Table 4
Contents
1. button to override the 4 second delay Speed Dial The Multi Purpose Key buttons located on the right hand side of the phone can be configured for speed dial Press the speed dial button to automatically call the assigned extension Grandstream Networks Inc GXP User Manual Page 17 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video Note The multi functional buttons will function as LINE keys when all LINEs are busy The LED will flash in red to indicate an incoming call Press the button to pick up the call If any one of the Multi Purpose Keys is associated with a call the button s speed dial BLF function will not work Making Calls using IP Addresses Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both phones have public IP addresses or e Both phones are on a same LAN VPN using private or public IP addresses or e 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 Select Direct IP Call using the arrow keys Press OK to select Input the 12 digit target IP address Please see example below Press OK key to initiate call SSN To make a quick IP call please see next section For example f the target2. 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 1 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 Set the LCD brightness level Range from 0 to 8 where 0 means off and 8 means the brightest Default is No This field is used to hide the keypad input during a call Default is No This field lets user
3. d et Innovative IP Voice amp Video Grandstream Networks Inc NENNEN GXP Series SIP Enterprise Phone TABLE OF CONTENTS GXP USER MANUAL WELCOME passende INSTALLATION e EQUIPMENT PAR NENNE eda CONNECTING YOUR PHONE sine GXP 2000 EXTENSION UNEE senere dee SAFETY COMPLINNORS RR TE VID WZRTURSKRSVAHTIAOI WE USING THE GXP SIP ENTERPRISE PHONE eeeeesssvvvvvveessenennneeeesnnnnnneeeennenens GETTING FAMILIAR WITH THE LCD eeeeeeeeeeeee eene nennen hene rentrer eun MAKING PHONE CALLS eden E at ui uU te DUERME end Uo UR ERH DU EP TEE RUE ANSWERING PHONE ALL PHONE FUNCTIONS DURING A PHONE CALL eene hee eene nenn nennen ee TN CUSTOMIZED LCD SCREEN amp SM CONFIGURATION GUIDE wsssvssssssscesscnsssnscessetsssssccvssaanssevesonvecenssseseectuanssessorensenses CONFIGURATION VIA KEY E KB CONFIGURATION VIA WEB BROWSER SAVING THE CONFIGURATION CHANGES cccccscccseccesccsccesccessceusccusscessseescsescseues REBOOTING THE PHONE REMOTELY ccccsccsccsccssccccscesccsccssceccessessenscsscescescessenss SOFTWARE UPGRADE amp CUSTOMIZATION cccccsssccssscccccreeesssssceee FIRMWARE UPGRADE THROUGH IEIROHTIIR nennen CONFIGURATION FILE DOWNLOAD RESTORE FACTORY DEFAULT SETTING eee e eee eere enne eene neenon TABLE OF FIGURES GXP USER MANUAL Figure 1 Connecting the GXP 2000 and the GXP Extension Figure 2 G
4. Basic Settings End User Password IP Address Multi Purpose Key X Time Zone Grandstream Networks Inc 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 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 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 informati
5. 2 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 EXT Draws power from PoE if provided by network PC 10 100Mbps RJ 45 ports for PC downlink connection LAN 10 100Mbps RJ 45 port for LAN uplink connection Supports PoE 802 3af Draws power from either spare line or signal line Power Jack 5V DC power port UL Certified RJ22 and 2 5mm for GXP 280 2010 2020 Headset Jack RJ22 for GXP 1200 2 5mm for GXP 2000 HW Rev1 0 or later Handset Jack RJ11 GXP 2000 EXTENSION UNIT GXP 2000 supports two 2 extension units providing up to 112 additional programmable extensions Each GXP Extension unit has 56 multi purpose keys dual color LEDs red green and support BLF Busy Lamp Field and Presence GXP 2000 Extension package contains Grandstream Networks Inc GXP User Manual Page 5 of 39 Firmware 1 1 6 16 Last Updated 05 2008 randstream Innovative IP Voice amp Video 1 One GXP Extension unit 2 One PS2 cable 3 One connection plate 4 One Universal Power Adaptor FIGURE 1 CONNECTING THE GXP 2000 AND THE GXP EXTENSION GXP 2000 w GXP Extension GXP Extension Connecting the GXP 2000 Reverse side of connection w GXP Extension w connection plate Connect the first GXP EXT to the GXP 20
6. Direct IP call Displays the IP call options menu Preference Press Menu button to enter this sub menu including e Do NOT Disturb DND Do NOT Disturb function could be turned on or off in the DO NOT Disturb menu e Ring Tone Choose different ring tones in the Ring Tone menu e Ring Volume Press Menu button to hear the selected ring volume press or to hear and adjust the ring tone volume e LCD Contrast e Download SCR XML The phone will download the custom idle screen if available e 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 Grandstream Networks Inc GXP User Manual Page 23 of 39 Firmware 1 1 6 16 Last Updated 05 2008 gt Frandstream Innovative IP Voice amp Video Configure Press Menu button to display the configuration selections e 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 Layer 2 QoS C
7. In the ADVANCED SETTINGS page enter the Upgrade Server s IP address or FQDN in the Firmware Server Path field Select TFTP or HTTP upgrade method Update the change by clicking the Update button Reboot or power cycle the phone to update the new firmware During this stage the LCD will display the firmware file downloading process If a firmware upgrade fails for any reason e g TFTP HTTP server is not responding there are no code image files available for upgrade or checksum test fails etc the phone will stop the upgrading process and re boot using the existing firmware software Firmware upgrades take around 60 seconds in a controlled LAN or 5 10 minutes over the Internet Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever possible No Local TFTP 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 New customerFree cfm Grandstream Networks Inc GXP User Manual Page 37 of 39 Firmware 1 1 6 16 Last Updated 05 2
8. ON is the period of ringing On time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Up to three cadences are supported This field is used to configure the Intercom key in the phone If the phone is working with a GS GXE502X IP PBX it can be configured in the following manner e To page an extension intercom feature code extension number e To page a group paging group feature code group extension Default is No If set to Yes the call waiting feature will be disabled Default is No If set to Yes the call waiting tone will be disabled 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 lt 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 Select either 2 5mm or RJ22 headset ports to be adjusted Headset TX Increases the selected headset s 2 5mm or RJ22 TX gain by or 6dB Default is 0dB gain dB Grandstream Network
9. 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 Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is required to support PSTN inter networking There are 4 uniquely defined ring tones e One 1 System Ring Tone when selected all calls
10. 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 6045062 The key represent the dot The key represent colon t Press OK to dial out Quick IP Call Mode The GXP also supports Quick IP call mode This enables the phone to make direct IP calls using only the last few digits last octet of the target phone s IP number This is possible only if both phones are in under the same LAN VPN This simulates a PBX function using the CMSA CD without a SIP server Controlled static IP usage is recommended Setting up the phone to make Quick IP calls To enable Quick IP calls the phone has to be setup first This is done through the web setup function In the Advanced Settings page set the Use Quick IP call mode to YES When xxx is dialed where x is 0 9 and xxx lt 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 423 follow by SEND or 192 168 0 2 calling 192 168 0 123 dial 4123 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 wi
11. lom ot O mate _ 1 37Ibs 19108 181ls 2 44bs 3 64bs Temperature 82 44 FIO AQG 0000000000000 Humidity 10 90 non condensing AM Compliance FCC CE C Tick Table 7 GXP Technical Specifications Grandstream Networks Inc GXP User Manual Page 10 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video Lines Multiple direct lines with independent SIP accounts programmable speed dial keys ossi XML programmable soft keys except on GXP 280 0 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 SUIS up to bs coming IM messages EE IM sk 101 plus em zm o om mm mm e zm o zm zm o e zm fr em em om mm em mm wm om em e em em em om mm wm om em e e em o e em vm e em vm 4 vm e e em wm em a Mp oc wm wm vm 4 4 em wm em wm 4m wm vm em wm wm em 4 em 4 wm ve wm mew 4 em 4 4 ve 4 wm vm vm 4 em 4 4 o 4 wm wm wm e 4 4 o 4 4 wm wm em 4 4m 4 ow 4 4 wm 4
12. 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 Layer 3 QoS This field defines the layer 3 QoS parameter It is the value used for IP Precedence or Diff Serv or MPLS Default value is 48 Grandstream Networks Inc GXP User Manual Page 29 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Layer 2 QoS No Key Entry Timeout Use as Dial Key Local RTP port Use Random Port Keep alive interval Use NAT IP STUN Server Firmware Upgrade and Provisioning Via TFTP Server Via HTTP Server Config Server Path Firmware File Prefix Postfix ndstream Innovative IP Voice amp Video This contains the value used for layer 2 VLAN tag Default setting is blank Default is 4 seconds This parameter allows users to configure the key as the Send or Dial key If set to Yes fe 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 T
13. or two expansion module The series is based on SIP standard and are interoperable with most 3rd party SIP platforms and open source 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 e 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 e 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 e 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 Grandstream Networks Inc GXP User Manual Page 4 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video Installation EQUIPMENT PACKAGING Table 1 Equipment Packaging pC GXP 280 GXP 1200 GXP 2000 GXP 2010 GXP 2020 Handset Yes Yes Yes Yes Yes Ethernet Cable High Phone Stand Low Phone Stand Wall Mount Spacers
14. phone e Make sure the phone is turned on and shows its IP address e Start a Web browser on your computer e Enter the phone s IP address in the address bar of the browser e Enter the administrator s password to access the Web Configuration Menu The Web enabled computer has to be connected to the same sub network as the phone This can easily be done by connecting the computer to the same hub or switch as the phone is connected to In absence of a hub switch or free ports on the hub switch please connect the computer directly to the phone using the PC port on the phone If the phone is properly connected to a working Internet connection the phone will display its IP address This address has the format xxx xxx xxx xxx where xxx stands for a number from 0 255 You will need this number to access the Web Configuration Menu e g if the phone shows 192 168 0 60 please use http 192 168 0 60 in the address bar your browser wo The default administrator password is admin the default end user password is 123 NOTE When changing any settings always SUBMIT them by pressing the button on the bottom of the page Reboot the phone to have the changes take effect If after having submitted some changes more settings have to be changed press the menu option needed Definitions This section will describe the options in the Web configuration user interface As mentioned a used can log in as an administrator or end user Func
15. service provider ITSP either an actual phone number or formatted like one SIP service subscriber s Authenticate ID used for authentication It can be identical to or different from SIP User ID SIP service subscriber s account password for GXP to register to SIP servers of ITSP SIP service subscriber s name that is used for Caller ID display 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 user s 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 GXP User Manual Firmware 1 1 6 16 Page 33 of 39 Last Updated 05 2008 Local SIP Port SIP Registration Failure Retry Wait Time SIP T1 Timeout SIP T2 Interval SIP Transport Use RFC3581 Symmetric Routing NAT Traversal STUN Subscribe for MWI PUBLISH for Presence Proxy Require Voice Mail UserID Send DTMF Early Dial Dial Plan Prefix Delayed Call Fo
16. 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 the document at http www grandstream com documents XMLApplicationGuide Rev1 1 pdf 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 e NEW CALL Press this button to make a new hand free call e FORWARD ALL Unconditionally forwards the main phone line to another phone e MISSED CALLS This option shows up there were unanswered calls to this phone The MissedCalls option shows a list of the missed calls e CALL RETURN Calls the phone that called tried to call your phone last e REDIAL Redials the last number e END CALL Hangs up phone icon LCD Icon Definitions Connectivity Status SIP Proxy Server Icon Grandstream Networks Inc Solid connected to SIP Server IP address received Blinkin
17. the display of the phone instead of the Grandstream logo In addition to the logo you can reprogram the soft keys on GXP 1200 GXP 2010 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 Grandstream Networks Inc GXP User Manual Page 22 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video 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 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 LDAP Directory Displays the LDAP directory Instant Messages Goes to voice messages
18. the name of the account as configured in the web interface is displayed on the LCD and a dial tone is heard For example Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as VoIP 1 VoIP 2 respectively and ensure that they are active and registered When LINET is pressed you will hear a dial tone and see VoIP 1 on the LCD display when LINE2 is pressed you will hear a dial tone and see VoIP 2 on the LCD display To make a call select the line you wish to use The corresponding LINE LED will light up in green User can switch lines before dialing any number by pressing the same LINE button one or more times If you continue to press a LINE button the selected account will circulate among the registered accounts For example when LINE1 is pressed the LCD displays VoIP 1 If LINE1 is pressed twice the LCD displays VoIP 2 and the subsequent call will be made through SIP account 2 Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use When the virtually mapped line is in use the GXP will flash the next available LINE from left to right or from top to bottom for Multi Purpose Keys in red A line is ACTIVE when it is in use and the corresponding LED is red Completing Calls There are six ways to complete a call 1 DIAL To make a phone call e Take Handset SPEAKER Headset off hook or press an available LINE key activates speakerphone or pres
19. 00 using the PS2 cable found in the GXP Extension package The first GXP Ext draws power directly from the phone Connect the second GXP Extension unit using the connection plate and the PS2 cable The GXP2000 will automatically reboot and power up the GXP Extensions Grandstream recommends though not required to use a separate power supply with the second GXP Ext NOTE should your system loose power please unplug your devices and power up the GXP 2000 first Powering up the system 1 The GXP 2000 will boot up first 2 The GXP LEDs will be solid red 3 The status light in the top right corner of the GXP Ext will blink red 4 Al of the LED indicators on the GXP Ext will flash three times 5 The status light at the top right corner of the GXP Ext will turn to solid green Grandstream Networks Inc GXP User Manual Page 6 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video Figure 2 GXP 2000 Internal Headset Wiring Schema BT100 handset jack BT100 handset plug abcd headset Jack BT100 power jack Bottom view ee NOTE For GXP 2000 HW REV 0 3 and 0 4 a 3 5mm to 2 5mm plug converter is required to use a 2 5mm headset The converter can be purchased at any electronics store SAFETY COMPLIANCES The GXP phone complies with FCC CE and various safety standards The GXP power adaptor is compliant with the UL standard Only use the universal power adaptor provided with the GXP packa
20. 008 Frandstream Innovative IP Voice amp Video Instructions for local TFTP Upgrade 1 Unzip the file and put all of them under the root directory of the TFTP server 2 The PC running the TFTP server and the GXP should be in the same LAN segment 3 Go to File gt Configure gt 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 e When GXP phone boots up it will send TFTP or HTTP request to download configuration file cfgO00b82xxxxxx where 000b82xxxxxx is the MAC address of the GXP phone This file is for provisioning purpose For normal TFTP or HTTP firmware upgrades the following error messages in a TFTP or HTTP server log can be ignored TFTP Error from IP ADRESS requesting cfq9000b82023d0d4 File does not exist Configuration File Download CONFIGURATION FILE DOWNLOAD The GXP can be configured via Web Interface as well as via Configuration File through TFTP or HTTP Config Server Path is the TFTP or HTTP server path for the configuration file It needs to be set to a valid URL either in FQDN or IP address format A configuration parameter is a
21. 4m 4 ow 4 wm wm em 4m 4 4 ve o wm wm em em 4 4 ve 4 wm wm wm ven wm wm wm em Feature Keys GXP 280 GXP i GXP 2000 GXP 2010 GXP 2020 EE EN EM E HOLD Yes 00 Ves ee Yes Yes SPEAKERPHONE Yes Yes Yes Yes Yes SEND Yves Yes Yes Yes e TRANSFER Yes e ies Yes es CONF ij Yes i Yes es e Yes MUTE Yes 1 Yes Yes Yess OS X DND LL Yes ves gt Yes Yes Ves 90 3 HEADSET Yes e ie e Yes INTERCOM No gt No No ee Yes gt PHONEBOOK No Mo Wo Yes X Yes MSG Yes gt Yes Yes e OS MENU Yes gt Yes ee gt Yes gt Yes gt sss NAVIGATION 4 Yes 3 Yes Yes Yes es 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 PNE 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 u 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 a
22. XP 2000 Internal Headset Wiring Schema Figure 3 Key Pad GUI Call Flow rannrrnnnnnrnnnnnrnnnnnnnnrannvnnnnnrnnnnnnnnnnnnennnnnene TABLE OF TABLES GXP USER MANUAL Table 1 Equipment Packaging ccccsscccsseeeseeeeseeeeseeseeeeeeeseeeeseeeeaeeeenes Table 2 GXP Connectors ceo eucse Ee Ed Table 3 GXP Product Models rrrrrrrnnrrnnnnnrvvvnrrrnnnnnvnnnnnnnnrssvnnvrnnnnssenenrnnnsnee Table 4 GXP Comparison Guide sse Table 5 GXP Key Features maGlance cece cecceseeeeeeeeeeeeeeeseeeseeeeaes Table 6 GXP Hardware Gpectcatons Table 7 GXP Technical Specifications 000n000n000nnannnannnnnnnnnnnennnneennnn Grandstream Networks Inc GXP User Manual Firmware 1 1 6 16 gt Frandstream Innovative IP Voice amp Video Page 2 of 39 Last Updated 05 2008 gt Frandstream Innovative IP Voice amp Video Fe PN Saag 13 FER Ree 13 Table 10 GXP Keypad Buttons 2 00 0 ceccccccceseeeeceeceaeeeeaeeeeseeceseeeeseeceseueessueeseeeeseueetausesseeesanees 15 Table 11 GXP Call E RE 21 Table 12 Key Pad Configuration Menu 23 Table 13 Device Configuration Status rrrrnnrrrannrnrnnnrrrnnnrrrnnnrrnrnnrrnnnrenrnenennnnnennnnnennnnsrnnnsnnnnnn 27 Table 14 Device Configuration Basic Gettngs 27 Table 15 Advanced RE dee E 29 Table 16 SIP Account EE e E 33 GUI INTERFACE EXAMPLES GXP USER MANUAL http Awww grandstream com
23. all made to the third SIP account the phone will use the third SIP account return the call e Press the MENU button to bring up the Main Menu e Select Call History and then Received Calls Missed Calls or Dialed Calls depending on your needs e Select phone number using the arrow keys e Press OK to select e 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 e Goto the phonebook by i Pressing the phonebook button bottom left hand side of phone or li Pressing the DOWN arrow key or iii Pressing the menu button and Selecting Phone book and Press MENU e Select the phone number by using the arrow keys e Press OK so select e 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 e Take the Handset SPEAKER Headset off hook e Select the LINE key associated with account e Press OK key to display LCD LINEx PAGE USING e Dial the phone number you want to Page Intercom e 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 or
24. alls 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 f after pressing the CONF button a user decides not to conference anyone press CONF again or the original LINE button his 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 Voice Messages Message Waiting Indicator A blinking red MWI Message Waiting Indicator indicates a message is waiting Press the MSG button to retrieve the message An IVR will prompt the user through the process of message retrieval Press a specific LINE to retrieve messages for a specific line account NOTE e Each line has a separate voicemail account Each account requires a voicemail portal number to be configured in the voicemail user id field e To check which line account has a message 1 press the mes
25. ands free speaker mode Press SEND to dial a new number or redial the last number dialed Press SEND send button to send a call immediately before no key entry timeout value expires P 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 SCH functions 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 TT Brings phonebook on screen Grandstream Networks Inc GXP User Manul Page 150f39 Firmware 1 1 6 16 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video 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 GXP can support up to six independent SIP accounts depending on the product model Each account is capable of independent SIP server user and NAT settings Each of the line buttons is virtually mapped to an individual SIP account The name of each account is conveniently printed next to its corresponding button In off hook state select an idle line and
26. at 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 Time waited before the call is forward to a number or VM Default is 20 seconds Default is No If set to Yes Call transfer Call Forwarding amp Do Not Disturb are supported locally provided ITSP support those features User can choose to disable Call Log and what kind of calls to log GXP User Manual Page 34 of 39 Grandstream Networks Inc Firmware 1 1 6 16 Last Updated 05 2008 Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Enable 100rel Account Ring Tone Send Anonymous Anonymous Method Anonymous Call Rejection Auto Answer Allow Auto Answer by Call Info ndstream Innovative IP Voice amp Video 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
27. e ew A M 4 vm ve ew eo gee ao vm meme wm wm vm vm m 4 em wm vm eee ow mmm 4 vm wm wow mmm wm vm wow ee 4 mmo vm ew i mmm wm vm mm 4 4 vm o 4 mee vm e vm vg memo ew ew e 4 vm mew we ew 4o 4 vm vm ew ow 4 oe wm vm oe wm Firmware Support firmware upgrade via TFTP or HTTP Upgrades Support for Authenticating configuration file before accepting changes SE User specific URL for configuration file and firmware files gt Advanced Message waiting indication support DNS SRV Look up and SIP Server Fail Over Server Features Support customizable idle screen via downloading XML by HTTPITFTP Security DIGEST authentication and encryption using MD5 and MD5 sess SRTP SIP over TLS pending Grandstream Networks Inc GXP User Manual Page 12 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video 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 Key Button Definitions LINE SELECTORS SIP PHONE LINES DATE AND TIME LOGO NETWORK STATUS STATUS BAR LINE STATUS INDICATOR SOFT BUTTONS Excluding GXP 2000 Table 9 LCD Icons Selects the phone line printed on its right hand side Displays the available phone lines Choose a phone line by pressing
28. firmware upgrade and configuration via HTTP For example http provisioning mycompany com 6688 Grandstream 1 1 6 16 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 GXP will only perform HTTP download once at boot up IP address or domain name of firmware server Default is blank If configured GXP will request the firmware file with the prefix postfix This setting is useful for ITSPs End user should keep it blank GXP User Manual Grandstream Networks Inc Page 30 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Config File Prefix Postfix Allow DHCP Option 66 to override server Authenticate Conf File Automatic Upgrade Phonebook XML Idle Screen XML Download XML Application Offhook Auto Dial DTMF Payload Type Syslog Server Syslog Level Grandstream Networks Inc ndstream Innovative IP Voice amp Video Default is blank End user should keep it blank Default is Yes This allows device gets provisioned automatically Default is No If set to Yes configuration file would be authenticated before acceptance End user should use default setting 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 ser
29. g physical connection failed Blank SIP Proxy Server not registered GXP User Manual Page 13 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Innovative IP Voice amp Video 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 DND Icon ON when the do not disturb is activated Activate by pressing MUTE DEL button twice Calls Forwarded Icon INDICATES calls are forwarded Follow call forwarding procedures Handset Speakerphone and Ring Volume Icon Each icon appears next to the volume icon To adjust volume use the up down button Real time Clock Synchronized to Internet time server Time zone configurable via web browser AM PM indicator Grandstream Networks Inc GXP User Manual Page 14 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video Table 10 GXP Keypad Buttons Key Button Key Button Definitions LINE BUTTONS Line keys with LED can be configured to different SIP profiles TRANSFER TRANSFER key Transfer an ACTIVE call to another number CONF Press CONF button to connect Calling Called party into conference Mute an active call or Delete a key entry E Also used to REJECT incoming call HOLD Place ACTIVE call on hold MSG Enter to retrieve voice mails or other messages BE Enable Disable h
30. ge 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 Grandstream Networks Inc GXP User Manual Page 7 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ream Innovative IP Voice amp Video Product Overview Table 3 GXP Product Models GXP280 is an entry level SIP phone It features e Single line e Three soft keys GXP 280 GXP1200 is an entry level SIP phone It features e Two lines e Three XML programmable soft keys GXP 1200 S3 GXP2000 is a mainstream SIP phone It features v i e Four lines w e leh k GXP 2000 lt gt Seven programmable hard keys KE GXP2010 is a key system SIP phone It features N e Four lines a e Eighteen programmable hard keys GXP 2010 up Three XML programmable soft keys Grandstream Networks Inc GXP User Manual Page 8 of 39 Firmware 1 1 6 16 Last Updated 05 2008 randstream Innovative IP Voice amp Video GXP2020 is an executive SIP phone It features e Six lines e Seven programmable hard
31. gle 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 GXP 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
32. he default value is 5004 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple GXPs are behind the same NAT Default is No This parameter specifies how often the GXP 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 Default method is HTTP Firmware upgrade may take up to 10 minutes depending on network environment Do not interrupt the firmware upgrading process This is the IP address of the configured TFTP server If selected and it is non zero or not blank the GXP 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
33. ice to your VoIP service provider INSTRUCTIONS FOR RESTORATION Step 1 Press OK button to bring up the keypad configuration menu select Config press OK to enter submenu select Factory Reset Please refer to Table 5 1 of keypad flow chart Step 2 Enter the MAC address printed on the bottom of the sticker Please use the following mapping 0 9 0 9 A 22 press the 2 key twice A will show on the LCD B 222 C 2222 D 33 press the 3 key twice D will show on the LCD E 333 F 3333 Example if the MAC address is 000582006395 it should be key in as 0002228200333395 NOTE If there are digits like 22 in the MAC you need to type 2 then press gt right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2 Step 3 Press the OK button to move the cursor to OK Press OK button again to confirm If the MAC address is correct the phone will reboot Otherwise it will exit to previous keypad menu interface Grandstream Networks Inc GXP User Manual Page 39 of 39 Firmware 1 1 6 16 Last Updated 05 2008
34. ious menu Back Clear All New Entry New Entry Back Instant Messages Back Clear All pr Do NOT Disturb Preference Ring Tone Ring Volume Download SCR XML Erase Custom SCR Config Network SIP Audio Upgrade Factory Reset Back Factory Functions Ethernet Loopback Audio Loopback Name Number Account Confirm Add Cancel amp Return Do NOT Disturb Enable DND Disable DND Back Ring Tone Default Ring Ring 1 Ring 2 Ring 3 Back SIP Primary A C ONLY p SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password Save Cancel Audio WR Select PCMU Select PCMA Select G 723 1 Select G 729 A B Select GSM Cancel Upgrade pr TFTP Server HTTP Server Save and use TFTP Save and use HTTP Cancel Keypad LED Diagnostic Mode Back GXP User Manual Firmware 1 1 6 16 Diagnostics C sen Innovative IP Voice amp Video Page 25 of 39 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video CONFIGURATION VIA WEB BROWSER The GXP embedded Web server responds to HTTP HTTPS GET POST requests Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft s IE or Mozilla Firefox Access the Web Configuration Menu To access the phone s Web Configuration Menu e Connect the computer to the same network as the
35. irm that reboot is underway Wait 30 seconds to log in again Grandstream Networks Inc GXP User Manual Firmware 1 1 6 16 Page 36 of 39 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video Software Upgrade amp 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 e firmware mycompany com 6688 Grandstream 1 1 6 16 e 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
36. keys e Four XML programmable soft keys GXP 2020 Table 4 GXP Comparison Guide Features GXP 280 GXP 1200 GXP 2000 GXP 2010 GXP 2020 LCD Display 128x32 pixel 128x32 pixel 130x64 pixel 240x120 pixel 320x160 pixel Number of Lines 1 88 200 0 0 MEM r r 6 Programmable No No 7 48 T ME Hard Keys SoftKeys 3 0 NN o F vw 4 Extension Module No No Yesupto2 Yes Yes Expansion Modules 56 nodes each Table 5 GXP Key Features in a Glance Features Benefits we wm wm e wm vm wm em wm em e em wm wm e wm wm wm wm wm wm e wm wm wm e wm wm vm wm wm wm ee wm em mm wm wm e ff em e wm e mm wm e wm wm wm e wm wm wm em e em wm wm em wm wm e em mm wm e wm em e wm em e wm e e wm em e wm wm e wm wm wm wm 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 pending Superb Audio Quality Advanced Digital Signal Processing DSP Silence suppression BS VAD CNG AGC Network Interfaces Dual 10 100mbps Ethernet ports headset jack RJ22 and or 2 5mm E SD Feature Rich Traditional voice features including caller ID call waiting hold septer transfer forward block autodial off hook dial and click to dial Advanced Features Multi line support with dual color LED except on GXP 280 multi pa
37. ll also use STUN Configure the Use Random Port to NO when completing Direct IP calls Grandstream Networks Inc GXP User Manual Page 18 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video 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 Re
38. nformation Default is No GXP supports up to 7 different Vocoder types including G 711 a u also known as PCMU PCMA GSM G 723 1 G 729A B G 726 32 iLBC G 722 wide band Configure Vocoders in a preference list that is included with the same preference order in SDP message Enter the first Vocoder in this list by choosing the appropriate option in Choice 1 Similarly enter the last Vocoder in this list by choosing the appropriate option in Choice 8 Enable SRTP mode based on selection Default is No If a server supports this feature user needs to configure an eventlist BLF URI on the service side i e BLE1006 myserver com On the GXP under Account page fill in the eventlist BLF field with the URI without the domain i e BLF1006 Under Basic Settings please select eventlist BLF choose account nubmer monitored number etc Default is Standard Choose the selection to meet special requirements from Soft Switch vendors SAVING THE CONFIGURATION CHANGES After the user makes a change to the configuration press the Update button in the Configuration Menu The web browser will then display a message window to confirm saved changes Grandstream recommends reboot or power cycle the IP phone after saving changes REBOOTING THE PHONE REMOTELY Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely The web browser will then display a message window to conf
39. o vm mm mm vm me vm vm e vm mm em vm mm mm vm mm vm mm mm vm mm om vm e wm mm mm dm mm mm mm mm wm mm mm mm mm mm vm mm em wm em wm wm wm wm wm wm mm mm wm mm mm mm mm mm mm mm wm mm mm mm mm mm om vm em mm em mm mm wm om mm mm mm mm wm mm mm mm mm mm vm mm wm om mm wm vm wm om mm wm vm mm em mm wm mm mm vm mm mm wm mm wm Headset Jack GXP 280 GXP 1200 GXP 2000 GXP 2010 GXP 2020 RJ22 RJ22 2 5mm 2 5mm and 2 5mm and and RJ22 RJ22 Call Appearance LED Dual color greenired l i EE GXP 280 E _ GXP 1200 E 2000 i GXP 2010 7 GXP 2020 Nu Power over Ethernet Built in teen l EG E C phone draws power from both spare lines or signal lines from Ethernet except on GXP 280 ewe ee em we we ew mm we ew ew ew ew ew ew ew ew ew mm mm ew zm ew mm ew mm ew ew ew ew mm mm mm mm mm ew ew ew ep em mm ew ew we em em we mm mm mm mm mm mm mm mm ew mm mm mm mm mm mm ew mm em ew ew ew mm mm mm mm 7 ew mm mm mm mm ew ee mm mm vm ee mm em mm mm mm mm ee ee mm Universal Switching Input 100 240VAC 50 60 HZ n PowerAdapto L un ue ee Dimension esses GXP 280 168mm l x 200mm w x 89 5mm h GXP 1200 210mm x 195mm w x 77mm h GXP 2000 i 220mm l x 215mm w x 57mm h GXP 2000 210mm l x 250mm w x 77mm ui GXP 2020 i251mm x202mm w x77mm h Weight 0000000000 ETERNI TRES IGNES TAE lone
40. on 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 Page 27 of 39 Firmware 1 1 6 16 Last Updated 05 2008 LCD Backlight Always On Time Display Format Date Display Format Display Clock instead of Date Daylight Savings Time LCD Backlight Brightness Disable in call DTMF display Mute Speaker Ringer in Headset Mode Disable Missed Call Backlight Grandstream Networks Inc ndstream Innovative IP Voice amp Video Turn on LC backlight at all times Default is No This option applies to GXP 1200 GXP 2000 only LCD time display in 12 hour or 24 hour format Choose one of the following formats e Year Month Day e Month Day Year e Day Month Year This option applies to GXP 1200 GXP 2000 only Choose to display clock or date on LCD This option applies to GXP 280 GXP 1200 GXP 2000 only 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 e Start time end time saving e Both start time and end time have the same syntax month day weekday hour minute o month 1 2 3 12 for Jan Feb
41. onfigure Vlan Tags Press to return the main menu Factory Functions Press Menu to display the factory function items including e 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 e Enable WDT Toggles the status of the Watchdog Timer Press to return to the main menu Reboot Press Menu button to reboot the device Display Exit Press Menu button to exit the menu Exit Exit from this menu Grandstream Networks Inc GXP User Manual Page 24 of 39 Firmware 1 1 6 16 Last Updated 05 2008 FIGURE 3 KEY PAD GUI CALL FLow Menu Call History Status Phone Book Instant Messages Direct IP Call Preference Config Factory Functions Reboot Exit Grandstream Networks Inc Y Received Calls Call History Dialed Calls Missed Calls Back Phone Book Any of prev
42. rty 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 Advanced Functionality Custom down loadable ring tones SRTP SIP over TLS pending multi language support and XML enabled adjustable positioning angles wall mountable AES encryption Grandstream Networks Inc GXP User Manual Page 9 of 39 Firmware 1 1 6 16 Last Updated 05 2008 dstream Innovative IP Voice amp Video Table 6 GXP Hardware Specifications LAN Interface Ethernet ports Two 2 10 100 Mbps Full Half Duplex Ethernet Switch with LAN and PC port with auto detection e mm wm ee eee wm e wm wm wm wm wm wm eee eee wm wm wm wm wm wm epee wm e wm wm wm wm wm ew wm mm ewww wm em vm wm vm em mm wm e em wm wm em wm wm wm www em em mm wm em www www ww www o ww wm gt gt wm wm em Graphic LCD Display es 00 00118 00 NEO ESESUES EE EEN Ie M EN GXP 280 GXP 1200 GXP 2000 GXP 2010 GXP 2020 128x32 128x32 130x64 240x120 320x160 pixel pixel pixel pixel pixel Expansion Module Support 0000000 SEXE GXP 1200 GXP 2000 Tin GXP 2020 No No Yes Pending Pending A mm om vm mm vm mm mm vm mm vm mm
43. rward Wait Time Enable Call Features Call Log ndstream Innovative IP Voice amp Video 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 Retry registration if the process failed Default is 20 seconds RFC 3261 SIP T1 timer Default is 1 second RFC 3261 SIP 12 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 th
44. s Inc GXP User Manual Page 32 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Headset RX gain dB Display Language ndstream Innovative IP Voice amp Video Increases the selected headset s 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 The user can only load one secondary language 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 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 Grandstream Networks Inc 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 Server s 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
45. s the NEW CALL soft key e 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 e Enter the phone number e 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 e Take Handset SPEAKER Headset off hook or press an available LINE key activates speakerphone the corresponding LED will be red e Press the SEND button or press the REDIAL soft key Grandstream Networks Inc GXP User Manual Page 16 of 39 Firmware 1 1 6 16 Last Updated 05 2008 gt Frandstream Innovative IP Voice amp Video 3 CALL RETURN To call the last phone number that called your phone When returning a call the phone will use the same SIP account as the call was made to Thus when returning a call made to the third SIP account the phone will use the third SIP account return the call i Hand free option 1 Press the CALL RETURN soft key ii Hand set option 1 Take the Handset off hook 2 Press the CALL RETURN soft key 4 USING THE CALL HISTORY To cal 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 c
46. sage button this always checks the primary account 2 check each line for stutter tone or 3 check missed calls using the menu Busy Lamp Field The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account When BLF is configured on one of the multi functional buttons the Speed Dial function will work when that line is not in use Call Pick Up is supported when user presses a flashing BLF key Grandstream Networks Inc GXP User Manual Page 20 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video CALL FEATURES The GXP supports traditional and advanced telephony features including caller ID caller ID w name call forward transfer park hold as well as intercom paging and BLF Table 11 GXP 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 50 Disable Call Waiting for all subsequent calls 51 Enable Call Waiting for all subsequent calls 70 Disable Call Waiting per Call 71 Enable Call Waiting per Call 72 Unconditional Call Forward Dial 72 for a dial tone Dial the forwarding number followed by 2 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 B
47. ssociated with each particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 4 digit numeric numbers i e P2 is associated with Admin Password in the ADVANCED SETTINGS page For a detailed parameter list please refer to the corresponding configuration template of the firmware Once the GXP boots up or re booted it will request a configuration file named cfgxxxxxxxxxxxx where XXXXXXXXXXXX is the MAC address of the device Le cf9000b820102ab 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 pre scheduled time intervals By defining different intervals in P193 for different devices a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time Grandstream Networks Inc GXP User Manual Page 38 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video Restore Factory Default Setting WARNING Restoring the Factory Default Setting will delete all configuration information of the phone Please backup or print all the settings before you restoring factory default settings Grandstream is not responsible for restoring lost parameters and cannot connect your dev
48. sume 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 wish 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 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 Grandstream Networks Inc GXP User Manual Page 19 of 39 Firmware 1 1 6 16 Last Updated 05 2008 gt Frandstream Innovative IP Voice amp Video 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 c
49. tions available for the end user are e Status Displays the network status account statuses software version and MAC address of the phone e 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 e Advanced Settings To set advanced network settings codec settings and XML configuration settings e Account X To configure each of the SIP accounts Grandstream Networks Inc GXP User Manual Page 26 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video Table 13 Device Configuration Status Hardware Revision MAC Address IP Address Product Model Part Number 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 GXP This field contains the product model information This field contains the product part number e Program This is the main software firmware release number always used to identify the software firmware system of the phone e 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 GXP can support four unique SIP profiles Indicates whether the PPPoE connection is enabled connected to a modem Table 14 Device Configuration
50. to choose whether to ring the phone Speaker when headset is connected Default is No By default LCD backlight will lit whenever there is a missed call GXP User Manual Page 28 of 39 Firmware 1 1 6 16 Last Updated 05 2008 ndstream Innovative IP Voice amp Video 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 15 Advanced Settings Admin Password G723 rate iLBC frame size ILBC payload type Silence Suppression Voice Frames per TX Administrator password Only the administrator can access the Advanced Settings and Account Settings page Password field is purposely blank for security reasons after clicking update and saved The maximum password length is 25 characters Encoding rate for G723 codec By default 6 3kbps rate is set ILBC packet frame size Default is 20ms For Asterisk PBX 30ms might be required Payload type for iLBC Default value is 97 The valid range is between 96 and 127 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 sin
51. user_manuals GUI GUI_GXP rar 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 i oc C dw Grandstream Networks Inc GXP User Manual Page 3 of 39 Firmware 1 1 6 16 Last Updated 05 2008 gt Frandstream Innovative IP Voice amp Video Welcome Your Grandstream GXP Series IP phone features a new sophisticated design and is very easy to use The GXP combines advanced feature functionality with the latest technology to offer excellent audio quality ease of use expandability and broad interoperability with 3 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 2010 Each model delivers superior audio quality using either a handset hands free speakerphone or headset and supports multi party conferencing multi languages dual color LEDs presence and BLF on most models Large easy to read backlit graphical displays except GXP 280 with multiple XML keys further enhance the user experience Some models GXP 2000 currently are expandable with one
52. usy Call Forward Dial 90 for a dial tone Dial the forwarding number followed by z 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 z 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 amp XML With its graphical LCD screen soft keys and XML support the GXP is a highly customizable phone XML is used to communicate between web servers and the GXP to dynamically update the phone s phonebook and idle screen logo and functions XML Phonebook The Grandstream GXP enables you to easily share and maintain a phonebook through the web The XML phonebook 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 Grandstream Networks Inc GXP User Manual Page 21 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Frandstream Innovative IP Voice amp Video Customizable Idle Screen and Soft buttons GXP also allows you to customize the idle screen using your own logo on
53. utomatic 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 amp 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 plan prefix off hook auto dial auto answer early dial and speed dial on some models Grandstream Networks Inc GXP User Manual Page 11 of 39 Firmware 1 1 6 16 Last Updated 05 2008 gt Frandstream Innovative IP Voice amp Video 1 e e e e e e rm e e e e em e e e e e e em e e ep e ee ee ee e e em ee em e e e e ee e ee ee e e em ee e ee e e e ee e ee e ew e e e e e ew e e o e o e e e e em e ee e e e e ee em e e ee ee ee ee ee ee ee 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 eee o e n e vm ew
54. ver 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 interval When the user downloads the XML phone the manually entered or edited entries will not be deleted unless this option is selected to Yes Enable XML Idle Screen download via TFTP or HTTP Define XML server path Enter server path for XML application This option applies to GXP 2020 only To configure a User ID extension to dial automatically when the phone is taken offhook 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 e product model version on boot up INFO level e NAT related info INFO level e sent or received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level e SLIC chip exception WARNING and ERROR levels e memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS LOG de
55. vice MAC addressj ferror 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 GXP User Manual Firmware 1 1 6 16 Page 31 of 39 Last Updated 05 2008 NTP server Distinctive Ring Tone System Ring Tone Call Progress Tones Intercom User ID Disable Call Waiting Disable Call Waiting Tone Use Quick IP Call Mode Lock keypad update Headset Port Type ndstream Innovative IP Voice amp Video 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 System ring tone Default is North American standard Adjust system ring tone frequencies and cadences based on local telecom standard Using these settings users can configure ring or tone frequencies based on parameters from local telecom By default they are set to North American standard Frequencies should be configured with known values to avoid uncomfortable high pitch sounds Syntax f1 val f2 val c on1 off1 on2 off2 on3 off3 Frequencies are in Hz and cadence on and off are in 10ms
56. will ring with system ring tone e 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 Whether to use sip anonymous Qanonymous invalid gt in the From Header or P Asserted Identity header Default is NO If set to YES anonymous call will be rejected Default is No If set to Yes GXP 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 If the Call Info header contains answer after 0 the call be answered automatically so called paging mode GXP User Manual Grandstream Networks Inc Page 35 of 39 Firmware 1 1 6 16 Last Updated 05 2008 Turn off speaker on remote disconnect Check SIP User ID for incoming INVITE Refer To Use Target Contact Disable Multiple Media Attribute in SDP Preferred Vocoder SRTP Mode eventlist BLF URI Special Feature ndstream Innovative IP Voice amp Video When BYE is received the phone will turn off its soeaker automatically Check the SIP User ID in Request URI If they don t match the call will be rejected Default is NO If set to YES then for Attended Transfer the Refer To header uses the transferred target s Contact header i
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