GXP User Manual
Contents
1. ndstream Innovative IP Voice amp Video Grandstream Networks Inc GXP1400 1405 Small Medium Business IP Phone Grandstream Networks Inc GXP1400 1405 User Manual Page 1 of 1 Firmware version 1 0 1 67 Last Updated 05 2011 TABLE OF CONTENTS GXP1400 1405 USER MANUAL NLM NNN 2 INSIESEGSTION eda 3 FOUPMENT ANN 3 COMET 3 TRONEN NN 3 VNR 3 PRODUCT OVERVIEW wosciceccsscccasecucevevesascescdseavesessccendascdevesessceveeddncevesevavacsvwsedevedessaceeddaseVesedavewavecseeveseseeuneddawavesetans 4 USING THE Ge PAN IE mica 7 GETTING FAMILIAR WITH THE LCD oesonnnenernvnnvnnennvnnvnnnnennennennennvnnnnennennennnnennennennenunnennennenuenennennennenunnsnnennenuenennenneneeneene 7 ENE SNOKER 8 ANSWERING PHONE CALLS erevnevnvvnevnvnevnevnunnvnnvnevnunnuennenevnennunnenevnennunnennvnennunnenneneenennunnenevnennunnennenennunnsnnennenennunnenneneenenn 10 PHONE FUNCTIONS DURING A PHONE CALL s oesnevnvnnennvnnvnvnnennennvnnnnvnnennennnnnnnennennenennennennvnnenennennennnnnnnennennenennennennennenenn 11 RNA 12 CUSTOMIZED LCD SCREEN SEXI rie TIRI TATA ENI 13 CONFIGURATION GUIDE REE EEE 14 CONFIGURE TINN MND 14 CONFIGURATION VIA WEB BROWSER iii 17 SAVING THE CONFIGURATION CHANGES EE EE EE ET EN ET RE ER 31 MONTE NVNv 31 SOFTWARE UPGRADE amp CUSTOMIZATION 2 cccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccecccccccccees 32 FIRMWARE UPGRADE THROUGH TFTP HTTP eesnenevnevnvnnvvnvnevnevnunnennvnevnunnennene2. 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 i RFC 3856 3863 for use of MFKs SIP Dialog package RFC 4235 Feature Keys HOLD TRANSFER CONF LINE 1 LINE 2 MSG SPEAKERPHONE HANDSET HEADSET MUTE DND NAVIGATION 5 VOLUME 3 XML Programmable Soft keys Device Management NAT friendly remote software upgrade via TFTP HTTP for deployed devices including behind firewall NAT Auto manual provisioning system Web GUI Interface Support Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS 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
3. GXP1400 1405 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 GXP1400 1405 User Manual Firmware version 1 0 1 67 Page 29 of 34 Last Updated 05 2011 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 Grandstream Networks Inc andstream Innovative IF Voice amp Video 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 GXP1400 1405 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
4. Secondary SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Grandstream Networks Inc 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 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 GXP1400 1405 to register to SIP servers of ITSP GXP1400 1405 User Manual Page 25 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IP Voice amp Video Name SIP service subscriber s name that is used for Caller ID display DNS Mode 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 wi
5. Spanish French German Portuguese Russian Croatian Hungarian Polish Slovenian which are built in the phone Users could select Automatic for local language based on IP location if available Also the phone will download secondary Grandstream Networks Inc GXP1400 1405 User Manual Page 14 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 Config Factory Functions Network Reboot Exit andstream Innovative IP Voice amp Video 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 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 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
6. 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 GXP1400 1405 User Manual Page 28 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 Anonymous Anonymous Call Rejection Auto Answer Allow Auto Answer by Call Info Grandstream Networks Inc CC isten Innovative IF 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
7. amp Video Welcome GXP1400 1405 is a next generation small to medium business IP phone that features 2 lines with 1 SIP account a 128x40 graphical LCD 3 XML programmable context sensitive soft keys dual network ports with integrated PoE GXP1405 only and 3 way conference The GXP1400 1405 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 3 party SIP devices and leading SIP NGN IMS platforms It is a perfect choice for small to medium businesses 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 GXP1400 1405 as it may cause damage to the products and void the manufacturer warranty Note 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 Grandstream Networks Inc GXP1400 1405 User Manual Page 2 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstre
8. angles wall mountable AES encryption automatic multimedia service eg weather information ee ee 2 gt gt ee e gt Table 5 GXP1400 1405 Hardware Specifications GXP1400 1405 LAN Interface 10 100MbpsFull Half Duplex Ethernet port with auto detection Integrated PoE GXP1405 only Graphic LCD Display 128x40 piel nn PN e DEN Call Appearance LED i 2 Dual color green red line keys Grandstream Networks Inc GXP1400 1405 User Manual j Page40f34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF Voice amp Video Universal Switching Input 100 240VAC 50 60 Hz Power Adaptor iets Output 5VDC 800mA 4 0 W UL certified Dimension 86mm W x 210mm L x 81mm D ST Weight 4 Unit weight 0 KG MM Package weight 1 1KG GXP1400 1 0KG GXP1405 Temperature ss 82 104F 0 4006 Humidity 40 90 non condensing 0 Compliance FCC Part 15 CFR 47 Class B EN55022 Class B EN55024 EN61000 3 2 EN61000 3 3 EN 60950 1 AS NZS CISPR 22 Class B AS NZS CISPR 24 RoHS UL 60950 power adapter Table 6 GXP1400 1405 Technical Specifications Lines 2 lines with 1 SIP account 3 XML programmable soft keys Protocol Support Support SIP 2 0
9. 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 To enable Quick IP calls the phone has to be setup first This is done through the web setup function In the Advanced Settings page set the Use Quick IP call mode to Yes When xxx is dialed where x is 0 9 and xxx lt 255 a direct IP call to aaa bbb ccc XXX is completed aaa bbb ccc is from the local IP address regardless of subnet mask The numbers xx or x are also valid The leading 0 is not required but OK For example 192 168 0 2 calling 192 168 0 3 dial 3 followed by 192 168 0 2 calling 192 168 0 23 dial 23 followed by 192 168 0 2 calling 192 168 0 123 dial 123 followed by 192 168 0 2 dial 43 and 403 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 ANSWERING PHONE CALLS Receiving Calls Grandstream Networks Inc GXP1400 1405 User Manual Page 10 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF Voice amp Video 1 Incoming single call Phone rings with selected ring tone The corresponding LINE flashes red Answer call by taking Handset off hook or pressing SPEAKER or HEADSET or by press
10. 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 and etc e Account To configure the SIP account Grandstream Networks Inc GXP1400 1405 User Manual Page 17 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IP Voice amp Video Table 13 Device Configuration Status MAC Address IP Address Product Model Part Number Software Version System Up Time System Time Registered PPPoE Link Up Service Status Core Dump The device ID in HEXADECIMAL format This field shows IP address of GXP1400 1405 This field contains the product model information This field contains the product part number 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 e Aux Aux code version number This field shows system up time since the last reboot This field shows the current time on the phone system Indicates whether accounts are registered to the related SIP server Indicates whether the PPPoE connection is enabled connected to a modem GUI shows the GUI status running or stopped Phone shows the
11. double talk performance Telephony Features Intuitive graphic user interface GUI downloadable phone book XML Grandstream Networks Inc GXP1400 1405 User Manual Page 5 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF Voice amp Video 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 EE eee auto answer early dial and speed dial ie Network and Provisioning Via keypad LCD Web browser or secure AES encrypted central configuration file manual or dynamic host configuration protocol DHCP network setup i Support NAT traversal using IETF STUN and Symmetric RTP Support for IEEE 802 1p Q tagging VLAN Layer 3 ToS Firmware Support firmware upgrade via TFTP or HTTP Upgrades i Support for Authenticating configuration file before accepting changes User specific URL for configuration file and firmware files Mass provisioning using TR 069 or encrypted XML configuration file te ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee ee rar Advanced Server Features Message waiting indication support DNS SRV Look up and SIP Server Fail O
12. number display occurs after the handset is off hook or handset button is pressed or speaker button is pressed or the line key is selected After dialing the number the phone waits 4 seconds by default No key Entry Timeout before sending and initiating the call Press button to override the 4 second delay Making Calls using IP Addresses Direct IP Call allows two phones to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both phones have public IP addresses or e Both phones are on a same LAN VPN using private or public IP addresses or e Both phones can be connected through a router using public or private IP addresses with necessary port forwarding or DMZ To make a direct IP call please follow these steps e Press MENU button to bring uo MAIN MENU e Select Direct IP Call using the arrow keys e Press OK to select e Input the 12 digit target IP address Please see example below e Press OK key to initiate call For example lf 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 represents the dot the key represents colon Press OK to dial out The GXP1400 1405 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
13. 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 GXP1400 1405 IP address Enter the admin password to access the web configuration interface In the ADVANCED SETTINGS page enter the Upgrade Server s IP address or FQDN in the 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 TFTP HTTP server is not responding there are no code image files available for upgrade or checksum test fails etc the phone will stop the upgrading process and re boot using the existing firmware software Firmware upgrades take around 60 seconds in a controlled LAN or 5 10 minutes over the Internet 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 prov
14. setting Default is No URL for TR 069 Auto Configuration Servers ACS GXP1400 1405 User Manual Page 21 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 Softkey Label Offhook Auto Dial Grandstream Networks Inc andstream Innovative IF 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 do
15. 0ms 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 GXP1400 1405 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 GXP1400 1405 User Manual Page 30 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 requirement
16. 1405 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 cfg000b82023dd4 File does not exist Configuration File Download CONFIGURATION FILE DOWNLOAD The GXP1400 1405 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 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 GXP1400 1405 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 cf9000b820102ab The configuration file name should be in lower cases For more details on XML provisioning please refer to http www grandstream com suppor
17. 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 Table 12 Keypad GUI Flow Grandstream Networks Inc GXP1400 1405 User Manual Page 15 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 I Clear All Missed Calls Transferred Calls Forwarded Calls New Entry Phone Book Name New Entry Number Download Phonebook XML Acct Back Confirm Add Cancel and Return LDAP Directory View Directory Search Configuration Download Directory Search Configuration Back Select Filter Filter Value Back Instant Message Do Not Disturb Clear All Back Enable DND Disable DND Back Preference Do Not Disturb ng ONE Ring Tone gt Default Ring LCD Contrast Ring1 LCD Brightness Ring2 Download SCR XML Ring 3 Erase Custom SCR Back Display Language Back LCD Brightness Active f Idle Config Back SIP Upgrade Fa
18. am Innovative IP Voice amp Video Installation EQUIPMENT PACKAGING Table 1 Equipment Packaging po GXP1400 1405 gt Handset Ves gt Yes Yes 4 CONNECTING YOUR PHONE The connectors of the GXP1400 1405 are located on the bottom of the device Table 2 GXP1400 1405 Connectors PC 10 100Mbps RJ 45 ports for PC downlink connection LAN 10 100Mbps RJ 45 port for LAN uplink connection integrated PoE GXP 1405 only Power Jack 5V DC power port UL Certified Handset Jack RJO Headset Jack RJ9 SAFETY COMPLIANCES The GXP1400 1405 phone complies with FCC CE and various safety standards The GXP1400 1405 power adaptor is compliant with the UL standard Please use the universal power adaptor provided with the GXP1400 1405 package only The manufacturer s warranty does not cover damages to the phone caused by unsupported power adaptors WARRANTY If you purchased your GXP1400 1405 from a reseller please contact the company where you purchased your phone for replacement repair or refund If you purchased the product directly from Grandstream contact your Grandstream Sales and Service Representative for a RMA Return Materials Authorization number before you return the product Grandstream reserves the right to remedy warranty policy without prior notification Grandstream Networks Inc GXP1400 1405 User Manual Page 3 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF Voice amp Vi
19. cate Remove OBP from The SIP Extension notifies the SIP server that it is behind a NAT firewall Route Validate Incoming This configuration selects whether or not the incoming messages should be Messages validated Support SIP Instance ID Selects whether or not SIP Instance ID is supported Grandstream Networks Inc GXP1400 1405 User Manual Page 26 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 NAT Traversal SUBSCRIBE for MWI PUBLISH for Presence Proxy Require Voice Mail UserlD Send DTMF DTMF Payload Type Early Dial Dial Plan Prefix andstream Innovative IP Voice amp Video This parameter activates 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 GXP1400 1405 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 Enabl
20. cted 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 GXP1400 1405 User Manual Firmware version 1 0 1 67 Page 23 of 34 Last Updated 05 2011 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 Grandstream Networks Inc 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 the call waiting feature will be disabled Default is No If set
21. ctory 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 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 802 1Q VLAN Tag Priority value Reset Vian Config Back GXP1400 1405 User Manual Grandstream Networks Inc Firmware version 1 0 1 67 Page 16 of 34 Last Updated 05 2011 andstream Innovative IF Voice amp Video CONFIGURATION VIA WEB BROWSER The GXP1400 1405 embedded Web server responds to HTTP HTTPS GET POST requests Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft s IE Mozilla Firefox and 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 M
22. d 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 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 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
23. day of March to the 1st Sunday of November By default Enable Weather Update is set to Yes If set to 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 This is displayed 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 Default is No By default LCD backlight will light up whenever there is a missed call Grandstream Networks Inc GXP1400 1405 User Manual Page 190f34 Grandstream Networks Inc Page 19 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 HEADSET Key Mode Headset TX gain dB Headset RX gain dB andstream Innovative IP Voice amp Video 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 15 Dev
24. deo Product Overview Table 3 GXP1400 1405 Feature Guide Features GXP1400 1405 LCD Display 128 x 40 pixel Number of Lines 2 Table 4 GXP1400 1405 Key Features in a Glance Features Benefits EE EEE EE pie pepe EE Open Standards SIP RFC3261 TCP IP UDP RTP RTCP HTTP HTTPS ARP RARP Compatibility ICMP DNS A record SRV and NAPTR DHCP both client and server PPPoE TELNET TFTP NTP STUN SIMPLE SIP over TLS 802 1x TR 069 Feature Rich Traditional voice features including caller ID call waiting hold transfer forward block auto dial off hook dial e ee cc gt gt gt x x x gt gt x Advanced Features 2 line keys with dual color LED and 1 SIP account 3 way conference graphic LCD 3 XML programmable context sensitive soft keys 5 navigation keys 8 dedicated buttons for HOLD TRANSFER CONFERENCE VOLUME HEADSET MUTE DND SPEAKERPHONE SEND REDIAL n _ n ccp ee e ee Advanced Functionality Customized downloadable ring tones SRTP SIP over TLS multi language support and XML enabled adjustable positioning
25. dial tone Dial the forwarding number followed by Wait for a dial tone Hang up Cancel Busy Call Forward dial 91 Wait for dial tone Hang up 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 Cancel Delayed Call Forward Dial 93 for a dial tone then hang up CUSTOMIZED LCD SCREEN amp XML GXP1400 1405 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 how to create a downloadable XML phonebook creating a custom idle screen and or reprogramming the soft keys on GXP1400 1405 please visit our website at htto Wwww grandstream com Grandstream Networks Inc GXP1400 1405 User Manual Page 13 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 Configuration Guide andstream Innovative IP Voice amp Video The GXP1400 1405 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 The phone automatically exits MENU mode with an incoming call the phone is off hook or the MENU mode if left idle for 20
26. e Presence feature SIP Extension to notify SIP server that the unit is behind the NAT Firewall When configured user can access messages by pressing MSG button This ID is usually the VM portal access number This parameter specifies the mechanism to transmit DTMF digit There are 3 supported modes in audio which means DTMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO 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 GXP1400 1405 User Manual Page 27 of 34 Grandstream Networks Inc Firmware version 1 0 1 67 Last Updated 05 2011 Dial Plan Delayed Call Forward Wait Time Enable Call Features Call Log Grandstream Networks Inc 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 a xx at least 2 digit numbers b xx only 2 digit numbers c exclude d 3 5 any digit of 3 4 or 5 e 147 any digit of 1 4 or 7 f lt 2 011 gt replace digit 2 with 011 when dialing g the OR operand e Example 1 369 11 1617xxxxxxx 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 diale
27. ed 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 the call is transferred NOTE To transfer calls across SIP domains SIP service providers must support transfer across SIP domains 3 Way Conferencing GXP1400 1405 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 Grandstream Networks Inc GXP1400 1405 User Manual Page 11 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IP Voice amp Video Choose the desired line to join the conference by pressing the corresponding LINE button 2 Cancel Conference f after pressing the CONF button a user decides not to conference anyone press HOLD 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 LINE key 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 M
28. enu 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 NM 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 htto 192 168 0 60 in the address bar of your browser Oo 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
29. ese 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 Language 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 Table 16 SIP Account Settings Account Active Account Name SIP Server
30. essages Message Waiting Indicator A blinking red MWI Message Waiting Indicator indicates a message is waiting Dial into the voicemail box to retrieve the message An IVR will prompt the user through the process of message retrieval Shared Call Appearance SCA The GXP1400 1405 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 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 membe
31. his field allows the user to choose the firmware upgrade method TFTP HTTP or HTTPS Defines the server path for the firmware server lt 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 GXP1400 1405 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 GXP1400 1405 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
32. ice Configuration Settings Advanced Settings Admin Password Layer 3 QoS Layer 2 QoS Local RTP port Use Random Port Keep alive interval Use NAT IP STUN Server Grandstream Networks Inc 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 GXP1400 1405 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
33. ide 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 solarwinds net updates New customerFree cfm Grandstream Networks Inc GXP1400 1405 User Manual Page 32 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF 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 TFIP server and the GXP1400 1405 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 GXP1400 1405 phone boots up it will send TFTP or HTTP request to download configuration file cf9000b82xxxxxx where 000b82xxxxxx is the MAC address of the GXP1400
34. ing the corresponding account LINE button 2 Incoming multiple calls When another call comes in while having an active call the phone will produce a Call Waiting tone stutter tone Answer the incoming call by pressing its corresponding LINE button The current active call will be put on hold Do Not Disturb 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 PS PHONE FUNCTIONS DURING A PHONE CALL Call Waiting Call Hold 1 Hold Placeacall 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 1 Press the MUTE button to enable disable muting the microphone 2 The Line Status Indicator will show LINEx TALKING or LINEx MUTE to indicate whether the microphone is muted Call Transfer GXP1400 1405 supports both Blind and Attended transfer 1 Blind Transfer Press TRANSFER button then dial the number and press the button to complete transfer of active call 2 Attend
35. le hands free speaker Enable Disable handset mode or used as SEND REDIAL Press the round button in the center to enter Keypad Configuration MENU mode when phone is idle Or use it as ENTER key when in Keypad Configuration Press the four navigation keys to move up down left right Adjust volume by pressing or o0808 0 9 Standard phone keypad press key to send call press key to for IVR functions MAKING PHONE CALLS Handset Headset and Speakerphone The GXP1400 1405 allows you to make phone calls via handset headset or speakerphone During the active calls the user can switch between the handset headset and the speakerphone by pressing the corresponding keys on the phone Dual Lines with SIP Account Grandstream Networks Inc GXP1400 1405 User Manual Page 8 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF Voice amp Video GXP1400 1405 can support up to two lines virtually mapped to a SIP account In off hook state select an idle line and the dial tone will be heard To make a call select the line you wish to use The user can switch lines before dialing any number by pressing the LINE button Completing Calls There are FOUR ways to complete a call 1 DIAL To make a phone call Take Handset off hook or press SPEAKER button or press HEADSET button or press an available LINE key to activate speakerphone The line will have a dial tone E
36. ll not send DNS query but use Primary IP or Secondary IP to send sip message if at least one of them are not empty Primary IP This option applies only if Use Configured IP is selected the phone will send DNS query to the Primary IP Insert IP address here Backup IP 1 Insert the first back up IP here Backup IP 2 Insert the second back up IP here SIP Registration This parameter controls sending REGISTER messages to the proxy server The default setting is Yes Unregister on Reboot Default is No If set to Yes the SIP user s registration information will be cleared on reboot Register Expiration This parameter allows user to specify the time frequency in minutes that GXP1400 1405 refreshes its registration with the specified registrar The default interval is 60 minutes The maximum interval is 65 535 minutes about 45 days Local SIP Port This parameter defines the local SIP port used to listen and transmit The default value for Account 1 is 5060 It is 5062 5064 5066 for Account 2 Account 3 and Account 4 respectively SIP Registration Failure Retry registration if the process failed Default is 20 seconds Retry Wait Time SIP T1 Timeout RFC 3261 SIP T1 timer Default is 0 5 second SIP T2 Interval RFC 3261 SIP T2 timer Default is 4 seconds SIP Transport Choose SIP Transport between UDP and TCP Default is UDP Check Domain Enable to check the domain certificate Default is No Certifi
37. log messages are sent based on the following events e product model version on boot up INFO level e NAT related info INFO level e sent or received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e Ethernet link up INFO level e SLIC chip exception WARNING and ERROR levels e memory exception ERROR level The Syslog uses USER facility In addition to standard Syslog payload it contains the following components GS LOG device MAC addressjferror 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 NTP 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 GXP1400 1405 will ONLY use selected ring tones for particular Caller IDs For all other calls the GXP1400 1405 will use System Ring Tone When sele
38. 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 GXP1400 1405 User Manual Page 20 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 Grandstream Networks Inc andstream Innovative IF 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 T
39. nter the phone number Press or HANDSET button to send REDIAL To redial the last dialed phone number Take Handset off hook or press the SPEAKER button or press an available LINE key to activate speakerphone or on idle screen Press the REDIAL soft key Via CALL HISTORY To call a phone number in the phone s history Press the MENU button to bring up the Main Menu Select Call History and then Answered Calls Missed Calls or Dialed Calls or etc depending on your needs Select phone number using the arrow keys Press OK to select Select and press Dial to dial out Via PHONEBOOK To Call a phone in from the phone s phonebook Go to the phonebook by pressing the DOWN arrow key or pressing the menu button and selecting Phone Book Select the phone number by using the arrow keys Press OK to select Select and press Dial to dial out VIA PAGE INTERCOM Server PBX has to support Page Intercom Also GXP1400 1405 and PBX have to be configured correctly Take Handset off hook or press SPEAKER button or press HEADSET button or press an available LINE key to activate speakerphone Press OK and the screen will display LINEx PAGE Dial the number to Page Intercom Press SEND button to dial out Grandstream Networks Inc GXP1400 1405 User Manual Page 9 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF Voice amp Video NOTE Dial tone and dialed
40. o 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 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 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 Sun
41. ow on the LCD 333 3333 if the MAC address is 00068200e395 it should be key in as 0002228200333395 NOTE If there are digits like 22 in the MAC you need to type 2 then press gt right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2 Step 3 Press the OK button to move the cursor to OK Press OK button again to confirm If the MAC address is correct the phone will reboot Otherwise it will exit to previous keypad menu interface Grandstream Networks Inc GXP1400 1405 User Manual Page 34 of 34 Firmware version 1 0 1 67 Last Updated 05 2011
42. phone status running or stopped Download core dump file for troubleshooting when necessary Table 14 Device Configuration Settings Basic Settings End User Password IP Address Grandstream Networks Inc This contains the password to access the Web Configuration Menu This field is case sensitive with a maximum length of 25 characters The GXP1400 1405 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 GXP1400 1405 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 GXP1400 1405 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 GXP1400 1405 User Manual Page 18 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 Disable Missed Call Backlight andstream Innovative IF Voice amp Vide
43. r 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 the phone In addition they would need to navigate to Settings gt Basic Settings on the web UI and set the line to Shared Line 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 GXP1400 1405 supports traditional and advanced telephony features including caller ID caller ID w name call forward transfer park hold as well as intercom paging Table 10 GXP1400 1405 Call Features Key Call Features 30 Block Caller ID for all subsequent calls Grandstream Networks Inc GXP1400 1405 User Manual Page 12 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 31 67 82 70 71 72 73 90 91 92 93 An Innovative IP Voice amp Video Send Caller ID for all subsequent calls Block Caller ID per call Send Caller ID per call Disable Call Waiting per Call Enable Call Waiting per Call 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 Cancel Unconditional Call Forward dial 73 and get the dial tone then hang up LCD will display Call FWD Activated Busy Call Forward Dial 90 for a
44. s 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 phone remotely The web browser will then display a message window to confirm that reboot is underway Wait 30 seconds to log in again Grandstream Networks Inc GXP1400 1405 User Manual Page 31 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 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 the ADVANCED SETTINGS configuration page FIRMWARE UPGRADE THROUGH TFTP HTTP To upgrade via TFTP or HTTP select TFTP or HTTP upgrade method Upgrade Server needs to be set to a valid URL of a HTTP server Server name can be in either FQDN or IP address format Here are examples of some valid URLs e firmware mycompany com 6688 Grandstream 1 2 3 5 e 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
45. 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 em fpeseription Displays histories of answered dialed missed and transferred and forwarded Call History Status Phone Book LDAP Directory Instant Messages Direct IP Call Preference calls Displays the network status account status software 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 lt or to hear and adjust the ring tone volume LCD Contrast Press or to adjust the LCD contrast LCD Brightness Press or to adjust the LCD brightness for active idle screen 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
46. shows a list of the missed SOFTKEYS 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 Table 8 LCD Icons Odeon So LCD Icon Definitions SIP Registration Status Icon Solid connected to SIP Server IP address received Blank SIP Proxy Server not registered Phone Status Icon OFF when the handset is on hook ON when the handset is off hook Speaker Phone Status Icon OFF when the speakerphone is off ON when the speakerphone is on DND Icon OFF when Do Not Disturb is disabled ON when Do Not Disturb is enabled Grandstream Networks Inc GXP1400 1405 User Manual Page 7 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IP Voice amp Video Calls Forwarded Icon INDICATES calls are forwarded Follow call forwarding procedures MUTE Icon INDICATES call is in MUTE during the call Table 9 GXP1400 1405 KEYPAD BUTTONS Key Button Key Button Definitions 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 LINE 1 LINE 2 Switch between Line 1 and Line 2 Mute an active call or use as DND button when the phone is in idle state Press HEADSET key to answer hang up phone calls while using headset It also allows user to toggle between headset and speaker Enable Disab
47. t Managing Firmware and Configuration File Download When Automatic Upgrade is set to Yes a Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check for upgrades at pre scheduled time intervals By defining different intervals in P193 for different devices a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time Grandstream Networks Inc GXP1400 1405 User Manual Page 33 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IF 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 Example A B C D F 0 9 22 press the 2 key twice A will show on the LCD 222 2222 33 press the 3 key twice D will sh
48. timer request If set to Yes the phone will use session timer even if the remote party does not support this feature If set to No the session timer is enabled only when the remote party supports this feature To turn off Session Timer select No for Caller Request Timer Callee Request Timer and Force Timer As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy server as the refresher As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is required to support PSTN inter networking There are 4 uniquely defined ring tones e One 1 System Ring Tone when selected all calls 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
49. 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 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 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 GXP1400 1405 User Manual Firmware version 1 0 1 67 Page 24 of 34 Last Updated 05 2011 Display Language andstream Innovative IF Voice amp Video Allows user to choose preferred display language in web UI and key pad Ul Currently the phone supports these languages English Simplified Chinese Traditional Chinese Korean Japan
50. ver Support customizable idle screen via downloading XML by HTTP TFTP Security User and administrator level passwords MD5 and MD5 sess based authentication AES based secure configuration file SRTP TLS 802 1x media access control Grandstream Networks Inc GXP1400 1405 User Manual Page 6 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 andstream Innovative IP Voice amp Video Using the GXP1400 1405 GETTING FAMILIAR WITH THE LCD GXP1400 1405 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 LCD Button Definitions DATE AND TIME 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 LOGO NAME ee i t customization The maximum size for logo name is 22 characters in English Shows the status of network in the middle of the screen It will indicate whether the NETWORK STATUS network is down or starting STATUS BAR Shows the status of the phone using icons as shown in the next table 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
51. vnennunnennvnnnnunnennenennunnennennenennunnenneneenenn 32 CONFIGURATION FO Wy De 33 RESTORE FACTORY DEFAULT SETTING essesssssssessessssessessosessecsessossosessessessosessessosessessessosessessessssessessesesse 34 TABLE OF TABLES GXP1400 1405 USER MANUAL Table 1 Equipment Packaging 3 Table 2 GXP1400 1405 Connectors 3 Table 3 GXP1400 1405 Feature Guide 4 Table 4 GXP1400 1405 Key Features in a Glance i 4 Table 5 GXP1400 1405 Hardware Specifications 4 Table 6 GXP1400 1405 Technical SpecificatioNS 5 Table 7 LCD ButtonS ee 7 Table 8 LCD ICONS NG 7 Table 9 GXP1400 1405 KEYPAD BUTTONS 8 Table 10 GXP1400 1405 Call Features i 12 Table 11 Key Pad Configuration MENU 14 Table 12 Keypad GUI FIOW ronnnnnrnrrnnnnnrvnrrnnnnnornnrnnnnnrrnnrnnnnnsnnnnnnnnnsennnnnnsnsrnnnnnnsnsnnnnnnnsnsennnnnnsnsnen 15 Table 13 Device Configuration Status i 18 Table 14 Device Configuration Settings Basic Settings rrrrrrnnnnrrrrnnrrovrnnnnrnnnnnnrnnnnnnnennnnnne 18 Table 15 Device Configuration Settings Advanced Settings ccccccsssescceeseeeeeeeeeeeeeeeeeeeeeas 20 Table 16 SIP Account Settings iii 25 Grandstream Networks Inc GXP1400 1405 User Manual Page 1 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 E sen Innovative IF Voice
52. 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 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 x1
53. wnloaded 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 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 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 GXP1400 1405 User Manual Page 22 of 34 Firmware version 1 0 1 67 Last Updated 05 2011 Syslog Server 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 Grandstream Networks Inc andstream Innovative IF Voice amp Video The IP address or URL of System log server This feature is especially useful for ITSPs Select the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Sys
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