Grandstream Networks, Inc.
GXP-2020
6-line SIP Enterprise Phone
GXP-2020 User Manual
Firmware Version 1.1.4.3
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TABLE OF FIGURES
GXP-2020 USER MANUAL
Figure 1: Connectors on the bottom of the GXP-2020 ................................................................... 6
Figure 2: Attach the wall-mount spacers to the GXP-2020............................................................. 7
Figure 3: Location of the fixed hangers on GXP-2020.................................................................... 7
Figure 4: Placing tab for mounting phone to wall............................................................................ 8
Figure 5: LCD in IDLE (on-hook) and ACTIVE (off-hook) mode................................................... 11
Figure 6: Getting Familiar with Key Pad .......................................................................................13
Figure 7: Keypad GUI Call Flow ................................................................................................... 22
TABLE OF TABLES
GXP-2020 USER MANUAL
Table 1: Definitions of the GXP-2020 Connectors.......................................................................... 6
Table 2: GXP-2020 Key Features in a Glance................................................................................ 9
Table 3: GXP-2020 Hardware Specifications ................................................................................. 9
Table 4: GXP-2020 Technical Specifications................................................................................ 10
Table 5: LCD Button Definitions.................................................................................................... 12
Table 6: LCD Icon Definitions .......................................................................................................12
Table 7: GXP-2020 Button Definitions..........................................................................................13
Table 8: GXP-2020 Call Features................................................................................................. 19
Table 9: Key Pad Configuration Menu.......................................................................................... 20
Table 10: Device Configuration - Status Definitions ..................................................................... 24
Table 11: Device Configuration – Basic Settings Definitions........................................................ 24
Table 12: Advanced User Configuration Page Definitions............................................................ 25
Table 13: SIP Account Configuration Page Definitions ................................................................ 28
GUI INTERFACE EXAMPLES
GXP-2020 USER MANUAL
1. SCREENSHOT OF CONFIGURATION LOGIN PAGE
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
3. SCREENSHOT OF STATUS CONFIGURATION PAGE
4. STATUS CONFIGURATION PAGE DEFINITIONS
5. SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE
6. SCREENSHOT OF SIP ACCOUNT CONFIGURATION
7. SCREENSHOT OF SAVED CONFIGURATION CHANGES
8. SCREENSHOT OF REBOOT PAGE
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Welcome
Your Grandstream GXP-2020 features a new sophisticated design and is very easy to use. The GXP-
2020 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.
This 6 line enterprise SIP phone offers more advanced functionality than the GXP2000. It features a
larger graphical LCD display, four (4) XML programmable soft-keys, seven (7) programmable hard keys,
5-way conferencing and three (3) adjustable positioning angles. The GXP-2020 also offers dual
10/100Mbps auto-sensing Ethernet ports with integrated P-o-E, secure central configuration with AES
encryption, and support for a broad range of voice codecs.
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-2020 as it may cause damage to
the products and void the manufacturer warranty.
•
•
This document is contains links to Grandstream GUI Interfaces. Please download these examples
http://www.grandstream.com/user_manuals/GUI/GUI_GXP-2020.rar for your reference.
This document is subject to change without notice. The latest electronic version of this user manual
is available for download @: http://www.grandstream.com/user_manuals/GXP-
2020_User_Manual.pdf
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•
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.
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Installation
EQUIPMENT PACKAGING
The GXP-2020 phone package contains:
1) One GXP-2020 Main Case
2) One Handset
3) One Phone Cord
4) One Universal Power Adaptor
5) One Ethernet Cable
6) One High Phone Stand
7) One Low Phone Stand
8) Two Wall Mount Spacers
CONNECTING YOUR PHONE
The connectors of the GXP-2020 are located on the bottom of the device.
Figure 1: Connectors on the bottom of the GXP-2020
GXP Ext
Connection
Headset and
handsetJet
WAN/LAN RJ-45
10/100Mbps ports
Power
Jack
Table 1: Definitions of the GXP-2020 Connectors
GXP
Ext Connection
Connects the GXP Ext directly to the GXP using connection cable. Draws power
from PoE if provided by network.
10/100Mbps RJ-45 ports for WAN (PC) and LAN connections (switched or routed).
Support PoE (802.3af). Draws power from either spare line or signal line
WAN/LAN
5V DC power port; UL Certified
Two headset jacks: RJ11/2.5mm
Power Jack
Headset Jack
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WALL-MOUNT
GXP-2020 can be wall mounted. For proper wall mounting please place the two wall-mount spacers into
the proper holes on the bottom of the GXP-2020 as show below:
Figure 2: Attach the wall-mount spacers to the GXP-2020
To position the phone on the wall, place two fixed hangers on the wall, hang the back of the phone on the
fixed hangers.
Figure 3: Location of the fixed hangers on GXP-2020
Fixed hangers
To use the handset, pull out the tab (extension downward) from the handset cradle, rotate the tab and plug it
back into the slot with the extension up to hold the handset.
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Figure 4: Placing tab for mounting phone to wall
`
Tab with
extension
Handset
Rest
Tab with
extension up
Tab
SAFETY COMPLIANCES
The GXP-2020 phone complies with FCC/CE and various safety standards. The GXP-2020 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 damages 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 the right to remedy warranty policy without prior
notification.
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Product Overview
GXP-2020 features 6 lines, each with an individual SIP account and offers a
large graphical LCD display, four (4) XML programmable soft keys, and seven
(7) programmable hard keys. It offers intuitive user interfaces, 5-way
conferencing, 2 headset jacks, and three (3) adjustable positioning angles. Like
the GXP2000, the GXP-2020 also offers dual 10/100Mbps auto-sensing
Ethernet ports with integrated P-o-E, secure central configuration with AES
Encryption, superior audio quality (using G.722 wide-band codec) and excellent
manageability. The GXP series is fully compatible and interoperable with most
SIP end-user devices, IPPBX and Softswitches.
Table 2: GXP-2020 Key Features in a Glance
Features
Benefits
Table 3: GXP-2020 Hardware Specifications
LAN Interface (Ethernet ports)
Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port
with auto detection
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Table 4: GXP-2020 Technical Specifications
Lines
6 direct lines with independent SIP accounts, 7 programmable speed dial keys
4 XML programmable soft-keys
Support SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856,
3863) for use of 7 MFKs, SIP Dialog package (RFC 4235)
Support for SIP MESSAGE method (RFC 3428)
Stores up to 100 incoming IM messages (drops IM message 101 plus)
Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Expansion interface, Address Book
Advanced Digital Signal Processing (DSP)
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), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control (patent-pending) and packet delay & loss concealment
Voice mail indicator, downloadable custom ring-tones, call hold, call transfer
(attended/blind), call forward, call waiting, caller ID, mute, redial, call log, caller ID
display or block, Do-Not-Disturb (DND) and volume control
Multi-party conferencing (up to 5), dial plans, off-hook auto dial, auto answer, early
dial and speed dial
Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS
User specific URL for configuration file and firmware files
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Using the GXP-2020 SIP Enterprise Phone
GETTING FAMILIAR WITH THE LCD
The GXP-2020 has a dynamic and customizable screen. The screen displays differently depending on
whether the phone is idle or in use (active screen). Figure 5 illustrates the IDLE and ACTIVE screens.
Figure 5: LCD in IDLE (on-hook) and ACTIVE (off-hook) mode.
SIP phone lines
Line Status Indicator and Account Name
Date and Time
6 line selectors
Customizable logo
Status Bar
Network Status (Phone IP Address)
Context Sensitive Soft Buttons
.
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Table 5: LCD Button Definitions
Key Button
Key Button Definitions
LINE SELECTORS (6)
Selects the phone line printed on its right-hand side.
Displays the available phone lines. Choose a phone line by pressing the
corresponding line selector on the left-hand side.
SIP PHONE LINES
DATE AND TIME
LOGO
Displays the current date and time. Can be synchronized with Internet time servers.
Displays company logo. This logo can be customized. For more information on
customizing the logo, please see the web-configuration menu (page 32).
Shows the status of the phone and network. It will indicate whether the network is
down, starting or is running (show IP-number). Other messages such as “DO NOT
DISTURB” or “## MISSED CALLS” are shown here too.
NETWORK STATUS
STATUS BAR
Shows the status of the phone, using icons as shown in the next table.
LINE STATUS
INDICATOR
Displays the name of the account that is in use. Select another account by pressing
the LINE SELECTOR BUTTONS
The soft-buttons are context sensitive and will change depending on the status of
the phone. Typical functions assigned to soft-buttons are:
•
•
NEW CALL
Press this button to make a new hand-free call.
FORWARD ALL Unconditionally forwards the main phone line to another
phone
•
MISSED CALLS This option shows up there were unanswered calls to this
SOFT-BUTTONS
phone. The MissedCalls option shows a list of the missed
calls
•
•
•
CALL RETURN Calls the phone that called/tried to call your phone last.
REDIAL
Redials the last number
Hangs up phone
END CALL
Table 6: LCD Icon Definitions
Icon
LCD Icon Definitions
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
Handset, Speakerphone and Ring Volume Icon:
Each icon appears next to the volume icon
To adjust volume, use the up/down button
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Figure 6: Getting Familiar with Key Pad
Message Waiting Indicator
Voice Message button
Menu Keys
Graphical LCD screen
Line 1-6 keys
Softkeys
Dedicated buttons
Phonebook button
Speed Dial Buttons
(Programmable Buttons)
Standard Keypad
Send Button
Hand Free Button
Hold Button
Table 7: GXP-2020 Button Definitions
Key Button
LINE BUTTONS (6)
TRANSFER
CONF
Key Button Definitions
6 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.
MUTE
DND
DO NOT DISTURB key; Press DND to turn “Do not disturb” function on or off.
Toggle between headset and speakerphone mode when in hands free mode
Turn intercom function on/off
HEADSET
INTERCOM
Brings phonebook on screen
HOLD
MSG
Place ACTIVE call on hold
Enter to retrieve (video) 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
SEND
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.
MENU
Standard phone keypad; press # key to send call; press * key to for IVR
functions
0 - 9, *, #
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MAKING PHONE CALLS
HANDSET, SPEAKERPHONE AND HEADSET MODE
Handset can be toggled between Speaker and Headset. To switch between Handset and
Speaker/Headset, press the Hook Flash in the handset cradle or press the SPEAKER button.
MULTIPLE SIP ACCOUNTS AND LINES
The GXP-2020 can support up to 6 independent SIP accounts. Each account is capable of independent
SIP server, user and NAT settings. Each of the 6 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 the first three of the six SIP accounts as “BenjiSIP”, “SIPnGO”, “IbenSIP”,
respectively and ensure each is active and registered. When LINE1 is pressed, you will hear a dial tone
and see “BenjiSIP” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see
“SIPnGo” on the LCD display; when LINE3 is pressed, you will hear a dial tone and see “IbenSIP” on the
LCD display.
To make a call, select the line you wish to use. The LED will light up green. Switch lines before dialing by
pressing the same LINE button one or more times. If you continue to press one LINE, the selected
account will circulate among the registered accounts.
For example: when LINE1 is pressed, the LCD displays “BenjiSIP”; If LINE1 is pressed twice, the LCD
displays “SIPnGo” and the subsequent call will be made through SIP account 2 - SIPnGo.
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-2020 will flash the next available LINE (from left to right) in
red. A line is ACTIVE when it is in use and the corresponding LED is solid 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 made 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.
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.
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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
4. USING THE CALL HISTORY: To call the a phone number in the phone’s history
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. USING 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 side of phone), or
ii. Pressing the DOWN arrow key, or
iii. Pressing 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. PAGING/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-hook 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” button to override the 4 second delay.
SPEED DIAL
The seven multi-functional buttons, located on the right-hand-side of the phone, can be configured for
speed dial. Press the speed dial button to automatically call the assigned extension.
Note: The multi-functional buttons will function as LINE keys when all six primary lines are busy. The
LED will flash indicating the incoming call. Press the button to pick up the call. If any one of the 7
functions keys is associated with a call, the speed dial/BLF function will not work.
For example: When first multi-functional button is in use, it cannot be used for speed dial/BLF.
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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 start “ * ” key represent the dot“.” ; The hash “#” key
represent colon “:”. Press OK to dial out.
QUICK IP CALL MODE
The GXP-2020 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 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
IP-IP call will also use STUN. Configure the “Use Random Port” to “NO” when completing Direct IP calls.
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ANSWERING PHONE CALLS
RECEIVING CALLS
1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE
flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER
or by pressing the corresponding account LINE button.
2. Incoming multiple calls: When another call comes in while having an active call, the phone will
produce a Call Waiting tone (stutter tone). Next available lines will flash red (as described in
section 4.3.2). Answer the incoming call by pressing its corresponding LINE button. The current
active call will be put on hold.
3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via
SPEAKER. (PBX (or Server) must also supports this feature)
DO NOT DISTURB
1. Press the “DND” button if you do not want to take a call. This will send the caller directly to
voicemail.
2. Press the “DND” button to set phone to ‘do not disturb’ (icon will be on the screen). The phone
will not ring and send caller directly to voicemail. (see note above)
PHONE FUNCTIONS DURING A PHONE CALL
CALL WAITING/ CALL HOLD
1. Hold: Place a call on ‘hold’ by pressing the “HOLD” button.
2. Resume: Resume call by pressing the corresponding blinking LINE.
3. Multiple Calls: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE
to place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.
MUTE/DELETE
1. Press the MUTE button to enable/disable muting the microphone.
2. The “Line Status Indicator” will show “LINEx: SPEAKING” or “LINEx: MUTE” to indicate whether
the microphone is muted.
CALL TRANSFER
GXP-2020 supports both blind and attended (or supervised) transfer:
1. Blind Transfer: Press “TRNF” 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 call is established, press “TRNF” key to transfer the
call and hang up.
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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5-WAY CONFERENCING
The GXP-2020 can host conference calls and supports 5-way conference calling.
1. Initiate a Conference Call:
ƒ
ƒ
ƒ
ƒ
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 step 2 and 3 for all parties that you want to join the conference. This can be done
at any time, thus also if a n
2. Cancel Conference: Canceling establishing conference call.
ƒ
If after pressing the “CONF” button, a user decides not to conference anyone, press
CONF again or the original LINE button.
ƒ
This will resume two-way conversation.
3. End Conference:
ƒ
ƒ
Press HOLD to end the conference call and put all parties on hold;
To speak with an individual party, select the corresponding blinking LINE.
CHECK VOICE MESSAGES (MESSAGE WAITING INDICATOR)
The 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.
ASTERISK BUSY LAMP FIELD
The seven multi-functional buttons can be configured for Asterisk Busy Field function with a specified
account. When Asterisk BLF is configured on one of the multi-functional buttons, the Speed Dial function
will work when it that line is not in use.
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CALL FEATURES
The GXP-2020 supports traditional and advanced telephony features including caller ID, caller ID w/name,
call forward/transfer/park/hold as well as intercom/paging and BLF.
Table 8: GXP-2020 Call Features
Key
*30
*31
*67
*82
*70
*71
*72
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
Disable Call Waiting (per Call)
Enable Call Waiting (per Call)
Unconditional Call Forward
Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wait for dial
tone. LCD will display “Call FWD Activated”.
*73
*90
Cancel Unconditional Call Forward: dial “*73” and get the dial tone, then hang up.
LCD will display “Call FWD Activated”.
Busy Call Forward
Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial
tone. Hang up.
*91
*92
Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
Delayed Call Forward
Dial “*92” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial
tone. Hang up. LCD will display “Call FWD Activated”.
*93
Cancel Delayed Call Forward
Dial “*93” for a dial tone, then hang up.
CUSTOMIZED LCD SCREEN & XML
With its graphical LCD screen, soft-keys and XML-support, the GXP-2020 is a highly customizable phone. XML
is used to communicate between web-servers and the GXP-2020 to dynamically update the phone’s
phonebook and idle screen logo and functions.
XML PHONEBOOK
The Grandstream GXP-2020 enables you to easily share and maintain a phonebook through the web. The
XML-phonebook must be stored on a web-server. For more information on how to setup the phone to download
the phonebook, please see the Configuration with Web Browser. For more information on how to create a
downloadable XML phonebook, detailed information and configuration guides are located on the website in the
CUSTOMIZABLE IDLE SCREEN AND SOFT-BUTTONS
The GXP-2020 also allows you to customize the idle-screen using your own logo on the display of the phone,
instead of the Grandstream logo. In addition to the logo, you can reprogram the soft-keys on the GXP-2020 for
your own customized applications. For more information about creating a custom idle screen and/or
reprogramming the soft-keys, please visit our website at: http://www.grandstream.com/resources.html.
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CONFIGURATION GUIDE
The GXP-2020 can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone;
secondly, through embedded web-configuration menu.
CONFIGURATION WITH KEY PAD
To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and
left/right. Press the OK button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button.
The phone automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if
left idle for 20 seconds.
Press the MENU button to enter the key the Key Pad Menu. The menu options available are listed in table 8.
Table 9: Key Pad Configuration Menu
Call History
Status
Displays histories of incoming, dialed and missed calls.
Displays the network status, account statuses, software version and
MAC-address of the phone.
Phone Book
Instant Messages
Direct IP call
Preference
Displays the phonebook
Goes to voice messages
Displays the IP-call options menu
Displays preference menu. Here ring tones, ring volume and XML screen
settings can be modified
Config
Display configuration menu. Here network, SIP and audio settings can be
modified. The FACTORY RESET function is stored here too.
Factory Functions
Extra diagnostic tools reside here.
Reboot
Makes the phone reboot itself.
Exit from this menu.
Exit
Display “Preference”
Press Menu button to enter this sub menu including
•
“Do NOT Disturb”
DND (Do NOT Disturb) function could be turned on or off in
the “DO NOT Disturb” menu.
•
•
Ring Tone
Choose different ring tones in the “Ring Tone” menu.
Ring Volume
Press Menu button to hear the selected ring volume, press
‘←’ or ’ →’ to hear and adjust the ring tone volume.
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.
Press Menu button to choose the menu item.
Press ‘←’ to return to the main menu.
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Display “Configure”
Press Menu button to display the configuration selections:
•
Network.
To enable/disable DHCP.
To setup IP-address, Net mask and Gateway address
•
SIP
To change SIP-server settings for primary account.
Audio
Upgrade
•
•
In this menu setting regarding the firmware server and config
server can be changed. It also enables the user to make the
phone attempt to download new firmware.
Factory Reset
•
Key in the physical/MAC address on back of the phone.
Press Menu button to reset FACTORY DEFAULT setting. Do not
use Factory Reset unless you want to restore factory settings
Press ‘←’ to return the main menu.
Display “Factory Functions”
Press Menu to display the factory function items including
•
Ethernet Loopback
Connect a cross Ethernet cable from your “PC” port, and the
“LAN” port. The test result is displayed on the screen. Use this
feature to diagnose the state of health of the RJ45 jacks. Press
Menu button to exit the diagnostic mode.
Note: Running the Ethernet Loopback mode with a normal
connection will cause IP loss.
•
•
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.
Press ‘←’ to return to the main menu.
Display “Reboot”
Press Menu button to reboot the device
Display “Exit”
Press Menu button to exit the menu
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Figure 7: Keypad GUI Call Flow
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CONFIGURATION WITH WEB BROWSER
The GXP-2020 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 phone1
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 the browser2
Enter the administrator’s password to access the Web Configuration Menu3
1
2
3
The Web-enabled computer has to be connected to the same sub-network as the phone. This can
easily be done by connecting the computer to the same hub or switch as the phone is connected to. In
absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the
phone using the PC-port on the phone.
If the phone is properly connected to a working Internet connection, the phone will display its IP
address. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. You
will need this number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60,
The default administrator password is “admin”; the default end-user password is “123”.
NOTE: When changing any settings, always SUBMIT them by pressing the button on the bottom of the
page. Reboot the phone to have the changes take effect. If, after having submitted some changes, more
settings have to be changed, press the menu option needed.
DEFINITIONS
This section will describe the options in the Web configuration user interface. As mentioned, a used can
log in as an administrator or end-user.
Functions available for the end-user are:
•
Status: Displays the network status, account statuses, software version and MAC-address of the
phone
•
Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings
can be set here.
Additional functions available to administrators are:
•
Advanced settings: To set advanced network settings, codec settings and XML configuration
settings.
•
Account (1-6): To configure each of the six SIP accounts.
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Table 10: Device Configuration - Status Definitions
Hardware Revision
MAC Address
IP Address
Hardware version number: Main Board, Interface Board
The device ID, in HEXADECIMAL format.
This field shows IP address of GXP-2020
This field contains the product model information.
Product Model
Software Version
•
•
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
System Up Time
Registered
This field shows system up time since the last reboot.
Indicates whether accounts are registered to the related SIP server(s). GXP-2020 can
support four unique SIP profiles.
PPPoE Link Up
Inidicates whether the PPPoE connection is enabled (connected to a modem).
Table 11: Device Configuration – Basic Settings Definitions
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
There GXP-2020 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-2020 acquires
its IP address from the first DHCP server it discovers on its LAN. The
DHCP option is reserved for NAT router mode. To use the PPPoE feature,
set the PPPoE account settings. The GXP-2020 establishes a PPPoE
session if any of the PPPoE fields are set.
2. Static IP mode: configure all of the following fields: IP address, Subnet
Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary). These fields are set to zero by default.
Multi purpose key 1-7
These options are used to assign a function to the corresponding multi purpose key.
Options available are: “Speed dial”, “Asterisk BLF”, “Presence Watcher” and
“Eventlist BLF”.
Each function is connected to one of the 6 accounts and has a target user ID.
Time Zone
This parameter controls the date/time display according to the specified time zone.
Daylight Savings Time
This parameter controls time displayed in daylight savings time or not. If set to
“Yes”, then the displayed time will be 1 hour ahead of normal time.
Time Display Format
Date Display Format
LCD time display in 12 hour or 24 hour format
Choose one of the following formats:
•
•
•
Year-Month-Day
Month-Day-Year
Day-Month-Year
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Advanced User configuration includes not only the end user configuration, but also advanced
configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous
configuration.
Table 12: Advanced User Configuration Page Definitions
Admin
Password
Administrator password. Only the administrator can configure the “Advanced Settings”
page. Password field is purposely blank for security reasons after clicking update and
saved. The maximum password length is 25 characters.
Silence
Suppression
This controls the silence suppression/VAD feature of the audio codec G.723 and G.729.
If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of
audio packets) will be sent during the period of no talking. If set to “No”, this feature is
disabled.
Voice Frames
per TX
This field contains the number of voice frames to be transmitted in a single Ethernet
packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500
byte (or 120kbps)).
When setting this value, be aware of the requested packet time (ptime, used in SDP
message) is a result of configuring this parameter. This parameter is associated with the
first codec in the above codec Preference List or the actual used payload type
negotiated between the 2 conversation parties at run time. e.g., if the first codec is
configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in
the SDP message of an INVITE request will be 60ms because each G.723 voice frame
contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or
G.711 or G.726, then the “ptime” value in the SDP message of an INVITE request will be
20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the IP phone
will use and save the maximum allowed value for the corresponding first codec choice.
The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20 (x10ms) frames;
for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames
respectively.
Please be careful when editing these parameters. Adjusting these parameters will also
change the dynamic jitter buffer. The GXP-2020 has a patent dynamic jitter buffer
handling algorithm. The jitter buffer range is 20 ~ 200 ms.
Grandstream recommends using the default settings provided. Grandstream does not
recommend adjusting these parameters if you are an average user. Incorrect settings
will affect the voice quality.
Please refer to the Codec FAQ at
http://www.grandstream.com/faqscodec.html for more technical detail.
Layer 3 QoS
Layer 2 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.
This contains the value used for layer 2 VLAN tag. Default setting is blank.
Default is 4 seconds.
No Key Entry
Timeout
Use # as
Dial Key
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.
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Local RTP port This parameter defines the local RTP-RTCP port pair used to listen and transmit. It is the
base RTP port for channel 0. When configured, channel 0 will use this port _value for
RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and
port_value+3 for its RTCP. The default value is 5004.
Use Random
Port
This parameter, when set to “Yes”, will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple GXP-2020s are behind the
same NAT. Default is No.
Keep-alive
interval
This parameter specifies how often the GXP-2020 sends a blank UDP packet to the SIP
server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
STUN Server
IP address or Domain name of the STUN server. STUN resolution result will display in
the STATUS page of the Web UI.
Firmware
Upgrade
This radio button will enable GXP-2020 to download firmware or configuration file
through either HTTP or local TFTP server. Choices are mutually exclusive.
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not
blank, the GXP-2020 will attempt to retrieve a new configuration file or new code image
from the specified TFTP server at boot time. It will make up to 3 attempts before timeout
and then it will start the boot process using the existing code image in the Flash memory.
If a TFTP server is configured and a new code image is retrieved, the new downloaded
image will be verified and then saved into the Flash memory.
Note: Grandstream strongly recommends that the user upgrade firmware locally in a
LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade
process (especially the power supply) as this will damage the device.
Via HTTP
Server
The HTTP server URL used for firmware upgrade and configuration via HTTP. For
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-2020 will only perform HTTP download once at
boot up.
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.
Phonebook
XML
Enable the XML phonebook via TFTP or HTTP. Define XML server path and download
speed.
Idle Screen XML Enable XML Idle Screen via TFTP or HTTP. Define XML server path.
DTMF Payload
Type
This parameter sets the payload type for DTMF using RFC2833. Default is 101.
Syslog Server
The IP address or URL of System log server. This feature is especially useful for ITSPs.
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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 following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the
following components: GS_LOG: [device MAC address][error code] error message
For example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000].
Ethernet link is up.
NTP server
This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve.
It is used to display the current date/time.
Distinctive Ring Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a particular
Tone
Caller ID. The GXP will ONLY use selected ring tones for particular Caller IDs. For all
other calls, 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.
Disable Call
Waiting
Default is No. If set to Yes, the call waiting will be disabled.
Use Quick IP
Call Mode
Dial an IP address under the same LAN/VPN segment by entering the last octet 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 0-9 and XXX <=255,
phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the
local IP address REGARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK). See Quick IP Call Mode
for details.
Lock keypad
update
If set to “Yes”, the configuration changes via keypad are disabled.
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INDIVIDUAL ACCOUNT SETTINGS
The GXP-2020 has 4 line appearances, each with an independent SIP account. Each SIP account
requires its own configuration page. Their configurations are identical.
Table 13: SIP Account Configuration Page Definitions
Account Active
This field indicates whether the account is active. The default value for the
primary account (Account 1) is Yes. The default value for the other two accounts
is No.
Account Name
SIP Server
The name associated with each account - displayed on LCD.
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
Border Controller. Used for firewall or NAT penetration in different network
environment. If the system detects symmetric NAT, STUN will not work. ONLY
outbound proxy can provide solution for symmetric NAT.
SIP User ID
User account information provided by VoIP service provider (ITSP); either an
actual phone number or formatted like one.
Authenticate ID
SIP service subscriber’s Authenticate ID used for authentication. It can be
identical to or different from SIP User ID.
Authenticate Password SIP service subscriber’s account password for GXP-2020 to register to (SIP)
servers of 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
Number
If the phone has an assigned PSTN telephone number, this field should be set to
“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
attached to the “From” header in SIP request
SIP Registration
This parameter controls sending REGISTER messages to the proxy server. The
default setting is “Yes”.
Un-register on Reboot
Register Expiration
Default is No. If set to “Yes”, the SIP user’s registration information will be
cleared on reboot.
This parameter allows user to specify the time frequency (in minutes) that GXP-
2020 refreshes its registration with the specified registrar. The default interval is
60 minutes. The maximum interval is 65,535 minutes (about 45 days).
Local SIP Port
This parameter defines the local SIP port used to listen and transmit. The default
value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
Account 4 respectively.
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NAT Traversal
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-2020 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.
Subscribe for MWI:
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail User ID
When configured, user can access messages by pressing “MSG” button. This ID
is usually the VM portal access number.
Send DTMF
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.
Early Dial
Default is No. Use only if proxy supports 484 response.
Dial Plan Prefix
Enable Call Features
Sets the prefix added to each dialed number.
Default is No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided ITSP support those features.
Session Expiration
The SIP Session Timer extension enables SIP sessions to be periodically
“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval
expires, if there is no refresh via a UPDATE or re-INVITE message, the session is
terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
Min-SE
Defines the minimum session expiration (in seconds). Default is 90 seconds.
Caller Request Timer
If set to “Yes”, the phone will use session timer when it makes outbound calls if
remote party supports session timer.
Callee Request Timer
Force 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.
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.
UAS Specify Refresher As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
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Force INVITE
Enable 100rel
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 inter-
networking..
Account Ring Tone
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.
Send Anonymous
Auto Answer
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.
Default is No. If set to “Yes”, GXP-2020 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.
Preferred Vocoder
The GXP-2020 supports up to 5 different Vocoder types including G.711(a/µ)
(also known as PCMU/PCMA), GSM, G.723.1, G.729A/B.
Configure Vocoders in a preference list that is included with the same preference
order in SDP message. Enter the first Vocoder in this list by choosing the
appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by
choosing the appropriate option in “Choice 8”.
Jitter Delay
Jitter Buffer Delay is pre-configured and can not be changed.
Special Feature
Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.
SAVING THE CONFIGURATION CHANGES
After the user makes a change to the configuration, press the “Update” button in the Configuration Menu.
The LCD will then display the following screen to confirm saved changes.
Grandstream recommends to 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 LCD
will then display the following screen to confirm reboot is underway.
Wait 30 seconds to log-in again.
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Software Upgrade & Customization
Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding
configuration settings are in the ADVANCED SETTINGS configuration page.
FIRMWARE UPGRADE THROUGH TFTP/HTTP
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set
to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are
examples of some valid URLs.
•
•
firmware.mycompany.com:6688/Grandstream/1.0.0.18
168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware: 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-2020 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 “Upgrade Server”
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 20 seconds in a controlled LAN or 2-3 minutes over the Internet.
Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever
possible.
GRANDSTREAM PUBLIC TFTP SERVER
For users who do not have local TFTP server, Grandstream provides a NAT-friendly TFTP server on the
public Internet for users to download the latest firmware upgrade automatically. Please check the
Support/Download section of our website to obtain this TFTP server IP address:
Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades.
customerFree.cfm.
Grandstream Networks, Inc.
GXP–2020 Users Manual
Page 31 of 33
Firmware 1.1.4.3
Last Updated: 05/2007
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INSTRUCTIONS TO CONFIGURE LOCAL TFTP SERVER:
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-2020 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 phone’s 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-2020 phone boots up, it will send TFTP or HTTP request to download configuration
file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP-2020 phone. This
file is for initial provisioning purpose only.
•
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-2020 can be configured via Web Interface as well as via Configuration File through TFTP or
HTTP. “Upgrade Server” 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 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding configuration template of the firmware.
Once the GXP-2020 boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”,
where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The configuration file
name should be in lower cases.
MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD
When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre-
scheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider
can manage and reduce the Firmware or Provisioning Server load at any given time.
Grandstream Networks, Inc.
GXP–2020 Users Manual
Page 32 of 33
Firmware 1.1.4.3
Last Updated: 05/2007
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Restore Factory Default Setting
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone.
Please backup or print all the settings before you restoring factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect to your VoIP service provider.
Disconnect network cable and power cycle the unit before resetting factory default settings.
1. Step 1: Press “OK” key to bring up the key pad configuration UI menu, select “Config”, press
“OK” to enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)
2. Step 2: Key in the MAC address printed on the bottom of the sticker. Please use the following
mapping:
a. 0-9: 0-9
b. A: 22 (press the “2” key twice, “A” will show on the LCD)
c. B: 222
d. C: 2222
e. D: 33 (press the “3” key twice, “D” will show on the LCD)
f. E: 333
g. F: 3333
For example: if the MAC address is 000b8200e395, it should be key in as “0002228200333395”.
NOTE: If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to
move the cursor or wait for 4 seconds to continue to key in another “2”.
3. Step 3: Press the “OK” key again to move the cursor to “OK” button. Press “OK” key again to
confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous
keypad menu interface.
Grandstream Networks, Inc.
GXP–2020 Users Manual
Page 33 of 33
Firmware 1.1.4.3
Last Updated: 05/2007
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