Grandstream GXP2100
Grandstream GXP2100
Grandstream GXP2100
WELCOME ................................................................................................................................................................. 4
INSTALLATION......................................................................................................................................................... 5
EQUIPMENT PACKAGING ............................................................................................................................................. 5
CONNECTING YOUR PHONE ........................................................................................................................................ 5
GXP2120/2110 EXTENSION UNIT ............................................................................................................................... 5
SAFETY COMPLIANCES ................................................................................................................................................ 7
WARRANTY ................................................................................................................................................................. 7
PRODUCT OVERVIEW ............................................................................................................................................ 8
USING THE GXP21XX SIP ENTERPRISE PHONE ........................................................................................... 12
GETTING FAMILIAR WITH THE LCD ..........................................................................................................................12
MAKING PHONE CALLS ............................................................................................................................................. 16
ANSWERING PHONE CALLS ....................................................................................................................................... 19
PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 19
CALL FEATURES ........................................................................................................................................................ 21
CUSTOMIZED LCD SCREEN & XML ......................................................................................................................... 22
CONFIGURATION GUIDE ...................................................................................................................................... 22
CONFIGURATION VIA KEYPAD .................................................................................................................................. 22
CONFIGURATION VIA WEB BROWSER ...................................................................................................................... 25
SAVING THE CONFIGURATION CHANGES ................................................................................................................... 39
REBOOTING THE PHONE REMOTELY ......................................................................................................................... 39
SOFTWARE UPGRADE & CUSTOMIZATION .................................................................................................. 40
FIRMWARE UPGRADE THROUGH TFTP/HTTP .......................................................................................................... 40
CONFIGURATION FILE DOWNLOAD ........................................................................................................................... 41
RESTORE FACTORY DEFAULT SETTING ....................................................................................................... 42
TABLE OF FIGURES
GXP21XX USER MANUAL
TABLE OF TABLES
GXP2120 USER MANUAL
The GXP21xx supports a broad range of codecs, security protection, PoE, dual 10/100mbps Ethernet
ports, along with customizable XML provisioning and application features. Users can expect superior
audio quality using the new high definition handset, hands-free speakerphone, or headset. Also, it can
support up to 5-way conferencing, multi-languages, dual-color LEDs, presence and Busy Lamp Field
(BLF). It presents a large easy-to-read backlit graphical display along with multiple XML keys to further
enhance the user experience. The GXP2120/2110 is also expandable with one to two expansion modules.
The GXP21xx is a perfect choice for enterprise users looking for a high quality, feature rich multi-line IP
phone with the best values.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation
of this product in any way other than as detailed by this User Manual, could void your manufacturer
warranty.
Warning: Please do not use a different power adaptor with the GXP as it may cause damage to the
products and void the manufacturer warranty.
• This document is contains links to Grandstream GUI Interfaces. Please download these examples
from http://www.grandstream.com/support/gxp_series/general/documents/gxp21xx_gui.zip for your
reference.
• This document is subject to change without notice. The latest electronic version of this user manual
is available for download @:
http://www.grandstream.com/support/gxp_series/general/documents/gxp21xx_usermanual_english.p
df
• 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.
The connectors of the GXP21xx are located on the bottom of the device.
Connects the GXP Extension unit directly to the GXP using connection cable.
EXT Draws power from PoE if provided by network. (Not applicable on GXP2100)
10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af).
LAN
Draws power from either spare line or signal line.
GXP2120/2110 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.
Connect the first GXP-EXT to the GXP2120/2110 using the connection 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 connection cable. The GXP2120/2110 will automatically reboot and
power up the GXP Extensions. Grandstream recommends, though not required, to use a separate power
supply with the second GXP-EXT.
NOTE: Should your system lose power, please unplug your devices and power up the GXP2120/2110 first.
NOTE:
1. Extension for GXP2120/2110 does not support hot-swap. Once connected, user should reboot the
phone to ensure the set up will work correctly.
2. GXP2120/2110 can drive 2 extension modules. Independent power adapters are not needed for
extension modules.
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.
Overview
Model Picture
GXP2120 is an executive SIP phone. It features:
y Six lines
y Seven programmable hard keys
GXP2120 y Four XML programmable soft keys
y Four lines
y Eighteen programmable hard keys
GXP2110 y Three XML programmable soft keys
y Four lines
y Seven programmable hard keys
GXP2100 y Three XML programmable soft keys
LAN Interface Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port with
(Ethernet ports) auto detection
Graphic LCD 320x160 pixel 240x120 pixel 180x90 pixel
Display
Lines Multiple direct lines with independent SIP accounts, programmable speed dial keys,
XML programmable soft-keys.
Protocol Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP,
Grandstream Networks, Inc. GXP21xx User Manual Page 9 of 42
Firmware 1.0.0.44 Last Updated: 10/2010
Support ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols, TR-
069, 802.1x.
Support multiple SIP accounts and up to 11 media channels concurrently
Support SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856,
3863) for use of MFKs, SIP Dialog package (RFC 4235)
Support for SIP MESSAGE method (RFC 3428)
GXP21xx has a dynamic and customizable screen. The screen displays differently depending on whether the
phone is idle or in use (active screen).
SIP PHONE Displays the available phone lines. Choose a phone line by pressing the corresponding line
LINES selector on the left-hand side.
DATE AND Displays the current date and time. Can be synchronized with Internet time servers.
TIME
Displays company logo. This logo can be customized. For more information on customizing the
LOGO
logo, please check page 24.
Shows the status of the phone and network. It will indicate whether the network is down, starting
NETWORK
or is running (show IP-number). Other messages such as “DO NOT DISTURB” or “## MISSED
STATUS
CALLS” are shown here too.
STATUS Shows the status of the phone, using icons as shown in the next table.
BAR
LINE Displays the name of the account that is in use. Select another account by pressing the LINE
STATUS SELECTOR BUTTONS
INDICATOR
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- • MISSED CALLS This option shows up there were unanswered calls to this phone. The
BUTTONS 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
• END CALL Hangs up phone
• CallPark When a GXP2120 dials out, the Call Park soft button will display
on screen. To park the call, press the ‘Call Park’ button.
• PickUp When another GXP2120 goes off-hook the Call Pickup soft button
will display on screen. To pickup the parked call, press the ‘Call
Pickup’ button.
Call Queue: FOR GXP2120/211 ONLY. Refer to the GXE5024/5028 Online User Manual for
more information.
SPECIAL • SignIn Press this button to sign in to the call queue. Agent will be prompted in
SOFT the LCD display to select the call queue to join. Press ‘menu’ button on
BUTTONS keypad to select ‘ok’. Once the agent completely signs in, the agent will
(Only When be brought back to the main screen.
Integrated • SignOut Press this button to sign out of the call queue. Press’ menu’ button on
with keypad to select ‘ok’. This will be displayed once the agent is signed in to
GXE5024/502 the call queue.
8)
PUBLIC MODE (Also mentioned on p.31 of this manual): This useful mode complements the
Call Queue feature by allowing various user agents to log in/log off, sharing the same phone.
When enabled, all other accounts on the phone will not be active. For more information, refer to
http://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_wit
h_gxp.pdf
• LogIn Press this button to log in the user agent into the call queue.
• Tab Press this button to jump to toggle between UserName and Password
entry fields.
• Backspace Press this button to erase the previously typed digit, letter, or character.
• LogOut Press this button to log out the user agent out of the call queue.
Real–time Clock:
AM Synchronized to Internet time server
Time zone configurable via web browser
PM AM/PM indicator
LINE BUTTONS Line keys with LED, can be configured to different SIP profiles
Press SEND to dial a new number or redial the last number dialed. Press
SEND send button to send a call immediately before “no key entry timeout” value
expires
Standard phone keypad; press # key to send call; press * key to for IVR
0 - 9, *, #
functions
DO NOT DISTURB key; Press DND to turn “Do not disturb” function on or off.
DND
(Excluded on GXP2100)
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:
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.
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.
• 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.
Grandstream Networks, Inc. GXP21xx User Manual Page 17 of 42
Firmware 1.0.0.44 Last Updated: 10/2010
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.
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.
The GXP21xx 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.
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)
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
GXP21xx supports both Blind and Attended (or supervised) transfer:
1. Blind Transfer: Press “TRANSFER” 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 “TRANSFER” key then the LINE
button of the waiting line to transfer the call. Hang up the phone call after “Transfer Successful” is
displayed in the screen.
5-Way Conferencing
GXP can host conference calls and supports up to 5-way conference calling (Excludes GXP2100 which
supports up to 4-way).
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. Also, this is not applicable when the feature
“Transfer on call hangup” is turned on.
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.
In the middle of the conversation, there are two types of hold: Public Hold and Private Hold. When a member
of the group places the call on public hold, the other users of the SCA group will be notified of this by the red-
flashing button and they will be able to resume the call from their phone by pressing the line button. However,
if this call is placed on private-hold, no other member of the SCA group will be able to resume that call.
To enable shared call appearance, the user would need to register the shared line account on one of the
accounts on the phone. In addition, they would need to navigate to “Settings”->”Basic Settings” on the web
UI and set the line to “Shared Line” with the corresponding account. If the user requires more shared call
appearances, the user can configure multiple line buttons to be “shared line” buttons associated with the
account.
CALL FEATURES
The GXP21xx 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.
*73 Cancel Unconditional Call Forward: dial “*73” and get the dial tone, then hang up.
LCD will display “Call FWD Activated”.
*91 Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
Grandstream GXP21xx Enterprise IP phone 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 GXP21xx, please visit our website at:
http://www.grandstream.com/support/gxp_series/general/gxp_support.html .
Configuration Guide
The GXP21xx can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone;
secondly, through embedded web-configuration menu.
Call History Displays histories of incoming, dialed, missed, and transferred calls.
Status Displays the network status, account statuses, software version and
MAC-address of the phone.
Phone Book Displays the phonebook
LDAP Directory Displays the LDAP directory
Instant Messages Goes to voice messages
Direct IP call Displays the IP-call options 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.
Call History
Any of previous menus
Answered Calls
Dialed Calls Back
Missed Calls Clear All
Transferred Calls
MENU Back
New Entry
Phone Book
Name:
New Entry Number:
Download Phonebook XML Acct:
Back Confirm Add:
Cancel & Return:
LDAP Directory
Search Configuration
View Directory
Call History Download Directory Select Filter
Search Configuration Filter Value
Back Back
Status
Instant Message
Do Not Disturb
Phone Book Clear All
Back Enable DND
Disable DND
LDAP Directory Back
Preference
Ring Tone
Do Not Disturb Network
Instant Default Ring
Ring Tone
Message Ring1
LCD Contrast IP Setting
LCD Brightness Ring2 IP
Download SCR XML Ring 3 Net Mask
Direct IP Call
Erase Custom SCR Back Gateway
Display Language DNS Server
Back LCD Brightness 1
Preference
DNS Server
Active
Config Idle SIP
Config Back
Account
Network SIP Proxy
Factory SIP
Functions Outbound
Upgrade Proxy
Factory Reset SIP User ID
Layer 2 QoS SIP Auth ID
Reboot Back Display Language
SIP Password
SIP Transport
English
Exit Audio
Factory Function Chinese
Save
French
Audio Loopback Spanish Upgrade
Diagnostic Mode German
Back Italian Firmware
Secondary Language Server
Language File Postfix Config Server
Back Upgrade Via
Diagnostic Mode Layer 2 QoS
Keypad/LED Diagnostic 802.1Q/VLAN Tag
Priority value
Reset Vlan Config
Back
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.
Software Version • Program: This is the main firmware release number, which is always used for
identifying the software (or firmware) system of the phone.
• Boot: Booting code version number
• Core: Core code version number
• Base: Base code version number
• DSP: DSP code version number
• Aux: Aux code version number
System Up Time This field shows system up time since the last reboot.
System Time This field shows the current time on the phone system.
Registered 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).
End User This contains the password to access the Web Configuration Menu. This field is
Password case sensitive with a maximum length of 25 characters.
802.1x Mode This option allows the user to enable/disable 802.1x mode on the phone. The
default value is disabled. To enable 802.1x mode, this field should be set to EAP-
MD5.
Once enabled, the user would be required to enter the following information below
to be authenticated on the network:
• Identity
• MD5 Password
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.
Self-Defined Time Zone This parameter allows the users to define their own time zone.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M3.2.0,M11.1.0
MTZ+6MDT+5,
This indicates a time zone with 6 hours offset with 1 hour ahead which is U.S
central time. If it is positive (+) if the local time zone is west of the Prime Meridian
(A.K.A: International or Greenwich Meridian) and negative (-) if it is east.
M3.2.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd
Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues,..,Sat)
Therefore, this example is the DST which starts from the second Sunday of March
to the 1st Sunday of November.
LCD Backlight Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means
Brightness the brightest.
Disable in-call DTMF Default is No. This field is used to hide the keypad input during a call.
display
Disable Missed Call Default is No. By default, LCD backlight will lit whenever there is a missed call.
Backlight
In “Default Mode”, only the speakerphone will ring for an incoming call. User can
use the headset key to pick-up, speak, and hang up calls through headset. The
headset icon will appear on the LCD when a call is in progress.
Headset Port Type Select either 2.5mm or RJ9 headset ports to be adjusted.
Headset TX Gain (dB) Increases the selected headset’s (2.5mm or RJ9) TX gain by + or – 6dB. Default is
0dB
Headset RX Gain (dB) Increases the selected headset’s (2.5mm or RJ9) RX gain by + or – 6dB. Default is
0dB
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.
Admin Administrator password. Only the administrator can access the “Advanced
Password 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.
Layer 3 QoS This field defines the layer 3 QoS parameter. It is the value used for IP
Precedence or Diff-Serv or MPLS. Default value is 48.
Layer 2 QoS This contains the value used for layer 2 VLAN tag. Default setting is blank.
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.
Firmware Upgrade and Allows the user to select the following options for firmware upgrade:
Provisioning • Always Check for New Firmware
• Check New Firmware only when F/W pre/suffix changes
• Always Skip the Firmware Check.
XML Config File The password used for encrypting the XML configuration file using OpenSSL.
Password This is required for the phone to decrypt the encrypted XML configuration file.
HTTP/HTTPS User Name The user name for the HTTP/HTTPS server.
Upgrade Via This field allows the user to choose the firmware upgrade method: TFTP, HTTP
or HTTPS.
Firmware Server Path Defines the server path for the firmware server. It can be different from the
Configuration server which is used for provisioning.
Config Server Path Defines the server path for provisioning; it can be different from the firmware
server.
Firmware File Default is blank. If configured, GXP21xx will request the firmware file with the
Prefix/Postfix prefix/postfix and only the firmware with the matching encrypted prefix will be
downloaded and flashed into the phone.
This setting is useful for ITSPs. End user should keep it blank.
Config File Default is blank. If configured, GXP21xx will request the config file with the
Prefix/Postfix prefix/postfix and only the file with the matching encrypted prefix will be
downloaded and flashed into the phone.
This setting is useful for ITSPs. End user should keep it blank.
Allow DHCP Option 43 Default is Yes. This allows device gets provisioned automatically.
and Option 66 to
override server
Authenticate Conf File Default is “No”. If set to “Yes”, configuration file would be authenticated before
acceptance. End user should use default setting.
Phonebook XML Selects the file download mode for the download server. Users can choose
Download from TFTP/HTTP/No.
Phonebook XML Server The URL/IP address of the phonebook download server
Path
Phonebook Download The interval at which the phonebook will be downloaded from the download
Interval server (in Minutes). The default setting is 0.
Remove Manually-edited If set to “Yes”, the phone will remove the manually-edited entries in the old
entries on Downloads phonebook list before downloading the new file. The default setting is set to
“Yes”.
Idle Screen XML Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use
Download Custom Filename” or not, and define the “XML server path”.
Softkey Label Defines the softkey label for the XML application
Offhook Auto Dial To configure a User ID/extension to dial automatically when the phone is taken
offhook.
Syslog Server The IP address or URL of System log server. This feature is especially useful
for ITSPs.
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.
SSL Certificate This defines the SSL certificate needed to access certain websites.
SSL Private Key This defines the SSL private key password.
Password
Distinctive Ring Tone Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a
particular Caller ID. The GXP will ONLY use selected ring tones for particular
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.
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.
Disable Call Waiting Default is No. If set to Yes, the call waiting feature will be disabled.
Disable Call Default is No. If set to Yes, the call waiting tone will be disabled.
Waiting Tone
Disable Direct IP Calls Default is No. If set to Yes, direct IP calls 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.
Enable MPK Sending Default is No. If set to “Yes”, Muti-Purpose keys can be sent as DTMF.
DTMF
Disable DND Button Default is No. If set to “Yes”, the “DND” button on keypad will be disabled.
Configuration via Configures the access control of configurations via the phone keypad menu.
Keypad Menu There are three modes:
• Unrestricted
• Basic Settings Only
• Constraint Mode
Language file postfix allows the language file to have different postfixes so the
phone can request a particular file. It will append an underscore "_" plus the
string in the language file postfix.
The default language file name is "gxp.lpf". If the field “Language File postfix
“has "NL" string in it, then the phone will request "gxp_NL.lpf" instead of
"gxp.lpf."
GXP21xx has up to six line appearances, each with an independent SIP account. Each SIP account
requires its own configuration page. Their configurations are identical.
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 The name associated with each account - displayed on LCD.
SIP Server SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy IP address or Domain name of Outbound Proxy, Media Gateway, or Session
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 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.
DNS Mode The default is set to A Record. If the user wishes to locate the server by DNS
SRV, the user may select SRV or NATPTR/SRV.
User ID is Phone If the phone has an assigned PSTN telephone number, this field should be set to
Number “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 Default is “No”. If set to “Yes”, the SIP user’s registration information will be
cleared on reboot.
Register Expiration 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).
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 Transport Choose SIP Transport between UDP and TCP. Default is UDP.
Check Domain When set to Yes/Enabled, it will check the domain certificate as defined in
Certificates RFC5922
Remove OBP from The SIP Extension notifies the SIP server that it is behind a NAT/firewall.
Route
Validate Incoming This configuration selects whether or not the incoming messages should be
Messages validated.
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 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.
DTMF Payload Type Sends DTMF using RFC2833. The default is 101.
Early Dial Default is No. Use only if proxy supports 484 responses.
Dial Plan Prefix Sets the prefix added to each dialed number.
BLF Call-pickup Prefix Default is ‘**”. This prefix is prepended when answering call with BLF key.
Delayed Call Forward Time waited before the call is forward to a number or VM.
Wait Time Default is 20 seconds.
Enable Call Features Default is Yes. If set to “No”, Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided ITSP support those features. In addition, “ForwardAll”
softkey will be hidden if call feature code is disabled for Account 1.
Call Log User can choose to disable Call Log and what kind of calls to log.
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 If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.
Force Timer If set to “Yes”, the phone will use session timer even if the remote party 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 Session Timer can be refreshed using INVITE method or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
Ring Timeout Defines how long ring will ring when receiving a call. Default is 60 seconds.
Send Anonymous If this parameter is set to “Yes”, the “From” header in outgoing INVITE message
will be set to anonymous, essentially blocking the Caller ID from displaying.
Anonymous Call Default is NO. If set to YES, anonymous call will be rejected
Rejection
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.
Allow Auto Answer by If the Call-Info header contains answer-after=0, the call be answered
Call-Info automatically (so called paging mode).
Transfer on Conference Defines whether or not the call is transferred to the other party if the initiator of the
Hangup conference hangs up.
Default setting is set to No.
Preferred Vocoder GXP supports up to 7 different Vocoder types including G.711(a/µ) (also known
as PCMU/PCMA), 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”.
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 has a patent dynamic jitter buffer
handling algorithm. The jitter buffer range is 20 ~ 200 ms.
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.
eventlist BLF URI If a server supports this feature, user needs to configure an "eventlist BLF" URI
on the service side (i.e.: BLF1006@myserver.com)
On the GXP, under Account page, fill in the ""eventlist BLF" field with the URI
without the domain. (i.e.: BLF1006). Under Basic Settings, please select "eventlist
BLF", choose account number, monitored number, etc.
Special Feature Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.
Grandstream recommends reboot or power cycle the IP phone after saving changes.
• firmware.mycompany.com:6688/Grandstream/1.2.3.5
• 72.172.83.110
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web
Configuration Interface.
During this stage, the LCD will display the firmware file downloading process. Please do NOT disrupt or
power down the unit. 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.
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: http://support.solarwinds.net/updates/New-
customerFree.cfm.
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”
The GXP21xx can be configured via Web Interface as well as via Configuration File (binary or XML) through
TFTP or HTTP/HTTPS. The “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. The “Config Server Path” can be the
same or different from the “Firmware Server Path”.
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 GXP21xx boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”
followed by a request for configuration XML file named “cfgxxxxxxxxxxxx.xml”, where “xxxxxxxxxxxx” is the
MAC address of the device, i.e., “cfg000b820102ab”. The configuration file name should be in lower cases.
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: 22 (press the “2” key twice, “A” will show on the LCD)
B: 222
C: 2222
D: 33 (press the “3” key twice, “D” will show on the LCD)
E: 333
F: 3333
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.