Grandstream Networks, Inc.
COPYRIGHT ©2019 Grandstream Networks, Inc. http://www.grandstream.com All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted. The latest electronic version of this guide is available for download here: http://www.grandstream.
U.S. FCC Statement Part 68 Regulatory Information This equipment complies with Part 68 of the FCC rules. Located on the equipment is a label that contains, among other information, the ACTA registration number and ringer equivalence number (REN). If requested, this information must be provided to the telephone company. The REN is used to determine the quantity of devices which may be connected to the telephone line.
Part 15 Regulatory Information This device complies with part 15 of the FCC Rules. Operation is subject to the following two condition: (1) this device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. Any Changes or modifications not expressly approved by the party responsible for compliance could void the user’s authority to operate the equipment.
Canada Regulatory Information Operation of 5150-5250 MHz is restricted to indoor use only. This device contains licence-exempt y=transmitter(s)/receiver(s) that comply with Innovation, Science and Economic Development Canada’s licence-exempt RSS(s). Operation is subject to the following two conditions: (1) This device may not cause interference. (2) This device must accept any interference, including interference that may cause undesired operation of the device.
EU Regulatory Information Support Frequency Bands and Power: WLAN/BT 2.4 GHz < 20 dBm; WLAN 5.2 GHz <23 dBm; WLAN 5.3/5.6 GHz <20 dBm; This equipment should be installed and operated with minimum distance 20cm between the radiator & your body. Transmitter must not be co-located or operating in conjunction wirh any other antenna or transmitter. The simplified EU declaration of conformity referred to in Article 10(9) shall be provided as follows: Hereby, Grandstream Networks, Inc.
Table of Contents DOCUMENT PURPOSE ............................................................................................... 12 CHANGE LOG .............................................................................................................. 13 Firmware Version 1.0.1.8......................................................................................................................... 13 PRODUCT OVERVIEW ......................................................................................
Language & keyboard .......................................................................................................................... 29 Date & Time ......................................................................................................................................... 30 System ..................................................................................................................................................... 31 Security Settings ...................................
Account/Advanced Settings ................................................................................................................. 54 Phone Settings Page Definitions ............................................................................................................. 56 Phone Settings/General Settings ......................................................................................................... 56 Phone Settings/Call Settings ......................................................
Upgrade via USB ..................................................................................................................................... 95 No Local Firmware Servers ..................................................................................................................... 95 Provisioning and Configuration File Download ....................................................................................... 96 FACTORY RESET .............................................................
Table of Tables Table 1: GXV3350 Features in a Glance .................................................................................................... 14 Table 2: GXV3350 Technical Specifications ................................................................................................ 15 Table 3: Equipment Packaging ................................................................................................................... 17 Table 4: GBX20 Extension Module Packaging ......................
DOCUMENT PURPOSE This document describes how to configure the GXV3350 via phone's LCD menu and web UI menu to fully manipulate phone's features. The intended audiences of this document are VoIP administrators. To learn the basic functions of GXV3350, please visit http://www.grandstream.com/support to download the latest “GXV3350 User Guide”.
CHANGE LOG This section documents significant changes from previous versions of administration guide for GXV3350. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. Firmware Version 1.0.1.8 This is the initial version. GXV3350 Administration Guide Version 1.0.1.
PRODUCT OVERVIEW Feature Highlights The following table contains the major features of the GXV3350: Table 1: GXV3350 Features in a Glance 16 lines with up to 16 SIP accounts, up to 6-way audio conference and 3-way 720p 30fps HD video conference, phonebook with up to 1000 contacts, call history with up to 1000 records. Dual switched 10/100/1000Mbps network ports, Dual-band 2.4GHz & 5GHz Wi-Fi (802.11a/b/g/n), PoE/PoE+, Bluetooth 4.2+EDR, USB, HDMI, EHS with Plantronics headsets support.
GXV3350 Technical Specifications The following table resumes all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages and upgrade/provisioning settings for the phone GXV3350. Table 2: GXV3350 Technical Specifications 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, LLDPMED, LDAP, TR-069, 802.
Applications Support Android 7.0 compliant applications to be developed, downloaded and run Deployment on the device with provisioning control HD Audio Yes, 2 omnidirectional microphones, HD handset and speakerphone with support for wideband audio Base Stand Yes, base stand with three adjustable levels Extension Module Yes, can power up to 4 GBX20 EXT modules which feature a 272x480 color LCD, 20 quick-dial/ BLF keys with dual-color LED, 2 navigation keys and less than 1.
GETTING STARTED This chapter provides basic installation instructions including the list of the packaging contents and also information for obtaining the best performance with the GXV3350. Equipment Packaging Table 3: Equipment Packaging GXV3350 1x GXV3350 Main Case. 1x Handset. 1x Phone Cord. 1x Phone Stand 1x Ethernet Cable. 1x 12V Power Adapter. 1x Quick Installation Guide. 1x GPL License. Figure 1: GXV3350 Package Content Note: Check the package before installation.
Using the Phone Stand The GXV3350 has a phone stand. To set up the GXV3350 as your desk phone, follow the steps below: 1. For installing the phone on the table with the phone stand, attach the phone stand to the bottom of the phone where there is a slot for the phone stand, (upper half, bottom part); 2. Connect the handset and main phone case with the phone cord; 3. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the router) using the Ethernet cable; 4.
Figure 3: Tab on The Handset Cradle Connecting the GXV3350 To setup your GXV3350, please follow the steps below: 1. Connect the handset and main phone case with the phone cord; 2. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the router) using the Ethernet cable; 3. Connect the 12V DC output plug to the power jack on the phone; plug the power adapter into an electrical outlet. If PoE switch is used in step 2, this step could be skipped; 4.
Table 4: GBX20 Extension Module Packaging 1x GBX20. 1x GBX20 Stand. 1x Connector. 1x TypeA-MicroB USB Cable. 1x 12V/1A Power Adapter. 1x Screw. 1x Quick Installation Guide. Note: 1. The GBX20 is an additional accessory for the GXV3350. Therefore, theGBX20 (including extension module accessories for installation) is not included in the GXV3350 package. 2.
5. Power up the GXV3350. The GBX20 will show the booting up screen with version information and connecting status. 6. After successfully booting up, the extension board will stay in idle. Press and hold the LEFT button for 3 seconds to check the version information and status. The GBX20 can be configured via the web GUI of the GXV3350 connected. After successfully configured, press LEFT or RIGHT button on the GBX20 and users could browse all the Programmable Keys' status in different pages.
GXV3350 LCD SETTINGS The GXV3350 LCD MENU provides an easy access to the settings on the phone. Some of the settings from Web GUI could be configured via the LCD as well. The following table shows the LCD setting menu options.
Access LCD Settings To open the settings menu, you should: Tap on Swipe down from the top of the home screen to open the notifications panel and hit the Settings app on the screen. Or; Settings icon in the top right corner.
Auto-Answer - IF Enabled and set to "Always", the phone will automatically turn on the speaker phone to answer all incoming calls. - If enabled and set to "Enable Intercom/Paging", the phone will answer the call based on the SIP info header sent from the server/proxy. By default, it's turned off.
- Show received files: Shows the Transfer history of Bluetooth files - Additional Settings: This menu is available only when the Bluetooth is enabled: Device Name. Tap to change the name of the GXV3350, which is displayed on other Bluetooth devices when discovered. By default, it's "GXV3350_XXXXXX" Where XXXXXX are the last 6 digits of the phone’s MAC address. Visibility timeout. Tap to select the timeout interval among "2 minutes", "5 minutes", "1 hour" or "never".
IPv6 Settings: Here user can configure the IPv6 address type only for data. If DHCP is selected, the phone will get an IP address automatically from the DHCP server in the network. This is the default mode. If Static IP is selected, manually enter the information for IP Address, Prefix Length, DNS Server and Alternative DNS server. 802.1x mode: This option allows the user to enable/disable 802.1x mode on the phone. The default setting is disabled. To enable 802.1x mode, select the 802.
o Network notification: If enabled, the phone will show notification on the top status bar indicating an open network is available. By default it's enabled. o MAC address: This shows the MAC address of the WiFi. o IP address: This shows the IP address of the phone from Wi-Fi network. VPN Enable / Disable VPN. General Network Settings LLDP Turn on/off LLDP on the GXV3350.
Proxy Settings For some network setup, it is required to connect to the Internet via proxy server. Manually configure "Proxy hostname", "Proxy port" and "Bypass proxy for" in proxy settings for the phone to get Internet connection successfully. Tethering & Portable Hotspot The GXV3350 can serve as a Wi-Fi access point for other devices to provide wireless access to the network if the Portable Wi-Fi hotspot is turned on. 1. Turn on hotspot by tapping on "Portable Wi-Fi hotspot".
Silent mode. Tap on it to turn on/off the sound from speaker when there is an incoming call. HDMI. Enable/disable audio switch between the phone and the HDMI output connected device (e.g. TV). When enabled, the TV will be used for audio output. Media Volume. Adjust the sound volume for media audio Alarm Volume. Adjust the alarm ring volume Ring Volume. Adjust the phone ringing volume Notification Volume. Adjust the notification sound volume Ringtone.
- Current Keyboard: support to change keyboard. - Virtual Keyboard: Android keyboard (AOSP): Set up the language used on Android keyboard and configure its different parameters including sound, auto-correction, word suggestion and so on. - Manage Keyboards. Tap on the + sign to choose which keyboard to use on the phone.
Use 24-hour format. Check/uncheck to display the time using 24-hour time format or not. For example, in 24-hour format, 13:00 will be displayed instead of 1:00 p.m. Select date format. Select the format of year, month and day for the date to be displayed. System Security Settings Device Security-Screen lock. Set up pattern or password for screen lock. Wizard will be provided to set up the pattern. The screen will be locked after booting up or the screen is off (i.e.
Peripherals Plug in RJ9/UHS Headset. Switch the media channel to RJ9 headset after plugging in the Accounts Add a system account to synchronize contacts calendars and other information. Power Information PoE Power Supply notification. If enabled, the phone would display a notification of “When using PoE power supply, if the power consumption of the device accessed through USB is too high, the phone may restart due to insufficient power supply.” If disabled, the notification will not be shown.
If an application is misbehaving, tap on "Report" softkey (if available) to send the developer information for the application. Memory will show the memories used on the phone by the applications Modify System settings gives the application the permission to modify the system settings Store provides Information about the Install source of the App Note: Stopping a built-in application, operating system processes or services might disable one or more dependent functions on the phone.
Show silently. Advanced Account Settings Account Settings page allows to configure SIP settings for each account. Tap on Account# to access the settings, when configured press ✔ sign (on the top right corner) to confirm the changes, or press back button to cancel them. Users can press Empty configuration on the bottom of the page to clear all the settings. Following settings can be configured for each account. Refer to [Account/General Settings] for description of each option. Account Activation.
o HTTP/HTTPS username: The user name for the HTTP/HTTPS server if set up on the server. o HTTP/HTTPS password: The password for the HTTP/HTTPS server if set up on the server. o Config Server Path: This defines the server path for the provisioning server. It can be different from the firmware server. Syslog Syslog level: Select the level of logging for syslog. The default setting is "None". There are 4 levels: DEBUG, INFO, WARNING and ERROR.
GXV3350 WEB GUI SETTINGS The GXV3350 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. Status Page Definitions Status/Account Status Account 16 SIP accounts on the phone. Number SIP User ID for the account. SIP Server URL or IP address, and port of the SIP server. Status Registration status for the SIP account.
Status/System Info Product Model Product model of the phone: GXV3350. Hardware Revision Hardware version number. Part Number Product part number. Serial Number Product serial number. System Version Firmware version ID. This is the main software release version, which is used to identify the software system of the phone. Recovery Version Recovery image version. Boot Version Booting code version. Kernel Version The kernel version. CPE Version The CPE version.
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. Disabled: Will use “SIP User ID” information in the Request-Line and “From” header. User=Phone: “User=Phone” parameter will be attached to the Request-Line and “From” header in the SIP request to indicate the E.164 number. If set to "Enable". Enabled: "Tel:" will be used instead of "sip:" in the SIP request.
Saved one until DNS TTL (Stay on responding IP until DNS timeout): On this mode, the phone will resolve DNS SRV records and tries to send the request to the server having lowest priority and if it doesn’t respond, it will move on to the next IP until one of the servers responds, once this happen the phone will keep contacting this responding IP until DNS timeout (30 minutes) before starting over.
Account/SIP Settings 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 Controls whether to clear SIP user’s information by sending un-register Registration request to the proxy 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. GXV3350 phone system supports both synchronized and non-synchronized MWI.
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. SIP URI Scheme When Defines which SIP header, "sip" or "sips", will be used if TLS is selected Using TLS for SIP Transport. T The default setting is "sip". Use Actual Ephemeral Determines the port information in the Via header and Contact header of Port in Contact with SIP message when the phone system use TCP or TLS.
If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800. Min-SE (s) Determines the minimum session expiration timer (in seconds) if the phone act as a timer refresher. The default setting is 90.
Account/Codec Settings Preferred Vocoder Preferred Vocoder 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. Codec Negotiation Configures the phone to use which codec sequence to negotiate as the Priority callee.
Audio RED Payload Configures audio RED payload type. The valid range is from 96 to 126. Type 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.
H.264 Image Size Sets the H.264 image size. It can be selected from the dropdown list. 720P 4CIF VGA CIF QVGA QCIF 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 Configures that whether to set H.264 constrained profiles.
None: 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. 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 126. Presentation Settings Enable BFCP If set to "Yes", the device will be able to receive the presentation stream in video calls and video meetings.
Account/RTP Settings SRTP Mode Sets if the phone system will enable the SRTP (Secured RTP) mode. It can be selected from dropdown list: Disable Enabled but not forced Enabled and forced 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”. The default setting is “Disable”.
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”. “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.
Note: If Auto Answer function has been enabled, this function does not take effect. Default Value: "No". 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 If set to "Yes", anonymous calls will be automatically blocked. The default Calls setting is “No”.
Ring Timeout (s) Defines the expiration timer (in seconds) for the rings with no answer. The default setting is 60. The valid range is from 10 to 300. Refer-To Use Target Sets the phone system to use the target’s Contact header tag to the Refer- Contact 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 Forward Call Forward Type Sets the Call Forwarding feature for this account.
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+ or 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 i) , - play second dial tone.
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.
Match Incoming Caller Specifies the rules for the incoming calls. If the incoming caller ID or Alert ID 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.
Check SIP User ID for Configures the phone system to check the SIP User ID in the Request URI Incoming INVITE 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".
Feature Key This Synchronization synchronization. When it's set to BroadSoft / Metaswitch, DND and Call feature is used for BroadSoft / Metaswitch call feature 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. Allow Sync Phonebook Allows users to synchronize XML phonebook upon receiving SIP NOTIFY via SIP Notify message with header Event: sync-contacts.
Enable Enterprise Configures whether to display the matched content automatically in search Contacts Timeout Auto of the LDAP contacts when timeout. If set to “No”, users need to click the Search “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.
Enable DND Reminder Enables the DND reminder ring. If set to "Yes", the ring splash that Ring indicates an incoming call when DND is enabled will be played. Default setting is “Yes” Enable Transfer Enables the Transfer function. Default settings is “Yes” Hold Call Before When set to "Yes" the phone holds the second call before completing the Completing Transfer attended transfer (it sends the INVITE method to hold the call before sending the REFER method).
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”. Auto Mute on Entry Configures whether to mute the call on entry automatically. If set to "Disable", then do not use auto mute function. If set to "Auto Mute on Outgoing Call", then mute automatically when the other party answers the outgoing call.
Virtual Account Group If set to "Yes", when processing SIP Register 3XX Response, it will parse Avaya Mode 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.
Enable Auto Record Configures whether to auto record when a call is established. If set to "Yes", When Call Established the call recording will start automatically when the call is established. Note: If Record Mode is set to “Disabled”, it will not be configured. Rejected Call Specifies whether to enable rejected call notification. Once enabled, a Notification missed call will prompt on LCD when rejecting an incoming call.
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. Phone Settings/Video Settings Video Frame Rate Configures video frame rate for SIP video call from “5 frames/second”, “15 frames/second”, “25 frames/second” and “30 frames/second”. The default setting is 15 frames/second.
Paging Priority Active 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 Enables the video feature to establish a multicast paging call. The default Video setting is disabled.
Different Networks for Configures whether to set up different networks for the phone data and the Data and VoIP Calls call. If set to "Yes", you need to configure the data network and VoIP network respectively. IPv4 IPv4 Address Type Configures the appropriate network settings on the phone. Users could select from "DHCP", "Static IP" or "PPPoE"(Point-to-point Protocol over Ethernet). By default, it is set to "DHCP". DHCP VLAN Override Selects the DHCP Option VLAN mode.
DNS Server 2 Configures the secondary DNS IP address. Static IPv6 Address Enter the static IPv6 address in "Statically configured" IPv6 address type. IPv6 Prefix Length Enter the IPv6 prefix length in "Statically configured" IPv6 address type. Default is 64. Preferred DNS Server IPv4 Address Type Users could select "DHCP" or "Static IP". • DHCP: Obtain IP address via a DHCP server in the LAN. All domain values for static IP/PPPoE are unavailable, even though the values have been saved in the flash.
Configures the Wi-Fi frequency band from the list: WiFi Band 2.4G 5G 2.4 G & 5G Default setting is 2.4G & 5G. Permits to scan and select the available Wi-Fi networks within the range if ESSID the Wi-Fi feature is enabled. Click on "Select" to select the Wi-Fi network to connect to. The ESSID will be auto filled in the ESSID filed. Add Network ESSID Configure the hidden ESSID name.
OpenVPN® CA OpenVPN® CA file (ca.crt) required by the OpenVPN® server for authentication purposes. Press "Upload" to upload the corresponding file to the device. OpenVPN® Client OpenVPN® Client certificate file (*.crt) required by the OpenVPN® server Certificate for authentication purposes. Press "Upload" to upload the corresponding file to the device. OpenVPN® Client Key The OpenVPN® Client key (*.key) required by the OpenVPN® server for authentication purposes.
SIP User-Agent Sets the user-agent for SIP. PC Port Mode PC Port Mode Enables and defines the PC port mode. If it is set to “Mirrored”, the traffic in the LAN port will go through PC port as well and packets can be captured by connecting a PC to the PC port. The default setting is "Enable". PC Port VLAN Tag Defines the VLAN Identifier of the Layer 2 frame for PC port.
DHCP Option 2 to override Time Zone setting Time Zone Time Display Format Obtains time zone setting (offset) from a DHCP server using DHCP Option 2; it will override selected time zone. If set to “No”, the phone will use selected time zone even if provided by DHCP server. The default setting is Yes. Specifies the local time zone for the phone. It covers the global time zones and user can selected the specific one from the drop-down list. Specifies which format will be used to display the time.
Note: When access control for keypad is limited to “Basic Settings Only” or “Constraint Mode”, the Admin authentication will be mandatory to start Factory Reset process. Permission to Install/Uninstall Apps Configures the permissions for users to install/uninstall the applications. If set to "Allow", the user is free to install/uninstall third-party apps. If set to "Require admin password", the user need to input the correct administrator password to install/uninstall third-party apps.
Certificate Management CA Certificate Import Trusted CA Certificates Trusted CA Certificates Allows to upload the CA Certificate file to phone. Lists trusted CA certificates previously uploaded. Administrator can delete a certificate from here. User Certificate Add Certificate Allows to upload & Install User Certificate file to phone. Custom Certificate Import Custom Certificate Custom Certificate Allows to upload a Custom Certificate file to phone. Lists trusted Custom Certificate previously uploaded.
If set to "Show Opposite Screen", HDMI device displays remote video screen in a video call while other screens display synchronization with LCD. This setting will take effect in the next call. Audio Control RJ9 Headset TX Gain Configures the Transmission Gain in RJ9 headset channel. (dB) It can be selected from the dropdown list. The default setting is 0dB: -24 -18 -12 -6 0 +6 +12 +18 +24 RJ9 Headset RX Gain Configures the Receive Gain in RJ9 headset channel.
Virtual Sound Card TX Configures the transmission gain of the virtual sound card, it can be gain(dB) selected from the dropdown list. -18 -15 -12 -9 -6 -3 0 +3 +6 +12 +15 +18 The default setting is “0dB”. Virtual Sound Card RX Configures the virtual sound card received audio signal, it can be selected gain(dB) from the dropdown list.
ACS URL Specifies URL of TR-069 ACS (e.g., http://acs.mycompany.com), or IP address. ACS User Name Enters username to authenticate to ACS. ACS Password Enters password to authenticate to ACS. Periodic Inform Enable Sends periodic inform packets to ACS. Default is “No”. Periodic Inform Interval Configures to sends periodic “Inform” packets to ACS based on specified (s) interval. Connection Request Enters user name for the ACS to connect to the phone.
Firmware Upgrade Click the "Update Detect" button to check whether the firmware in the firmware server has an updated version, if so, update immediately. Config File Config File: Configure Manually Download Device Downloads the phone's configuration file in text format. The config file Configuration includes all the P value parameters for phone's current settings except password for security purpose.
XML Config File Decrypts XML configuration file when encrypted. The password used for Password encrypting the XML configuration file is using OpenSSL. CUST File GUI Customization File Selects download method: TFTP, HTTP or HTTPS. Download Mode Default setting is “HTTPS”. GUI Customization File Sets IP address or domain name of the GUI customization file server. The URL server hosts a copy of the file to be installed on the phone. The Default setting is fm.grandstream.com/gs.
Firmware Upgrade and Defines the phone system’s rules for automatic upgrade. It can be selected Provisioning from: Always Check at bootup Always Check at bootup, when F/W pre/suffix changes, Skip the Firmware Check. The default setting is “Always Check at bootup”. Upgrade with Prompt If set to "No", the phone will automatically start upgrading after downloading the firmware file. Otherwise, users would need to confirm in the prompted message on the LCD screen to start upgrading process.
The URL will be included in the SIP NOTIFY message. PnP(3CX) Auto Sets the phone system to broadcast the SIP SUBSCRIBE message during Provision booting up to allow itself to be discovered and be configured by the SIP platform. The default setting is "Yes". Advanced Settings Enable SIP NOTIFY Enables the phone to challenge SIP NOTIFY with 401. Authentication The default setting is “Yes”.
Selects the level of logging for syslog. The default setting is "None". There are 4 levels from the dropdown list: DEBUG, INFO, WARNING and ERROR. The following information will be included in the syslog packet: DEBUG (Sent or received SIP messages). INFO (Product model/version on boot up, NAT related info, SIP Syslog Level message summary, Inbound and outbound calls, Registration status change, negotiated codec, Ethernet link up). WARNING (SLIC chip exception).
Click the "Delete" button on the right to delete the file. Core Dump Enable Core Dump Configures whether to generate and save the core dump file when the Generation program crashes. The default setting is “No”. Core Dump List Selects the existing core dump file in the drop-down box. Users could delete the file by pressing on “Delete” button. View Core Dump Press “List” button to view all existing core dump files.
3. If users need to add multiple dynamic variables in the same event, users could use "&" to connect with different dynamic variables. For example: 192.168.40.207/mac=$mac&local=$local 4. When the corresponding event occurs on the phone, the phone will send the MAC address and phone number to server address 192.168.40.207. Bootup Completed Configures the event URL when phone boots up. Incoming Call Configures the event URL when phone has an incoming call.
Programmable key Format Display Format Configures the display format for the MPK. Users could select "Name", "User ID" or "Name(User ID)". "Name" is the one saved in phone contacts. The default setting is "Name, User ID, Key mode”. Show Display Name If selected, the display name on the server will replace the name users from Server configured. BLF Key Mode The key modes are: Speed Dial: Press to dial the User ID when the accounts being configured as VPK.
Display Name Configures the display name when the accounts being configured as VPK. User ID Configures the User ID for the corresponding VPK mode when the accounts being configured as VPK. DTMF Content When key mode is set to Dial DTMF it configures the dialed DTMF content. Address When key mode is set to Multicast Paging, it configures the multiple broadcast address. Conference name Set the name of speed meeting when the key mode is set to Quick Conference.
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 User ID when the accounts being configured as VPK. Dial DTMF: Dial the DTMF digits of the User ID when the accounts being configured as VPK during the call. Call Park: Configure the call park feature code to park or retrieve the call.
BLF Call-pick Prefix Configures the prefix prepended to the BLF extension if the phone answers a call to the monitored party by the BLF key. Default setting is ** for each account. Determines the event list BLF URI on the phone to monitor the extensions in the list with MPK keys. This feature is based on BroadSoft standard. It Event List URI requires filling in the BLF ID to the box. For example, if the server provides the URI: BLF123@myserver.com, this field should be filled with BLF123.
Replace Duplicate Items Configures the phone system to keep the original contact entries when duplicated contact entries are included in the contact file. If set to "Yes", the phone will replace the original entries to the new one. Otherwise, the phone system will save both contact entries. The default setting is "No". Replace Duplicate If set to "Replace by name", replace the records of the same name automatically when importing new records.
File Type Sets the type format for phonebook file importing. It can be selected from the dropdown list. XML VCard The default setting is "XML". Export Downloads the phonebook file from the phone to PC. Download Contacts Clear The Old List Sets the phone system to delete the previous contacts when a new contact file is downloaded. If "Yes", the previous contacts will be removed. The default setting is "No".
The phone system will send a request to the server to download the phonebook file with filename phonebook.xml. HTTP/HTTPS User Name Configures user name for HTTP/HTTPS server to download the phonebook file. HTTP/HTTPS Password Specifies password for HTTP/HTTPS server to download phonebook file. Automatic Download Determines how the phone system to send the request to the server to Interval download the phonebook file.
LDAP Number Configures the "number" attributes of each record which are returned in the Attributes LDAP search result. This field allows the users to configure multiple space separated number attributes. Example: telephoneNumber telephoneNumber Mobile LDAP Mail Attributes Determines the "mail" attributes of each record which are returned in the LDAP search result. Example: mail LDAP Name Filter Configures the filter used for name lookups.
System Application/Recording File name Displays the name of the recording file Duration Displays the duration of the phone call Date Displays the date the call was recorded on Operation Delete, Modify or download the recording file Value-added Service Page Definitions Value-added Service/Value-added Service (0/10) Service Type Users can set the service type to “Door System” to configure the door system options, or to “DTMF” to set DTMF content to send it during calls.
Determines the door system password which should match the one Access password configured on the used door system settings. In case the GDS is set as ‘Door System type’ parameter, the password should match the one configured on the GDS to open door 1. Related Display Name2 Indicates the name that will be displayed on LCD for door 2 when the call matches the configured system number. Access password The configured password should match the one configured on the GDS to open door 2.
Auto-filling CallPark If it is set to "Yes", the configured "Call Park Service Code" will be Feature Code automatically filled in on the phone's dial pad when picking up the parked call. This option will be active only if "Special Mode" is set to "Broadsoft" and "Enable SCA" is set to "Yes". The default setting is "Yes". CallPark Feature Code Configures the pickup feature code for call park.
UPGRADING AND PROVISIONING The GXV3350 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP, HTTP or HTTPS; the server name can be FQDN or IP address. Examples of valid URLs: firmware.grandstream.com/BETA fw.mycompany.com Upgrade and Provisioning Configuration There are two ways to setup upgrade and provisioning on GXV3350. They are Keypad Menu and Web GUI.
Figure 10: GXV3350 Upgrade Configuration via LCD Configure via Web GUI Open a web browser on PC and enter the IP address for the GXV3350. Then login with the administrator username and password (that needs to be at least 6 characters). Go to Maintenance Upgrade. In the Upgrade web page, enter the IP address or the FQDN for the upgrade server and choose to upgrade via TFTP, HTTP or HTTPS (The default setting is HTTPS).
Upload Firmware Locally If there is no HTTP/TFTP server, users could also upload the firmware to the GXV3350 directly via Web GUI. Please follow the steps below to upload firmware to GXV3350 locally. 1. Download the latest GXV3350 firmware file from the following link and save it in your PC. http://www.grandstream.com/support/firmware 2. Log in the Web GUI as administrator in the PC. 3. Go to Web GUIMaintenanceUpgrade. 4.
Please check our web site at http://www.grandstream.com/support/firmware for latest firmware. Instructions for local firmware upgrade via TFTP: 1. Unzip the firmware files and put all of them in the root directory of the TFTP server; 2. Connect the PC running the TFTP server and the GXV3350 device to the same LAN segment; 3. Launch the TFTP server and go to the File menuConfigureSecurity to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade; 4.
FACTORY RESET Restore to Factory Default via LCD Menu Warning: Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider. In order to restore the GXV3350 unit to factory reset via the LCD Menu, please refer to the following steps: 1.
3. A dialog box will pop up to confirm factory reset; 4. Click OK to restore the phone to factory settings. Figure 15: GXV3350 Web GUI - Confirm Factory Reset Restore to Factory Default via Hard Keys For users that could not restore the GXV3350 to factory reset via LCD Menu or the Web GUI, restoring the unit via Hard keys is an alternative. Please, follow the steps below to restore the GXV3350 via Hard Keys: 1. Power cycle the GXV3350. 2.
SAFE MODE The GXV3350 allows users to enter safe mode by enabling safe mode option from WEB UI under MaintenanceUpgrade/Advanced Settingssafe mode. If enabled, the phone will enter safe mode after rebooting. Users can alternatively enter safe mode by pressing the Menu button during bootup.
EXPERIENCING THE GXV3350 APPLICATION PHONE Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products. We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
