CONNECTING YOUR PHONE ........................................................................................................................................ 4
USING THE GXP1400/1405 ....................................................................................................................................... 8
GETTING FAMILIAR WITH THE LCD ............................................................................................................................ 8
MAKING PHONE CALLS ............................................................................................................................................... 9
PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 12
CALL FEATURES ........................................................................................................................................................ 15
CONFIGURATION VIA KEYPAD .................................................................................................................................. 17
Grandstream Networks, Inc. GXP1400/1405 User Manual Page 2 of 41
Firmware version: 1.0.1.110 Last Updated: 01/2012
Welcome
GXP1400/1405 is a next generation small-to-medium business IP phone that features 2 lines with 2 SIP
accounts, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports
with integrated PoE (GXP1405 only), and 3-way conference. The GXP1400/1405 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 3
is a perfect choice for small-to-medium businesses 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 GXP1400/1405 as it may cause damage
to the products and void the manufacturer warranty.
Note:
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 perm itted.
rd
party SIP devices and leading SIP/NGN/IMS platforms. It
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Installation
EQUIPMENT PACKAGING
Table 1: Equipment Packaging
Main Case
Handset
Phone Cord
Power Adaptor
Ethernet Cable
Base Stand
Quick Start Guide
GXP1400/1405
Yes
Yes
Yes
Yes
Yes
Yes
Yes
CONNECTING YOUR PHONE
The connectors of the GXP1400/1405 are located on the bottom of the device.
Table 2: GXP1400/1405 Connectors
PC
LAN
Power Jack
Handset Jack
Headset Jack
10/100Mbps RJ-45 ports for PC (downlink) connection
10/100Mbps RJ-45 port for LAN (uplink) connection, integrated PoE (GXP1405 only)
5V DC power port; UL Certified
RJ9
RJ9
SAFETY COMPLIANCES
The GXP1400/1405 phone complies with FCC/CE and various safety standards. The GXP1400/1405 power
adaptor is compliant with the UL standard. Please use the universal power adaptor provided with the
GXP1400/1405 package only. The manufacturer’s warranty does not cover damages to the phone caused by
unsupported power adaptors.
ARRANTY
W
If you purchased your GXP1400/1405 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: GXP1400/1405 Feature Guide
Features GXP1400/1405
LCD DisplayNumber of LinesProgrammable Soft KeysExtension Module
Table 4: GXP1400/1405 Key Features in a Glance
Features Benefits
Open Standards Compatibility
Superb Audio Quality
Network InterfacesFeature Rich
Advanced Features
Advanced Functionality
128 x 40 pixel
2
3
N/A
SIP RFC3261, TCP/IP/UDP, RTP, HTTP/HTTPS, ARP/RARP, ICMP,DNS (A record, SRV and NAPTR), DHCP (both client and server),PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, 802.1x, TR-069
Advanced Digital Signal Processing (DSP), Silence Suppression, VAD, CNG, AGC
Traditional voice features including caller ID, call waiting, hold, transfer,forward, block, auto answer, off-hook dial and etc
2 line keys with dual-color LED and 2 SIP accounts, 3 way conference, graphic LCD, 3 XML programmable context sensitive soft keys, 5navigation keys, 8 dedicated buttons for HOLD, TRANSFER, CONFERENCE, VOLUME, HEADSET, MUTE/DND, SPEAKERPHONE, SEND/REDIAL
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: GXP1400/1405 Hardware Specifications
GXP1400/1405
LAN Interface
10/100 Mbps Full/Half Duplex Ethernet port with auto detection Integrated PoE (GXP1405 only)
Graphic LCD DisplayExpansion ModuleCall Appearance LED Universal Switching
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128 x 40 pixel
N/A
2 Dual color (green/red) line keys
Input: 100-240VAC 50-60 Hz
Power Adaptor
DimensionWeight
Output: +5VDC, 800mA, 4.0 W, UL certified
186mm (W) x 210mm (L) x 81mm (D) Unit weight: 0.7KG Package weight: 1.1KG (GXP1400), 1.0KG (GXP1405)
TemperatureHumidityCompliance
32 -104
10% - 90% (non-condensing)
FCC Part 15 (CFR 47) Class B
°
F/ 0 - 40°C
EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1 AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS UL 60950 (power adapter)
Table 6: GXP1400/1405 Technical Specifications
Lines
2 lines with 2 SIP accounts, 3 XML programmable soft-keys
Protocol Support SIP RFC3261, TCP/IP/UDP, RTP, HTTP/HTTPS, ARP/RARP, ICMP,
DNS (A record, SRV and NAPTR), DHCP (both client and server),PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, 802.1x, TR-069
Support 2 SIP accounts Support SIP Dialog package (RFC 4235) Support for SIP MESSAGE method (RFC 3428)
DisplayFeature Keys
Graphic LCD display, up to 4 level grayscaleHOLD, TRANSFER, CONF, LINE 1, LINE 2, MSG, SPEAKERPHONE,
HANDSET, HEADSET, MUTE/DND, NAVIGATION(5), VOLUME, 3 XML Programmable Soft keys
Device ManagementNAT-friendly remote software upgrade (via TFTP/HTTP) for deployed
devices including behind firewall/NAT Auto/manual provisioning system, Web GUI InterfaceSupport Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS,
DiffServ, MPLS)
Audio Features
Full-duplex hands-free speakerphoneAdvanced 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 toneAdaptive jitter buffer control (patent-pending) and packet delay and loss concealmentHD audio handset with HD wideband audio codecs for excellent double-talk performance
Telephony FeaturesIntuitive 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
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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
3-way conference, dial plan prefix, dial-plan support, off-hook auto dial,
auto answer and early dial
Network and ProvisioningVia keypad/LCD, Web browser, orsecure (AES encrypted) central
configuration file, manual or dynamichost 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 Upgrades
Support firmware upgrade via TFTP or HTTPSupport for Authenticating configuration file before accepting changes
User specific URL for configuration file and firmware files Mass provisioning using TR-069 orencrypted 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 basedauthentication, AES based secure configuration file, SRTP, TLS, 802.1xmedia access control
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Firmware version: 1.0.1.110 Last Updated: 01/2012
Using the GXP1400/1405
GETTING FAMILIAR WITH THE LCD
GXP1400/1405 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 Display Definition
Display Item Definitions
DATE AND TIME
LOGO NAME
NETWORK
STATUS
STATUS BAR
SOFTKEYS
Table 8: LCD Icons
Displays the current date and time. It can be synchronized with Internet time
servers
Displays company logo name. This logo name can be customized via xml screen
customization. The maximum size for logo name is 26 characters in English
(approximately)
Shows the status of network in the middle of the screen. It will indicate whether
the network is down or starting
Shows the status of the phone, using icons as shown in the next table
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 unanswered calls to this phone.
NEXTSCR Press this button to toggle between idle screen, weather
and IP Address.
REDIAL Redials the last dialed-out number
END CALL Hangs up the call
LCD Icons Descriptions
SIP Registration Status Icon:
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Solid – connected to SIP Server/IP address received
SIP Registration Status Icon:
Blank – SIP Proxy/Server not registered
Handset Status Icon:
OFF - handset on-hook ON - handset off-hook
Speaker Phone Status Icon:
OFF - speakerphone off ON - speakerphone on
Headset Status Icon:
OFF - headset off ON - headset on
DND Icon:
OFF - “Do Not Disturb” disabled ON - “Do Not Disturb” enabled
Calls Forwarded Icon:
INDICATES calls are forwarded. Please refer to call forwarding procedures
MUTE Icon:
INDICATES call is on MUTE during the call
SRTP Icon:
INDICATES SRTP is enabled for the call
Table 9: GXP1400/1405 KEYPAD BUTTONS
Button Descriptions
HOLD
TRANSFER
CONF
LINE 1 / LINE 2
0 - 9, *, #
Place active call on hold
Transfer an active call to another number
Press CONF button to connect Calling/Called party into conference
Switch between Line 1 and Line 2
Mute an active call; or use as DND button when the phone is in idle state.
Press HEADSET key to answer/hang up phone calls when using headset. It also
allows user to toggle between headset and speaker
Enable/Disable hands-free speaker
Enable/Disable handset mode; or used as SEND/REDIAL
Press the four navigation keys to move up/down/left/right
Press the round button in the center to enter Keypad Configuration “MENU”
mode when phone is idle. Or use it as ENTER key when in Keypad
Configuration
Adjust volume by pressing “– “or “+”
Standard phone keypad; press # key to send call; press * key to for IVR
functions
MAKING PHONE CALLS
Handset, Headset and Speakerphone
The GXP1400/1405 allows you to make phone calls via handset, headset or speakerphone. During the
active calls the user can switch between the handset, headset and the speakerphone by pressing the
corresponding keys on the phone.
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2 SIP Accounts and Lines
GXP1400/1405 can support up to 2 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. In
off-hook state, select an idle line and the dial tone will be heard.
To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. The user
can switch lines before dialing any number by pressing the LINE buttons.
For example: when LINE 1 is pressed, the LINE 1 LED will light up in green. If LINE 2 is pressed, the LINE 2
LED will light up in green 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 GXP140x will flash the other 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:
1. D
IAL: To make a phone call.
Take Handset off hook
or press SPEAKER button
or press HEADSET button
or press an available LINE key to activate speakerphone
The line will have a dial tone
Enter the phone number
Press “#” or HANDSET button to send
EDIAL: To redial the last dialed phone number.
2. R
Take Handset off-hook
or press the SPEAKER button
or press an available LINE key to activate speakerphone
or on idle screen
Press the REDIAL soft-key
IA CALL HISTORY:To call a phone number in the phone’s history.
3. V
Press the MENU button to bring up the Main Menu.
Select Call History and then “Answered Calls”, “Missed Calls” or “Dialed Calls” or etc
depending on your needs
Select phone number using the arrow keys
Press OK to select
Select and press “Dial” to dial out
IA PHONEBOOK: To Call a phone in from the phone’s phonebook.
4. V
Go to the phonebook by pressing the DOWN arrow key or pressing the menu button and
selecting “Phone Book”
Select the phone number by using the arrow keys
Press OK to select
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Select and press “Dial” to dial out
5. V
IA PAGE/INTERCOM: Server/PBX has to support Page/Intercom. Also, GXP1400/1405 and PBX have
to be configured correctly.
Take Handset off hook
or press SPEAKER button
or press HEADSET button
or press an available LINE key to activate speakerphone
Press OK button (the round button in the center of navigation keys) and the screen will display
“LINEx: PAGE”
Dial the number to Page/Intercom
Press “SEND” button to dial out
NOTE:
Dial-tone and dialed number display occurs after the handset is off-hook, or handset button is
pressed, or speaker button is pressed, or the line key is selected. After dialing the number, the
phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call.
Press “#” button to override the 4 second delay.
Making Calls using IP Addresses
Direct IP Call 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:
Press MENU button to bring up MAIN MENU
Select “Direct IP Call” using the arrow-keys
Press OK to select
Input the 12-digit target IP address. (Please see example below)
Press OK key to initiate call.
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 represents the dot “.”; the “#” key represents colon “:”. Press
OK to dial out.
The GXP1400/1405 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.
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).
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Firmware version: 1.0.1.110 Last Updated: 01/2012
For example:
192.168.0.2 calling 192.168.0.3 -- offhook the phone, dial #3 followed by #
192.168.0.2 calling 192.168.0.23 -- offhook the phone, dial #23 followed by #
192.168.0.2 calling 192.168.0.123 -- offhook the phone, dial #123 followed by #
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.
ANSWERING PHONE CALLS
Receiving Calls
1. Incoming single call: Phone rings with selected ring-tone. The corresponding LINE flashes in red.
Answer call by taking Handset off hook or pressing SPEAKER or HEADSET 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). Answer the incoming call by pressing its corresponding
LINE button. The current active call will be put on hold.
Do Not Disturb
Do Not Disturb can be enabled/disabled by pressing the MUTE/DND button on the phone when the phone is
in idle screen. Or users could set it from the MENU following the steps below.
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.
HONE FUNCTIONS DURING A PHONE CALL
P
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) will be heard when line is in use.
Mute
1. During the call, press the MUTE button to enable/disable muting the microphone.
2. The “Line Status Indicator” will show “LINEx: TALKING” or “LINEx: MUTE” to indicate whether the
microphone is muted.
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Firmware version: 1.0.1.110 Last Updated: 01/2012
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