Grandstream Networks, Inc.
GXP2120/GXP2110/GXP2100/GXP1450 User Manual Index GNU GPL INFORMATION .......................................................................... 5 CHANGE LOG ........................................................................................... 6 FIRMWARE VERSION 1.0.4.23 ............................................................................................................ 6 FIRMWARE VERSION 1.0.4.9 ..................................................................................................
VOICE MESSAGES (MESSAGE WAITING INDICATOR) ........................................................... 29 SHARED CALL APPEARANCE (SCA) ........................................................................................ 29 CALL FEATURES ................................................................................................................................ 30 CUSTOMIZED LCD SCREEN & XML .................................................................................................
Table 6: LCD DISPLAY DEFINITIONS........................................................................................................ 15 Table 7: LCD ICONS ................................................................................................................................... 17 Table 8: KEYPAD DEFINITIONS ................................................................................................................ 20 Table 9: CALL FEATURES ...................................................
GNU GPL INFORMATION GXP2120/GXP2110/GXP2100/GXP1450 firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu_gpl. FIRMWARE VERSION 1.0.4.
CHANGE LOG This section documents significant changes from previous versions of user manuals for GXP2120/GXP2110/GXP2100/GXP1450. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. FIRMWARE VERSION 1.0.4.23 • Updated XML Application documentation link. [CUSTOMIZED LCD SCREEN & XML] • Added "Use Privacy Header" and "Use P-Preferred-Identity Header" options in web GUI.
WELCOME Thank you for purchasing Grandstream GXP2120/GXP2110/GXP2100/GXP1450 SIP Enterprise Phones. Your Grandstream GXP2120/GXP2110/GXP2100/GXP1450 Enterprise IP phone is feature-enriched, sophisticated, yet simple to use.
PRODUCT OVERVIEW FEATURE HIGHTLIGHTS Table 1: GXP2120/GXP2110/GXP2100/GXP1450 FEATURES IN A GLANCE GXP2120 is an executive SIP phone. It features: GXP2120 6 lines 7 programmable Multi Purpose Keys 4 XML programmable soft keys GXP2110 is an executive SIP phone. It features: GXP2110 4 lines 18 programmable Multi Purpose Keys 3 XML programmable soft keys GXP2100 is an executive SIP phone.
Table 2: GXP2120/GXP2110/GXP2100/GXP1450 COMPARISON GUIDE Features GXP2120 GXP2110 GXP2100 GXP1450 LCD Display 320x160 pixel 240x120 pixel 180x90 pixel 180x60 pixel Number of Lines 6 4 4 2 Programmable 7 18 7 N/A 4 3 3 3 Yes, Yes, N/A N/A up to 2 Expansion up to 2 Expansion Modules, Modules, 56 nodes each 56 nodes each Hard Keys Programmable Soft Keys Extension Module GXP2120/GXP2110/GXP2100/GXP1450 TECHNICAL SPECIFICATIONS Table 3: GXP2120/GXP2110/GXP2100/GXP1450 TECHNIC
Feature Keys Voice Codec Feature key comparison: GXP2120 GXP2110 GXP2100 GXP1450 HOLD Yes Yes Yes Yes SPEAKERPHONE Yes Yes Yes Yes SEND Yes Yes Yes Yes TRANSFER Yes Yes Yes Yes CONF Yes Yes Yes Yes MUTE Yes Yes Yes Yes DND Yes Yes No No HEADSET Yes Yes Yes Yes INTERCOM Yes Yes Yes No PHONEBOOK Yes Yes Yes Yes MSG Yes Yes Yes Yes MENU Yes Yes Yes Yes NAVIGATION (4) Yes Yes Yes Yes Support for G.723.1, G.729A/B, G.711u/a, G.726-32, G.
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Multi-language Simplified Chinese, traditional Chinese, Korean, Japanese, and etc Upgrade and Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or Provisioning AES encrypted XML configuration file Power and Green Universal power adapter: Energy Efficiency Input: 100-240VAC 50-60Hz; Output: 5VDC, 800mA Integrated Power-over-Ethernet (Built-in auto-sensing: Cisco and IEEE 802.
INSTALLATION EQUIPMENT PACKAGING Table 4: GXP2120/GXP2110/GXP2100/GXP1450 EQUIPMENT PACKAGING Main Case Yes (1) Handset Yes (1) Phone Cord Yes (1) Power Adaptor Yes (1) Ethernet Cable Yes (1) Phone Stand Yes (2) Wall Mount Spacers Yes (2) Quick Start Guide Yes (1) CONNECTING YOUR PHONE Table 5: GXP2120/GXP2110/GXP2100/GXP1450 CONNECTORS Handset Port RJ9 handset connector port Headset Port 2.
6. Using the keypad configuration menu or phone's embedded web server (Web GUI) by entering the IP address in web browser, you can further configure the phone. GXP2120/GXP2110 EXTENSION MODULE GXP2120/2110 supports two extension units, providing up to 112 additional programmable extensions. Each GXP Extension unit has 56 multi-purpose keys, dual color LEDs (red/green) and support BLF (Busy Lamp Field) and Presence.
Note: • Should your system lose power, please unplug your devices and power up the GXP2120/2110 first. • Extension for GXP2120/2110 is the same for GXP2020/2010 models. However, GXP2120/2110 uses a different-shaped connector for the special port (as shown above). Extension cables will be included with the extension board. • Extension for GXP2120/2110 does not support hot-swap. Once connected, user should reboot the phone to ensure the set up will work correctly.
USING THE GXP2120/GXP2110/GXP2100/GXP1450 GETTING FAMILAR WITH THE LCD GXP2120/GXP2110/GXP2100/GXP1450 has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active). The following table describes the items displayed on the GXP2120/GXP2110/GXP2100/GXP1450 idle screen. Table 6: LCD DISPLAY DEFINITIONS DATE AND TIME Displays the current date and time. It can be synchronized with Internet time servers. Displays company logo/name.
Note: If XML application is used for GXP2120/GXP2110/GXP2100, the softkey for XML application will show up in the default idle screen as configured. The softkeys are context sensitive and will change depending on the call status of the phone. • Redial Redials the last dialed number after off hook when there is existed call log. • Dial Dials the call out after off hook and entering the number. • Hold Puts the current active call on hold.
Call Parking: Please refer to GXE5024/5028 Online User Manual for more information. • CallPark When the phone dials out, the Call Park softkey will display on screen. To park the call, press the "Call Park" softkey and select MPK to park the call. • PickUp When the phone goes off-hook, the Call Pickup softkey will display on SPECIAL screen. To pick up the parked call, press the "Call Pickup" softkey.
Call Forward No Answer Status. OFF - Call Forward No Answer feature disabled ON - Call Forward No Answer feature enabled Call Forward All and Call Forward No Answer Status. OFF - Call Forward All and Call Forward No Answer feature disabled ON - Call Forward All and Call Forward No Answer feature enabled Keypad Status. OFF - keypad is unlocked ON - keypad is locked Enter Keypad Unlock Password. Voicemail Status. OFF - No new voicemail ON - New voicemail Instant Message.
Speaker Status. OFF - speaker off ON - speaker on Headset Status. OFF - headset off ON - headset on Calling Out. The phone is calling out Calling In. Phone is ringing with incoming call Incoming Call. The current call is an incoming call Outgoing Call. The current call is an outgoing call Call Failed Fail to establish call SRTP Status. OFF - SRTP is not used ON - SRTP is used MUTE Status. OFF - No muted ON - Muted Call On Hold. Call Active. Conference Call. Core Dump.
GETTING FAMILAR WITH THE KEYPAD The following table describes the buttons used on the GXP2120/GXP2110/GXP2100/GXP1450 keypad. Table 8: KEYPAD DEFINITIONS GXP21xx GXP1450 Definition Open or switch line. LINE KEYS Place active call on hold, or resume the call on hold. HOLD Send/Redial. • SEND Send. Enter the digits and then press Send to dial out the number; • TRANSFER TRANSFER CONF CONF MUTE Redial. Redial when there is a previously dialed call. Transfer an active call to another number.
PHONEBOOK/ CONTACT Display phonebook list and options. Volume (For GXP2100/GXP1450 only). On GXP2100/GXP1450, press "-" or "+" to adjust the volume; On GXP2120/GXP2110, when the phone is off hook, press UP or DOWN button to adjust the volume. 0 - 9, *, # Multi Purpose Keys Standard phone keypad. Configure key mode and User ID to use the Multi Purpose key as Speed Dial, BLF, Call Park and etc.
COMPLETING CALLS There are several ways to complete a call on GXP2120/GXP2110/GXP2100/GXP1450. • On hook dialing. Enter the number when the phone is on hook and then send out. When the phone is in idle, enter the number to be dialed out; Take handset off hook; or Press Speaker button; or Press Headset button with headset plugged in; or Select an available LINE key; • The call will be dialed out. Off hook and dial. Off hook the phone, enter the number and send out.
Select and enter Phonebook; Select the phonebook entry you would like to call using the navigation arrow keys. Press MENU button to enter the entry detail; • Select option "Dial" using the navigation arrow keys and then press MENU button again to dial out. Speed Dial. Dial the number configured as Speed Dial on Line Key. Go to phone's Web GUI->Basic Settings, configure the Line Key's Key Mode as Speed Dial.
MAKING CALLS USING IP ADDRESSES Direct IP Call allows two phones to talk to each other in an ad-hoc fashion without a SIP proxy. VoIP calls can be made between two phones if: • Both phones have public IP addresses; or • Both phones are on the same LAN/VPN using private or public IP addresses; or • Both phones can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
For example: • 192.168.0.2 calling 192.168.0.3 -- dial #3 followed by # or “SEND”; • 192.168.0.2 calling 192.168.0.23 -- dial #23 followed by # “SEND”; • 192.168.0.2 calling 192.168.0.123 -- dial #123 followed by # “SEND”; • 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3. Note: • The # will represent colon ":" in direct IP call rather than SEND key as in normal phone call; • If you have a SIP server configured, direct IP call still works.
DURING A PHONE CALL CALL WAITING/CALL HOLD • Hold. Place a call on hold by pressing the HOLD button. The active LINE key will blink in green; • Resume. Resume call by pressing the blinking LINE key; • Multiple calls. Automatically place active call on hold or switch between calls by pressing the LINE key. Call waiting tone (stutter tone) will be audible on incoming call during the active call. MUTE During an active call, press the MUTE button to mute/unmute the microphone.
Establish one call first; During the call, press TRAN key. A new line will be brought up and the first call will be automatically placed on hold; Enter the number and press SEND key to establish the second call; After the second call is established, press TRAN key again. The call will be transferred; If users press the SPLIT softkey before the call is transferred in the step above, the second call will be resumed.
Select 1 LINE key and press to resume the 2-way conversation; If users would like to re-establish conference call, before 1 separate LINE is selected, press the ReConf softkey right after the conference call is held/split; • End Conference. Press HOLD key to split the conference call. The conference call will be ended with both calls on hold; Or Users could press the EndCall softkey or simply hang up the call to terminate the conference call.
• End Conference. Press HOLD key to split the conference call. The conference call will be ended with both calls on hold; Or Users could press the EndCall softkey or simply hang up the call to terminate the conference call. Note: • The party that starts the conference call has to remain in the conference for its entire duration, you can put the party on mute but it must remain in the conversation. Also, this is not applicable when the feature "Transfer on Conference Hangup" is turned on.
by the red-flashing button and they will be able to resume the call from their phone by pressing the line button. However, if this call is placed on private-hold, no other member of the SCA group will be able to resume that call. To enable shared call appearance, the user would need to register the shared line account on the phone. In addition, they would need to navigate to “Settings”->”Basic Settings” on the web UI and set the line to "Shared Line".
• Dial *71 and then enter the number to dial out. Unconditional Call Forward. To set up unconditional call forward: *72 • Off hook the phone; • Dial *72 and then enter the number to forward the call; • Press OK softkey or SEND key. Cancel Unconditional Call Forward. To cancel the unconditional call forward: *73 • Off hook the phone; • Dial *73; • Hang up the call. Busy Call Forward.
• XML custom idle screen (customize idle screen logo, softkey layout, and etc.) http://www.grandstream.com/products/gxp_series/general/documents/GXP21xx_14xx_XML_Screen_ Customization.zip • XML downloadable phonebook http://www.grandstream.com/products/gxp_series/general/documents/gxp_wp_xml_phonebook.pdf • XML Application (For GXP21xx only) http://www.grandstream.com/products/gxp_series/general/documents/xml_application_guide.zip FIRMWARE VERSION 1.0.4.
CONFIGURATION GUIDE The GXP2120/GXP2110/GXP2100/GXP1450 can be configured via two ways: • LCD Configuration Menu using the phone's keypad; • Web GUI embedded on the phone using PC's web browser. CONFIGURATION VIA KEYPAD To configure via the LCD configuration menu using phone's keypad, follow the instructions below: • Enter MENU options. When the phone is in idle, press the round MENU button to enter the • Navigate in the menu options.
searching. Instant Messages Displays received instant messages. Direct IP Call Makes direct IP call. Preference Preference sub menu includes the following options: • Do Not Disturb Enables/disables Do Not Disturb on the phone. • Forward Call Configures call forward feature on selected account, forward type and number. • Ring Tone Configures different ring tones for incoming call. • Ring Volume Adjusts ring volume by pressing left/right arrow key.
Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and Audio information to register SIP account on the phone. • Upgrade Configures firmware server and config server for upgrading and provisioning the phone. • Factory Reset Resets the phone to factory default settings. • Layer 2 QoS Configures 802.1Q/VLAN Tag and priority value.
MENU Call History Status Phone Book LDAP Directory Instant Messages Direct IP Call Preference Config Factory Answered Calls Dialed Calls Missed Calls Transferred Calls Forwarded Calls Clear All Back First Name Last Name Number Acct Groups Confirm Add Cancel & Return Groups New Entry Search Download Phonebook XML Delete All Entries Back Server Address Port Base User Name Password LDAP Number Filter LDAP Name Filter LDAP Version ...
CONFIGURATION VIA WEB BROWSER The GXP2120/GXP2110/GXP2100/GXP1450 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft’s IE, Mozilla Firefox and Google Chrome. To access the 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. You may check the IP address by pressing SwitchSCR softkey or go to MENU->Status; 3.
• Status: Displays the Account status, Network status, and System Info of the phone; • Account: To configure the SIP account; • Basic Settings: To configure basic network settings, time settings, Line keys, and etc; • Advanced Settings: To configure advanced network settings, upgrading and provisioning, language settings, call features, and etc. • Extension Boards: To configure the Multi Purpose Keys on the connected extension boards for GXP2120/GXP2110.
SIP Server Secondary SIP Server The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (ITSP). The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails. IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session Border Controller. It's used by the phone for Outbound Proxy Firewall or NAT penetration in different network environments.
notify the server to unbind the connection. The default setting is "No". Specifies the frequency (in minutes) in which the phone refreshes its Register Expiration registration with the specified registrar. The default value is 60 minutes. The maximum value is 64800 minutes (about 45 days). Specifies the time frequency (in seconds) that the phone sends Reregister Before Expiration re-registration request before the Register Expiration. The default value is 0.
"Keep-Alive". Configure this to be "No" if an outbound proxy is used. "STUN" cannot be used if the detected NAT is symmetric NAT. When set to "Yes", a SUBSCRIBE for Message Waiting Indication will SUBSCRIBE for MWI be sent periodically. The phone supports synchronized and non-synchronized MWI. The default setting is "No". SUBSCRIBE for Registration When set to "Yes", a SUBSCRIBE for Registration will be sent out periodically. The default setting is "No".
e) [147] - any digit of 1, 4, or 7 f) <2=011> - replace digit 2 with 011 when dialing g) | - the OR operand • 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+} Allows any number with leading digit 1 followed by a 3 digit number, followed by any number between 2 and 9, fo
picks up a call with BLF key. The default setting is **. (Not applicable to GXP1450) Delayed Call Forward Wait Defines the timeout (in seconds) before the call is forwarded on no Time answer. The default value is 20 seconds. When enabled, Do No Disturb, Call Forward and other call features will Enable Call Features be supported locally provided ITSP support those features. The default setting is "Yes". If set to "No", ForwardAll softkey will be hidden for Account 1.
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. Account Ring Tone Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu. Specifies matching rules with number, pattern or Alert Info text.
Contact header information for attended transfer. The default setting is "No". Transfer on Conference Defines whether or not the call is transferred to the other party if the Hangup initiator of the conference hangs up. The default setting is "No". Check SIP User ID for incoming INVITE If set to "Yes", SIP User ID will be checked in the Request URI of the incoming INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected. The default setting is "No".
between 96 and 127. Jitter Buffer Type Jitter Buffer Length Selects either Fixed or Adaptive based on network conditions. The default setting is "Adaptive". Selects Low, Medium, or High based on network conditions. The default setting is "Medium". Configures the eventlist BLF URI on the phone to monitor the extensions in the list with Multi Purpose Key. If the server supports this feature, users need to configure an eventlist BLF URI on the service side first (i.e., BLF1006@myserver.
characters. Confirm Password Confirms the end user password field to be the same as above. Internet Protocol Selects Prefer IPv4 or Prefer IPv6. Allows users to configure the appropriate network settings on the phone IPv4 Address Type to obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE". By default, it is set to "DHCP". DHCP Host name (Option 12) DHCP Vendor Class ID (Option 60) Specifies the name of the client.
to EAP-MD5. Identity Enter the Identity for the 802.1x mode. MD5 Password Enter the MD5 Password for the 802.1x mode. Specifies the HTTP proxy URL for the phone to send packets to. The HTTP Proxy proxy server will act as an intermediary to route the packets to the destination. Specifies the HTTPS proxy URL for the phone to send packets to. The HTTPS Proxy proxy server will act as an intermediary to route the packets to the destination. Assigns a function to the corresponding line key.
• Voice Mail Select Account and enter the Voice Mail access number in the UserID field. • Call Return The last answered calls can be dialed out by using Call Return. The Name and UserID field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call. • Transfer Select Account, and enter the number in the UserID field to be transferred (blind transfer) during the call.
• 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 Speed Dial number using account 2. • Dial DTMF Enter a series of DTMF digits in the UserID field to be dialed during the call. "Enable MPK Sending DTMF" (under Advanced Setting) has to be set to "Yes" first. • Voice Mail Select Account and enter the Voice Mail access number in the UserID field.
U.S central time. If it is positive (+) if the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is east. M4.1.0,M11.1.0 The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec) The 2nd number indicates the nth iteration of the weekday: (1st Sunday, rd 3 Tuesday…) The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, ...
Configures the date display format on the LCD. The following formats are supported: Date Display Format Time Display Format Disable in-call DTMF Display Disable Missed Call Backlight • yyyy-mm-dd: 2012-07-02 • mm-dd-yyyy: 07-02-2012 • dd-mm-yyyy: 02-07-2012 • dddd, MMMM dd: Friday, October 12 (Not applicable to GXP1450) • MMMM dd, dddd: October 12, Friday (Not applicable to GXP1450) Configures the time display in 12-hour or 24-hour format on the LCD. The default setting is in 12-hour format.
between Headset and Speaker. Call History Flash Writing: Defines the interval (in seconds) to save the call history to phone's flash. Write Timeout The default value is 300 seconds. Max Unsaved Log Headset TX gain Headset RX gain Handset TX gain Defines the number of unsaved logs before written to phone's flash. The default value is 200 entries. Configures the transmission gain of the headset. The default value is 0dB. Configures the receiving gain of the headset. The default value is 0dB.
The IP address or Domain name of the STUN server. STUN resolution STUN Server results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN. Firmware Upgrade and Provisioning Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, Always Skip the Firmware Check. The password for encrypting the XML configuration file using OpenSSL.
TR-069 Username ACS username for TR-069. TR-069 Password ACS password for TR-069. Periodic Inform Enable Periodic Inform Interval Connection Request Username 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. The user name for the ACS to connect to the phone. Connection Request Password The password for the ACS to connect to the phone.
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 name lookups.
Search Timeout Sort Results LDAP Lookup 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. Specifies whether the searching result is sorted or not. The default setting is "No". Configures to enable LDAP number searching when dialing and receiving calls. Configures the display name when LDAP looks up the name for incoming call or outgoing call.
are 4 levels: DEBUG, INFO, WARNING AND ERROR.
Call-Waiting Tone Busy Tone Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]; Reorder Tone (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.
NETWORK options will not display for users to access in keypad menu. If set to "Yes", the keypad can be locked by pressing and holding the STAR * key for about 4 seconds. A lock icon will show indicating the keypad is locked. The default setting is "Yes". Enable STAR key Keypad locking Note: When the keypad is locked, users would need press and hold the STAR * key for about 4 seconds again and then enter the password to unlock it.
• NAT Traversal (under Account Setting page) Default setting is "No". Enable the device to use NAT traversal when it is behind firewall on a private network. Select Keep-Alive, Auto, STUN (with STUN server path configured too) or other option according to the network setting.
PUBLIC MODE The GXP2120/GXP2110/GXP2100/GXP1450 supports hot desking using public mode. Under public mode, users could login the phone with the SIP account User ID and password. Please follow the steps below to configure the phone for public mode: • Under Web GUI->Account 1 setting page, fill up the SIP server address for account 1. Click "Update" on the bottom of the page; • Under Web GUI->Advanced setting page, set Public Mode option to "Yes".
• phonenumber=1234: The number for the phone to dial out • account=0: The account index for the phone to make call. The index is 0 for account 1, 1 for account 2, 2 for account 3, and etc. • password=admin: The admin login password of phone's Web GUI. Click group to in select the dropdown menu. Click to Click to input number search in and dial from available phonebook. lines. Click to edit this contact.
Figure 4: Click-to-Dial SAVING THE CONFIGURATION CHANGES After users makes changes to the configuration, press the Update button on the bottom of the Web GUI page. We recommend rebooting or powering cycle the IP phone after saving changes. REBOOTING FROM REMOTE LOCATIONS Press the Reboot button on the bottom of the web GUI page to reboot the phone remotely. The web browser will then display a reboot page with message "The device is rebooting now...". Wait for about 1 minute to log in again.
UPGRADING AND PROVISIONING The GXP2120/GXP2110/GXP2100/GXP1450 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 or HTTP; the server name can be FQDN or IP address. Examples of valid URLs: firmware.grandstream.com fw.ipvideotalk.com/gs There are two ways to setup a software upgrade server: The IVR Menu or the Web Configuration Interface.
Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We recommend completing firmware upgrades in a controlled LAN environment whenever possible. NO LOCAL TFTP/HTTP SERVERS For users that would like to use remote upgrading without a local TFTP/HTTP server, Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via this server. Please refer to the webpage: http://www.grandstream.
server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The "Config Server Path" can be the same or different from the "Firmware Server Path". A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with the “Admin Password” in the Web GUI->Settings->Advanced Settings.
RESTORE FACTORY DEFAULT SETTINGS 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.
EXPERIENCING THE GXP2120/GXP2110/GXP2100/GXP1450 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.
