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... 12 GETTING FAMILIAR WITH THE LCD.... 12 MAKING PHONE CALLS....15 ANSWERING PHONE CALLS.... 18 PHONE FUNCTIONS DURING A PHONE CALL.... 18 CALL FEATURES..... 20 CUSTOMIZED LCD SCREEN & XML.... 20 CONFIGURATION GUIDE..... 21 CONFIGURATION VIA KEYPAD..... 21 CONFIGURATION VIA WEB BROWSER.... 24 SAVING THE CONFIGURATION CHANGES.... 35 REBOOTING THE PHONE REMOTELY... 36 SOFTWARE UPGRADE & CUSTOMIZATION... 37 FIRMWARE UPGRADE THROUGH TFTP/HTTP... 37 CONFIGURATION FILE DOWNLOAD.... 38 RESTORE FACTORY DEFAULT SETTING.... 39
TABLE OF FIGURES GXP USER MANUAL Figure 1: Connecting the GXP2000 and the GXPExtension.. 6 Figure 2: GXP2000 Internal Headset Wiring Schema... 7 Figure 3: Key Pad GUI Call Flow... 44 TABLE OF TABLES GXP USER MANUAL Table 1: Table 2: Table 3: Table 4: Table 5: Table 6: Table 7: Equipment Packaging... 5 GXP Connectors.... 5 GXP Product Models.... 8 GXP Comparison Guide.... 9 GXP Key Features in a Glance.... 9 GXP Hardware Specifications... 9 GXP Technical Specifications... 9
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Grandstream Networks, Inc.
Table 8: LCD Buttons.... 12 Table 9: LCD Icons.... 12 Table 10: GXP Keypad Buttons.... 14 Table 11: GXP Call Features.... 21 Table 12: Key Pad Configuration Menu... 42 Table 13: Device Configuration - Status.... 46 Table 14: Device Configuration Basic Settings... 46 Table 15: Advanced Settings.... 48 Table 16: SIP Account Settings.... 53
GUI INTERFACE EXAMPLES GXP USER MANUAL (http://www.grandstream.com/user_manuals/GUI/GUI_GXP.rar) 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
GXP User Manual Firmware 1.1.6.46
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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 (except on GXP-280), dual 10/100mbps Ethernet ports and are very easy to manage. Currently, the GXP Series consists of the following five models: GXP-280, GXP-1200, GXP-2000, GXP-2010 and GXP-2020. Each model delivers superior audio quality using either a handset, hands-free speakerphone or headset and supports multiparty conferencing, multi-languages, dual-color LEDs, presence and BLF (on most models). Large easyto-read backlit graphical displays (except GXP-280) with multiple XML keys further enhance the user experience. 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 http://www.grandstream.com/user_manuals/GUI/GUI_GXP.rar 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/user_manuals/GXP_User_Manual.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.
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Installation

EQUIPMENT PACKAGING
Table 1: Equipment Packaging Main Case Handset Phone Cord Power Adaptor Ethernet Cable High Phone Stand Low Phone Stand Wall Mount Spacers (2) GXP-280 Yes Yes Yes Yes Yes No Yes No GXP-1200 Yes Yes Yes Yes Yes Yes No Yes GXP-2000 Yes Yes Yes Yes Yes No No No GXP-2010 Yes Yes Yes Yes Yes Yes Yes Yes GXP-2020 Yes Yes Yes Yes Yes Yes Yes Yes

CONNECTING YOUR PHONE

The connectors of the GXP1200/2010/2020 are located on the bottom of the device while they are located on the back side of the GXP280/2000.
Table 2: GXP Connectors Connects the GXP Extension unit directly to the GXP using connection cable. Draws power from PoE if provided by network. 10/100Mbps RJ-45 ports for PC (downlink) connection. 10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af). Draws power from either spare line or signal line. 5V DC power port; UL Certified RJ22 and 2.5mm for GXP-280/2010/2020 RJ22 for GXP-1200 2.5mm for GXP-2000 HW Rev1.0 or later RJ11

EXT PC LAN Power Jack Headset Jack Handset Jack

GXP-2000 EXTENSION UNIT

GXP2000 supports two (2) extension units, providing up to 112 additional programmable extensions. Each GXP Extension unit has 56 multipurpose keys, dual color LEDs (red/green) and support BLF (Busy Lamp Field) and Presence. GXP2000 Extension package contains:
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1) 2) 3) 4)

One GXP Extension unit One PS2 cable One connection plate One Universal Power Adaptor
FIGURE 1: CONNECTING THE GXP2000 AND THE GXPEXTENSION GXP2000 w/GXPExtension GXP Extension
Connecting the GXP2000 w/GXPExtension
Reverse side of connection w/connection plate
Connect the first GXP EXT to the GXP2000 using the PS2 cable found in the GXP Extension package. The first GXPExt draws power directly from the phone. Connect the second GXP Extension unit using the connection plate and the PS2 cable. The GXP2000 will automatically reboot and power up the GXP Extensions. Grandstream recommends, though not required, to use a separate power supply with the second GXP Ext. NOTE: should your system loose power, please unplug your devices and power up the GXP2000 first. Powering up the system: 1. 2. 3. 4. 5. The GXP2000 will boot up first; The GXP LEDs will be solid red; The status light in the top right corner of the GXPExt will blink red; All of the LED indicators on the GXPExt will flash three times; The status light at the top right corner of the GXPExt will turn to solid green.
Note: Extension for GXP2010 and GXP2020 does not support hot-swap. Once connected, user should reboot the phone to ensure the set up will work correctly.
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Figure 2: GXP2000 Internal Headset Wiring Schema
NOTE: For GXP-2000 HW REV. 0.3 and 0.4, a 3.5mm to 2.5mm plug converter is required to use a 2.5mm headset. The converter can be purchased at any electronics store.

SAFETY COMPLIANCES

The GXP phone complies with FCC/CE and various safety standards. The GXP power adaptor is compliant with the UL standard. Only use the universal power adaptor provided with the GXP package. The manufacturers warranty does not cover damages to the phone caused by unsupported power adaptors.

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.
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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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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
*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

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.

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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.
Display Clock instead of Choose to display clock or date on LCD. This option applies to GXP-280/GXP1200/GXP-2000 only. Date Daylight Savings Time
This parameter controls time displayed in daylight savings time. If set to Yes, then the displayed time will be 1 hour ahead of normal time. The Optional Rule is configured to automatically adjust the Daylight Savings Time (DST) based on the rule set in this field. Rule Syntax: start-time; end-time; saving Both start-time and end-time have the same syntax: month,day,weekday,hour,minute o month: 1,2,3,.,12 (for Jan, Feb,., Dec) o day: [+|-]1,2,3,.,31 o weekday: 1, 2, 3,., 7 (for Mon, Tue,., Sun), or 0 which means the daylight saving rule is not based on week days but based on the day of the month. o hour: hour (0-23), minute: minute (0-59) If weekday is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the day value must not be negative. If weekday is not zero and day is positive, then the daylight saving starts on the first day the iteration of the weekday (e.g.: 1st Sunday, 3rd Tuesday etc). If weekday is not zero and day is negative, then the daylight saving starts on the last day the iteration of the weekday (e.g.: last Sunday, 3rd last Tuesday etc). The saving is in the unit of minutes. The saving time may also be preceded by a negative (-) sign if subtraction is desired instead of addition. The default value is set for US, the Automatic Daylight Saving Time Rule shall be set to 3,2,7,2,0;11,1,7,2,0;60 Examples US/Canada where daylight saving time is applicable: 03,02,7,02,00;11,1,7,02,00;60 This means the daylight saving time starts from the second Sunday of March at 2AM and ends the first Sunday of November at 2AM. The saving is 60 minutes.
LCD Backlight Brightness LCD Contrast Disable in-call DTMF display Mute Speaker Ringer in Headset Mode

Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means the brightest. For GXP2010 and GXP2020. Set LCD contrast. Range from 0 to 20. Not for GXP280 Default is No. This field is used to hide the keypad input during a call. Default is No. This field lets user to choose whether to ring the phone Speaker when headset is connected.
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Disable Missed Call Backlight HEADSET Key Mode
Default is No. By default, LCD backlight will lit whenever there is a missed call. Not for GXP280. Set Default mode or choose Toggle Headset/Speaker. Not for GXP2000
Advanced User configuration includes not only the end user configuration, but also advanced configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.
Table 15: Advanced Settings Admin Password G723 rate Administrator password. Only the administrator can access the Advanced Settings and Account Settings page. Password field is purposely blank for security reasons after clicking update and saved. The maximum password length is 25 characters. Encoding rate for G723 codec. By default, 6.3kbps rate is set.
iLBC frame size iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required. iLBC payload type Silence Suppression Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. This controls the silence suppression/VAD feature of the audio codec G.723 and G.729. If set to Yes, when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to No, this feature is disabled.
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Voice Frames per TX

This field contains the number of voice frames to be transmitted in a single Ethernet packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500 byte (or 120kbps)). When setting this value, be aware of the requested packet time (ptime, used in SDP message) is a result of configuring this parameter. This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. E.g., if the first codec is configured as G.723 and the Voice Frames per TX is set to 2, then the ptime value in the SDP message of an INVITE request will be 60ms because each G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or G.711 or G.726, then the ptime value in the SDP message of an INVITE request will be 20ms. If the configured voice frames per TX exceeds the maximum allowed value, the IP phone will use and save the maximum allowed value for the corresponding first codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames respectively. Please be careful when editing these parameters. Adjusting these parameters will also change the dynamic jitter buffer. The GXP has a patent dynamic jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms. Grandstream recommends using the default settings provided. Grandstream does not recommend adjusting these parameters if you are an average user. Incorrect settings will affect the voice quality. Please refer to the Codec FAQ at http://www.grandstream.com/pdf/FAQ-Codec.pdf for more technical detail.

Layer 3 QoS Layer 2 QoS No Key Entry Timeout Use # as Dial Key
This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or Diff-Serv or MPLS. Default value is 48. This contains the value used for layer 2 VLAN tag. Default setting is blank. Default is 4 seconds. This parameter allows users to configure the # key as the Send (or Dial) key. If set to Yes, the # key will immediately send the call. In this case, this key is essentially equivalent to the (Re)Dial key. If set to No, the # key is included as part of the dial string. This parameter defines the local RTP-RTCP port pair used to listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004. This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GXPs are behind the same NAT. Default is No. 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.
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Local RTP port

Use Random Port Keep-alive interval Use NAT IP

STUN Server

Firmware Upgrade and Provisioning
IP address or Domain name of the STUN server. STUN resolution result will display 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.
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero 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 HTTP Server The HTTP server URL used for firmware upgrade and configuration via HTTP. For example: http://provisioning.mycompany.com:6688/Grandstream/1.1.6.44. 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 Firmware File Prefix/Postfix
IP address or domain name of firmware server. 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. Default is Yes. This allows device gets provisioned automatically.
Config File Prefix/Postfix Allow DHCP Option 66 to override server
Authenticate Conf File Automatic Upgrade
Default is No. If set to Yes, configuration file would be authenticated before acceptance. End user should use default setting. 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.
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Idle Screen XML Enable XML Idle Screen download via TFTP or HTTP. Define XML server path. Download XML Application Offhook Auto Dial DTMF Payload Type Onhook Threshhold Syslog Server Syslog Level Enter server path for XML application. This option applies to GXP-2020 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 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. NTP server This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve. It is used to display the current date/time.

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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 Register Expiration
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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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.
Grandstream Networks, Inc. GXP User Manual Firmware 1.1.6.46 Page 35 of 39 Last Updated: 03/2008
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/1.1.6.37 168.75.215.189
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.

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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