GXP User Manual - Cheap IP Phones
Contents
1. 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 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 GXP1450 User Manual Page 14 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 N Innovative IP Voice amp Video will2. 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 URL 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 cfgXXXXXXXXXXXX followed by a request for configuration XML file named cfgxxxxxxxxxxxx xml where XXXXXXXXXXXX is the MAC address of the device i e cfg000b820102ab The configuration file name should be in lower cases For more details on XML provisioning please refer to http www grandstream com support 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 th
3. 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 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 Default is No If set to Yes conference will be disabled 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 Default is No If set to Yes the phone will use attended transfer by default Configures the access control of configurations via the phone keypad menu There are three modes e Unrestricted e Basic Settings Only e Constraint Mode Default is No If set to Yes when pressing STAR key for 4 5 seconds there will be a lock icon shown in the right side of the screen indication the keypad is locked To unlock pressing STAR key for 4 5 seconds and there will be a window prompted asking for passw
4. 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Dial Plan Delayed Call Forward Wait Time Enable Call Features Call Log Andstream Innovative 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 xx 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 with 011 when dialing the OR operand a b GO OO CO e Example 1 369 11 161 7xxxxxXX Allow 311 611 and 911 or any 10 digit numbers with leading digits 1617 e Example 2 1900x 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 8 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 with 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 3469 11 Explanation of example rule reading from left to right e 1900x prevents dialing any number started with 1900
5. IP Voice amp Video 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 7 LCD Buttons LCD Button DATE AND TIME LOGO NAME NETWORK STATUS STATUS BAR LINE STATUS INDICATOR SOFTKEYS Table 8 LCD Icons message LCD Button Definitions Displays the current date and time Can be synchronized with Internet time servers Displays company logo name This logo name can be customized via xml screen customization Displays the status of the phone and network It will indicate whether the network is down starting or running IP address MISSED CALLS is 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 key on the left side The softkeys are context sensitive and will change depending on the status of the phone Typical functions assigned to soft buttons are e FORWARD ALL Unconditionally forwards the phone line to another phone e MISSED CALL This option shows up there were unanswered calls to this phone The Missed Calls option shows a list of the missed calls e NEXTSCR Press this button to toggle between idle screen weather and IP Address e REDIAL Redials the last number e END CALL Hangs up phone LC
6. SIP account to make the phone call e Go to the phonebook by Pressing the phonebook button bottom left hand side of phone or i 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 5 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 GXP1450 User Manual Page 11 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video e Press SEND key NOTE Dialtone and dialed number display occurs after the phone 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 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
7. Table 13 Device Configuration Settings Basic Settings End User Password This contains the password to access the Web Configuration Menu This field is case sensitive with a maximum length of 25 characters IP Address The GXP1450 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 GXP1450 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 GXP1450 establishes a PPPoE session if any of the PPPoE fields Is set 2 PPPoE mode configure all of the following fields PPPoE account ID PPPoE password and PPPoE service name 3 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 GXP1450 User Manual Page 22 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 802 1x Mode Line Keys x Time Zone Self Defined Time Zone Weather Update LCD Backlight Brightness LCD Contrast Time Display Format Disable in call DTMF display Andstream Innovative IP Voice amp Video This option allows the user to enable disable 802 1x mode on the phone The default value is disabled To enable
8. 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 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 accounts on the phone In addition they would need to navigate to Settings gt Basic Settings on the web GUI 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 includin
9. 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 GXP1450 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 0 5 second RFC 3261 SIP T2 timer Default is 4 seconds Choose SIP Transport between UDP and TCP Default is UDP Enable to check the domain certificate Default is No 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 Support SIP Instance ID Selects whether or not SIP Instance ID is supported GXP1450 User Manual Page 30 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 NAT Traversal SUBSCRIBE for MWI PUBLISH for Presence Proxy Require Voice Mail UserID Send DTMF DTMF Payload Type Early Dial Dial Plan Prefix Andstream Innovative IP Voice amp Video This parameter acti
10. 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 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 ForwardAll 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 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 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 Ring Timeout Send Anonym
11. language based on IP location if available Also the phone will download secondary language if available e Time Settings Press Menu button to choose the menu item Press or follow the soft keys to return to the main menu Press Menu button to display the configuration selections e SIP To change SIP server settings for SIP accounts 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 802 1Q VLAN Tag and priority value 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 in the LCD Lift and put back the handset or press Menu button to exit the diagnostic mode Press to return the main menu To enable disable DHCP to setup IP address Net mask and Gateway address Press Menu button to reboot the device Exit from this menu FIGURE 2 KEYPAD GUI FLOW GXP1450 User Manual Page 18 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www In
12. 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 GARE w To make a quick IP call please refer to 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 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 addr
13. 800 088 4846 andstream 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 power adaptor with the GXP1450 as it may cause damage to the products and void the manufacturer warranty 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 3 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales inter
14. 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 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 This parameter controls the date time display according to the specified time zone If Allow DHCP Option 2 to override Time Zone setting is checked the time zone will be overridden by the DHCP server This parameter allows the users to define their own time zone The syntax is std offset dst offset start time end time Default is set to MTZ 6MDT 5 M3 2 0 M11 1 0 MTZ 6MDT 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 1st number indicates Month 1 2 3 12 for Jan Feb Dec The 2nd number indicates the nth iteration of the weekday 1st Sunday rd Tuesday The ard 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 By default Enable Weather Update is set to Yes If set t
15. D Icon Definitions DND Icon ON when the Do Not Disturb is activated Calls Forwarded Icon INDICATES calls are forwarded Key pad lock Icon ON when using STAR key to lock the keypad Enter Password to unlock the keypad Voice Mail Message Waiting Indicator ON when there is new voice mail Network Status Network is down GXP1450 User Manual Page 8 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www lnternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 D Missed Call Icon Indicates missed call s Andstream Innovative IP Voice amp Video a Save Call Record Indicates phone system writing the call records into the flash It might take 10 to 20 seconds to finish the process FIGURE 1 GXP1450 KEYPAD LAYOUT Keypad Table 9 GXP1450 Keypad Buttons Message Waiting Indicator Line Keys Message Phonebook Soft Keys Conference Navigation Keys Menu OK Key Volume Headset Mute Speaker Send Key Button Key Button Definitions LINE BUTTONS 2 Line keys with LED can be configured to different SIP profiles HOLD Place ACTIVE call on hold TRANSFER Transfer an ACTIVE call to another number CONF Press CONF button to connect Calling Called party into conference FAKE Brings phonebook on screen Mute an active call Enable Disable hands free speaker Enter to retrieve voice mails or other messages Press HEADSET key to answer hang up phone calls while using headset It also allows user t
16. EDIAL PHONEBOOK MESSAGE 3 XML Programmable Softkeys 5 Navigation keys Device NAT friendly remote software upgrade via TFTP HTTP for deployed Management devices including behind firewall NAT 3 Auto manual provisioning system Web GUI Interface E Address Book Audio Features Full duplex hands free speakerphone 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 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 and loss concealment HD audio handset with HD wideband audio codecs for excellent double talk performance Telephony Features i Intuitive graphic user interface GUI downloadable phone book XML i 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 GXP1450 User Manual Page 6 of 38 Firmware 1 0 1 66 La
17. GH Innovative IP Voice amp Video Grandstream Networks Inc GXP1450 SIP Enterprise Phone Grandstream Networks Inc GXP1450 User Manual Page 1 of 1 Firmware 1 0 1 66 Last Updated 05 2011 TABLE OF CONTENTS GXP1450 USER MANUAL id DER SV OO EE sveuceunssuedalceckcnasedascdebudcususuedabccatesuvaucusseubieeeessyscontcseieaucens 3 INSTALLATION lla 4 FOUPMENT EE eg 4 CONNECTING YOUR Le SE 4 SAFETY COMPLIANCE Saia aiar 4 VORREI lin 4 PRODUCT OVERVIEW iss nss ie ses ies ees ide n oe oe sd ee eed dd os de oe sd di oe Ge Geo si ed oe Ge od be Ge Ge ed 5 USING THE GXP1450 SIP ENTERPRISE PHONE ccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccecs 8 GETTING FAMILIAR WITH THE ECD oriali TAO TI 8 MAKING PHONE CALI Sas See Se Se ee ee Ge ee De Ge eo Ge Ge NT ratio 10 ANSWERING PHONE CALLS sesse ee Ge TR Ge Ge a Dee GE Ge ATI ee EE Se Ge GO Gie 13 PHONE FUNCTIONS DURING A PHONE CALL ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee 13 CALP dd EE ER N EE OR OE OO OE OE EE OO 15 CUSTOMIZED LCD SCREEN amp d BE 16 CONFIGURA TION GUIDE all Jia 17 OIG WR ISCH 17 CONFIGURATION VIA WEB BROWSER a ra 21 SAVING THE CONFIGURATION CHANGES EE 35 REBOOTING THE PHONE REMOTEI SY root dae 35 SOFTWARE UPGRADE amp CUSTOMIZATION ccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccecs 36 FIRMWARE UPGRADE THROUGH NN CO NAH TP LE 36 CONFIGURAT
18. ION FIEEDOWNEOADr srl 37 RESTORE FACTORY DEFAULT SETTING cccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccsccccccecs 38 TABLE OF FIGURES GXP1450 USER MANUAL Figure 1 GXP1450 Keypad Layout EE ER RE ER RR ER RE Re ee Re RR ee EE Re Ee ee ee EE ee Ee ee ee ee 10 FOUG ELE E IR e RE EO ER EER OO ER ER OE OE 18 TABLE OF TABLES GXP1450 USER MANUAL Fable Ge ein Ed Paokad ING SE DR EE n ere eee eA 4 Table 2 GRP 1490 COMECOU Sia 4 Table GRR 1450 Feature Udi 5 Table 4 GXP1450 Key Features in a Glance i 5 Table 5 GXP1450 Hardware SpecificationS iii 5 Table 6 GXP1450 Technical Gpoecfcatons i 6 Eelere COD BUO oee E 8 talle 8 Ree pp 8 Table 9 GXP1450 Keypad Buttons 9 Grandstream Networks Inc GXP1450 User Manual Page 1 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Table 10 Table 11 Table 12 Table 13 Table 14 Table 15 Andstream Innovative IP Voice amp Video FF Gall se OS RE EE OA 15 Key Pad Conu ation Meli 17 Device Connguration E 22 Device Configuration Settings Basic Gettnges ie 22 Device Configuration Settings Advanced Gettmges iss see Ee Ee Ee Ee ee 24 ACCIONES arean EE E E E ee 29 GXP1450 User Manual Page 2 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0
19. Manual Firmware 1 0 1 66 Page 25 of 38 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 TR 069 Username TR 069 Password Save Credentials Auto Login Periodic Inform Enable Periodic Inform Interval Connection Request Username Connection Request Password Authentication Method Connection Request Port Phonebook XML Download Phonebook XML Server Path Phonebook Download Interval Remove Manually edited entries on Downloads LDAP Directory Idle Screen XML Download Download Screen XML At Boot up Use custom filename Idle Screen XML Server Path XML Application Offhook Auto Dial Syslog Server andstream Innovative IP Voice amp Video Enter username for TR 069 Enter password for TR 069 Save TR 069 credentials Default is No Auto Login TR 069 account Default is No Enable periodic inform Default is No When enabling periodic inform set up the periodic inform interval Enter the connection request username Enter the connection request password Select the authentication method among No authentication Basic or Digest Enter the connection request port 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 def
20. 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 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 GXP1450 can host conference calls and supports up to 3 way conference calling 1 Initiate a Conference Call Establish a connection with two 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 host 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 f 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
21. XML applications 1 XML Custom Screen 2 XML Downloadable Phonebook and 3 Advanced XML Survey Application For more information on how to create a downloadable XML phonebook creating a custom idle screen and or reprogramming the softkeys on GXP1450 please visit our website at hitp www grandstream com support GXP1450 User Manual Page 16 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Configuration Guide Andstream Innovative IP Voice amp Video 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 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 11 Table 11 Key Pad Configuration Menu en Description Call History Status Phone Book LDAP Directory Instant Messages Direct IP Call Preference Displays histories of answered dialed missed and transferred and forwarded calls Displays the network status account status softwa
22. XP1450 User Manual Page 29 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Authenticate ID Authenticate Password Name DNS Mode Primary IP Backup IP 1 Backup IP 2 SIP Registration Unregister on Reboot Register Expiration Local SIP Port SIP Registration Failure Retry Wait Time SIP T1 Timeout SIP T2 Interval SIP Transport Check Domain Certificate Remove OBP from Route Validate Incoming Messages Andstream Innovative IP Voice amp Video 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 GXP1450 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 Secondary 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 Insert the first back up IP here Insert the second back up IP here This parameter controls sending REGISTER messages to the proxy server The
23. ault 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 The phone will download the idle screen xml file if set to Yes The default setting is No The phone will use custom filename specified in XML server path if set to Yes The default setting is No Specify the idle screen XML server path Server path enter server path for XML application Softkey Label define 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 Firmware 1 0 1 66 Page 26 of 38 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Syslog Level Send SIP Log NTP server Allow DHCP Option 42 to override NTP server SSL Certificate SSL Private Key SSL Private Key Password Distinctive Ring Tone System Ring Tone Andstream Innovative IP Voice amp Video Select the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog mes
24. 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 to 255 You will need this number to access the Web Configuration Menu For example if the phone shows 192 168 0 60 please use Attp 192 168 0 60 in the address bar of your browser Co The default administrator password is admin the default end user password is 123 NOTE When changing any settings always SUBMIT them by pressing UPDATE 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 user can log in as an administrator or end user Functions available for the end user are e Status Displays the network status account status software version and MAC address of the phone and service status e Basic Settings Basic preferences such as date and time settings multi purpose keys and LCD settings can be set here Additional functions availabl
25. e Firmware or Provisioning Server load at any given time GXP1450 User Manual Page 37 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream 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 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 A 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 a e od Example if the MAC address is 000b8200 395 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 Pre
26. e to administrators are e Advanced Settings To set advanced network settings codec settings and XML configuration settings and etc e Account To configure the SIP account GXP1450 User Manual Page 21 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video Table 12 Device Configuration Status MAC Address The device ID in HEXADECIMAL format IP Address This field shows IP address of GXP1450 Product Model This field contains the product model information Part Number This field contains the product part number Software Version Program This is the main firmware release number which is always used for identifying the software or firmware system of the phone Boot Booting code version number Core Core code version number Base Base code version number DSP DSP code version number Aux Aux code version number System Up Time This field shows system up time since the last reboot System Time This field shows the current time on the phone system Registered Indicates whether accounts are registered to the related SIP server PPPoE Link Up Indicates whether the PPPoE connection is enabled connected to a modem Service Status e GUI shows the GUI status running or stopped Phone shows the phone status running or stopped Core Dump Download core dump file for troubleshooting when necessary
27. e will send out the dialed number GXP1450 User Manual Page 34 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Use as Dial Key G723 Rate G726 32 Packing Mode ILBC Frame Size iLBC Payload Type Special Feature andstream Innovative IP Voice amp Video This parameter allows users to configure the key as the Send or Dial key If set to Yes the key will immediately send the call In this case this key is essentially equivalent to the Re Dial key If set to No the key is included as part of the dial string Encoding rate for G723 codec By default 6 3kbps rate is set Select ITU or IETF for G726 32 packing mode 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 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 rebooting or powering 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 ph
28. en the web browser Enter the GXP1450 IP address Enter the admin password to access the web configuration interface In the ADVANCED SETTINGS page enter the Upgrade Servers IP address or FQDN in the Firmware Server Path field Select TFTP or HTTP upgrade method Update the change by clicking the Update button Reboot or power cycle the phone to update the new firmware During this stage the LCD will display the firmware file downloading process Please do NOT disrupt or power down the unit If a firmware upgrade fails for any reason e g IFTP 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 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 HTTP server on the public Internet for users to download the latest firmware upgrade automatically Please check the Support Download section of our website to obtain this HTTP server IP address http www grandstream com support firmware 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 sola
29. ess 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 GXP1450 User Manual Page 12 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 andstream innovative IP Voice amp Video 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 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 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 a
30. freshed 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 GXP1450 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 66 Page 33 of 38 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Refer To Use Target Contact Transfer on Conference Hangup Preferred Vocoder SRTP Mode Symmetric RTP Silence Suppression Voice Frame
31. g caller ID caller ID w name call forward transfer park hold as well as intercom paging and BLF Table 10 GXP1450 Call Features Key Call Features 30 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 70 Disable Call Waiting per Call 71 Enable Call Waiting per Call 72 Unconditional Call Forward Dial 72 for a dial tone Dial the forwarding number followed by Wait for dial tone LCD will display Call FWD Activated 73 Cancel Unconditional Call Forward dial 73 and get the dial tone then hang up LCD will display Call FWD Activated 90 Busy Call Forward Dial 90 for a dial tone Dial the forwarding number followed by Wait for a dial tone Hang up 91 Cancel Busy Call Forward dial 91 Wait for dial tone Hang up 92 Delayed Call Forward Dial 92 for a dial tone Dial the forwarding number followed by Wait for a dial tone Hang up LCD will display Call FWD Activated GXP1450 User Manual Page 15 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video 93 Cancel Delayed Call Forward Dial 93 for a dial tone then hang up CUSTOMIZED LCD SCREEN amp XML GXP1450 Enterprise IP phone support both simple and advanced
32. ge of the Web UI GXP1450 User Manual Page 24 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 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 Allow DHCP Option 43 and Option 66 to override server Automatic Upgrade Authenticate Conf File Enable TR 069 ACS URL andstream innovative IP Voice amp Video 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 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 HTTP HTTPS server The password for the HTTP HTTPS server This field allows the user to choose the firmware upgrade method TFTP HTTP or HTTPS Defi
33. ideo Product Overview Table 3 GXP1450 Feature Guide Features GXP1450 LCD Display 180x60 pixel lt Number of Unes BR Programmable Soft Keys 3 Extension Module NA Table 4 GXP1450 Key Features in a Glance Features Benefits eee Open Standards Compatible SIP RFC3261 TCP IP UDP RTP RTCP HTTP HTTPS ARP RARP ICMP DNS A record SRV and NAPTR DHCP both client and server PPPoE TELNET TFTP NTP STUN SIMPLE SIP over TLS I 802 1x TR 069 eee Superb Audio Quality Advanced Digital Signal Processing DSP Silence Suppression VAD EE ETA SE Network Interfaces Dual 10 100mbps Ethernet ports with integrated PoE Feature Rich Traditional voice features including caller ID call waiting hold transfer forward block auto dial off hook dial Advanced Features 2 line keys with dual color LED and 2 SIP accounts 3 way conferencing backlit graphic 180x60 LCD 3 XML programmable context sensitive soft keys 5 navigation keys 10 dedicated buttons for HOLD TRANSFER CONFERENCE VOLUME HEADSET MUTE SPEAKERPHONE SEND REDIAL PHONEBOOK MESSAGE Customized downloadable ring tones SRTP SIP over TLS multi language support and XML enabled adjustable positioning angles wall mountable AES encryption automatic multimedia service eg weather i information Table 5 GXP1450 Hardware Specifications GXP1450 LAN Interface Ethernet ports Two 2 10 100 Mbps Full Half Duplex Ether
34. ing2 Download SCR XML Ring 3 Erase Custom SCR Back Display Language Back LCD Brightness Active f Idle Config Back SIP Upgrade Factory Reset Layer 2 QoS Back Display Language Factory Function English Chinese Audio Loopback French Diagnostic Mode Spanish Back German Italian Secondary Language Language File Postfix Network Back IP Setting Diagnostic Mode PPPoE Settings IP Keypad LED Diagnostic Netmask Gateway DNS Server 1 DNS Server 2 Back GXP1450 User Manual Firmware 1 0 1 66 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 802 1Q VLAN Tag Priority value Reset Vlan Config Back Page 20 of 38 Last Updated 05 2011 Andstream Innovative IP Voice amp Video CONFIGURATION VIA WEB BROWSER The GXP1450 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 Microsofts 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 administrators password to access the Web Configuration Menu The Web enabled computer has to be connected to the same sub network as the phone This
35. le LINE key activates soeakerphone 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 softkey 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 second SIP account was made for the last call call attempt the phone will use the second 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 softkey 3 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 second SIP account the phone will use the second 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 4 USING THE PHONEBOOK To call 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
36. nd automatically establishes the call via SPEAKER PBX or Server must also supports this feature Do Not Disturb DM rr 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 i Hold Place a call on hold by pressing the HOLD button Resume Resume call by pressing the corresponding blinking LINE 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 Press the MUTE button to enable disable muting the microphone 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 transfer 1 Blind Transfer Press TRANSFER button then dial the number and press the SEND button to complete transfer of active call GXP1450 User Manual Page 13 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 andstream innovative IP Voice amp Video 2 Attended
37. nes 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 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 Default is Yes This allows device gets provisioned from the server 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 HTTP 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 use default setting Default is No URL for TR 069 Auto Configuration Servers ACS GXP1450 User
38. net Switch with LAN and PC port with auto detection Expansion Module Support no NO Headset Jack E RJS Call Appearance LED 2 Dual color genre Power over Ethernet 0 Built in auto sensing Cisco and IEEE 802 3af standard Universal Switching i Input 100 240VAC 50 60Hz i tts GXP1450 User Manual Page 5 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 andstream Innovative IP Voice amp Video Power Adaptor Output 5VDC 800mA UL certified Dimension di 186mm W x 210mm L x 81mm D Weight I OBKG Temperature 382104 F0 40C Humidity Ee 10 90 non condensing Compliance a FCC CE C Tick Table 6 GXP1450 Technical Specifications Lines e 2 lines with 2 independent SIP accounts XML programmable soft keys Ed E Ehe he E sala r 71 7 7 Protocol Support Support SIP 2 0 TCP UDP IP PPPoE RTP RTCP SRTP by SDES HTTP ARP RARP ICMP DNS DHCP NTP TFTP SIMPLE PRESENCE protocols 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 HOLD TRANSFER CONF VOLUME HEADSET MUTE SPEAKERPHONE SEND R
39. netvoipphone co uk 0800 088 4846 andstream Innovative IP Voice amp Video Installation EQUIPMENT PACKAGING Table 1 Equipment Packaging XP1450 Handset Ve CONNECTING YOUR PHONE The connectors of the GXP1450 are located on the bottom of the device Table 2 GXP1450 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 RJ9 SAFETY COMPLIANCES The GXP1450 complies with FCC CE and various safety standards The GXP1450 power adaptor is compliant with the UL standard Only use the universal power adaptor provided with the GXP1450 package The manufacturer s warranty does not cover damages to the phone caused by unsupported power adaptors WARRANTY If you purchased your GXP1450 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 4 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp V
40. nguage This is similar to updating firmware in your local network environment 1 Get the language file gxp txt ready Make sure the file is using UTF 8 encoding 2 Copy gxp txt to the firmware server directory using your local TFTP or HTTP server 3 Access the advanced settings of the Web GUI set Display Language to Download Language and enter the server path in Firmware Server Path Select TFTP or HTTP for firmware upgrade 4 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 15 SIP Account Settings Account Active Account Name SIP Server Secondary SIP Server Outbound Proxy SIP User ID This field indicates whether the account is active The default value is Yes The name associated with each account displayed on LCD SIP Server s IP address or Domain name provided by VoIP service provider This field allows administrator to configure a backup SIP Server 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 G
41. o No weather information will not display on the phone Settings to customize the display of weather via e City Code Enter city code e Update Interval Refresh time in minutes e Degree Unit Select Automatic Fahrenheit or Celsius Weather information is displayed on GXP1450 LCD when Enable Weather Update is set to Yes and pressing the SwitchSCR soft key once Set the LCD brightness level for idle state and active state Range from 0 to 8 where 0 means off and 8 means the brightest Set LCD contrast Range from 0 to 20 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 GXP1450 User Manual Firmware 1 0 1 66 Page 23 of 38 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Disable Missed Call Backlight HEADSET Key Mode Headset TX gain dB Headset RX gain dB Andstream Default is No By default LCD backlight will light up whenever there is a missed call Default Mode Toggle to Headset when using Speaker Handset Dial pick up call or hang up call using Headset Toggle Headset Speaker toggle between using Headset and using Speaker Set headset TX gain to 6 0 or 6 Default is 0 db Set headset RX gain to 6 0 or 6 Default is 0 db Table 14 Device Configuration Settings Advanced Settings Admin Password Layer 3 QoS Layer 2 Q
42. o toggle between headset and speaker GXP1450 User Manual Firmware 1 0 1 66 Page 9 of 38 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video Enable Disable handset mode or used as SEND REDIAL Press or to adjust the volume for handset speakerphone headset E gt Enter Keypad Configuration MENU mode when phone is in IDLE mode dd f Use as ENTER key when in Keypad Configuration Standard phone keypad press key to send call press key to for IVR 0 9 functions 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 and a dial
43. oS Local RTP port Use Random Port Keep alive interval Use NAT IP STUN Server 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 This field defines the layer 3 QoS parameter It is the value used for IP Precedence or Diff Serv or MPLS Default value is 12 This contains the value used for layer 2 802 1Q VLAN tag and 802 1p priority value Default setting is 0 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 Local RTP port ranges from 1024 to 65400 and must be even 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 This parameter specifies how often the GXP1450 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 pa
44. one 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 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IF Voice amp Video Software Upgrade amp Customization Software or firmware upgrades are completed via either TFTP or HTTP The corresponding configuration settings are in tte 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 2 3 5 72 172 83 110 There are two ways to set up the Upgrade Server to upgrade firmware via Key Pad Menu or 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 op
45. ord Enter the password to lock the keypad in web GUI To unlock the keypad enter the password in the prompted window in the phones LCD screen GXP1450 User Manual Firmware 1 0 1 66 Page 28 of 38 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Display Language Andstream Innovative IP Voice amp Video Allows user to choose preferred display language in web UI and key pad UI Currently the phone supports these languages English Simplified Chinese Traditional Chinese Korean Japanese Italian Spanish French German Portuguese Russian Croatian Hungarian Polish and Slovenian 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 file name is gxp txt If the field Language File postfix has NL string in it then the phone will request gxp_NL txt instead of gxp txt User can only load one secondary language Supported downloadable language Czech Dutch Estonian French German Italian Polish Portuguese Slovak Slovenian and Spanish How to set up Download La
46. ous Anonymous Call Rejection Auto Answer Allow Auto Answer by Call Info Andstream 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 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 re
47. r 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 GXP1450 User Manual Page 27 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Call Progress Tones Intercom User ID Disable Call Waiting Disable Call Waiting Tone Disable Direct IP Calls Use Quick IP Call Mode Disable Conference Disable DND Button Disable Transfer Auto Attended Transfer Configuration via Keypad Menu Enable STAR key Keypad locking Password to lock unlock Andstream Innovative IP Voice amp Video 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 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 Configure intercom user ID when intercom is used Default is No If set to Yes
48. re version MAC address and hardware version of the phone Displays the phonebook and downloads phonebook XML Displays the LDAP directory and downloads directory Goes to instant messages Dials IP address for direct IP call Press Menu button to enter this sub menu including Do NOT Disturb DND Do Not Disturb function could be turned on or off in the Do Not Disturb menu Ring Tone Choose different ring tones in the Ring Tone menu Ring Volume Press Menu button to hear the selected ring volume press or gt to hear and adjust the ring tone volume LCD Contrast Press or to adjust the LCD contrast LCD Brightness Press lt or to adjust the LCD brightness for active idle screen Download SCR XML The phone will download the custom idle screen if available GXP1450 User Manual Page 17 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Config Factory Functions Network Reboot Exit Andstream Innovative IP Voice amp Video e Erase Custom SCR Custom idle screen will be erased and will be replaced with default logo e Display Language You can choose English Simplified Chinese Traditional Chinese Korean Japanese Italian Spanish French German Portuguese Russian Croatian Hungarian Polish Slovenian which are built in the phone Users could select Automatic for local
49. rwinds net updates New customerFree cfm GXP1450 User Manual Page 36 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream 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 GXP1450 should be in the same LAN segment 3 Go to File gt Configure gt Security to change the TFIP 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 GXP1450 phone boots up it will send TFTP or HTTP request to download configuration file cfg000b82xxxxxx where 000b82xxxxxx is the MAC address of the GXP1450 phone This file is for provisioning purpose For normal TFIP or HITP firmware upgrades the following error messages in a TETP or HTTP server log can be ignored TFTP Error from IP ADRESS requesting cfg000b82023d44 File does not exist Configuration File Download CONFIGURATION FILE DOWNLOAD The GXP1450 can be configured via Web Interface as well as via Configuration File
50. s per TX No Key Entry Timeout Andstream Innovative IP Voice amp Video Default is No If set to Yes then for Attended Transfer the Refer To header uses the transferred targets 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 GXP1450 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 reque
51. sages 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 device MAC addressjlerror 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 When setting the Yes phone will send out SIP Log to syslog server Default setting is No This parameter defines the URI or IP address of the NIP Network Time Protocol serve It is used to display the current date time Default is Yes This allows device gets provisioned for DHCP Option 42 from the server automatically 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 GXP1450 will ONLY use selected ring tones for particular Caller IDs For all other calls the GXP1450 will use System Ring Tone When selected and no Calle
52. ss 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 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846
53. st Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video i 3 way conference dial plan prefix dial plan support off hook auto dial auto answer early dial and speed dial EER EER EER EER EE EER EE EER EE EER RE EE EER EER EER EER EER EER Network and Provisioning 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 7 Support for IEEE 802 1p Q tagging VLAN Layer 3 ToS Firmware Upgrade Support firmware upgrade via TFTP or HTTP i Support for Authenticating configuration file before accepting changes i User specific URL for configuration file and firmware files Mass provisioning using TR 069 or encrypted XML configuration file ee a a TEE TERT EETL ELE ELE TELL EE MMMM Advanced Server Features Message waiting indication support DNS SRV Look up and SIP Server Fail Over Support customizable idle screen via downloading XML by HTTP TFTP Security User and administrator level passwords MD5 and MDS sess based authentication AES based secure configuration file SRTP TLS 802 1x media access control GXP1450 User Manual Page 7 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative
54. sted 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 GXP1450 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 After the timeout the phon
55. ternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video GXP1450 User Manual Page 19 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 MENU Call History Status Phone Book LDAP Directory Instant Message Direct IP Call Preference Config Factory Functions Network Reboot Exit Call History Any of previous menus Answered Calls Back Dialed Calls Clear All Missed Calls Transferred Calls Forwarded Calls New Entry Phone Book Name New Entry gt Number Download Phonebook XML Acct Back Confirm Add Cancel and Return LDAP Directory i I Search Configuration View Directory Download Directory Select Filter Search Configuration P Filter Value Back Back Instant Message Do Not Disturb ndstream Innovative IP Voice amp Video SIP Account SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio Save Upgrade Firmware Server Config Server Upgrade Via Layer 2 QoS Clear All Back Enable DND Disable DND Back Preference Ring T Do Not Disturb ARA Ring Tone m Default Ring LCD Contrast Ring1 LCD Brightness R
56. tone is heard For example Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as VoIP 1 VoIP 2 respectively and ensure that they are active and registered When LINE1 is pressed you will hear a dial tone and see VoIP 1 on the LCD display when LINE2 is pressed you will hear a dial tone and see VoIP 2 on the LCD display To make a call select the line you wish to use The corresponding LINE LED will light up in green User can switch lines before dialing any number by pressing the same LINE button one or more times If you continue to press a LINE button the selected account will circulate among the 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 GXP1450 will flash the next available LINE in red A line is ACTIVE when it is in use and the corresponding LED is red Completing Calls There are five ways to complete a call GXP1450 User Manual Page 10 of 38 Firmware 1 0 1 66 Last Updated 05 2011 www InternetVoipPhone co uk sales internetvoipphone co uk 0800 088 4846 Andstream Innovative IP Voice amp Video 1 DIAL To make a phone call e Take Handset SPEAKER Headset off hook or press an availab
57. vates the NAT traversal mechanism It has options No STUN Keep Alive UPnP Auto VPN If selecting STUN 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 selecting Keep Alive with no specified STUN server the GXP1450 will periodically every 20 seconds or so send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Enable Presence feature SIP Extension to notify SIP server that the unit is behind the NAT Firewall When configured user can access messages by pressing MSG button This ID is usually the VM portal access number This parameter specifies the mechanism to transmit DTMF digit There are 3 supported modes in audio which means 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 supports 484 responses Sets the prefix added to each dialed number GXP1450 User Manual Page 31 of 38 Firmware
Download Pdf Manuals
Related Search