Grandstream Networks ENTERPRISE GXP1200, GXP2000, GXP2010, GXP2020, ENTERPRISE GXP280/GXP285 User Manual

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Grandstream Networks, Inc.

GXP2020/GXP2010/GXP2000/GXP1200/GXP280/GXP285 Enterprise IP Phones

GXP-1200

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GXP-2020 GXP-2010

GXP-280/GXP-285 GXP-2000

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TABLE OF CONTENTS

GXP USER MANUAL

WELCOME ................................................................................................................................................................. 4

INSTALLATION......................................................................................................................................................... 5

EQUIPMENT PACKAGING ............................................................................................................................................. 5

CONNECTING YOUR PHONE ........................................................................................................................................ 5

GXP-2000 EXTENSION UNIT ....................................................................................................................................... 5

SAFETY COMPLIANCES ................................................................................................................................................ 7

WARRANTY ................................................................................................................................................................. 7

PRODUCT OVERVIEW ............................................................................................................................................ 8

USING THE GXP SIP ENTERPRISE PHONE ..................................................................................................... 13

GETTING FAMILIAR WITH THE LCD .......................................................................................................................... 13

MAKING PHONE CALLS ............................................................................................................................................. 17

ANSWERING PHONE CALLS ....................................................................................................................................... 20

PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 20

CALL FEATURES ........................................................................................................................................................ 23

CUSTOMIZED LCD SCREEN & XML ......................................................................................................................... 23

CONFIGURATION GUIDE ...................................................................................................................................... 24

CONFIGURATION VIA KEYPAD .................................................................................................................................. 24

CONFIGURATION VIA WEB BROWSER ...................................................................................................................... 27

SAVING THE CONFIGURATION CHANGES ................................................................................................................... 41

REBOOTING THE PHONE REMOTELY ......................................................................................................................... 41

SOFTWARE UPGRADE & CUSTOMIZATION .................................................................................................. 42

FIRMWARE UPGRADE THROUGH TFTP/HTTP .......................................................................................................... 42

CONFIGURATION FILE DOWNLOAD ........................................................................................................................... 43

RESTORE FACTORY DEFAULT SETTING ....................................................................................................... 44

TABLE OF FIGURES

GXP USER MANUAL

Figure 1: Connecting the GXP–2000 and the GX P –Extension Figure 2: GXP–2000 Internal Headset Wiring Schema Table 10: GXP Keypad Buttons Figure 3: Keypad GUI Flow

TABLE OF TABLES

GXP USER MANUAL

Table 1: Equipment Packaging Table 2: GXP Connectors

Table 3: GXP Product Models Table 4: GXP Comparison Guide

Table 5: GXP Key Features in a Glance Table 6: GXP Hardware Specifications

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Table 7: GXP Technical Specifications ......................................................................................... 11

Table 8: LCD Buttons Table 9: LCD Icons Table 11: GXP Call Features Table 12: Key Pad Configuration Menu Table 13: Device Configuration - Status Table 14: Device Configuration – Basic Settings Table 15: Advanced Settings Table 16: SIP Account Settings

GUI INTERFACE EXAMPLES

GXP USER MANUAL

(http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip)

1. S

CREENSHOT OF CONFIGURATION LOGIN PAGE

2. S

CREENSHOT OF STATUS PAGE

3. S

CREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE

4. S

CREENSHOT OF ADVANCED USER CONFIGURATION PAGE

5. S

CREENSHOT OF SIP ACCOUNT CONFIGURATION

6. S

CREENSHOT OF SAVED CONFIGURATION CHANGES

7. S

CREENSHOT OF REBOOT PAGE

.................................................................................................................... 13

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

........................................................................................................ 23

........................................................................................ 24

....................................................................................... 28

.......................................................................... 28

........................................................................................................ 31

.................................................................................................... 37

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Welcome

Your Grandstream GXP Series IP phone features a new sophisticated design and is very easy to use. The GXP combines advanced feature functionality with the latest technology to offer excellent audio quality, ease of use, expandability, and broad interoperability with 3 the enterprise customer.

The GXP Series supports a broad range of codecs, security protection, PoE (not supported on GXP-280), dual 10/100mbps Ethernet ports and are very easy to manage. Currently, the GXP Series consists of the following six models: GXP-280, GXP-285, GXP-1200, GX P-2000, GXP-2010 and GXP-2020. Each model delivers superior audio quality using either a handset, hands-free speakerphone or headset (except for GXP2000) and supports multi-party conferencing, multi-languages, dual-color LEDs, presence and BLF (on most models). Large easy-to-read backlit graphical displays with multiple XML keys further enhance the user experience (not supported on GXP-280/285). Some models (GXP-2000, GXP2010 and GXP2020 currently) are expandable with one or two expansion module.

The series is based on SIP standard and are interoperable with most 3rd party SIP platforms and opensource platforms.

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 GXP as it may cause damage to the

products and void the manufacturer warranty.

• This document is contains links to Grandstream GUI Interfaces. Please download these exam ples

http://www.grandstream.com/support/gxp_series/general/documents/gxp_gui.zip for your

from reference.

• This document is subject to change without notic e. The latest electronic version of this user manual

is available for download

http://www.grandstream.com/support/gxp_series/general/documents/gxp_usermanual_english.pdf

• Reproduction or transmittal of the enti re or any part, in any form or by any means, electronic or print,

for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.

party SIP platforms. It is ideal for

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GXP-280/285

GXP-1200

GXP-2000

GXP-2010

GXP-2020

Main Case

Yes

Handset

Yes

Phone Cord

Yes

Power Adaptor

Yes

Ethernet Cable

Yes

High Phone Stand

Yes

Low Phone Stand

Yes

Wall Mount Spacers (2)

Yes

EXT

Connects the GXP Extension unit directly to the GXP using connection cable. Draws power from PoE if provided by network.

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

10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af). Draws power from either spare line or signal line.

Power Jack

5V DC power port; UL Certified

RJ22 and 2.5mm for GXP-280/285/2010/2020

2.5mm for GXP-2000 HW Rev1.0 or later

Handset Jack

RJ11

Installation

EQUIPMENT PACKAGING

Table 1: Equipment Packaging

CONNECTING YOUR PHONE

The connectors of the GXP1200/2010/2020 are locat ed on the bottom of the device while they are located on the back side of the GXP280/285/2000.

Table 2: GXP Connectors

LAN

Headset Jack

RJ22 for GXP-1200

GXP-2000 EXTENSION UNIT

GXP–2000 supports two (2) extension units, providing up to 112 additi onal programmable extensions. Each GXP Extension unit has 56 multi–purpose keys, dual color LEDs (red/green) and support BLF (Busy Lamp Field) and Presence.

GXP–2000 Extension package contains:

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1) One GXP Extension unit

GXP–2000 w/GXP–Extension

GXP Extension

2) One PS2 ca ble

3) One conn ect ion plate

4) One Univ ersal Power Adaptor

FIGURE 1: CONNECTING THE GXP–2000 AND THE GXP–EXTENSION

Connecting the GXP–2000 w/GXP–Extension

Reverse side of connection w/connection plate

Connect the first GXP –EXT to the GXP–2000 using the PS2 cable found in the GXP Extension package. The first GXP–Ext draws power directly from the phone. Connect the second GXP Extension unit using the connection plate and the PS2 cable. The GXP2000 will automatically reboot and power up the GXP Extensions. Grandstream recommends, though not required, to use a separate power supply with the second GXP Ext.

NOTE: should your system lose power, please unplug your devices and power up the GXP–2000 first. Powering up the system:

1. The GXP–2000 will boot up first;

2. The GXP LEDs will be solid red;

3. The status light in the top right corner of the GXP–Ext will blink red;

4. All of the LED indicators on the GXP–Ext will flash three times;

5. The status light at the top right corner of the GXP–E xt will turn to solid green.

NOTE: 1. Extension for GXP2010 and GXP2020 does not support hot-swap. Once connected, user should reboot the phone to ensure the set up will work correctly.

2. GXP2010/GXP2020 can drive 2 extension modules. I ndependent power adapters are not needed for extension modules.

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Figure 2: GXP–2000 Internal Headset Wiring Schema

NOTE: For GXP-2000 HW REV. 0.3 and 0.4, a 3.5mm to 2.5mm plug converter is required to use a 2.5mm

headset. The converter can be purchased at any electronics store.

SAFETY COMPLIANCES

The GXP phone complies with FCC/CE and various safety standards. The GXP power adaptor is compliant with the UL standard. Only use the universal power adaptor provided with the GXP package. The manufacturer’s warranty does not cover dam ages to the phone caused by unsupported power adaptors.

WARRANTY

If you purchased your GXP 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 t he right to remedy warranty policy without prior notification.

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Product Overview

GXP280 is an entry-level SIP phone. It features:

GXP285 is an entry-level SIP phone. It features:

GXP1200 is an entry-level SIP phone. It features: GXP2000 is a mainstream SIP phone. It feature s:

GXP2010 is a key system SIP phone. It features:

Table 3: GXP Product Models

Model Picture Overview

 Single line  Three soft keys

GXP-280

GXP-285

 Single line  Three soft keys  PoE Supported

GXP-1200

GXP-2000

GXP-2010

 Two lines  Three soft keys  PoE Supported

 Four lines  Seven programmable hard keys  PoE Supported

 Four lines  Eighteen programmable hard keys  Three XML programmable soft keys  PoE Supported

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GXP2020 is an executive SIP phone. It features:

, up to 2

Modules, 56

, up to 2

Modules, 56

Silence suppression,

Traditional voice features including caller ID, call waiting, hold,

navigation keys,

ble positioning angles, wall mountable, AES

 Six lines  Seven programmable hard keys  Four XML programmable soft keys

GXP-2020

 PoE Supported

Table 4: GXP Comparison Guide

Features GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

LCD Display Number of Lines Programmable

Hard Keys Soft Keys

Extension Module

128x32 pixel 128x32 pixel 130x64 pixel 240x 120 pixel 320x160 pixel 1 2 4 4 6 No No 7 18 7

3 3 No 3 4 No No

Yes, up to 2 Expansion

Yes Expansion

Yes

Expansion

Modules, 56 nodes each

nodes each

Table 5: GXP Key Features in a Glance

Features Benefits

Open Standards Compatible

SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record and SRV), DHCP (both client and server), PPPoE, TFTP, NTP, Telnet.

Superb Audio Quality

Advanced Digital Signal Processing (DSP), VAD, CNG, AGC.

Network Interfaces

Dual 10/100mbps Ethernet ports, headset jack (RJ22 and/or 2.5mm jack).

Feature Rich

transfer, forward, block, and off-hook dial, click to dial

Advanced Features

Multi-line support with dual-color LED (except on GXP-280/285), multi-party conferencing, line extension interface, large back-lit (except on GXP-280/285) graphic LCD, 5 or 3 dedicated buttons for hold, send, speakerphone, headset, transfer, conference (for up to 5 parties depending on model), mute, message, Do-not-disturb, phone book, intercom/paging.

Advanced Functionality

Custom downloadable ring-tones, SRTP, multi-language support and XML enabled, adjusta encryption.

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Table 6: GXP Hardware Specifications

LAN Interface (Ethernet ports)

Graphic LCD Display

Expansion Module Support

Headset Jack

Call Appearance LED

Power over Ethernet

Universal Switching Power Adaptor Dimension

Weight

Temperature Humidity Compliance

Two (2) 10/100 Mbps Full/H alf Duplex Ethernet Switch with LAN and PC port with auto detection

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

128x32 pixel

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

No No Yes Yes Yes

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

2.5mm and RJ22

128x32 pixel 130x64

pixel

240x120 pixel

RJ22 2.5mm 2.5mm and

RJ22

320x160 pixel

2.5mm and RJ22

Dual color (green/red)

GXP-280/285 GXP-

1200

GXP-2000 GXP-2010 GXP-2020

No 3 11 22 13

Built-in auto-sensing: Cisco and IEEE 802.3af standard: phone draws power from Ethernet (except on GXP-280)

Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA, UL certified

GXP-280/285 168mm(l) x 200mm(w) x 89.5mm(h) GXP-1200 210mm(l) x 195mm(w) x 77mm(h) GXP-2000 220mm(l) x 215mm(w) x 57mm(h) GXP-2010 210mm(l) x 250mm(w) x 77mm (h) GXP-2020 251mm(l) x 202mm(w) x 77mm(h)

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020

0.62kg (1.37lbs)

0.86kg (1.91lbs)

0.82kg (1.81lbs)

1.1kg (2.44lbs)

1.66kg (3.64lbs)

32 –104° F/ 0 – 40°C 10% – 90% (non-condensing) FCC / CE / C-Tick

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Table 7: GXP Technical Specifications

friendly remote software upgrade (via TFTP/HTTP) for deployed devices including

tones, call hold, call transfer

Lines

Multiple direct lines with independent SIP accounts, program mable speed dial keys, XML programmable soft-keys (non programmable on GXP-280/285, GXP1200, GXP2000).

Protocol Support

Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols

Supports multiple SIP accounts

Supports SIP PUBLISH method (RFC 3903), SIP Presence package (RFC 3856, 3863) for

use of 7 MFKs, SIP Dialog package (RFC 4235) Supports SIP MESSAGE method (RFC 3428) Stores up to 100 incoming IM messages

Display Feature

Keys

Back-lit graphic LCD display. (GXP-280/285 display is not back-lit)

GXP-280/285 GXP-1200 GXP-2000 GXP-2010 GXP-2020 HOLD Yes Yes Yes Yes Yes SPEAKERPHONE Yes Yes Yes Yes Yes SEND Yes Yes Yes Yes Yes TRANSFER Yes Yes Yes Yes Yes CONF Yes Yes Yes Yes Yes MUTE Yes Yes Yes Yes Yes DND Yes Yes Yes Yes Yes HEADSET Yes Yes Yes Yes Yes INTERCOM No No No Yes Yes PHONEBOOK No No No Yes Yes MSG Yes Yes Yes Yes Yes MENU Yes Yes Yes Yes Yes NAVIGATION (4) Yes (3) Yes Yes Yes Yes

Device Management

NATbehind firewall/NAT

Auto/manual provisioning system, GUI Interface Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Expansion interface, Address Book

Audio Features

Full-duplex hands-free speakerphone, headset enabled

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),

GSM 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), AGC

(automatic gain control) Acoustic Echo Cancellation (AEC) with Automati c Gain Control (AGC) for speakerphone

mode, Support side tone

Adaptive jitter buffer control and packet delay & loss concealment

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-

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(attended/blind), call forward, call waiting, caller ID, mute, redial, call log, caller ID display or

block, Do-Not-Disturb (DND) and volume control

Provisioning

Features

Security

Multi-party conferencing (up to 5), dial plan prefix, off-hook auto dial, auto answer, early dial

and speed dial (on some models)

Network and

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 Sy m m etric 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

Advanced Server

Message waiting indication, support DNS SRV Look up and SIP Server Fail Over, Support

customizable idle screen via downloading XML by HT T P /TFTP

DIGEST authentication and encryption using MD5 and MD5-sess, SRTP

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Using the GXP SIP Enterprise Pho ne

Key Button

Key Button Definitions

LINE SELECTORS

Selects the phone line printed on it s right-hand side.

LINES

selector on the left-hand side.

DATE AND TIME

logo, please check page 24.

Shows the status of the phone and network. It will indicate whether the ne twork is down, starting or CALLS” are shown here too.

BAR

LINE INDICATOR

The soft-buttons are context sensitive and will change depending on the st atus of the phone.

GETTING FAMILIAR WITH THE LCD

GXP-2xxx has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active screen).

Table 8: LCD Buttons

SIP PHONE

LOGO

NETWORK STATUS

STATUS

SOFTBUTTONS (Excluding GXP-2000)

Displays the available phone lines. Choose a phone line by pressing the co rresponding line

Displays the current date and time. Can be synchronized with Internet time servers.

Displays company logo. This logo ca n be cu stomized. For more information on customizing the

is running (show IP address). Other messages such as “DO NOT DISTURB” or “## MISSED

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

Displays the name of the account that is i n use. Select another account by pressing the LINE SELECTOR BUTTONS

Typical functions assigned to soft-buttons are:

• NEW CALL Press this button to make a new hand-free call .

• FORWARD ALL Unconditionally forwards the main phone line to another phone

• MISSED CALLS This option shows up there were unanswered calls to this phone. The

MissedCalls option shows a list of the missed calls

• CALL RETURN Calls the phone that called/tried to call your phone last.

• REDIAL Redials the last num ber

• END CALL Hangs up the call

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SPECIAL

Icon

LCD Icon Definitions

Connectivity Status / SIP Proxy/Server Icon:

Blank – SIP Proxy/Server not registered

Phone Status Icon:

Speaker Phone Status Icon:

ON when the speakerphone is on

DND Icon:

Activate by pressing MUTE/DEL button once

SOFT BUTTONS (Excluding GXP2000 and Only When Integrated with GXE5024/50

28)

Call Parking: FOR GXP2020/GXP2010 ONLY. Refer to the GXE5024/5028 Online User Manual

for more information.

• CallPark When a GXP2020 dials out, the Call Park soft button will display

on screen. To park the call, press the ‘Call Park’ button.

• PickUp When another GXP2020 goes off-hook the Call Pickup soft button

will display on screen. To pickup the parked call, press the ‘Call Pickup’ button.

Call Queue:

Manual for more information.

• SignIn Press this button to sign in to the call queue. Agent will be prompted in

• SignOut Press this button to sign out of the call queue. Press’ m enu’ button on

PUBLIC MODE (Also mentioned on p.31 of t his manual): This useful mode complements the Call Queue feature by allowing va r i ous user agents to log in/log off, shar i ng the same phone. When enabled, all other accounts on t he phone will not be active. For more information, refer to http://www.grandstream.com/support/gxe_series/gxe502x/documents/gxe502X_call_queue_

with_gxp.pdf

• LogIn Press this button to log i n the user agent into the call queue.

• Tab Press this button to jump to toggle between UserName and Password

• Backspace Press this button to erase the previo usly typed digit, letter, or character.

• LogOut Press this button to log out the user agent out of the call queue.

FOR GXP2020/2010 and 1200 only. Refer to the GXE5024/5028 Online User

the LCD display to select the call queue to join. Press ‘menu’ button on keypad to select ‘ok’. Once the agent completely signs in, the agent will be brought back to the main screen.

keypad to select ‘ok’. This will be displayed once the agent is signed in to the call queue.

entry fields.

Table 9: LCD Icons

Solid – connected to SIP Server/IP address recei ved Blinking – physical connection failed

OFF when the handset is on-hook

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ON when the handset is off-hook

FLASH when phone rings or a call is pe nding OFF when the speakerphone is off

ON when the “do not disturb” is activated

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Calls Forwarded Icon:

Handset, Speakerphone and Ring Volume Icon:

AM/PM indicator

INDICATES calls are forwarded Follow ‘call forwarding’ procedures

Each icon appears next to the volume icon

To adjust volume, use the up/down button

Real–time Clock:

Synchronized to Internet time server Time zone configurable via web browser

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TABLE 10: GXP KEYPAD BUTTONS

Key Button

Key Button Definitions

LINE BUTTONS

Line keys with LED, can be configured to different SIP profiles

TRANSFER

TRANSFER key: Transfer an ACTIVE call to another number

CONF

Press CONF button to connect Calling/Called party into conference

Also used to ‘REJECT’ incoming call.

HOLD

Place ACTIVE call on hold

MSG

Enter to retrieve voice mails or other messages

expires

Enter Keypad Configuration “MENU” mode when phone is in IDLE mode. Use as ENTER key when in Keypad Configuration.

Standard phone keypad; press # ke y to send call; press * key to for IVR functions

DND

DO NOT DISTURB key; Press DND to turn “Do not disturb” function on or off.

GXP2000.

INTERCOM

Turn intercom function on/off

MUTE

SEND

0 - 9, *, #

HEADSET

Mute an active call; or Delete a key entry

Enable/Disable hands-free speaker mode

Press SEND to dial a new number or red i al the last number dialed. Press send button to send a call immediatel y before “no key entry timeout” valu e

Enter to retrieve voice mails or ot her messages

Press HEADSET key to answer/hang u p phone calls while using headset. It also allows user to toggle between headset and speaker. Not available on

Brings phonebook on screen

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MAKING PHONE CALLS

Handset, Speakerphone an d He ad set Mo de

The GXP series phones allow you make phone calls via handset, speakerphone, or headset mode. During the active calls the user can switch between the handset and the speaker by pressing the speaker key. For headsets to operate, the user must plug the headset to an RJ22 or 2.5mm port on the phone, which allows the user to pick-up, speak, or hang-up calls.

Multiple SIP Accounts and Lines

GXP can support up to six independent SIP accounts depending on the product model. Each account is capable of independent SIP server, user and NAT settings. Each of the line buttons is “virtually” mapped to an individual SIP account. The name of each account is conveniently printed next to its corresponding button. In off-hook state, select an idle line and the name of the account (as configured in the web interface) is displayed on the LCD and a dial tone is heard.

For example: Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as “VoIP 1”, “VoIP 2”, respectively and ensure that they are active and registered. When LINE1 is pressed, you will hear a dial tone and see “VoIP 1” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see “VoIP 2” on the LCD display.

To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. User can switch lines before dialing any number by pressing the same LINE button one or more times. If you continue to press a LINE button, the selected account will circulate among the regi st ered accounts.

For example: when LINE1 is pressed, the LCD displays “VoIP 1”; If LINE1 is pressed twice, the LCD displays “VoIP 2” and the subsequent call will be made through SIP account 2.

Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When the “virtually” mapped line is in use, the GXP will flash the next available LINE (from left to right or from top to bottom for Multi Purpose Keys) in red. A line is A CTIVE when it is in use and the corresponding LED is red.

Completing Calls

There are six ways to complete a call:

IAL: To make a phone call.

1. D

• Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates speakerphone) or press the NEW CALL soft-key.

• The line will have a dial tone and the primary line (LI NE1) LED is red. If you wish, select another LINE key (alternati ve SIP account).

• Enter the phone number

• Press the SEND key

or press the “DIAL” soft-key.

EDIAL: To redial the last dialed phone number.

2. R When redialing, the phone will use the same SI P acc ount as was used for the last call. Thus, when the third SIP account was used for the last call/call attempt, the phone will use the third acco unt to redial.

• Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates speakerphone), the corresponding LED will be red.

• Press the SEND button or press the REDIAL soft-key.

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3. CALL RETURN: To call the last phone number that called your phone. When returning a call, the phone will use the same SIP account as the call was made to. Thus, when returning a call made to the third SIP account, the phone will use the third SIP account return the call.

i. Hand-free option

1. Press the CALL RETURN soft-key

ii. Hand-set option

1. Take the Handset off-hook

2. Press the CALL RETURN soft-key

SING THE CALL HISTORY: To call a phone number in the phone’s history

4. U When using the call history, the phone will use the same SIP account as was used for the last call/call attempt. Thus, when returning a call made to the third SIP account, the phone will use the third SIP account return the call.

• Press the MENU button to bring up the Main Menu.

• Select Call History and then “Received Calls”, “Missed Calls” or “Dialed Calls” depending on

your needs

• Select phone number using the arrow keys

• Press OK to select

• Press OK again to dial.

5. U

SING THE PHONEBOOK: Calling a phone in from the phone’s phonebook.

Each entry in the phonebook can be attached to an individual SIP account. The phone will use that SIP account to make the phone call.

• Go to the phonebook by: i. Pressing the phonebook button (bottom, left-hand sid e of phone), or

ii. Pres sing the DOWN arrow key, or

iii. Pres sing the menu button and

Selecting “Phone book” and Press MENU

• Select the phone number by using the arrow keys

• Press OK so select

• Press OK again to dial.

6. P

AGING/INTERCOM:

The paging/intercom function can only be used if the SERVER/PBX supports this feature and both the phones and PBX are correctly configured.

• Take the Handset/SPEAKER/Headset off-hook,

• Select the LINE key associated with account

• Press OK key to display LCD: LINEx: PAGE USING.

• Dial the phone number you want to Page/Intercom

• Press SEND key.

NOTE: Dial-tone and dialed number display occurs after the handset is off-ho ok and the line key is selected. The phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call. Press the “SEND” or “#” button to override the 4 second delay.

Speed Dial

The Multi Purpose Key buttons, located on the right-hand-side of the phone, can be configured for speed dial. Press the speed dial button to automaticall y call the assign ed ext ension.

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Note: The multi-functional buttons will function as LINE keys when all LINEs are busy. The LED will flash in red to indicate an incoming call. Press the button to pick up the call. If any one of the Multi Purpose Keys is associated with a call, the button’s speed di al /BLF function will not work.

Making Calls using IP Addresses

Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy. VoIP calls can be made between two phones if:

• Both phones have public IP addresses, or

• Both phones are on a same LAN/VPN using private or public IP addresses, or

• Both phones can be connected through a router using public or private IP addresses (with necessary

port forwarding or DMZ)

To make a direct IP call, please follow these steps:

1. Press MENU button to bring up MAIN MENU.

2. Select “Direct IP Call” using the arrow-keys.

3. Press OK to select.

4. Input the 12-digit target IP address. (Please see example below).

5. Press OK key to initiate call.

To make a quick IP call, please see next section. For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input

the following: 192*168*1*60#5062 - The “ * ” key represent the dot“.” ; The “#” ke y represent colon “:”. Press OK to dial out.

Quick IP Call Mode

The GXP also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phone’s IP-number. This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended.

Setting up the phone to make Quick IP calls

To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the “Advanced Settings” page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x is 0-9 and

xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. “aaa.bbb.ccc” is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK).

For example:

192.168.0.2 calling 192.168.0.3 -- dial #3 foll ow by S END or #

192.168.0.2 calling 192.168.0.23 -- dial #23 f ol l ow by SEND or #

192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or #

192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3

NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IPIP call will also use STUN. Configure the “Use R andom 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 account LINE flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER or by pressing the corresponding account LINE button.

2. Incoming multiple calls: When another call comes in while having an active call, the phone will produce a Call Waiting tone (stutter tone). Next available lines will flash red (as described in section

4.3.2). Answer the incoming call by pressing its corresponding LINE button. The current active call will be put on hold.

3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via SPEAKER. (PBX (or Server) must also supports this feature)

Do Not Disturb

1. Press the “DND” or “MUTE” button if you do not want to take a call. This will send the caller directly to voicemail.

2. Press the “DND” or “MUTE” button to set phone to ‘do not disturb’ (icon will be on the screen). The phone will not ring and send caller directly to voicemail. (see note above)

PHONE FUNCTIONS DURING A PHONE CALL

Call Waiting/ Call Hold

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

2. Resume: Resume call by pressing the correspondin g bl inking LINE.

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

: Automatically place ACTIVE call on ‘HOLD’ by selecting another available LINE to

Mute/Delete

1. Press the MUTE button to enable/disable m uting the microphone.

2. The “Line Status Indicator” will show “LINEx: SPEAKING” or “LINEx: MUTE” to indicate whether the microphone is muted.

NOTE: Pressing MUTE button for an incoming call will reject the call. MUTE button also functions as delete key when user wishes to delete the last entered digit.

Call Transfer

GXP supports Blind, Attended (or supervised), Semi-Attended and Auto-attended (for GXP1200 only) transfer:

NOTE: To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains. Blind transfer will usually use the primary account SIP profile.

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1. Blind Transfer: Press “TRANSFER (or TRNF for GXP-2000)” button, then dial the number and press the “SEND” button to complete transfer of active call.

2. Attended (or Supervised) Transfer: Press “LINEx” button to make a call and automatically place the ACTIVE LINE on HOLD. Once the second call is established, press “TRANSFER (or TRNF)” key then the LINE button of the waiting line to transfer the call. Hang up the phone call after “Transfer Successful” is displayed in the screen.

3. Semi-Attended Transfer: In the web UI, under “Account Settings”, select the Semi-attended Transfer Mode. There are two modes to select for this feature. RFC5589 and Send Refer with early dialog:

a. With RFC5589 phone will not send out REFER to transferee until 200OK is received from

transfer target. Like an attended transfer, press “LINEx” button to make a call and automatically place the ACTIVE LINE on HOLD. Dial the second call, and as it is ringing press the “TRANSFER (or TRNF)” key and then the blinking LINE button of the call on hold to transfer the call.

b. With "Send REFER with early dialog" mode phone will send out REFER as soon as the

transfer target is selected (in the early dialog stage). Like an attended transfer, press “LINEx” button to make a call and automatically place the ACTIVE LINE on HOLD. Dial the s econd call, and as it is ringing press the “TRANSFER (or TRNF)” key and then the blinking LINE button of the call on hold to transfer the call. Tran sf eror will hang up.

4. Auto-Attended Transfer (only on GXP1200): In the web UI, under “Advanced Settings”, set “Autoattended Transfer” to “Yes”.

 Establish a call between two phones  Press TRANSFER button on GXP1200  Select another line on GXP1200, call another ph one and press SEND button  If the remote phone answers, press TRANSFER but ton on GXP1200 to transfer the call

NOTE: If there is no extra line available, GXP1200 will do the transfer as if Auto-attended transfer is disabled.

5-Way Conferencing

GXP can host conference calls and supports up to 5-way conference call ing.

2. Initiate a Conference Call

 Establish a connection with two or more parties  Press CONF button  Choose the desired line to join the conferen ce by pressing the corresponding LINE button.  Repeat previous two steps for all other parties that would like to join the conference. This

can be done at any time. However, if a new call comes in, the other calls will be placed on hold and the host will have to individually re-join the held lines back into the conference by repeating the previous two steps again.

3. Cancel Conference: Canceling establishing conference call.

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

again or the original LINE button.

 This will resume two-way conversation.

4. End Conference:

 Press HOLD to end the conference call and put all parties on hold;  To speak with an individual party, select the corresponding blinking LINE.

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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.

Voice Messages (Mess age Waiting Indicator)

A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Press the MSG button to retrieve the message. An IVR will prompt the user through the process of message retrieval. Press a specific LINE to retrieve messages for a specif ic li ne account.

NOTE:

• Each line has a separate voicemail account. Each account requires a voicemail portal number to be configured in the “voicemail user id” field.

• To check which line account has a message 1) press the message button (this always checks the primary account), 2) check each line for stutter tone or 3) check missed calls using the menu.

Busy Lamp Field

The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account. When BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when that line is not in use. Call Pick Up is supported when user presses a flashing BLF key.

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CALL FEATURES

The GXP supports traditional and advanced telephony features including caller ID, caller ID w/name, call forward/transfer/park/hold as well as inter com/paging and BLF.

Table 11: GXP Call Features

Key Call Features

*30 Block Caller ID (for all subs equent calls) *31 Send Caller ID (for all subsequent calls) *67 Block Caller ID (per call) *82 Send Caller ID (per call) *50 Disable Call Waiting (for all subsequent calls) *51 Enable Call Waiting (for all subsequent calls) *70 Disable Call Waiting (per Call) *71 Enable Call Waiting (per Call) *72 Unconditional Call Forward

Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wai t for dial tone. LCD will display “Call FWD Activated”.

*73 Cancel Unconditional Call For ward: dial “*73” and get the dial tone, then hang up.

LCD will display “Call FWD Activated”.

*90 Busy Call Forward

Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wai t for a dial tone. Hang up.

*91 Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up. *92 Delayed Call Forward

Dial “*92” for a dial tone. Dial the forwarding number followed by “#”. Wai t for a dial tone. Hang up. LCD will display “Call FWD Activated”.

*93 Cancel Delayed Call Forward

Dial “*93” for a dial tone, then hang up.

CUSTOMIZED LCD SCREEN & XML

Grandstream GXP Series phones support both simple and advanced XML appli cations: 1) XML Cust om Scree n,

2) XML Downloadable Phoneboo k and 3) Adv anced XML S urvey Application. For more i nformation on how to create a downloadable XML pho nebook, creating a cust om idle screen and/ or reprogrammi ng the soft-keys on GXP-1200/GXP-2010/GXP2020, please visit our website

http://www.grandstream.com/support/gxp_series/general/gxp_support.html .

at:

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Preference

Press

button to enter this sub menu including

• “Do NOT Disturb”

Press Menu button to choose the menu item.

Press ‘←’ to return to the main menu.

Configure

Press Menu button to display the configuration selections:

Network.

To setup IP-address, Net mask and Gateway address

Configuration Guide

The GXP can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly, through embedded web-configuration menu.

CONFIGURATION VIA KEYPAD

To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and left/right. Press the OK button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button. The phone automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if left idle for 20 seconds.

Press the MENU button to e nter the key t he Key Pad Menu. The m enu option s available a re listed in table 8.

Table 12: Key Pad Configuration Menu

Call History Status

Phone Book LDAP Directory Instant Messages Direct IP call

Displays histories of incoming, dialed and missed calls. Displays the network status, account stat uses, software version and

MAC-address of the phone. Displays the phonebook

Displays the LDAP directory Goes to voice messages Displays the IP-call options menu

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” m enu.

• Ring Volume Press Menu button to hear the selected ring volume, press ‘←’ or ’ →’ to hear and adjust the ring tone volume.

• LCD Contrast

• LCD Brightness

• 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 Grandstream logo.

• Display Language You can choose English, Chinese or Seconda ry Language

• To enable/disable DHCP.

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• SIP

Press ‘←’ to return the main menu.

Factory Functions

Press Menu to display the factory function items including

Audio Loopback

Toggles the status of the Watchdog Timer.

Press ‘←’ to return to the main menu.

Reboot

Press Menu button to reboot the device

Display “Exit”

Press Menu button to exit the menu

To change SIP-server settings for primary account.

• Upgrade In this menu setting regarding the firmware serv er and Config server can be changed. It also enables the user to make the phone attempt to download new firmware.

• Multi Purpose Key (On GXP2000/2010/2020 only) To configure multi-purpose keys.

• Factory Reset Key in the physical/MAC address on back of the ph one. Press Menu button to reset FACTORY DEFAULT setting. Do not use Factory Reset unless you want to restore f actory settings

• Layer 2 QoS Configure Vlan Tags

• Speak into the handset. If you hear your voice in t he handset, your audio works fine. Press Menu button to exit the mode.

• Diagnostic Mode All LEDs will light up Press any key on the keypad, to display the button name in the LCD. Lift and put back the handset or press Menu b utton to exit the diagnostic mode.

• Enable WDT

Exit

Exit from this menu.

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FIGURE 3: KEYPAD GUI FLOW

Answered Calls

Back

Call History

Back Any of previous menus

Phone Book

New Entry

View Directory

LDAP Directory

Clear All Preference

Do Not Disturb

Back

Network

Config

Enable DND

Default Ring

English

Do Not Disturb

Display Language

IP Setting

Back

Account

Save

Network

Firmware

Upgrade Via

802.1Q/VLAN Tag

Back

Upgrade

Layer 2 QoS

Instant Message

Diagnostic Mode

SIP

Dialed Calls Missed Calls

New Entry Download Phonebook XML Back

Clear All

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

Call History

Status

Phone Book

LDAP Directory

Instant Message

Direct IP Call

Preference

Config

Factory Functions

Reboot

Exit

Download Directory Search Configuration Back

Back

Ring Tone Ring Volume LCD Contrast LCD Brightness Download SCR XML Erase Custom SCR Display Language

SIP Upgrade Multi-Purpose Keys Factory Reset Layer 2 QoS

Factory Function

Audio Loopback Diagnostic Mode Enable WDT Back

Search Configuration

Select Filter Filter Value Back

Disable DND Back

Ring Tone

Ring1 Ring2 Ring 3 Back

LCD Brightness

Active Idle Back

Chinese Secondary Language Language File Postfix Back

IP Net Mask Gateway DNSServer1 DNSServer2

SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio

Server Config Server

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Keypad/LED Diagnostic

Priority value Reset Vlan Config

Page 27

CONFIGURATION VIA WEB BROWSER

The GXP 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 or Mozilla Firefox.

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 t he browser

• Enter the administrator’s password to access t he Web Configuration Menu

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.

If the phone is properly connected to a working Internet connection, the phone will display its IP address. This address has the format: xxx.xxx. xxx.xxx, where xxx stands for a number from 0-255. You will need this number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60, please use “http://192.168.0.60” in the address bar y our browser.

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

NOTE: When changing any settings, always SUBMIT them by pressing the 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.

Definitions

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

Functions available for the end-user are:

• Status: Displays the network status, account statuses, software version and MAC-address of the phone

• Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings can be set here.

Additional functions available to administrat ors are:

• Advanced Settings: To set advanced network settings, codec settings and XML configuration settings.

• Account X: To configure each of the SIP accounts.

• EXT X: To configure setting on extension module

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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 GXP

Product Model

This field contains the product model informati on.

Part Number

This field contains the product part number

Software Version

System Up Time

This field shows system up time since the last reboot.

System Time

This field shows the current time on the phone sy st em.

Registered

Indicates whether accounts are registered to the related SIP server(s). GXP can

PPPoE Link Up

Indicates whether the PPPoE connection is ena bl ed (connected to a modem).

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

(secondary). These fields are set to zero by default.

• Program: This is the main software (firmware) release number, always used to identify the software (firmware) system of the phone.

• Boot: Booting code version number

support four unique SIP profiles.

Table 14: Device Configuration – Basic Settings

The GXP operates in two modes:

1. DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The GXP acquires its IP address from the first DHCP server it discovers on its LAN. The DHCP option is reserved for NAT router mode. To use t he PPPoE feature, set the PPPoE account settings. The GXP establishe s a PPPoE session if any of the PPPoE fields is set.

2. Static IP mode: configure all of the following fields: IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2

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on the PBX and it

Each function is connected to one of the accounts and has a target user ID.

Time Zone

This parameter controls the date/time display according to the specified time zone.

LCD Backlight Always On

1200/GXP-2000 only.

Time Display Format

LCD time display in 12 hour or 24 hour format

Date Display Format

Choose one of the following formats:

This option applies to GXP280/GXP285/GXP1200/GXP2000 only.

Display Clock instead of Date

Choose to display clock or date on LCD. This option appli es to GXP-280/GXP285/GXP-1200/GXP-2000 only.

Multi Purpose Key X

These options are used to assign a function to t he cor responding multi purpose key. Options available are:

1. “Speed Dial”.

2. “BLF” (Busy Lamp Field). This option has to be supported indicates the status of the extension. The t hree possible states are idle (green), busy (red), ringing (blinking red).

3. “Presence Watcher”. This option has to be supported by a presence server and it is tied to the “Do not disturb” status of the phone.

4. “Eventlist BLF”. This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message.

5. “Speed Dial Via Active Account”. This option will act just like speed dial, but based on the current active account. For instance, if the phone is offhook and account 4 is active, it will call the configured speed dial number using account 4.

Turn on LCD backlight at all times. Default is No. T hi s option applies to GXP-

• Year-Month-Day

• Month-Day-Year

• Day-Month-Year

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ends the first Sunday of November at 2AM. The saving is 60 minutes.

Brightness

the brightest. For GXP2010 and GXP2020 only.

LCD Contrast

Set LCD contrast. Range from 0 to 20. Not for GXP280/285

Disable in-call DTMF display

Mute Speaker Ringer

Default is No. When it’s enabled, speaker won’t ring on an incoming call.

Disable Missed Call Backlight

Default is No. By default, LCD backlight will l it whenever there is a missed call. Not for GXP280/285.

Daylight Savings Time

This parameter controls time displayed in daylight savings time. I f set to “Yes”, then the displayed time will be 1 hour ahead of normal time.

The “Optional Rule” is configured to automatically adjust the Daylight Savings Time (DST) based on the rule set in this field. Rule Syntax:

• start-time; end-time; saving

• Both start-time and end-time have the same syntax: month,day,weekday,hour,minute

o month: 1,2,3,..,12 (for Jan, Feb, .., Dec) o day: [+|-]1,2,3,..,31 o weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight

saving rule is not based on week days but based on the day of the month.

o hour: hour (0-23), minute: minute (0-59)

If “weekday” is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the “day” value must not be negative. If “weekday” is not zero and “ day” is positive, then the daylight saving starts on the first “day” the iter ation of the weekday (e.g.: 1st Sunday, 3rd Tuesday etc). If “weekday” is not zero and “day” is negativ e, then the daylight saving starts on the last “day” the iteration of the weekday (e.g.: last Sunday, 3rd last Tuesday etc).

The saving is in the unit of minutes. The saving time may also be preceded b y a n egative (-) sign if subtraction is desired instead of addition.

The default value is set for US, the “Automatic Daylight Saving Time Rule” shall be set to

“3,2,7,2,0;11,1,7,2,0;60”

Examples US/Canada where daylight saving time is applicable:

03,02,7,02,00;11,1,7,02,00;60 This means the daylight saving time st ar ts from the second Sunday of March at 2AM and

LCD Backlight

Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means

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

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NOTE: This is not available for GXP2000

PC Port Mode

For GXP1200/GXP280 only. Default is ‘Switch’. Choose to toggle between Switch/Hub mode.

Admin

the “Advanced

page. Password field is purposely blank for

king update and saved. The maximum password

length is 25 characters.

G723 rate

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

iLBC frame size

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

127.

Silence Suppression

This controls the silence suppression/VAD feature of the audio codec G.723 and

of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to “No”, this feature is disabled.

HEADSET Key Mode

Select either “Default mode” or “Toggle Speaker(default )/Hea dset ” or “Toggle Speaker/Headset(default)”.

In “Default Mode”, only the speakerphone will ring for an incoming call. User ca n use the headset key to pick-up, speak, and hang up calls throu gh headset . The headset icon will appear on the LCD when a call is in progress.

If “Toggle Speaker/Headset(default)”is checked, onl y the headset will ring for an incoming call.

If “Toggle Speaker(default)/Headset” is checked, only the speakerphone will ring for an incoming call but the user can make the phone rings using the headset by pressing the HEADSET key while phone is idle. The headset icon will appear on the idle LCD screen.

Advanced User configuration includes not only the end user configuration, but also advanced configuration such as SIP configuration, Codec select ion, NAT Traversal Setting and other miscellaneous configuration.

Table 15: Advanced Settings

Administrator password. Only the administrator can access

Password

Settings” and “Account Settings” security reasons after clic

iLBC payload type

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

G.729. If set to “Yes”, when silence is detected, a small quantity

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Voice Frames per TX

This field contains the number of voice frames to be transmitted in a single

d the IS limit is based on the maximum size of

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

codec in the above codec Preference List or the actual

used payload type negotiated between the 2 conversation parties at run time.

, if the first codec is configured as G.723 and the “Voice Frames per TX” is

value in the SDP message of an INVITE request will

larly, if

this field is set to 2 and the first codec is G.729 or G.711 or G.726, then the

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

s 20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for

Please be careful when editing these parameters. Adjusting these parameters

a patent dynamic jitter

Grandstream recommends using the default settings provided. Grandstream

mmend adjusting these parameters if you are an average user.

fect the voice quality. Please refer to the Codec FAQ

at http://www.grandstream.com/pdf/FAQ-Codec.pdf for more technical detail.

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 48.

Layer 2 QoS

This contains the value used for layer 2 VLAN tag. Def ault setting is blank.

Data VLAN Tag

Default is 0. Enabling the Data VLAN filtering will help reduce the load on the

primarily for VLAN

filtering where tagged traffic will be forwarded to the DSP .

No Key Entry Timeout

Default is 4 seconds.

Use # as

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

lent to the “(Re)Dial” key. If set to “No”, the “#” key is

included as part of the dial string.

Local RTP port

This parameter defines the starting local RTP-RTCP port pair used to listen and

mit. It is the base RTP port for channel 0. When configured, channel 0 will

use port_value+2 for RTP and port_value+3 for its RTCP. The default value is

5004.

Use Random Port

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

the same NAT. Default is No.

Ethernet packet (be advise Ethernet packet is 1500 byte (or 120kbps)).

associated with the first

E.g.

set to 2, then the “ptime” be 60ms because each G.723 voice frame contains 30ms of audio. Simi

“ptime” value in the SDP message of an INVITE reque st will be 20ms .

G.726, it i G.729/G.728, 64 (x10ms) and 64 (x2.5ms) fram es respectively.

will also change the dynamic jitter buffer. The GXP has buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.

does not reco Incorrect settings will af

phone, but it isn’t necessary in most environments. This is

Dial Key

key is essentially equiva

trans use this port _value for RTP and the port_value+1 for its RTCP; channel 1 wil

SIP and RTP ports. This is usually necessary when multiple GXPs are behi

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sends a blank UDP packet to the

SIP server in order to keep the “hole” on the NA T open. Default is 20 seconds.

Use NAT IP

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

STUN Server

displayed in the STATUS page of the Web UI.

Firmware Upgrade and

Default method is HTTP. Firmware upgrade m ay take up to 10 minutes process.

Via TFTP Server

This is the IP address of the configured TFTP server. If selected and it is non-

le or new code image from the specified TFTP server at boot time. It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code

retrieved, the new downloaded image will be verified and then saved

Grandstream strongly recommends that the user upgrade firmware locally

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

pecially the power supply) as this will damage the

device.

The HTTP server URL used for firmware upgrade and configuration via HTTP.

” is the specific TCP port that the HTTP server is using; omit if using

will only perform HTTP download once

at boot up.

Config Server Path

IP address or domain name of firmware server.

XML Config File Password

Only)

Firmware File

Default is blank. If configured, GXP will request the firmware file with the

Config File

Default is blank. End user should keep it blank.

Allow DHCP Option 43 and server

Default is Yes. This allows the device to get provisioned automatically.

Authenticate Conf File

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

Keep-alive interval

Provisioning

Via HTTP Server

This parameter specifies how often the GXP

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

depending on network environment. Do not interrupt the firmware upgrading

zero or not blank, the GXP will attempt to retrieve a new configuration fi

image is into the Flash memory.

Note:

TFTP upgrade process (es

For example: http://provisioning.mycompany.com:6688/Grandstream/1.2.5.3 Here “:6688 default port 80.

Note: If Auto Upgrade is set to No, GXP

(For GXP280/GXP/285/GXP1200

Prefix/Postfix

Option 66 to override

configuration updates via XML configuration files. Users can set the XML config file password in the web UI of the phone.

prefix/postfix. This setting is useful for ITS Ps. End user should keep it blank.

The XML provisioning system allows Grandstream phones to perform

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. Choose “Yes” to enable automatic HTTP upgrade and

every” field, enter the number of minutes to check the

HTTP server for firmware upgrade or configuration changes. When set to “No”,

form HTTP upgrade and configuration check once at boot

up.

LDAP Directory

IP address or domain name of LDAP script server

Phonebook XML

L server path and

When the user downloads the XML phone the manually

entered or edited entries will not be delet ed unl ess this option is selected to Yes.

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 serve r path”.

XML Application

Enter server path for XML application. This option applies to GXP-2020 and GXP-2010 only.

DTMF Payload Type

This parameter sets the payload type for DTMF using RFC2833. Default is 101.

Onhook Threshhold

es the time handset has to be down to be recognized it’s onhook.

Default is 800ms. For GXP280/285 only.

Syslog Server

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

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

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

GS_LOG: [device MAC address][error

: May 19 02:40:38 192.168.1.14 GS_LOG:

[00:0b:82:00:a1:be][000]. Ethernet lin k i s up.

Automatic Upgrade

Offhook Auto Dial

This function is used by ITSP. End user should NOT touch these parameters. Default is No

provisioning. In “Check for upgrade

the phone will only per

Enable the XML phonebook via TFTP or HTTP. Define XM download interval.

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

It determin

following events:

• product model/version on boot up (INFO lev el )

• 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 Syslo contains the following components: code] error message For example

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NTP server

This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve. It is used to display the current dat e/time.

Public Mode

Default is ‘No’ (standard mode). Supported only on GXP1200, GXP2010, and

signed specifically to support the Call Queue feature implemented on Grandstream’s GXE5024 or GXE5028. If set to ‘Yes’, this feature will allow other call queue agents to login/logout using their own

, all other

accounts will not be active.

Distinctive Ring Tone

Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a particular Caller ID. The GXP will ONLY use selected ring tones for particular

, the GXP 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. standard.

Call Progress Tones

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

are supported.

To page a group : [paging group feature code] +[ * ]+[group extension]

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

GXP2020. This feature is de

Caller IDs. For all other calls

Adjust system ring tone frequencies and cad ences based on local telecom

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 i n 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 b e zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pat tern. Up to three cadences

Intercom User ID:

This field is used to configure the Intercom key in the phone. For GXP2010 and GXP2020 only If the phone is working with a GS GXE502X IP-PBX it can be configured in the following manner:

• To page an extension : [intercom feature code] +[ *]+[extension number]

•

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Use Quick IP Call Mode

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

9 and

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

Quick IP

Call Mode

Disable Conference

Default is No. If set to Yes, conference wi ll be di sabled.

Lock keypad update

If set to “Yes”, the configuration changes vi a key pad are disabled.

Enable MPK Sending DTMF

Default is No. If set to “Yes”, Muti-Purpose keys can be sent as DTMF. For GXP2020/2010/2000.

For GXP2000, MUTE/DEL button functions as DND button when pressed while phone is idle.

Disable Transfer

Default is No. If set to Yes, transfer will be di sabled.

Disable Multicast Filter

Default is No. If set to Yes, the phone will not filter out (discard) multicast packets.

Semi-Attended Transfer

Default is RFC5589 mode, which allows the REFER m essag e to be sent af ter

TRANSFER is not finished until it receives

(e.g. 4xx, 2xx) from the SIP server.

Disable Headset Button

ill

have no effect.

Enable Constraint Mode

Default is No. If set to Yes, the phone will limit the end-user’s access to the configuration of the phone.

Display CID instead of Name

If set to “Yes”, CID will be displayed in the screen instead of Name. Default is ”No”. For GXP280/285 only.

Headset Port Type

Select either 2.5mm or RJ22 headset ports to be adj ust ed.

Headset TX gain (dB)

Increases the selected headset’s (2.5mm or RJ22) TX gain by + or – 6dB. Default is 0dB

Headset RX gain (dB)

Increases the selected headset’s (2.5mm or RJ22) RX gain by + or – 6dB. Default is 0dB

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 0aaa.bbb.ccc comes from the local IP address RE GARDLESS of subnet mask. #XX or #X are also valid so leading 0 is not required (but OK). See

for details.

Disable DND

Mode

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

INVITE is answered so that the responses

Default is No. If set to Yes, Headset button will be disabled and pressing it w

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Language file postfix allows the language file to have different postfixes so the

ne can request a particular file. It will append an underscore "_" plus the

The default language file name is "gxp.lpf". If the field “Language File postfix

lpf" instead of

Supported Secondary language: Czech, Dutch, Estonian, French, German,

environment.

using your local TFTP or HTTP

dvanced settings of the Web GUI, set “Display Language” to

6. Update and reboot the phone

Account Active

This field indicates whether the account is active. The default value for the is No.

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.

Outbound Proxy

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

outbound proxy can provide solution for symmet ric NAT.

actual phone number or formatted like one.

Authenticate ID

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

Display Language

Allows user to choose preferred display langu age i n web UI and key pad UI.

pho string in the language file postfix.

“has "NL" string in it, then the phone will request "gxp_NL. "gxp.lpf."

User can only load one secondary language. Italian, Polish, Portuguese, Slovak, S l ovenian and Spanish. How to set up Secondary Language:

Note: This is similar to updating firmware in your local network Please refer to http://www.grandstream.com/faqsfirmware.html#4 for details.

1. Download the language package from http://www.grandstream.com/firmware.html

2. Unzip the language package

3. Open the desired language zip file

4. Copy gxp.lpf to the firmware server directory server.

5. Access the a “Secondary Language”

GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires it s own configuration page. Their configurat i ons are identical.

Table 16: SIP Account Settings

primary account (Account 1) is Yes. The default value for the other two accounts

Border Controller. Used for firewall or NAT penet rat i on in different network environment. If the system detects symmetric NAT, STUN will not work. ONLY

SIP User ID

User account information provided by VoIP servi ce provider (ITSP); either an

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ITSP.

Name

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

Use DNS SRV:

Default is No. If set to “Yes”, the client will use DNS SRV to look up server.

User ID is Phone

If the phone has an assigned PSTN telephone numb er, this field should be set to attached to the “From” header in SIP request

SIP Registration

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

Un-register on Reboot

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

SIP Instance ID

Default is set “No.” If set to Yes it will be enabled and will add reg-ID and Instance

phones to register.

This parameter allows user to specify the time frequency (in minutes) that GXP minutes. The maximum interval is 65,535 minutes (about 45 days).

Local SIP Port

This parameter defines the local SIP port used to l i st en and transmit. The default Account 4 respectively.

SIP Registration Failure Retry Wait Time

Retry registration if the process failed. Default is 20 seco nds.

SIP T1 Timeout

RFC 3261 SIP T1 timer. Default is 1 second.

SIP T2 Interval

RFC 3261 SIP T2 timer. Default is 0.5 seconds.

SIP Transport

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

Use RFC3581 Symmetric Routing

Default No. When selected the phone will follow t he routing procedures specified in RFC3581.

NAT Traversal (STUN)

This parameter activates the NAT traversal mec hanism. If activated (by choosing

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.

Authenticate Password

Number

SIP service subscriber’s account password for GXP to register to (SIP) servers of

“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be

ID on contact header in the REGISTER messages. This feature is mainly provided for servers that don't support SIP Instance ID feature, but will still allow

refreshes its registration with the specified registrar. The default interval is 60

value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and

“Yes”) 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 configurat ion 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 the NAT Traversal field is set to “Yes” with no specified STUN server, the GXP will periodically (every 20 seconds or so) send a blank UDP packet (with no payload

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PUBLISH for Presence

Enable Presence feature.

Proxy-Require

SIP Extension to notify SIP server that t he uni t is behind the NAT/Firewall.

Voice Mail UserID

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

Send DTMF

This parameter specifies the mechanism to transmi t DTMF digit. There are 3 very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.

Early Dial

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

Dial Plan Prefix

Sets the prefix added to each dialed number.

BLF Call-pickup Prefix

Default is ‘**”. This prefix is prepended when answering call with BLF key.

Delayed Call Forward Wait Time

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

Enable Call Features

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

Session Expiration

The SIP Session Timer extension enables SIP ses sions to be periodically

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 wh en 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 doe s not

Caller Request Timer, Callee Request Timer, and F orce 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.

SUBSCRIBE for Registration Event

Default is No. This is mainly used for IMS purposes. When enabled, the terminals should store the Service-Route header values after successfully registered, and thereafter add a route header with the values stored in the Service-Route when initiating a request excluding REGISTER.

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

Default is 20 seconds.

supported locally provided ITSP support t hose f eatures. In addition, “ForwardAll”

“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVI T E m essage, 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.

support this feature. If set to “No”, the session t i m er is enabled only when the remote party supports this feature. To tu rn off Session Timer, select “No” for

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use the phone as the refresher.

Force INVITE

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

Enable 100rel

PRACK (Provisional Acknowledgment) method enables reliability to SIP networking..

Account Ring Tone

There are 4 uniquely defined ring tones:

designated account will play selected ring tone.

Ring Timeout

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

Whether to use “<sip:anonymous@anonym ous. i nvalid>” in the From Header or PAsserted-Identity header.

Anonymous Call Rejection

Default is NO. If set to YES, anonymous call will be rejected

Auto Answer

Default is No. If set to “Yes”, GXP will automatically switch on speaker to answer the SIP info header from the server.

Allow Auto Answer by Call-Info

If the Call-Info header contains answer-after=0, the call be answ ered automatically (so called paging mode).

Turn off speaker on remote disconnect

When BYE is received, the phone will turn off its spe aker automatically.

Check SIP User ID for incoming INVITE

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

Refer-To Use Target Contact

Attribute in SDP

Preferred Vocoder

GXP supports up to 7 different Vocoder types incl udi ng G.711(a/µ) (also known

choosing the appropriate option in “Choice 8”.

SRTP Mode

Enable SRTP mode based on selection. Default is No.

UAS Specify Refresher

As a Callee, select UAC to use caller or proxy se rv er as the refresher, or UAS to

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

• 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

Disable Multiple Media

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

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

Default is No.

as PCMU/PCMA), GSM, G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band). Configure Vocoders in a preference list that i s included with the same preference

order in SDP message. Enter the first Vocod er in this list by choosing the appropriate option in “Choice 1”. Similarly , enter the last Vocoder in this list by

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BLF", choose account number, monitored number, etc.

Special Feature

Default is Standard. Choose the selection t o m eet special requirements from Soft

eventlist BLF URI

If a server supports this feature, user need s to configure an "eventlist BLF" URI on the service side (i.e.: BLF1006@myserver.com) On the GXP, under Account page, fill in the " "eventlist BLF" field with the URI without the domain. (i.e.: BLF1006). Under Ba sic Settings, please select "eventlist

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.

Grandstream recommends reboot or power 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/1.2.5.3

• 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 Interfac e

To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXP 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. 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. Grandstream recommends 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 ser ver, Grandstream provides 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:

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:

customerFree.cfm.

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http://www.grandstream.com/firmware.html .

http://support.solarwinds.net/updates/New-

Page 43

Instructions for local TFTP Upgrade:

1. Unzip the file and put all of them under the root dire ct ory of the TFTP server.

2. The PC running the TFTP server and the GXP should be in the sam e LA N 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 I P 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 Microso ft IIS

web server.

NOTE:

• When GXP phone boots up, it will send TFTP or HTTP request to download configuration file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP 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 GXP can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. “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.

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 f i rmware.

Once the GXP boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The configuration file name should be in lower cases.

NOTE : Since firmware 1.2.4.3, GXP280/GXP285/GXP1200 can be provisioned using XM L conf i guration file.

Please refer to our XML provisioning guide using this link

http://www.grandstream.com/support/gxp_series/general/documents/GS-XML_Provisioning_Guide.pdf

Managing Firmware an d 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 Provisioni ng 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. Grandstream is 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” (P l ease 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 shoul d be key in as “0002228200333 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 cont inue 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 m enu interface.

395”.

Grandstream Networks, Inc. GXP User Manual Page 44 of 44

Firmware 1.2.5.3 Last Updated: 03/2011