Grandstream Networks GXP1450 User manual

Grandstream Networks, Inc. GXP1450 User Manual Page 1 of 1
Firmware 1.0.1.66 Last Updated: 05/2011
Grandstream Networks, Inc.
GXP1450 SIP Enterprise Phone

Grandstream Networks, Inc. GXP1450 User Manual Page 1 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
TABLE OF CONTENTS
GXP1450 USER MANUAL
WELCOME..........................................................................................................................................................3
INSTALLATION.................................................................................................................................................4
EQUIPMENT PACKAGING ......................................................................................................................................4
CONNECTING YOUR PHONE..................................................................................................................................4
SAFETY COMPLIANCES.........................................................................................................................................4
WARRANTY .........................................................................................................................................................4
PRODUCT OVERVIEW.....................................................................................................................................5
USING THE GXP1450 SIP ENTERPRISE PHONE ..........................................................................................8
GETTING FAMILIAR WITH THE LCD ......................................................................................................................8
MAKING PHONE CALLS ......................................................................................................................................10
ANSWERING PHONE CALLS ................................................................................................................................13
PHONE FUNCTIONS DURING A PHONE CALL ........................................................................................................13
CALL FEATURES ................................................................................................................................................15
CUSTOMIZED LCD SCREEN &XML ...................................................................................................................16
CONFIGURATION GUIDE................................................................................................................................17
CONFIGURATION VIA KEYPAD............................................................................................................................17
CONFIGURATION VIA WEB BROWSER ................................................................................................................21
SAVING THE CONFIGURATION CHANGES .............................................................................................................35
REBOOTING THE PHONE REMOTELY....................................................................................................................35
SOFTWARE UPGRADE & CUSTOMIZATION.............................................................................................36
FIRMWARE UPGRADE THROUGH TFTP/HTTP.....................................................................................................36
CONFIGURATION FILE DOWNLOAD .....................................................................................................................37
RESTORE FACTORY DEFAULT SETTING..................................................................................................38
TABLE OF FIGURES
GXP1450 USER MANUAL
Figure 1: GXP1450 Keypad Layout……………………………………………………………………. 10
Figure 2: Keypad GUI Flow........................................................................................................18
TABLE OF TABLES
GXP1450 USER MANUAL
Table 1: Equipment Packaging...................................................................................................4
Table 2: GXP1450 Connectors ...................................................................................................4
Table 3: GXP1450 Feature Guide...............................................................................................5
Table 4: GXP1450 Key Features in a Glance..............................................................................5
Table 5: GXP1450 Hardware Specifications................................................................................5
Table 6: GXP1450 Technical Specifications................................................................................6
Table 7: LCD Buttons .................................................................................................................8
Table 8: LCD Icons.....................................................................................................................8
Table 9: GXP1450 Keypad Buttons.............................................................................................9

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Table 10: GXP1450 Call Features ............................................................................................15
Table 11: Key Pad Configuration Menu.....................................................................................17
Table 12: Device Configuration - Status....................................................................................22
Table 13: Device Configuration – Settings/Basic Settings..........................................................22
Table 14: Device Configuration – Settings /Advanced Settings..................................................24
Table 15: SIP Account Settings.................................................................................................29

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Welcome
GXP1450 is a next generation enterprise grade IP phone that features 2 lines with 2 SIP accounts, a
180x60 backlit graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with
integrated PoE, and 3-way conference.
The GXP1450 delivers superior HD audio quality, rich and leading edge telephony features, personalized
information and customizable application service, automated provisioning for easy deployment, advanced
security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading
SIP/NGN/IMS platforms. It is a perfect choice for enterprise users looking for a high quality, feature rich IP
phone with affordable cost.
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 GXP1450 as it may cause damage to the
products and void the manufacturer warranty.
•This document is subject to change without notice.
•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 is not permitted.

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Installation
EQUIPMENT PACKAGING
Table 1: Equipment Packaging
GXP1450
Main Case
Yes
Handset
Yes
Phone Cord
Yes
Power Adaptor
Yes
Ethernet Cable
Yes
Base Stand
Yes
Quick Start Guide
Yes
CONNECTING YOUR PHONE
The connectors of the GXP1450 are located on the bottom of the device.
Table 2: GXP1450 Connectors
PC
10/100Mbps RJ-45 ports for PC (downlink) connection.
LAN
10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af).
Power Jack
5V DC power port; UL Certified
Headset Jack
RJ9
Handset Jack
RJ9
SAFETY COMPLIANCES
The GXP1450 complies with FCC/CE and various safety standards. The GXP1450 power adaptor is
compliant with the UL standard. Only use the universal power adaptor provided with the GXP1450 package.
The manufacturer’s warranty does not cover damages to the phone caused by unsupported power adaptors.
WARRANTY
If you purchased your GXP1450 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: GXP1450 Feature Guide
Features GXP1450
LCD Display 180x60 pixel
Number of Lines 2
Programmable Soft Keys 3
Extension Module N/A
Table 4: GXP1450 Key Features in a Glance
Features Benefits
Open Standards Compatible SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP,
ICMP, DNS (A record, SRV and NAPTR), DHCP (both client and
server), PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, SIP over TLS,
802.1x, TR-069
Superb Audio Quality Advanced Digital Signal Processing (DSP), Silence Suppression, VAD,
CNG, AGC
Network Interfaces Dual 10/100mbps Ethernet ports with integrated PoE
Feature Rich Traditional voice features including caller ID, call waiting, hold, transfer,
forward, block, auto-dial, off-hook dial
Advanced Features 2 line keys with dual-color LED and 2 SIP accounts, 3 way
conferencing, backlit graphic 180x60 LCD,
3 XML programmable
context sensitive soft keys, 5 navigation keys, 10 dedicated buttons for
HOLD, TRANSFER, CONFERENCE, VOLUME, HEADSET, MUTE,
SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE
Advanced Functionality Customized downloadable ring-tones, SRTP, SIP over TLS, multi-
language support and XML enabled, adjustable positioning angles, wall
mountable, AES encryption, automatic multimedia service (eg., weather
information)
Table 5: GXP1450 Hardware Specifications
GXP1450
LAN Interface (Ethernet ports) Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and
PC port with auto detection
Graphic LCD Display 180x60 pixel
Expansion Module Support No
Headset Jack RJ9
Call Appearance LED 2 Dual color (green/red)
Power over Ethernet Built-in auto-sensing: Cisco and IEEE 802.3af standard
Universal Switching Input: 100-240VAC 50-60 Hz

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Power Adaptor Output: +5VDC, 800mA, UL certified
Dimension 186mm (W) x 210mm (L) x 81mm (D)
Weight 0.8KG
Temperature 32 –104
°
F/ 0 – 40
°
C
Humidity 10% – 90% (non-condensing)
Compliance FCC / CE / C-Tick
Table 6: GXP1450 Technical Specifications
Lines
2 lines with 2 independent SIP accounts, XML programmable soft-keys.
Protocol Support
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES,
HTTP, ARP/RARP, ICMP, DNS, DHCP, NT
P, 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)
Display
Backlit graphic LCD display, up to 4 level grayscale
Feature Keys HOLD, TRANSFER, CONF, VOLUME, HEADSET, MUTE,
SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE, 3 XML
Programmable Softkeys, 5 Navigation keys,
Device
Management NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed
devices including behind firewall/NAT
Auto/manual provisioning system, Web GUI Interface
Address Book
Audio Features
Full-duplex hands-free speakerphone
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), 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 and loss
concealment
HD audio handset with HD wideband audio codecs for excellent double-
talk performance
Telephony Features Intuitive graphic user interface (GUI), downloadable phone book (XML,
LDAP), support for anonymous call using privacy header, MLS (multi
language support)
Voice mail indicator, downloadable custom ring-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

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3-way conference, dial plan prefix, dial-plan support, off-hook auto dial,
auto answer, early dial and speed dial
Network and Provisioning Via keypad/LCD, Web browser, or secure (AES encrypted) central
configuration file, manual or dynamic host configuration protocol (DHCP)
network setup
Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 ToS
Firmware Upgrade Support firmware upgrade via TFTP or HTTP
Support for Authenticating configuration file before accepting changes
User specific URL for configuration file and firmware files
Mass provisioning using TR-069 or encrypted XML configuration file
Advanced Server Features Message waiting indication, support DNS SRV Look up and SIP Server
Fail Over, Support customizable idle screen via downloading XML by
HTTP/TFTP
Security User and administrator level passwords, MD5 and MD5-sess based
authentication, AES based secure configuration file, SRTP, TLS, 802.1x
media access control

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Using the GXP1450 SIP Enterprise Phone
GETTING FAMILIAR WITH THE LCD
GXP1450 has a dynamic and customizable screen. The screen displays differently depending on whether
the phone is idle or in use (active screen).
Table 7: LCD Buttons
LCD Button
LCD Button Definitions
DATE AND TIME
Displays the current date and time. Can be synchronized with Internet time servers.
LOGO/NAME
Displays company logo name. This logo name can be customized via xml screen
customization.
NETWORK STATUS
Displays the status of the phone and network. It will indicate whether the network is
down, starting or running (IP address). “## MISSED CALLS” is shown here too.
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 key on the left side.
SOFTKEYS
The softkeys are context sensitive and will change depending on the status of the
phone. Typical functions assigned to soft-buttons are:
•FORWARD ALL Unconditionally forwards the phone line to another phone
•MISSED CALL This option shows up there were unanswered calls to this
phone. The Missed Calls option shows a list of the missed
calls
•NEXTSCR Press this button to toggle between idle screen, weather
and IP Address.
•REDIAL Redials the last number
•END CALL Hangs up phone
Table 8: LCD Icons
Icon
LCD Icon Definitions
DND Icon: ON when the “Do Not Disturb” is activated
Calls Forwarded Icon: INDICATES calls are forwarded
Key pad lock Icon: ON when using STAR key to lock the keypad
Enter Password to unlock the keypad
Voice Mail / Message Waiting Indicator:ON when there is new voice mail
/ message
Network Status: Network is down

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Missed Call Icon: Indicates missed call(s)
Save Call Record:
Indicates phone system writing the call records into the
flash. It might take 10 to 20 seconds to finish the process
FIGURE 1: GXP1450 KEYPAD LAYOUT
Table 9: GXP1450 Keypad Buttons
Key Button
Key Button Definitions
LINE BUTTONS
2 Line keys with LED, can be configured to different SIP profiles
HOLD
Place ACTIVE call on hold
TRANSFER
Transfer an ACTIVE call to another number
CONF
Press CONF button to connect Calling/Called party into conference
Enter to retrieve voice mails or other messages
Brings phonebook on screen
Mute an active call
Press HEADSET key to answer/hang up phone calls while using headset. It
also allows user to toggle between headset and speaker
Enable/Disable hands-free speaker

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Enable/Disable handset mode ; or used as SEND/REDIAL
Press “–” or “+” to adjust the volume for handset/speakerphone/headset
Enter Keypad Configuration “MENU” mode when phone is in IDLE mode.
Use as ENTER key when in Keypad Configuration.
0 - 9, *, #
Standard phone keypad; press # key to send call; press * key to for IVR
functions
MAKING PHONE CALLS
Handset, Speakerphone and Headset Mode
The GXP1450 allows you to make phone calls via handset, speakerphone, or headset mode. During the
active calls the user can switch between the handset and the speaker by pressing the speaker key. For
headsets to operate, the user must plug the headset to an RJ9 port on the phone, which allows the user to
pick-up, speak, or hang-up calls.
Multiple SIP Accounts and Lines
GXP1450 can support up to two independent SIP accounts. 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 GXP1450 will flash the next available LINE in red. A line is ACTIVE
when it is in use and the corresponding LED is red.
Completing Calls
There are five ways to complete a call:

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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” softkey
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 second SIP account was made for the last call/call attempt, the phone will use the second
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 softkey
3. 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 second SIP account, the phone will use the
second 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
4. USING THE PHONEBOOK:To call 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
5. 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

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•Press SEND key.
NOTE: Dialtone and dialed number display occurs after the phone 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.
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 refer to 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 GXP1450 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

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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.
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 menu button, and scroll down to “Preference”.
2. Select “Do Not Disturb” by pressing menu button.
3. Use arrow keys to either enable or disable “Do Not Disturb” feature.
4. When enabled, there will be a special “Do Not Disturb” icon appearing on the display. This will send
the incoming caller directly to voicemail.
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
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
GXP1450 supports both Blind and Attended transfer:
1. Blind Transfer: Press “TRANSFER” button, then dial the number and press the “SEND” button to
complete transfer of active call.

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2. Attended 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.
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.
3-Way Conferencing
GXP1450 can host conference calls and supports up to 3-way conference calling.
1. Initiate a Conference Call:
Establish a connection with two parties
Press CONF button
Choose the desired line to join the conference by pressing the corresponding LINE button
Repeat previous two steps for all other parties that would like to join the conference. This
can be done at any time. However, if a new call comes in, the other calls will be placed on
hold and the host will have to individually re-join the held lines back into the conference by
repeating the previous two steps again
2. Cancel Conference: Canceling establishing conference call
If after pressing the “CONF” button, a user decides not to conference anyone, press CONF
again or the original LINE button
This will resume two-way conversation
3. End Conference:
Press HOLD to end the conference call and put all parties on hold
To speak with an individual party, select the corresponding blinking LINE
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.
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.
Shared Call Appearance (SCA)
The GXP1450 phone supports shared call appearance by Broadsoft standard. This feature allows members
of the SCA group to shared SIP lines and provides status monitoring (idle, active, progressing, hold) of the
shared line. When there is an incoming call designated for the SCA group, all of the members of the group

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will be notified of an incoming call and will be able to answer the call from the phone with the SCA extension
registered.
All the users that belong to the same SCA group will be notified by visual indicator when a user seizes the
line and places an outgoing call, and all the users of this group will not be able to seize the line until the line
goes back to an idle state or when the call is placed on hold. (With the exception of when multiple call
appearances are enabled on the server side)
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
GUI 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 GXP1450 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 10: GXP1450 Call Features
Key Call Features
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*67 Block Caller ID (per call)
*82 Send Caller ID (per call)
*70 Disable Call Waiting (per Call)
*71 Enable Call Waiting (per Call)
*72 Unconditional Call Forward
Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wait for dial
tone. LCD will display “Call FWD Activated”.
*73 Cancel Unconditional Call Forward: dial “*73” and get the dial tone, then hang up.
LCD will display “Call FWD Activated”.
*90 Busy Call Forward
Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial
tone. Hang up.
*91 Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
*92 Delayed Call Forward
Dial “*92” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial
tone. Hang up. LCD will display “Call FWD Activated”.

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*93 Cancel Delayed Call Forward
Dial “*93” for a dial tone, then hang up.
CUSTOMIZED LCD SCREEN &XML
GXP1450 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 softkeys on GXP1450,
please visit our website at: http://www.grandstream.com/support

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Firmware 1.0.1.66 Last Updated: 05/2011
Configuration Guide
The GXP1450 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 11.
Table 11: Key Pad Configuration Menu
Item Description
Call History Displays histories of answered, dialed, missed, and transferred and forwarded
calls
Status Displays the network status, account status, software version, MAC-address and
hardware version of the phone
Phone Book Displays the phonebook and downloads phonebook XML
LDAP Directory Displays the LDAP directory and downloads directory
Instant Messages Goes to instant messages
Direct IP Call Dials IP address for direct IP call
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
Press “←”or “→” to adjust the LCD contrast.
•LCD Brightness
Press “←”or “→” to adjust the LCD brightness for active/idle screen.
•Download SCR XML
The phone will download the custom idle screen if available.

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Firmware 1.0.1.66 Last Updated: 05/2011
•Erase Custom SCR
Custom idle screen will be erased and will be replaced with default
logo.
•Display Language
You can choose English, Simplified Chinese, Traditional Chinese,
Korean, Japanese, Italian, Spanish, French, German, Portuguese,
Russian, Croatian, Hungarian, Polish, Slovenian which are built in the
phone. Users could select Automatic for local language based on IP
location if available. Also, the phone will download secondary
language if available.
•Time Settings
Press Menu button to choose the menu item
Press “←” or follow the soft keys to return to the main menu
Config Press Menu button to display the configuration selections:
•SIP
To change SIP server settings for SIP accounts.
•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 802.1Q/VLAN Tag and priority value.
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.
Press “←” to return the main menu
Network To enable/disable DHCP; to setup IP-address, Net mask and Gateway address
Reboot Press Menu button to reboot the device
Exit Exit from this menu
FIGURE 2: KEYPAD GUI FLOW

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Firmware 1.0.1.66 Last Updated: 05/2011