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... 12 GETTING FAMILIAR WITH THE LCD.... 12 MAKING PHONE CALLS....15 ANSWERING PHONE CALLS.... 18 PHONE FUNCTIONS DURING A PHONE CALL.... 18 CALL FEATURES..... 20 CUSTOMIZED LCD SCREEN & XML.... 20 CONFIGURATION GUIDE..... 21 CONFIGURATION VIA KEYPAD..... 21 CONFIGURATION VIA WEB BROWSER.... 24 SAVING THE CONFIGURATION CHANGES.... 35 REBOOTING THE PHONE REMOTELY... 36 SOFTWARE UPGRADE & CUSTOMIZATION... 37 FIRMWARE UPGRADE THROUGH TFTP/HTTP... 37 CONFIGURATION FILE DOWNLOAD.... 38 RESTORE FACTORY DEFAULT SETTING.... 39
TABLE OF FIGURES GXP USER MANUAL Figure 1: Connecting the GXP2000 and the GXPExtension.. 6 Figure 2: GXP2000 Internal Headset Wiring Schema... 7 Figure 3: Key Pad GUI Call Flow... 44 TABLE OF TABLES GXP USER MANUAL Table 1: Table 2: Table 3: Table 4: Table 5: Table 6: Table 7: Equipment Packaging... 5 GXP Connectors.... 5 GXP Product Models.... 8 GXP Comparison Guide.... 9 GXP Key Features in a Glance.... 9 GXP Hardware Specifications... 9 GXP Technical Specifications... 9
GXP User Manual Firmware 1.1.6.46 Page 2 of 39 Last Updated: 03/2008
Grandstream Networks, Inc.
Table 8: LCD Buttons.... 12 Table 9: LCD Icons.... 12 Table 10: GXP Keypad Buttons.... 14 Table 11: GXP Call Features.... 21 Table 12: Key Pad Configuration Menu... 42 Table 13: Device Configuration - Status.... 46 Table 14: Device Configuration Basic Settings... 46 Table 15: Advanced Settings.... 48 Table 16: SIP Account Settings.... 53
GUI INTERFACE EXAMPLES GXP USER MANUAL (http://www.grandstream.com/user_manuals/GUI/GUI_GXP.rar) 1. 2. 3. 4. 5. 6. 7. SCREENSHOT OF CONFIGURATION LOGIN PAGE SCREENSHOT OF STATUS PAGE SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE SCREENSHOT OF SIP ACCOUNT CONFIGURATION SCREENSHOT OF SAVED CONFIGURATION CHANGES SCREENSHOT OF REBOOT PAGE
GXP User Manual Firmware 1.1.6.46
Page 3 of 39 Last Updated: 03/2008

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 3rd party SIP platforms. It is ideal for the enterprise customer. The GXP Series supports a broad range of codecs, security protection, PoE (except on GXP-280), dual 10/100mbps Ethernet ports and are very easy to manage. Currently, the GXP Series consists of the following five models: GXP-280, GXP-1200, GXP-2000, GXP-2010 and GXP-2020. Each model delivers superior audio quality using either a handset, hands-free speakerphone or headset and supports multiparty conferencing, multi-languages, dual-color LEDs, presence and BLF (on most models). Large easyto-read backlit graphical displays (except GXP-280) with multiple XML keys further enhance the user experience. 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 examples http://www.grandstream.com/user_manuals/GUI/GUI_GXP.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/GXP_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.

GXP User Manual Firmware 1.1.6.46 Page 4 of 39 Last Updated: 03/2008

Installation

EQUIPMENT PACKAGING
Table 1: Equipment Packaging Main Case Handset Phone Cord Power Adaptor Ethernet Cable High Phone Stand Low Phone Stand Wall Mount Spacers (2) GXP-280 Yes Yes Yes Yes Yes No Yes No GXP-1200 Yes Yes Yes Yes Yes Yes No Yes GXP-2000 Yes Yes Yes Yes Yes No No No GXP-2010 Yes Yes Yes Yes Yes Yes Yes Yes GXP-2020 Yes Yes Yes Yes Yes Yes Yes Yes

CONNECTING YOUR PHONE

The connectors of the GXP1200/2010/2020 are located on the bottom of the device while they are located on the back side of the GXP280/2000.
Table 2: GXP Connectors 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. 5V DC power port; UL Certified RJ22 and 2.5mm for GXP-280/2010/2020 RJ22 for GXP-1200 2.5mm for GXP-2000 HW Rev1.0 or later RJ11
EXT PC LAN Power Jack Headset Jack Handset Jack

GXP-2000 EXTENSION UNIT

GXP2000 supports two (2) extension units, providing up to 112 additional programmable extensions. Each GXP Extension unit has 56 multipurpose keys, dual color LEDs (red/green) and support BLF (Busy Lamp Field) and Presence. GXP2000 Extension package contains:
Grandstream Networks, Inc. Page 5 of 39 Last Updated: 03/2008

1) 2) 3) 4)

One GXP Extension unit One PS2 cable One connection plate One Universal Power Adaptor
FIGURE 1: CONNECTING THE GXP2000 AND THE GXPEXTENSION GXP2000 w/GXPExtension GXP Extension
Connecting the GXP2000 w/GXPExtension
Reverse side of connection w/connection plate
Connect the first GXP EXT to the GXP2000 using the PS2 cable found in the GXP Extension package. The first GXPExt 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 loose power, please unplug your devices and power up the GXP2000 first. Powering up the system: 1. 2. 3. 4. 5. The GXP2000 will boot up first; The GXP LEDs will be solid red; The status light in the top right corner of the GXPExt will blink red; All of the LED indicators on the GXPExt will flash three times; The status light at the top right corner of the GXPExt will turn to solid green.
Note: 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.

GXP-2010 240x120 pixel 3 Yes, up to 2 Expansion Modules, 56 nodes each
GXP-2020 320x160 pixel 4 Yes, up to 2 Expansion Modules, 56 nodes each
Table 5: GXP Key Features in a Glance Features Open Standards Compatible Benefits 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 (pending). Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG, AGC. Dual 10/100mbps Ethernet ports, headset jack (RJ22 and/or 2.5mm jack). Traditional voice features including caller ID, call waiting, hold, transfer, forward, block, autodial, off-hook dial, and click to dial. Multi-line support with dual-color LED (except on GXP-280), multiparty conferencing, line extension interface, large back-lit (except on GXP-280) graphic LCD, 5 or 3 navigation keys, 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. Custom down-loadable ring-tones, SRTP, SIP over TLS (pending), multi-language support and XML enabled, adjustable positioning angles, wall mountable, AES encryption.
Superb Audio Quality Network Interfaces Feature Rich Advanced Features

Advanced Functionality

Table 6: GXP Hardware Specifications
LAN Interface (Ethernet ports) Graphic LCD Display
Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port with auto detection GXP-280 GXP-1200 GXP-2000 GXP-2010 GXP-2020

128x32 pixel

Expansion Module Support GXP-280

GXP-1200

130x64 pixel

240x120 pixel

320x160 pixel

GXP-2020

No Headset Jack
Page 9 of 39 Last Updated: 03/2008

GXP-280

2.5mm and RJ22 Call Appearance LED

2.5mm and RJ22

Dual color (green/red)
GXP-280 GXP-1200 GXP-2000 GXP-2010 GXP-2020

No Power over Ethernet

Universal Switching Power Adaptor Dimension
Built-in auto-sensing: Cisco and IEEE 802.3af standard: phone draws power from both spare lines or signal lines from Ethernet (except on GXP-280) Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA, UL certified GXP-280 GXP-1200 GXP-2000 GXP-2010 GXP-2020 168mm(l) x 200mm(w) x 89.5mm(h) 210mm(l) x 195mm(w) x 77mm(h) 220mm(l) x 215mm(w) x 57mm(h) 210mm(l) x 250mm(w) x 77mm (h) 251mm(l) x 202mm(w) x 77mm(h)
GXP-1200 0.86kg (1.91lbs) GXP-2000 GXP-2010 1.1kg (2.44lbs) GXP-2020

Weight

GXP-280 0.62kg (1.37lbs)

0.82kg (1.81lbs)

1.66kg (3.64lbs)
Temperature Humidity Compliance
F/ 0 40C 10% 90% (non-condensing) FCC / CE / C-Tick
Table 7: GXP Technical Specifications Multiple direct lines with independent SIP accounts, programmable speed dial keys, XML programmable soft-keys. (except on GXP-280) Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, 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) Back-lit graphic LCD display. (GXP-280 display is not back-lit) GXP-280 HOLD

Lines Protocol Support

Display Feature Keys

GXP1200 Yes

GXP-2000 Yes

GXP-2010 Yes

GXP-2020 Yes
Page 10 of 39 Last Updated: 03/2008

Device Management

Audio Features

Telephony Features

Network and Provisioning
Firmware Upgrades Advanced Server Features Security
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 NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed devices 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 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/-law, G.726-32, G.722 (wideband), 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 Intuitive graphic user interface (GUI), downloadable phone book (XML, LDAP), support for anonymous call using privacy header, MLS (multi language support) Voice mail indicator, downloadable custom ring-tones, call hold, call 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 plan prefix, off-hook auto dial, auto answer, early dial and speed dial (on some models) Via keypad/LCD, Web browser, or secure (AES encrypted) central configuration file, manual or dynamic host configuration protocol (DHCP) network setup Support NAT traversal using IETF STUN and Symmetric RTP Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS Support firmware upgrade via TFTP or HTTP, Support for Authenticating configuration file before accepting changes User specific URL for configuration file and firmware files Message waiting indication, support DNS SRV Look up and SIP Server Fail Over, Support customizable idle screen via downloading XML by HTTP/TFTP DIGEST authentication and encryption using MD5 and MD5-sess, SRTP, SIP over TLS (pending)

Page 11 of 39 Last Updated: 03/2008
Using the GXP SIP Enterprise 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 Key Button
LINE SELECTORS SIP PHONE LINES DATE AND TIME LOGO

Key Button Definitions

Selects the phone line printed on its right-hand side. Displays the available phone lines. Choose a phone line by pressing the corresponding line selector on the left-hand side. Displays the current date and time. Can be synchronized with Internet time servers. Displays company logo. This logo can be customized. For more information on customizing the logo, please check page 24. 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. Shows the status of the phone, using icons as shown in the next table. Displays the name of the account that is in use. Select another account by pressing 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 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

NETWORK STATUS

STATUS BAR LINE STATUS INDICATOR
SOFT-BUTTONS (Excluding GXP-2000)

Table 9: LCD Icons

LCD Icon Definitions
Connectivity Status / SIP Proxy/Server Icon: Solid connected to SIP Server/IP address received Blinking physical connection failed Blank SIP Proxy/Server not registered Phone Status Icon: OFF when the handset is on-hook ON when the handset is off-hook

Page 17 of 39 Last Updated: 03/2008

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

Call Transfer

GXP supports both Blind and Attended (or supervised) transfer: 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.
Page 18 of 39 Last Updated: 03/2008
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 TRANSFER (or 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.

5-Way Conferencing

GXP can host conference calls and supports up to 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. 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.

*73 *90

*91 *92
CUSTOMIZED LCD SCREEN & XML
Grandstream GXP Series phones support both simple and advanced XML applications: 1) XML Custom Screen, 2) XML Downloadable Phonebook and 3) Advanced XML Survey Application. For more information on how to create a downloadable XML phonebook, creating a custom idle screen and/or reprogramming the soft-keys on GXP1200/GXP-2010/GXP2020, please visit our website at: http://www.grandstream.com/resources.html.
Page 20 of 39 Last Updated: 03/2008

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 enter the key the Key Pad Menu. The menu options available are listed in table 8.
Table 12: Key Pad Configuration Menu Call History Status Phone Book LDAP Directory Instant Messages Direct IP call Preference Displays histories of incoming, dialed and missed calls. Displays the network status, account statuses, 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 Press Menu button to enter this sub menu including Do NOT Disturb DND (Do NOT Disturb) function could be turned on or off in the DO NOT Disturb menu. Ring Tone Choose different ring tones in the Ring Tone menu. Ring Volume Press Menu button to hear the selected ring volume, press or to hear and adjust the ring tone volume. LCD Contrast 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 Secondary Language Press Menu button to choose the menu item. Press to return to the main menu.
Page 21 of 39 Last Updated: 03/2008

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. 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 Layer 2 QoS Configure Vlan Tags Press to return the main menu.

Factory Functions

Press Menu to display the factory function items including Audio Loopback Speak into the handset. If you hear your voice in the handset, your audio 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. Enable WDT Toggles the status of the Watchdog Timer. Press to return to the main menu. Press Menu button to reboot the device Display Exit Press Menu button to exit the menu Exit from this menu.

Reboot

Page 22 of 39 Last Updated: 03/2008
FIGURE 3: KEYPAD GUI FLOW
Call History Answered Calls Dialed Calls Missed Calls Back Phone Book New Entry Download Phonebook XML Back LDAP Directory View Directory Download Directory Search Configuration Back Instant Message Do Not Disturb Phone Book Clear All Back Preference Do Not Disturb Ring Tone Ring Volume LCD Contrast LCD Brightness Download SCR XML Erase Custom SCR Display Language Back Config Network SIP Upgrade Factory Reset Layer 2 QoS Back Display Language Exit Factory Function Audio Loopback Diagnostic Mode Enable WDT Back English Chinese Secondary Language Language File Postfix Back Ring Tone Default Ring Ring1 Ring2 Ring 3 Back LCD Brightness Active Idle Back Network IP Setting IP NetMask Gateway DNS Server 1 DNS Server SIP Account SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio Save Upgrade Firmware Server Config Server Upgrade Via Layer 2 QoS 802.1Q/VLAN Tag Priority value Reset Vlan Config Back Enable DND Disable DND Back Search Configuration Select Filter Filter Value Back Name: Number: Acct: Confirm Add: Cancel & Return: Any of previous menus Back Clear All New Entry

Call History

Status

LDAP Directory

Instant Message

Direct IP Call

Preference

Config

Diagnostic Mode Keypad/LED Diagnostic
Page 23 of 39 Last Updated: 03/2008

Page 24 of 39 Last Updated: 03/2008
Table 13: Device Configuration - Status MAC Address IP Address Product Model Part Number Software Version The device ID, in HEXADECIMAL format. This field shows IP address of GXP This field contains the product model information. This field contains the product part number System Up Time System Time Registered PPPoE Link Up 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. This field shows the current time on the phone system. Indicates whether accounts are registered to the related SIP server(s). GXP can support four unique SIP profiles. Indicates whether the PPPoE connection is enabled (connected to a modem).
Table 14: Device Configuration Basic Settings End User Password IP Address This contains the password to access the Web Configuration Menu. This field is case sensitive with a maximum length of 25 characters. There 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 the PPPoE feature, set the PPPoE account settings. The GXP establishes 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 (secondary). These fields are set to zero by default. These options are used to assign a function to the corresponding multi purpose key. Options available are: 1. Speed Dial. 2. BLF (Busy Lamp Field). This option has to be supported on the PBX and it indicates the status of the extension. The three 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. Each function is connected to one of the accounts and has a target user ID. This parameter controls the date/time display according to the specified time zone.
GXP User Manual Firmware 1.1.6.46 Page 25 of 39 Last Updated: 03/2008

Multi Purpose Key X

Time Zone

LCD Backlight Always On Time Display Format Date Display Format
Turn on LC backlight at all times. Default is No. This option applies to GXP1200/GXP-2000 only. LCD time display in 12 hour or 24 hour format Choose one of the following formats: Year-Month-Day Month-Day-Year Day-Month-Year This option applies to GXP280/GXP-1200/GXP-2000 only.
Display Clock instead of Choose to display clock or date on LCD. This option applies to GXP-280/GXP1200/GXP-2000 only. Date Daylight Savings Time
This parameter controls time displayed in daylight savings time. If 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 iteration of the weekday (e.g.: 1st Sunday, 3rd Tuesday etc). If weekday is not zero and day is negative, 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 by a negative (-) 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 starts from the second Sunday of March at 2AM and ends the first Sunday of November at 2AM. The saving is 60 minutes.
LCD Backlight Brightness LCD Contrast Disable in-call DTMF display Mute Speaker Ringer in Headset Mode
Set the LCD brightness level. Range from 0 to 8 where 0 means off and 8 means the brightest. For GXP2010 and GXP2020. Set LCD contrast. Range from 0 to 20. Not for GXP280 Default is No. This field is used to hide the keypad input during a call. Default is No. This field lets user to choose whether to ring the phone Speaker when headset is connected.
GXP User Manual Firmware 1.1.6.46 Page 26 of 39 Last Updated: 03/2008
Disable Missed Call Backlight HEADSET Key Mode
Default is No. By default, LCD backlight will lit whenever there is a missed call. Not for GXP280. Set Default mode or choose Toggle Headset/Speaker. Not for GXP2000

Disable Conference Lock keypad update Enable MPK Sending DTMF Disable DND Disable Transfer
Default is No. If set to Yes, conference will be disabled. If set to Yes, the configuration changes via keypad are disabled. Default is No. If set to Yes, Muti-Purpose keys can be sent as DTMF. For GXP2020/2010/2000. Default is No. If set to Yes, the DND button on keypad will be disabled. For GXP2000, MUTE/DEL button functions as DND button when pressed while phone is idle. Default is No. If set to Yes, transfer will be disabled.
If set to Yes, CID will be displayed in the screen instead of Name. Default is No. For Display CID instead of Name GXP280 only.
Grandstream Networks, Inc. GXP User Manual Firmware 1.1.6.46 Page 31 of 39 Last Updated: 03/2008
Headset Port Type Headset TX gain (dB) Headset RX gain (dB) Display Language
Select either 2.5mm or RJ22 headset ports to be adjusted. Increases the selected headsets (2.5mm or RJ22) TX gain by + or 6dB. Default is 0dB Increases the selected headsets (2.5mm or RJ22) RX gain by + or 6dB. Default is 0dB Allows user to choose preferred display language in web UI and key pad UI. User can only load one secondary language. Supported Secondary language: Czech, Dutch, Estonian, French, German, Italian, Polish, Portuguese, Slovak, Slovenian and Spanish. How to set up Secondary Language: 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 5. Reboot the phone. 6. Access the advanced settings of the Web GUI, set Display Language to Secondary Language 7. Update and reboot the phone
GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires its own configuration page. Their configurations are identical. Table 16: SIP Account Settings Account Active 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. The name associated with each account - displayed on LCD. SIP Servers IP address or Domain name provided by VoIP service provider. IP address or Domain name of Outbound Proxy, Media Gateway, or Session Border Controller. Used for firewall or NAT penetration in different network environment. If the system detects symmetric NAT, STUN will not work. ONLY outbound proxy can provide solution for symmetric NAT. User account information provided by VoIP service provider (ITSP); either an actual phone number or formatted like one. SIP service subscribers Authenticate ID used for authentication. It can be identical to or different from SIP User ID. SIP service subscribers account password for GXP to register to (SIP) servers of ITSP. SIP service subscribers name that is used for Caller ID display.

GXP User Manual Firmware 1.1.6.46 Page 32 of 39 Last Updated: 03/2008
Account Name SIP Server Outbound Proxy
SIP User ID Authenticate ID Authenticate Password Name
Use DNS SRV: User ID is Phone Number SIP Registration Un-register on Reboot Register Expiration
Default is No. If set to Yes, the client will use DNS SRV to look up server. If the phone has an assigned PSTN telephone number, this field should be set to Yes. Otherwise, set it to No. If Yes is set, a user=phone parameter will be attached to the From header in SIP request This parameter controls sending REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to Yes, the SIP users registration information will be cleared on reboot. This parameter allows user to specify the time frequency (in minutes) that GXP refreshes its registration with the specified registrar. The default interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days). This parameter 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.

Local SIP Port

SIP Registration Failure Retry registration if the process failed. Default is 20 seconds. Retry Wait Time SIP T1 Timeout SIP T2 Interval SIP Transport Use RFC3581 Symmetric Routing NAT Traversal (STUN) RFC 3261 SIP T1 timer. Default is 1 second. RFC 3261 SIP T2 timer. Default is 0.5 seconds. Choose SIP Transport between UDP and TCP. Default is UDP. Default No. When selected the phone will follow the routing procedures specified in RFC3581. 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 GXP 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. Enable Presence feature. 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.

Page 36 of 39 Last Updated: 03/2008
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/1.1.6.37 168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web Configuration Interface.

Key Pad Menu

To configure the Upgrade Server via Key Pad Menu options, select Config from the Main Menu, then select Upgrade. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN format. Choose Save and use TFTP or Save and use HTTP to select upgrade method. Select Reboot from the Main Menu to reboot the phone.
Web Configuration Interface
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXP IP address. Enter the admin password to access the web configuration interface. In the ADVANCED SETTINGS page, enter the Upgrade Servers 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 Server

For users who do not have a 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/firmware.html. 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.

Avaya Solution & Interoperability Test Lab
Application Notes for the Grandstream Networks SIP Telephones with Avaya Communication Manager 4.0.1 and Avaya SIP Enablement Services 4.0 Issue 0.1

Abstract

These Application Notes describe a solution comprised of Avaya Communication Manager 4.0.1, Avaya SIP Enablement Services 4.0, and Grandstream Networks SIP telephones. Grandstream GXP2020 and GXP1200 are SIP-based VoIP telephones. Grandstream GXP2020 telephone is typically used in an enterprise or small business environment and Grandstream GXP1200 telephone is used by residential or Small Office and Home Office users. During compliance testing, Grandstream telephones successfully registered with Avaya SES, placed and received calls to and from SIP and non-SIP telephones, and executed other telephony features like three-way conference, transfers, holds, etc. Information in these Application Notes has been obtained through compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab.

AT; Reviewed: RRR m/d/y

Solution & Interoperability Test Lab Application Notes 2007 Avaya Inc. All Rights Reserved.
1 of 30 GrandstreamSIP.doc

1. Introduction

These Application Notes describe a solution comprised of Avaya Communication Manager 4.0.1, Avaya SIP Enablement Services (SES) 4.0, and Grandstream Networks SIP telephones. Grandstream GXP2020 and GXP1200 are SIP-based VoIP telephones. Grandstream GXP2020 telephone is typically used in an enterprise or small business environment and Grandstream GXP1200 telephone is used by residential or Small Office and Home Office (SoHo) users. During compliance testing, Grandstream telephones successfully registered with Avaya SES, placed and received calls to and from SIP and non-SIP telephones, and executed other telephony features like three-way conference, transfers, holds, etc. Grandstream telephones can bridge calls on a single line to establish a three-party conference. Grandstream GXP2020 supports up to six and GXP1200 is a single line telephone. Grandstream telephones support IM and Presence but no testing was done because of incompatibility with Avayas implementation. Figure 1 illustrates a sample configuration consisting of a pair of Avaya S8710 Media Servers, an Avaya G650 Media Gateway, an Avaya SIP Enablement Services (SES) server, and the Grandstream telephones. Avaya Communication Manager is installed on the S8710 Media Servers. The solution described herein is also extensible to other Avaya Media Servers and Media Gateways. For completeness, Avaya 4600 Series SIP IP Telephones, Avaya 4600 Series H.323 IP Telephones, and Avaya 6400 and 8400 Series Digital Telephones, are included in Figure 1 to demonstrate calls between the SIP-based Grandstream telephones and Avaya SIP, H.323, and digital telephones. The analog PSTN telephone is also included to demonstrate calls routed by Avaya Communication Manager between the Grandstream telephones and the PSTN. The Grandstream telephone originates a call by sending a call request (SIP INVITE message) to the Avaya SES server. The Avaya SES server routes the call over a SIP trunk to Avaya Communication Manager for origination services. If the call is destined for another local SIP telephone, then Avaya Communication Manager routes the call back over the SIP trunk to Avaya SES server for delivery to destination SIP telephone. Otherwise, Avaya Communication Manager routes the call to the PSTN, a local Avaya H.323, digital, or analog telephone, an adjunct, a vector, a hunt group, etc., depending on the destination number. For a call arriving at Avaya Communication Manager that is destined for the Grandstream telephone, Avaya Communication Manager routes the call over the SIP trunk to the Avaya SES server for delivery to Grandstream telephone. These application notes assume that Avaya Communication Manager and Avaya SES are already installed and basic configuration steps have been performed. Only steps relevant to this compliance test will be described in this document. For further details on configuration steps not covered in this document, consult [3] and [4].
2 of 30 GrandstreamSIP.doc
Figure 1: Sample configuration
3 of 30 GrandstreamSIP.doc
2. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration provided: Equipment Avaya S8710 Media Server Software/Firmware Avaya Communication Manager 4.0.1 (R014x.00.1.731.2) Avaya G650 Media Gateway TN2312BP IP Server Interface HW12, FW040 TN799DP C-LAN Interface HW01, FW024 TN2302AP IP Media Processor HW20, FW117 Avaya SIP Enablement Services Server SES 4.0 (SES-4.0.0.0-033.6) Avaya 4600 Series IP Telephones 2.2.3 (4610SW SIP) 2.3 (4602SW H.323) 2.6 (4610SW H.323) 2.5 (4625SW H.323) Avaya 6400 and 8400 Series Digital Telephones Avaya Analog Telephone Grandstream Networks GXP2020 Telephone 1.1.5.15 Grandstream Networks GXP1200 Telephone 1.1.5.15

3. Configure Avaya Communication Manager
This section describes a procedure for setting up a SIP trunk between Avaya Communication Manager and Avaya SES which includes steps for setting up a list of IP code set, an IP network region, a signaling group and its interface. Before a trunk can be configured, it is necessary to verify if there is enough capacity to setup an additional trunk. Also, a procedure is described here to configure SIP telephones in Avaya Communication Manager. Configuration in the following sections is only for the fields where a value needs to be entered or modified. Default values are used for all other fields. For further information related to configure Avaya Communication Manager refer to [1] and [2]. These steps are performed from the Avaya Communication Manager System Access Terminal (SAT) interface. Grandstream and other SIP telephones are configured as off-PBX telephones in Avaya Communication Manager. Avaya Communication Manager does not directly control an off-PBX telephone but its features and calling privileges can be applied to it by associating a local, on-PBX telephone with the off-PBX telephone. Similarly, a SIP telephone in Avaya SES is associated with an on-PBX telephone on Avaya Communication Manager. SIP Telephones register with the Avaya SES and use Avaya Communication Manager for call origination and termination services. Throughout the rest of this document, on-PBX telephones associated with SIP telephones in such a manner will be referred to as Outboard Proxy SIP (OPS) stations.
4 of 30 GrandstreamSIP.doc
3.1. Capacity Verification
Step Description 1. Enter the display system-parameters customer-options command. Verify that there are sufficient Maximum Off-PBX Telephones OPS licenses. If not, contact an authorized Avaya account representative to obtain additional licenses.
display system-parameters customer-options OPTIONAL FEATURES G3 Version: V13 Location: 1 Platform: 8 Page 1 of 10
RFA System ID (SID): 1 RFA Module ID (MID): 1 USED 50 0
Platform Maximum Ports: Maximum Stations: Maximum XMOBILE Stations: Maximum Off-PBX Telephones - EC500: Maximum Off-PBX Telephones OPS: Maximum Off-PBX Telephones - SCCAN:

Proceed to Page 2 of OPTIONAL FEATURES form. Verify that the number of SIP trunks supported by the system is sufficient for the number of SIP trunks needed. If not, contact an authorized Avaya account representative to obtain additional licenses. Note: Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. The license file installed on the system controls the maximum permitted.
display system-parameters customer-options OPTIONAL FEATURES IP PORT CAPACITIES Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP eCons: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable H.323 Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: 200 Page 2 of 10

USED 153

Maximum Number of DS1 Boards with Echo Cancellation: Maximum TN2501 VAL Boards: Maximum G250/G350/G700 VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: (NOTE: You must logoff & login to effect the permission changes.)
5 of 30 GrandstreamSIP.doc

3.2. IP Codec Set

This section describes the steps for administering codec set in Avaya Communication Manager. This codec set is used in the IP network region for communications between Avaya Communication Manager and Avaya SES. Step 1. Description Enter the change ip-codec-set <c> command, where c is a number between 1 and 7, inclusive. IP codec sets are used in Section 3.3 for configuring IP network region to specify which codec sets may be used within and between network regions. For the compliance testing, G.711MU and G.729AB were used and Media Encryption was set to none as encryption is currently not supported for SIP telephony.
change ip-codec-set 2 IP Codec Set Codec Set: 2 Audio Codec 1: G.711MU 2: G.729AB 3: 4: 5: 6: 7: Silence Suppression n n Frames Per Pkt Packet Size(ms) Page 1 of 2
Media Encryption 1: none 2: 3:
6 of 30 GrandstreamSIP.doc

3.3. IP Network Region

This section describes the steps for administering an IP network region in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SES. Step Description 1. Enter the change ip-network-region <n> command, where n is a number between 1 and 250 inclusive and configure the following: Authoritative Domain Set to the devconnect.com. This should match the SIP Domain value in Section 4, Step 2. Intra-region IP-IP Direct Audio Set to yes to allow direct IP-to-IP audio connectivity between endpoints registered to Avaya Communication Manager or Avaya SES in the same IP network region. Codec Set Set the codec set number as provisioned in Section 3.2. Inter-region IP-IP Direct Audio Set to yes to allow direct IP-to-IP audio connectivity between endpoints registered to Avaya Communication Manager or Avaya SES in different IP network regions.

change ip-network-region 2 IP NETWORK REGION Region: 2 Location: Authoritative Domain: devconnect.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 2 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 65535 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 Page 1 of 19
7 of 30 GrandstreamSIP.doc
Step Description 2. Proceed to Page 3 of IP network region configuration and enable inter-region connectivity between regions as per below. For this compliance testing, codec set was set to the IP codec set configured in Section 3.2.
Page 3 of 19 Inter Network Region Connection Management src rgn 2 dst rgn 15 codec set direct Total WAN WAN-BW-limits y :NoLimit Video Dyn WAN-BW-limits Intervening-regions CAC IGAR n

3.4. IP Node Names

This section describes the steps for setting IP node name for Avaya SES in Avaya Communication Manager. Step Description 1. Enter the change node-names ip command and add a node name for Avaya SES along with its IP address. change node-names ip
IP NODE NAMES Name CLAN-1A06 MEDPRO-1A13 SES IP Address 192.45.100.147 192.45.103.148 192.45.52.160 Page 1 of 1
8 of 30 GrandstreamSIP.doc

3.5. SIP Signaling

This section describes the steps for administering a signaling group in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SIP Enablement Services. Step Description 1. Issue the command add signaling-group <s>, where s is an available signaling group and configure the following: Group Type Set to sip. Transport Method Set to tls. Near-end Node Name - Set to CLAN name as displayed in Section 3.4. Far-end Node Name - Set to Avaya SES name configured in Section 3.4. Far-end Network Region - Set to the region configured in Section 3.3. Far-end Domain - Set to the devconnect.com. This should match the SIP Domain value in Section 4, Step 2.

add signaling-group 10 SIGNALING GROUP Group Number: 10 Group Type: sip Transport Method: tls Page 1 of 5
Near-end Node Name: CLAN-1A06 Near-end Listen Port: 5061 Far-end Domain:devconnect.com
Far-end Node Name: SES Far-end Listen Port: 5061 Far-end Network Region: 2
Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Session Establishment Timer(min): 120 Direct IP-IP Audio Connections? y IP Audio Hairpinning? n
9 of 30 GrandstreamSIP.doc

3.6. SIP Trunking

This section describes the steps for administering a trunk group in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SES. Step Description 1. Issue the command add trunk-group <t>, where t is an unallocated trunk group and configure the following: Group Type Set to the Group Type field value configured in Section 3.5. TAC (Trunk Access Code) Set to any available trunk access code. Signaling Group Set to the Group Number field value configured in Section 3.5. Number of Members Allowed values are between 0 and 255. Set to a value large enough to accommodate the number of SIP telephone extensions being used. Group Name Enter any descriptive name. Note: Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. The license file installed on the system controls the maximum permitted.
add trunk-group 10 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 10 Group Type: SIP-SES-DevCon1 COR: two-way Outgoing Display? n 0 tie Auth Code? sip 1 n CDR Reports: y TAC: 110 Page 1 of 21
Night Service: n Signaling Group: 10 Number of Members: 150
10 of 30 GrandstreamSIP.doc

3.7. SIP Stations

This section describes the steps for administering OPS stations in Avaya Communication Manager and associating the OPS station extensions with the telephone numbers of the Grandstream telephones. Step Description 1. Enter the add station <s> command, where s is an available extension in the dial plan, to administer an OPS station. On Page 1 of the station form configure the following fields: Type Set to 6408D+. Port Set to X. Name Enter any descriptive name.
add station 54007 STATION Extension: Type: Port: Name: 54007 6408D+ X GXP2020 Lock Messages? n Security Code: Coverage Path 1: Coverage Path 2: Hunt-to Station: BCC: TN: COR: COS: Page 1 of 4
STATION OPTIONS Loss Group: Data Module? Speakerphone: Display Language:

2 n 2-way english

Personalized Ringing Pattern: 1 Message Lamp Ext: 54007 Mute Button Enabled? y
Media Complex Ext: IP SoftPhone? n
Proceed to Page 3 of the STATION form and add the required number of call-appr entries in BUTTON ASSIGNMENT field. The number of call appearances should match the Call Limit field value in Step 4.

add station 54007 STATION SITE DATA Room: Jack: Cable: Floor: Building: ABBREVIATED DIALING LIST1: BUTTON ASSIGNMENTS 1: call-appr 2: call-appr 3: call-appr 4: Headset? Speaker? Mounting: Cord Length: Set Color: n n d 0 Page 3 of 3

List2:

List3:

5: 6: 7: 8:

11 of 30 GrandstreamSIP.doc
Step Description 2. Enter the add off-pbx-telephone station-mapping command and configure the following: Station Extension Set the extension of the OPS station as configured above. Application Set to OPS. Phone Number Enter the number that the Grandstream telephone will use for registration and call termination. In the example below, the Phone Number is the same as the Station Extension, but is not required to be the same. Trunk Selection Set to the trunk group number configured in Section 3.6.
add off-pbx-telephone station-mapping Page STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Extension 54007 Application OPS Dial Phone Number Prefix - 54007 Trunk Selection of 2

Configuration Set 1

Proceed to Page 2 of station mapping form and verify that the Call Limit field value matches the number of call appearances configured in Step 2.
add off-pbx-telephone station-mapping 54008 STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Extension 54008 Call Limit 2 Mapping Mode both Calls Allowed Allowed all Page 2 of 2

Bridged Calls both 1

Repeat Steps 1 and 2 as necessary to administer additional OPS stations and associations for Grandstream telephones.
12 of 30 GrandstreamSIP.doc
4. Configure Avaya SIP Enablement Services
This section describes the steps for creating SIP trunk between Avaya SES and Avaya Communication Manager. Also, SIP user accounts are configured in Avaya SES and associated with an Avaya Communication Manager OPS station extension. The Grandstream telephone will register with Avaya SES using the SIP user accounts. For further information related to configure Avaya SES refer to [5] and [6]. Configuration in the following steps is only for the fields where a value needs to be entered or modified. Default values are used for all other fields. Step Description 1. Open a web browser, enter http://<IP address of Avaya SES server>/admin for the URL, and log in with the appropriate credentials. Click on the Launch Administration Web Interface link upon successful login. 2. On the SIP Server Management page: Click the + sign to expand the options under Server Configuration. Click System Properties. Verify the SIP Domain matches the Far-end Domain field value configured for the signaling group on Avaya Communication Manager in Section 3.5.
13 of 30 GrandstreamSIP.doc
Step Description 3. To enable secure SIP trunking between Avaya SES and Avaya Communication Manager, add a media server corresponding to Avaya Communication Manager from the SIP Server Management page: Click the + sign to expand the options under Media Servers. Click Add.

14 of 30 GrandstreamSIP.doc
Step Description 4. At the Add Media Server Interface page, provision SIP Trunk parameters as follows for connectivity to Avaya Communications Manager: SIP Trunk Link Type - Set to the Transport Method field value in Section 3.5. SIP Trunk IP Address - Set to the CLAN IP address as displayed in Section 3.4. Click Add when finished and then click Continue on the confirmation page [not shown].
15 of 30 GrandstreamSIP.doc
Step Description 5. In the left pane of the SIP Server Management page, expand Users and click Add.
16 of 30 GrandstreamSIP.doc
Step Description 6. At the Add User page, configure the following: Primary Handle Enter the phone number of the Grandstream telephone. This number was configured in Section 3.7, Step 3. User ID Set to any descriptive name. Password and Confirm Password Specify a password that the Grandstream telephone will use to register with Avaya SES. Host Select the IP address or Fully Qualified Domain Name (FQDN) of the Avaya SES server. First Name and Last Name Enter descriptive names. Check the Add Media Server Extension checkbox. Click Add when finished and then click Continue on the next page [not shown].
17 of 30 GrandstreamSIP.doc
Step Description 7. At the Add Media Server Extension page, configure the following: Extension Set to Phone Number field value configured in Section 3.7, Step 3. Media Server Set to the media server where this OPS station is configured. Click Add and then click Continue on the next page [not shown]. Note: Media Server was previously configured on SES
Repeat Steps as necessary to configure additional Grandstream telephones.
18 of 30 GrandstreamSIP.doc
Step Description 9. Click Update at the bottom of the left panel to save the configuration completed in the above steps.
10. Click Continue at the bottom of the right panel.
19 of 30 GrandstreamSIP.doc
5. Configure Grandstream Telephones
This section describes the steps for configuring the Grandstream telephones. Grandstream GXP2020 and GXP1200 have similar configuration steps except Grandstream GXP2020 supports up to six separate SIP accounts whereas GXP1200 supports up to two separate SIP Accounts. This section assumes that the Grandstream telephones IP address is already configured. Configuration steps described in this section apply only to the fields where a value needs to be modified or entered. Default values are used for all other fields. Screens shots shown here are for GXP2020 but GXP1200 has similar screens. For further information on Grandstream telephones refer to [5] and [6]. Note: Due to the page size, only the most relevant fields have been included in the screen shots. Step Description 1. Open a web browser and enter http://a.b.c.d for the URL, where a.b.c.d is the IP address of the Grandstream telephone. Enter the password and click Login to proceed to the next screen.

20 of 30 GrandstreamSIP.doc
Step Description 2. Select the BASIC SETTINGS tab and check the statically configured as option to configure as follows: IP Address Set the IP address. Subnet Mask Set the subnet mask. Default Router Set the default router. Click Update to modify the values.
21 of 30 GrandstreamSIP.doc
Step Description 3. Select the ADVANCED SETTINGS tab and configure as follows: Layer 3 QoS Set to the desired value between 0 and 63. For compliance testing, a value of 48 was used. 802.1p priority value Set to the desired value between 0 and 7. For compliance testing, a value of 6 was used. Click Update to modify the values.
22 of 30 GrandstreamSIP.doc
Step Description 4. Select the ACCOUNT1 tab and configure as follows: Account Name Set to the Primary Handle field value configured in Section 4, Step 6. SIP Server Set to the SIP Domain field value configured in Section 4, Step 2. Outbound Proxy Set to the Avaya SES server IP address. SIP User ID Set to the User Id field value configured in Section 4, Step 6. Authenticate ID Set to the User Id field value configured in Section 4, Step 6. Authenticate Password Set to the Password field value configured in Section 4, Step 6. Name Enter any descriptive name. SIP Transport Set to UDP. Send DTMF set to via RTP. Turn off speaker on remote disconnect Set the value to Yes. Special Vocoder This should have at least one of the codecs configured in Section 3.2. Click Update. Repeat this step to configure additional accounts. For GXP2020, up to six accounts can be configured and for GXP1200, up to two accounts can be configured.
23 of 30 GrandstreamSIP.doc
6. Interoperability Compliance Testing
The focus of the interoperability compliance testing was primarily on verifying call establishment on the Grandstream telephones and operations such as dialing methods (manual, re-dial, and phone book), hold, mute, transfer and conference. Grandstream telephones interactions with SES, Avaya Communication Manager, and Avaya SIP, H.323, and digital telephones were also verified.
6.1. General Test Approach
The general test approach was to place calls to and from the Grandstream GXP2020 and GXP1200 telephones and exercise basic telephone operations. The main objectives were to verify that: Grandstream telephones successfully register with Avaya SES. Grandstream telephones successfully establish calls with Avaya SIP, H.323, and digital telephones attached to Avaya SES or Avaya Communication Manager. Grandstream telephones successfully establish calls with PSTN telephones through Avaya Communication Manager. Grandstream telephones successfully handle concurrent calls. Grandstream telephones successfully negotiate the right codec. Grandstream telephones successfully shuffle for VoIP calls. Grandstream telephones successfully transmit DTMF during a call. Grandstream telephones successfully hold and transfer a call. Grandstream telephones establish a three party conference call, and display calling party number. Grandstream telephones successfully tags layer-2 (802.1p) and layer-3 (DiffServ) QoS packets.

For serviceability testing, failures such as cable pulls and hardware resets were applied. For performance testing, a conference call involving two Grandstream telephones and two Avaya telephones was formed as follows: A call was established between an Avaya telephone and a Grandstream telephone. The Grandstream telephone then used its second call appearance to establish a call with another Grandstream telephone, and bridged the two lines together, forming a 3-party conference. The second Grandstream telephone then used its second call appearance to establish a call with another Avaya telephone, and bridged its two lines together, effectively forming a 4party conference.
24 of 30 GrandstreamSIP.doc

6.2. Test Results

The test objectives of Section 6.1 were verified. For serviceability testing, the Grandstream telephones operated properly after recovering from failures such as cable disconnects, and resets of the Grandstream telephones, the Avaya SES server, and Avaya Communication Manager. For performance testing, the conference call was successfully maintained for approximately two hours. Grandstream telephones successfully shuffled to communicate directly with the other telephones. Grandstream telephones successfully negotiated the codec to be used and properly tagged layer-2 and layer-3 QoS packets. The following observations were made during testing: Grandstream telephones cannot mute all parties if it initiates the conference. Only the last party added is muted. Grandstream telephones only support UDP as SIP transport. Grandstream Networks will address and attempt to resolve the above observations in future firmware releases. Contact Grandstream Networks (www.grandstream.com) for further updates.
25 of 30 GrandstreamSIP.doc

7. Verification Steps

The following steps may be used to verify the configuration: Verify that the Grandstream telephones successfully register with the Avaya SES server by following the Users -> Registered Users links on the SES Administration Web Interface. Place calls to and from the Grandstream telephone and verify that the calls are successfully established with two-way talk path. From the Avaya Communication Manager System Access Terminal (SAT) interface, perform the following steps to verify: - Audio codec used between two telephones - Shuffling between two telephones Step Description 1. Enter status trunk <t> command, where t is the SIP trunk configured in Section 3.6. Note down the Member with Service State set to in-service/active. In this example, 0010/002 and 0010/006 are active and either member can be used to verify whether calls shuffled and which codec was used.
Status trunk 10 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy no no no no no no no no no no
0010/001 0010/002 0010/003 0010/004 0010/005 0010/006 0010/007 0010/008 0010/009 0010/010

T00046 T00047 T00048 T00049 T00050 T00051 T00052 T00053 T00054 T00055
in-service/idle in-service/active in-service/idle in-service/idle in-service/idle in-service/active in-service/idle in-service/idle in-service/idle in-service/idle
26 of 30 GrandstreamSIP.doc
Step Description 2. Enter status trunk <m>, where m is the member in active state as noted in the previous step for verification of codec used and shuffling status: Codec The codec used for Audio is G.711MU in this example. Shuffling - If the Near-end IP Addr and Far-end IP Addr for Audio are using the same port and the Audio Connection Type is ip-direct, it signifies that shuffling was successful. In this example, shuffling was successful.
status trunk 10/2 TRUNK STATUS Trunk Group/Member: 0010/002 Port: T00047 Signalling Group ID: Service State: in-service/active Maintenance Busy? No Page 1 of 2
Connected Ports: T0051 Port Signaling: 01A0617 G.711MU Audio: Video: Video Codec: Near-end IP Addr 192. 45.100.147 192. 45. 53.101 : Port : 5061 : 34008 Far-end IP Addr 192. 45. 52.160 192. 45. 53.102 : Port : 5061 : 34008
Authentication Type: None Audio Connection Type: ip-direct
27 of 30 GrandstreamSIP.doc
Step Description 3. Select the STATUS tab at the Grandstream Device Configuration screen and verify the following: Verify the IP Address is correct. Verify the Software Version is correct. Verify the Accounts configured in Section 5, Step 3 are registered with Avaya SES.
28 of 30 GrandstreamSIP.doc

8. Support

For technical support on Grandstream Networks telephones, consult the support pages at http://www.grandstream.com/contact_us.html or contact Grandstream Networks technical support at:
Telephone: 1- (617) E-mail: support@grandstream.com

9. Conclusion

These Application Notes describe a solution comprised of Avaya Communication Manager 4.0.1, Avaya SES 4.0, and Grandstream Networks SIP telephones. Grandstream GXP2020 and GXP1200 are SIP-based VoIP telephones. Grandstream GXP2020 telephone is typically used in an enterprise or small business environment and Grandstream GXP1200 telephone is used by residential or SoHo users. During compliance testing, Grandstream telephones successfully registered with Avaya SES, placed and received calls to and from SIP and non-SIP telephones, and executed other telephony features like three-way conference, transfers, hold, etc. The objective of Section 6.1 were met with some exceptions noted in Section 6.2.
10. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com/. [1] Administrator Guide for Avaya Communication Manager, Issue 2.1, May 2006, Document Number 03-300509 [2] Administration for Network Connectivity for Avaya Communication Manager, Issue 11, February 2006, Document Number 555-233-504 [3] SIP Support in Release 3.1 of Avaya Communication Manager, Issue 6, February 2006, Document Number 555-245-206 [4] Installing and Administering SIP Enablement Services R4.0, Issue 2.0, August 2006, Document Number 03-600768 Product documentation for Grandstream Networks products may be found at http://www.grandstream.com. [5] Grandstream GXP2020 user manual GXP2020UsersManual.pdf [6] Grandstream GXP1200 user manual GXP1200UserManual.pdf

29 of 30 GrandstreamSIP.doc
2007 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please e-mail any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya DevConnect Program at devconnect@avaya.com.
30 of 30 GrandstreamSIP.doc