Grandstream Networks Gxp1400 Users Manual GXP User

Grandstream Networks, Inc. GXP1400/1405 User Manual Page 1 of 1

Firmware version 1.0.1.83 Last Updated: 08/2011

Grandstream Networks, Inc.

GXP1400/1405 Small-Medium Business IP Phone

Grandstream Networks, Inc. GXP1400/1405 User Manual Page 1 of 36

Firmware version 1.0.1.83 Last Updated: 08/2011

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 ........................................................................................................................................................ 14

CUSTOMIZED LCD SCREEN & XML ......................................................................................................................... 15

CONFIGURATION GUIDE ...................................................................................................................................... 16

CONFIGURATION VIA KEYPAD .................................................................................................................................. 16

CONFIGURATION VIA WEB BROWSER ...................................................................................................................... 19

SAVING THE CONFIGURATION CHANGES ................................................................................................................... 33

REBOOTING THE PHONE REMOTELY.......................................................................................................................... 33

SOFTWARE UPGRADE & CUSTOMIZATION .................................................................................................. 34

FIRMWARE UPGRADE THROUGH TFTP/HTTP .......................................................................................................... 34

CONFIGURATION FILE DOWNLOAD ........................................................................................................................... 35

RESTORE FACTORY DEFAULT SETTING ....................................................................................................... 36

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 ....................................................................................... 14

Table 11: Key Pad Configuration Menu ........................................................................................ 16

Table 12: Keypad GUI Flow .......................................................................................................... 17

Table 13: Device Configuration - Status ....................................................................................... 20

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Table 14: Device Configuration – Settings/Basic Settings ............................................................ 20

Table 15: Device Configuration – Settings /Advanced Settings ................................................... 22

Table 16: SIP Account Settings .................................................................................................... 27

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 1 SIP

account, 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 3rd party SIP devices and leading SIP/NGN/IMS platforms. It

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 permitted.

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Installation

EQUIPMENT PACKAGING

Table 1: Equipment Packaging

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

CONNECTING YOUR PHONE

The connectors of the GXP1400/1405 are located on the bottom of the device.

Table 2: GXP1400/1405 Connectors

PC 10/100Mbps RJ-45 ports for PC (downlink) connection

LAN 10/100Mbps RJ-45 port for LAN (uplink) connection, integrated PoE (GXP1405

only)

Power Jack 5V DC power port; UL Certified

Handset Jack RJ9

Headset Jack 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.

WARRANTY

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 128 x 40 pixel

Number of Lines 2

Programmable Soft Keys 3

Extension Module N/A

Table 4: GXP1400/1405 Key Features in a Glance

Features Benefits

Open Standards

Compatibility 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

Superb Audio Quality Advanced Digital Signal Processing (DSP), Silence Suppression, VAD,

CNG, AGC

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

Feature Rich Traditional voice features including caller ID, call waiting, hold, transfer,

forward, block, auto-dial, off-hook dial

Advanced Features 2 line keys with dual-color LED and 1 SIP account, 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

Advanced Functionality Customized downloadable ring-tones, SRTP, SIP over TLS, multi-

language support and XML enabled, adjustable positioning angles, wall

mountable, AES encryption, automatic multimedia service (eg., weather

information)

Table 5: 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 128 x 40 pixel

Expansion Module N/A

Call Appearance LED 2 Dual color (green/red) line keys

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Universal Switching Input: 100-240VAC 50-60 Hz

Power Adaptor Output: +5VDC, 800mA, 4.0 W, UL certified

Dimension 186mm (W) x 210mm (L) x 81mm (D)

Weight

Unit weight: 0.7KG

Package weight: 1.1KG (GXP1400), 1.0KG (GXP1405)

Temperature 32 -104° F/ 0 - 40°C

Humidity 10% - 90% (non-condensing)

Compliance FCC Part 15 (CFR 47) Class B

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 1 SIP account, 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)

Display

Graphic LCD display, up to 4 level grayscale

Feature Keys

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

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language support)

Voice mail indicator, downloadable custom ring-tones, call hold, call

transfer (attended/blind), call forward, call waiting, caller ID, mute, redial,

call log, caller ID display or block, Do-Not-Disturb (DND) and volume

control

3-way conference, dial plan prefix, dial-plan support, off-hook auto dial,

auto answer and early dial

Network and Provisioning Via keypad/LCD, Web browser, or secure (AES encrypted) central

configuration file, manual or dynamic host configuration protocol (DHCP)

network setup

Support NAT traversal using IETF STUN and Symmetric RTP

Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 ToS

Firmware

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

Displays the current date and time. It can be synchronized with Internet time

servers

LOGO NAME 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

NETWORK

STATUS Shows the status of network in the middle of the screen. It will indicate whether

the network is down or starting

STATUS BAR Shows the status of the phone, using icons as shown in the next table

SOFTKEYS

The softkeys are context sensitive and will change depending on the status of

the phone. Typical functions assigned to soft-buttons are:

• FORWARD ALL Unconditionally forwards the phone line to another

phone

• MISSED CALL This option shows 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

Table 8: LCD Icons

LCD Icons Descriptions

SIP Registration Status Icon:

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

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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 Place active call on hold

TRANSFER Transfer an active call to another number

CONF Press CONF button to connect Calling/Called party into conference

LINE 1 / LINE 2 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 “+”

0 - 9, *, # 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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Dual Lines with SIP Account

GXP1400/1405 can support up to two lines “virtually” mapped to a 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 user can switch

lines before dialing any number by pressing the LINE button.

Completing Calls

There are FIVE ways to complete a call:

1. DIAL: 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

2. REDIAL: To redial the last dialed phone number.

• 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

3. VIA CALL HISTORY: To call a phone number in the phone’s history.

• 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

4. VIA PHONEBOOK: To Call a phone in from the phone’s phonebook.

• 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. VIA 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 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 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).

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.

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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. 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.

PHONE FUNCTIONS DURING A PHONE CALL

Call Waiting/Call Hold

1. Hold: Place a call on ‘hold’ by pressing the “HOLD” button.

2. Resume: Resume call by pressing the corresponding blinking LINE.

3. Multiple Calls: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE to

place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.

Mute

1. 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.

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.

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2. Attended Transfer: Press “LINEx” button to make a call and automatically place the ACTIVE LINE

on HOLD. Once the call is established, press “TRANSFER” key then the LINE button of the waiting

line to transfer the call. Hang up the phone call after 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 HOLD

or the original LINE button

 This will resume two-way conversation

3. End Conference:

 Press HOLD to end the conference call and put all parties on hold

 To speak with an individual party, select the corresponding 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:

 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:

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 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 red-

flashing button and they will be able to resume the call from their phone by pressing the line button. However,

if this call is placed on private-hold, no other member of the SCA group will be able to resume that call.

To enable shared call appearance, the user would need to register the shared line account on 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.

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

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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 and press “#” or OK

softkey.

*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 and press “#” or OK

softkey.

*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 and press “#” or OK

softkey.

*93 Cancel Delayed Call Forward

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

CUSTOMIZED LCD SCREEN & XML

GXP1400/1405 IP phone support both simple and advanced XML applications: 1) XML Custom Screen and 2)

XML Downloadable Phonebook. For more information on how to create a downloadable XML phonebook, creating

a custom idle screen and/or reprogramming the soft-keys on GXP1400/1405, please visit our website at

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

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

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

Status 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.

Phone Book Displays the phonebook and downloads phonebook XML

LDAP Directory Displays the LDAP directory and downloads directory

Instant Messages Goes to instant messages

Direct IP Call Dials IP address for direct IP call

Preference Press Menu button to enter this sub menu including:

• Do NOT Disturb

DND (Do Not Disturb) function could be turned on or off in the “Do Not

Disturb” menu.

• Ring Tone

Choose different ring tones in the “Ring Tone” menu.

• Ring Volume

Press Menu button to hear the selected ring volume, press ‘←’ or ’ →’

to hear and adjust the ring tone volume.

• LCD Contrast

Press ‘←’ or ’ →’ to adjust the LCD contrast.

• 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 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.

Factory Functions

Press Menu to display the factory function items including

• Audio Loopback

Speak into the handset. If you hear your voice in the handset, your audio

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

Network

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

Netmask, Gateway address and DNS Server 1 and DNS Server 2.

Call Features

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

Exit Exit from this menu.

Table 12: Keypad GUI Flow

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Call History

Status

Phone Book

LDAP Directory

Instant

Message

Direct IP Call

Preference

Config

Factory

Functions

Network

Call Features

Reboot

Exit

MENU

Answered Calls

Dialed Calls

Missed Calls

Transferred Calls

Forwarded Calls

Clear All

Back

Delete All Entries

New Entry

Download Phonebook XML

Delete All Entries

Back

Phone Book

First Name:

Last Name

Number:

Acct:

Confirm Add:

Cancel & Return:

New Entry

View Directory

Download Directory

Search Configuration

Back

LDAP Directory

Select Filter

Filter Value

Back

Search Configuration

Clear All

Back

Preference

Do Not Disturb

Ring Tone

Ring Volume

LCD Contrast

Download SCR XML

Erase Custom SCR

Display Language

Time Settings

Back

SIP

Upgrade

Factory Reset

Layer 2 QoS

Back

Audio Loopback

Diagnostic Mode

Back

Factory Function

Config

Enable DND

Disable DND

Back

Default Ring

Ring1

Ring2

Ring 3

Back

Do Not Disturb

Ring Tone

Account

SIP Proxy

Outbound Proxy

SIP User ID

SIP Auth ID

SIP Password

SIP Transport

Audio

Save

Cancel

Firmware Server

Config Server

Upgrade Via

Back

802.1Q/VLAN Tag

Priority value

Reset Vlan Config

Back

Upgrade

Layer 2 QoS

Instant Message

Keypad/LED Diagnostic

Diagnostic Mode

SIP

IP Setting

PPPoE Settings

IP

Netmask

Gateway

DNS Server 1

DNS Server 2

Back

Network

Call History Items

Account 1

Call History

Call Features

Forward All

Forward Busy

Forward No Answer

No Answer Timeout

Account 1

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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 phone1

• 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 browser2

• Enter the administrator’s password to access the Web Configuration Menu3

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.

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

The device ID, in HEXADECIMAL format.

IP Address

This field shows IP address of GXP1400/1405.

Product Model

This field contains the product model information.

Part Number

This field contains the product part number.

Software Version

• Program: This is the main firmware release number, which is always used for

identifying the software (or firmware) system of the phone.

• Boot: Booting code version number

• Core: Core code version number

• Base: Base code version number

• DSP: DSP code version number

• Aux: Aux code version number

System Up Time

This field shows system up time since the last reboot.

System Time

This field shows the current time on the phone system.

Registered

Indicates whether accounts are registered to the related SIP server.

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 to access the Web Configuration Menu. This field is case

sensitive with a maximum length of 25 characters.

IP Address

The GXP1400/1405 operates in three modes:

1. DHCP mode: The GXP1400/1405

acquires its IP address from the first

DHCP server it discovers on its LAN. The DHCP option is

reserved for NAT

router mode. In DHCP mode, a

ll the field values for the Static IP mode are

not used (even though they are still saved in the Flash memory).

2. PPPoE mode: To use the PPPoE feature, set the PPPoE account settings

(PPPoE account ID, PPPoE password and PPPoE service name).

The

GXP1400/1405 establishes a PPPoE session if any of the PPPoE fields is

set.

3. Static IP mode: Configure all of the following fie

lds: IP address, Subnet

Mask, Gateway, DNS Server 1, DNS Server 2 and Preferred DNS Server.

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802.1x Mode

This option allows the user to enable/disable 802.1x mode on the phone. The default

value is disabled. To enable 802.1x mode, this field should be set to EAP-MD5.

Once

enabled, the user would be required to enter the following information below to be

authenticated on the network:

• Identity

•

MD5 Password

Line Keys x

This allows the user to configure the account mapped to each line key, as well as

enabling SCA (Shared Call Appearance) for the line.

Options available for Key Mode are :

1. Line

2. Shared Line

Time Zone

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.

Self-Defined Time

Zone This parameter allows the users to define their own time zone.

The syntax is: std offset dst [offset], start [/time], end [/time]

Default is set to: MTZ+6MDT+5,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, 3rd

Tuesday…)

The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, … ,Sat)

Therefore, this example is the DST which starts from the 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

Set LCD contrast. Range from 0 to 20.

Time Display Format

LCD time display in 12 hour or 24 hour format.

Disable in-call DTMF

display

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

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HEADSET Key Mode

Default Mode:

- Toggle to Headset when using Speaker/Handset

- Dial, pick up call or hang up call using Headset

Toggle Headset/Speaker:

- toggle between using Headset and using Speaker

Headset TX gain (dB)

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

Headset RX gain (dB)

Set headset 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.

Layer 3 QoS

This field defines the layer 3 QoS parameter. It is the value used for IP Precedence

or Diff-Serv or MPLS. Default value is 12.

Layer 2 QoS

This contains the value used for layer 2 802.1Q/VLAN tag and 802.1p priority value.

Default setting is 0.

Local RTP port

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.

Use Random Port

This parameter, when set to “Yes”, will force random generation of both the local

SIP and RTP ports. This is usually necessary when multiple GXP

s are behind the

same NAT. Default is “No”.

Keep-alive interval

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.

Use NAT IP

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

STUN Server

IP address or Domain name of the STUN server. STUN resolution result will display

in the STATUS page of the Web UI.

Firmware Upgrade and

Provisioning

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.

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XML Config File

Password

The password used for encrypting the XML configuration file using OpenSSL. This

is required for the phone to decrypt the encrypted XML configuration file.

HTTP/HTTPS User Name

The user name for the HTTP/HTTPS server.

HTTP/HTTPS Password

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 method: TFTP, HTTP or

HTTPS.

Firmware Server Path

Defines the server path for the firmware server. It can be different from the

Configuration server which is used for provisioning.

Config Server Path

Defines the config server path for provisioning; it can be different from the Firmware

server.

Firmware File

Prefix/Postfix Default is blank. If configured, GXP1400/1405 will request the firmware file with the

prefix/postfix and only the firmware with the ma

tching encrypted prefix will be

downloaded and flashed into the phone.

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

Config File

Prefix/Postfix

Default is blank. If configured, GXP1400/1405 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.

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

Allow DHCP Option 43

and Option 66 to

override server

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

Automatic Upgrade

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

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.

Authenticate Conf File

Default is “No”. If set to “Yes”, configuration file would be authenticated before

acceptance. End user should use default setting.

Enable TR-069

Default is “No”.

ACS URL

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

TR-069 Username

Enter username for TR-069.

TR-069 Password

Enter password for TR-069.

Periodic Inform Enable

Enable periodic inform. Default is “No”.

Periodic Inform Interval

When enabling periodic inform, set up the periodic inform interval.

Connection Request

Username

Enter the connection request username.

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Connection Request

Password

Enter the connection request password.

Authentication Method

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

Connection Request

Port

Enter the connection request port.

Phonebook XML

Download

Selects the file download mode for the download server. Users can choose from

TFTP/HTTP/No.

Phonebook XML Server

Path

The URL/IP address of the phonebook download server.

Phonebook Download

Interval

The interval at which the phonebook will be downloaded from the download server

(in Minutes). The default setting is 0.

Remove Manually-edited

entries on Downloads

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”.

LDAP Directory

IP address or domain name of LDAP script server.

Idle Screen XML

Download

Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use

Custom Filename” or not, and define the “XML server path”.

Download Screen XML

At Boot-up

The phone will download the idle screen xml file if set to “Yes”. The default setting

is “No”.

Use custom filename

The phone will use custom filename specified in XML server path if set to “Yes”.

The default setting is “No”.

Idle Screen XML Server

Path

Specify the idle screen XML server path.

Offhook Auto Dial

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

offhook.

Syslog Server

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

ITSPs.

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Syslog Level

Select the ATA to report the log level. 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: GS_LOG: [device MAC address][error code]

error

message.

For example: May 19 02:40:38 192.168.1.14 GS

_LOG: [00:0b:82:00:a1:be][000].

Ethernet link is up.

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

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

server automatically.

SSL Certificate

This defines the SSL certificate needed to access certain websites.

SSL Private Key

This defines the SSL Private key.

SSL Private Key

Password

This defines the SSL private key password.

Distinctive Ring Tone

Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a

particular Caller ID. The GXP1400/1405 will ONLY use selected ring to

nes for

particular Caller IDs. For all other calls, the GXP1400/1405

will use System Ring

Tone. When selected and no Caller ID is configured, the selected ring tone will be

used for all incoming calls.

System Ring Tone

System ring tone. Default is North American standard.

Adjust system ring tone frequencies and cadences based on local telecom

standard.

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Call Progress Tones

Using these settings, users can configure ring or tone frequencies based on

parameters from local telecom. By default, they are set to North American standard.

Frequencies should be configured with known values to avoid uncomfortable high

pitch sounds.

Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];

(Frequencies are in Hz and cadence on and off are in 10ms)

ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In

order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms

and a pause of OFF ms and then repeat the pattern. Up to three cadences are

supported.

Disable Call Waiting

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

Disable Call

Waiting Tone

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

Disable Direct IP Calls

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

Use Quick IP Call Mode

Dial an IP address under the same LAN/VPN segment by entering the last octet in

the IP address.

In the Advanced Settings page there is an option “Use Quick IP-call mode”.

Default

setting is “No”. When set to “Yes”, and #XXX is dialed, where X is 0-

9 and XXX

<=255, phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc

comes from the local IP address REGARDLESS of subnet mask.

#XX or #X are also valid so leading 0 is not required (but OK). See

Quick IP Call

Mode for details.

Disable Conference

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

Disable DND Button

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

Disable Transfer

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

Auto-Attended Transfer

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

Configuration via

Keypad Menu

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

Enable STAR key

Keypad locking

If enabled, when the phone is in idle screen, press and hold STAR key for 4

seconds and the keypad will be locked. The password to lock/unlock can be

configured.

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.

Table 16: SIP Account Settings

Account Name

The name associated with each account - displayed on LCD.

SIP Server

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

Secondary SIP Server

This field allows administrator to configure a backup SIP Server.

Outbound Proxy

IP address or Domain name of Outbound Proxy, Media Gateway, or Session Border

Controller. Used for firewall or NAT penetration in different network environment. If

the system detects symmetric NAT, STUN will not work. ONLY outbound proxy can

provide solution for symmetric NAT.

SIP User ID

User account information provided by VoIP service provider (ITSP); either an actual

phone number or formatted like one.

Authenticate ID

SIP service subscriber’s Authenticate ID used for authentication. It can be identical

to or different from SIP User ID.

Authenticate Password

SIP service subscriber’s account password for GXP1400/1405 to register to (SIP)

servers of ITSP.

Name

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

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DNS Mode

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.

Primary IP

This option applies only if “Use Configured IP” is selected, the phone will send DNS

query to the Primary IP. Insert IP address here.

Backup IP 1

Insert the first back up IP here.

Backup IP 2

Insert the second back up IP here.

TEL URI

Default is “Disabled”. Users can enable it or select USER=PHONE.

SIP Registration

This parameter controls sending REGISTER messages to the proxy server.

The

default setting is “Yes”.

Unregister on Reboot

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

on reboot.

Register Expiration

This parameter allows user to specify the time frequency (in minutes) that

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).

Reregister Before

Expiration

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.

Local SIP Port

This parameter defines the local SIP port used to listen and transmit. The default

value is 5060.

SIP Registration Failure

Retry Wait Time

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

SIP T1 Timeout

RFC 3261 SIP T1 timer. Default is 0.5 second.

SIP T2 Interval

RFC 3261 SIP T2 timer. Default is 4 seconds.

SIP Transport

Choose SIP Transport between UDP and TCP. Default is UDP.

SIP URI Scheme when

using TLS

Select “sip:” or “sips:”. Default is “sips:”.

Use Actual Ephemeral

Port in Contact with

TCP/TLS

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

Check Domain

Certificates

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

Remove OBP from

Route

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

Validate Incoming

Messages

This configuration selects whether or not the incoming messages should be

validated.

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Support SIP Instance ID

Selects whether or not SIP Instance ID is supported.

NAT Traversal

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.

SUBSCRIBE for MWI

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

will be sent periodically.

SUBSCRIBE for

Registration

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

periodically.

Feature Key

Synchronization

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

Broadsoft. 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 the Broadsoft server side, those features will be synchronized on

the phone side and the Broadsoft server side.

PUBLISH for Presence

Enable Presence feature.

Proxy-Require

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

Voice Mail UserID

When configured, user can access messages by pressing “MSG” button. This ID is

usually the VM portal access number.

Send DTMF

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

supported modes: in audio which means DTMF is combined in audio signal (not

very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.

DTMF Payload Type

Sends DTMF using RFC2833. The default is 101.

Early Dial

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

Dial Plan Prefix

Sets the prefix added to each dialed number.

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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+}

Allows 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

Wait Time

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

Enable Call Features

Default is “Yes”. If set to “No”, Call transfer, Call Forwarding & Do-Not-Disturb are

supported locally provided ITSP support those features. In addition, “ForwardAll”

softkey will be hidden if call feature code is disabled for Account 1.

Call Log

User can choose to disable Call Log and what kind of calls to log.

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Session Expiration

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.

Min-SE

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

Caller Request Timer

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

remote party supports session timer.

Callee Request Timer

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

with session timer request.

Force Timer

If set to “Yes”, the phone will use session timer even if the remote party does not

support this feature. If set to “No

”, the session timer is enabled only when the

remote party supports this feature. To turn off Session Timer, select “No” for Caller

Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher

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

or proxy server as the refresher.

UAS Specify Refresher

As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to

use the phone as the refresher.

Force INVITE

Session Timer can be refreshed using INVITE method or UPDATE method. Select

“Yes” to use INVITE method to refresh the session timer.

Enable 100rel

PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional

responses (1xx series). This is required to support PSTN inter-networking.

Account Ring Tone

There are 4 uniquely defined ring tones:

• One (1) System Ring Tone: when selected, all calls will ring with system

ring tone.

•

Three (3) Customer Ring Tones: when selected, incoming calls from

designated account will play selected ring tone.

Ring Timeout

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

Line-seize Timeout

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

15 seconds.

Send Anonymous

If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will

be set to anonymous, essentially blocking the Caller ID from displaying.

Anonymous Call

Rejection

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

Auto Answer

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

to answer the incoming call. Set to Intercom/Paging mode, it will answer the call

based on the SIP info header from the server.

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Allow Auto Answer by

Call-Info

If the Call-Info header contains answer-after=0, the call be answered automatically

(so called paging mode).

Refer-To Use Target

Contact

Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header

uses the transferred target’s Contact header information.

Transfer on Conference

Hangup

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”.

Preferred Vocoder

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 ch

oosing the

appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by

choosing the appropriate option in “Choice 8”.

SRTP Mode

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

Symmetric RTP

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.

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 an

d 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 fo

r 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 reco

mmend

adjusting these parameters if you are an average user.

Incorrect settings will affect

the voice quality.

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Firmware version: 1.0.1.83 Last Updated: 08/2011

No Key Entry Timeout

Default is 4 seconds.

Use # as Dial Key

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.

G723 Rate

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

G726-32 Packing Mode

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

ilbc Frame Size

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

ilbc Payload Type

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

Conference URI

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

Special Feature

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

Switch vendors.

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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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 corresponding

configuration template of the firmware.

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 pre-

scheduled 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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Firmware version: 1.0.1.83 Last Updated: 08/2011

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 000b8200e395, it should be key in as “0002228200333395”.

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.

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