Grandstream GXP1400, GXP1405 User Manual

Grandstream Networks, Inc.

GXP1400/1405 Small-Medium Business IP Phone

TABLE OF CONTENTS

GXP1400/1405 USER MANUAL

WELCOME ................................................................................................................................................................. 3

INSTALLATION......................................................................................................................................................... 4

EQUIPMENT PACKAGING ............................................................................................................................................. 4

CONNECTING YOUR PHONE ........................................................................................................................................ 4

SAFETY COMPLIANCES ................................................................................................................................................ 4

WARRANTY ................................................................................................................................................................. 4

PRODUCT OVERVIEW ............................................................................................................................................ 5

USING THE GXP1400/1405 ....................................................................................................................................... 8

GETTING FAMILIAR WITH THE LCD ............................................................................................................................ 8

MAKING PHONE CALLS ............................................................................................................................................... 9

ANSWERING PHONE CALLS ....................................................................................................................................... 12

PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 12

CALL FEATURES ........................................................................................................................................................ 15

CUSTOMIZED LCD SCREEN & XML ......................................................................................................................... 16

CONFIGURATION GUIDE ...................................................................................................................................... 17

CONFIGURATION VIA KEYPAD .................................................................................................................................. 17

CONFIGURATION VIA WEB BROWSER ...................................................................................................................... 20

SAVING THE CONFIGURATION CHANGES ................................................................................................................... 38

REBOOTING THE PHONE REMOTELY ......................................................................................................................... 38

SOFTWARE UPGRADE & CUSTOMIZATION .................................................................................................. 39

FIRMWARE UPGRADE THROUGH TFTP/HTTP .......................................................................................................... 39

CONFIGURATION FILE DOWNLOAD ........................................................................................................................... 40



RESTORE FACTORY DEFAULT SETTING ....................................................................................................... 41

TABLE OF TABLES

GXP1400/1405 USER MANUAL

Table 1: Equipment Packaging ....................................................................................................... 4



Table 2: GXP1400/1405 Connectors .............................................................................................. 4

Table 3: GXP1400/1405 Feature Guide ......................................................................................... 5

Table 4: GXP1400/1405 Key Features in a Glance ........................................................................ 5

Table 5: GXP1400/1405 Hardware Specifications ......................................................................... 5

Table 6: GXP1400/1405 Technical Specifications .......................................................................... 6

Table 7: LCD Display Definition ...................................................................................................... 8

Table 8: LCD Icons ......................................................................................................................... 8

Table 9: GXP1400/1405 KEYPAD BUTTONS................................................................................ 9

Table 10: GXP1400/1405 Call Features ....................................................................................... 15

Table 11: Key Pad Configuration Menu ........................................................................................ 17

Table 12: Keypad GUI Flow .......................................................................................................... 18

Table 13: Device Configuration - Status ....................................................................................... 21

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Table 14: Device Configuration – Settings/Basic Settings ............................................................ 21

Table 15: Device Configuration – Settings /Advanced Settings ................................................... 23

Table 16: SIP Account Settings .................................................................................................... 30

GUI INTERFACE EXAMPLES

GXP1400/1405 USER MANUAL

http://www.grandstream.com/products/gxp_series/general/documents/gxp21xx_gui.zip

1. Screenshot of Configuration Login Page

2. Screenshot of Status Page

3. Screenshot of Basic Setting Configuration Page

4. Screenshot of Advanced User Configuration Page

5. Screenshot of SIP Account Configuration Page

6. Screenshot of Saved Configuration Changes Page

7. Screenshot of Reboot Page

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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 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, 802.1x, TR-069

Advanced Digital Signal Processing (DSP), Silence Suppression, VAD,
CNG, AGC

10/100 Mbps Ethernet port, integrated PoE (GXP1405 only)

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, 5
navigation 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 Display
Expansion Module
Call 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
Dimension
Weight

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)

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

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)

Display
Feature Keys

Graphic LCD display, up to 4 level grayscale
HOLD, TRANSFER, CONF, LINE 1, LINE 2, MSG, SPEAKERPHONE,

HANDSET, HEADSET, MUTE/DND, NAVIGATION(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 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

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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 Pr ovisioning 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
Upgrades

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 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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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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Call Transfer

GXP1400/1405 supports both Blind and Attended transfer. Also, users could make auto-attended transfer

when this feature is enabled from web GUI.

1. Blind Transfer: Press “TRANSFER” button, then dial the number and press the # button to

complete transfer of active call.

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 the call is transferred.

3. Auto-Attended Transfer: Users could enable Auto-Attended Transfer under Web GUI->Advanced

Setting Page. During the first call, press “TRANSFER” hard button and it will bring up another line.
The first call will be on hold. Enter the number and press SEND or “#” key to establish the second
call. After the second call is established, users could press “TRANSFER” hard button to transfer the
call, or press the SPLIT soft key so the second call will be resumed.

NOTE:

 To transfer calls across SIP domains, SIP service providers must support transfer across SIP

domains.

3-Way Conferencing

GXP1400/1405 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

2. Cancel Conference:

 If after pressing the “CONF” button, a user decides not to conference anyone, press the

current active LINE button

 This will resume two-way conversation with the current line

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 LINE key

GXP1400/1405 also supports Easy Conference mode. In Easy Conference mode, users can initiate

conference by calling another number when the current line is in talking or conference. Also the conference
can be re-established by pressing the ReConf softkey when the conference is on hold. Easy Conference
mode can be used combined with the traditional ways to establish 3-way conference.

1. Initiate a Conference Call:

 Establish one call
 Press CONF button and a new line will be brought up
 Dial the number and press SEND button to establish the second call
 Press CONF button again or press the ConfCall softkey to establish the 3-way conference

2. Hold Conference:

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 During the conference, press HOLD button and the conference will be put on hold

- To resume the conference, press the ReConf softkey

- To split the conference and resume the call with each party, press the
corresponding line key

3. End Conference:

 If the users decide not to conference after establishing the second call, press EndCall

softkey instead of ConfCall softkey/CONF button. It will end the second call and the screen
will show the first call is on hold.

 During the conference, press EndCall softkey or hang up to end the conference

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.

 When using Easy Conference mode, press SEND button to establish the second call after entering

the number instead of using “#”.

Voice Messages (Message Waiting Indicator)

A blinking red MWI (Message Waiting Indicator) on the top right corner of the GXP1400/1405 indicates a
message is waiting. Dial into the voicemail box to retrieve the message. An IVR will prompt the user through
the process of message retrieval.

Shared Call Appearance (SCA)

The GXP1400/1405 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 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 red­flashing 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 the phone. In
addition, they would need to navigate to “Settings”->”Basic Settings” on the web UI and set the line to
“Shared Line”. If the user requires more shared call appearances, the user can configure multiple line
buttons to be “shared line” buttons associated with the account.

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Firmware version: 1.0.1.110 Last Updated: 01/2012

CALL FEATURES

The GXP1400/1405 supports traditional and advanced telephony features including caller ID, caller ID
w/name, call forward/transfer/park/hold as well as intercom/paging.

Table 10: GXP1400/1405 Call Features

Key Call Features

*30

Block Caller ID (for all subsequent calls)
Offhook and dial “*30”.

*31

Send Caller ID (for all subsequent calls)
Offhook and dial “*31”.

*67

Block Caller ID (per call)
Offhook, dial “*67” and then enter the number to dial out.

*82

Send Caller ID (per call)
Offhook, dial “*82” and then enter the number to dial out.

*70

Disable Call Waiting (per Call)
Offhook, dial “*70” and then enter the number to dial out.

*71

Enable Call Waiting (per Call)
Offhook, dial “*71” and then enter the number to dial out.

*72 Unconditional Call Forward

Offhook, dial “*72”. Then enter the number to forward the call. Press OK softkey or
SEND button.

*73 Cancel Unconditional Call Forward

Offhook, dial “*73” and the phone will hang up.

*90 Busy Call Forward

Offhook, dial “*90”. Then enter the number to forward the call. Press OK softkey or
SEND button.

*91 Cancel Busy Call Forward

Offhook, dial “*91” and the phone will hang up.

*92 Delayed Call Forward

Offhook, dial “*92”. Then enter the number to forward the call. Press OK softkey or
SEND button.

*93 Cancel Delayed Call Forward

Offhook, dial “*93” and the phone will hang up.

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Firmware version: 1.0.1.110 Last Updated: 01/2012

CUSTOMIZED LCD SCREEN & XML

GXP1400/1405 IP phone supports 1) XML Custom Screen and 2) XML Downloadable Phonebook. Please refer to
the following link for documentation and templates.

1) XML Custom Screen (custom idle screen logo, softkey layout and etc.)

http://www.grandstream.com/products/gxp_series/general/documents/GXP140x_XML_Screen_Customization.zip

2) XML Downloadable Phonebook

http://www.grandstream.com/products/gxp_series/general/documents/gxp_wp_xml_phonebook.pdf

Grandstream Networks, Inc. GXP1400/1405 User Manual Page 16 of 41

Firmware version: 1.0.1.110 Last Updated: 01/2012

Configuration Guide

The GXP1400/1405 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 softkey to confirm a menu selection. Press left arrow key can exit to the previous menu. 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 Pad Menu. The menu options available are listed in table 11.

Table 11: Key Pad Configuration Menu

Item Description

Call History

Status

Phone Book
LDAP Directory
Instant Messages
Direct IP Call
Preference

Displays histories of answered, dialed, missed, and transferred and forwarded

calls. Select “Clear All” to clear all the call history entries.

Displays the network status, account status, software version and hardware
version of the phone.

Press network status to enter the sub menu for IP setting information

(DHCP/Static IP/PPPoE), Subnet Mask, Gateway and DNS server.
Displays the phonebook and downloads phonebook XML
Displays the LDAP directory and downloads directory
Goes to instant messages
Dials IP address for direct IP call
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.

 Download SCR XML

The phone will download the custom idle screen if available.

 Erase Custom SCR

Custom idle screen will be erased and will be replaced with default
logo.

 Display Language

Users can choose English, Simplified Chinese, Traditional Chinese,
Korean, Japanese, Italian, Spanish, French, German, Portuguese,
Russian, Croatian, Hungarian, Polish, Slovenian, Arabic, Hebrew or

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Dutch 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

Users can set the date and time on the phone.

Press Menu button to choose the menu item

Press ‘←’ or follow the soft keys to return to the main menu

Config

Factory Functions

Network

Call Features

Press Menu button to display the configuration selections:

 SIP

To change SIP server settings for SIP account (SIP Proxy, Outbound
Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and
Audio).

 Upgrade

To configure the firmware server and Config server for upgrading or
provisioning the phone.

 Factory Reset

Key in the physical/MAC address on the back of the phone.
Press OK softkey to reset to 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.

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

is working 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

To select IP mode (DHCP/Static IP/PPPoE); to setup PPPoE, IP address,

Netmask, Gateway address and DNS Server 1 and DNS Server 2.
To enable/disable and configure Forward All, Forward Busy, Forward No Answer,

No Answer Timeout, select Call Features and press Account 1 to set the forward
call features.

Reboot
Exit

Select on Reboot and press Menu button to reboot the device.

Exit from this menu.

Table 12: Keypad GUI Flow

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y

Call Histor

Call History Items

MENU

Call History

Status

Phone Book

LDAP Directory

Instant
Message

Direct IP Call

Preference

Config

Factory
Functions

Network

Call Features

Reboot

Exit

Answered Calls
Dialed Calls
Missed Calls
Transferred Calls
Forwarded Calls
Clear All
Back

Phone Book

New Entry
Download Phonebook XML
Delete All Entries
Back

LDAP Directory

View Directory
Download Directory
Search Configuration
Back

Instant Message

Clear All
Back

Preference

Do Not Disturb
Ring Tone
Ring Volume
LCD Contrast
Download SCR XML
Erase Custom SCR
Display Language
Time Settings
Back

Config

SIP
Upgrade
Factory Reset
Layer 2 QoS
Back

Factory Function

Audio Loopback
Diagnostic Mode
Back

Network

IP Setting
PPPoE Settings
IP
Netmask
Gateway
DNS Server 1
DNS Server 2

Delete All Entries

New Entry

First Name:
Last Name
Number:
Acct:
Confirm Add:
Cancel & Return:

Search Configuration

Select Filter
Filter Value
Back

Do Not Disturb

Enable DND
Disable DND
Back

Ring Tone

Default Ring
Ring1
Ring2
Ring 3
Back

SIP

Account
SIP Proxy
Outbound Proxy
SIP User ID
SIP Auth ID
SIP Password
SIP Transport
Audio
Save
Cancel

Upgrade

Firmware Server
Config Server
Upgrade Via
Back

Layer 2 QoS

802.1Q/VLAN Tag
Priority value
Reset Vlan Config
Back

Diagnostic Mode

Keypad/LED Diagnostic

Account 1

Call Features

Account 1

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Forward All
Forward Busy
Forward No Answer
No Answer Timeout

Firmware version: 1.0.1.110 Last Updated: 01/2012

CONFIGURATION VIA WEB BROWSER

The GXP1400/1405 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded
HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft’s IE, Mozilla
Firefox and Google Chrome.

Access the Web Configuration Menu

To access the phone’s Web Configuration Menu

 Connect the computer to the same network as the phone
 Make sure the phone is turned on and shows its IP address
 Start a Web browser on your computer
 Enter the phone’s IP address in the address bar of the browser
 Enter the administrator’s password to access the Web Configuration Menu

1

The Web-enabled computer has to be connected to the same sub-network as the phone. This can easily

be done by connecting the computer to the same hub or switch as the phone is connected to. In absence
of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone using
the PC port on the phone.

2

If the phone is properly connected to a working Internet connection, the phone will display its IP address in
Menu->Status. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0 to 255.
You will need this number to access the Web Configuration Menu. For example, if the phone shows

192.168.0.60, please use “http://192.168.0.60” in the address bar of your browser.

3

The default administrator password is “admin”; the default end-user password is “123”.

NOTE:

 When changing any settings, always SUBMIT them by pressing “UPDATE” 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.

 All the options under Basic Setting and Account Setting, and most of the options under Advanced

Setting do not require reboot after submitting the changes. Under Advanced Setting, the parameters
on network configuration require reboot after update.

1

2

3

Definitions

This section will describe the options in the Web configuration user interface. As mentioned, a user can log in
as an administrator or end-user.

Functions available for the end-user are:

 Status: Displays the network status, account status, software version and MAC address of the

phone, and service status.

 Basic Settings: Basic preferences such as date and time settings, line keys and LCD settings can

be set here.

Additional functions available to administrators are:

 Advanced Settings: To set advanced network settings, codec settings, XML configuration settings

and etc.

 Account: To configure the SIP account.

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Table 13: Device Configuration - Status

MAC Address

IP Address
Product Model
Part Number
Software Version

System Up Time
System Time
Registered

The device ID, in HEXADECIMAL format.

This will be used for provisioning and is written on the label in the original box as well
as on the label located on the back panel of the device.

This field shows IP address of GXP1400/1405.

This field contains the product model information.

This field contains the product part number.

• 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

This field shows system up time since the last reboot.

This field shows the current time on the phone system.

Indicates whether accounts are registered to the related SIP server.

PPPoE Link Up

Indicates whether the PPPoE connection is enabled (connected to a modem) and the
NAT type.

Service Status

• GUI: shows the GUI status: running or stopped

• Phone: shows the phone status: running or stopped

Core Dump

Download core dump file for troubleshooting when necessary.

Table 14: Device Configuration – Settings/Basic Settings

End User Password

This contains the password for end user to access the Web Configuration Menu.
Users can enter new password here. This field is case sensitive with a maximum
length of 25 characters.

Confirm Password
IP Address

Enter the end user password again as above to confirm new password.

The GXP140x operates in three modes:

1. DHCP mode: The GXP140x acquires IP address from the first DHCP server

it discovers on the LAN. The DHCP option is reserved for NAT router mode.
In DHCP mode, all the field values for the Static IP mode are not used.

2. PPPoE mode: Set PPPoE account ID, PPPoE password and PPPoE service

name for the GXP140x to establish PPPoE sesstion.

3. Static IP mode: Configure IP address, Subnet Mask, Gateway, DNS Server

1, DNS Server 2 and Preferred DNS Server if Static IP mode is used.

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802.1x Mode

Line Keys x

Time Zone

Self-Defined Time
Zone

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:

1. Identity

2. MD5 Password

This allows the user to configure each line key and enable Shared Call Appearance
for the line. Options available for Key Mode are :

1. Line

2. Shared Line

This parameter controls the date/time display according to the specified time zone.
If “Allow DHCP Option 2 to override Time Zone setting” is checked, the time zone will
be overridden by the DHCP server.

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,M4.1.0,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.

M4.1.0,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, 3

rd

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 first Sunday of April to the
1st Sunday of November.

Weather Update

By default, “Enable Weather Update:” is set to “Yes”. If set to “No”, weather
information will not display on the phone.

Settings to customize the display of weather via:

 City Code – Automatic or enter city code (default is Automatic)
 Update Interval – Refresh time in minutes (default is 5 mins)
 Degree Unit – Select Automatic, Fahrenheit or Celsius (default is Automatic)

This is displayed when “Enable Weather Update” is set to “Yes” and pressing the
‘SwitchSCR’ soft-key once.

LCD Contrast

Date Display Format

Set LCD contrast. Range from 0 to 20.

LCD date display in the idle screen. The following formats are supported:

• yyyy-mm-dd: 2011-11-17

• mm-dd-yyyy: 11-17-2011

• dd-mm-yyyy: 17-11-2011

• dddd, MMMM dd: Thursday, November 17

• MMMM dd, dddd: November 17, Thursday

Time Display Format

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Firmware version: 1.0.1.110 Last Updated: 01/2012

LCD time display in 12 hour or 24 hour format.

Disable in-call DTMF
display

Always Ring Speaker

HEADSET Key Mode

Default is “No”. This field is used to hide the keypad input during a call.

Default is "No". This options is used when Headset is used on "Toggle
Headset/Speaker" mode.

If selected to "Yes", when the phone is in Headset mode, both headset and speaker
will ring on incoming call.

When headset is connected to GXP140x, users could use the HEADSET button in
two different modes:

• Default Mode:

- When GXP140x is in idle, press HEADSET button to offhook the phone and make
calls by using headset. Headset icon will display on the top of the call screen in
dialing/talking status

- When there is an incoming call, press HEADSET button to pick the call by using
headset

- When there is an active call using headset, press HEADSET button to hang up the
call

- When Speaker/Handset is used in dialing/talking status, press HEADSET button to
switch to headset. Press it again to hang up the call, or press Speaker/Handset to
switch back to previous mode

• Toggle Headset/Speaker

- When GXP140x is in idle, press HEADSET button to switch to Headset mode. The
idle screen will display a Headset icon. In this mode, if pressing Speaker button or
softkey, headset will be used by default

- When there is an incoming call, press LINE button, “Answer” softkey or Speaker
button, headset will be used

- When there is an active call, press HEADSET button to toggle between Headset
and Speaker

Headset TX gain (dB)
Headset RX gain (dB)
Handset TX gain (dB)

Set headset TX gain to -6, 0 or +6. Default is 0 db.

Set headset RX gain to -6, 0 or +6. Default is 0 db.

Set handset RX gain to -6, 0 or +6. Default is 0 db.

Table 15: Device Configuration – Settings /Advanced Settings

Admin
Password

Administrator password. Only the administrator can access the “Advanced Settings”
and “Account Settings” page. Password field is purposely blank for security reasons
after clicking update and saved. The maximum password length is 25 characters.

Confirm Password

Layer 3 QoS

Enter the end user password again as above to confirm new password.

This field defines the layer 3 QoS parameter. It is the value used for IP Precedence
or Diff-Serv or MPLS. Default value is 12.

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Layer 2 QoS

Local RTP port

Use Random Port

Keep-alive interval

Use NAT IP
STUN Server

Firmware Upgrade and
Provisioning

This contains the value used for layer 2 802.1Q/VLAN tag and 802.1p priority value.
Default setting is 0. VLAN supported equipment is required when configuring these
settings.

This parameter defines the local RTP 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; channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024
to 65400 and must be even. The default value is 5004.

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”.

This parameter specifies how often the GXP1400/1405 sends a blank UDP packet
to the SIP server in order to keep the “hole” on the NAT open. Default is 20
seconds.

NAT IP address used in SIP/SDP message. Default is blank.

IP address or Domain name of the STUN server. STUN resolution result will display
in the STATUS page of the Web UI.

Allows the user to select the following options for firmware upgrade:

 Always Check for New Firmware
 Check New Firmware only when F/W pre/suffix changes
 Always Skip the Firmware Check.

Firmware upgrade may take up to 10 minutes depending on network environment.
Do not interrupt the firmware upgrading process.

Note: Grandstream strongly recommends that the user upgrade firmware locally in

a LAN environment if using TFTP to upgrade. Please DO NOT interrupt the

upgrade process (especially the power supply) as this will damage the device.

XML Config File
Password

HTTP/HTTPS User Name

HTTP/HTTPS Password

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The password used for encrypting the XML configuration file using OpenSSL. This
is required for the phone to decrypt the encrypted XML configuration file.

The user name for the HTTP/HTTPS server.

The password for the HTTP/HTTPS server. It won’t display for security protection.

Upgrade Via

This field allows the user to choose the firmware upgrade/config server path
method: TFTP, HTTP or HTTPS.

• TFTP:

GXP140x retrieves the new firmware files or new configuration file from the
specified TFTP server path at boot time. If there is no new firmware file or
configuration file, the system will start the boot process using the existing firmware
or config file. If a TFTP server is configured and new firmware files are retrieved,
the new downloaded image is saved into the Flash memory. (Please do NOT
interrupt the TFTP upgrade process (especially the power supply) as this will
damage the device.

• HTTP:

GXP140x retrieves the new firmware files or new configuration file from the
specified URL or IP for the HTTP server. For example:

provisioning.mycompany.com:6688/Grandstream/1.0.0.6

Note: “:6688” is the specific TCP port where the HTTP server is listening; Omit if

using default port 80.

• HTTPS:

GXP140x retrieves the new firmware files or new configuration file from the
specified URL or IP for the HTTP server via a secured HTTP connection. For
example:

provisioning.mycompany.com

Note: HTTPS default port is 443.

Firmware Server Path

Defines the server path for the firmware server. It can be different from the
Configuration server which is used for provisioning.

For example:
firmware.mycompany.com:6688/Grandstream/1.0.0.6

Config Server Path

Defines the server path for provisioning; it can be different from the firmware server.

For example:
provisioning.mycompany.com:6688/Grandstream/gxp1400

Firmware File
Prefix/Postfix

Default is blank. If configured, GXP140x will request the firmware file with the
prefix/postfix and only the firmware with the matching encrypted prefix will be
downloaded and flashed into the phone.

Note: This setting is useful for ITSPs. End user should keep it blank.

Config File
Prefix/Postfix

Default is blank. If configured, GXP140x will request the config file with the
prefix/postfix and only the file with the matching encrypted prefix will be downloaded
and flashed into the phone.

Note: This setting is useful for ITSPs. End user should keep it blank.

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Allow DHCP Option 43
and Option 66 to
override server

Automatic Upgrade

Authenticate Conf File

Enable TR-069
ACS URL
TR-069 Username
TR-069 Password
Periodic Inform Enable

Periodic Inform Interval

Default is “Yes”. This allows device to get provisioned from the server automatically.

Default is “No”. Choose “Yes” to enable automatic HTTP upgrade and provisioning.

In “Check for upgrade every” field, enter the number of minutes to check the HTTP
server for firmware upgrade or configuration changes. When set to “No”, the phone
will only perform HTTP upgrade and configuration check once at boot up.

Note: This function is used by ITSP. End user should NOT touch these parameters.

Default is “No”. If set to “Yes”, configuration file would be authenticated before
acceptance. End user should use default setting.

Default is “No”.

URL for TR-069 Auto Configuration Servers (ACS).

Enter ACS username for TR-069.

Enter ACS password for TR-069.

Enable periodic inform. Default is “No”. If set to YES, device will send inform
packets to the ACS.

When enabling periodic inform, set up the periodic inform interval to send the inform
packets to the ACS.

Connection Request
Username

Connection Request
Password

Authentication Method
Connection Request

Port
Phonebook XML

Download
Phonebook XML Server

Path
Phonebook Download

Interval
Remove Manually-edited

entries on Downloads

Enter the connection request username. This is the user name for the ACS to
connect to this device.

Enter the connection request password. This is the password for the ACS to
connect to this device.

Select the authentication method among “No authentication”, “Basic” or “Digest”.

Enter the connection request port. This is the port for the ACS to connect to this
device.

Selects the file download mode for the download server. Users can choose from
TFTP/HTTP/No.

The URL/IP address of the phonebook download server.

The interval at which the phonebook will be downloaded from the download server
(in Minutes). The default setting is 0.

If set to “Yes”, the phone will remove the manually-edited entries in the old
phonebook list before downloading the new file. The default setting is set to “Yes”.

Note:

If there is a duplicate entry (same name, same number and same index) added
both manually and from phonebook xml file, only 1 entry will be saved and
displayed.

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Firmware version: 1.0.1.110 Last Updated: 01/2012

LDAP Directory
Idle Screen XML

Download
Download Screen XML

At Boot-up
Use custom filename

Idle Screen XML Server
Path

Offhook Auto Dial

Syslog Server

Syslog Level

IP address or domain name of LDAP script server.

Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use
Custom Filename” or not, and define the “XML server path”.

The phone will download the idle screen xml file if set to “Yes”. The default setting
is “No”.

The phone will use custom filename specified in XML server path if set to “Yes”.
The default setting is “No”.

Specify the idle screen XML server path.

To configure a User ID/extension to dial automatically when the phone is taken
offhook.

The IP address or URL of System log server. This feature is especially useful for
ITSPs.

Select the syslog level for GXP140x to report. Default is NONE.

The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are
sent based on the following events:

 product model/version on boot up (INFO level)
 NAT related info (INFO level)
 sent or received SIP message (DEBUG level)
 SIP message summary (INFO level)
 inbound and outbound calls (INFO level)
 registration status change (INFO level)
 negotiated codec (INFO level)
 Ethernet link up (INFO level)
 SLIC chip exception (WARNING and ERROR levels)
 memory exception (ERROR level)

The Syslog uses USER facility. In addition to standard Syslog payload, it contains

the following components: GXP_1400: [device MAC address][firmware version]

error message.

For example:

Nov 30 07:55:46 10.131.28.27 GXP1400_GUI:[00:0b:82:2b:3c:a4][1.0.1.108][-LCD­] IdleScreen/XmlApp is now visible

Send SIP Log

When setting the “Yes”, phone will send out SIP Log to syslog server. Default
setting is “No”.

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.

Allow DHCP Option 42
to override NTP server

SSL Certificate
SSL Private Key

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Default is “Yes”. This allows device gets provisioned for DHCP Option 42 from the
server automatically.

The user specified SSL certificate used for SIP over TLS.

The user specified SSL private key used for SIP over TLS.

SSL Private Key
Password

Distinctive Ring Tone

System Ring Tone

Call Progress Tones

User specified password to protect the private key above.

Select a Distinctive Ring Tone 1 through 3 for the caller ID. The Caller ID can be
configured as a particular caller ID or Alert-Info text.

When a particular call ID is configured in the Caller ID, the selected ring tone will be
used when the incoming call is from the Caller ID. System Ring Tone will be used
for all other calls.

If server supports Alert-Info, the Alert-Info text can be mapped to the 3 customized
ring tones. For example, if you configure the custom ring tone 1 user ID to “priority”,
that ring tone will be used if we receive INVITE with Alert-Info header in the
following format:
Alert-Info:;info=priority

System ring tone. Default is North American standard. User could adjust system
ring tone frequencies and cadences based on local telecom standard.

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
Disable Call

Default is “No”. If set to “Yes”, the call waiting feature will be disabled.

Default is “No”. If set to “Yes”, the call waiting tone will be disabled.

Waiting Tone
Disable Direct IP Calls
Use Quick IP Call Mode

Default is “No”. If set to “Yes”, direct IP calls will be disabled.

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”, offhook the phone and dial #XXX (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 section for details.
Disable Conference
Disable DND Button
Disable Transfer

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Default is “No”. If set to “Yes”, conference will be disabled.

Default is “No”. If set to “Yes”, the “DND” button on keypad will be disabled.

Default is “No”. If set to “Yes”, transfer will be disabled.

Auto-Attended Transfer
Configuration via

Keypad Menu

Enable STAR key
Keypad locking

Password to lock/unlock

Default is “No”. If set to “Yes”, the phone will use attended transfer by default.

Configures the access control of configurations via the phone keypad menu. There

are three modes:

 Unrestricted
 Basic Settings Only:

CONFIG option will not display in keypad MENU

 Constraint Mode:

CONFIG, FACTORY FUNCTIONS and NETWORK options will not display
in keypad MENU

Default is enabled. When the phone is in idle screen, press and hold STAR key for

4 seconds and the keypad will be locked.

To unlock the keypad, press and hold STAR key for 4 seconds. Enter the password

to unlock in the prompted window, and then press MENU key to enter.

The password to lock/unlock can be configured. If users forget the password to

unlock it, log into web GUI and change the password. Update and reboot the

phone. When the phone boots up, press and hold STAR key for 4 seconds and

enter the new password to unlock it.

Enter the password to lock the keypad in web GUI.

To unlock the keypad, enter the password in the prompted window in the phone’s

LCD screen.

Do not escape “#” as
%23 in SIP URI

Default is “No”. By default, # will be replaced as %23 in SIP URI.

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Display Language

Allows user to choose preferred display language in web UI and keypad UI.

Currently, the phone supports these languages: Arabic, German, English, Spanish,

French, Hebrew, Croatian, Hungarian, Italian, Japanese, Korean, Dutch, Polish,

Portuguese, Russian, Slovenian, Simplified Chinese and Traditional Chinese.

Note: The “Automatic” setting in language refers to Grandstream’s IP2Location

client which when connected to Internet would attempt to lookup a database (driven

by Grandstream) with the IP address for its geographical location.

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.txt". If the field “Language File postfix “has

"NL" string in it, then the phone will request "gxp_NL.txt" instead of "gxp.txt".

User can only load one secondary language.

Supported downloadable language: Czech, Croatian, Estonian, French, German,

Italian, Polish, Portuguese, Slovak, Slovenian and Spanish.

How to set up Download Language:

This is similar to updating firmware in your local network environment.

1. Get the language file gxp.txt ready. Make sure the file is using UTF-8 encoding.

2. Copy gxp.txt to the firmware server directory using your local TFTP or HTTP

server.

3. Access the advanced settings of the Web GUI, set “Display Language” to

“Download Language” and enter the server path in Firmware Server Path. Select

TFTP or HTTP for firmware upgrade.

4. Update and reboot the phone.

Download Device
Configuration

Download the current device configuration txt file. In the txt file, all the P values will

be displayed except for the password fields.

GXP140x has up to two line appearances, each with an independent SIP account. Each SIP account
requires its own configuration page. Their configurations are identical.

Table 16: SIP Account Settings

Account Active
Account Name
SIP Server

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This field indicates whether the account is active. The default value is “Yes”.

The name associated with each account.

SIP Server’s IP address or Domain name provided by VoIP service provider.

Secondary SIP Server

Outbound Proxy

SIP User ID

Authenticate ID

This field contains the URL or the IP address of a second SIP server.

When this field is configured, GXP140x will send out Registration requests and
Subscribe messages (except for message waiting) to the “SIP Server” and
“Secondary SIP Server” for the same account.

When making a call, GXP140x will use the registered primary “SIP server” first. If
this primary “SIP Server” is not available, the registered “Secondary SIP Server” will
be used.

If the primary “SIP Server” is not registered but “Secondary SIP Server” can be
registered, GXP140x will use the “Secondary SIP Server” directly when making a
call.

Note: Please do not configure duplicate SIP Server address in “SIP server” and

“Secondary SIP Server”.

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.

User account information provided by VoIP service provider (ITSP); either an actual
phone number or formatted like one.

SIP service subscriber’s Authenticate ID used for authentication. It can be identical
to or different from SIP User ID.

Authenticate Password

Name
DNS Mode

Primary IP

Backup IP 1
Backup IP 2
TEL URI

SIP Registration

Unregister on Reboot

SIP service subscriber’s account password for GXP1400/1405 to register to (SIP)
servers of ITSP.

SIP service subscriber’s name that is used for Caller ID display.

The default is set to A Record. If users wish to locate the server by DNS SRV, users
may select SRV or NATPTR/SRV. When "Use Configured IP" option is selected, if
SIP server is configured as domain name, phone will not send DNS query, but use
"Primary IP" or "Secondary IP" to send sip message if at least one of them are not
empty.

This option applies only if “Use Configured IP” is selected, the phone will send DNS
query to the Primary IP. Insert IP address here.

Insert the first back up IP here.

Insert the second back up IP here.

Default is “Disabled”. If the phone has an assigned PSTN telephone number, this
field should be turned on and a “User=Phone” parameter will be attached to the
“From” header in SIP request.

This parameter controls sending REGISTER messages to the proxy server. The
default setting is “Yes”.

Default is “No”. If set to “Yes”, the SIP user’s registration information will be cleared
on reboot.

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Register Expiration

Reregister Before
Expiration

Local SIP Port

SIP Registration Failure
Retry Wait Time

SIP T1 Timeout
SIP T2 Interval
SIP Transport
SIP URI Scheme when

using TLS
Use Actual Ephemeral

Port in Contact with
TCP/TLS

This parameter allows user to specify the time frequency (in minutes) that
GXP1400/1405 refreshes its registration with the specified registrar. The default
interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days).

This parameter allows user to specify the time frequency (in seconds) that
GXP1400/1405 sends out a re-registration request before the Register Expiration.
By default is 0 second.

This parameter defines the local SIP port used to listen and transmit. The default
value is 5060 for Account 1 and 5062 for Account 2.

Retry registration if the process failed. Default is 20 seconds.

RFC 3261 SIP T1 timer. Default is 0.5 second.

RFC 3261 SIP T2 timer. Default is 4 seconds.

Choose SIP Transport between UDP, TCP and TLS/TCP. Default is UDP.

When TLS/TCP is used (in “SIP Transport”), select “sip:” or “sips:” in this mode.
Default is “sips:”.

Enable to use actual ephemeral port in contact with TCP/TLS. Default is “No”.

Check Domain
Certificates

Remove OBP from
Route

Validate Incoming
Messages

Support SIP Instance ID

NAT Traversal

SUBSCRIBE for MWI

Enable to check the domain certificate. Default is “No”.

The SIP Extension notifies the SIP server that it is behind a NAT/firewall.

This configuration selects whether or not the incoming messages should be
validated.

Selects whether or not SIP Instance ID is supported.

This parameter activates the NAT traversal mechanism. It has options: No, STUN,
Keep-Alive, UPnP, Auto, VPN.

If selecting STUN and a STUN server is also specified, the phone performs
according to the STUN client specification. Using this mode, the embedded STUN
client detects if and what type of NAT/Firewall configuration is used. If the detected
NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use
its mapped public IP address and port in all of its SIP and SDP messages.

If selecting Keep-Alive with no specified STUN server, the GXP1400/1405 will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload
data) to the SIP server to keep the “hole” on the NAT open.

Default is “No”. When set to “Yes”, a SUBSCRIBE for Message Waiting Indication
will be sent periodically.

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SUBSCRIBE for
Registration

Feature Key
Synchronization

Proxy-Require
Voice Mail UserID

Send DTMF

DTMF Payload Type

Default is “No”. When set to “Yes” a SUBSCRIBE for Registration will be sent
periodically.

Default is “No”. This option is to synchronize DND/Call Forward features with
Broadsoft platform.

When set to “Yes”, a SUBSCRIBE will be sent out periodically to the server. Then
when DND/Call Forward features (Call Forward No Answer, Unconditional Call
Forward and Call Forward on Busy) are configured or changed on the phone and
Broadsoft server side, those features will be synchronized on the phone side and
Broadsoft server side.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

When configured, user can access messages by pressing “MSG” button. This ID is
usually the VM portal access number.

This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes:

- In audio: DTMF is combined in audio signal (not very reliable with low-bit-rate
codec)

- via RTP (RFC2833)

- via SIP INFO

Sends DTMF using RFC2833. The default is 101.

Early Dial
Dial Plan Prefix

Default is “No”. Use only if proxy supports 484 responses.

Sets the prefix added to each dialed number.

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A

Dial Plan

Dial Plan Rules:

1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d

2. Grammar: x - any digit from 0-9;
a) xx+ - at least 2 digit numbers
b) xx. - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
e) [147] - any digit of 1, 4, or 7
f) <2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand

• Example 1: {[369]11 | 1617xxxxxxx}

Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617

• Example 2: {^1900x+ | <=1617>xxxxxxx}

Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit
numbers

• Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}

llows any number with leading digit 1 followed by a 3 digit number, followed by any
number between 2 and 9, followed by any 7 digit number OR Allows any length of
numbers with leading digit 2, replacing the 2 with 011 when dialed.

3. Default: Outgoing – {x+}
Allow any length of numbers.

Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):

• ^1900x. - prevents dialing any number started with 1900

• <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing 7
numbers and 1617 area code will be added automatically

• 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits
length

• 011[2-9]x. - allows international calls starting with 011

• [3469]11 - allow dialing special and emergency numbers 311, 411, 611 and 911

Note: In some cases where the user wishes to dial strings such as *123 to activate

voice mail or other applications provided by their service provider, the * should be
predefined inside the dial plan feature. An example dial plan will be: { *x+ } which
allows the user to dial * followed by any length of numbers.

Delayed Call Forward

Time waited before the call is forward to a number or VM. Default is 20 seconds.

Wait Time
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

Enable/disable Call Log and select type of calls to log:

- Log All Calls

- Log Incoming/Outgoing only (Missed calls NOT recorded)

- Disable Call Log

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A

Session Expiration

Min-SE
Caller Request Timer

Callee Request Timer

Force Timer

UAC Specify Refresher

UAS Specify Refresher

The SIP Session Timer extension enables SIP sessions to be periodically
“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval
expires, if there is no refresh via a UPDATE or re-INVITE message, the session is
terminated.

Session Expiration is the time (in seconds) at which the session is considered timed
out, provided no successful session refresh transaction occurs beforehand. The
default value is 180 seconds.

Defines the minimum session expiration (in seconds). Default is 90 seconds.

If set to “Yes”, the phone will use session timer when it makes outbound calls if
remote party supports session timer.

If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.

If set to “Yes”, the phone will use session timer even if the remote party does not
support this feature. If set to “No”, the session timer is enabled only when the
remote party supports this feature. To turn off Session Timer, select “No” for Caller
Request Timer, Callee Request Timer, and Force Timer.

As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee
or proxy server as the refresher.

s a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.

Force INVITE

Enable 100rel

Account Ring Tone

Ring Timeout

Line-seize Timeout

Send Anonymous

Anonymous Call
Rejection

Auto Answer

Session Timer can be refreshed using INVITE method or UPDATE method. Select
“Yes” to use INVITE method to refresh the session timer.

PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional
responses (1xx series). This is required to support PSTN inter-networking.

There are 4 uniquely defined ring tones:

 System Ring Tone: when selected, all calls will ring with system ring tone.
 3 Customer Ring Tones: when selected, incoming calls from designated

account will play selected ring tone.

Defines how long ring will ring when receiving a call. Default is 60 seconds.

Defines how long before the line can be seized when Share Line is used. Default is
15 seconds.

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.

Default is “No”. If set to “Yes”, anonymous call will be rejected.

Default is “No”. If set to “Yes”, GXP1400/1405 will automatically switch on speaker
to answer the incoming call.

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Allow Auto Answer by
Call-Info

Refer-To Use Target
Contact

Transfer on Conference
Hangup

Check SIP User ID for
Incoming INVITE

Preferred Vocoder

SRTP Mode

Symmetric RTP

If the Call-Info header contains answer-after=0, the call be answered automatically.
This fields need to be set to Yes if users would like to have the phone to be
paged/intercom.

Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header
uses the transferred target’s Contact header information.

Defines whether or not the call is transferred to the other party if the initiator of the
conference hangs up.
Default setting is set to “No”.

Check the SIP User ID in Request URI. If they don’t match, the call will be rejected.

GXP1400/1405 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”.

Enable SRTP mode based on selection. Default is “No”.

Selects whether or not symmetric RTP is supported.

Silence Suppression

This controls the silence suppression/VAD feature of the audio codec G.723 and
G.729. If set to “Yes”, when silence is detected, a small quantity of VAD packets
(instead of audio packets) will be sent during the period of no talking. If set to “No”,
this feature is disabled.

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Voice Frames per TX

This field contains the number of voice frames to be transmitted in a single Ethernet
packet (be advised the IS limit is based on the maximum size of Ethernet packet is
1500 byte (or 120kbps)).

When setting this value, be aware of the requested packet time (ptime, used in SDP
message) is a result of configuring this parameter. This parameter is associated
with the first codec in the above codec Preference List or the actual used payload

type negotiated between the 2 conversation parties at run time. E.g., if the first

codec is configured as G.723 and the “Voice Frames per TX” is set to 2, then the
“ptime” value in the SDP message of an INVITE request will be 60ms because each
G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the
first codec is G.729 or G.711 or G.726, then the “ptime” value in the SDP message
of an INVITE request will be 20ms.

If the configured voice frames per TX exceeds the maximum allowed value, the IP
phone will use and save the maximum allowed value for the corresponding first
codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20
(x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms)
and 64 (x2.5ms) frames respectively.

Please be careful when editing these parameters. Adjusting these parameters will
also change the dynamic jitter buffer. The GXP1400/1405 has a patent dynamic
jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.

We recommend using the default settings provided. We do not recommend
adjusting these parameters if you are an average user. Incorrect settings will affect
the voice quality.

No Key Entry Timeout
Use # as Dial Key

G723 Rate
G726-32 Packing Mode
iLbc Frame Size
iLbc Payload Type
Jitter Buffer Type
Jitter Buffer Length

Conference URI
Special Feature

Default is 4 seconds.

This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If
set to “Yes”, the “#” key will immediately send the call. In this case, this key is
essentially equivalent to the “(Re)Dial” key. If set to “No”, the “#” key is included as
part of the dial string.

Encoding rate for G723 codec. By default, 6.3kbps rate is set.

Select “ITU” or “IETF” for G726-32 packing mode.

ilbc packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required.

Payload type for Ilbc. Default value is 97. The valid range is between 96 and 127.

Jitter buffer type: Fixed or Adaptive. Default value is Adaptive.

Jitter buffer length. Default value is 300ms. The valid range is between 100ms and
800ms.

Configure the conference URI when using Broadsoft N-way calling feature.

Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.

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SAVING THE CONFIGURATION CHANGES

After the user makes a change to the configuration, press the “Update” button in the Configuration Menu.
The web browser will then display a message window to confirm saved changes.

We recommend rebooting or powering cycle the IP phone after saving changes.

REBOOTING THE PHONE REMOTELY

Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The web
browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in
again.

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Firmware version: 1.0.1.110 Last Updated: 01/2012

Software Upgrade & Customization

Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configuration
settings are in the ADVANCED SETTINGS configuration page.

FIRMWARE UPGRADE THROUGH TFTP/HTTP

To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set to
a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples
of some valid URLs.

 firmware.mycompany.com:6688/Grandstream/1.2.3.5
 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.

Key Pad Menu

To configure the Upgrade Server via Key Pad Menu options, select “Config” from the Main Menu, then select
“Upgrade”. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN

format. Choose “Save and use TFTP” or “Save and use HTTP” to select upgrade method. Select “Reboot”

from the Main Menu to reboot the phone.

Web Configuration Interface

To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the
GXP1400/1405 IP address. Enter the admin password to access the web configuration interface. In the
ADVANCED SETTINGS page, enter the Upgrade Server’s IP address or FQDN in the “Firmware Server
Path” field. Select TFTP or HTTP upgrade method. Update the change by clicking the “Update” button.
“Reboot” or power cycle the phone to update the new firmware.

During this stage, the LCD will display the firmware file downloading process. 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. We
recommend completing firmware upgrades in a controlled LAN environment whenever possible.

No Local TFTP/HTTP Server

For users who do not have a local TFTP/HTTP server, we provide a HTTP server on the public Internet for
users to download the latest firmware upgrade automatically. Please check the Support/Download section of
our website to obtain this HTTP server IP address: http://www.grandstream.com/support/firmware

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

.

.

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INSTRUCTIONS FOR LOCAL TFTP UPGRADE:

1. Unzip the file and put all of them under the root directory of the TFTP server.

2. The PC running the TFTP server and the GXP1400/1405 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 GXP1400/1405 phone boots up, it will send TFTP or HTTP request to download configuration

file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP1400/1405 phone.
This file is for provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following

error messages in a TFTP or HTTP server log can be ignored: “TFTP Error from [IP ADRESS]
requesting cfg000b82023dd4 : File does not exist. Configuration File Download”

CONFIGURATION FILE DOWNLOAD

The GXP1400/1405 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 link below to download the corresponding configuration
template of the firmware.

http://www.grandstream.com/support/tools

Once the GXP1400/1405 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.

For more details on XML provisioning, please refer to http://www.grandstream.com/support

Managing Firmware and Configuration File Download

When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre­scheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider can
manage and reduce the Firmware or Provisioning Server load at any given time.

.

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Restore Factory Default Setting

WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone.

Please backup or print all the settings before you restoring factory default settings. We are not responsible
for restoring lost parameters and cannot connect your device to your VoIP service provider.

INSTRUCTIONS FOR RESTORATION:

Step 1: Press “OK” button to bring up the keypad configuration menu, select “Config”, press “OK” to

enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)

Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following mapping:

0-9: 0-9
A: 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

Example: if the MAC address is 000b
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.

8200e395, it should be key in as “0002228200333395”.

Grandstream Networks, Inc. GXP1400/1405 User Manual Page 41 of 41

Firmware version: 1.0.1.110 Last Updated: 01/2012