Grandstream Networks Telephone GXP 280 User Manual

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Table 8: LCD Buttons.................................................................................................................... 12  
Table 9: LCD Icons ....................................................................................................................... 12  
Table 10: GXP Keypad Buttons.................................................................................................... 14  
Table 11: GXP Call Features........................................................................................................ 21  
Table 12: Key Pad Configuration Menu........................................................................................ 42  
Table 13: Device Configuration - Status .......................................................................................46  
Table 14: Device Configuration – Basic Settings.......................................................................... 46  
Table 15: Advanced Settings........................................................................................................ 48  
Table 16: SIP Account Settings .................................................................................................... 53  
GUI INTERFACE EXAMPLES  
GXP USER MANUAL  
1. SCREENSHOT OF CONFIGURATION LOGIN PAGE  
2. SCREENSHOT OF STATUS PAGE  
3. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE  
4. SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE  
5. SCREENSHOT OF SIP ACCOUNT CONFIGURATION  
6. SCREENSHOT OF SAVED CONFIGURATION CHANGES  
7. SCREENSHOT OF REBOOT PAGE  
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Welcome  
Your Grandstream GXP Series IP phone features a new sophisticated design and is very easy to use.  
The GXP combines advanced feature functionality with the latest technology to offer excellent audio  
quality, ease of use, expandability, and broad interoperability with 3rd party SIP platforms. It is ideal for  
the enterprise customer.  
The GXP Series supports a broad range of codecs, security protection, PoE (except on GXP-280), dual  
10/100mbps Ethernet ports and are very easy to manage. Currently, the GXP Series consists of the  
following five models: GXP-280, GXP-1200, GXP-2000, GXP-2010 and GXP-2020. Each model delivers  
superior audio quality using either a handset, hands-free speakerphone or headset and supports multi-  
party conferencing, multi-languages, dual-color LEDs, presence and BLF (on most models). Large easy-  
to-read backlit graphical displays (except GXP-280) with multiple XML keys further enhance the user  
experience. Some models (GXP-2000, GXP2010 and GXP2020 currently) are expandable with one or  
two expansion module.  
The series is based on SIP standard and are interoperable with most 3rd party SIP platforms and open-  
source platforms.  
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation  
of this product in any way other than as detailed by this User Manual, could void your manufacturer  
warranty.  
Warning: Please do not use a different power adaptor with the GXP as it may cause damage to the  
products and void the manufacturer warranty.  
This document is contains links to Grandstream GUI Interfaces. Please download these examples  
This document is subject to change without notice. The latest electronic version of this user manual  
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,  
for any purpose without the express written permission of Grandstream Networks, Inc. is not  
permitted.  
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Installation  
EQUIPMENT PACKAGING  
Table 1: Equipment Packaging  
GXP-280  
GXP-1200  
Yes  
GXP-2000  
Yes  
GXP-2010  
Yes  
GXP-2020  
Yes  
Main Case  
Handset  
Phone Cord  
Yes  
Yes  
Yes  
Yes  
Yes  
No  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Power Adaptor  
Ethernet Cable  
High Phone Stand  
Low Phone Stand  
Wall Mount Spacers (2)  
Yes  
No  
Yes  
Yes  
Yes  
No  
No  
Yes  
No  
No  
Yes  
Yes  
Yes  
Yes  
CONNECTING YOUR PHONE  
The connectors of the GXP1200/2010/2020 are located on the bottom of the device while they are located on  
the back side of the GXP280/2000.  
Table 2: GXP Connectors  
Connects the GXP Extension unit directly to the GXP using connection cable.  
EXT  
PC  
Draws power from PoE if provided by network.  
10/100Mbps RJ-45 ports for PC (downlink) connection.  
10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af).  
Draws power from either spare line or signal line.  
LAN  
5V DC power port; UL Certified  
Power Jack  
RJ22 and 2.5mm for GXP-280/2010/2020  
RJ22 for GXP-1200  
2.5mm for GXP-2000 HW Rev1.0 or later  
Headset Jack  
Handset Jack  
RJ11  
GXP-2000 EXTENSION UNIT  
GXP–2000 supports two (2) extension units, providing up to 112 additional 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  
2) One PS2 cable  
3) One connection plate  
4) One Universal Power Adaptor  
FIGURE 1: CONNECTING THE GXP–2000 AND THE GXP–EXTENSION  
GXP–2000 w/GXP–Extension  
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 loose 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–Ext will turn to solid green.  
Note: Extension for GXP2010 and GXP2020 does not support hot-swap. Once connected, user should  
reboot the phone to ensure the set up will work correctly.  
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Figure 2: 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 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  
Table 3: GXP Product Models  
Overview  
Model  
Picture  
GXP280 is an entry-level SIP phone. It features:  
y
y
Single line  
Three soft keys  
GXP-280  
GXP1200 is an entry-level SIP phone. It features:  
y
y
Two lines  
Three soft keys  
GXP-1200  
GXP2000 is a mainstream SIP phone. It features:  
y
y
Four lines  
Seven programmable hard keys  
GXP-2000  
GXP-2010  
GXP2010 is a key system SIP phone. It features:  
y
y
y
Four lines  
Eighteen programmable hard keys  
Three XML programmable soft keys  
GXP2020 is an executive SIP phone. It features:  
y
y
y
Six lines  
Seven programmable hard keys  
Four XML programmable soft keys  
GXP-2020  
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Table 4: GXP Comparison Guide  
Features  
GXP-2020  
Table 5: GXP Key Features in a Glance  
Features  
Benefits  
Table 6: GXP 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 7: GXP Technical Specifications  
Lines  
Multiple direct lines with independent SIP accounts, programmable speed dial keys,  
XML programmable soft-keys. (except on GXP-280)  
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)  
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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 plan prefix, off-hook auto dial, auto answer,  
early dial and speed dial (on some models)  
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 SIP Enterprise Phone  
GETTING FAMILIAR WITH THE LCD  
GXP-2xxx has a dynamic and customizable screen. The screen displays differently depending on whether  
the phone is idle or in use (active screen).  
Table 8: LCD Buttons  
Key Button  
Key Button Definitions  
LINE SELECTORS  
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 check page 24.  
Shows the status of the phone and network. It will indicate whether the network is  
down, starting or is running (show IP-number). Other messages such as “DO NOT  
DISTURB” or “## MISSED CALLS” are shown here too.  
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  
SOFT-BUTTONS  
(Excluding GXP-2000)  
MISSED CALLS This option shows up there were unanswered calls to this  
phone. The MissedCalls option shows a list of the missed  
calls  
CALL RETURN Calls the phone that called/tried to call your phone last.  
REDIAL  
Redials the last number  
Hangs up phone  
END CALL  
Table 9: LCD Icons  
Icon  
LCD Icon Definitions  
Connectivity Status / SIP Proxy/Server Icon:  
Solid – connected to SIP Server/IP address received  
Blinking – physical connection failed  
Blank – SIP Proxy/Server not registered  
Phone Status Icon:  
OFF when the handset is on-hook  
ON when the handset is off-hook  
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Speaker Phone Status Icon:  
FLASH when phone rings or a call is pending  
OFF when the speakerphone is off  
ON when the speakerphone is on  
DND Icon:  
ON when the “do not disturb” is activated  
Activate by pressing MUTE/DEL button once  
Calls Forwarded Icon:  
INDICATES calls are forwarded  
Follow ‘call forwarding’ procedures  
Handset, Speakerphone and Ring Volume Icon:  
Each icon appears next to the volume icon  
To adjust volume, use the up/down button  
Real–time Clock:  
Synchronized to Internet time server  
Time zone configurable via web browser  
AM/PM indicator  
AM  
PM  
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TABLE 10: GXP KEYPAD BUTTONS  
Key Button  
LINE BUTTONS  
TRANSFER  
CONF  
Key Button Definitions  
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  
HOLD  
MSG  
Place ACTIVE call on hold  
Enter to retrieve voice mails or other messages  
Enable/Disable hands-free speaker mode  
Press SEND to dial a new number or redial the last number dialed. Press  
send button to send a call immediately before “no key entry timeout” value  
expires  
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, *, #  
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  
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MAKING PHONE CALLS  
Handset, Speakerphone and Headset Mode  
Handset can be toggled between Speaker and Headset. To switch between Handset and Speaker/Headset,  
press the Hook Flash in the handset cradle or press the SPEAKER button.  
Multiple SIP Accounts and Lines  
GXP can support up to six independent SIP accounts depending on the product model. Each account is  
capable of independent SIP server, user and NAT settings. Each of the line buttons is “virtually” mapped to  
an individual SIP account. The name of each account is conveniently printed next to its corresponding  
button. In off-hook state, select an idle line and the name of the account (as configured in the web interface)  
is displayed on the LCD and a dial tone is heard.  
For example: Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as “VoIP 1”, “VoIP 2”,  
respectively and ensure that they are active and registered. When LINE1 is pressed, you will hear a dial  
tone and see “VoIP 1” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see “VoIP 2”  
on the LCD display.  
To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. User can  
switch lines before dialing any number by pressing the same LINE button one or more times. If you continue  
to press a LINE button, the selected account will circulate among the registered accounts.  
For example: when LINE1 is pressed, the LCD displays “VoIP 1”; If LINE1 is pressed twice, the LCD  
displays “VoIP 2” and the subsequent call will be made through SIP account 2.  
Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When the  
“virtually” mapped line is in use, the GXP will flash the next available LINE (from left to right or from top to  
bottom for Multi Purpose Keys) in red. A line is ACTIVE when it is in use and the corresponding LED is red.  
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.  
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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  
4. USING THE CALL HISTORY:  
To call  
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” 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 automatically call the assigned extension.  
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Note: The multi-functional buttons will function as LINE keys when all LINEs are busy. The LED will flash in  
red to indicate an incoming call. Press the button to pick up the call. If any one of the Multi Purpose Keys is  
associated with a call, the button’s speed dial/BLF function will not work.  
Making Calls using IP Addresses  
Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls  
can be made between two phones if:  
Both phones have public IP addresses, or  
Both phones are on a same LAN/VPN using private or public IP addresses, or  
Both phones can be connected through a router using public or private IP addresses (with necessary  
port forwarding or DMZ)  
To make a direct IP call, please follow these steps:  
1. Press MENU button to bring up MAIN MENU.  
2. Select “Direct IP Call” using the arrow-keys.  
3. Press OK to select.  
4. Input the 12-digit target IP address. (Please see example below).  
5. Press OK key to initiate call.  
To make a quick IP call, please see next section.  
For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input  
the following: 192*168*1*60#5062 - The “ * ” key represent the dot“.” ; The “#” key represent colon “:”.  
Press OK to dial out.  
Quick IP Call Mode  
The GXP also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the  
last few digits (last octet) of the target 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” or “MUTE” button if you do not want to take a call. This will send the caller directly  
to voicemail.  
2. Press the “DND” or “MUTE” button to set phone to ‘do not disturb’ (icon will be on the screen). The  
phone will not ring and send caller directly to voicemail. (see note above)  
PHONE FUNCTIONS DURING A PHONE CALL  
Call Waiting/ Call Hold  
1. Hold: Place a call on ‘hold’ by pressing the “HOLD” button.  
2. Resume: Resume call by pressing the corresponding blinking LINE.  
3. Multiple Calls: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE to  
place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.  
Mute/Delete  
1. Press the MUTE button to enable/disable muting the microphone.  
2. The “Line Status Indicator” will show “LINEx: SPEAKING” or “LINEx: MUTE” to indicate whether the  
microphone is muted.  
NOTE: Pressing MUTE button for an incoming call will reject the call. MUTE button also functions as delete  
key when user wishs to delete the last entered digit.  
Call Transfer  
GXP supports both Blind and Attended (or supervised) transfer:  
1. Blind Transfer: Press “TRANSFER (or TRNF for GXP-2000)” button, then dial the number and  
press the “SEND” button to complete transfer of active call.  
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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 “TRANSFER (or 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.  
5-Way Conferencing  
GXP can host conference calls and supports up to 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.  
NOTE: The party that starts the conference call has to remain in the conference for its entire duration, you  
can put the party on mute but it must remain in the conversation.  
Voice Messages (Message Waiting Indicator)  
A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Press the MSG button to  
retrieve the message. An IVR will prompt the user through the process of message retrieval. Press a  
specific LINE to retrieve messages for a specific line account.  
NOTE:  
Each line has a separate voicemail account. Each account requires a voicemail portal number to be  
configured in the “voicemail user id” field.  
To check which line account has a message 1) press the message button (this always checks the  
primary account), 2) check each line for stutter tone or 3) check missed calls using the menu.  
Busy Lamp Field  
The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account.  
When BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when that  
line is not in use. Call Pick Up is supported when user presses a flashing BLF key.  
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CALL FEATURES  
The GXP supports traditional and advanced telephony features including caller ID, caller ID w/name, call  
forward/transfer/park/hold as well as intercom/paging and BLF.  
Table 11: GXP Call Features  
Key  
*30  
*31  
*67  
*82  
*50  
*51  
*70  
*71  
*72  
Call Features  
Block Caller ID (for all subsequent calls)  
Send Caller ID (for all subsequent calls)  
Block Caller ID (per call)  
Send Caller ID (per call)  
Disable Call Waiting (for all subsequent calls)  
Enable Call Waiting (for all subsequent calls)  
Disable Call Waiting (per Call)  
Enable Call Waiting (per Call)  
Unconditional Call Forward  
Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wait for dial  
tone. LCD will display “Call FWD Activated”.  
*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  
Grandstream GXP Series phones support both simple and advanced XML applications: 1) XML Custom Screen, 2)  
XML Downloadable Phonebook and 3) Advanced XML Survey Application. For more information on how to create  
a downloadable XML phonebook, creating a custom idle screen and/or reprogramming the soft-keys on GXP-  
1200/GXP-2010/GXP2020, please visit our website at: http://www.grandstream.com/resources.html.  
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Configuration Guide  
The GXP can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly,  
through embedded web-configuration menu.  
CONFIGURATION VIA KEYPAD  
To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and left/right.  
Press the OK button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button. The phone  
automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if left idle for 20  
seconds.  
Press the MENU button to enter the key the Key Pad Menu. The menu options available are listed in table 8.  
Table 12: Key Pad Configuration Menu  
Call History  
Status  
Displays histories of incoming, dialed and missed calls.  
Displays the network status, account statuses, software version and  
MAC-address of the phone.  
Phone Book  
Displays the phonebook  
LDAP Directory  
Instant Messages  
Direct IP call  
Displays the LDAP directory  
Goes to voice messages  
Displays the IP-call options menu  
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.  
LCD Contrast  
LCD Brightness  
Download SCR XML  
The phone will download the custom idle screen (if available)  
Erase Custom SCR  
Custom idle screen will be erased and will be replaced with  
default Grandstream logo.  
Display Language  
You can choose English, Chinese or Secondary Language  
Press Menu button to choose the menu item.  
Press ‘’ to return to the main menu.  
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Configure  
Press Menu button to display the configuration selections:  
Network.  
To enable/disable DHCP.  
To setup IP-address, Net mask and Gateway address  
SIP  
To change SIP-server settings for primary account.  
Upgrade  
In this menu setting regarding the firmware server and Config  
server can be changed. It also enables the user to make the  
phone attempt to download new firmware.  
Factory Reset  
Key in the physical/MAC address on back of the phone.  
Press Menu button to reset FACTORY DEFAULT setting. Do not  
use Factory Reset unless you want to restore factory settings  
Layer 2 QoS  
Configure Vlan Tags  
Press ‘’ to return the main menu.  
Factory Functions  
Press Menu to display the factory function items including  
Audio Loopback  
Speak into the handset. If you hear your voice in the handset,  
your audio works fine.  
Press Menu button to exit the mode.  
Diagnostic Mode  
All LEDs will light up  
Press any key on the keypad, to display the button name in the  
LCD. Lift and put back the handset or press Menu button to exit  
the diagnostic mode.  
Enable WDT  
Toggles the status of the Watchdog Timer.  
Press ‘’ to return to the main menu.  
Reboot  
Exit  
Press Menu button to reboot the device  
Display “Exit”  
Press Menu button to exit the menu  
Exit from this menu.  
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FIGURE 3: KEYPAD GUI FLOW  
Call History  
Any of previous menus  
Answered Calls  
Dialed Calls  
Missed Calls  
Back  
Back  
Clear All  
MENU  
New Entry  
Phone Book  
Name:  
New Entry  
Number:  
Download Phonebook XML  
Back  
Acct:  
Confirm Add:  
Cancel & Return:  
LDAP Directory  
Search Configuration  
View Directory  
Download Directory  
Search Configuration  
Back  
Call History  
Select Filter  
Filter Value  
Back  
Status  
Instant Message  
Do Not Disturb  
Clear All  
Back  
Phone Book  
LDAP Directory  
Enable DND  
Disable DND  
Back  
Preference  
Ring Tone  
Do Not Disturb  
Ring Tone  
Ring Volume  
Network  
IP Setting  
Instant  
Message  
Default Ring  
Ring1  
Ring2  
Ring 3  
Back  
LCD Contrast  
LCD Brightness  
Download SCR XML  
Erase Custom SCR  
Display Language  
Back  
IP  
NetMask  
Gateway  
DNS Server  
1
Direct IP Call  
Preference  
Config  
LCD Brightness  
DNS Server  
Active  
Idle  
Back  
SIP  
Config  
Account  
Network  
SIP  
Upgrade  
Factory Reset  
Layer 2 QoS  
Back  
SIP Proxy  
Outbound  
Proxy  
SIP User ID  
SIP Auth ID  
SIP Password  
SIP Transport  
Audio  
Factory  
Functions  
Reboot  
Exit  
Display Language  
Factory Function  
Save  
English  
Chinese  
Audio Loopback  
Diagnostic Mode  
Enable WDT  
Back  
Upgrade  
Secondary Language  
Language File Postfix  
Back  
Firmware  
Server  
Config Server  
Upgrade Via  
Diagnostic Mode  
Layer 2 QoS  
Keypad/LED Diagnostic  
802.1Q/VLAN Tag  
Priority value  
Reset Vlan Config  
Back  
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CONFIGURATION VIA WEB BROWSER  
The GXP embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages  
allow a user to configure the IP phone through a Web browser such as 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, please use  
http://192.168.0.60” in the address bar your browser.  
The default administrator password is “admin”; the default end-user password is “123”.  
NOTE: When changing any settings, always SUBMIT them by pressing the button on the bottom of the page.  
Reboot the phone to have the changes take effect. If, after having submitted some changes, more settings  
have to be changed, press the menu option needed.  
Definitions  
This section will describe the options in the Web configuration user interface. As mentioned, a used can log  
in as an administrator or end-user.  
Functions available for the end-user are:  
Status: Displays the network status, account statuses, software version and MAC-address of the  
phone  
Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings can  
be set here.  
Additional functions available to administrators are:  
Advanced Settings: To set advanced network settings, codec settings and XML configuration  
settings.  
Account X: To configure each of the SIP accounts.  
EXT X: To configure setting on extension module  
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Table 13: Device Configuration - Status  
MAC Address  
IP Address  
The device ID, in HEXADECIMAL format.  
This field shows IP address of GXP  
Product Model  
Part Number  
Software Version  
This field contains the product model information.  
This field contains the product part number  
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  
System Time  
Registered  
This field shows system up time since the last reboot.  
This field shows the current time on the phone system.  
Indicates whether accounts are registered to the related SIP server(s). GXP can  
support four unique SIP profiles.  
PPPoE Link Up  
Indicates whether the PPPoE connection is enabled (connected to a modem).  
Table 14: Device Configuration – Basic Settings  
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 operates in two modes:  
1. DHCP mode: all the field values for the Static IP mode are not used (even  
though they are still saved in the Flash memory.) The GXP acquires its IP  
address from the first DHCP server it discovers on its LAN. The DHCP  
option is reserved for NAT router mode. To use the PPPoE feature, set the  
PPPoE account settings. The GXP establishes a PPPoE session if any of  
the PPPoE fields is set.  
2. Static IP mode: configure all of the following fields: IP address, Subnet  
Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2  
(secondary). These fields are set to zero by default.  
Multi Purpose Key X  
These options are used to assign a function to the corresponding multi purpose key.  
Options available are:  
1. “Speed Dial”.  
2. “BLF” (Busy Lamp Field). This option has to be supported on the PBX and it  
indicates the status of the extension. The three possible states are idle  
(green), busy (red), ringing (blinking red).  
3. “Presence Watcher”. This option has to be supported by a presence server  
and it is tied to the “Do not disturb” status of the phone.  
4. “Eventlist BLF”. This option is similar to the BLF option but in this case the  
PBX collects the information from the phones and sends it out in one single  
notify message.  
Each function is connected to one of the accounts and has a target user ID.  
Time Zone  
This parameter controls the date/time display according to the specified time zone.  
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LCD Backlight Always  
On  
Turn on LC backlight at all times. Default is No. This option applies to GXP-  
1200/GXP-2000 only.  
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  
This option applies to GXP280/GXP-1200/GXP-2000 only.  
Display Clock instead of Choose to display clock or date on LCD. This option applies to GXP-280/GXP-  
Date  
1200/GXP-2000 only.  
This parameter controls time displayed in daylight savings time. If set to “Yes”, then the  
displayed time will be 1 hour ahead of normal time.  
Daylight Savings 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
o
o
month: 1,2,3,..,12 (for Jan, Feb, .., Dec)  
day: [+|-]1,2,3,..,31  
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.  
hour: hour (0-23), minute: minute (0-59)  
o
If “weekday” is 0, it means the date to start or end daylight saving is at exactly the given date.  
In that case, the “day” value must not be negative. If “weekday” is not zero and “day” is  
positive, then the daylight saving starts on the first “day” the iteration of the weekday (e.g.:  
1st Sunday, 3rd Tuesday etc). If “weekday” is not zero and “day” is negative, then the  
daylight saving starts on the last “day” the iteration of the weekday (e.g.: last Sunday, 3rd last  
Tuesday etc).  
The saving is in the unit of minutes. The saving time may also be preceded by a negative (-)  
sign if subtraction is desired instead of addition.  
The default value is set for US, the “Automatic Daylight Saving Time Rule” shall be set to  
“3,2,7,2,0;11,1,7,2,0;60”  
Examples  
US/Canada where daylight saving time is applicable:  
03,02,7,02,00;11,1,7,02,00;60  
This means the daylight saving time starts from the second Sunday of March at 2AM and  
ends the first Sunday of November at 2AM. The saving is 60 minutes.  
LCD Backlight  
Brightness  
Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means  
the brightest. For GXP2010 and GXP2020.  
LCD Contrast  
Set LCD contrast. Range from 0 to 20. Not for GXP280  
Disable in-call DTMF  
display  
Default is No. This field is used to hide the keypad input during a call.  
Mute Speaker Ringer  
in Headset Mode  
Default is No. This field lets user to choose whether to ring the phone Speaker  
when headset is connected.  
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Disable Missed Call  
Backlight  
Default is No. By default, LCD backlight will lit whenever there is a missed call.  
Not for GXP280.  
HEADSET Key Mode  
Set Default mode or choose Toggle Headset/Speaker. Not for GXP2000  
Advanced User configuration includes not only the end user configuration, but also advanced configuration  
such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.  
Table 15: Advanced Settings  
Admin  
Password  
Administrator password. Only the administrator can access the “Advanced Settings” and  
“Account Settings” page. Password field is purposely blank for security reasons after  
clicking update and saved. The maximum password length is 25 characters.  
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.  
iLBC payload  
type  
Payload type for iLBC. Default value is 97. The valid range is between 96 and 127.  
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.  
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Voice Frames  
per TX  
This field contains the number of voice frames to be transmitted in a single Ethernet  
packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500  
byte (or 120kbps)).  
When setting this value, be aware of the requested packet time (ptime, used in SDP  
message) is a result of configuring this parameter. This parameter is associated with the  
first codec in the above codec Preference List or the actual used payload type  
negotiated between the 2 conversation parties at run time. E.g., if the first codec is  
configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in  
the SDP message of an INVITE request will be 60ms because each G.723 voice frame  
contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or  
G.711 or G.726, then the “ptime” value in the SDP message of an INVITE request will be  
20ms.  
If the configured voice frames per TX exceeds the maximum allowed value, the IP phone  
will use and save the maximum allowed value for the corresponding first codec choice.  
The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20 (x10ms) frames;  
for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames  
respectively.  
Please be careful when editing these parameters. Adjusting these parameters will also  
change the dynamic jitter buffer. The GXP has a patent dynamic jitter buffer handling  
algorithm. The jitter buffer range is 20 ~ 200 ms.  
Grandstream recommends using the default settings provided. Grandstream does not  
recommend adjusting these parameters if you are an average user. Incorrect settings  
will affect the voice quality.  
Please refer to the Codec FAQ at  
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.  
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 GXPs are behind the same  
NAT. Default is No.  
Keep-alive  
interval  
This parameter specifies how often the GXP sends a blank UDP packet to the SIP  
server in order to keep the “hole” on the NAT open. Default is 20 seconds.  
Use NAT IP  
NAT IP address used in SIP/SDP message. Default is blank.  
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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 and  
Provisioning  
Default method is HTTP. Firmware upgrade may take up to 10 minutes depending on  
network environment. Do not interrupt the firmware upgrading process.  
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not  
blank, the GXP will attempt to retrieve a new configuration file or new code image from  
the specified TFTP server at boot time. It will make up to 3 attempts before timeout and  
then it will start the boot process using the existing code image in the Flash memory. If a  
TFTP server is configured and a new code image is retrieved, the new downloaded  
image will be verified and then saved into the Flash memory.  
Note: Grandstream strongly recommends that the user upgrade firmware locally in a  
LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade  
process (especially the power supply) as this will damage the device.  
Via HTTP  
Server  
The HTTP server URL used for firmware upgrade and configuration via HTTP. For  
Here “:6688” is the specific TCP port that the HTTP server is using; omit if using default  
port 80.  
Note: If Auto Upgrade is set to No, GXP will only perform HTTP download once at boot  
up.  
Config Server  
Path  
IP address or domain name of firmware server.  
Firmware File  
Prefix/Postfix  
Default is blank. If configured, GXP will request the firmware file with the prefix/postfix.  
This setting is useful for ITSPs. End user should keep it blank.  
Config File  
Prefix/Postfix  
Default is blank. End user should keep it blank.  
Allow DHCP  
Option 66 to  
override server  
Default is Yes. This allows device gets 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.  
Automatic  
Upgrade  
This function is used by ITSP. End user should NOT touch these parameters.  
Default is No. Choose “Yes” to enable automatic HTTP upgrade and provisioning.  
In “Check for upgrade every” field, enter the number of minutes to check the HTTP  
server for firmware upgrade or configuration changes. When set to “No”, the phone will  
only perform HTTP upgrade and configuration check once at boot up.  
LDAP Directory IP address or domain name of LDAP script server  
Phonebook  
XML  
Enable the XML phonebook via TFTP or HTTP. Define XML server path and download  
interval. When the user downloads the XML phone the manually entered or edited  
entries will not be deleted unless this option is selected to Yes.  
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Idle Screen XML Enable XML Idle Screen download via TFTP or HTTP. Define XML server path.  
Download  
XML  
Application  
Enter server path for XML application. This option applies to GXP-2020 only.  
To configure a User ID/extension to dial autpomatically when the phone is taken offhook.  
This parameter sets the payload type for DTMF using RFC2833. Default is 101.  
Offhook Auto  
Dial  
DTMF Payload  
Type  
Onhook  
Threshhold  
It determines the time handset has to be down to be recognized it’s onhook. Default is  
800ms. For GXP280 only.  
Syslog Server  
Syslog Level  
The IP address or URL of System log server. This feature is especially useful for ITSPs.  
Select the ATA to report the log level. Default is NONE. The level is one of DEBUG,  
INFO, WARNING or ERROR. Syslog messages are sent based on the following events:  
product model/version on boot up (INFO level)  
NAT related info (INFO level)  
sent or received SIP message (DEBUG level)  
SIP message summary (INFO level)  
inbound and outbound calls (INFO level)  
registration status change (INFO level)  
negotiated codec (INFO level)  
Ethernet link up (INFO level)  
SLIC chip exception (WARNING and ERROR levels)  
memory exception (ERROR level)  
The Syslog uses USER facility. In addition to standard Syslog payload, it contains the  
following components: GS_LOG: [device MAC address][error code] error message  
For example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000].  
Ethernet link is up.  
NTP server  
This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve.  
It is used to display the current date/time.  
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.  
System Ring  
Tone  
System ring tone. Default is North American standard.  
Adjust system ring tone frequencies and cadences based on local telecom standard.  
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Call Progress  
Tones  
Using these settings, users can configure ring or tone frequencies based on 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 in 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 be zero. Otherwise it will ring ON ms and a pause  
of OFF ms and then repeat the pattern. Up to three cadences are supported.  
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]  
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.  
Default is No. If set to Yes, the call waiting tone will be disabled.  
Default is No. If set to Yes, direct IP calls will be disabled  
Disable Call  
Waiting Tone  
Disable Direct  
IP Calls  
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.  
Disable  
Default is No. If set to Yes, conference will be disabled.  
Conference  
Lock keypad  
update  
If set to “Yes”, the configuration changes via keypad are disabled.  
Default is No. If set to “Yes”, Muti-Purpose keys can be sent as DTMF.  
Enable MPK  
Sending DTMF For GXP2020/2010/2000.  
Disable DND  
Default is No. If set to “Yes”, the “DND” button on keypad will be disabled. For GXP2000,  
MUTE/DEL button functions as DND button when pressed while phone is idle.  
Headset Port  
Type  
Select either 2.5mm or RJ22 headset ports to be adjusted.  
Headset TX  
gain (dB)  
Increases the selected headset’s (2.5mm or RJ22) TX gain by + or – 6dB. Default is 0dB  
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Headset RX  
gain (dB)  
Increases the selected headset’s (2.5mm or RJ22) RX gain by + or – 6dB. Default is 0dB  
Display  
Language  
Allows user to choose preferred display language in web UI and key pad UI.  
The user can only load one secondary language  
GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires its  
own configuration page. Their configurations are identical.  
Table 16: SIP Account Settings  
Account Active  
This field indicates whether the account is active. The default value for the  
primary account (Account 1) is Yes. The default value for the other two accounts  
is No.  
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 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  
refreshes its registration with the specified registrar. The default interval is 60  
minutes. The maximum interval is 65,535 minutes (about 45 days).  
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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.  
SIP Registration Failure Retry registration if the process failed. Default is 20 seconds.  
Retry Wait Time  
SIP T1 Timeout  
SIP T2 Interval  
SIP Transport  
RFC 3261 SIP T1 timer. Default is 1 second.  
RFC 3261 SIP T2 timer. Default is 0.5 seconds.  
Choose SIP Transport between UDP and TCP. Default is UDP.  
Use RFC3581  
Symmetric Routing  
Default No. When selected the phone will follow the routing procedures specified  
in RFC3581.  
NAT Traversal (STUN)  
This parameter activates the NAT traversal mechanism. If activated (by choosing  
“Yes”) and a STUN server is also specified, the phone performs according to the  
STUN client specification. Using this mode, the embedded STUN client detects if  
and what type of NAT/Firewall configuration is used. If the detected NAT is a Full  
Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped  
public IP address and port in all of its SIP and SDP messages. If the NAT  
Traversal field is set to “Yes” with no specified STUN server, the GXP will  
periodically (every 20 seconds or so) send a blank UDP packet (with no payload  
data) to the SIP server to keep the “hole” on the NAT open.  
Subscribe for MWI:  
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication  
will be sent periodically.  
PUBLISH for Presence Enable Presence feature.  
Proxy-Require  
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.  
Voice Mail UserID  
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 responses.  
Sets the prefix added to each dialed number.  
Dial Plan Prefix  
Time waited before the call is forward to a number or VM.  
Default is 20 seconds.  
Delayed Call Forward  
Wait Time  
Enable Call Features  
Default is No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are  
supported locally provided ITSP support those features.  
Call Log  
User can choose to disable Call Log and what kind of calls to log.  
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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.  
Ring Timeout  
Caller Request Timer  
Defines how long ring will ring when receiving a call. Default is 60 seconds.  
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.  
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  
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:[email protected]>” in the From Header or P-  
Asserted-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 incoming call. Set to Intercom/Paging mode, it will answer the call based on  
the SIP info header from the server.  
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Allow Auto Answer by  
Call-Info  
If the Call-Info header contains answer-after=0, the call be answered  
automatically (so called paging mode).  
Turn off speaker on  
remote disconnect  
When BYE is received, the phone will turn off its speaker 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.  
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the  
transferred target’s Contact header information.  
Refer-To Use Target  
Contact  
Disable Multiple Media Default is No.  
Attribute in SDP  
Preferred Vocoder  
GXP supports up to 7 different Vocoder types including G.711(a/µ) (also known  
as PCMU/PCMA), GSM, G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).  
Configure Vocoders in a preference list that is included with the same preference  
order in SDP message. Enter the first Vocoder in this list by choosing the  
appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by  
choosing the appropriate option in “Choice 8”.  
SRTP Mode  
Enable SRTP mode based on selection. Default is No.  
eventlist BLF URI  
If a server supports this feature, user needs to configure an "eventlist BLF" URI  
on the service side (i.e.: [email protected])  
On the GXP, under Account page, fill in the ""eventlist BLF" field with the URI  
without the domain. (i.e.: BLF1006). Under Basic Settings, please select "eventlist  
BLF", choose account nubmer, monitored number, etc.  
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 web browser will then display a message window to confirm saved changes.  
Grandstream recommends reboot or power cycle the IP phone after saving changes.  
REBOOTING THE PHONE REMOTELY  
Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The web  
browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in  
again.  
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Software Upgrade & Customization  
Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configuration  
settings are in the ADVANCED SETTINGS configuration page.  
FIRMWARE UPGRADE THROUGH TFTP/HTTP  
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set to  
a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples  
of some valid URLs.  
firmware.mycompany.com:6688/Grandstream/1.1.6.37  
168.75.215.189  
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web  
Configuration Interface.  
Key Pad Menu  
To configure the Upgrade Server via Key Pad Menu options, select “Config” from the Main Menu, then select  
“Upgrade”. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN  
format. Choose “Save and use TFTP” or “Save and use HTTP” to select upgrade method. Select “Reboot”  
from the Main Menu to reboot the phone.  
Web Configuration Interface  
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXP  
IP address. Enter the admin password to access the web configuration interface. In the ADVANCED  
SETTINGS page, enter the Upgrade 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 Server  
For users who do not have a local TFTP server, Grandstream provides a NAT-friendly TFTP server on the  
public Internet for users to download the latest firmware upgrade automatically. Please check the  
Support/Download section of our website to obtain this TFTP server IP address:  
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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Instructions for local TFTP Upgrade:  
1. Unzip the file and put all of them under the root directory of the TFTP server.  
2. The PC running the TFTP server and the GXP should be in the same LAN segment.  
3. Go to File -> Configure -> Security to change the TFTP server's default setting from  
"Receive Only" to "Transmit Only" for the firmware upgrade.  
4. Start the TFTP server, in the 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 phone boots up, it will send TFTP or HTTP request to download configuration file  
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP phone. This file is for  
provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following error messages in  
a TFTP or HTTP server log can be ignored: TFTP Error from [IP ADRESS] requesting  
cfg000b82023dd4 : File does not exist. Configuration File Download”  
CONFIGURATION FILE DOWNLOAD  
The GXP can be configured via Web Interface as well as via Configuration File through TFTP or HTTP.  
“Config Server Path” is the TFTP or HTTP server path for the configuration file. It needs to be set to a valid  
URL, either in FQDN or IP address format.  
A configuration parameter is associated with each particular field in the web configuration page.  
A
parameter consists of a Capital letter P and 2 to 4 digit numeric numbers. i.e., P2 is associated with “Admin  
Password” in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the  
corresponding configuration template of the firmware.  
Once the GXP boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”, where  
“xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The configuration file name  
should be in lower cases.  
Managing Firmware and Configuration File Download  
When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in minutes,  
default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre-scheduled  
time intervals. By defining different intervals in P193 for different devices, a Server Provider can manage  
and reduce the Firmware or Provisioning Server load at any given time.  
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Restore Factory Default Setting  
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone.  
Please backup or print all the settings before you restoring factory default settings. Grandstream is not  
responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.  
INSTRUCTIONS FOR RESTORATION:  
Step 1: Press “OK” button to bring up the keypad configuration menu, select “Config”, press “OK” to  
enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)  
Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following mapping:  
0-9: 0-9  
A:  
B:  
22 (press the “2” key twice, “A” will show on the LCD)  
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 should be key in as “0002228200333395”.  
NOTE: If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to  
move the cursor or wait for 4 seconds to continue to key in another “2”.  
Step 3: Press the “OK” button to move the cursor to “OK”. Press “OK” button again to confirm. If the  
MAC address is correct, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.  
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