this key is essentially equivalent to the "Send" key. If set to "No", the "#" key is included as part of the dialing string and please make sure the dial plan is properly configured to allow dialing # out.

Configures Conference URI for N-way conference (Broadsoft standard).

The default setting is "No". When set to "Yes", a soft key "CallCenter" is displayed on LCD dial screen for the Broadsoft account. User can access different Broadsoft Call Center agent features via this softkey.

Please note that "Feature Key Synchronization" will be enabled regardless of this setting.

Broadsoft Hoteling event feature. Users can configure this setting if

"Broadsoft Call Center" is enabled. The default setting is "No". With

"Hoteling Event" enabled, user can access the Hoteling feature option by pressing the "CallCenter" softkey.

When set to "Yes", the phone will send SUBSCRIBE to the server to obtain call center status. Users can configure this setting if "Broadsoft

Call Center" is enabled. The default setting is "No".

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu. If there is custom ring tone imported from GXV3275 LCD menu, the ring tone file name will be displayed in this dropdown menu as well for users to select.

Specifies the Call Forward Type:

1. None: Disable Call Forward

2. Unconditional: Forward all calls to a particular number



All To

Time Period

In Time Forward To

Out Time Forward To

Busy To

No Answer To

No Answer Timeout (s)

Matching Incoming Caller ID

3. Time based: Set a time period. In this Time Period, calls are forwarded to the number specified in In Time Forward To; out of this Time Period, calls are forwarded to the number specified in Out

Time Forward To.

4. Others: when phone is busy, calls are forwarded to the number specified in Busy To; when incoming calls are not answered, those calls are forwarded to the number specified in No Answer To; the waiting time for answering calls is specified in No Answer Timeout

(s). The valid range for No Answer Timeout (s) is from 1 to 120.

Specifies the number to be forwarded to when "Unconditional" Call

Forward Type is used.

Configures the period of time to forward the call when "Time based" Call

Forward Type is used.

When "Time based" Call Forward Type is used, specifies the number to be forwarded to within the configured Time Period above.

When "Time based" Call Forward Type is used, specifies the number to be forwarded to when it's not within the configured Time Period.

Specifies the number to be forwarded to for Call Forward On Busy.

Specifies the number to be forwarded to for Call Forward On No Answer.

Defines the timeout (in seconds) before the call is forwarded on no answer. The default value is 20 seconds. The valid range is from 1 to

120.

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



Specific caller ID number. For example, 8321123;



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

xx+ : at least 2-digit number;

xx : only 2-digit number;

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

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



Alert Info text

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

Alert-Info: <http://127.0.0.1>; info=priority



Distinctive Ring Tone

Upload Local MOH Audio File

Enable Local MOH

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

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.

If set to "Yes", local MOH will be turned on. Users will need upload a local MOH audio file so it can be played on the other party when the user puts the other party on hold. The default setting is "No".

ADVANCED SETTINGS PAGE DEFINATIONS

ADVANCED SETTINGS/GENERAL SETTINGS

Local RTP Port

Use Random Port

Disable in-call DTMF display

Keep-alive Interval (s)

Defines the local RTP-RTCP port pair used to listen and transmit. It is the base RTP port for channel 0. When configured, for audio, channel 0 will use this port_value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+4 for RTP and port_value+5 for its RTCP.

For video, channel 0 will use port_value+2 for RTP and port_value+3 for its RTCP; channel 1 will use port_value+6 for RTP and port_value+7 for

RTCP. The default value is 5004. The valid range is from 1024 to 65400.

When set to "Yes", this parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is "Yes".

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

When it's set to "Yes", the DTMF digits entered during the call will not display. The default setting is "No".

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



STUN Server

Use NAT IP default setting is 20 seconds. The valid range is from 10 to 160.

The IP address or Domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN. The default setting is

"stun.ipvideotalk.com".

Specify the NAT IP address used in SIP/SDP messages. This field is blank at the default settings. It should ONLY be used if it's required by your ITSP.

SIP TLS Certificate

SIP TLS Private Key

SSL certificate used for SIP TLS Transport.

SSL Private key used for SIP TLS Transport.

SIP TLS Private Key Password SSL Private key password used for SIP TLS Transport.

ADVANCED SETTINGS/CALL FEATURES

Always Ring Speaker

Auto Conference

Disable Call-Waiting

Disable DialPlan

Disable Call-Waiting Tone

Disable DND Reminder Ring

Disable Direct IP Call

Use Quick IP-Call mode

This setting is used when the headset is connected. If set to "Yes", the phone will always ring speaker when headset is used at the same time.


If set to "Yes", users could press CONFERENCE button to establish conference call when there are 2 or more calls on the phone, instead of adding each party one by one. The default setting is "No".

Disables the call waiting feature. The default setting is "No".

Disables dial plan check on the GXV3275 when dialing out. This can be used when dialing out phone number which contains alphabet characters from imported contacts. The default setting is "No".

Disables the call waiting tone when call waiting is on. The default setting is "No".

Disables the DND reminder ring. If set to "Yes", the ring splash that indicates an incoming call when DND is enabled will not be played. The default setting is "No".

Disables Direct IP Call. The default setting is "No".

When set to "Yes", users can dial an IP address under the same

LAN/VPN segment by entering the last octet in the IP address. To dial quick IP call, offhook the phone and dial #XXX (X is 0-9 and XXX

<=255), phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask. #XX or #X are also valid so leading 0 is not required (but OK). No

SIP server is required to make quick IP call. The default setting is "No".



Disable Conference

Disable Transfer

Default Transfer Mode

Escape # as %23 in SIP URI

Offhook Auto Dial

Offhook Timeout (s)

Disables the Conference function. The default setting is "No".

Disables the Transfer function. The default setting is "No".

Selects default transfer mode as "Blind Transfer" or "Attended Transfer".

The default setting is "Blind Transfer".

Replaces "#" by "%23" for some special situations. The default setting is

"Yes".

Configures a User ID/extension to dial automatically when the phone is offhook. The phone will use the first account to dial out. The default setting is "No".

If configured, when the phone is on hook, it will go off hook after the timeout (in seconds). The default setting is 30. The valid range is from

10 to 60.

ADVANCED SETTINGS/VIDEO SETTINGS

Video Frame Rate

Display Video Information

Enable Frame Skipping in

Video Decoder

Configures video frame rate for SIP video call from "5 frames/second",

"15 frames/second", "25 frames/second", "30 frames/second" and

"Variable frames rate". 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.

Configure whether to display video stream sending and receiving information during video call. The default setting is "No".

Configure to enable frame skipping in video decoder. If 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".



ADVANCED SETTINGS/TONE GENERATOR

Call Progress Tones:

Dial Tone

Ring Back Tone

Busy Tone

Reorder Tone

Confirmation Tone

Call-Waiting Tone

PSTN Disconnect Tone

Call-Waiting Tone Gain

Default Ring Cadence

Using these settings, users can configure ring or tone frequencies based on parameters from the local telecom provider. By default, they are set to the North American standard.

Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.

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

(Frequencies are in Hz and cadence on and off are in 10ms)

ON is the period of ringing ("On time" in 'ms') while OFF is the period of silence. 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 repeat the pattern.

Up to three cadences are supported.

This adjusts the call waiting tone volume. Users can select "Low",

"Medium" or "High". The default setting is "Low".

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

ADVANCED SETTING/MPK GENERAL SETTINGS (FOR MPK APP)

BLF Call-pick Prefix

Event List URI

Configures the prefix prepended to the BLF extension when the phone picks up a call with BLF key. The default setting is ** for each account.

Configures the eventlist BLF URI on the phone to monitor the extensions in the list with multi-purpose keys. The server side has to support this feature. Users need configure an eventlist BLF URI on the service side first (i.e., [email protected]) with a list of extension included. On the phone, in this "eventlist BLF URI" field, fill in the URI without the domain (i.e., BLF1006).

ADVANCED SETTING/MPK LCD SETTINGS (FOR MPK APP)

Key Mode

Assigns a function to the Multi Purpose Key from the MPK App installed on the GXV3275. The MPK App can be downloaded from GS Market.

The key mode options are:



Speed Dial

Press to dial the extension configured in UserID field.



MPK List

Account

Name

UserID

Display Format



Busy Lamp Field (BLF)

Monitor the extension status as configured in UserID field.



Call Transfer

Transfer the current active call to the extension configured in UserID field.



Call Intercom

Intercom/paging to the extension configured in UserID field.



Speed Dial via Active Account

Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured extension using account 2.



Dial DTMF

Dial the DTMF digits filled in UserID field during the call.



Call Park

Configure the call park feature code to park or pick up the call.

The configured MPK will show in the MPK List. Users could select "Up"

"Down" and "More" to select, edit and delete the MPK.

Selects the SIP Account used for the multi-purpose keys.

Configures the display name for the multi-purpose key.

Configures the UserID for the corresponding multi-purpose key mode.

Configures the display format for the multi-purpose keys. Users could select display "Name", "UserID", or "Name (UserID)". "Name" is the one saved in GXV3275 Contacts. The default setting is "Name".

Configure whether the display name from server side will be shown or not. The default setting is "No".

Show DisplayName from

Server

MAINTENANCE PAGE DEFINATION

MAINTENANCE/NETWORK SETTINGS

Address Type

DHCP Host name (Option 12)

DHCP Vendor Class ID

(Option 60)

Allows users to configure the appropriate network settings on the phone.

Users could select "DHCP", "Static IP" or "PPPoE". By default, it is set to

"DHCP".

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

Used by clients and servers to exchange vendor class ID. The default setting is GXV.



IP Address

Subnet Mask

Default Gateway

DNS Server 1

DNS Server 2

PPPoE Account ID

PPPoE Password

Alternate DNS Server

Enter the IP address when static IP is used.

Enter the Subnet Mask when static IP is used.

Enter the Default Gateway when static IP is used.

Enter the DNS Server 1 address when static IP is used.

Enter the DNS Server 2 address when static IP is used.

Enter the PPPoE account ID.

Enter the PPPoE Password.

Enter the alternate DNS server address.

Second Alternate DNS Server Enter the second alternate DNS server address.

Enable LLDP

Layer 3 QoS for SIP

Enable/Disable LLDP on the phone. The default setting is "Yes".

Defines the Layer 3 QoS parameter for SIP packets. This value is used for IP Precedence, Diff-Serv or MPLS. The default setting is 48.

Layer 3 QoS for Audio

Layer 3 QoS for Video

Layer 2 QoS 802.1Q/VLAN

Tag

Layer 2 QoS 802.1p Priority

Value

User-Agent

Defines the Layer 3 QoS parameter for audio packets. This value is used for IP Precedence, Diff-Serv or MPLS. The default setting is 48.

Defines the Layer 3 QoS parameter for video packets. This value is used for IP Precedence, Diff-Serv or MPLS. The default setting is 48.

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

Assigns the priority value of the Layer2 QoS packets for LAN port. The

Default value is 0.

This is a configurable User Agent string used in phonebook and screensaver downloading.

802.1x mode

Identity

MD5 Password

CA Certificate

Allows the user to enable/disable 802.1x mode on the phone. The default setting is "Disable".

Enter the Identity information for the 802.1x mode.

Enter the MD5 Password for the 802.1x mode.

Enter the CA Certificate for the 802.1x mode.

Client Certificate Enter the Client Certificate for the 802.1x mode.

HTTP/HTTPS Proxy Hostname

Specifies the HTTP/HTTPS proxy URL for the phone to send packets to.

The proxy server will act as an intermediary to route the packets to the destination.

HTTP/HTTPS Proxy Port

Bypass Proxy For

Specifies the HTTP/HTTPS proxy URL for the phone to send packets to.

The proxy server will act as an intermediary to route the packets to the destination.

Defines the destination IP address where no proxy server is needed.

The phone will not use a proxy server when sending packets to the



specified destination IP address.

MAINTENANCE/WI-FI SETTINGS

Wi-Fi Basics

Wi-Fi Function

ESSID

Enable/Disable Wi-Fi. The default setting is "Enable".

Click on "Scan" to detect the available Wi-Fi network within the area.

The ESSID(s) will show in the box. Click on "Select" to select the Wi-Fi network to connect to.

Wi-Fi Basics

ESSID

Password

Fill in the ESSID you would like to save the Wi-Fi password.

Enter the password for the ESSID.

MAINTENANCE/TIME SETTINGS

Assign NTP Server Address

DHCP Option 42 override NTP server

DHCP Option 2 to override

Time Zone setting

Time Zone

Time Display Format

Date Display Format

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

Defines whether DHCP Option 42 should override NTP server or not.

When enabled, DHCP Option 42 will override the NTP server if it's set up on the LAN. The default setting is "Yes".

Allows device to get provisioned for Time Zone from DHCP Option 2 in the local server automatically. The default setting is "No".

Controls the date/time display according to the specified time zone.

12-hour or 24-hour time display format

Configures the date display format on the LCD. The following formats are supported:



Normal (M/DD/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 DD/MM/YYYY.

MAINTENANCE/WEB/SSH ACCESS

Disable SSH

Access Method

If set to "Yes", the phone will not allow any SSH access to the phone.


Allows users to select HTTP or HTTPS for Web Access. The default



Port

Admin Password

Confirm Admin Password

User Password

Confirm User Password

MAINTENANCE/UPGRADE

setting is HTTP.

By default, HTTP uses port 80 and HTTPS uses port 443. This field is for customizing the web port.

Allows users to change the admin password. The password field is purposely hidden after clicking "Save" button for security purpose. This field is case sensitive with a maximum length of 32 characters. The default admin password is "admin".

Type the new Admin Password again to confirm.

Allows the administrator to set the password for user-level web GUI access. This field is case sensitive with a maximum length of 32 characters. The default user password is "123".

Type the new User Password again to confirm.

Download Device

Configuration

Configure via LCD Menu

XML Config File Password

HTTP/HTTPS User Name

HTTP/HTTPS Password

Upgrade Via

Firmware Server Path


Click on "Download" to download the phone's config file in .txt format.

The config file in .txt format has 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 using this txt file.

Configure the access control for the users to configure from keypad men. There are two different options:



Unrestricted. All the options under settings can be accessed in keypad menu. This is the default setting.



Basic settings only. "Advanced" menu under settings will not be available in keypad menu.

Caution:

When set to "Basic settings only", the phone could get LOCKED if the administrator password is lost or forgotten.

The password for encrypting the XML configuration file using OpenSSL.

This is required for the phone to decrypt the encrypted XML configuration file.

The user name for the HTTP/HTTPS server.

The password for the HTTP/HTTPS server.

Allows users to choose the firmware upgrade method: TFTP, HTTP or

HTTPS.

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


Config Server Path

Firmware File Prefix

Firmware File Postfix

Config File Prefix the configuration server for provisioning.

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

Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone.

Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the phone.

Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone.

Config File Postfix mDNS Override Server

Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.

Allows mDNS override config/firmware server setting on the phone if present. The default setting is "Yes".

DHCP Option 66 Override

Server

DHCP option 66 originally was only designed for TFTP server. Later on it was extended to support an HTTP URL. The GXV3275 supports both

TFTP and HTTP server via option 66. The default setting is "Yes".

DHCP Option 120 Override SIP

Server

Enables DHCP Option 120 from local server to override the SIP Server on the phone. The default setting is "Yes".

3CX Auto Provision

Automatic Upgrade

Enables automatic provision feature on the phone when 3CX is used as the SIP server. The default setting is "Yes".

Enables automatic upgrade and provisioning. The default setting is "No".

Automatic Upgrade Check

Interval (m)

Hour of the Day (0-23)

Day of the Week (0-6)

Firmware Upgrade and

Provisioning

Auto Reboot to Upgrade

Without Prompt

Specifies the time period to check for firmware upgrade (in minutes).

The default setting is 10080 minutes (7 days).

Defines the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes.

Defines the day of the week to check the HTTP/TFTP server for firmware upgrades or configuration files changes.

Defines the rules for automatic upgrade: Always Check at bootup, when

F/W pre/suffix changes, Skip the Firmware Check.

If set to "Yes", the phone will automatically start upgrading after downloading the firmware files. Otherwise, users would need confirm in the prompted message in LCD to start upgrading process. The default value is "Yes".



Authenticate Conf File

Factory Reset

MAINTENANCE/SYSLOG

Syslog Server

Authenticates configuration file before acceptance. The default setting is

"No".

Click "Reset" to restore to factory default settings. Users can select to clear the SD card in phone' storage for factory reset.

Syslog Level

The URL/IP address for the syslog server. The default setting is

"log.ipvideotalk.com".

Selects the level of logging for syslog. The default setting is "None".

There are 4 levels: DEBUG, INFO, WARNING AND ERROR. Syslog messages are sent based on the following events:



Product model/version on boot up (INFO level);



NAT related info (INFO level);



Sent or received SIP message (DEBUG level);



SIP message summary (INFO level);



Inbound and outbound calls (INFO level);



Registration status change (INFO level);



Negotiated codec (INFO level);



Ethernet link up (INFO level);



SLIC chip exception (WARNING and ERROR levels);



Memory exception (ERROR level).

MAINTENANCE/LOGCAT

Clear Log

Log Tag

Log Priority

MAINTENANCE/DEBUG

Capture Trace

Press the clear log button to clear log.

Specifies the log tag to filter the log.

Selects the log priority. The log priority options are: Verbose, Debug,

Info, Warn, Error, Fatal, Silent (suppress all output).

View Trace


Press START to start capturing a trace, and press STOP to stop the capture process.

Press LIST to view the list of captured traces in web browser. It will show the time when the trace was captured. Click on the name of the file to download and view the trace.


Note:

The trace files are saved in phone’s internal storage. Users can access or delete the trace files from LCD->FileManager->Internal

Storage->ppp directory.

MAINTENANCE/LANGUAGE

Language Select the language displayed on the phone's LCD.

MAINTENANCE/TR-069

Enable TR-069

ACS URL

ACS Username

ACS Password

Periodic Inform Enable

Periodic Inform Interval (s)

Enables TR-069. The default setting is "No".

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

ACS username for TR-069.

ACS password for TR-069.

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

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

ACS.

Connection Request

Username

CPE Cert File

CPE Cert Key

The user name for the ACS to connect to the phone.

Connection Request Password The password for the ACS to connect to the phone.

Connection Request Port The port for the ACS to connect to the phone.

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

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

MAINTENANCE/CONTACTS

Import/Export

File Encoding

File Type

Save Phonebook to PC

Clear The Old List

Replace Duplicate Items

Selects the file encoding format for phonebook file import/export. The default setting is UTF-8.

Selects the file format for phonebook file import/export. The default setting is "XML".

Press "Save" to open the XML file and save it to PC.

If set to "Yes", when the phonebook file is imported to the phone, the old phonebook entries will be removed. The default setting is "No".

If set to "Yes", when the phonebook file is imported to the phone, the



Local File

Download (XML Phonebook)

duplicate entries will be removed. The default setting is "No".

Press "Browse" to select phonebook file from local PC and upload to the phone.

Clear The Old List

Replace Duplicate Items

Download Mode

File Encoding

Download Server

Download Now

Download Interval

If set to "Yes", when the phonebook XML file is downloaded to the phone, the old phonebook entries will be removed. The default setting is

"No".

If set to "Yes", when the XML phonebook file is downloaded to the phone, the duplicate entries will be removed. The default setting is "No".

Selects "OFF" to disable XML phonebook download, or "TFTP"/"HTTP" to download the phonebook from a TFTP/HTTP remote server. The default setting is "OFF".

Selects the file encoding format for XML phonebook download. The default setting is UTF-8.

Specifies server address to download the XML phonebook file. The default XML phonebook file name the phone sends request for is

phonebook.xml

.

Click "Download" button to start downloading the XML phonebook to the phone immediately.

Select download interval from "2 hours", "4 hours", "6 hours", "8 hours" and "12 hours". The phone will automatically send download request to the download server based on the download interval.

Note:

For more information on XML Based Downloadable Phonebook on GXV3275, please refer to the

GXV3275 XML Phonebook Guide in the following link: http://www.grandstream.com/products/gxv_series/gxv3275/documents/gxv3275_xml_phonebook_guide.p

df



Base DN

User Name

Password

LDAP Name Attributes

LDAP Number Attributes

LDAP Mail Attributes

LDAP Name Filter

MAINTENANCE/LDAP BOOK

Server Address

Port

Configures the IP address or domain name of the LDAP server.

Configures the LDAP server port. The default port number is 389.

Configures the LDAP search base. This is the location in the directory where the search is requested to begin.

Example: dc=grandstream, dc=com ou=Boston, dc=grandstream, dc=com

Configures the bind "Username" for querying LDAP servers. Some

LDAP servers allow anonymous binds in which case the setting can be left blank.

Configures the bind "Password" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

Specifies the "name" attributes of each record which are returned in the

LDAP search result. This field allows the users to configure multiple space separated name attributes.

Example: gn cn sn description

Specifies the "number" attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated number attributes.

Example: telephoneNumber telephoneNumber Mobile

Specifies the "mail" attributes of the each record which are returned in the LDAP search result.

Example: mail

Configures the filter used for name lookups.

Examples:

(|(cn=%)(sn=%)) returns all records which has the "cn" or "sn" field starting with the entered prefix;

(!(sn=%)) returns all the records which do not have the "sn" field starting with the entered prefix;

(&(cn=%) (telephoneNumber=*)) returns all the records with the "cn"



LDAP Number Filter

LDAP Mail Filter

LDAP Displaying Name

Attributes

Max Hits

Search Timeout (ms) field starting with the entered prefix and "telephoneNumber" field set.

Configures the filter used for number lookups.

Examples:

(|(telephoneNumber=%)(Mobile=%) returns all records which has the

"telephoneNumber" or "Mobile" field starting with the entered prefix;

(&(telephoneNumber=%) (cn=*)) returns all the records with the

"telephoneNumber" field starting with the entered prefix and "cn" field set.

Configures the filter used for mail lookups.

Example:

(mail=%)

Configures the entry information to be shown on phone's LCD. Up to 3 fields can be displayed.

Example:

%cn %sn %telephoneNumber

Specifies the maximum number of results to be returned by the LDAP server. If set to 0, server will return all search results. The default setting is 50.

Specifies the interval (in seconds) for the server to process the request and client waits for server to return. The default setting is 30 seconds.

Configures to enable LDAP number searching when making outgoing calls. The default setting is "No".

Configures to enable LDAP number searching when receiving incoming calls. The default setting is "No".

LDAP Lookup For Dial

LDAP Lookup For Incoming

Call

Note:

For more information on how to configure LDAP on GXV3275, please refer to the GXV3275 LDAP

Configuration Guide in the following link: http://www.grandstream.com/products/gxv_series/gxv3275/documents/gxv3275_ldap_book_guide.pdf



MAINTENANCE/DEVICE MANAGER

AE Mode

Disable Missed Call Backlight

Disable Missed Call Indicator

Disable MWI Indicator

Disable New Message

Indicator

Disable Contact or Message

Full Indicator

Disable Indicator When LCD is

Off

RJ9 Headset TX Gain (dB)

RJ9 Headset RX Gain (dB)

AE mode (Automatic Exposure) has two options "Dark Environment" and "Bright Environment". Users could select the mode according to the actual environment (dark or bright) where the camera is being used. The phone's camera settings including exposure, brightness, contrast and

Gamma value will be automatically adjusted to achieve better visual effect under the selected environment. The default setting is "Bright

Environment".

If set to "Yes", the screen won't turn off on screen timeout when there is unchecked missed call on the phone. The default setting is "No".

If set to "Yes", the LED indicator on the upper right corner of the

GXV3275 will not light up when there is missed call on the phone. The default setting is "No".


GXV3275 will not light up when there is new voicemail on the phone.



GXV3275 will not light up when there is new message on the phone.



GXV3275 will not light up when the contact storage or message storage is full. The default setting is "No".


GXV3275 will not light up when the LCD screen is off. The default setting is "No".

Select the transmission gain of the RJ9 headset. The default setting is

+6dB.

Select the receiving gain of the RJ9 headset. The default setting is

+6dB.

3.5mm Earphone TX Gain (dB)

Select the transmission gain of the 3.5mm Earphone. The default setting is +6dB.

3.5mm Earphone RX Gain (dB)

Select the receiving gain of the 3.5mm Earphone. The default setting is

+6dB.

Headset Type

Selects whether the connected headset is normal RJ11 headset,

Plantronics EHS headset or Jabra EHS headset.

Accept Call and Speaker Dial This option controls the audio channel behavior when handset is not



in RJ9 Headset Automatically picked up for incoming/outgoing calls. If set to "Yes", RJ9 headset will be automatically used for the answered and dialed calls. Otherwise, speakerphone will be used. The default setting is "No".



GXV3275 WEB GUI SETTINGS. Grandstream Networks GXV3275

GXV3275 WEB GUI SETTINGS

ACCESS GXV3275 WEB GUI

SAVING CHANGES

DEFINITIONS

TOOLBAR

STATUS PAGE DEFINITIONS

ACCOUNT PAGE DEFINITIONS

ADVANCED SETTINGS PAGE DEFINATIONS

MAINTENANCE PAGE DEFINATION

Related manuals

Table of contents

GXV3275 WEB GUI SETTINGS. Grandstream Networks GXV3275