CONNECTING YOUR PHONE ........................................................................................................................................ 5
GXP-2000EXTENSION UNIT ....................................................................................................................................... 5
USING THE GXP SIP ENTERPRISE PHONE ..................................................................................................... 13
GETTING FAMILIAR WITH THE LCD .......................................................................................................................... 13
MAKING PHONE CALLS ............................................................................................................................................. 17
PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 20
CALL FEATURES ........................................................................................................................................................ 23
CONFIGURATION VIA KEYPAD .................................................................................................................................. 24
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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 3
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, GX P-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 exam ples
http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip for your
from
reference.
•This document is subject to change without notic e. The latest electronic version of this user manual
•Reproduction or transmittal of the enti re 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.
rd
party SIP platforms. It is ideal for
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GXP-280/285
GXP-1200
GXP-2000
GXP-2010
GXP-2020
Main Case
Yes
Yes
Yes
Yes
Yes
Handset
Yes
Yes
Yes
Yes
Yes
Phone Cord
Yes
Yes
Yes
Yes
Yes
Power Adaptor
Yes
Yes
Yes
Yes
Yes
Ethernet Cable
Yes
Yes
Yes
Yes
Yes
High Phone Stand
No
Yes
No
Yes
Yes
Low Phone Stand
Yes
No
No
Yes
Yes
Wall Mount Spacers (2)
No
Yes
No
Yes
Yes
EXT
Connects the GXP Extension unit directly to the GXP using connection cable.
Draws power from PoE if provided by network.
PC
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.
Power Jack
5V DC power port; UL Certified
RJ22 and 2.5mm for GXP-280/285/2010/2020
2.5mm for GXP-2000 HW Rev1.0 or later
Handset Jack
RJ11
Installation
EQUIPMENT PACKAGING
Table 1: Equipment Packaging
CONNECTING YOUR PHONE
The connectors of the GXP1200/2010/2020 are locat ed on the bottom of the device while they are located on
the back side of the GXP280/285/2000.
Table 2: GXP Connectors
LAN
Headset Jack
RJ22 for GXP-1200
GXP-2000EXTENSION UNIT
GXP–2000 supports two (2) extension units, providing up to 112 additi onal programmable extensions. Each
GXP Extension unit has 56 multi–purpose keys, dual color LEDs (red/green) and support BLF (Busy Lamp
Field) and Presence.
GXP–2000 Extension package contains:
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1) One GXP Extension unit
GXP–2000 w/GXP–Extension
GXP Extension
2) One PS2 ca ble
3) One conn ect ion plate
4) One Univ ersal Power Adaptor
FIGURE 1:CONNECTING THE GXP–2000 AND THE GXP–EXTENSION
Connecting the GXP–2000
w/GXP–Extension
Reverse side of connection
w/connection plate
Connect the first GXP –EXT to the GXP–2000 using the PS2 cable found in the GXP Extension package.
The first GXP–Ext 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 lose power, please unplug your devices and power up the GXP–2000 first.
Powering up the system:
1. The GXP–2000 will boot up first;
2. The GXP LEDs will be solid red;
3. The status light in the top right corner of the GXP–Ext will blink red;
4. All of the LED indicators on the GXP–Ext will flash three times;
5. The status light at the top right corner of the GXP–E xt will turn to solid green.
NOTE: 1. 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.
2. GXP2010/GXP2020 can drive 2 extension modules. I ndependent power adapters are not needed
for extension modules.
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Figure 2: GXP–2000 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
manufacturer’s warranty does not cover dam ages to the phone caused by unsupported power adaptors.
WARRANTY
If you purchased your GXP from a reseller, please contact the company where you purchased your phone
for replacement, repair or refund. If you purchased the product directly from Grandstream, contact your
Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number before
you return the product. Grandstream reserves t he right to remedy warranty policy without prior notification.
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Product Overview
GXP280 is an entry-level SIP phone. It features:
GXP285 is an entry-level SIP phone. It features:
GXP1200 is an entry-level SIP phone. It features:
GXP2000 is a mainstream SIP phone. It feature s:
GXP2010 is a key system SIP phone. It features:
Table 3: GXP Product Models
Model Picture Overview
Single line
Three soft keys
GXP-280
GXP-285
Single line
Three soft keys
PoE Supported
GXP-1200
GXP-2000
GXP-2010
Two lines
Three soft keys
PoE Supported
Four lines
Seven programmable hard keys
PoE Supported
Four lines
Eighteen programmable hard keys
Three XML programmable soft keys
PoE Supported
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GXP2020 is an executive SIP phone. It features:
, up to 2
Modules, 56
, up to 2
Modules, 56
Silence suppression,
Traditional voice features including caller ID, call waiting, hold,
navigation keys,
ble positioning angles, wall mountable, AES
Six lines
Seven programmable hard keys
Four XML programmable soft keys
GXP-2020
PoE Supported
Table 4: GXP Comparison Guide
Features GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020
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.
Superb Audio Quality
Advanced Digital Signal Processing (DSP),
VAD, CNG, AGC.
Network Interfaces
Dual 10/100mbps Ethernet ports, headset jack (RJ22 and/or 2.5mm
jack).
Feature Rich
transfer, forward, block, and off-hook dial, click to dial
Advanced Features
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
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.
Advanced Functionality
Custom downloadable ring-tones, SRTP, multi-language support and
XML enabled, adjusta
encryption.
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Table 6: GXP Hardware Specifications
LAN Interface (Ethernet
ports)
Graphic LCD Display
Expansion Module Support
Headset Jack
Call Appearance LED
Power over Ethernet
Universal Switching
Power Adaptor
Dimension
Weight
Temperature
Humidity
Compliance
Two (2) 10/100 Mbps Full/H alf Duplex Ethernet Switch with LAN and PC port with
auto detection
GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020
128x32
pixel
GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020
No NoYesYesYes
GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020
2.5mm and
RJ22
128x32 pixel130x64
pixel
240x120
pixel
RJ222.5mm2.5mm and
RJ22
320x160
pixel
2.5mm and
RJ22
Dual color (green/red)
GXP-280/285 GXP-
1200
GXP-2000 GXP-2010 GXP-2020
No 3112213
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 168mm(l) x 200mm(w) x 89.5mm(h)
GXP-1200 210mm(l) x 195mm(w) x 77mm(h)
GXP-2000 220mm(l) x 215mm(w) x 57mm(h)
GXP-2010 210mm(l) x 250mm(w) x 77mm (h)
GXP-2020 251mm(l) x 202mm(w) x 77mm(h)
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Table 7: GXP Technical Specifications
friendly remote software upgrade (via TFTP/HTTP) for deployed devices including
tones, call hold, call transfer
Lines
Multiple direct lines with independent SIP accounts, program mable speed dial keys,
XML programmable soft-keys (non programmable on GXP-280/285, GXP1200, GXP2000).
Protocol
Support
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP,
ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols
Dynamic negotiation of codec and voice payload length
Support for G.723,1 (5.3/6.3K), G.729A/B, G.711 a/µ-law, G.726-32, G.722 (wide-band),
GSM and iLBC codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation), AGC
(automatic gain control)
Acoustic Echo Cancellation (AEC) with Automati c Gain Control (AGC) for speakerphone
mode, Support side tone
Adaptive jitter buffer control and packet delay & loss concealment
Telephony
Features
Intuitive graphic user interface (GUI), downloadable phone book (XML, LDAP), support for
anonymous call using privacy header, MLS (multi language support )
Voice mail indicator, downloadable custom ring-
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(attended/blind), call forward, call waiting, caller ID, mute, redial, call log, caller ID display or
block, Do-Not-Disturb (DND) and volume control
Provisioning
Features
Security
Multi-party conferencing (up to 5), dial plan prefix, off-hook auto dial, auto answer, early dial
and speed dial (on some models)
Network and
Via keypad/LCD, Web browser, or secure (AES encrypted) central configuration file, manual
or dynamic host configuration protocol (DHCP ) network setup
Support NAT traversal using IETF STUN and Sy m m etric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS
Firmware
Upgrades
Support firmware upgrade via TFTP or HTTP,
Support for Authenticating configuration file before accepting changes
User specific URL for configuration file and firmware files
Advanced
Server
Message waiting indication, support DNS SRV Look up and SIP Server Fail Over, Support
customizable idle screen via downloading XML by HT T P /TFTP
DIGEST authentication and encryption using MD5 and MD5-sess, SRTP
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Using the GXP SIP Enterprise Pho ne
Key Button
Key Button Definitions
LINE
SELECTORS
Selects the phone line printed on it s right-hand side.
LINES
selector on the left-hand side.
DATE AND
TIME
logo, please check page 24.
Shows the status of the phone and network. It will indicate whether the ne twork is down, starting or
CALLS” are shown here too.
BAR
LINE
INDICATOR
The soft-buttons are context sensitive and will change depending on the st atus of the 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
SIP PHONE
LOGO
NETWORK
STATUS
STATUS
STATUS
SOFTBUTTONS
(Excluding
GXP-2000)
Displays the available phone lines. Choose a phone line by pressing the co rresponding line
Displays the current date and time. Can be synchronized with Internet time servers.
Displays company logo. This logo ca n be cu stomized. For more information on customizing the
is running (show IP address). Other messages such as “DO NOT DISTURB” or “## MISSED
Shows the status of the phone, using icons as shown in the next table.
Displays the name of the account that is i n use. Select another account by pressing the LINE
SELECTOR BUTTONS
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 num ber
• END CALL Hangs up the call
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SPECIAL
Icon
LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon:
Blank – SIP Proxy/Server not registered
Phone Status Icon:
Speaker Phone Status Icon:
ON when the speakerphone is on
DND Icon:
Activate by pressing MUTE/DEL button once
SOFT
BUTTONS
(Excluding
GXP2000
and Only
When
Integrated
with
GXE5024/50
28)
Call Parking: FOR GXP2020/GXP2010 ONLY. Refer to the GXE5024/5028 Online User Manual
for more information.
•CallParkWhen a GXP2020 dials out, the Call Park soft button will display
on screen. To park the call, press the ‘Call Park’ button.
•PickUpWhen another GXP2020 goes off-hook the Call Pickup soft button
will display on screen. To pickup the parked call, press the ‘Call
Pickup’ button.
Call Queue:
Manual for more information.
• SignInPress this button to sign in to the call queue. Agent will be prompted in
• SignOut Press this button to sign out of the call queue. Press’ m enu’ button on
PUBLIC MODE (Also mentioned on p.31 of t his manual): This useful mode complements the
Call Queue feature by allowing va r i ous user agents to log in/log off, shar i ng the same phone.
When enabled, all other accounts on t he phone will not be active. For more information, refer
tohttp://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_
with_gxp.pdf
• LogIn Press this button to log i n the user agent into the call queue.
• Tab Press this button to jump to toggle between UserName and Password
• Backspace Press this button to erase the previo usly typed digit, letter, or character.
• LogOut Press this button to log out the user agent out of the call queue.
FOR GXP2020/2010 and 1200 only. Refer to the GXE5024/5028 Online User
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.
keypad to select ‘ok’. This will be displayed once the agent is signed in to
the call queue.
entry fields.
Table 9: LCD Icons
Solid – connected to SIP Server/IP address recei ved
Blinking – physical connection failed
OFF when the handset is on-hook
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ON when the handset is off-hook
FLASH when phone rings or a call is pe nding
OFF when the speakerphone is off
ON when the “do not disturb” is activated
Page 15
Calls Forwarded Icon:
Handset, Speakerphone and Ring Volume Icon:
AM/PM indicator
PM
AM
INDICATES calls are forwarded
Follow ‘call forwarding’ procedures
Each icon appears next to the volume icon
To adjust volume, use the up/down button
Real–time Clock:
Synchronized to Internet time server
Time zone configurable via web browser
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TABLE 10:GXPKEYPADBUTTONS
Key Button
Key Button Definitions
LINE BUTTONS
Line keys with LED, can be configured to different SIP profiles
TRANSFER
TRANSFER key: Transfer an ACTIVE call to another number
CONF
Press CONF button to connect Calling/Called party into conference
Also used to ‘REJECT’ incoming call.
HOLD
Place ACTIVE call on hold
MSG
Enter to retrieve voice mails or other messages
expires
Enter Keypad Configuration “MENU” mode when phone is in IDLE mode.
Use as ENTER key when in Keypad Configuration.
Standard phone keypad; press # ke y to send call; press * key to for IVR
functions
DND
DO NOT DISTURB key; Press DND to turn “Do not disturb” function on or off.
GXP2000.
INTERCOM
Turn intercom function on/off
MUTE
SEND
MENU
0 - 9, *, #
HEADSET
Mute an active call; or Delete a key entry
Enable/Disable hands-free speaker mode
Press SEND to dial a new number or red i al the last number dialed. Press
send button to send a call immediatel y before “no key entry timeout” valu e
Enter to retrieve voice mails or ot her messages
Press HEADSET key to answer/hang u p phone calls while using headset. It
also allows user to toggle between headset and speaker. Not available on
Brings phonebook on screen
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MAKINGPHONE CALLS
Handset, Speakerphone an d He ad set Mo de
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 theline 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 regi st ered 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 A CTIVE when it is in use and the corresponding LED is red.
Completing Calls
There are six ways to complete a call:
IAL: To make a phone call.
1. D
•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 (LI NE1) LED is red.
If you wish, select another LINE key (alternati ve SIP account).
• Enter the phone number
• Press the SEND key
or press the “DIAL” soft-key.
EDIAL: To redial the last dialed phone number.
2. R
When redialing, the phone will use the same SI P acc ount 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 acco unt 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.
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3. CALL RETURN:To call the last phone number that called your phone.
When returning a call, the phone will use the same SIP account as the call was made to. Thus, when
returning a call made to the third SIP account, the phone will use the third SIP account return the
call.
i. Hand-free option
1. Press the CALL RETURN soft-key
ii. Hand-set option
1. Take the Handset off-hook
2. Press the CALL RETURN soft-key
SING THE CALL HISTORY:To call a phone number in the phone’s history
4. U
When using the call history, the phone will use the same SIP account as was used for the last
call/call attempt. Thus, when returning a call made to the third SIP account, the phone will use the
third SIP account return the call.
• Press the MENU button to bring up the Main Menu.
• Select Call History and then “Received Calls”, “Missed Calls” or “Dialed Calls” depending on
your needs
• Select phone number using the arrow keys
• Press OK to select
• Press OK again to dial.
5. U
SING THE PHONEBOOK: Calling a phone in from the phone’s phonebook.
Each entry in the phonebook can be attached to an individual SIP account. The phone will use that
SIP account to make the phone call.
•Go to the phonebook by:
i. Pressing the phonebook button (bottom, left-hand sid e of phone), or
ii. Pres sing the DOWN arrow key, or
iii. Pres sing the menu button and
Selecting “Phone book” and
Press MENU
• Select the phone number by using the arrow keys
• Press OK so select
• Press OK again to dial.
6. P
AGING/INTERCOM:
The paging/intercom function can only be used if the SERVER/PBX supports this feature and both
the phones and PBX are correctly configured.
• Take the Handset/SPEAKER/Headset off-hook,
• Select the LINE key associated with account
• Press OK key to display LCD: LINEx: PAGE USING.
• Dial the phone number you want to Page/Intercom
• Press SEND key.
NOTE: Dial-tone and dialed number display occurs after the handset is off-ho ok and the line key is selected.
The phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call. Press
the “SEND” or “#” button to override the 4 second delay.
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 automaticall y call the assign ed ext ension.
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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 button’s speed di al /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 “#” ke y 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 phone’s 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 foll ow by S END or #
192.168.0.2 calling 192.168.0.23 -- dial #23 f ol l ow 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 R andom 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” butt on.
2. Resume: Resume call by pressing the correspondin g bl inking LINE.
3. Multiple Calls
place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.
: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE to
Mute/Delete
1. Press the MUTE button to enable/disable m uting 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 wishes to delete the last entered digit.
NOTE: To transfer calls across SIP domains, SIP service providers must support transfer across SIP
domains. Blind transfer will usually use the primary account SIP profile.
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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.
2. Attended (or Supervised) Transfer: Press “LINEx” button to make a call and automatically place
the ACTIVE LINE on HOLD. Once the second call is established, press “TRANSFER (or TRNF)”
key then the LINE button of the waiting line to transfer the call. Hang up the phone call after
“Transfer Successful” is displayed in the screen.
3. Semi-Attended Transfer: In the web UI, under “Account Settings”, select the Semi-attended
Transfer Mode. There are two modes to select for this feature. RFC5589 and Send Refer with early
dialog:
a. With RFC5589 phone will not send out REFER to transferee until 200OK is received from
transfer target. Like an attended transfer, press “LINEx” button to make a call and
automatically place the ACTIVE LINE on HOLD. Dial the second call, and as it is ringing
press the “TRANSFER (or TRNF)” key and then the blinking LINE button of the call on hold
to transfer the call.
b. With "Send REFER with early dialog" mode phone will send out REFER as soon as the
transfer target is selected (in the early dialog stage). Like an attended transfer, press “LINEx”
button to make a call and automatically place the ACTIVE LINE on HOLD. Dial the s econd
call, and as it is ringing press the “TRANSFER (or TRNF)” key and then the blinking LINE
button of the call on hold to transfer the call. Tran sf eror will hang up.
4. Auto-Attended Transfer (only on GXP1200): In the web UI, under “Advanced Settings”, set “Autoattended Transfer” to “Yes”.
Establish a call between two phones
Press TRANSFER button on GXP1200
Select another line on GXP1200, call another ph one and press SEND button
If the remote phone answers, press TRANSFER but ton on GXP1200 to transfer the call
NOTE: If there is no extra line available, GXP1200 will do the transfer as if Auto-attended
transfer is disabled.
5-Way Conferencing
GXP can host conference calls and supports up to 5-way conference call ing.
2. Initiate a Conference Call
Establish a connection with two or more parties
Press CONF button
Choose the desired line to join the conferen ce 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.
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 (Mess age 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 specif ic li ne 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 inter com/paging and BLF.
Table 11: GXP Call Features
Key Call Features
*30 Block Caller ID (for all subs equent calls)
*31 Send Caller ID (for all subsequent calls)
*67 Block Caller ID (per call)
*82 Send Caller ID (per call)
*50 Disable Call Waiting (for all subsequent calls)
*51 Enable Call Waiting (for all subsequent calls)
*70 Disable Call Waiting (per Call)
*71 Enable Call Waiting (per Call)
*72 Unconditional Call Forward
Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wai t for dial
tone. LCD will display “Call FWD Activated”.
*73 Cancel Unconditional Call For ward: dial “*73” and get the dial tone, then hang up.
LCD will display “Call FWD Activated”.
*90 Busy Call Forward
Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wai t for a dial
tone. Hang up.
*91 Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
*92 Delayed Call Forward
Dial “*92” for a dial tone. Dial the forwarding number followed by “#”. Wai t for a dial
tone. Hang up. LCD will display “Call FWD Activated”.
*93 Cancel Delayed Call Forward
Dial “*93” for a dial tone, then hang up.
CUSTOMIZED LCDSCREEN &XML
Grandstream GXP Series phones support both simple and advanced XML appli cations: 1) XML Cust om Scree n,
2) XML Downloadable Phoneboo k and 3) Adv anced XML S urvey Application. For more i nformation on how to
create a downloadable XML pho nebook, creating a cust om idle screen and/ or reprogrammi ng the soft-keys on
GXP-1200/GXP-2010/GXP2020, please visit our website
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Preference
Press
button to enter this sub menu including
•“Do NOT Disturb”
Press Menu button to choose the menu item.
Press ‘←’ to return to the main menu.
Configure
Press Menu button to display the configuration selections:
Network.
To setup IP-address, Net mask and Gateway address
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 e nter the key t he Key Pad Menu. The m enu option s available a re listed in table 8.
Table 12: Key Pad Configuration Menu
Call History
Status
Phone Book
LDAP Directory
Instant Messages
Direct IP call
Displays histories of incoming, dialed and missed calls.
Displays the network status, account stat uses, 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
Menu
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” m enu.
•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 Seconda ry Language
•
To enable/disable DHCP.
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•SIP
Press ‘←’ to return the main menu.
Factory Functions
Press Menu to display the factory function items including
Audio Loopback
Toggles the status of the Watchdog Timer.
Press ‘←’ to return to the main menu.
Reboot
Press Menu button to reboot the device
Display “Exit”
Press Menu button to exit the menu
To change SIP-server settings for primary account.
•Upgrade
In this menu setting regarding the firmware serv er 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 ph one.
Press Menu button to reset FACTORY DEFAULT setting. Do not
use Factory Reset unless you want to restore f actory settings
•Layer 2 QoS
Configure Vlan Tags
•
Speak into the handset. If you hear your voice in t he 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 b utton to exit
the diagnostic mode.
• Enable WDT
Exit
Exit from this menu.
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FIGURE 3:KEYPAD GUIFLOW
MENU
Answered Calls
Back
Call History
Back
Any of previous menus
Phone Book
New Entry
View Directory
LDAP Directory
Clear All
Preference
Do Not Disturb
Back
Network
Config
Enable DND
Default Ring
English
Do Not Disturb
Display Language
IP Setting
Back
Account
Save
Network
Firmware
Upgrade Via
802.1Q/VLAN Tag
Back
Upgrade
Layer 2 QoS
Instant Message
Diagnostic Mode
SIP
Dialed Calls
Missed Calls
New Entry
Download Phonebook XML
Back
Clear All
Name:
Number:
Acct:
Confirm Add:
Cancel & Return:
Call History
Status
Phone Book
LDAP Directory
Instant
Message
Direct IP Call
Preference
Config
Factory
Functions
Reboot
Exit
Download Directory
Search Configuration
Back
Back
Ring Tone
Ring Volume
LCD Contrast
LCD Brightness
Download SCR XML
Erase Custom SCR
Display Language
Chinese
Secondary Language
Language File Postfix
Back
IP
Net Mask
Gateway
DNSServer1
DNSServer2
SIP Proxy
Outbound
Proxy
SIP User ID
SIP Auth ID
SIP Password
SIP Transport
Audio
Server
Config Server
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Keypad/LED Diagnostic
Priority value
Reset Vlan Config
Page 27
CONFIGURATION VIAWEB 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 Microsoft’s IE or Mozilla Firefox.
Access the Web Configuration Menu
To access the phone’s Web Configuration Menu
• Connect the computer to the same network as the phone
• Make sure the phone is turned on and shows its IP-address
• Start a Web-browser on your computer
• Enter the phone’s IP-address in the address bar of t he browser
• Enter the administrator’s password to access t he Web Configuration Menu
1
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.
2
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 y our browser.
3
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.
1
2
3
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 administrat ors 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
The device ID, in HEXADECIMAL format.
IP Address
This field shows IP address of GXP
Product Model
This field contains the product model informati on.
Part Number
This field contains the product part number
Software Version
System Up Time
This field shows system up time since the last reboot.
System Time
This field shows the current time on the phone sy st em.
Registered
Indicates whether accounts are registered to the related SIP server(s). GXP can
PPPoE Link Up
Indicates whether the PPPoE connection is ena bl ed (connected to a modem).
End User
Password
This contains the password to access the Web Configuration Menu. This field
is case sensitive with a maximum length of 25 characters.
IP Address
(secondary). These fields are set to zero by default.
•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
support four unique SIP profiles.
Table 14: Device Configuration – Basic Settings
The 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 t he PPPoE feature, set the
PPPoE account settings. The GXP establishe s 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
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on the PBX and it
Each function is connected to one of the accounts and has a target user ID.
Time Zone
This parameter controls the date/time display according to the specified time zone.
LCD Backlight Always
On
1200/GXP-2000 only.
Time Display Format
LCD time display in 12 hour or 24 hour format
Date Display Format
Choose one of the following formats:
This option applies to GXP280/GXP285/GXP1200/GXP2000 only.
Display Clock instead of
Date
Choose to display clock or date on LCD. This option appli es to GXP-280/GXP285/GXP-1200/GXP-2000 only.
Multi Purpose Key X
These options are used to assign a function to t he cor responding multi purpose key.
Options available are:
1. “Speed Dial”.
2. “BLF” (Busy Lamp Field). This option has to be supported
indicates the status of the extension. The t hree 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.
5. “Speed Dial Via Active Account”. This option will act just like speed dial, but
based on the current active account. For instance, if the phone is offhook
and account 4 is active, it will call the configured speed dial number using
account 4.
Turn on LCD backlight at all times. Default is No. T hi s option applies to GXP-
• Year-Month-Day
• Month-Day-Year
• Day-Month-Year
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ends the first Sunday of November at 2AM. The saving is 60 minutes.
Brightness
the brightest. For GXP2010 and GXP2020 only.
LCD Contrast
Set LCD contrast. Range from 0 to 20. Not for GXP280/285
Disable in-call DTMF
display
Mute Speaker Ringer
Default is No. When it’s enabled, speaker won’t ring on an incoming call.
Disable Missed Call
Backlight
Default is No. By default, LCD backlight will l it whenever there is a missed call.
Not for GXP280/285.
Daylight Savings Time
This parameter controls time displayed in daylight savings time. I f 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.
ohour: 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 iter ation of the weekday (e.g.:
1st Sunday, 3rd Tuesday etc). If “weekday” is not zero and “day” is negativ e, 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 b y a n egative (-)
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 st ar ts from the second Sunday of March at 2AM and
LCD Backlight
Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means
Default is No. This field is used to hide the keypad input during a call.
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NOTE: This is not available for GXP2000
PC Port Mode
For GXP1200/GXP280 only. Default is ‘Switch’. Choose to toggle between
Switch/Hub mode.
Admin
the “Advanced
page. Password field is purposely blank for
king update and saved. The maximum password
length is 25 characters.
G723 rate
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.
127.
Silence Suppression
This controls the silence suppression/VAD feature of the audio codec G.723 and
of VAD packets
(instead of audio packets) will be sent during the period of no talking. If set to
“No”, this feature is disabled.
HEADSET Key Mode
Select either “Default mode” or “Toggle Speaker(default )/Hea dset ” or “Toggle
Speaker/Headset(default)”.
In “Default Mode”, only the speakerphone will ring for an incoming call. User ca n
use the headset key to pick-up, speak, and hang up calls throu gh headset . The
headset icon will appear on the LCD when a call is in progress.
If “Toggle Speaker/Headset(default)”is checked, onl y the headset will ring for an
incoming call.
If “Toggle Speaker(default)/Headset” is checked, only the speakerphone will ring for
an incoming call but the user can make the phone rings using the headset by
pressing the HEADSET key while phone is idle. The headset icon will appear on
the idle LCD screen.
Advanced User configuration includes not only the end user configuration, but also advanced configuration
such as SIP configuration, Codec select ion, NAT Traversal Setting and other miscellaneous configuration.
Table 15: Advanced Settings
Administrator password. Only the administrator can access
Password
Settings” and “Account Settings”
security reasons after clic
iLBC payload type
Payload type for iLBC. Default value is 97. The valid range is between 96 and
G.729. If set to “Yes”, when silence is detected, a small quantity
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Voice Frames per TX
This field contains the number of voice frames to be transmitted in a single
d the IS limit is based on the maximum size of
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
codec in the above codec Preference List or the actual
used payload type negotiated between the 2 conversation parties at run time.
, if the first codec is configured as G.723 and the “Voice Frames per TX” is
value in the SDP message of an INVITE request will
larly, if
this field is set to 2 and the first codec is G.729 or G.711 or G.726, then the
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
s 20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for
Please be careful when editing these parameters. Adjusting these parameters
a patent dynamic jitter
Grandstream recommends using the default settings provided. Grandstream
mmend adjusting these parameters if you are an average user.
fect 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
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.
Layer 2 QoS
This contains the value used for layer 2 VLAN tag. Def ault setting is blank.
Data VLAN Tag
Default is 0. Enabling the Data VLAN filtering will help reduce the load on the
primarily for VLAN
filtering where tagged traffic will be forwarded to the DSP .
No Key Entry Timeout
Default is 4 seconds.
Use # as
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
lent to the “(Re)Dial” key. If set to “No”, the “#” key is
included as part of the dial string.
Local RTP port
This parameter defines the starting local RTP-RTCP port pair used to listen and
mit. It is the base RTP port for channel 0. When configured, channel 0 will
l
use port_value+2 for RTP and port_value+3 for its RTCP. The default value is
5004.
Use Random Port
This parameter, when set to “Yes”, will force random generation of both the local
set to 2, then the “ptime”
be 60ms because each G.723 voice frame contains 30ms of audio. Simi
“ptime” value in the SDP message of an INVITE reque st will be 20ms .
G.726, it i
G.729/G.728, 64 (x10ms) and 64 (x2.5ms) fram es respectively.
will also change the dynamic jitter buffer. The GXP has
buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.
does not reco
Incorrect settings will af
phone, but it isn’t necessary in most environments. This is
Dial Key
key is essentially equiva
trans
use this port _value for RTP and the port_value+1 for its RTCP; channel 1 wil
SIP and RTP ports. This is usually necessary when multiple GXPs are behi
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sends a blank UDP packet to the
SIP server in order to keep the “hole” on the NA T open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
STUN Server
be
displayed in the STATUS page of the Web UI.
Firmware Upgrade and
Default method is HTTP. Firmware upgrade m ay take up to 10 minutes
process.
Via TFTP Server
This is the IP address of the configured TFTP server. If selected and it is non-
le 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
retrieved, the new downloaded image will be verified and then saved
Grandstream strongly recommends that the user upgrade firmware locally
in a LAN environment if using TFTP to upgrade. Please do NOT interrupt the
pecially the power supply) as this will damage the
device.
The HTTP server URL used for firmware upgrade and configuration via HTTP.
” is the specific TCP port that the HTTP server is using; omit if using
will only perform HTTP download once
at boot up.
Config Server Path
IP address or domain name of firmware server.
XML Config File Password
Only)
Firmware File
Default is blank. If configured, GXP will request the firmware file with the
Config File
Default is blank. End user should keep it blank.
Allow DHCP Option 43 and
server
Default is Yes. This allows the device to get provisioned automatically.
Authenticate Conf File
Default is “No”. If set to “Yes”, configuration file would be authenticated before
acceptance. End user should use default setting.
Keep-alive interval
Provisioning
Via HTTP Server
This parameter specifies how often the GXP
IP address or Domain name of the STUN server. STUN resolution result will
depending on network environment. Do not interrupt the firmware upgrading
zero or not blank, the GXP will attempt to retrieve a new configuration fi
image is
into the Flash memory.
Note:
TFTP upgrade process (es
For example: http://provisioning.mycompany.com:6688/Grandstream/1.2.5.3
Here “:6688
default port 80.
Note: If Auto Upgrade is set to No, GXP
(For
GXP280/GXP/285/GXP1200
Prefix/Postfix
Prefix/Postfix
Option 66 to override
configuration updates via XML configuration files. Users can set the XML config
file password in the web UI of the phone.
prefix/postfix. This setting is useful for ITS Ps. End user should keep it blank.
The XML provisioning system allows Grandstream phones to perform
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. Choose “Yes” to enable automatic HTTP upgrade and
every” field, enter the number of minutes to check the
HTTP server for firmware upgrade or configuration changes. When set to “No”,
form HTTP upgrade and configuration check once at boot
up.
LDAP Directory
IP address or domain name of LDAP script server
Phonebook XML
L server path and
When the user downloads the XML phone the manually
entered or edited entries will not be delet ed unl ess this option is selected to Yes.
Idle Screen XML
Download
Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use
Custom Filename” or not, and define the “XML serve r path”.
XML Application
Enter server path for XML application. This option applies to GXP-2020 and
GXP-2010 only.
DTMF Payload Type
This parameter sets the payload type for DTMF using RFC2833. Default is 101.
Onhook Threshhold
es the time handset has to be down to be recognized it’s onhook.
Default is 800ms. For GXP280/285 only.
Syslog Server
The IP address or URL of System log server. This feature is especially useful for
ITSPs.
Syslog Level
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
g uses USER facility. In addition to standard Syslog payload, it
GS_LOG: [device MAC address][error
: May 19 02:40:38 192.168.1.14 GS_LOG:
[00:0b:82:00:a1:be][000]. Ethernet lin k i s up.
Automatic Upgrade
Offhook Auto Dial
This function is used by ITSP. End user should NOT touch these parameters.
Default is No
provisioning.
In “Check for upgrade
the phone will only per
Enable the XML phonebook via TFTP or HTTP. Define XM
download interval.
To configure a User ID/extension to dial automatically when the phone is taken
offhook.
It determin
following events:
• product model/version on boot up (INFO lev el )
• 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 Syslo
contains the following components:
code] error message
For example
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NTP server
This parameter defines the URI or IP address of the NTP (Network Time
Protocol) serve. It is used to display the current dat e/time.
Public Mode
Default is ‘No’ (standard mode). Supported only on GXP1200, GXP2010, and
signed specifically to support the Call Queue
feature implemented on Grandstream’s GXE5024 or GXE5028. If set to ‘Yes’,
this feature will allow other call queue agents to login/logout using their own
, all other
accounts will not be active.
Distinctive Ring Tone
Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a
particular Caller ID. The GXP will ONLY use selected ring tones for particular
, the GXP will use System Ring Tone. When
selected and no Caller ID is configured, the selected ring tone will be used for all
incoming calls.
System Ring Tone
System ring tone. Default is North American standard.
standard.
Call Progress Tones
Using these settings, users can configure ring or tone frequencies based on
are supported.
To page a group : [paging group feature code] +[ * ]+[group extension]
Disable Call Waiting
Default is No. If set to Yes, the call waiting feature will be disabled.
Disable Call
Waiting Tone
Default is No. If set to Yes, the call waiting tone will be disabled.
Disable Direct IP Calls
Default is No. If set to Yes, direct IP calls will be disabled
GXP2020. This feature is de
login name and password on the same phone. When enabled
Caller IDs. For all other calls
Adjust system ring tone frequencies and cad ences based on local telecom
parameters from local telecom. By default, they are set to North American
standard.
Frequencies should be configured with known values to avoid uncomfortable
high pitch sounds.
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are i n 10ms)
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence.
In order to set a continuous ring, OFF should b e zero. Otherwise it will ring ON
ms and a pause of OFF ms and then repeat the pat tern. Up to three cadences
Intercom User ID:
This field is used to configure the Intercom key in the phone. For GXP2010 and
GXP2020 only
If the phone is working with a GS GXE502X IP-PBX it can be configured in the
following manner:
• To page an extension : [intercom feature code] +[ *]+[extension number]
•
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Use Quick IP Call Mode
Dial an IP address under the same LAN/VPN segment by entering the last octet
.
9 and
XXX <=255, phone will make direct IP call to aaa.bbb.ccc.XXX where
Quick IP
Call Mode
Disable Conference
Default is No. If set to Yes, conference wi ll be di sabled.
Lock keypad update
If set to “Yes”, the configuration changes vi a key pad are disabled.
Enable MPK Sending
DTMF
Default is No. If set to “Yes”, Muti-Purpose keys can be sent as DTMF.
For GXP2020/2010/2000.
For
GXP2000, MUTE/DEL button functions as DND button when pressed while
phone is idle.
Disable Transfer
Default is No. If set to Yes, transfer will be di sabled.
Disable Multicast Filter
Default is No. If set to Yes, the phone will not filter out (discard) multicast
packets.
Semi-Attended Transfer
Default is RFC5589 mode, which allows the REFER m essag e to be sent af ter
TRANSFER is not finished until it receives
(e.g. 4xx, 2xx) from the SIP server.
Disable Headset Button
ill
have no effect.
Enable Constraint Mode
Default is No. If set to Yes, the phone will limit the end-user’s access to the
configuration of the phone.
Display CID instead of
Name
If set to “Yes”, CID will be displayed in the screen instead of Name. Default is
”No”. For GXP280/285 only.
Headset Port Type
Select either 2.5mm or RJ22 headset ports to be adj ust ed.
Headset TX gain (dB)
Increases the selected headset’s (2.5mm or RJ22) TX gain by + or – 6dB.
Default is 0dB
Headset RX gain (dB)
Increases the selected headset’s (2.5mm or RJ22) RX gain by + or – 6dB.
Default is 0dB
in the IP address.
In the Advanced Settings page there is an option “Use Quick IP-call mode”
Default setting is No. When set to YES, and #XXX is dialed, where X is 0aaa.bbb.ccc comes from the local IP address RE GARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK). See
for details.
Disable DND
Mode
Default is No. If set to “Yes”, the “DND” button on keypad will be disabled.
INVITE is answered so that the
responses
Default is No. If set to Yes, Headset button will be disabled and pressing it w
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Language file postfix allows the language file to have different postfixes so the
ne can request a particular file. It will append an underscore "_" plus the
The default language file name is "gxp.lpf". If the field “Language File postfix
dvanced settings of the Web GUI, set “Display Language” to
6. Update and reboot the phone
Account Active
This field indicates whether the account is active. The default value for the
is No.
Account Name
The name associated with each account - displayed on LCD.
SIP Server
SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy
IP address or Domain name of Outbound Proxy , Media Gateway, or Session
outbound proxy can provide solution for symmet ric NAT.
actual phone number or formatted like one.
Authenticate ID
SIP service subscriber’s Authenticate I D used for authentication. It can be
identical to or different from SIP User ID.
Display Language
Allows user to choose preferred display langu age i n web UI and key pad UI.
pho
string in the language file postfix.
“has "NL" string in it, then the phone will request "gxp_NL.
"gxp.lpf."
User can only load one secondary language.
Italian, Polish, Portuguese, Slovak, S l ovenian and Spanish.
How to set up Secondary Language:
Note: This is similar to updating firmware in your local network
Please refer to http://www.grandstream.com/faqsfirmware.html#4 for details.
1. Download the language package
from http://www.grandstream.com/firmware.html
2. Unzip the language package
3. Open the desired language zip file
4. Copy gxp.lpf to the firmware server directory
server.
5. Access the a
“Secondary Language”
GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires it s
own configuration page. Their configurat i ons are identical.
Table 16: SIP Account Settings
primary account (Account 1) is Yes. The default value for the other two accounts
Border Controller. Used for firewall or NAT penet rat i on in different network
environment. If the system detects symmetric NAT, STUN will not work. ONLY
SIP User ID
User account information provided by VoIP servi ce provider (ITSP); either an
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ITSP.
Name
SIP service subscriber’s name that is used for Caller ID display.
Use DNS SRV:
Default is No. If set to “Yes”, the client will use DNS SRV to look up server.
User ID is Phone
If the phone has an assigned PSTN telephone numb er, this field should be set to
attached to the “From” header in SIP request
SIP Registration
This parameter controls sending REGISTER messages to the proxy server. The
default setting is “Yes”.
Un-register on Reboot
Default is “No”. If set to “Yes”, the SIP user’s registration information will be
cleared from the server when the phone reboots.
SIP Instance ID
Default is set “No.” If set to Yes it will be enabled and will add reg-ID and Instance
phones to register.
Register Expiration
This parameter allows user to specify the time frequency (in minutes) that GXP
minutes. The maximum interval is 65,535 minutes (about 45 days).
Local SIP Port
This parameter defines the local SIP port used to l i st en and transmit. The default
Account 4 respectively.
SIP Registration Failure
Retry Wait Time
Retry registration if the process failed. Default is 20 seco nds.
SIP T1 Timeout
RFC 3261 SIP T1 timer. Default is 1 second.
SIP T2 Interval
RFC 3261 SIP T2 timer. Default is 0.5 seconds.
SIP Transport
Choose SIP Transport between UDP and TCP. Default is UDP.
Use RFC3581
Symmetric Routing
Default No. When selected the phone will follow t he routing procedures specified
in RFC3581.
NAT Traversal (STUN)
This parameter activates the NAT traversal mec hanism. If activated (by choosing
data) to the SIP server to keep the “hole” on the NAT open.
SUBSCRIBE for MWI:
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
Authenticate Password
Number
SIP service subscriber’s account password for GXP to register to (SIP) servers of
“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
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
refreshes its registration with the specified registrar. The default interval is 60
value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
“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 configurat ion 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
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PUBLISH for Presence
Enable Presence feature.
Proxy-Require
SIP Extension to notify SIP server that t he uni t is behind the NAT/Firewall.
Voice Mail UserID
When configured, user can access message s by pressing “MSG” button. This ID
is usually the VM portal access number.
Send DTMF
This parameter specifies the mechanism to transmi t DTMF digit. There are 3
very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
Early Dial
Default is No. Use only if proxy supports 484 responses.
Dial Plan Prefix
Sets the prefix added to each dialed number.
BLF Call-pickup Prefix
Default is ‘**”. This prefix is prepended when answering call with BLF key.
Delayed Call Forward
Wait Time
Time waited before the call is forward to a number or VM.
Enable Call Features
Default is Yes. If set to “No”, Call transfer, Call Forwarding & Do-Not-Disturb are
softkey will be hidden if call feature code is disabled for Account 1.
Call Log
User can choose to disable Call Log and what kind o f calls to log.
Session Expiration
The SIP Session Timer extension enables SIP ses sions to be periodically
The default value is 180 seconds.
Min-SE
Defines the minimum session expiration (in seconds). Default is 90 seconds.
Caller Request Timer
If set to “Yes”, the phone will use session timer wh en it makes outbound calls if
remote party supports session timer.
Callee Request Timer
If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.
Force Timer
If set to “Yes”, the phone will use session timer even if the remote party doe s not
Caller Request Timer, Callee Request Timer, and F orce Timer.
UAC Specify Refresher
As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
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.
supported modes: in audio which means DTMF is combined in audio signal (not
Default is 20 seconds.
supported locally provided ITSP support t hose f eatures. In addition, “ForwardAll”
“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval
expires, if there is no refresh via a UPDATE or re-INVI T E m essage, 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.
support this feature. If set to “No”, the session t i m er is enabled only when the
remote party supports this feature. To tu rn off Session Timer, select “No” for
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use the phone as the refresher.
Force INVITE
Session Timer can be refreshed using INVITE m ethod or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
Enable 100rel
PRACK (Provisional Acknowledgment) method enables reliability to SIP
networking..
Account Ring Tone
There are 4 uniquely defined ring tones:
designated account will play selected ring tone.
Ring Timeout
Defines how long ring will ring when receiving a call . Default is 60 seconds.
Send Anonymous
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.
Anonymous Method
Whether to use “<sip:anonymous@anonym ous. i nvalid>” in the From Header or PAsserted-Identity header.
Anonymous Call
Rejection
Default is NO. If set to YES, anonymous call will be rejected
Auto Answer
Default is No. If set to “Yes”, GXP will automatically switch on speaker to answer
the SIP info header from the server.
Allow Auto Answer by
Call-Info
If the Call-Info header contains answer-after=0, the call be answ ered
automatically (so called paging mode).
Turn off speaker on
remote disconnect
When BYE is received, the phone will turn off its spe aker automatically.
Check SIP User ID for
incoming INVITE
Check the SIP User ID in Request URI. If they don’t match, the call will be
rejected.
Refer-To Use Target
Contact
Attribute in SDP
Preferred Vocoder
GXP supports up to 7 different Vocoder types incl udi ng G.711(a/µ) (also known
choosing the appropriate option in “Choice 8”.
SRTP Mode
Enable SRTP mode based on selection. Default is No.
UAS Specify Refresher
As a Callee, select UAC to use caller or proxy se rv er as the refresher, or UAS to
provisional responses (1xx series). This is required to support PSTN inter-
•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
Disable Multiple Media
the incoming call. Set to Intercom/Paging mode, it will answer the call based on
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred target’s Contact header information.
Default is No.
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 i s included with the same preference
order in SDP message. Enter the first Vocod er in this list by choosing the
appropriate option in “Choice 1”. Similarly , enter the last Vocoder in this list by
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Page 41
BLF", choose account number, monitored number, etc.
Special Feature
Default is Standard. Choose the selection t o m eet special requirements from Soft
eventlist BLF URI
If a server supports this feature, user need s 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 Ba sic Settings, please select "eventlist
Switch vendors.
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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Page 42
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 Interfac e
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 Server’s 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 ser ver, 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:
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:
customerFree.cfm.
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Firmware 1.2.5.3 Last Updated: 03/2011
http://www.grandstream.com/firmware.html .
http://support.solarwinds.net/updates/New-
Page 43
Instructions for local TFTP Upgrade:
1. Unzip the file and put all of them under the root dire ct ory of the TFTP server.
2. The PC running the TFTP server and the GXP should be in the sam e LA N 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 phone’s web configuration page
5. Configure the Firmware Server Path with the I P 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 Microso ft 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 f i rmware.
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 XM L conf i guration file.
Please refer to our XML provisioning guide using this link
Managing Firmware an d 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 Provisioni ng Server load at any given time.
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Page 44
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” (P l ease 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: 0-9
A: 22 (press the “2” key twice, “A” will show on the LCD)
B: 222
C: 2222
D: 33 (press the “3” key twice, “D” will show on the LCD)
E: 333
F: 3333
Example: if the MAC address is 000b8200e395, it shoul d be key in as “0002228200333
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 cont inue 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 m enu interface.
395”.
Grandstream Networks, Inc. GXP User Manual Page 44 of 44
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