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GXP User Manual - RES Communications.com

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2. LDAP Directory Search Configuration View Directory Call History Dow nload Directory Select Filter Filter Value Back Search Configuration Back Status Instant Message Do Not Disturb Clear All Phone Book Back Enable DND Disable DND LDAP Directory Saman Back Ring Tone Default Ring Ring1 Ring2 Ring 3 Back Do Not Disturb Ring Tone LCD Contrast LCD Brightness l Dow nload SCR XML Directia Cai Erase Custom SCR Gateway Display Language DNS Server Preference Back LCD Brightness i 5 erver Instant Network Message IP Setting IP Net Mask Config SIP Account SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio Save Network Factory SIP Functions Upgrade Factory Reset Layer 2 QoS Reboot Back Dis play Language English Chinese French Spanish German Exit Factory Function Audio Loopback Diagnostic Mode Upgrade Italian Secondary Language Language File Postfix Back Back Firmware Server Config Server Upgrade Via Diagnostic Mode Layer 2 QoS Keypad LED Diagnostic 802 1Q VLAN Tag Priority value Reset Vlan Config Back GXP1450 User Manual Page 20 of 38 Firmware 1 0 1 26 Last Updated 122010 LE 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
3. The GXP1450 phone supports shared call appearance by Broadsoft standard This feature allows members of the SCA group to shared SIP lines and provides status monitoring idle active progressing hold of the shared line When there is an incoming call designated for the SCA group all of the members of the group will be notified of an incoming call and will be able to answer the call from the phone with the SCA extension registered All the users that belong to the same SCA group will be notified by visual indicator when a user seizes the line and places an outgoing call and all the users of this group will not be able to seize the line until the line goes back to an idle state or when the call is placed on hold With the exception of when multiple call appearances are enabled on the server side GXP1450 User Manual Page 16 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video In the middle of the conversation there are two types of hold Public Hold and Private Hold When a member of the group places the call on public hold the other users of the SCA group will be notified of this by the red flashing button and they will be able to resume the call from their phone by pressing the line button However if this call is placed on private hold no other member of the SCA group will be able to resume that call To enable shared call appearance the user would need to register the shared line account on one of the
4. 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 will ring with system ring tone e Three 3 Customer Ring Tones when selected incoming calls from designated account will play selected ring tone Defines how long ring will ring when receiving a call Default is 60 seconds If this parameter is set to Yes the From header in outgoing INVITE message will be set to anonymous essentially blocking the Caller ID from displaying Default is NO If set to YES 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 GXP1450 User Manual Firmware 1 0 1 26 Page 33 of 38 Last Updated 12 2010 Refer To Use Target Contact Transfer on Conference Hangup Preferred Vocoder SRTP Mode Symmetric RTP Silence Suppression Voice Frames per TX No Key Entry Timeout Gin Innova tive IP Voice amp
5. 2 The Line Status Indicator will show LINEx SPEAKING or LINEx MUTE to indicate whether the microphone is muted Call Transfer GXP1450 supports both Blind and Attended or supervised transfer 1 Blind Transfer Press TRANSFER button then dial the number and press the SEND button to complete transfer of active call 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 key then the LINE GXP1450 User Manual Page 15 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video button of the waiting line to transfer the call Hang up the phone call after Transfer Successful is displayed in the screen 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 3 Way Conferencing GXP can host conference calls and supports up to 3 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 previous two steps for all other parties that would like to join the conference This can be done at any time However if a new call comes in the other calls will be placed on hold and the h
6. 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 GXP1450 User Manual Page 5 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video Product Overview Table 3 GXP1450 Product Models GXP1450 is an executive SIP phone It features e Two lines e Three XML programmable soft keys GXP1450 Table 4 GXP1450 Feature Guide Features GXP1450 LCD Display 1 180x60 pixel Number of Lies ss L Programmable Hard Keys Q 0 TTT Programmable Soft Keys 13 Extension Module gt NA Table 5 GXP1450 Key Features in a Glance Traditional voice features including caller ID call waiting hold transfer forward block autodial off hook dial Multi line support with dual color LED multi party conferencing line extension interface large back lit graphic LCD 5 navigation keys dedicated buttons for hold send speakerphone headset transfer 3 way conference mute message i Do not disturb phone book intercom paging Features f Benefits Open Standards SIP 2 0 TCP IP UDP RTP RTCP HTTP HTTPS ARP RARP ICMP DNS A
7. ndstream Innovative IP Voice amp Video Grandstream Networks Inc GXP1450 SIP Enterprise Phone Grandstream Networks Inc GXP1450 User Manual Page 1 of 38 Firmware 1 0 1 26 Last Updated 12 2010 TABLE OF CONTENTS GXP1450 USER MANUAL WEERCOSNIE cerc INSTA CLA HON ssr FOUI MENT La L eT CONNECTING YOUR PHONE eee anne SAFETY COMPLIANCES a PP Be PRODUCTOVERVIE ss USING THE GXP1450 SIP ENTERPRISE PHONE eseeseveeveeveeveesersevnevnenees GETTING FAMILIAR WITH THE LCD cece c cece ecceceeeceeseeeeeeeeeseeseeeeeees MAKING PHONE CALLS ccccsssscnssscacacsssdesessanamsnrnaantarsananraanenaanianaananaannbeseonent ANSWERING PHONE CALLS xe PHONE FUNCTIONS DURING A PHONE CALL xe CALLI ATURE TT CUSTOMIZED LCD SCREEN r X NL CONFIGURA TION GOME uk a CONFIGURATION VIA KEYPAD CONFIGURATION VIA WEB BROWSER ee SAVING THE CONFIGURATION CHANGES REBOOTING THE PHONE REMOTE LY ccccccceecescceececceceeeseeesecseuseeseeeeueenss SOFTWARE UPGRADE amp CUSTOMIZATION cccccccccccccccscceees FIRMW ARE UPGRADE THROUGH TFTP HTTP ccc cece ccc ee ccccceenccceeecs CONFIGURATION FILE DOWNLOAD cccccceccccecccccecucccceccccceeeccceesueceeeuecs RES TORE FACTORY DEFA ULT SETTING x eeeeveeveesevnevnevneveeveenevnevnenneneee TABLE OF FIGURES GXP1450 USER MANUAL Table 10 GXP1450 Keypad ButtOns eee Table 10 GXP1450 Keypad BUuTIOnS ese Fig
8. Compatible record and SRV DHCP both client and server PPPoE TFTP NTP Telnet and SIP over TLS 802 1x TR 069 Superb Audio Advanced Digital Signal Processing DSP Silence suppression VAD CNG Quality AGC Network Interfaces i Dual 10 100mbps Ethernet ports Advanced Custom down loadable ring tones SRTP SIP over TLS multi language support Functionality and XML enabled adjustable positioning angles wall mountable AES encryption s m m m m m m m m m m m m m m m m m d m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m m EEE GXP1450 User Manual Page 6 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video Table 6 GXP1450 Hardware Specifications GXP1450 LAN Interface Two 2 10 100 Mbps Full Half Duplex Ethernet Switch with LAN and PC port with Call Appearance Ethernetports ee auto detection ee Graphic LCD 180x60 pixel Display Expansion Module No Support L Headset Jack RJ9 LED Powerover as Built in auto sensing Cisco and IEEE 802 3af standard t S Ethernet L Universal Input 100 240VAC 50 60 Hz Switching Power Adaptor 1 Output 5VDC 800mA ULcertified Dimension gt NN 186mm W x 210mm L x8imm D Weight poe OSKG Temperature 82 MONFO AOC oaaao Humidity 10 90 non condensing 122222 Compliance FCC CE C Tick Table 7 GXP1450 Technical Specifications Multiple direct lines w
9. accounts on the phone In addition they would need to navigate to Settings gt Basic Settings on the web Ul and set the line to Shared Line with the corresponding account If the user requires more shared call appearances the user can configure multiple line buttons to be shared line buttons associated with the account CALL FEATURES The GXP1450 supports traditional and advanced telephony features including caller ID caller ID wname call forward transfer park nold 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 70 Disable Call Waiting per Call 71 Enable Call Waiting per Call 72 Unconditional Call Forward Dial 72 for a dial tone Dial the forwarding number followed by Wait for dial tone LCD will display Call FWD Activated 73 Cancel Unconditional Call Forward dial 73 and get the dial tone then hang up LCD will display Call FWD Activated 90 Busy Call Forward Dial 90 for a dial tone Dial the forwarding number followed by Wait for a dial tone Hang up 91 Cancel Busy Call Forward dial 91 Wait for dial tone Hang up 92 Delayed Call Forward Dial 92 for a dial tone Dial the forwarding number followed by Wait for a dial to
10. prevents dialing any number started with 1900 e lt 1617 gt 2 9 xxxxxx allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically e 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length e 011 2 9 x allows international calls starting with 011 e 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Note In some cases where the user wishes to dial strings such as 123 to activate voice mail or other applications provided by their service provider the should be predefined inside the dial plan feature An example dial plan will be x which allows the user to dial followed by any length of numbers Default is This prefix is prepended when answering call with BLF key Time waited before the call is forward to a number or VM Default is 20 seconds Default is Yes If set to No Call transfer Call Forwarding amp Do Not Disturb are supported locally provided ITSP support those features In addition Forward Al softkey will be hidden if call feature code is disabled for Account 1 User can choose to disable Call Log and what kind of calls to log GXP1450 User Manual Page 32 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Session Expiration Min SE Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Enable 100rel Account Ring T
11. LE 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 nece ssary 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 SN e a a To make a quick IP call please see next section For example If the target IP address is 192 168 1 60 and the port is 5062 e g 192 168 1 60 5062 input the following 192 168 1 60 5062 The key represent the dot The key represent colon Press OK to dial out Quick IP Call Mode The GXP1450 also supports Quick IP call mode This enables the phone to make direct IP calls using only the
12. LE Innovative IP Voice amp Video Restore Factory Default Setting WARNING Restoring the Factory Default Setting wil delete all configuration information of the phone Please backup or print all the settings before you restoring factory default settings We are not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider INSTRUCTIONS FOR RESTORATION Step 1 Press OK button to bring up the keypad configuration menu select Config press OK to enter submenu select Factory Reset Please refer to Table 5 1 of keypad flow chart Step 2 Enter the MAC address printed on the bottom of the sticker Please use the following mapping 0 9 0 9 22 press the 2 key twice A will show on the LCD 222 2222 33 press the 3 key twice D will show on the LCD 333 3333 pes or Example if the MAC address is 000682006395 it should be key in as 000 2228200 333395 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 GXP1450 User Manual Page 38 of 38 Firmware 1 0 1 26 Last Updated 12 2010
13. Time This field shows the current time on the phone system Registered Indicates whether accounts are registered to the related SIP server s GXP can support four unique SIP profiles PPPoE Link Up Indicates whether the PPPoE connection is enabled connected to a modem Table 14 Device Configuration Settings Basic Settings End User This contains the password to access the Web Configuration Menu This field is Password case sensitive wth a maximum length of 25 characters IP Address The GXP operates in two modes 1 DHCP mode all the field values for the Static IP mode are not used even though they are still saved in the Flash memory The GXP acquires its IP address from the first DHCP server it discovers on its LAN The DHCP option is reserved for NAT router mode To use 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 802 1x Mode This option allows the user to enable disable 802 1x mode on the phone The default value is disabled To enable 802 1x mode this field should be set to EAP MD5 Once enabled the user would be required to enter the following information below to be authenticated on the network e Identity e MD5 Password GXP1450 User Manual Pa
14. Vi Default is NO If set to YES then for Attended Transfer the Refer To header uses the transferred target s Contact header information Defines whether or not the call is transferred to the other party if the initiator of the conference hangs up Default setting is set to No GXP supports up to 7 different Vocoder types including G 711 a u also known as PCMU PCMA G 723 1 G 729A B G 726 32 iLBC G 722 wide band Configure Vocoders in a preference list that is included with the same preference order in SDP message Enter the first Vocoder in this list by 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 Selects whether or not symmetric RTP is supported This controls the silence suppression VAD feature of the audio codec G 723 and G 729 If set to Yes when silence is detected a small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled This field contains the number of voice frames to be transmitted in a single Ethernet packet be advised the IS limit is based on the maximum size of Ethernet packet is 1500 byte or 120kbps When setting this value be aware of the requested packet time ptime used in SDP message is a result of configuring this parameter This parameter is
15. and flashed into the phone This setting is useful for ITSPs End user should keep it blank Default is blank If configured GXP1450 will request the config file with the prefix postfix and only the file with the matching encrypted prefix will be downloaded and flashed into the phone This setting is useful for ITSPs End user should keep it blank GXP1450 User Manual Page 25 of 38 Firmware 1 0 1 26 Last Updated 122010 Allow DHCP Option 43 and Option 66 to override server Automatic Upgrade Authenticate Conf File TR 069 Username TR 069 Password ACS URL Phonebook XML Download Phonebook XML Server Path Phonebook Download Interval Remove Manually edited entries on Downloads LDAP Directory Idle Screen XML Download XML Application Softkey Label Offhook Auto Dial Syslog Server GE seven Innova tive IP Voice amp Vi Default is Yes This allows device gets provisioned automatically This function is used by ITSP End user should NOT touch these parameters Default is No Choose Yes to enable automatic HTTP upgrade and provisioning In Check for upgrade every field enter the number of minutes to check the HTTP server for firmware upgrade or configuration changes When set to No the phone will only perform HI TP upgrade and configuration check once at boot up Default is No If set to Yes configuration file would be authenticated before acceptance End user should u
16. 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 We recommend using the default settings provided We do not recommend adjusting these parameters if you are an average user Incorrect settings will affect the voice quality Default is 4 seconds GXP1450 User Manual Page 34 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Use as Dial Key G723 Rate iLBC Frame Size iLBC Paylo
17. field allows administrator to configure a backup SIP Server GXP1450 User Manual Page 29 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name DNS Mode Primary IP SIP Registration Un register on Reboot Register Expiration Local SIP Port SIP Registration Failure Retry Wait Time SIP T1 Timeout SIP T2 Interval SIP Transport Remove OBP from Route Validate Incoming Messages GE seven Innova tive IP Voice amp Vi 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 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 The default is set to A Record If user wishes to locate the server by DNS SRV the user may select SRV or NATPTR SRV When Use Configured IP option is selected if SIP server is configured as domain name phone will not send DNS query but use Primary IP or Seco
18. the speakerphone is on DND Icon ON when the do not disturb is activated Activate by pressing MUTE DEL button once Calls Forwarded Icon INDICATES calls are forwarded Follow call forwarding procedures Handset Speakerphone and Ring Volume Icon Each icon appears next tothe 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 GXP1450 User Manual Page 10 of 38 Firmware 1 0 1 26 Last Updated 12 2010 andstream Innovative IP Voice amp Video TABLE 10 GXP1450 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 akey entry HIE Also used to REJECT incoming call HOLD Place ACTIVE call on hold MSG Enter to retrieve voice mails or other messages I Enable Disable hands 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 Enter to retrieve voice mails or other messages Enter Keypad Configuration MENU mode when phone is in IDLE mode WEN Use as ENTER key when in Keypad Configuration 0 9 Standard phone keypad press key to sen
19. u law G 726 32 G 722 wide band and iLBC codecs In band and out of band DTMF in audio RFC2833 SIP INFO Silence Suppression VAD voice activity detection CNG comfort noise generation ANG automatic gain control Acoustic Echo Cancellation AEC with Acoustic Gain Control AGC for speakerphone mode Support side tone Adaptive jitter buffer control patent pending and packet delay amp loss concealment HD audio handset with HD wideband audio codecs for excellent double talk performance Telephony Features Network and Provisioning Firmware Upgrades Security tr rc ctr c ccc 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 3 way conference dial plan prefix dial plan support off hook auto dial auto answer early dial and speed dial eee 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 Layer3 TOS 1 112 120 Support firmware upgrade via TFTP or HTTP Support for Authenticating configuratio
20. 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 Allows the user to select the following options for firmware upgrade e Always Check for New Firmware e Check New Firmware only when F W pre suffix changes e Always Skip the Firmware Check Firmware upgrade may take up to 10 minutes depending on network environment Do not interrupt the firmware upgrading process Note Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade Please DO NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device The password used for encrypting the XML configuration file using OpenSSL This is required for the phone to decrypt the encrypted XML configuration file The user name for the HI TP HT TPS server The password for the HTTP HTTPS server This field allows the user to choose the firmware upgrade method TFTP HI TP or HTTPS Defines the server path for the firmware server It can be different from the Configuration server which is used for provisioning Defines the server path for provisioning it can be different from the firmware server Default is blank If configured GXP1450 will request the firmware file with the prefix postfix and only the firmware with the matching encrypted prefix will be downloaded
21. 5 This indicates a time zone with 6 hours offset with 1 hour ahead which is U S central time If it is positive if the local time zone is west of the Prime Meridian A K A International or Greenwich Meridian and negative if it is east M3 2 0 M11 1 0 The ist number indicates Month 1 2 3 12 for Jan Feb Dec The 2nd number indicates the nth iteration of the weekday 1st Sunday 3 Tuesday The 3rd number indicates weekday 0 1 2 6 for Sun Mon Tues Sat Therefore this example is the DST which starts from the second Sunday of March to the 1st Sunday of November Settings to customize the dispay of weather via e City Code Enter city zip code e Update Interval Refresh time in minutes e Degree Unit Select either Fahrenheit or Celsius This is displayed when pressing the SwitchSCR soft key once Settings to customize the display order of major stock indices This is displayed when pressing the SwitchSCR soft key twice Settings to customize the display order of currency updates of foreign currency into US dollars This is displayed when pressing the Switch SCR soft key three times Set the LCD brightness level Range from 0 to 8 where 0 means off and 8 means the brightest Set LCD contrast Range from 0 to 20 GXP1450 User Manual Page 23 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Time Display Format Disable in call DIMF display Disable Missed Call Backlight H
22. CALLS are shown here too STATUS Shows the status of the phone using icons as shown in the next table BAR LINE Displays the name of the account that is in use Select another account by pressing the LINE STATUS SELECTOR BUTTONS INDICATOR 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 pa Missed Calls option shows a list of the missed calls e SWITCHSCR Press this button to toggle between XML Applications such as weather stocks and currency 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 Table 9 LCD Icons lon gt 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 GXP1450 User Manual Firmware 1 0 1 26 Page 9 of 38 Last Updated 12 2010 ndstream 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
23. EADSET Key Mode Headset Port Type Headset TX Gain dB Headset RX Gain dB LE Innova tive IP Voice amp Video LCD time display in 12 hour or 24 hour format Default is No This field is used to hide the keypad input during a call Default is No By default LCD backlight will lit whenever there is a missed call Select either Default mode or Toggle Headset Speaker In Default Mode only the speakerphone will ring for an incoming call User can use the headset key to pick up speak and hang up calls through headset The headset icon will appear on the LCD when a call is in progress If user wishes to ring the headset Toggle Headset Speaker option shall be checked User will also need to press the HEADSET key while phone is idle The headset icon will appear on the idle LCD screen Select either RJ9 headset ports to be adjusted Increases the selected headset s 2 5mm or RJ22 TX gain by or 6dB Default is 0dB Increases the selected headset s 2 5mm or RJ22 RX gain by or 6dB Default is 0dB Advanced User configuration includes not only the end user configuration but also advanced configur ation such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous configuration Table 15 Device Configuration Settings Advanced Settings Admin Password Layer 3 QoS Layer 2 QoS Local RTP port Use Random Port Administrator password Only the admin
24. L either in FQDN or IP address format The Config Server Path can be the same or different from the Firmware Server Path A configuration parameter is associated with each particular field in the web configuration page A parameter consists of a Capital letter P and 2 to 4 digit numeric numbers i e P2 is associated with Admin Password in the ADVANCED SETTINGS page For a detailed parameter list please refer to the corresponding configuration template of the firmware Once the GXP1450 boots up or re booted it will request a configuration file named cCfgxXXXXXXXXXXX followed by a request for configuration XML file named cfgxxxxxxxxxxxx xml where XxXXXXXXXXXXX is the MAC address of the device i e cf9000b820102ab The configuration file name should be in lower cases For more details on XML provisioning please refer to http www grandstream com support general gs_ xml provisioning pdf 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 GXP1450 User Manual Page 37 of 38 Firmware 1 0 1 26 Last Updated 12 2010
25. NTERFACE EXAMPLES GXP1450 USER MANUAL http www grandstream com support gxp_series general documents gxp_qui Zip 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 SEE ae S GXP1450 User Manual Page 3 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video Welcome GXP1450 is a next generation enterprise grade IP phone that features 2 lines with 2 SIP accounts a 180x60 backlit graphical LCD 3 XML programmable context sensitive soft keys dual network ports with integrated PoE and 3 way conference The GXP1450 delivers superior HD audio quality rich and leading edge telephony features personalized information and customizable application service automated provisioning for easy deployment advanced security protection for privacy and broad interoperability with most 3rd party SIP devices and leading SIP NGN IMS platforms It is a perfect choice for enterprise users looking for a high quality feature rich IP phone with affordable cost Caution Changes or modifications to this product not expressly approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty Warning Please do not use a different p
26. ad Type eventlist BLF URI Special Feature LE Innovative IP Voice amp Video This parameter allows users to configure the T key as the Send or Dial key If set to Yes the key will immediately send the call In this case this key is essentially equivalent to the Re Dial key If set to No the key is included as part of the dial string 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 If a server supports this feature user needs to configure an eventlist BLF URI on the service side i e BLF1006 myserver com On the GXP under Account page fill in the eventlist BLF field with the URI without the domain i e BLF1006 Under Basic Settings please select eventlist BLF choose account number 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 We recommend to reboot or power cycle the IP phone after saving changes REBOOTING THE PHONE REMOTELY Press the Reboot button at the bottom of the configuration
27. arly Dial Dial Plan Prefix LE Innovative IP Voice amp Video Selects whether or not SIP Instance ID is supported This parameter activates the NAT traversal mechanism If activated by choosing Yes anda 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 DIMF digit There are 3 supported modes in audio which means DT MF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO Sends DTMF using RFC2833 The default is 101 Default is No Use only if proxy suppor
28. cal TFTP Upgrade GXP1450 User Manual Page 36 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video 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 httod apache org or use Microsoft IIS web server NOTE e When GXP phone boots up it will send TFTP or HI TP request to download configuration file cfg000b82xxxxxx where 000b82xxxxxx is the MAC address of the GXP phone This file is for provisioning purpose For normal TFTP or HTTP firmware upgrades the following error messages in a TFTP or HTTP server log can be ignored TFTP Error from IP ADRESS requesting cfg000b82023a04 File does not exist Configuration File Download CONFIGURATION FILE DOWNLOAD The GXP1450 can be configured via Web Interface as well as via Configuration File binary or XML through TFTP or HTTP HTTPS The Config Server Path is the TFTP or HTTP server path for the configuration file It needs to be set to a valid UR
29. dcall press key to for IVR 3 functions DND DO NOT DISTURB key Press DND to turn Do not disturb function on or off HEADSET Press HEADSET key to answer hang up phone calls while using headset It A also allows user to toggle between headset and speaker Brings phonebook on screen GXP1450 User Manual Page 11 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video MAKING PHONE CALLS Handset Speakerphone and Headset Mode The GXP1450 allows you to make phone calls via handset speakerphone or headset mode During the active calls the user can switch between the handset and the speaker by pressing the speaker key For headsets to operate the user must plug the headset to an RJ9 port on the phone which allows the user to pick up speak or hang up calls Multiple SIP Accounts and Lines GXP1450 can support up to two independent SIP accounts Each account is capable of independent SIP server user and NAT settings Each of the line buttons is virtually mapped to an individual SIP account The name of each account is conveniently printed next to its corresponding button In off hook state select an idle line and the name of the account as configured in the web interface is displayed on the LCD anda dial tone is heard For example Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as VoIP 1 VoIP 2 respectively and ensure that they are active and registered When LINE1 is press
30. e aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask XX or HX are also valid so leading 0 is not required but OK See Quick IP Call Mode for details Default is No If set to Yes conference will be disabled Default is No If set to Yes Muti Purpose keys can be sent as DTMF Default is No If set to Yes the DND button on keypad will be disabled Default is No If set to Yes transfer will be disabled Configures the access control of configurations via the phone keypad menu There are three modes e Unrestricted e Basic Settings Only e Constraint Mode GXP1450 User Manual Firmware 1 0 1 26 Page 28 of 38 Last Updated 12 2010 Display Language LE Innova tive IP Voice amp Video Allows user to choose preferred display language in web Ul and key pad Ul Currently the phone supports these languages English Simplified Chinese Traditional Chinese Korean Japanese Italian Spanish French and German Portuguese Russian Croatian Note The Automatic setting in language refers to Grandstream s IP2Location client which when connected to Internet would attempt to lookup a database driven by Grandstream with the IP address for its geographical location Language file postfix allows the language file to have different postfixes so the phone can request a particular file It will append an underscore _ plus the string in the language file postfix The default language f
31. e 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 GXP1450 User Manual Page 12 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE 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 li Hand set option 1 Take the Handset off hook 2 Press the CALL RETURN soft key 4 USING THE CALL HISTORY To call a phone number in the phone s history When using the call history the phone will use the same SIP account as was used for the last call call attempt Thus when returning a call made to the third SIP account the phone will use the third SIP account return the call 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
32. ed 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 VolP 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 aphone call e Take Handset SPEAKER Headset off hook or press an available LINE key activates speakerphone or press 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 phon
33. g or tone frequencies based on parameters from local telecom By default they are set to North American standard Frequencies should be configured with known values to avoid uncomfortable high pitch sounds Syntax f1 val f2 val c on1 off1 on2 off2 on3 off3 Frequencies are in Hz and cadence on and off are in 10ms ON is the period of ringing On time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Up to three cadences are supported GXP1450 User Manual Page 27 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Disable Call Waiting Disable Call Waiting Tone Disable Direct IP Calls Use Quick IP Call Mode Disable Conference Enable MPK Sending DTMF Disable DND Button Disable Transfer Configuration via Keypad Menu LE Innovative IP Voice amp Video 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 Default is No If set to Yes direct IP calls 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 wher
34. ge 22 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Line Keys x Multi Purpose Key X Time Zone Self Defined Time Zone Weather Update Stock Update Currency Update LCD Backlight Brightness LCD Contrast GE seven Innova tive IP Voice amp Vi This allows the user to configure the account mapped to each line key as well as enabling SCA Shared Call Appearance for the line Options available for Key Mode are 1 Line 2 Shared Line 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 This parameter allows the users to define their own time zone The syntax is std offset dst offset start time end Aime Default is set to MTZ 6M DT 5 M3 2 0 M11 1 0 MTZ 6M DT
35. ile name is gxp Ipf If the field Language File postfix has NL string in it then the phone will request gxp NL Ipf instead of gxp Ipf 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 Note This is similar to updating firmware in your local network environment Please refer to htto www grandstream com fagsfirmware himl 4 for details ke 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 Ipf to the firmware server directory using your local TFTP or HTTP server 5 Access the advanced settings of the Web GUI set Display Language to Secondary Language 6 Update and reboot the phone GXP1450 has up to two 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 Secondary SIP Server 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 This
36. ings have to be changed press the menu option needed Definitions This section will describe the options in the Web configuration user interface As mentioned a used can log in aS an administrator or end user Functions available for the end user are 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 GXP1450 User Manual Page 21 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Gin Innova tive IP Voice amp Vi Table 13 Device Configuration Status MAC Address The device ID in HEXADECIMAL format IP Address This field shows IP address of GXP Product Model This field contains the product model information Part Number This field contains the product part number Software Version e Program This is the main firmware release number which is always used for identifying the software or firmware system of the phone e Boot Booting code version number e Core Core code version number e Base Base code version number e DSP DSP code version number e Aux Aux code version number System Up Time This field shows system up time since the last reboot System
37. istrator 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 This field defines the layer 3 QoS parameter It is the value used for IP Precedence or Diff Serv or MPLS Default value is 48 This contains the value used for layer 2 VLAN tag Default setting is blank 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 RICP The default value is 5004 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple GXPs are behind the same NAT Default is No GXP1450 User Manual Page 24 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Keep alive interval Use NAT IP STUN Server Firmware Upgrade and Provisioning XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Upgrade Via Firmware Server Path Config Server Path Firmware File Prefix Postfix Config File Prefix Postfix GE seven Innova tive IP Voice amp Vi 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
38. ith independent SIP accounts programmable speed dial keys XML programmable soft keys Protocol Support SIP 2 0 TCP UDP IP PPPoE RIP RTCP SRTP by SDES HTTP Support ARP RARP ICMP DNS DHCP NTP TFTP SIMPLE PRESENCE protocols TR 069 802 1x Support multiple SIP accounts and up to 11 media channels concurrently Support SIP PUBLISH method RFC 3903 SIP Presence package RFC 3856 3863 for use of MFKs SIP Dialog package RFC 4235 Support for SIP MESSAGE method RFC 3428 Feature Keys BIR ER ao ma m m m m m m m a m n mn n mn m mn ms v mn mn m m m m dda m mn m m m m v mn m m m m m m m d m m m m m m m m m mm m mma mm m m ma m m m m ma Tm m ma mm ma m m m xx m m m m m mm m m m m m m m m m m m m m m m m m m m m m m GXP1450 User Manual Page 7 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video PHONEBOOK Ves 1 MG Yes MENU Yes NAVIGATION 4 Yes G 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 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
39. 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 xis 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 23 follow by SEND or 192 168 0 2 calling 192 168 0 123 dial 123 follow by SEND or 192 168 0 2 dial 3 and 03 and 003 results in the same call call 192 168 0 3 NOTE If you have a SIP Server configured a Direct IP IP still works If you are using STUN the Direct IP IP call will also use STUN Configure the Use Random Port to NO when completing Direct IP calls GXP1450 User Manual Page 14 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video ANSWERING PHONE CALLS Receiving Calls 1 Incoming single call Phone rings with selected ring tone The corre
40. menu to reboot the phone remotely The web browser will then display a message window to confirm that reboot is underway Wait 30 seconds to log in again GXP1450 User Manual Page 35 of 38 Firmware 1 0 1 26 Last Updated 12 2010 LE Innovative IP Voice amp Video Software Upgrade amp Customization Software or firmware upgrades are completed via either TFTP or HTTP The corresponding configur ation 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 e xamples of some valid URLs e firmware mycompany com 6688 Grandstream 1 2 3 5 72 172 83 110 There are two ways to set up the Upgrade Server to upgrade firmware via Key Pad Menu and Web Configuration Interface Key Pad Menu To configure the Upgrade Server via Key Pad Menu options select Config from the Main Menu then select Upgrade Under this sub Menu user can edit Upgrade Server in either an IP address format or FQDN format Choose Save and use TFTP or Save and use HTTP to select upgrade method Select Reboot from the Main Menu to reboot the phone Web Configuration Interface To configure the Upgrade Server via the Web configuration interface open the web browser Enter the GXP IP address Enter the admin
41. n file before accepting changes User specific URL for configuration file and firmware files Mass provisioning using TR 069 or encrypted XML configuration file 2222 Message waiting indication support DNS SRV Look up and SIP Server Fail Over Support customizable idle screen via downloading XML by HTTP TFTP User and administrator level passwords MD5 and MD5 sessbased authentication AES based secure configuration file SRTP TLS 802 1x media access control GXP1450 User Manual Page 8 of 38 Firmware 1 0 1 26 Last Updated 12 2010 andstream Using the GXP1450 SIP Enterprise Phone GETTING FAMILIAR WITH THE LCD GXP1450 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 Selects the phone line printed on its right hand side SELECTORS SIP PHONE Displays the available phone lines Choose a phone line by pressing the corresponding line LINES selector on the left hand side DATE AND Displays the current date and time Can be synchronized with Internet time servers TIME LOGO Displays company logo This logo can be customized For more information on customizing the logo please check page 24 NETWORK Shows the status of the phone and network It will indicate whether the network is down starting STATUS or is running show IP number Other messages such as DO NOT DISTURB or MISSED
42. ndary IP to send sip message if at least one of them are not empty This option applies only if Use Configured IP is selected the phone will send DNS query to the Primary IP Insert IP address here 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 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 T2 timer Default is 0 5 seconds Choose SIP Transport between UDP and TCP Default is UDP The SIP Extension notifies the SIP server that it is behind a NAT firewall This configuration selects whether or not the incoming messages should be validated GXP1450 User Manual Page 30 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Support SIP Instance ID NAT Traversal STUN SUBSCRIBE for MWI PUBLISH for Presence Proxy Require Voice Mail UserID Send DTMF DTMF Payload Type E
43. ne 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 GXP1450 User Manual Page 17 of 38 Firmware 1 0 1 26 Last Updated 12 2010 C istan Innovative IP Voice amp Video GXP1450 Enterprise IP phone 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 howto create a downloadable XML phonebook creating a custom idle screen and or reprogramming the soft keys on GXP1450 please visit our website at htto www grandstream com support gxp_series general gxp support html Configuration Guide The GXP1450 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 f 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 missed and transfer
44. nts 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 device MAC address lerror code error message For example May 19 02 40 38 192 168 1 14 GS LOG 00 0b 82 00 a1 be 000 Ethernet link is up This parameter defines the URI or IP address of the NTP Network Time Protocol serve It is used to display the current date time This defines the SSL certificate needed to access certain websites This defines the SSL Private key This defines the SSL private key password 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 rin
45. one Ring Timeout Send Anonymous Anonymous Call Rejection Auto Answer Allow Auto Answer by Call Info GE seven Innova tive IP Voice amp Vi The SIP Session Timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the session is terminated Session Expiration is the time in seconds at which the session is considered timed out provided no successful session refresh transaction occurs beforehand The default value is 180 seconds Defines the minimum session expiration in seconds Default is 90 seconds If set to Yes the phone will use session timer when it makes outbound calls if remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request If set to Yes the phone will use session timer even if the remote party does not support this feature If set to No the session timer is enabled only when the remote party supports this feature To turn off Session Timer select No for Caller Request Timer Callee Request Timer and Force Timer As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy server as the refresher As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher
46. or primary account e 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 e 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 e Layer 2 QoS Configure VLAN Tags Press to return the main menu Factory Functions Press Menu to display the factory function items including e 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 e Diagnostic Mode All LEDs will light up Press any key on the keypad to display the button name inthe LCD Lift and put back the handset or press Menu button to exit the diagnostic mode 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 GXP1450 User Manual Page 19 of 38 Firmware 1 0 1 26 Last Updated 12 2010 andstream Innovative IP Voice amp Video FIGURE 2 KEYPAD GUI FLow Call History Any of previous menus Answered Calls Dialed Calls Back Missed Calls Clear All Transferred Calls MENU Back Phone Book New Entry Name Number Acct New Entry Download Phonebook XML Back Confirm Add Cancel amp Return
47. ost will have to individually re join the held lines back into the conference by repeating the previous two steps again 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 Also this is not applicable when the feature Transfer on call hangup is turned on 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 message button this always checks the primary account 2 check each line for stutter tone or 3 check missed calls using the menu Shared Call Appearance SCA
48. ower adaptor wth 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 from http www grandstream com Support qxp series general documents gxp dqui zip for your reference e This document is subject to change without notice 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 is not permitted GXP1450 User Manual Page 4 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Gs Innovative IP Voice amp Video Installation EQUIPMENT PACKAGING Table 1 Equipment Packaging TSS XP Yes es Power Adaptor Ethernet Cable Base Stand Handset Y Phone Cord Y CONNECTING YOUR PHONE The connectors of the GXP1450 are located on the bottom of the device Table 2 GXP Connectors PC 10 100Mbps RJ 45 ports for PC downlink connection LAN 10 100Mbps RJ 45 port for LAN uplink connection Supports PoE 802 3af Power Jack 5V DC power port UL Certified Headset Jack RJ9 Handset Jack RJ1 1 SAFETY COMPLIANCES The GXP phone complies wth FCC CE and various safety standards The GXP power adaptor is compliant with the UL standard Only use the universal power adaptor provided with the GXP package The manufacturer s warranty does not cover damages to the phone caused by unsupported power adaptors
49. 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 Please do NOT disrupt or power down the unit If a firmware upgrade fails for any reason e g TF TP HT TP server is not responding there are no code image files available for upgrade or checksum test fails etc the phone will stop the Upgrading process and re boot using the existing firmware software Firmware upgrades take around 60 seconds in a controlled LAN or 5 10 minutes over the Internet We recommend completing firmware upgrades in a controlled LAN environment whenever possible No Local TFTP HTTP Server For users who do not have a local TFTP HTTP server we provide a HI TP 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 HI TP server IP address htto 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 Instructions for lo
50. phone through a Web browser such as Microsoft s IE Mozilla Firefox Google Chrome Access the Web Configuration Menu To access the phone s Web Configuration Menu e Connect the computer to the same network as the 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 htto 1 92 168 0 60 in the address bar your browser The default administrator password is admin the default end user password is 123 NOTE When changing any settings always SUBMIT them by pressing the button on the bottom of the page Reboot the phone to have the changes take effect If after having submitted some changes more sett
51. red 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 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 Ring Tone Choose different ring tones in the Ring Tone menu Ring Volume Press Menu button to hear the selected ring volume press lt 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 logo Display Language You can choose English Simplified Chinese Traditional Chinese Korean Japanese Italian Spanish French GXP1450 User Manual Page 18 of 38 Firmware 1 0 1 26 Last Updated 12 2010 ndstream Innovative IP Voice amp Video German Portuguese Russian Croatian or Secondary Language Press Menu button to choose the menu item Press to return to the main menu Config Press Menu button to display the configuration selections e Network To enable disable DHCP To setup IP address Net mask and Gateway address e SIP To change SIP server settings f
52. se default setting Enter username for TR 069 Enter password for TR 069 URL for TR 069 Auto Configuration Servers ACS selects the file download mode for the download server Users can choose from TFTP HTTP No The URL IP address of the phonebook download server The interval at which the phonebook will be downloaded from the download server in Minutes The default setting is 0 If set to Yes the phone will remove the manually edited entries in the old phonebook list before downloading the new file The default setting is set to Yes IP address or domain name of LDAP script server Enable XML Idle Screen download via TFTP or HTTP Select whether to Use Custom Filename or not and define the XML server path Enter server path for XML application Defines the softkey label for the XML application To configure a User ID extension to dial automatically when the phone is taken offhook The IP address or URL of System log server This feature is especially useful for ITSPs GXP1450 User Manual Page 26 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Syslog Level NTP server SSL Certificate SSL Private Key SSL Private Key Password Distinctive Ring Tone System Ring Tone Call Progress Tones GE seven Innova tive IP Voice amp Vi 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 eve
53. select e Press OK again to dial 5 USING THE PHONEBOOK Calling a phone in from the hones 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 Go to the phonebook by i Pressing the phonebook button bottom left hand side of phone or li Pressing the DOWN arrow key or ili 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 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 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 GXP1450 User Manual Page 13 of 38 Firmware 1 0 1 26 Last Updated 12 2010
54. sponding account LINE flashes red Answer call by taking Handset SPEAKER Headset off hook or pressing SPEAKER or by pressing the corresponding account LINE button 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 Paging Intercom Enabled Phone beeps once and automatically establishes the call via SPEAKER PBX or Server must also supports this feature Do Not Disturb ee T Press the menu button and scroll down to Preference Select Do Not Disturb by pressing menu button Use arrow keys to either enable or disable Do Not Disturb feature When enabled there will be a special Do Not Disturb icon appearing on the display This will send the incoming caller directly to voicemail PHONE FUNCTIONS DURING A PHONE CALL Call Waiting Call Hold 1 Hold Place a call on hold by pressing the HOLD button 2 Resume Resume call by pressing the corresponding blinking LINE 3 Multiple Calls Automatically place ACTIVE call on HOLD by selecting another available LINE to place or receive another call Call Waiting tone stutter tone audible when line is in use Mute Delete 1 Press the MUTE button to enable disable muting the microphone
55. ts 484 responses Sets the prefix added to each dialed number GXP1450 User Manual Page 31 of 38 Firmware 1 0 1 26 Last Updated 12 2010 Dial Plan BLF Call pickup Prefix Delayed Call Forward Wait Time Enable Call Features Call Log C istan Innova tive IP Voice amp Video Dial Plan Rules 1 Accepted Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 XXL at least 2 digit numbers xx only 2 digit numbers exclude 3 5 any digit of 3 4 or 5 147 any digit of 1 4 or 7 lt 2 011 gt replace digit 2 wth 011 when dialing the OR operand a D Q 3000 e Example 1 369111 1617xxxxxxx Allow 311 611 and 911 or any 10 digit numbers with leading digits 1617 e Example 2 1 900x lt 1617 gt xxxxxxx Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers e Example 3 1xxx 2 9 xxxxxx lt 2 011 gt x Allows any number with leading digit 1 followed by a 3 digit number followed by any number between 2 and 9 followed by any 7 digit number OR Allows any length of numbers with leading digit 2 replacing the 2 wth 011 when dialed 3 Default Outgoing x Allow any length of numbers Example of a simple dial plan used in a Home Office in the US 1900x lt 1617 gt 2 9 xxxxxx 1 2 9 xx 2 9 xxxxxx 011 2 9 x 8469 11 Explanation of example rule reading from left to right e M900x
56. ure 2 Keypad GUI FI0W rrannnnnnnonnnnnnnnnnvnnrnnnnennnnennanennnnennnnennanennssnnnn TABLE OF TABLES GXP1450 USER MANUAL Table 1 Equipment Packaging eee ee eee ee eee Table 2 GXP ConnectorS eee Table 3 GXP1450 Product Models sss sese Table 4 GXP1450 Feature Guide Table 5 GXP1450 Key Features in a Glance ranannnrnnnnnnnnennnrnnnennnrnnnnn Table 6 GXP1450 Hardware Specifications cccsececsseeeeeeeeeeeeeeeeens Table 7 GXP1450 Technical Specifications rrrrrrrnarerarernrrnanennnennnnnr GXP1450 User Manual Firmware 1 0 1 26 LE Innovative IP Voice amp Video EEE E 4 PETE 5 EE DN 5 EN 5 S 5 S 5 S E O A 6 EEE R T 9 EPE E P 9 EEE 12 NN 15 EEE 15 EEE 17 EEE 17 EN REE E RS 18 NN 18 EEE 21 NN 35 EE ETE 35 NN 36 ERE 36 37 EEE 38 EEE 11 EEE 11 EE 20 KN 9 S 5 E E E EE 6 R E A AEE N 6 S D S 7 EEE 7 Page 2 of 38 Last Updated 12 2010 Gin nnovative IP Voice amp Video FE PENN 9 TE PT 9 Table 11 GXP Call dT lt T 17 Table 12 Key Pad Configuration Menu sese eee ee ee eee 18 Table 13 Device Configuration Status ccccccccseccseceseeeeeeeseeeteeeeeeeeseeeseeeteeeeseetuetsneeseeesaeess 22 Table 14 Device Configuration Settings Basic SettingS ccccceccseceseeeteeeseeeeeeeeeeeeseeeeeeess 22 Table 15 Device Configuration Settings Advanced SettingS ccccccseccseeeseeeseeeeeeseeeseeens 24 TEN Pr NN 29 GUI I

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