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.

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
Multiple direct lines with independent SIP accounts, programmable speed dial keys, XML programmable soft-keys (non programmable on GXP-280/285, GXP1200, GXP2000). Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols Supports multiple SIP accounts Supports SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856, 3863) for use of 7 MFKs, SIP Dialog package (RFC 4235) Supports SIP MESSAGE method (RFC 3428) Stores up to 100 incoming IM messages Back-lit graphic LCD display. (GXP-280/285 display is not back-lit) HOLD SPEAKERPHONE SEND TRANSFER CONF MUTE DND HEADSET INTERCOM PHONEBOOK MSG MENU NAVIGATION (4) GXP-280/285 Yes Yes Yes Yes Yes Yes Yes Yes No No Yes Yes Yes (3) GXP-1200 Yes Yes Yes Yes Yes Yes Yes Yes No No Yes Yes Yes GXP-2000 Yes Yes Yes Yes Yes Yes Yes Yes No No Yes Yes Yes GXP-2010 Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes GXP-2020 Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes

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
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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. Press HEADSET key to answer/hang up phone calls while using headset. It also allows user to toggle between headset and speaker. Not available on GXP2000. 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
The GXP series phones allow you make phone calls via handset, speakerphone, or headset mode. During the active calls the user can switch between the handset and the speaker by pressing the speaker key. For headsets to operate, the user must plug the headset to an RJ22 or 2.5mm port on the phone, which allows the user to pick-up, speak, or hang-up calls.
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.

Completing Calls

There are six ways to complete a call: 1. DIAL: To make a phone call. Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates speakerphone) or press the NEW CALL soft-key. The line will have a dial tone and the primary line (LINE1) LED is red. If you wish, select another LINE key (alternative SIP account). Enter the phone number Press the SEND key or press the DIAL soft-key. 2. REDIAL: To redial the last dialed phone number. When redialing, the phone will use the same SIP account as was used for the last call. Thus, when the third SIP account was used for the last call/call attempt, the phone will use the third account to redial. Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates speakerphone), the corresponding LED will be red. Press the SEND button or press the REDIAL soft-key.

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)

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.

GXP User Manual Firmware 1.2.5.3 Page 24 of 44 Last Updated: 03/2011
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. Multi Purpose Key (On GXP2000/2010/2020 only) To configure multi-purpose keys. 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 Multi-Purpose Keys Factory Reset Layer 2 QoS 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 Net Mask Gateway DNSServer1 DNSServer2 Back 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

Factory Functions
Diagnostic Mode Keypad/LED Diagnostic

Time Zone LCD Backlight Always On Time Display Format Date Display Format
Display Clock instead of Choose to display clock or date on LCD. This option applies to GXP-280/GXP285/GXP-1200/GXP-2000 only. Date
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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 Disable Missed Call Backlight
Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means the brightest. For GXP2010 and GXP2020 only. Set LCD contrast. Range from 0 to 20. Not for GXP280/285 Default is No. This field is used to hide the keypad input during a call. Default is No. When its enabled, speaker wont ring on an incoming call. Default is No. By default, LCD backlight will lit whenever there is a missed call. Not for GXP280/285.

Layer 3 QoS Layer 2 QoS Data VLAN Tag
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 0. Enabling the Data VLAN filtering will help reduce the load on the phone, but it isnt necessary in most environments. This is primarily for VLAN filtering where tagged traffic will be forwarded to the DSP. 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 starting 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.
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.

Register Expiration

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.

SUBSCRIBE for MWI:

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SUBSCRIBE for Registration Event
Default is No. This is mainly used for IMS purposes. When enabled, the terminals should store the Service-Route header values after successfully registered, and thereafter add a route header with the values stored in the Service-Route when initiating a request excluding REGISTER. 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. Default is No. Use only if proxy supports 484 responses. Sets the prefix added to each dialed number. Default is **. This prefix is prepended when answering call with BLF key.

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.

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