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Using the GXP SIP Enterprise Phone
GETTING FAMILIAR WITH THE LCD
GXP-2xxx has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active screen). Table 8: LCD Buttons Key Button
LINE SELECTORS SIP PHONE LINES DATE AND TIME LOGO

Key Button Definitions

Selects the phone line printed on its right-hand side. Displays the available phone lines. Choose a phone line by pressing the corresponding line selector on the left-hand side. Displays the current date and time. Can be synchronized with Internet time servers. Displays company logo. This logo can be customized. For more information on customizing the logo, please check page 24. Shows the status of the phone and network. It will indicate whether the network is down, starting or is running (show IP-number). Other messages such as DO NOT DISTURB or ## MISSED CALLS are shown here too. Shows the status of the phone, using icons as shown in the next table. Displays the name of the account that is in use. Select another account by pressing the LINE SELECTOR BUTTONS The soft-buttons are context sensitive and will change depending on the status of the phone. Typical functions assigned to soft-buttons are: NEW CALL Press this button to make a new hand-free call. FORWARD ALL Unconditionally forwards the main phone line to another phone MISSED CALLS This option shows up there were unanswered calls to this phone. The MissedCalls option shows a list of the missed calls CALL RETURN Calls the phone that called/tried to call your phone last. REDIAL Redials the last number END CALL Hangs up phone

NETWORK STATUS

STATUS BAR LINE STATUS INDICATOR
SOFT-BUTTONS (Excluding GXP-2000)

Table 9: LCD Icons

LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon: Solid connected to SIP Server/IP address received Blinking physical connection failed Blank SIP Proxy/Server not registered Phone Status Icon: OFF when the handset is on-hook ON when the handset is off-hook

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Speaker Phone Status Icon: FLASH when phone rings or a call is pending OFF when the speakerphone is off ON when the speakerphone is on DND Icon: ON when the do not disturb is activated Activate by pressing MUTE/DEL button once Calls Forwarded Icon: INDICATES calls are forwarded Follow call forwarding procedures Handset, Speakerphone and Ring Volume Icon: Each icon appears next to the volume icon To adjust volume, use the up/down button
Realtime Clock: Synchronized to Internet time server Time zone configurable via web browser AM/PM indicator
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TABLE 10: GXP KEYPAD BUTTONS Key Button
LINE BUTTONS TRANSFER CONF MUTE HOLD MSG
Line keys with LED, can be configured to different SIP profiles TRANSFER key: Transfer an ACTIVE call to another number Press CONF button to connect Calling/Called party into conference Mute an active call; or Delete a key entry Also used to REJECT incoming call. Place ACTIVE call on hold Enter to retrieve voice mails or other messages Enable/Disable hands-free speaker mode Press SEND to dial a new number or redial the last number dialed. Press send button to send a call immediately before no key entry timeout value expires Enter to retrieve voice mails or other messages Enter Keypad Configuration MENU mode when phone is in IDLE mode. Use as ENTER key when in Keypad Configuration. Standard phone keypad; press # key to send call; press * key to for IVR functions DO NOT DISTURB key; Press DND to turn Do not disturb function on or off. Toggle between headset and speakerphone mode when in hands free mode Turn intercom function on/off Brings phonebook on screen
0 - 9, *, # DND HEADSET INTERCOM
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MAKING PHONE CALLS

Handset, Speakerphone and Headset Mode
Handset can be toggled between Speaker and Headset. To switch between Handset and Speaker/Headset, press the Hook Flash in the handset cradle or press the SPEAKER button.
Multiple SIP Accounts and Lines
GXP can support up to six independent SIP accounts depending on the product model. Each account is capable of independent SIP server, user and NAT settings. Each of the line buttons is virtually mapped to an individual SIP account. The name of each account is conveniently printed next to its corresponding button. In off-hook state, select an idle line and the name of the account (as configured in the web interface) is displayed on the LCD and a dial tone is heard. For example: Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as VoIP 1, VoIP 2, respectively and ensure that they are active and registered. When LINE1 is pressed, you will hear a dial tone and see VoIP 1 on the LCD display; when LINE2 is pressed, you will hear a dial tone and see VoIP 2 on the LCD display. To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. User can switch lines before dialing any number by pressing the same LINE button one or more times. If you continue to press a LINE button, the selected account will circulate among the registered accounts. For example: when LINE1 is pressed, the LCD displays VoIP 1; If LINE1 is pressed twice, the LCD displays VoIP 2 and the subsequent call will be made through SIP account 2. Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When the virtually mapped line is in use, the GXP will flash the next available LINE (from left to right or from top to bottom for Multi Purpose Keys) in red. A line is ACTIVE when it is in use and the corresponding LED is red.

Speed Dial

The Multi Purpose Key buttons, located on the right-hand-side of the phone, can be configured for speed dial. Press the speed dial button to automatically call the assigned extension.
Grandstream Networks, Inc. GXP User Manual Firmware 1.1.6.46 Page 16 of 39 Last Updated: 03/2008
Note: The multi-functional buttons will function as LINE keys when all LINEs are busy. The LED will flash in red to indicate an incoming call. Press the button to pick up the call. If any one of the Multi Purpose Keys is associated with a call, the buttons speed dial/BLF function will not work.
Making Calls using IP Addresses
Direct IP calling 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 a 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)
To make a direct IP call, please follow these steps: 1. Press MENU button to bring up MAIN MENU. 2. Select Direct IP Call using the arrow-keys. 3. Press OK to select. 4. Input the 12-digit target IP address. (Please see example below). 5. Press OK key to initiate call. To make a quick IP call, please see next section. For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input the following: 192*168*1*60#5062 - The * key represent the dot. ; The # key represent colon :. Press OK to dial out. Quick IP Call Mode The GXP also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phones IP-number. This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended. Setting up the phone to make Quick IP calls To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the Advanced Settings page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x is 0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. aaa.bbb.ccc is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK). For example: 192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or # 192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or # 192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or # 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3 NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IPIP call will also use STUN. Configure the Use Random Port to NO when completing Direct IP calls.

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ANSWERING PHONE CALLS

Receiving Calls
1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER or by pressing the corresponding account LINE button. 2. Incoming multiple calls: When another call comes in while having an active call, the phone will produce a Call Waiting tone (stutter tone). Next available lines will flash red (as described in section 4.3.2). Answer the incoming call by pressing its corresponding LINE button. The current active call will be put on hold. 3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via SPEAKER. (PBX (or Server) must also supports this feature)

Do Not Disturb

1. Press the DND or MUTE button if you do not want to take a call. This will send the caller directly to voicemail. 2. Press the DND or MUTE button to set phone to do not disturb (icon will be on the screen). The phone will not ring and send caller directly to voicemail. (see note above)
PHONE FUNCTIONS DURING A PHONE CALL

Call Waiting/ Call Hold

1. Hold: Place a call on hold by pressing the HOLD button. 2. Resume: Resume call by pressing the corresponding blinking LINE. 3. Multiple Calls: Automatically place ACTIVE call on HOLD by selecting another available LINE to place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.

Mute/Delete

1. Press the MUTE button to enable/disable muting the microphone. 2. The Line Status Indicator will show LINEx: SPEAKING or LINEx: MUTE to indicate whether the microphone is muted. NOTE: Pressing MUTE button for an incoming call will reject the call. MUTE button also functions as delete key when user wishs to delete the last entered digit.

Call Transfer

GXP supports both Blind and Attended (or supervised) transfer: 1. Blind Transfer: Press TRANSFER (or TRNF for GXP-2000) button, then dial the number and press the SEND button to complete transfer of active call.
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The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account. When BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when that line is not in use. Call Pick Up is supported when user presses a flashing BLF key.
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CALL FEATURES

The GXP supports traditional and advanced telephony features including caller ID, caller ID w/name, call forward/transfer/park/hold as well as intercom/paging and BLF. Table 11: GXP Call Features Key
*30 *31 *67 *82 *50 *51 *70 *71 *72

Call Features

Block Caller ID (for all subsequent calls) Send Caller ID (for all subsequent calls) Block Caller ID (per call) Send Caller ID (per call) Disable Call Waiting (for all subsequent calls) Enable Call Waiting (for all subsequent calls) Disable Call Waiting (per Call) Enable Call Waiting (per Call) Unconditional Call Forward Dial *72 for a dial tone. Dial the forwarding number followed by #. Wait for dial tone. LCD will display Call FWD Activated. Cancel Unconditional Call Forward: dial *73 and get the dial tone, then hang up. LCD will display Call FWD Activated. Busy Call Forward Dial *90 for a dial tone. Dial the forwarding number followed by #. Wait for a dial tone. Hang up. Cancel Busy Call Forward: dial *91. Wait for dial tone. Hang up. Delayed Call Forward Dial *92 for a dial tone. Dial the forwarding number followed by #. Wait for a dial tone. Hang up. LCD will display Call FWD Activated. Cancel Delayed Call Forward Dial *93 for a dial tone, then hang up.

*73 *90

*91 *92
CUSTOMIZED LCD SCREEN & XML
Grandstream GXP Series phones support both simple and advanced XML applications: 1) XML Custom Screen, 2) XML Downloadable Phonebook and 3) Advanced XML Survey Application. For more information on how to create a downloadable XML phonebook, creating a custom idle screen and/or reprogramming the soft-keys on GXP1200/GXP-2010/GXP2020, please visit our website at: http://www.grandstream.com/resources.html.
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Configuration Guide

The GXP can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly, through embedded web-configuration menu.

CONFIGURATION VIA KEYPAD

To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and left/right. Press the OK button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button. The phone automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if left idle for 20 seconds. Press the MENU button to enter the key the Key Pad Menu. The menu options available are listed in table 8.

Table 12: Key Pad Configuration Menu Call History Status Phone Book LDAP Directory Instant Messages Direct IP call Preference Displays histories of incoming, dialed and missed calls. Displays the network status, account statuses, software version and MAC-address of the phone. Displays the phonebook Displays the LDAP directory Goes to voice messages Displays the IP-call options menu Press Menu button to enter this sub menu including Do NOT Disturb DND (Do NOT Disturb) function could be turned on or off in the DO NOT Disturb menu. Ring Tone Choose different ring tones in the Ring Tone menu. Ring Volume Press Menu button to hear the selected ring volume, press or to hear and adjust the ring tone volume. LCD Contrast LCD Brightness Download SCR XML The phone will download the custom idle screen (if available) Erase Custom SCR Custom idle screen will be erased and will be replaced with default Grandstream logo. Display Language You can choose English, Chinese or Secondary Language Press Menu button to choose the menu item. Press to return to the main menu.
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Configure

Press Menu button to display the configuration selections: Network. To enable/disable DHCP. To setup IP-address, Net mask and Gateway address SIP To change SIP-server settings for primary account. Upgrade In this menu setting regarding the firmware server and Config server can be changed. It also enables the user to make the phone attempt to download new firmware. Factory Reset Key in the physical/MAC address on back of the phone. Press Menu button to reset FACTORY DEFAULT setting. Do not use Factory Reset unless you want to restore factory settings Layer 2 QoS Configure Vlan Tags Press to return the main menu.

Factory Functions

Press Menu to display the factory function items including Audio Loopback Speak into the handset. If you hear your voice in the handset, your audio works fine. Press Menu button to exit the mode. Diagnostic Mode All LEDs will light up Press any key on the keypad, to display the button name in the LCD. Lift and put back the handset or press Menu button to exit the diagnostic mode. Enable WDT Toggles the status of the Watchdog Timer. Press to return to the main menu. Press Menu button to reboot the device Display Exit Press Menu button to exit the menu Exit from this menu.

Reboot

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FIGURE 3: KEYPAD GUI FLOW
Call History Answered Calls Dialed Calls Missed Calls Back Phone Book New Entry Download Phonebook XML Back LDAP Directory View Directory Download Directory Search Configuration Back Instant Message Do Not Disturb Phone Book Clear All Back Preference Do Not Disturb Ring Tone Ring Volume LCD Contrast LCD Brightness Download SCR XML Erase Custom SCR Display Language Back Config Network SIP Upgrade Factory Reset Layer 2 QoS Back Display Language Exit Factory Function Audio Loopback Diagnostic Mode Enable WDT Back English Chinese Secondary Language Language File Postfix Back Ring Tone Default Ring Ring1 Ring2 Ring 3 Back LCD Brightness Active Idle Back Network IP Setting IP NetMask Gateway DNS Server 1 DNS Server SIP Account SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio Save Upgrade Firmware Server Config Server Upgrade Via Layer 2 QoS 802.1Q/VLAN Tag Priority value Reset Vlan Config Back Enable DND Disable DND Back Search Configuration Select Filter Filter Value Back Name: Number: Acct: Confirm Add: Cancel & Return: Any of previous menus Back Clear All New Entry

Call History

Status

LDAP Directory

Instant Message

Direct IP Call

Preference

Config

Diagnostic Mode Keypad/LED Diagnostic
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CONFIGURATION VIA WEB BROWSER
The GXP 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 Microsofts IE or Mozilla Firefox.
Access the Web Configuration Menu
To access the phones Web Configuration Menu Connect the computer to the same network as the phone1 Make sure the phone is turned on and shows its IP-address Start a Web-browser on your computer Enter the phones IP-address in the address bar of the browser2 Enter the administrators password to access the Web Configuration Menu3
The Web-enabled computer has to be connected to the same sub-network as the phone. This can easily be done by connecting the computer to the same hub or switch as the phone is connected to. In absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone using the PC-port on 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. You will need this number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60, please use http://192.168.0.60 in the address bar your browser. The default administrator password is admin; the default end-user password is 123.
NOTE: When changing any settings, always SUBMIT them by pressing the button on the bottom of the page. Reboot the phone to have the changes take effect. If, after having submitted some changes, more settings have to be changed, press the menu option needed.

Definitions

This section will describe the options in the Web configuration user interface. As mentioned, a used can log in as an administrator or end-user. Functions available for the end-user are: Status: Displays the network status, account statuses, software version and MAC-address of the phone Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings can be set here. Additional functions available to administrators are: Advanced Settings: To set advanced network settings, codec settings and XML configuration settings. Account X: To configure each of the SIP accounts. EXT X: To configure setting on extension module
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Table 13: Device Configuration - Status MAC Address IP Address Product Model Part Number Software Version The device ID, in HEXADECIMAL format. This field shows IP address of GXP This field contains the product model information. This field contains the product part number System Up Time System Time Registered PPPoE Link Up Program: This is the main software (firmware) release number, always used to identify the software (firmware) system of the phone. Boot: Booting code version number

This field shows system up time since the last reboot. This field shows the current time on the phone system. Indicates whether accounts are registered to the related SIP server(s). GXP can support four unique SIP profiles. Indicates whether the PPPoE connection is enabled (connected to a modem).
Table 14: Device Configuration Basic Settings End User Password IP Address This contains the password to access the Web Configuration Menu. This field is case sensitive with a maximum length of 25 characters. There GXP operates in two modes: 1. DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The GXP acquires its IP address from the first DHCP server it discovers on its LAN. The DHCP option is reserved for NAT router mode. To use the PPPoE feature, set the PPPoE account settings. The GXP establishes a PPPoE session if any of the PPPoE fields is set. 2. Static IP mode: configure all of the following fields: IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary). These fields are set to zero by default. These options are used to assign a function to the corresponding multi purpose key. Options available are: 1. Speed Dial. 2. BLF (Busy Lamp Field). This option has to be supported on the PBX and it indicates the status of the extension. The three possible states are idle (green), busy (red), ringing (blinking red). 3. Presence Watcher. This option has to be supported by a presence server and it is tied to the Do not disturb status of the phone. 4. Eventlist BLF. This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message. Each function is connected to one of the accounts and has a target user ID. This parameter controls the date/time display according to the specified time zone.
GXP User Manual Firmware 1.1.6.46 Page 25 of 39 Last Updated: 03/2008

Multi Purpose Key X

Time Zone
LCD Backlight Always On Time Display Format Date Display Format
Turn on LC backlight at all times. Default is No. This option applies to GXP1200/GXP-2000 only. LCD time display in 12 hour or 24 hour format Choose one of the following formats: Year-Month-Day Month-Day-Year Day-Month-Year This option applies to GXP280/GXP-1200/GXP-2000 only.

Default is No. If set to Yes, the client will use DNS SRV to look up server. If the phone has an assigned PSTN telephone number, this field should be set to Yes. Otherwise, set it to No. If Yes is set, a user=phone parameter will be attached to the From header in SIP request This parameter controls sending REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the SIP users registration information will be cleared on reboot. This parameter allows user to specify the time frequency (in minutes) that GXP refreshes its registration with the specified registrar. The default interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days). This parameter defines the local SIP port used to listen and transmit. The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and Account 4 respectively.

Local SIP Port

SIP Registration Failure Retry registration if the process failed. Default is 20 seconds. Retry Wait Time SIP T1 Timeout SIP T2 Interval SIP Transport Use RFC3581 Symmetric Routing NAT Traversal (STUN) RFC 3261 SIP T1 timer. Default is 1 second. RFC 3261 SIP T2 timer. Default is 0.5 seconds. Choose SIP Transport between UDP and TCP. Default is UDP. Default No. When selected the phone will follow the routing procedures specified in RFC3581. This parameter activates the NAT traversal mechanism. If activated (by choosing Yes) and a STUN server is also specified, the phone performs according to the STUN client specification. Using this mode, the embedded STUN client detects if and what type of NAT/Firewall configuration is used. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped public IP address and port in all of its SIP and SDP messages. If the NAT Traversal field is set to Yes with no specified STUN server, the GXP will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the hole on the NAT open. Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically. Enable Presence feature. SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. When configured, user can access messages by pressing MSG button. This ID is usually the VM portal access number. This parameter specifies the mechanism to transmit DTMF digit. There are 3 supported modes: in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
Subscribe for MWI: PUBLISH for Presence Proxy-Require Voice Mail UserID Send DTMF
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Early Dial Dial Plan Prefix Delayed Call Forward Wait Time Enable Call Features Call Log Session Expiration
Default is No. Use only if proxy supports 484 responses. Sets the prefix added to each dialed number.
Time waited before the call is forward to a number or VM. Default is 20 seconds.
Default is No. If set to Yes, Call transfer, Call Forwarding & Do-Not-Disturb are supported locally provided ITSP support those features. User can choose to disable Call Log and what kind of calls to log. The SIP Session Timer extension enables SIP sessions to be periodically refreshed via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session is terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default value is 180 seconds.
Min-SE Caller Request Timer Callee Request Timer Force Timer
Defines the minimum session expiration (in seconds). Default is 90 seconds. If set to Yes, the phone will use session timer when it makes outbound calls if remote party supports session timer. If selecting Yes, the phone will use session timer when it receives inbound calls with session timer request. If set to Yes, the phone will use session timer even if the remote party does not support this feature. If set to No, the session timer is enabled only when the remote party supports this feature. To turn off Session Timer, select No for Caller Request Timer, Callee Request Timer, and Force Timer. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Session Timer can be refreshed using INVITE method or UPDATE method. Select Yes to use INVITE method to refresh the session timer. PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is required to support PSTN internetworking. There are 4 uniquely defined ring tones: One (1) System Ring Tone: when selected, all calls will ring with system ring tone. Three (3) Customer Ring Tones: when selected, incoming calls from designated account will play selected ring tone. Defines how long ring will ring when receiving a call. Default is 60 seconds.
UAC Specify Refresher UAS Specify Refresher Force INVITE Enable 100rel

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Software Upgrade & Customization
Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.
FIRMWARE UPGRADE THROUGH TFTP/HTTP
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. Upgrade Server needs to be set to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples of some valid URLs. firmware.mycompany.com:6688/Grandstream/1.1.6.37 168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web Configuration Interface.

Key Pad Menu

To configure the Upgrade Server via Key Pad Menu options, select Config from the Main Menu, then select Upgrade. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN format. Choose Save and use TFTP or Save and use HTTP to select upgrade method. Select Reboot from the Main Menu to reboot the phone.
Web Configuration Interface
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXP IP address. Enter the admin password to access the web configuration interface. In the ADVANCED SETTINGS page, enter the Upgrade Servers IP address or FQDN in the Firmware Server Path field. Select TFTP or HTTP upgrade method. Update the change by clicking the Update button. Reboot or power cycle the phone to update the new firmware. During this stage, the LCD will display the firmware file downloading process. If a firmware upgrade fails for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the phone will stop the upgrading process and re-boot using the existing firmware/software. Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever possible.

No Local TFTP Server

For users who do not have a local TFTP server, Grandstream provides a NAT-friendly TFTP server on the public Internet for users to download the latest firmware upgrade automatically. Please check the Support/Download section of our website to obtain this TFTP server IP address: http://www.grandstream.com/firmware.html. Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades. A free Windows version TFTP server is available: http://support.solarwinds.net/updates/NewcustomerFree.cfm.
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Instructions for local TFTP Upgrade: 1. Unzip the file and put all of them under the root directory of the TFTP server. 2. The PC running the TFTP server and the GXP should be in the same LAN segment. 3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server, in the phones web configuration page 5. Configure the Firmware Server Path with the IP address of the PC 6. Update the change and reboot the unit User can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server. NOTE: When GXP phone boots up, it will send TFTP or HTTP request to download configuration file cfg000b82xxxxxx, where 000b82xxxxxx is the MAC address of the GXP phone. This file is for provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following error messages in a TFTP or HTTP server log can be ignored: TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does not exist. Configuration File Download

CONFIGURATION FILE DOWNLOAD
The GXP can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. Config Server Path is the TFTP or HTTP server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. 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 4 digit numeric numbers. i.e., P2 is associated with Admin Password in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding configuration template of the firmware. Once the GXP boots up (or re-booted), it will request a configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases.
Managing Firmware and Configuration File Download
When Automatic Upgrade is set to Yes, a Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre-scheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time.
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Restore Factory Default Setting
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone. Please backup or print all the settings before you restoring factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
INSTRUCTIONS FOR RESTORATION: Step 1: Press OK button to bring up the keypad configuration menu, select Config, press OK to enter submenu, select Factory Reset (Please refer to Table 5-1 of keypad flow chart) Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following mapping: 0-9: A: B: C: D: E: F: 0-(press the 2 key twice, A will show on the LCD) 33 (press the 3 key twice, D will show on the LCD) 333 3333

Example: if the MAC address is 000b8200e395, it should be key in as 0002228200333395. NOTE: If there are digits like 22 in the MAC, you need to type 2 then press -> right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2. Step 3: Press the OK button to move the cursor to OK. Press OK button again to confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.
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Grandstream Networks, Inc.
GXP2020/GXP2010/GXP2000/GXP1200/GXP280/GXP285 Enterprise IP Phones

GXP-2020

GXP-2010

GXP-1200

GXP-2000

GXP-280/GXP-285

GXP User Manual Firmware 1.2.5.3
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TABLE OF CONTENTS GXP USER MANUAL WELCOME...... 4 INSTALLATION..... 5 EQUIPMENT PACKAGING..... 5 CONNECTING YOUR PHONE..... 5 GXP-2000 EXTENSION UNIT..... 5 SAFETY COMPLIANCES.... 7 WARRANTY...... 7 PRODUCT OVERVIEW..... 8 USING THE GXP SIP ENTERPRISE PHONE.... 13 GETTING FAMILIAR WITH THE LCD.... 13 MAKING PHONE CALLS..... 17 ANSWERING PHONE CALLS..... 20 PHONE FUNCTIONS DURING A PHONE CALL.... 20 CALL FEATURES..... 23 CUSTOMIZED LCD SCREEN & XML.... 23 CONFIGURATION GUIDE.... 24 CONFIGURATION VIA KEYPAD..... 24 CONFIGURATION VIA WEB BROWSER..... 27 SAVING THE CONFIGURATION CHANGES.... 41 REBOOTING THE PHONE REMOTELY.... 41 SOFTWARE UPGRADE & CUSTOMIZATION... 42 FIRMWARE UPGRADE THROUGH TFTP/HTTP.... 42 CONFIGURATION FILE DOWNLOAD..... 43 RESTORE FACTORY DEFAULT SETTING... 44
TABLE OF FIGURES GXP USER MANUAL Figure 1: Connecting the GXP2000 and the GXPExtension.. 6 Figure 2: GXP2000 Internal Headset Wiring Schema... 7 Table 10: GXP Keypad Buttons.... 16 Figure 3: Keypad GUI Flow... 26 TABLE OF TABLES GXP USER MANUAL Table 1: Equipment Packaging... 5 Table 2: GXP Connectors.... 5 Table 3: GXP Product Models.... 8 Table 4: GXP Comparison Guide.... 9 Table 5: GXP Key Features in a Glance.... 9 Table 6: GXP Hardware Specifications.... 10
Grandstream Networks, Inc. GXP User Manual Firmware 1.2.5.3 Page 2 of 44 Last Updated: 03/2011
Table 7: GXP Technical Specifications.... 11 Table 8: LCD Buttons.... 13 Table 9: LCD Icons.... 14 Table 11: GXP Call Features.... 23 Table 12: Key Pad Configuration Menu... 24 Table 13: Device Configuration - Status.... 28 Table 14: Device Configuration Basic Settings... 28 Table 15: Advanced Settings.... 31 Table 16: SIP Account Settings.... 37
GUI INTERFACE EXAMPLES GXP USER MANUAL (http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip)

1. 2. 3. 4. 5. 6. 7.

SCREENSHOT OF CONFIGURATION LOGIN PAGE SCREENSHOT OF STATUS PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE SCREENSHOT OF SIP ACCOUNT CONFIGURATION SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE
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Welcome

Your Grandstream GXP Series IP phone features a new sophisticated design and is very easy to use. The GXP combines advanced feature functionality with the latest technology to offer excellent audio quality, ease of use, expandability, and broad interoperability with 3rd party SIP platforms. It is ideal for the enterprise customer. The GXP Series supports a broad range of codecs, security protection, PoE (not supported on GXP-280), dual 10/100mbps Ethernet ports and are very easy to manage. Currently, the GXP Series consists of the following six models: GXP-280, GXP-285, GXP-1200, GXP-2000, GXP-2010 and GXP-2020. Each model delivers superior audio quality using either a handset, hands-free speakerphone or headset (except for GXP2000) and supports multi-party conferencing, multi-languages, dual-color LEDs, presence and BLF (on most models). Large easy-to-read backlit graphical displays with multiple XML keys further enhance the user experience (not supported on GXP-280/285). Some models (GXP-2000, GXP2010 and GXP2020 currently) are expandable with one or two expansion module. The series is based on SIP standard and are interoperable with most 3rd party SIP platforms and opensource platforms.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty. Warning: Please do not use a different power adaptor with the GXP as it may cause damage to the products and void the manufacturer warranty.
This document is contains links to Grandstream GUI Interfaces. Please download these examples from http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip for your reference. This document is subject to change without notice. The latest electronic version of this user manual is available for download @: http://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdf 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.

Table 5: GXP Key Features in a Glance Features Open Standards Compatible Benefits SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record and SRV), DHCP (both client and server), PPPoE, TFTP, NTP, Telnet. Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG, AGC. Dual 10/100mbps Ethernet ports, headset jack (RJ22 and/or 2.5mm jack). Traditional voice features including caller ID, call waiting, hold, transfer, forward, block, and off-hook dial, click to dial Multi-line support with dual-color LED (except on GXP-280/285), multi-party conferencing, line extension interface, large back-lit (except on GXP-280/285) graphic LCD, 5 or 3 navigation keys, dedicated buttons for hold, send, speakerphone, headset, transfer, conference (for up to 5 parties depending on model), mute, message, Do-not-disturb, phone book, intercom/paging. Custom downloadable ring-tones, SRTP, multi-language support and XML enabled, adjustable positioning angles, wall mountable, AES encryption.
Superb Audio Quality Network Interfaces Feature Rich Advanced Features

Advanced Functionality

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Table 6: GXP Hardware Specifications
LAN Interface (Ethernet ports) Graphic LCD Display
Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port with auto detection GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

128x32 pixel

Expansion Module Support GXP-280/285

130x64 pixel

240x120 pixel

320x160 pixel

No Headset Jack

GXP-280/285

2.5mm RJ22

2.5mm and RJ22

Call Appearance LED

Dual color (green/red)

GXP-280/285 GXP1200 GXP-2000 GXP-2010 GXP-2020
No Power over Ethernet Universal Switching Power Adaptor Dimension
Built-in auto-sensing: Cisco and IEEE 802.3af standard: phone draws power from Ethernet (except on GXP-280) Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA, UL certified GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020 168mm(l) x 200mm(w) x 89.5mm(h) 210mm(l) x 195mm(w) x 77mm(h) 220mm(l) x 215mm(w) x 57mm(h) 210mm(l) x 250mm(w) x 77mm (h) 251mm(l) x 202mm(w) x 77mm(h)
GXP-1200 0.86kg (1.91lbs) GXP-2000 GXP-2010 1.1kg (2.44lbs) GXP-2020

Weight

GXP-280/285 0.62kg (1.37lbs)

0.82kg (1.81lbs)

1.66kg (3.64lbs)
Temperature Humidity Compliance
F/ 0 40C 10% 90% (non-condensing) FCC / CE / C-Tick
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Table 7: GXP Technical Specifications

Lines Protocol Support

Display Feature Keys

NETWORK STATUS STATUS BAR LINE STATUS INDICATOR
Displays the name of the account that is in use. Select another account by pressing the LINE SELECTOR BUTTONS
SOFTBUTTONS (Excluding GXP-2000)
The soft-buttons are context sensitive and will change depending on the status of the phone. Typical functions assigned to soft-buttons are: NEW CALL Press this button to make a new hand-free call. FORWARD ALL Unconditionally forwards the main phone line to another phone MISSED CALLS This option shows up there were unanswered calls to this phone. The MissedCalls option shows a list of the missed calls CALL RETURN Calls the phone that called/tried to call your phone last. REDIAL Redials the last number END CALL Hangs up the call
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Call Parking: FOR GXP2020/GXP2010 ONLY. Refer to the GXE5024/5028 Online User Manual for more information. CallPark PickUp
When a GXP2020 dials out, the Call Park soft button will display on screen. To park the call, press the Call Park button. When another GXP2020 goes off-hook the Call Pickup soft button will display on screen. To pickup the parked call, press the Call Pickup button.
SPECIAL SOFT BUTTONS (Excluding GXP2000 and Only When Integrated with GXE5024/50 28)
Call Queue: FOR GXP2020/2010 and 1200 only. Refer to the GXE5024/5028 Online User Manual for more information. SignIn Press this button to sign in to the call queue. Agent will be prompted in the LCD display to select the call queue to join. Press menu button on keypad to select ok. Once the agent completely signs in, the agent will be brought back to the main screen. SignOut Press this button to sign out of the call queue. Press menu button on keypad to select ok. This will be displayed once the agent is signed in to the call queue. PUBLIC MODE (Also mentioned on p.31 of this manual): This useful mode complements the Call Queue feature by allowing various user agents to log in/log off, sharing the same phone. When enabled, all other accounts on the phone will not be active. For more information, refer to http://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_ with_gxp.pdf

Tab Backspace LogOut

Press this button to log in the user agent into the call queue. Press this button to jump to toggle between UserName and Password entry fields. Press this button to erase the previously typed digit, letter, or character. Press this button to log out the user agent out of the call queue.

Table 9: LCD Icons

LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon: Solid connected to SIP Server/IP address received Blinking physical connection failed Blank SIP Proxy/Server not registered Phone Status Icon: OFF when the handset is on-hook ON when the handset is off-hook Speaker Phone Status Icon: FLASH when phone rings or a call is pending OFF when the speakerphone is off ON when the speakerphone is on DND Icon: ON when the do not disturb is activated Activate by pressing MUTE/DEL button once

Speed Dial

The Multi Purpose Key buttons, located on the right-hand-side of the phone, can be configured for speed dial. Press the speed dial button to automatically call the assigned extension.
Grandstream Networks, Inc. GXP User Manual Firmware 1.2.5.3 Page 18 of 44 Last Updated: 03/2011
Note: The multi-functional buttons will function as LINE keys when all LINEs are busy. The LED will flash in red to indicate an incoming call. Press the button to pick up the call. If any one of the Multi Purpose Keys is associated with a call, the buttons speed dial/BLF function will not work.
Making Calls using IP Addresses
Direct IP calling 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 a 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)
To make a direct IP call, please follow these steps: 1. Press MENU button to bring up MAIN MENU. 2. Select Direct IP Call using the arrow-keys. 3. Press OK to select. 4. Input the 12-digit target IP address. (Please see example below). 5. Press OK key to initiate call. To make a quick IP call, please see next section. For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input the following: 192*168*1*60#5062 - The * key represent the dot. ; The # key represent colon :. Press OK to dial out. Quick IP Call Mode The GXP also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phones IP-number. This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended. Setting up the phone to make Quick IP calls To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the Advanced Settings page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x is 0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. aaa.bbb.ccc is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK). For example: 192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or # 192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or # 192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or # 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3 NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IPIP call will also use STUN. Configure the Use Random Port to NO when completing Direct IP calls.

5-Way Conferencing

GXP can host conference calls and supports up to 5-way conference calling. 2. Initiate a Conference Call: Establish a connection with two or more parties Press CONF button Choose the desired line to join the conference by pressing the corresponding LINE button. Repeat previous two steps for all other parties that would like to join the conference. This can be done at any time. However, if a new call comes in, the other calls will be placed on hold and the host will have to individually re-join the held lines back into the conference by repeating the previous two steps again. 3. Cancel Conference: Canceling establishing conference call. If after pressing the CONF button, a user decides not to conference anyone, press CONF again or the original LINE button. This will resume two-way conversation. 4. End Conference: Press HOLD to end the conference call and put all parties on hold; To speak with an individual party, select the corresponding blinking LINE.
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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.
Voice Messages (Message Waiting Indicator)
A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Press the MSG button to retrieve the message. An IVR will prompt the user through the process of message retrieval. Press a specific LINE to retrieve messages for a specific line account. NOTE: Each line has a separate voicemail account. Each account requires a voicemail portal number to be configured in the voicemail user id field. To check which line account has a message 1) press the message button (this always checks the primary account), 2) check each line for stutter tone or 3) check missed calls using the menu.

Busy Lamp Field

The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account. When BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when that line is not in use. Call Pick Up is supported when user presses a flashing BLF key.
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CALL FEATURES

The GXP supports traditional and advanced telephony features including caller ID, caller ID w/name, call forward/transfer/park/hold as well as intercom/paging and BLF. Table 11: GXP Call Features Key

No Key Entry Timeout Use # as Dial Key

Local RTP port

Use Random Port
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Keep-alive interval Use NAT IP STUN Server
Firmware Upgrade and Provisioning
This parameter specifies how often the GXP sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open. Default is 20 seconds. NAT IP address used in SIP/SDP message. Default is blank. IP address or Domain name of the STUN server. STUN resolution result will be displayed in the STATUS page of the Web UI. Default method is HTTP. Firmware upgrade may take up to 10 minutes depending on network environment. Do not interrupt the firmware upgrading process. This is the IP address of the configured TFTP server. If selected and it is nonzero or not blank, the GXP will attempt to retrieve a new configuration file or new code image from the specified TFTP server at boot time. It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image will be verified and then saved into the Flash memory. Note: Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device.

Via TFTP Server

Via HTTP Server
The HTTP server URL used for firmware upgrade and configuration via HTTP. For example: http://provisioning.mycompany.com:6688/Grandstream/1.2.5.3 Here :6688 is the specific TCP port that the HTTP server is using; omit if using default port 80. Note: If Auto Upgrade is set to No, GXP will only perform HTTP download once at boot up.

Config Server Path

IP address or domain name of firmware server.
XML Config File Password The XML provisioning system allows Grandstream phones to perform (For configuration updates via XML configuration files. Users can set the XML config GXP280/GXP/285/GXP1200 file password in the web UI of the phone. Only) Firmware File Prefix/Postfix
Default is blank. If configured, GXP will request the firmware file with the prefix/postfix. This setting is useful for ITSPs. End user should keep it blank. Default is blank. End user should keep it blank.

Config File Prefix/Postfix
Allow DHCP Option 43 and Default is Yes. This allows the device to get provisioned automatically. Option 66 to override server

Authenticate Conf File

Default is No. If set to Yes, configuration file would be authenticated before acceptance. End user should use default setting.
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Automatic Upgrade

This function is used by ITSP. End user should NOT touch these parameters. Default is No. Choose Yes to enable automatic HTTP upgrade and provisioning. In Check for upgrade every field, enter the number of minutes to check the HTTP server for firmware upgrade or configuration changes. When set to No, the phone will only perform HTTP upgrade and configuration check once at boot up.
LDAP Directory Phonebook XML
IP address or domain name of LDAP script server Enable the XML phonebook via TFTP or HTTP. Define XML server path and download interval. When the user downloads the XML phone the manually entered or edited entries will not be deleted unless this option is selected to Yes. Enable XML Idle Screen download via TFTP or HTTP. Select whether to Use Custom Filename or not, and define the XML server path. Enter server path for XML application. This option applies to GXP-2020 and GXP-2010 only. To configure a User ID/extension to dial automatically when the phone is taken offhook. This parameter sets the payload type for DTMF using RFC2833. Default is 101. It determines the time handset has to be down to be recognized its onhook. Default is 800ms. For GXP280/285 only. The IP address or URL of System log server. This feature is especially useful for ITSPs. Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or 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) The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error message For example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000]. Ethernet link is up.

GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires its own configuration page. Their configurations are identical. Table 16: SIP Account Settings Account Active This field indicates whether the account is active. The default value for the primary account (Account 1) is Yes. The default value for the other two accounts is No. The name associated with each account - displayed on LCD. SIP Servers IP address or Domain name provided by VoIP service provider. IP address or Domain name of Outbound Proxy, Media Gateway, or Session Border Controller. Used for firewall or NAT penetration in different network environment. If the system detects symmetric NAT, STUN will not work. ONLY outbound proxy can provide solution for symmetric NAT. User account information provided by VoIP service provider (ITSP); either an actual phone number or formatted like one. SIP service subscribers Authenticate ID used for authentication. It can be identical to or different from SIP User ID.
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Account Name SIP Server Outbound Proxy
SIP User ID Authenticate ID
Authenticate Password Name Use DNS SRV: User ID is Phone Number SIP Registration Un-register on Reboot SIP Instance ID
SIP service subscribers account password for GXP to register to (SIP) servers of ITSP. SIP service subscribers name that is used for Caller ID display. Default is No. If set to Yes, the client will use DNS SRV to look up server. If the phone has an assigned PSTN telephone number, this field should be set to Yes. Otherwise, set it to No. If Yes is set, a user=phone parameter will be attached to the From header in SIP request This parameter controls sending REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the SIP users registration information will be cleared from the server when the phone reboots. Default is set No. If set to Yes it will be enabled and will add reg-ID and Instance ID on contact header in the REGISTER messages. This feature is mainly provided for servers that don't support SIP Instance ID feature, but will still allow phones to register. This parameter allows user to specify the time frequency (in minutes) that GXP refreshes its registration with the specified registrar. The default interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days). This parameter defines the local SIP port used to listen and transmit. The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and Account 4 respectively.

Time waited before the call is forward to a number or VM. Default is 20 seconds.
PUBLISH for Presence Proxy-Require Voice Mail UserID Send DTMF
Early Dial Dial Plan Prefix BLF Call-pickup Prefix Delayed Call Forward Wait Time Enable Call Features
Default is Yes. If set to No, Call transfer, Call Forwarding & Do-Not-Disturb are supported locally provided ITSP support those features. In addition, ForwardAll softkey will be hidden if call feature code is disabled for Account 1. User can choose to disable Call Log and what kind of calls to log. The SIP Session Timer extension enables SIP sessions to be periodically refreshed via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session is terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default value is 180 seconds.
Call Log Session Expiration
Min-SE Caller Request Timer Callee Request Timer Force Timer
Defines the minimum session expiration (in seconds). Default is 90 seconds. If set to Yes, the phone will use session timer when it makes outbound calls if remote party supports session timer. If selecting Yes, the phone will use session timer when it receives inbound calls with session timer request. If set to Yes, the phone will use session timer even if the remote party does not support this feature. If set to No, the session timer is enabled only when the remote party supports this feature. To turn off Session Timer, select No for Caller Request Timer, Callee Request Timer, and Force Timer. As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher.

UAC Specify Refresher

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UAS Specify Refresher Force INVITE Enable 100rel
As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher. Session Timer can be refreshed using INVITE method or UPDATE method. Select Yes to use INVITE method to refresh the session timer. PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is required to support PSTN internetworking. There are 4 uniquely defined ring tones: One (1) System Ring Tone: when selected, all calls will ring with system ring tone. Three (3) Customer Ring Tones: when selected, incoming calls from designated account will play selected ring tone. Defines how long ring will ring when receiving a call. Default is 60 seconds. If this parameter is set to Yes, the From header in outgoing INVITE message will be set to anonymous, essentially blocking the Caller ID from displaying. Whether to use <sip:anonymous@anonymous.invalid> in the From Header or PAsserted-Identity header. Default is NO. If set to YES, anonymous call will be rejected Default is No. If set to Yes, GXP will automatically switch on speaker to answer the incoming call. Set to Intercom/Paging mode, it will answer the call based on the SIP info header from the server. If the Call-Info header contains answer-after=0, the call be answered automatically (so called paging mode). When BYE is received, the phone will turn off its speaker automatically. Check the SIP User ID in Request URI. If they dont match, the call will be rejected.

Default is NO. If set to YES, then for Attended Transfer, the Refer-To header uses the transferred targets Contact header information.

Account Ring Tone

Ring Timeout Send Anonymous Anonymous Method Anonymous Call Rejection Auto Answer
Allow Auto Answer by Call-Info Turn off speaker on remote disconnect Check SIP User ID for incoming INVITE Refer-To Use Target Contact Disable Multiple Media Attribute in SDP Preferred Vocoder
Default is No. GXP supports up to 7 different Vocoder types including G.711(a/) (also known as PCMU/PCMA), GSM, G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band). Configure Vocoders in a preference list that is included with the same preference order in SDP message. Enter the first Vocoder in this list by choosing the appropriate option in Choice 1. Similarly, enter the last Vocoder in this list by choosing the appropriate option in Choice 8.

SRTP Mode

Enable SRTP mode based on selection. Default is No.
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eventlist BLF URI

If a server supports this feature, user needs to configure an "eventlist BLF" URI on the service side (i.e.: BLF1006@myserver.com) On the GXP, under Account page, fill in the ""eventlist BLF" field with the URI without the domain. (i.e.: BLF1006). Under Basic Settings, please select "eventlist BLF", choose account number, monitored number, etc. Default is Standard. Choose the selection to meet special requirements from Soft Switch vendors.

Special Feature

SAVING THE CONFIGURATION CHANGES
After the user makes a change to the configuration, press the Update button in the Configuration Menu. The web browser will then display a message window to confirm saved changes. Grandstream recommends reboot or power cycle the IP phone after saving changes.
REBOOTING THE PHONE REMOTELY
Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely. The web browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in again.
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Software Upgrade & Customization
Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.

FIRMWARE UPGRADE THROUGH TFTP/HTTP
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. Upgrade Server needs to be set to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples of some valid URLs. firmware.mycompany.com:6688/Grandstream/1.2.5.3 72.172.83.110
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web Configuration Interface.

Key Pad Menu

To configure the Upgrade Server via Key Pad Menu options, select Config from the Main Menu, then select Upgrade. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN format. Choose Save and use TFTP or Save and use HTTP to select upgrade method. Select Reboot from the Main Menu to reboot the phone.
Web Configuration Interface
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXP IP address. Enter the admin password to access the web configuration interface. In the ADVANCED SETTINGS page, enter the Upgrade Servers IP address or FQDN in the Firmware Server Path field. Select TFTP or HTTP upgrade method. Update the change by clicking the Update button. Reboot or power cycle the phone to update the new firmware. During this stage, the LCD will display the firmware file downloading process. If a firmware upgrade fails for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the phone will stop the upgrading process and re-boot using the existing firmware/software. Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever possible.
No Local TFTP/HTTP Server
For users who do not have a local TFTP/HTTP server, Grandstream provides a HTTP server on the public Internet for users to download the latest firmware upgrade automatically. Please check the Support/Download section of our website to obtain this HTTP server IP address: http://www.grandstream.com/firmware.html. Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades. A http://support.solarwinds.net/updates/Newfree Windows version TFTP server is available: customerFree.cfm.
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Instructions for local TFTP Upgrade: 1. Unzip the file and put all of them under the root directory of the TFTP server. 2. The PC running the TFTP server and the GXP should be in the same LAN segment. 3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server, in the phones web configuration page 5. Configure the Firmware Server Path with the IP address of the PC 6. Update the change and reboot the unit User can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server. NOTE: When GXP phone boots up, it will send TFTP or HTTP request to download configuration file cfg000b82xxxxxx, where 000b82xxxxxx is the MAC address of the GXP phone. This file is for provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following error messages in a TFTP or HTTP server log can be ignored: TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does not exist. Configuration File Download

CONFIGURATION FILE DOWNLOAD
The GXP can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. Config Server Path is the TFTP or HTTP server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. 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 4 digit numeric numbers. i.e., P2 is associated with Admin Password in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding configuration template of the firmware. Once the GXP boots up (or re-booted), it will request a configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases. NOTE : Since firmware 1.2.4.3, GXP280/GXP285/GXP1200 can be provisioned using XML configuration file. Please refer to our XML provisioning guide using this link : http://www.grandstream.com/support/gxp_series/general/documents/GS-XML_Provisioning_Guide.pdf
Managing Firmware and Configuration File Download
When Automatic Upgrade is set to Yes, a Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at prescheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time.
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Restore Factory Default Setting
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone. Please backup or print all the settings before you restoring factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
INSTRUCTIONS FOR RESTORATION: Step 1: Press OK button to bring up the keypad configuration menu, select Config, press OK to enter submenu, select Factory Reset (Please refer to Table 5-1 of keypad flow chart) Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following mapping: 0-9: A: B: C: D: E: F: 0-(press the 2 key twice, A will show on the LCD) 33 (press the 3 key twice, D will show on the LCD) 333 3333
Example: if the MAC address is 000b8200e395, it should be key in as 0002228200333395. NOTE: If there are digits like 22 in the MAC, you need to type 2 then press -> right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2. Step 3: Press the OK button to move the cursor to OK. Press OK button again to confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.

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