CONNECTING YOUR PHONE ........................................................................................................................................ 4
USING THE GXP1400/1405 ....................................................................................................................................... 8
GETTING FAMILIAR WITH THE LCD ............................................................................................................................ 8
MAKING PHONE CALLS ............................................................................................................................................... 9
PHONE FUNCTIONS DUR IN G A PHONE CALL ............................................................................................................. 12
CALL FEATURES ........................................................................................................................................................ 14
CONFIGURATION VIA KEYPAD .................................................................................................................................. 16
SAVING THE CONFIGURATION CHANGES ................................................................................................................... 33
REBOOTING THE PHONE REMOTELY.......................................................................................................................... 33
Grandstream Networks, Inc. GXP1400/1405 User Manual Page 2 of 36
Firmware ver s ion: 1.0 .1.83 Last Updated: 08/2011
Welcome
GXP1400/1405 is a next generation small-to-m edium business IP phone that feat ures 2 lines with 1 SIP
account, a 128x40 graphic al LCD, 3 XML programm able context-sensitive sof t keys, dual network ports
with integrated PoE (GXP1405 only), and 3-wa y c onference. The GXP1400/1405 delivers super ior 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 interop erabilit y with m ost 3
is a perfect choice for small-to-medium businesses looking for a high quality, feature r ich IP phone with
affordable cost.
Caution: Changes or modificati ons to this prod uct not ex pressly approv ed by Grandstream, or operati on
of this product in any way other than as detailed by this User Manual, could void your manufacturer
warranty.
Warning: P lease do not u se a different power adapt or 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 trans mittal of the ent ire or a ny part, i n any f orm or by any means , elec tronic or pr int,
for any purpose without the express written permission is not permitted.
rd
party SIP devices and leading SI P/NGN/IMS p latform s. It
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GXP1400/1405
Main Case
Yes
Handset
Yes
Phone Cord
Yes
Power Adaptor
Yes (GXP1400 only)
Ethernet Cable
Yes
Base Stand
Yes
Quick Start Guide
Yes
Installation
EQUIPMENT PACKAGING
Table 1: Equipment Packaging
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 an d various saf ety 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.
WARRANTY
If you purchased your GXP1400/1405 from a reseller, please contact the compan y where you purchase d
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 Display
Number of Lines
Programmable Soft Keys
Extension Module
Table 4: GXP1400/1405 Key Features in a Glance
Features Benefits
Open Standards
Compatibility
Superb Audio Quality
Network Interfaces
Feature 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, SIP over TLS, 802.1x,
TR-069
Advanced Digital Signal Processing (DSP), Silence Suppression, VAD,
CNG, AGC
10/100 Mbps Ethernet port, integrate d Po E ( GX P140 5 only)
Traditional voice featur es including caller ID, call waiting, hold, transfer,
forward, block, auto-dial, off-hook dial
2 line keys with dual-color LED and 1 SIP account, 3 way conference,
Customized downloadable ring-tones, SRTP, SIP over TLS, multilanguage 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 por t with aut o detec ti on
Integrated PoE (GXP1 405 only)
Graphic LCD Display
Expansion Module
Call Appearance LED
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Firmware ver s ion: 1.0 .1.83 Last Updated: 08/2011
128 x 40 pixel
N/A
2 Dual color (green/red) line keys
Unit weight:0.7KG
Lines
Display
LDAP), support for anonymous call using privacy header, MLS (multi
Universal Switching
Power Adaptor
Dimension
Input: 100-240VAC 50-60 Hz
Output: +5VDC, 800mA, 4.0 W, UL certified
186mm (W) x 210mm (L) x 81mm (D)
Weight
Package weight: 1.1KG (GXP1400), 1.0KG (GXP1405)
Temperature
Humidity
Compliance
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
2 lines with 1 SIP ac count, 3 XML programmable soft-keys
Protocol Support
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP, SRTP by SDES, HTTP,
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)
Graphic LCD display, up to 4 level grayscale
Feature Keys
HOLD, TRANSF ER, CONF, LINE 1, LIN E 2, MSG, SPEAKERPHON E,
HANDSET, HEADSET, MUTE/DND, NAVIGAT ION(5), VOLUME, 3 XML
Programmable Soft 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
Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS,
DiffServ, MPLS)
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 ac ti vit y detec tio n), C NG (c om f ort 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 doubletalk performance
Telephony Features
Intuitive graphic user inter face (GUI), download able phone book (XML,
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Firmware ver s ion: 1.0 .1.83 Last Updated: 08/2011
language support)
auto answer and early dial
Voice mail indicator, downloadable custom ring-tones, call hold, call
transfer (attended/ blind), call f orward, c all waiti ng, call er ID, m ute, r edia l,
call log, caller ID display or block, Do-Not-Disturb (DND) and volume
control
3-way conference, d ial plan prefix, dial-plan support, off-hook auto dia l,
Network and Provisioning
Firmware
Upgrades
Advanced Server Features
Security
Via keypad/LCD, Web browser, or secure (AES encrypted) central
configuration file, m anual o r d ynamic hos t conf igurati on protoc ol (D HCP)
network setup
Support NAT traversal using IETF STUN and Symmetric RTP
Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 ToS
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
Message waiting indication, support DNS SRV Look up and SIP Server
Fail Over, Support customizable idle screen via downloading XML by
HTTP/TFTP
User and administrator level pass words , MD5 and MD5-sess based
authentication, AES based secure configuration file, SRTP, TLS, 802.1x
media access control
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Displays the current date and time. It can be synchronized with Internet time
The softkeys are context sensitive and will change depending on the status of
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
servers
Displays company logo name. This logo name can be customized via xml screen
customization. The maximum size for logo name is 22 characters in English
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 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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Firmware ver s ion: 1.0 .1.83 Last Updated: 08/2011
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 I con:
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 ent er 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
MAKINGPHONE 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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Dual Lines with SIP Account
GXP1400/1405 can support up to two lines “virtu ally” m apped to a SIP account. In off-hook state, se lect an
idle line and the dial ton e will be heard. To m ake a call, select the line you wish to use. The user can s witch
lines before dialing a n y number by pressing the LINE button.
Completing Calls
There are FIVE ways to complete a call:
IAL: To make a phone call.
1. D
•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 Ma in Men u.
• 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
• Select and press “Dial” to dial out
5. V
IA PAGE/INTERCOM: Server/PBX has to support Page/I nterc om. Also, GXP1400/1 405 and P BX h av e
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 and the screen will display “LINEx : PAGE”
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• 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 with out 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 conne ct ed throu gh a r out er usin g pub lic or pri vat e IP a ddr ess es (with nec es s ary
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 tar get IP address is 192.168.1.60 a nd the port is 5062 (e.g. 19 2.168.1.60:50 62), input
the following: 192*168*1*60#5062. The “*” key repres ents the dot “.”; the “#” key repres ents 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 fe w digit s ( last octet) of the t arget phone ’s IP-number. T his is pos sible on ly if both ph ones
are in under the same LA N/VPN. This simulates a PBX function using the C MSA/CD without a SIP server.
Controlled static IP usage is recommended.
To enable Quick IP calls , the pho ne has to be setup f ir st. T his is done thr oug h the web-setup function. In th e
“Advanced Settings” p age, set the "Use Quick IP-call mode” to “Yes”. When #xxx is dialed, where x is 0-9
and xxx <=255, a direc t IP call to aaa.bbb.cc c.XXX is complete d. “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 followed by #
192.168.0.2 calling 192.168.0.23 -- dial #23 followed by #
192.168.0.2 calling 192.168.0.123 -- 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.
Grandstream Networks, Inc. GXP1400/1405 User Manual Page 11 of 36
Firmware ver s ion: 1.0 .1.83 Last Updated: 08/2011
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