Grandstream Networks ADAPTER GXP2020 User Manual.pdf

Grandstream Networks, Inc.

GXP2020

6-line SIP Enterprise Phone

GXP2020 User Manual	www.grandstream.com
Firmware Version 1.1.4.3	[e-mail]

TABLE OF CONTENTS
GXP2020 USER MANUAL
WELCOME....................................................................................................................................................	4
INSTALLATION............................................................................................................................................	5
EQUIPMENT PACKAGING ...............................................................................................................................	5
CONNECTING YOUR PHONE ..........................................................................................................................	5
WALL-MOUNT...............................................................................................................................................	6
SAFETY COMPLIANCES .................................................................................................................................	7
WARRANTY ..................................................................................................................................................	7
PRODUCT OVERVIEW ................................................................................................................................	8
USING THE GXP2020 SIP ENTERPRISE PHONE ...................................................................................	10
GETTING FAMILIAR WITH THE LCD...............................................................................................................	10
MAKING PHONE CALLS ...............................................................................................................................	13
Handset, Speakerphone and Headset Mode........................................................................................	13
Multiple SIP Accounts and Lines...........................................................................................................	13
Completing Calls ...................................................................................................................................	13
Speed Dial.............................................................................................................................................	14
Making Calls using IP Addresses..........................................................................................................	15
Quick IP Call Mode................................................................................................................................	15
ANSWERING PHONE CALLS .........................................................................................................................	16
Receiving Calls......................................................................................................................................	16
Do Not Disturb.......................................................................................................................................	16
PHONE FUNCTIONS DURING A PHONE CALL .................................................................................................	16
Call Waiting/ Call Hold...........................................................................................................................	16
Mute/delete............................................................................................................................................	16
Call Transfer..........................................................................................................................................	16
5-Way Conferencing..............................................................................................................................	17
Check Voice Messages (Message Waiting Indicator)...........................................................................	17
Asterisk Busy Lamp Field......................................................................................................................	17
CALL FEATURES .........................................................................................................................................	18
CUSTOMIZED LCD SCREEN & XML .............................................................................................................	18
XML Phonebook....................................................................................................................................	18
Customizable Idle Screen and Soft-buttons..........................................................................................	18
CONFIGURATION GUIDE...........................................................................................................................	19
CONFIGURATION WITH KEY PAD ..................................................................................................................	19
CONFIGURATION WITH WEB BROWSER ........................................................................................................	22
Access the Web Configuration Menu....................................................................................................	22
Definitions..............................................................................................................................................	22
Individual Account Settings ...................................................................................................................	27
Saving the Configuration Changes .......................................................................................................	29
Rebooting the Phone Remotely ............................................................................................................	29
SOFTWARE UPGRADE & CUSTOMIZATION..........................................................................................	30
FIRMWARE UPGRADE THROUGH TFTP/HTTP..............................................................................................	30
Key Pad Menu.......................................................................................................................................	30
Web Configuration Interface .................................................................................................................	30
No Local TFTP Server...........................................................................................................................	30
CONFIGURATION FILE DOWNLOAD ...............................................................................................................	31
Managing Firmware and Configuration File Download .........................................................................	31
RESTORE FACTORY DEFAULT SETTING..............................................................................................	32

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TABLE OF FIGURES
GXP2020 USER MANUAL
FIGURE 1: CONNECTORS ON THE BOTTOM OF THE GXP2020............................................................................	5
FIGURE 2: ATTACHING THE WALL-MOUNT SPACERS.........................................................................................	6
FIGURE 3: LOCATION OF THE FIXED HANGERS ON GXP2020.............................................................................	6
FIGURE 4: WALL MOUNTING THE GXP2020.....................................................................................................	7
FIGURE 5: LCD IN IDLE (ON-HOOK) AND ACTIVE (OFF-HOOK) MODE. ............................................................	10
FIGURE 6: GETTING FAMILIAR WITH KEY PAD .................................................................................................	12
FIGURE 7: KEY PAD GUI CALL FLOW.............................................................................................................	21
TABLE OF TABLES
GXP2020 USER MANUAL
TABLE 1: DEFINITIONS OF THE GXP2020 CONNECTORS...................................................................................	5
TABLE 2: GXP2020 KEY FEATURES IN A GLANCE ............................................................................................	8
TABLE 3: GXP2020 HARDWARE SPECIFICATIONS ............................................................................................	8
TABLE 4: GXP2020 TECHNICAL SPECIFICATIONS.............................................................................................	9
TABLE 5: LCD BUTTON DEFINITIONS .............................................................................................................	11
TABLE 6: LCD ICON DEFINITIONS ..................................................................................................................	11
TABLE 7: GXP2020 BUTTON DEFINITIONS.....................................................................................................	12
TABLE 8: GXP2020 CALL FEATURES ............................................................................................................	18
TABLE 9: KEY PAD CONFIGURATION MENU ....................................................................................................	19
TABLE 10: DEVICE CONFIGURATION - STATUS DEFINITIONS ............................................................................	23
TABLE 11: DEVICE CONFIGURATION – BASIC SETTINGS DEFINITIONS ..............................................................	23
TABLE 12: ADVANCED USER CONFIGURATION PAGE DEFINITIONS...................................................................	24
TABLE 13: SIP ACCOUNT CONFIGURATION PAGE DEFINITIONS .......................................................................	27

GUI INTERFACE EXAMPLES

GXP2020 USER MANUAL (http://www.grandstream.com/user_manuals/GUI/GUI_GXP2020.rar)

1.SCREENSHOT OF CONFIGURATION LOGIN PAGE

2.SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE

3.SCREENSHOT OF STATUS CONFIGURATION PAGE

4.STATUS CONFIGURATION PAGE DEFINITIONS

5.SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE

6.SCREENSHOT OF SIP ACCOUNT CONFIGURATION

7.SCREENSHOT OF SAVED CONFIGURATION CHANGES

8.SCREENSHOT OF REBOOT PAGE

Grandstream Networks, Inc.	GXP–2020 User Manual	Page 3 of 32
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Welcome

Your Grandstream GXP2020 features a new sophisticated design and is very easy to use. The GXP2020 combines advanced feature functionality with the latest technology to offer excellent audio quality, ease of use, expandability, and broad interoperability with 3rd party SIP platforms. It is ideal for the enterprise customer.

This 6 line enterprise SIP phone offers more advanced functionality than the GXP2000. It features a larger graphical LCD display, four (4) XML programmable soft-keys, seven (7) programmable hard keys, 5-way conferencing and three (3) adjustable positioning angles. The GXP2020 also offers dual 10/100Mbps auto-sensing Ethernet ports with integrated P-o-E, secure central configuration with AES encryption, and support for a broad range of voice codecs.

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 GXP2020 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 examples http://www.grandstream.com/user_manuals/GUI/GUI_GXP2020.rar for your reference.

•This document is subject to change without notice. The latest electronic version of this user manual is available for download @: http://www.grandstream.com/user_manuals/GXP2020_User_Manual.pdf

•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 of Grandstream Networks, Inc. is not permitted.

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Installation

EQUIPMENT PACKAGING

The GXP2020 phone package contains:

1)One GXP2020 Main Case

2)One Handset

3)One Phone Cord

4)One Universal Power Adaptor

5)One Ethernet Cable

6)One High Phone Stand

7)One Low Phone Stand

8)Two Wall Mount Spacers

CONNECTING YOUR PHONE

The connectors of the GXP2020 are located on the bottom of the device.

Figure 1: Connectors on the bottom of the GXP2020

			GXP Ext			Headset and
			Connection	WAN/LAN RJ-45	Power	handsetJet
				10/100Mbps ports	Jack
Table 1: Definitions of the GXP2020 Connectors
	GXP		Connects the GXP Ext directly to the GXP using connection cable. Draws power
	Ext Connection		from PoE if provided by network.
	WAN/LAN		10/100Mbps RJ-45 ports for WAN (PC) and LAN connections (switched or routed)				.
			Support PoE (802.3af). Draws power from either spare line or signal line
	Power Jack		5V DC power port; UL Certified
	Headset Jack		Two headset jacks: RJ11/2.5mm
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WALL-MOUNT

GXP2020 can be wall mounted. For proper wall mounting please place the two wall-mount spacers into the proper holes on the bottom of the GXP2020 as show below:

Figure 2: Attaching the Wall-Mount Spacers

To position the phone on the wall, place two fixed hangers on the wall, hang the back of the phone on the fixed hangers.

Figure 3: Location of the fixed hangers on GXP2020

Fixed hangers

To use the handset, pull out the tab (extension downward) from the handset cradle, rotate the tab and plug it back into the slot with the extension up to hold the handset.

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Figure 4: Wall Mounting the GXP2020

`

Tab with

Handset extension

Rest

Tab with extension up

Tab

SAFETY COMPLIANCES

The GXP2020 phone complies with FCC/CE and various safety standards. The GXP2020 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 damages 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 the right to remedy warranty policy without prior notification.

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Product Overview

GXP2020 features 6 lines, each with an individual SIP account and offers a large graphical LCD display, four (4) XML programmable soft keys, and seven

(7) programmable hard keys. It offers intuitive user interfaces, 5-way conferencing, 2 headset jacks, and three (3) adjustable positioning angles. Like the GXP2000, the GXP2020 also offers dual 10/100Mbps auto-sensing Ethernet ports with integrated P-o-E, secure central configuration with AES Encryption, superior audio quality (using G.722 wide-band codec) and excellent manageability. The GXP series is fully compatible and interoperable with most SIP end-user devices, IPPBX and Softswitches.

Table 2: GXP2020 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,
	and TLS.
Superb Audio Quality	Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG,
	AGC.
Network Interfaces	Dual 10/100mbps Ethernet ports; 2 USB (2.0) host ports, headset jack (RJ11
	and 2.5mm jack).
Feature Rich	Traditional voice features including caller ID, call waiting, hold, transfer,
	forward, block, autodial, off-hook dial, and click to dial.
Advanced Features	Multi-line support with dual-color LED, multi-party conferencing, line extension
	interface, large back-lit graphic LCD, 5 navigation keys, eleven (11) dedicated
	buttons for hold, send, speakerphone, headset, transfer, conference (for up to
	4 parties), mute, message, Do-not-disturb, phone book, intercom/paging.
Advanced Functionality	Custom down-loadable ring-tones, SRTP, SIP over TLS (pending), multi-
	language support and XML enabled, 3 adjustable positioning angles, wall
	mountable, AES encryption.

Table 3: GXP2020 Hardware Specifications

LAN Interface (Ethernet ports)	Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port
	with auto detection
Graphic LCD Display	320 X 160 pixels
Expansion Module Support	Yes (pending)
Headset Jack	2.5mm headset port and RJ11 jack
LED	13 LED with different light pattern in dual color (g/r)
Power over Ethernet	Built-in auto-sensing: Cisco and IEEE	802.3af standard: phone
	draws power from both spare lines or signal lines from Ethernet
Universal Switching	Input: 100-240VAC 50-60 Hz
Power Adaptor	Output: +5VDC, 1200mA, UL certified
Dimension	251mm(l) x 202mm(w) x 77mm(h)
Weight	1.66kg (3.64lbs)
Temperature	32 –104° F/ 0 – 40°C
Humidity	10% – 90% (non-condensing)
Compliance	FCC / CE / C-Tick
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Table 4: GXP2020 Technical Specifications

Lines	6 direct lines with independent SIP accounts, 7 programmable speed dial keys
	4 XML programmable soft-keys
Protocol	Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP,
Support	ARP/RARP, ICMP, DNS, DHCP, NTP/SNTP, TFTP, SIMPLE/PRESENCE protocols
	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 7 MFKs, SIP Dialog package (RFC 4235)
	Support for SIP MESSAGE method (RFC 3428)
	Stores up to 100 incoming IM messages (drops IM message 101 plus)
Display	Back-lit 320 X 160 pixels graphic LCD
Feature Keys	11 dedicated keys: hold, send, speakerphone, headset, transfer, conference (up to
	5 parties), mute, message, Do-Not-Disturb (DND), phone book, intercom/paging, 5
	display/menu navigation keys, dual color (g/r) LEDs
Device	NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed devices
Management	including behind 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), 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 & loss concealment
Telephony	Intuitive graphic user interface (GUI), downloadable phone book (XML, LDAP),
Features	support for anonymous call using privacy header, MLS (multi 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
	Multi-party conferencing (up to 5), dial plans, off-hook auto dial, auto answer, early
	dial and speed dial
Network and	Via keypad/LCD, Web browser, or secure (AES encrypted) central configuration file,
Provisioning	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	Support firmware upgrade via TFTP or HTTP,
Upgrades	Support for Authenticating configuration file before accepting changes
	User specific URL for configuration file and firmware files
Advanced	Message waiting indication, support DNS SRV Look up and SIP Server Fail Over,
Server Features	Support customizable idle screen via downloading XML by HTTP/TFTP
Security	DIGEST authentication and encryption using MD5 and MD5-sess, SRTP, SIP over
	TLS (pending)

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Using the GXP2020 SIP Enterprise Phone

GETTING FAMILIAR WITH THE LCD

The GXP2020 has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active screen). Figure 5 illustrates the IDLE and ACTIVE screens.

Figure 5: LCD in IDLE (on-hook) and ACTIVE (off-hook) mode.

	SIP phone lines		Line Status Indicator and Account Name
	6 line selectors	Date and Time
		Customizable logo
			Status Bar

Network Status (Phone IP Address)

Context Sensitive Soft Buttons

.

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Table 5: LCD Button Definitions
	Key Button			Key Button Definitions
	LINE SELECTORS (6)			Selects the phone line printed on its right-hand side.
	SIP PHONE LINES			Displays the available phone lines. Choose a phone line by pressing the
				corresponding line selector on the left-hand side.
	DATE AND TIME			Displays the current date and time. Can be synchronized with Internet time servers.
	LOGO			Displays company logo. This logo can be customized. For more information on
				customizing the logo, please see the web-configuration menu (page 31).
	NETWORK STATUS			Shows the status of the phone and network. It will indicate whether the network is
				down, starting or is running (show IP-number). Other messages such as “DO NOT
				DISTURB” or “## MISSED CALLS” are shown here too.
	STATUS BAR			Shows the status of the phone, using icons as shown in the next table.
	LINE STATUS			Displays the name of the account that is in use. Select another account by pressing
	INDICATOR			the LINE SELECTOR BUTTONS
				The soft-buttons are context sensitive and will change depending on the status of
				the phone. 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
	SOFT-BUTTONS			•	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 number
				•	END CALL	Hangs up phone
Table 6: LCD Icon Definitions
	Icon	LCD Icon Definitions

Phone Status Icon:

OFF when the handset is on-hook

ON when the handset is off-hook

Speaker Phone Status Icon:

FLASH when phone rings or a call is pending

OFF when the speakerphone is off

ON when the speakerphone is on

Handset, Speakerphone and Ring Volume Icon:

Each icon appears next to the volume icon

To adjust volume, use the up/down button

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Figure 6: Getting Familiar with Key Pad

Message Waiting Indicator

Voice Message button

Graphical LCD screen

Menu Keys

Line 1-6 keys

Softkeys

Dedicated buttons

Speed Dial Buttons (Programmable Buttons)

Phonebook button

Standard Keypad

Send Button

Hand Free Button

Hold Button

Table 7: GXP2020 Button Definitions
	Key Button		Key Button Definitions
	LINE BUTTONS (6)		6 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
	MUTE		Mute an active call; or Delete a key entry
			Also used to ‘REJECT’ incoming call.
	DND		DO NOT DISTURB key; Press DND to turn “Do not disturb” function on or off.
	HEADSET		Toggle between headset and speakerphone mode when in hands free mode
	INTERCOM		Turn intercom function on/off
			Brings phonebook on screen
	HOLD		Place ACTIVE call on hold
	MSG		Enter to retrieve (video) voice mails or other messages
			Enable/Disable hands-free speaker mode
	SEND		Press SEND to dial a new number or redial the last number dialed. Press
			send button to send a call immediately before “no key entry timeout” value
			expires
			Enter to retrieve voice mails or other messages
	MENU		Enter Keypad Configuration “MENU” mode when phone is in IDLE mode.
			Use as ENTER key when in Keypad Configuration.
	0 - 9, *, #		Standard phone keypad; press # key to send call; press * key to for IVR
			functions
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MAKING PHONE CALLS

HANDSET, SPEAKERPHONE AND HEADSET MODE

Handset can be toggled between Speaker and Headset. To switch between Handset and Speaker/Headset, press the Hook Flash in the handset cradle or press the SPEAKER button.

MULTIPLE SIP ACCOUNTS AND LINES

The GXP2020 can support up to 6 independent SIP accounts. Each account is capable of independent SIP server, user and NAT settings. Each of the 6 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 the first three of the six SIP accounts as “BenjiSIP”, “SIPnGO”, “IbenSIP”, respectively and ensure each is active and registered. When LINE1 is pressed, you will hear a dial tone and see “BenjiSIP” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see “SIPnGo” on the LCD display; when LINE3 is pressed, you will hear a dial tone and see “IbenSIP” on the LCD display.

To make a call, select the line you wish to use. The LED will light up green. Switch lines before dialing by pressing the same LINE button one or more times. If you continue to press one LINE, the selected account will circulate among the registered accounts.

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

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 GXP2020 will flash the next available LINE (from left to right) in red. A line is ACTIVE when it is in use and the corresponding LED is solid red.

COMPLETING CALLS

There are six ways to complete a call:

1.DIAL: To make a phone call.

•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 (LINE1) LED is red. If you wish, select another LINE key (alternative SIP account).

•Enter the phone number

•Press the SEND key

or press the “DIAL” soft-key.

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

When redialing the phone will use the same SIP account as was used for the last call. Thus, when the third SIP account was made for the last call/call attempt, the phone will use the third account 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.

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.

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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

4.USING THE CALL HISTORY: To call the a phone number in the phone’s history

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.USING 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 side of phone), or

ii.Pressing the DOWN arrow key, or

iii.Pressing 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.PAGING/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-hook 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” button to override the 4 second delay.

SPEED DIAL

The seven multi-functional buttons, located on the right-hand-side of the phone, can be configured for speed dial. Press the speed dial button to automatically call the assigned extension.

Note: The multi-functional buttons will function as LINE keys when all six primary lines are busy. The LED will flash indicating the incoming call. Press the button to pick up the call. If any one of the 7 functions keys is associated with a call, the speed dial/BLF function will not work.

For example: When first multi-functional button is in use, it cannot be used for speed dial/BLF.

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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 start “ * ” key represent the dot“.” ; The hash “#” key represent colon “:”. Press OK to dial out.

QUICK IP CALL MODE

The GXP2020 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 follow by SEND or # 192.168.0.2 calling 192.168.0.23 -- dial #23 follow 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 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 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” button if you do not want to take a call. This will send the caller directly to voicemail.

2.Press the “DND” 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” 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/DELETE

1.Press the MUTE button to enable/disable muting the microphone.

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

CALL TRANSFER

GXP2020 supports both blind and attended (or supervised) transfer:

1.Blind Transfer: Press “TRNF” 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 call is established, press “TRNF” key to transfer the call and hang up.

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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5-WAY CONFERENCING

The GXP2020 can host conference calls and supports 5-way conference calling.

1.Initiate a Conference Call:

Establish a connection with two or more parties

Press CONF button

Choose the desired line to join the conference by pressing the corresponding LINE button.

Repeat step 2 and 3 for all parties that you want to join the conference. This can be done at any time, thus also if a n

2.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.

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 blinking LINE.

CHECK VOICE MESSAGES (MESSAGE WAITING INDICATOR)

The 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 specific line 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.

ASTERISK BUSY LAMP FIELD

The seven multi-functional buttons can be configured for Asterisk Busy Field function with a specified account. When Asterisk BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when it that line is not in use.

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CALL FEATURES

The GXP2020 supports traditional and advanced telephony features including caller ID, caller ID w/name, call forward/transfer/park/hold as well as intercom/paging and BLF.

Table 8: GXP2020 Call Features

Key		Call Features
*30		Block Caller ID (for all subsequent calls)
*31		Send Caller ID (for all subsequent calls)
*67		Block Caller ID (per call)
*82		Send Caller ID (per call)
*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 “#”. Wait for dial
		tone. LCD will display “Call FWD Activated”.
*73		Cancel Unconditional Call Forward: 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 “#”. Wait 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 “#”. Wait 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

With its graphical LCD screen, soft-keys and XML-support, the GXP2020 is a highly customizable phone. XML is used to communicate between web-servers and the GXP2020 to dynamically update the phone’s phonebook and idle screen logo and functions.

XML PHONEBOOK

The Grandstream GXP2020 enables you to easily share and maintain a phonebook through the web. The XMLphonebook must be stored on a web-server. For more information on how to setup the phone to download the phonebook, please see the Configuration with Web Browser. For more information on how to create a downloadable XML phonebook, detailed information and configuration guides are located on the website in the GXP tab @ http://www.grandstream.com/resources.html.

CUSTOMIZABLE IDLE SCREEN AND SOFT-BUTTONS

The GXP2020 also allows you to customize the idle-screen using your own logo on the display of the phone, instead of the Grandstream logo. In addition to the logo, you can reprogram the soft-keys on the GXP2020 for your own customized applications. For more information about creating a custom idle screen and/or reprogramming the soft-keys, please visit our website at: http://www.grandstream.com/resources.html.

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CONFIGURATION GUIDE

The GXP2020 can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly, through embedded web-configuration menu.

CONFIGURATION WITH KEY PAD

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 enter the key the Key Pad Menu. The menu options available are listed in table 8.

Table 9: Key Pad Configuration Menu

Call History		Displays histories of incoming, dialed and missed calls.
Status		Displays the network status, account statuses, software version and
		MAC-address of the phone.
Phone Book	Displays the phonebook
Instant Messages		Goes to voice messages
Direct IP call		Displays the IP-call options menu
Preference		Displays preference menu. Here ring tones, ring volume and XML screen
		settings can be modified
Config		Display configuration menu. Here network, SIP and audio settings can be
		modified. The FACTORY RESET function is stored here too.
Factory Functions		Extra diagnostic tools reside here.
Reboot		Makes the phone reboot itself.
Exit		Exit from this menu.

Display “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.

•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. Press Menu button to choose the menu item. Press ‘←’ to return to the main menu.

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Display “Configure” Press Menu button to display the configuration selections:

•Network.

To enable/disable DHCP.

To setup IP-address, Net mask and Gateway address

•SIP

To change SIP-server settings for primary account.

•Audio

•Upgrade

In this menu setting regarding the firmware server and config server can be changed. It also enables the user to make the phone attempt to download new firmware.

•Factory Reset

Key in the physical/MAC address on back of the phone.

Press Menu button to reset FACTORY DEFAULT setting. Do not use Factory Reset unless you want to restore factory settings

Press ‘←’ to return the main menu.

Display “Factory Functions” Press Menu to display the factory function items including

•Ethernet Loopback

Connect a cross Ethernet cable from your “PC” port, and the “LAN” port. The test result is displayed on the screen. Use this feature to diagnose the state of health of the RJ45 jacks. Press Menu button to exit the diagnostic mode.

Note: Running the Ethernet Loopback mode with a normal connection will cause IP loss.

•Audio Loopback

Speak into the handset. If you hear your voice in the 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 button to exit

the diagnostic mode.

Press ‘←’ to return to the main menu.

Display “Reboot” Press Menu button to reboot the device

Display “Exit”

Press Menu button to exit the menu

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Figure 7: Key Pad GUI Call Flow

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CONFIGURATION WITH WEB BROWSER

The GXP2020 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 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

1The 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.

2If 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 your browser.

3The 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 administrators are:

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

•Account (1-6): To configure each of the six SIP accounts.

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Table 10: Device Configuration - Status Definitions

Hardware Revision

MAC Address

IP Address

Product Model

Software Version

System Up Time

Registered

PPPoE Link Up

Hardware version number: Main Board, Interface Board

The device ID, in HEXADECIMAL format.

This field shows IP address of GXP2020

This field contains the product model information.

•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

This field shows system up time since the last reboot.

Indicates whether accounts are registered to the related SIP server(s). GXP2020 can support four unique SIP profiles.

Inidicates whether the PPPoE connection is enabled (connected to a modem).

Table 11: Device Configuration – Basic Settings Definitions

	End User		This contains the password to access the Web Configuration Menu. This field is
	Password		case sensitive with a maximum length of 25 characters.
			There GXP2020 operates in two modes:
	IP Address
			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 GXP2020 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 the PPPoE feature, set the
			PPPoE account settings. The GXP2020 establishes a PPPoE session if
			any of the PPPoE fields are 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
			(secondary). These fields are set to zero by default.
			These options are used to assign a function to the corresponding multi purpose key.
	Multi purpose key 1-7
			Options available are: “Speed dial”, “Asterisk BLF”, “Presence Watcher” and
			“Eventlist BLF”.
			Each function is connected to one of the 6 accounts and has a target user ID.
			This parameter controls the date/time display according to the specified time zone.
	Time Zone
			This parameter controls time displayed in daylight savings time or not. If set to
	Daylight Savings Time
			“Yes”, then the displayed time will be 1 hour ahead of normal time.
			LCD time display in 12 hour or 24 hour format
	Time Display Format

Date Display Format Choose one of the following formats:

•Year-Month-Day

•Month-Day-Year

•Day-Month-Year

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

Table 12: Advanced User Configuration Page Definitions

Admin

Password

Silence

Suppression

Voice Frames per TX

Layer 3 QoS

Layer 2 QoS

No Key Entry

Timeout

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

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.

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

When setting this value, be aware of the requested packet time (ptime, used in SDP message) is a result of configuring this parameter. This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. e.g., if the first codec is configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms because each G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or G.711 or G.726, then the “ptime” value in the SDP message of an INVITE request will be 20ms.

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

Please be careful when editing these parameters. Adjusting these parameters will also change the dynamic jitter buffer. The GXP2020 has a patent dynamic jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.

Grandstream recommends using the default settings provided. Grandstream does not recommend adjusting these parameters if you are an average user. Incorrect settings will affect the voice quality. Please refer to the Codec FAQ at http://www.grandstream.com/FAQ/FAQ-Codec.pdf for more technical detail.

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.

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

Default is 4 seconds.

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Use # as

Dial Key

Local RTP port

Use Random

Port

Keep-alive interval

Use NAT IP

STUN Server

Firmware

Upgrade

Via TFTP Server

Via HTTP

Server

Automatic

Upgrade

Phonebook

XML

Idle Screen XML

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.

This parameter defines the local RTP-RTCP 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 and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.

This parameter, when set to “Yes”, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GXP2020s are behind the same NAT. Default is No.

This parameter specifies how often the GXP2020 sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.

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

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

This radio button will enable GXP2020 to download firmware or configuration file through either HTTP or local TFTP server. Choices are mutually exclusive.

This is the IP address of the configured TFTP server. If selected and it is non-zero or not blank, the GXP2020 will attempt to retrieve a new configuration file 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 image is retrieved, the new downloaded image will be verified and then saved into the Flash memory.

Note: Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device.

The HTTP server URL used for firmware upgrade and configuration via HTTP. For example: http://provisioning.mycompany.com:6688/Grandstream/1.1.1.14.

Here “:6688” is the specific TCP port that the HTTP server is using; omit if using default port 80.

Note: If Auto Upgrade is set to No, GXP2020 will only perform HTTP download once at boot up.

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.

Enable the XML phonebook via TFTP or HTTP. Define XML server path and download speed.

Enable XML Idle Screen via TFTP or HTTP. Define XML server path.

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DTMF Payload

Type

Syslog Server

Syslog Level

NTP server

Distinctive Ring

Tone

Disable Call

Waiting

Use Quick IP

Call Mode

Lock keypad update

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

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

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.

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

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 Caller IDs. For all other calls, 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.

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

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

In the Advanced Settings page there is an option "Use Quick IP-call mode". Default setting is No. When set to YES, 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.

If set to “Yes”, the configuration changes via keypad are disabled.

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INDIVIDUAL ACCOUNT SETTINGS

The GXP2020 has 6 line appearances, each with an independent SIP account. Each SIP account requires its own configuration page. Their configurations are identical.

Table 13: SIP Account Configuration Page Definitions
Account Active	This field indicates whether the account is active. The default value for the
	primary account (Account 1) is Yes. The default value for the other two accounts
	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
	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 GXP2020 to register to (SIP)
	servers of 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 number, this field should be set to
Number	“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
	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 on reboot.
Register Expiration	This parameter allows user to specify the time frequency (in minutes) that
	GXP2020 refreshes its registration with the specified registrar. The default interval
	is 60 minutes. The maximum interval is 65,535 minutes (about 45 days).
Local SIP Port	This parameter defines the local SIP port used to listen and transmit. The default
	value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
	Account 4 respectively.

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NAT Traversal

Subscribe for MWI:

Proxy-Require

Voice Mail User ID

Send DTMF

Early Dial

Dial Plan Prefix

Enable Call Features

Session Expiration

Min-SE

Caller Request Timer

Callee Request Timer

Force Timer

UAC Specify Refresher

UAS Specify Refresher

This parameter activates the NAT traversal mechanism. If activated (by choosing “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 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 the NAT Traversal field is set to “Yes” with no specified STUN server, the GXP2020 will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open.

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

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

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

This parameter specifies the mechanism to transmit DTMF digit. There are 3 supported modes: in audio which means DTMF is combined in audio signal (not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.

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

Sets the prefix added to each dialed number.

Default is No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are supported locally provided ITSP support those features.

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

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

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

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

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

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

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

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

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Force INVITE

Enable 100rel

Account Ring Tone

Send Anonymous

Auto Answer

Preferred Vocoder

Jitter Delay

Special Feature

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

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

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.

If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will be set to anonymous, essentially blocking the Caller ID from displaying.

Default is No. If set to “Yes”, GXP2020 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.

The GXP2020 supports up to 5 different Vocoder types including G.711(a/µ) (also known as PCMU/PCMA), GSM, G.723.1, G.729A/B.

Configure Vocoders in a preference list that is included with the same preference order in SDP message. Enter the first Vocoder in this list by choosing the appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by choosing the appropriate option in “Choice 8”.

Jitter Buffer Delay is pre-configured and can not be changed.

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 LCD will then display the following screen to confirm saved changes.

Grandstream recommends to 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 LCD will then display the following screen to confirm reboot is underway.

Wait 30 seconds to log-in again.

Grandstream Networks, Inc.	GXP–2020 User Manual	Page 29 of 32
	Firmware 1.1.4.3	Last Updated: 06/2007

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/1.0.0.18

•168.75.215.189

There are two ways to set up the Upgrade Server to upgrade firmware: 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 GXP2020 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 “Upgrade Server” 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 20 seconds in a controlled LAN or 2-3 minutes over the Internet. Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever possible.

NO LOCAL TFTP SERVER

For users who do not have local TFTP server, Grandstream provides a NAT-friendly TFTP 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 TFTP server IP address: http://www.grandstream.com/y-firmware.htm . Each product has a unique TFTP server 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: http://support.solarwinds.net/updates/NewcustomerFree.cfm.

Grandstream Networks, Inc.	GXP–2020 User Manual	Page 30 of 32
	Firmware 1.1.4.3	Last Updated: 06/2007

DIRECTIONS TO CONFIGURE LOCAL TFTP:

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 GXP2020 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 GXP2020 phone boots up, it will send TFTP or HTTP request to download configuration file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP2020 phone. This file is for initial provisioning purpose only.

•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 GXP2020 can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. “Upgrade Server” 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 3 (Could be extended to 4 in the future) 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 GXP2020 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.

MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD

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

Grandstream Networks, Inc.	GXP–2020 User Manual	Page 31 of 32
	Firmware 1.1.4.3	Last Updated: 06/2007

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 to your VoIP service provider.

DIRECTIONS FOR RESTORATION:

Disconnect network cable and power cycle the unit before resetting factory default settings.

1.Step 1: Press “OK” key to bring up the key pad configuration UI menu, select “Config”, press “OK” to enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)

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

a.0-9: 0-9

b.A: 22 (press the “2” key twice, “A” will show on the LCD)

c.B: 222

d.C: 2222

e.D: 33 (press the “3” key twice, “D” will show on the LCD)

f.E: 333

g.F: 3333

For 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”.

3.Step 3: Press the “OK” key again to move the cursor to “OK” button. Press “OK” key again to confirm. If the MAC address is correct, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.

Grandstream Networks, Inc.	GXP–2020 User Manual	Page 32 of 32
	Firmware 1.1.4.3	Last Updated: 06/2007