Home

Grandstream Networks, Inc.

Contents

1. 30 STATUS PAGE DEFINITIONS 2000000000000000000 21 ACCOUNT PAGE DEFINITIONS 220200000000 31 SETTINGS BASIC SETTINGS 2 02000000000000000 39 SETTINGS ADVANCED SETTINGS 43 5 5 50 PUBLIG MODE 50 EDITING CONTACTS AND CLICK TO DIAL 2 51 UPGRADING AND PROVISIONING 55 UPGRADE VIA KEYPAD MENU 55 UPGRAGE VIA WEB 55 NO LOCAL TFTP HTTP SERVERS 56 CONFIGURATION FILE 56 RESTORE FACTORY DEFAULT 58 EXPERIENCING THE GXP1160 GXP1165 59 Table of Tables GXP1160 GXP1165 User Manual Table 1 GXP1160 GXP1165 TECHNICAL 8 8 Table 2 GXP1160 GXP1165 EQUIPMENT PACKAGING esee nennen 10 Table 3 GXP1160 GXP1165 6 4 0 11 Table 4 GXP1160 GXP1165 DI
2. Innovative IP Voice amp Video Grandstream Networks Inc GXP1160 GXP1165 Small Medium Business IP Phone GXP1160 GXP1165 USER MANUAL Innovative Voice amp Video GXP1160 GXP1165 User Manual Index GNU GPL 8 7 5 CHANGE LOG aciadedsmwsdsscediadiadecunsdestaauidadedwadsiadaeuna 6 FIRMWARE VERSION 1 0 5 2 6 5 Ie Oe rae 7 PRODUCT 2 10 11111 RR RR RR saa Dua 8 FEATURE HIGHT LIGHTS 0cc cccccccscccecccsaccececnoneccesceceaccusaccesccceneccssucaudctesccadesbeneccustececccnenscessuancestesses 8 GXP1160 GXP1165 TECHNICAL 5 8 lt lt 2 lt gt lt lt lt 10 EQUIPMENT 10 CONNECTING YOUR 0 10 SAFETY COMPLIANCE S ccc ccccccsscccccccosccccscececcccseccecccccneccusenceccstenecesceaeeccssdenaeecseeeecesseccesseseeccesseees 11 WARBRAWN EY E 11 USING THE 1160 1165 0 42 4 13 GETTING FAMILAR WITH THE 13 GETTING FAMILAR WITH THE 14 MAKING PHONE 15 HAN
3. Link Up Service Status Core Dump ACCOUNT PAGE DEFINITIONS Account Name SIP Server Global unique ID of device in HEX format The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device The IPv4 address obtained on the phone The IPv6 address obtained on the phone Product model of the phone Product part number e boot boot version number e core version number e base base version number e prog program version number This is the main firmware release number which is always used for identifying the software system of the phone e dsp DSP version number System up time since the last reboot Current system time on the phone system SIP account registration status PPPoE connection status GUI and Phone service status running or stopped Core dump file that could be downloaded for troubleshooting purpose The name associated with the SIP account The URL or IP address and port of the SIP server This is provided by your VoIP service provider ITSP FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 31 of 59 Secondary SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name DNS Mode TEL URI SIP Registration Unregister On Reboot Register Expiration Innovative IP Voice amp Video The URL or IP address and port of
4. If set to Yes the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep based on the SIP info header sent from the server proxy The default setting is If set to Yes the Refer To header uses the transferred target s Contact header information for attended transfer The default setting is No Defines whether or not the call is transferred to the other party if the initiator of the conference hangs up The default setting is If set to Yes SIP User ID will be checked in the Request URI of the incoming INVITE If it doesn t match the phone s SIP User ID the call will be rejected The default setting is Defines whether the phone will challenge INVITE requests or not When set to Yes the phone will challenge the INVITE for authentication with FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 37 of 59 Preferred Vocoder SRTP Mode Symmetric RTP Silence Suppression Voice Frames Per TX No Key Entry Timeout s Use as Dial Key G723 Rate G 726 32 Packing Mode iLBC Frame Size iLBC Payload Type Jitter Buffer Type Jitter Buffer Length Gorse Innovative IP Voice SIP 401 Unauthorized response The PBX will need resend the SIP INVITE request with authentication credentials The default setting is No 7 different vocoder types are supported on the phone including G 711 U law PCMU 9 711 A law G 723
5. DO NOT DISTURB Do Not Disturb can be enabled disabled in Menu gt Preference e Press the Menu button and select Preference using navigation keys e Press Menu button again to get into Preference options e Select Do Not Disturb and press Menu button e Use UP and DOWN arrow keys to select and press Menu button to enable or disable Do Not Disturb feature When Do Not Disturb feature is turned on the DND icon will appear on the right side of the LCD The incoming call will not be accepted or directly go into voicemail DURING A PHONE CALL CALL WAITING CALL HOLD FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 19 of 59 E sen Innovative IP Voice amp Video e Hold Place a call on hold by pressing the HOLD key e Resume Resume call by pressing the HOLD key j again e Multiple calls Automatically place active call hold or switch between two calls by pressing the FLASH key 7 Call waiting tone stutter tone will be audible on incoming call during the active call MUTE During an active call press the MUTE softkey to mute unmute the microphone The LCD will show LINEx TALKING or LINEx MUTE to indicate the mute status with Mute icon displayed on the right side of the screen CALL TRANSFER GXP1160 GXP1165 supports Blind Transfer Attended Transfer and Auto Attended Transfer e Blind Transfer gt During the first active call press TRANSFER key and dial the num
6. innovative IP Voice 8 Video searching Instant Messages Displays received instant messages Direct IP Call Makes direct IP call Preference Preference sub menu includes the following options Do Not Disturb Enables disables Do Not Disturb on the phone Forward Call Configures call forward feature on selected account forward type and number Ring Tone Configures different ring tones for incoming call Ring Volume Adjusts ring volume by pressing left right arrow key LCD Contrast Adjusts LCD contrast by pressing left right arrow key Download SCR XML Triggers the phone to download the XML idle screen file immediately The XML idle screen server path and downloading method need to be set up correctly in Web GUI gt Advanced Settings Erase Custom SCR Erases custom XML idle screen previously loaded on the phone After erasing it the phone will show default idle screen Display Language Selects the language to be displayed on the phone Users could select Automatic for local language based on IP location if available Time Settings Configures date and time on the phone Config Config sub menu includes the following options SIP Configures SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport and Audio information to register SIP account on the phone FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 27 of 59 Factory Functions Network Call Features Voice Mails
7. gt handset off hook or Press Speaker button or Press Headset softkey with headset plugged in gt You shall hear dial tone after off hook gt Press MENU button to switch the call screen from Line x Caller DIAL to Line x Caller Paging gt Enter the number gt Press SEND key or to dial out Note After entering the number the phone waits for the No Key Entry Timeout Default timeout is 4 seconds configurable via Web GUI before dialing out Press SEND key or key to override the No Key Entry Timeout If digits have been entered after handset is off hook the SEND key will works as SEND instead of REDIAL By default can be used as SEND to dial the number out Users could disable it by setting Use as Dial Key to No from Web GUI gt Account page For Paging Intercom if the SIP Server PBX supports the feature and has Paging Intercom feature code set up already users might not necessarily need toggle to paging mode in the call screen on GXP1160 GXP1165 Simply dial the feature code with extension as a normal call 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 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 17 of 59 E sen Innovative IP Voice amp Video can be made between two phones if e Both phones have public IP addresses or e Both phones are on the same LAN VPN using private
8. Reboot Exit Innovative IP Voice Upgrade Configures firmware server and config server for upgrading and provisioning the phone Factory Reset Resets the phone to factory default settings Layer 2 QoS Configures 802 1Q VLAN Tag and priority value Factory Functions sub menu includes the following options Audio Loopback Speak to the phone using speaker handset headset If you can hear your voice your audio is working fine Press Menu button to exit audio loopback mode Diagnostic Mode All LEDs will light up Press any key except MENU key on the keypad to display the button name in the LCD Lift and put back the handset or press Menu button to exit diagnostic mode Keyboard Diagnostic Press all the available keys on the phone The LCD will display the name for the keys to be pressed to finish the keyboard diagnostic mode Selects IP mode DHCP Static IP PPPoE Configures PPPoE account ID and password Configures IP address Netmask Gateway DNS Server 1 and DNS Server 2 Configures 802 1x mode Configures call forward features for Forward All Forward Busy Forward No Answer and No Answer Timeout Displays voicemail message information in the format below new messages all messages urgent messages all urgent messages Reboot the phone Exit from this menu The following picture shows the keypad MENU configuration flow on GXP1160 GXP1165 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165
9. 2 GXP1160 GXP1165 Keypad MENU Flow FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 29 of 59 E sen Innovative IP Voice amp Video CONFIGURATION VIA WEB BROWSER The GXP1160 GXP1165 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 To access the GXP1160 GXP1165 Web GUI 1 Connect the computer to the same network as the phone 2 Make sure the phone is turned on and shows its IP address You may check the IP address by pressing NextScr softkey or go to MENU gt Status 3 Open Web browser on your computer 4 Enter the phone s IP address in the address bar of the browser 5 Enter the administrator s login and password to access the Web Configuration Menu Note e The computer has to be connected to the same sub network as the phone This can be easily done by connecting the computer to the same hub or switch as the phone connected to In absence of a hub switch or free ports on the hub switch please connect the computer directly to the PC port on the back of the phone e f the phone is properly connected to a working Internet connection the IP address of the phone will display in MENU gt Status This address has the format xxx xxx xxx xxx where xxx stands for a number from 0 255 Users will need this number to access the Web GUI For example if the
10. IPv6 Address IPv6 Prefix Length IPv6 Prefix DNS Server 1 DNS Server 2 Preferred DNS server 802 1x mode Identity 802 1x Secret Private Key Password 802 1x CA Certificate 802 1x Client Certificate HTTP Proxy HTTPS Proxy Gorse Innovative IP Voice Enables DHCP Option 120 from local server to override the SIP Server on the phone The default setting is Enter the PPPoE account ID Enter the PPPoE Password Enter the PPPoE Service Name Enter the IP address when static IP is used Enter the Subnet Mask when static IP is used for IPv4 Enter the Default Gateway when static IP is used for IPv4 Enter the DNS Server 1 when static IP is used for IPv4 Enter the DNS Server 2 when static IP is used for IPv4 Enter the Preferred DNS Server for IPv4 Allows users to configure the appropriate network settings on the phone to obtain IPv6 address Users could select Auto configured or Statically configured for the IPv6 address type Enter the static IPv6 address when Full Static is used in Statically configured IPv6 address type Enter the IPv6 prefix length when Full Static is used in Statically configured IPv6 address type Enter the IPv6 Prefix 64 bits when Prefix Static is used in Statically configured IPv6 address type Enter the DNS Server 1 for IPv6 Enter the DNS Server 2 for IPv6 Enter the Preferred DNS Server for IPv6 Allows the user to set 802 1x mode on the phone The default v
11. Telecom Mode to use China Telecom special FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 49 of 59 Do Not Escape as 9623 in SIP URI Disable Telnet PC Port Mode Display Language Download Device Configuration NAT SETTINGS Innovative IP Voice amp Video features on the phone Specifies whether to replace by 23 or not for some special situations The default setting is No Disables Telnet access The default setting is No Configures the PC port mode The default setting is Enabled When set to Disabled the PC port is turned off When set to Mirrored the traffic in the LAN port will go through PC port as well so users could capture phone s trace by connecting a PC to the phone s PC port Selects display language on the phone Click to download the device configuration file in txt format If the devices are kept within a private network behind a firewall we recommend using STUN Server The following settings are useful in the STUN Server scenario e STUN Server under Advanced Settings page Enter a STUN Server IP or FQDN that you may have or look up a free public STUN Server on the internet and enter it on this field If using Public IP keep this field blank e Use Random Ports under Advanced Settings page This setting depends on your network settings When set to Yes it will force random generation of both the local SIP and RTP ports This is usually necessary
12. established Cancel Conference gt If after press the CONFERENCE key 4 the user decides to conference press Cancel softkey FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 21 of 59 Innovative Voice amp Video gt This will resume the 2 way conversation with the current line e Split and Re conference gt During the 3 way conference press HOLD key The conference call will be split and both calls will be put on hold separately gt Press FLASH key to resume the 2 way conversation with the second established call gt If users would like to re establish conference call press the ReConf softkey e End Conference gt Press HOLD key to split the conference call The conference call will be ended with both calls on hold Or gt Users could press the EndCall softkey or simply hang up the call to terminate the conference call GXP1160 GXP1165 supports Easy Conference Mode which can be used combined with the traditional way to establish the conference e Initiate a conference call gt Establish 1 call Press CONFERENCE key j and a new line will be brought up using the same account gt Dial the number and press SEND key to establish the second call gt Press CONFERENCE key j or press the ConfCall softkey to establish the conference e Split and Re conference gt During the 3 way conference press HOLD key Cu The conference c
13. format on the LCD The default setting is in 12 hour format When it s set to Yes the DTMF digits entered during the call will not display The default setting is No Configures to enable or disable the speaker to ring when headset is used on Toggle Headset Speaker mode If set to Yes when the phone is in Headset Toggle Headset Speaker mode both headset and speaker will ring on incoming call The default setting is When headset is connected to the phone users could use the HEADSET button in Default Mode or Toggle Headset Speaker e Default Mode gt When the phone is in idle press HEADSET button to off hook the phone and making calls by using headset Headset icon will display on the left side of the screen in dialing talking status gt When there is an incoming call press HEADSET button to pick up the call using headset gt When there is an active call using headset press HEADSET button to hang up the call When Speaker Handset is being used in dialing talking status press HEADSET button to switch to headset Press it again to hang up the call Or press speaker Handset to switch back to the previous mode e Toggle Headst Speaker gt When the phone is in idle press HEADSET button to switch to Headset mode The headset icon will display on the left side of the screen In this mode if pressing Speaker button or Line key to off hook the phone headset will be used gt When there is an active ca
14. information service e g local weather personalized music sing tone ring back tone e Dual switched auto sensing 10 100Mbps network ports integrated PoE GXP1165 only e Automated provisioning using TR 069 or AES encrypted XML configuration file SRTP and TLS for advanced security protection 802 1x for media access control GXP1160 GXP1165 TECHNICAL SPECIFICATIONS Protocols and Standards Network Interfaces Graphic Display Feature Keys Voice Codec Telephony Features Headset Jack Base Stand Table 1 GXP1160 GXP1165 TECHNICAL SPECIFICATIONS SIP RFC3261 TCP IP UDP RTP RTCP HTTP HTTPS ARP RARP ICMP DNS A record SRV NAPTR DHCP PPPoE NTP STUN SIMPLE TR 069 802 1x 6 Dual switched 10 100Mbps ports integrated PoE GXP1165 only 128 x 40 graphical LCD display 1 SIP account 3 XML programmable context sensitive soft keys 5 Navigation Menu Volume keys 9 dedicated function keys for PHONEBOOK MESSAGE with LED indicator HOLD TRANSFER CONFERENCE FLASH SPEAKERPHONE VOLUME SEND REDIAL Support for G 723 1 G 729A B G 711u a G 726 32 G 722 wide band iLBC in band and out of band DTMF in audio RFC2833 SIP INFO Hold transfer forward 3 way conference downloadable phone book XML LDAP up to 500 items call waiting call log up to 200 records off hook auto dial auto answer click to dial flexible dial plan hot desking personalized music ringtones server redund
15. is in idle enter the number to be dialed out Take handset off hook or Press Speaker button or Press Headset softkey with headset plugged in gt The call will be dialed out e Off hook and dial Off hook the phone enter the number and send out gt Take handset off hook or Press Speaker button or Press Headset softkey with headset plugged in You shall hear dial tone after off hook Enter the number Press SEND key or to dial out e Redial Redial the last dialed number gt handset off hook or Press Speaker button or Press Headset softkey with headset plugged in or When the phone is in idle gt Press SEND key Ce or the REDIAL softkey e Via Call History Dial the number logged in phone s call history gt Press MENU button to bring up the main menu gt Enter Call History and select Answered Calls Missed Calls Transferred Calls or Forwarded Calls gt Select the entry you would like to call using the navigation UP and DOWN arrow keys FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 16 of 59 sen Innovative IP Voice amp Video gt Press SEND key to dial out Via Phonebook Dial the number from the phonebook gt Press MENU button to bring up the main menu gt Select and enter Phonebook gt Select the phonebook entry you would like to call using the navigation UP and DOWN arrow keys gt Press SEND key to dial out Via Page Intercom
16. mask XX or X are also valid so leading 0 is not required but OK No SIP server is required to make quick IP call The default setting is Disables the Conference function The default setting is Disables the Transfer function The default setting is If set to Yes the phone will use attended transfer by default The default setting is Configures the number for the phone to dial as DTMF during the call using TRANSFER button Configures the access control for the users to configure from keypad Menu There are three different options e Unrestricted All the options can be accessed in keypad Menu Basic settings only The CONFIG option will not display for users to access in keypad Menu e Constraint Mode CONFIG FACTORY FUNCTIONS NETWORK options will not display for users to access in keypad menu If set to Yes the keypad can be locked by pressing and holding the STAR key for about 4 seconds A lock icon will show indicating the keypad is locked The default setting is Yes Note When the keypad is locked users would need press and hold the STAR key for about 4 seconds again and then enter the password to unlock it Configures the password to lock unlock the keypad The password field allows number with up to 32 characters If configured when the phone is on hook it will go off hook after the timeout in seconds The default value is 30 seconds Enables Disables China
17. 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 the steps below e Press MENU button to bring up main menu e Select Direct IP Call using the navigation arrow keys e Press MENU to enter the Direct IP Call mode e Input the 12 digit target IP address Please see example below e Press the More softkey to make sure the softkey selection IPv4 or IPv6 is correctly selected depending on your network environment e Press OK softkey to dial For example If the target IP address is 192 168 1 60 and the port is 5062 i e 192 168 1 60 5062 input the following 192 168 1 60 5062 The key represents the dot the key represents colon Wait for about 4 seconds and the phone will initiate the call Quick IP Call Mode The GXP1160 GXP1165 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 address This is possible only if both phones are 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 Call Mode go to GXP1160 GXP1165 Web GUI gt Advanced Setting page set Use Quick IP Call Mode to Yes Click on Update on the bottom of the Web GUI page to take the change To make Quick I
18. under the GNU General Public License GPL Grandstream uses software under the specific terms of the GPL Please see the GNU General Public License GPL for the exact terms and conditions of the license Grandstream GNU GPL related source code can be downloaded from Grandstream web site from http www grandstream com support faq qnu_qpl FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 5 of 59 Innovative Voice amp Video CHANGE LOG This section documents significant changes from previous versions of GXP1160 GXP1165 user manuals Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here FIRMWARE VERSION 1 0 5 2 e This is the initial version FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 6 of 59 E sen Innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream GXP1160 GXP1165 Small Medium Business IP Phone GXP1160 GXP1165 is a next generation small to medium business IP phone that features single SIP account up to 2 call appearances a 128 x 40 graphical LCD 3 XML programmable context sensitive soft keys dual network ports with integrated PoE GXP1165 only 3 way conference and Electronic Hook Switch EHS with Plantronics headset The GXP1160 1165 delivers superior audio quality rich and leading edge telephony features personalized information and customizable application service
19. up the keypad configuration menu e Select Config and enter e Select Factory Reset e Awarning window will pop out to make sure a reset is requested and confirmed e Press the OK softkey to confirm and the phone will reboot To cancel the Reset press Cancel softkey instead FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 58 of 59 Innovative IP Voice amp Video EXPERIENCING THE GXP1160 GXP1165 Please visit our website http Awww grandstream com to receive the most up to date updates on firmware releases additional features FAQs documentation and news on new products We encourage you to browse our product related documentation FAQs and User and Developer Forum for answers to your general questions If you have purchased our products through a Grandstream Certified Partner or Reseller please contact them directly for immediate support Our technical support staff is trained and ready to answer all of your questions Contact a technical support member or submit a trouble ticket online to receive in depth support Thank you again for purchasing Grandstream IP phone it will be sure to bring convenience and color to both your business and personal life FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 59 of 59
20. 000 3 2 EN61000 3 3 EN60950 1 AS NZS CISPR 22 Class B AS NZS CISPR 24 RoHS UL 60950 power adapter FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 9 of 59 ream Innovative IP Voice amp Video INSTALLATION EQUIPMENT PACKAGING Table 2 GXP1160 GXP1165 EQUIPMENT PACKAGING Main Case Yes 1 Handset Yes 1 Phone Cord Yes 1 Power Adaptor Yes 1 Ethernet Cable Yes 1 Phone Stand Yes 1 Quick Start Guide Yes 1 CONNECTING YOUR PHONE Power Port Headset Port Handset Port Figure 1 GXP1160 GXP1165 Ports FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 10 of 59 E sen Innovative IP Voice amp Video Table 3 GXP1160 GXP1165 CONNECTORS Handset Port RJ9 handset connector port Headset Port RJ9 headset connector port supporting EHS Electronic Hook Switch with Plantronics headsets LAN Port 10 100Mbps 45 port connecting to Ethernet integrated PoE GXP1165 only PC Port 10 100Mbps RJ 45 port for PC connection Power Jack 5V DC Power connector port To set up the GXP1160 GXP1165 follow the steps below 1 Attach the phone stand to the back of the phone where there are slots 2 Connect the handset and main phone case with the phone cord 3 Connect the LAN port of the phone to the RJ 45 socket of a hub switch or a router LAN side of the router using the Ethernet cable 4 Connect the 5V DC output plug to the power jack on the phone plug the power adapter into an elect
21. 1 G 729A B G 722 wide band G72 32 Users can configure vocoders a preference list that is included with the same preference order in SDP message Enables the SRTP mode based on your selection The default setting is Disabled Defines whether symmetric RTP is supported or not The default setting is No 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 The default setting is No Configures the number of voice frames transmitted per packet When configuring this it should be noted that the ptime value for the SDP will change with different configurations here This value is related to the codec used and the actual frames transmitted during the in payload call For end users it is recommended to use the default setting as incorrect settings may influence the audio quality Defines the timeout in seconds for no key entry If no key is pressed after the timeout the digits will be sent out The default value is 4 seconds Allows users to configure the key as the Send key If set to Yes the key will immediately dial out the input digits In this case this key is essentially equivalent to the Send key If set to No the key is included as part of the dialing string Sele
22. 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 Example of a simple dial plan used in a Home Office in the US 71900 lt 1617 gt 2 9 xxxxxx 1 2 9 2 9 011 2 9 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 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 allows international calls starting with 011 e 3469 11 allows 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 Defines the timeout in seconds before the call is forwarded on no answer The default value is 20 seconds When enabled Do No Disturb Call Forward and other call features will be supported locally provided ITSP supp
23. DSET SPEAKER AND HEADSET 2 2 040 000000800500000 15 2 CALLS WITH 1 2 2 1 72222000000000000 15 COMPLETING 16 2 2 00000 00000 rennen serene 16 MAKING CALLS USING 55 65 20200020 2 21 71211100000 17 ANSWERING PHONE 19 RECEIVING CALLS EMEN 19 DO 5 2 20202000000000 19 DURING A PHONE 19 CALL WAITING CALL HOLD 2 2000 2 070000000 19 MOTE P tae PA EM EUN 20 CALL TRANSFER 2 2 2727012100000 20 3 WAY CONFERENCING 1 1 010000050 21 VOICE MESSAGES MESSAGE WAITING INDICATOR esses sese 23 GALL FEATURES 23 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 2 of 59 Innovative IP Voice amp Video CUSTOMIZED LCD SCREEN 22 8 25 CONFIGURATION 26 CONFIGURATION VIA 0 40 1 050000060 a E a aa a 26 CONFIGURATION VIA WEB BROWSER dass ds sd nna 30
24. 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 First Sunday of April to the 1st Sunday of November Configures to enable or disable weather update on the phone The default setting is Yes If set to No the weather information screen will not show Configures weather city code for the phone to look up the weather information The default setting is Automatic and the weather information will be obtained based on the IP location of the phone if available Otherwise specify the self defined city code For example USCA0638 is the city code for Los Angeles CA United States Specifies the weather update interval in minutes The default value is 15 minutes Specifies the degree unit for the weather information to display on the phone Configures the LCD contrast level from 0 to 20 The default value is 10 Configures the date display format on the LCD The following formats are supported e 2012 07 02 e mm dd yyyy 07 02 2012 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 41 of 59 Time Display Format Disable in call DTMF Display Always Ring Speaker Headset Key Mode Write Timeout Max Unsaved Log Headset TX gain e dd mm yyyy 02 07 2012 Configures the time display in 12 hour or 24 hour
25. E key 2 Voice Mail User ID has to be properly configured as the voice mail number under Web GUI gt Account page An IVR will prompt the user through the process of message retrieval CALL FEATURES The GXP1160 GXP1165 supports traditional and advanced telephony features including caller ID caller ID with caller Name call forward and etc Table 7 CALL FEATURES 30 Block Caller ID for all subsequent calls e Off hook the phone e Dial 30 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 23 of 59 31 67 82 70 71 72 78 90 91 C iin Innovative IP Voice Send Caller ID for all subsequent calls e Off hook the phone e Dial 31 Block Caller ID per call e Off hook the phone Dial 67 and then enter the number to dial out Send Caller ID per call e Off hook the phone e Dial 82 and then enter the number to dial out Disable Call Waiting per Call e Off hook the phone e Dial 70 and then enter the number to dial out Enable Call Waiting per Call e Off hook the phone e Dial 71 and then enter the number to dial out Unconditional Call Forward To set up unconditional call forward e Off hook the phone Dial 72 and then enter the number to forward the call e Press OK softkey or SEND key Cancel Unconditional Call Forward To cancel the unconditional call forward e Off hook the phone e Dial 73 e Hang up the call B
26. IRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 12 of 59 Innovative Voice amp Video USING THE GXP1160 GXP1165 GETTING FAMILAR WITH THE LCD GXP1160 GXP1165 has a dynamic and customizable screen The screen displays differently depending on whether the phone is idle or in use active The following table describes the items displayed on the GXP1160 GXP1165 idle screen DATE AND TIME LOGO NAME NETWORK STATUS STATUS BAR SOFTKEYS V Table 4 GXP1160 GXP1165 DISPLAY DEFINITIONS Displays the current date and time It can be synchronized with Internet time servers Displays company logo name This logo name can be customized via xml screen customization The maximum size for logo name is 26 characters in English approximately Shows the status of network in the middle of the screen It will indicate whether the network is down or starting Shows the status of the phone for registration status call features and etc 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 softkeys are e NextScr Toggles among idle screen weather information IP Address and extension number e Headset Onhook offhook using headset or toggle to headset mode e FwdAIl Unconditionally forwards the calls to another number e Missed Shows unanswered calls to this phone e Redial Redials the las
27. P Call take the phone off hook first Then dial xxx where x is 0 9 and xxx 255 Press or SEND and a direct IP call to aaa bbb ccc XXX will be completed aaa bbb ccc is from the local IP address regardless of subnet mask The number xx or x are also valid The leading 0 is not required but it s OK For example e 192 168 0 2 calling 192 168 0 3 dial 3 followed by or SEND e 192 168 0 2 calling 192 168 0 23 dial 23 followed by SEND FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 18 of 59 E sen Innovative IP Voice amp Video e 192 168 0 2 calling 192 168 0 123 dial 123 followed by SEND 192 168 0 2 dial 3 and 03 and 003 results in the same call call 192 168 0 3 Note e The 4 will represent colon in direct IP call rather than SEND key as in normal phone call e f you have SIP server configured direct IP call still works If you are using STUN direct IP call will also use STUN e Configure the Use Random Port to No when completing direct IP calls ANSWERING PHONE CALLS RECEIVING CALLS e Single incoming call Phone rings with selected ring tone Answer call by taking handset off hook or using Speaker Headset e Multiple incoming 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 the FLASHING key The current active call will be put on hold automatically
28. RMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 54 of 59 Innovative Voice amp Video UPGRADING AND PROVISIONING The GXP1160 GXP1165 can be upgraded via TFTP HTTP HTTPS by configuring the URL IP Address for the TFTP HTTP HTTPS server and selecting a download method Configure a valid URL for TFTP or HTTP the server name can be FQDN or IP address Examples of valid URLs firmware grandstream com fw ipvideotalk com gs There are two ways to setup a software upgrade server The IVR Menu or the Web Configuration Interface UPGRADE VIA KEYPAD MENU Follow the steps below to configure the upgrade server path via phone s keypad menu e Press MENU button and navigate using Up Down arrow to select Config e Inthe Config options select Upgrade e Enter the firmware server path and select upgrade method The server path could be in IP address format or FQDN format e Press the OK softkey A reboot message window will be prompt e Reboot the phone to have the change take effect When upgrading starts the screen will show upgrading progress When done you will see the phone restarts again Please do not interrupt or power cycle the phone when the upgrading process is on UPGRAGE VIA WEB GUI Open a web browser on PC and enter the IP address of the phone Then login with the administrator username and password Go to Settings gt Advanced Settings page enter the IP address or the FQDN for the upgrade server in F
29. SPLAY 0400000000 13 Table 5 GXP1160 GXP1165 5 2 20 000008 13 Table 6 GXP1160 GXP1165 KEYPAD 6 14 Table 7 5 23 Table 8 GXP1160 GXP1165 CONFIGURATION 0 000060000 26 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 3 of 59 E sen Table of Figures GXP1160 GXP1165 User Manual Figure 1 GXP1160 GXP1165 8 10 Figure 2 GXP1160 GXP1165 Keypad MENU 29 Figure 3 GXP1160 GXP1165 Web GUI 52 Figure 4 GXP1160 GXP1165 53 GUI Interface Examples GXP1160 GXP1165 User Manual http www grandstream com products gxp series general documents gxp21xx gui zip Screenshot of Configuration Login Page Screenshot of Status Page Screenshot of Basic Setting Configuration Page Screenshot of Advanced User Configuration Page Screenshot of SIP Account Configuration Page Screenshot of Saved Configuration Changes Page Screenshot of Reboot Page NOOO fF FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 4 of 59 Innovative IP Voice amp Video GNU GPL INFORMATION GXP1160 GXP1165 firmware contains third party software licensed
30. USER MANUAL Page 28 of 59 E sen Innovative IP Voice amp Video Answered Calls First Name Dialed Calls Last Name Missed Calls Number Transferred Calls Acct Forwarded Calls Groups MENU Clear All Confirm Add Back Cancel amp Return Groups Server Address Call History N Port lew Entry Base Search Download Phonebook XML Username Status Delete All Entries LDAP Number Filter LDAP Name Filter Phone Book LDAP Version Search LDAP Configuration Back LDAP Enable DND Directory Disable DND Do Not Disturb Esc Forward Call Instant Ring Tone Ring Messages Ring Volume Ring1 LCD Contrast Ring2 Download SCR XML Ring 3 Direct IP Call Erase Custom SCR Back Display Language Time Settings Preference ERE ERE SIP Proxy SIP Outbound Proxy Config gt Upgrade SIP User ID Factory Reset SIP Auth ID Layer 2 QoS SIP Password Factory Back SIP Transport Functions 252 Audio Loopback Cancel Diagnostic Mode Network Keyboard Diagnostic Back Firmware Server Call Features Config Server IP Setting Upgrades PPPoE Settings Beck Voice Mails aiie 802 1Q VLAN Tag Priority value Reboot DNS Server 1 Reset Vlan Config DNS Server 2 Back 802 1X Exit Back Forward All Forward Busy Account 1 m Forward No Answer No Answer Timeout Figure
31. acts edit contacts or dial out with Click to Dial feature on the top of the Web GUI In the following figure the Contact page shows all the added contacts manually or downloaded via XML phonebook Here users could add new contact export phonebook XML import phonebook XML search contact filter contacts by group edit selected contact or dial the contact number FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 51 of 59 Innovative Voice amp Video Click to select Click to Click to input number group the search in dial from available dropdown menu phonebook lines GXP1160 Executive IP Phone Device Configuration ndstieam Status Settings Contacts Accounts Innovative IP Voice i Video Software Version 1 0 5 2 PHONEBDOK First Name Last Name Phone Number 6372344564 Click to edit this SEEN contact Bob Lee Jane White 1 Add New Contact port Phonebook Import Phonebook XML Click to Click to export Click to import Click to call Click to download add new phonebook in phonebook this contact the contact contacts XML format XML file from the information in vcf phone format Figure 3 GXP1160 GXP1165 Web GUI Contacts When clicking on the icon on the top menu of the Web GUI a new dialing window will show for you to enter the number Once Dial is clicked the phone will go off hook and dial out the number f
32. all will be split and both calls will be put on hold separately Press FLASH key to resume the 2 way conversation with the second established call gt If users would like to re establish conference call press the ReConf softkey FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 22 of 59 E sen Innovative IP Voice amp Video e Cancel Conference gt If users decides to conference after establishing the second call press EndCall softkey gt This will end the second call and the screen will show the first call on hold e End Conference gt Press HOLD key to split the conference call The conference call will be ended with both calls on hold Or gt Users could press the EndCall softkey or simply hang up the call to terminate the conference call 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 e The option Disable Conference under GXP1160 GXP1165 Web GUI gt Settings gt Advanced Settings has to be set to No to establish conference VOICE MESSAGES MESSAGE WAITING INDICATOR A blinking red MWI Message Waiting Indicator indicates a message is waiting Dial into the voicemail box to retrieve the message by entering the voice mail number of the server or pressing the MESSAG
33. alue is disabled Enter the Identity for EAP PEAPvO MSCHAPv2 802 1x mode EAP MD5 Enter the Secret Private Key Password for 802 1x mode It won t be displayed for security protection purpose Upload the CA Certificate file for 802 1x mode Upload the Client Certificate for 802 1x mode Specifies the HTTP proxy URL for the phone to send packets to The proxy server will act as an intermediary to route the packets to the destination Specifies the HTTPS proxy URL for the phone to send packets to The FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 40 of 59 Time Zone Self Defined Time Zone Enable Weather Update City Code Update Interval Degree Unit LCD Contrast Date Display Format Innovative IP Voice 8 Video proxy server will act as an intermediary to route the packets to the destination Configures the date time used on the phone according to the specified time zone This parameter allows the users to define their own time zone The syntax is std offset dst offset start time end time Default is set to MTZ 6MDT 5 M4 1 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 M4 1 0 M11 1 0 The 1st number indicates Month 1 2 3 12 for Jan Feb
34. ancy and fail over RJ9 supporting Electronic Hook Switch EHS with Plantronics headsets Yes 1 angle position available FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 8 of 59 Wall Mountable QoS Security Multi language Upgrade and Provisioning Power and Green Energy Efficiency Physical Temperature and Humidity Package Content Compliance Coe Innovative IP Voice Yes Layer 2 802 1Q 802 1p and Layer 3 ToS DiffServ MPLS QoS User and administrator level passwords MD5 and MbD5 sess based authentication AES encrypted configuration file SRTP TLS 802 1x media access control English German Italian French Spanish Portuguese Russian Croatian Simplified and Traditional Chinese Korean Japanese and etc Firmware upgrade via TFTP HTTP HTTPS mass provisioning using TR 069 or AES encrypted XML configuration file Universal power adapter included Input 100 240VAC 50 60Hz Output 5VDC 800mA Integrated Power over Ethernet 802 3af GXP1165 only Max power consumption 2 5W universal power adapter or 3W PoE Unit dimension 154mm W x 200mm L x 79mm D handset onhook Unit weight 0 6kg Package weight 1 03kg Operating 32 104 F 0 40 C 10 90 non condensing Storage 14 140 F 10 60 GXP1160 GXP1165 phone handset with cord base stand universal power supply network cable quick start guide FCC Part 15 CFR 47 Class B EN55022 Class B EN55024 EN61
35. 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 small to medium businesses looking for a high quality feature rich IP phone with highly 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 GXP1160 GXP1165 as it may cause damage to the products and void the manufacturer warranty This document is subject to change without notice The latest electronic version of this user manual is available for download here http www grandstream com support Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 7 of 59 E sen innovative IP Voice amp Video PRODUCT OVERVIEW FEATURE HIGHTLIGHTS e 128 x 40 pixel graphical LCD display e Single SIP account up to 2 call appearances 3 XML programmable context sensitive soft keys 3 way conference e Phonebook with up to 500 contacts and call history with up to 200 records e Automated personal
36. ber to transfer to Press SEND key or to complete transfer of active call e Attended Transfer gt During the first active call press FLASH key C The first call will be put on hold gt Enter the number for the second call and establish the call Press TRANSFER key gt Press FLASH key to transfer the call FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 20 of 59 Innovative Voice amp Video Auto Attended Transfer gt Note Set Auto Attended Transfer to Yes under Web GUI gt Advanced Settings page And then click Update on the bottom of the page Establish one call first During the call press TRANSFER key 2 line will be brought and the first call will be automatically placed on hold Enter the number and press SEND key to establish the second call After the second call is established press TRANSFER key 2 again The will be transferred If users press the SPLIT softkey before the call is transferred in the step above the second call will be resumed To transfer calls across SIP domains SIP service providers must support transfer across SIP domains 3 WAY CONFERENCING GXP1160 GXP1165 can host 3 way conference call with another 2 parties Initiate a conference call gt gt gt Establish 2 calls with 2 parties respectively Press CONFERENCE key Press FLASH key CZ 3 way conference will be
37. cts encoding rate for G723 codec The default value is 5 3kbps Selects ITU or IETF for G726 32 packing mode Selects iLBC packet frame size The default value is 30ms Specifies Payload type The default value is 97 The valid range is between 96 and 127 Selects either Fixed or Adaptive based on network conditions The default setting is Adaptive Selects Low Medium or High based on network conditions The default FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 38 of 59 Conference URI DND Call Feature On DND Call Feature Off Use Privacy Header Use P Preferred Identity Header Special Feature setting is Medium Configures the conference URI when using Broadsoft N way calling feature Configures DND feature code to turn on DND Configures DND feature code to turn off DND Controls whether the Privacy Header will present in the SIP INVITE message or not The default setting is default which is when Huawei IMS special feature is on the Privacy Header will not show in INVITE If set to Yes the Privacy Header will always show in INVITE If set to No the Privacy Header will not show in INVITE Controls whether the P Preferred Identity Header will present in the SIP INVITE message or not The default setting is default which is when Huawei IMS special feature is on the P Preferred Identity Header will not show in INVITE If set to Yes the P Preferred Identity H
38. d should be set to UserzPhone Then a User Phone parameter will be attached to the Request Line and TO header in the SIP request to indicate the E 164 number If set to Enable Tel will be used instead of SIP in the SIP request The default setting is Disable Selects whether or not the phone will send SIP Register messages to the proxy server The default setting is Yes If set to Yes the SIP user s registration information will be cleared when the phone reboots The SIP Contact header will contain to notify the server to unbind the connection The default setting is No Specifies the frequency in minutes in which the phone refreshes its FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 32 of 59 Reregister Before Expiration Local SIP Port SIP Registration Failure Retry Wait Time SIP T1 Timeout SIP T2 interval SIP Transport SIP URI Scheme when using TLS Use Actual Ephemeral Port in Contact with TCP TLS Check Domain Certificates Remove OBP from route Validate Incoming Messages Support SIP Instance ID NAT Traversal Innovative IP Voice amp Video registration with the specified registrar The default value is 60 minutes The maximum value is 64800 minutes about 45 days Specifies the time frequency in seconds that the phone sends re registration request before the Register Expiration The default value is 0 Defines the local SIP port used to liste
39. e Call Waiting Tone Busy Tone Reorder Tone Disable Call Waiting Gee Innovative IP Voice e ethernet link up INFO level SLIC chip exception WARNING and ERROR levels e memory exception ERROR level Configures whether the SIP log will be included in the syslog messages or not The default setting is No Defines the URL or IP address of the NTP server The phone may obtain the date and time from the server Defines whether DHCP Option 42 should override NTP server or not When enabled DHCP Option 42 will override the NTP server if it s set up on the LAN The default setting is Yes Configures to turn on off public mode for hot desking feature on the phone If set to Yes users would need fill in the SIP Server address for account 1 as well Then reboot the phone When the phone boots up users will need enter SIP User ID and Password on the LCD to login and use the phone Note When the phone is in public mode login screen press HOLD button will have the IP address of the phone displayed SSL Certificate used for SIP Transport in TLS TCP SSL Private key used for SIP Transport in TLS TCP SSL Private key password used for SIP Transport in TLS TCP Configures system ring tone The default value is North American standard Users could adjust system ring tone frequencies and cadences based on local telecom standard Configures ring or tone frequencies based on parameters from local telecom The default value
40. e space separated number attributes Example telephoneNumber telephoneNumber Mobile Configures the entry information to be shown on phone s LCD Up to 3 fields can be displayed Example cn 9esn YtelephoneNumber Specifies the maximum number of results to be returned by the LDAP server If set to 0 server will return all search results The default setting is 50 Specifies the interval in seconds for the server to process the request and client waits for server to return The default setting is 30 seconds Specifies whether the searching result is sorted or not The default FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 46 of 59 LDAP Lookup Lookup Display Name Use Phonebook Key for LDAP Search Idle Screen XML Download Download Screen XML At Bootup User Custom Filename Idle Screen XML Server Path Offhook Auto Dial Auto Recover From Abnormal Syslog Server Syslog Level setting is Configures to enable LDAP number searching when dialing and receiving calls Configures the display name when LDAP looks up the name for incoming call or outgoing call This field must be a subset of the LDAP Name Attributes Example gn cn sn description If set to Yes the Phonebook Key f pressing will bring up LDAP search screen Configures to enable idle screen XML download Users could select HTTP HTTPS TFTP to download the XML idle screen file The default sett
41. e the phone will try to use public IP addresses and port number in all the SIP amp SDP messages The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be Keep Alive Configure this to be No if an outbound proxy is used STUN cannot be used if the detected NAT is symmetric NAT FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 33 of 59 SUBSCRIBE for MWI SUBSCRIBE for Registration Feature Key Synchronization Proxy Require Voice Mail UserlD Send DTMF DTMF Payload Type Early Dial Dial Plan Prefix Dial Plan Gosia Innovative When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically The phone supports synchronized and non synchronized MWI The default setting is When set to Yes a SUBSCRIBE for Registration will be sent out periodically The default setting is No This feature is used for Broadsoft call feature synchronization When it s enabled DND and Call Forward features can be synchronized with Broadsoft server The default setting is Disabled A SIP Extension to notify the SIP server that the phone is behind a NAT Firewall Do not configure this parameter unless this feature is supported on the SIP server Allows you to access voice messages by pressing the MESSAGE button on the phone This ID is usually the VM portal access number For example in Asterisk server 8500 could be u
42. e phone to the same LAN segment 3 Launch the TFTP server and go to the File menu gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade Start the TFTP server and configure the TFTP server in the phone s web configuration interface 5 Configure the Firmware Server Path to the IP address of the PC Update the changes and reboot the phone End users can also choose to download a free HTTP server from hittp httod apache org or use Microsoft IIS web server Note When the phone boots up it will send a TFTP or HTTP request to download the configuration file where is the MAC address of the phone If it is a normal TFTP or HTTP upgrade the following messages TFTP Error from IP ADRESS requesting cfg000b82023dd4 File does not exist Configuration File Download can be ignored in the TFTP HTTP server log CONFIGURATION FILE DOWNLOAD Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File binary or XML through TFTP or HTTP HTTPS The Config Server Path is the TFTP or HTTP HTTPS FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 56 of 59 Innovative IP Voice amp Video 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 f
43. eader will always show in INVITE If set to No the P Preferred Identity Header will not show in INVITE Different soft switch vendors have special requirements Therefore users may need select special features to meet these requirements Users can choose from Standard Nortel MCS Broadsoft CBCOM RNK Sylantro or Huawei IMS depending on the server type The default setting is Standard SETTINGS BASIC SETTINGS PAGE End User Password Confirm Password Internet Protocol IPv4 Address Type DHCP Host name Option 12 DHCP Vendor Class ID Option 60 Allows the administrator to set the password for user level web GUI access This field is case sensitive with a maximum length of 30 characters Confirms the end user password field to be the same as above Selects Prefer IPv4 or Prefer IPv6 Allows users to configure the appropriate network settings on the phone to obtain IPv4 address Users could select DHCP Static IP or PPPoE By default it is set to DHCP Specifies the name of the client This field is optional but may be required by some Internet Service Providers Used by clients and servers to exchange vendor class ID FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 39 of 59 Allow DHCP Option 120 to override SIP Server PPPoE Account ID PPPoE Password PPPoE Service Name IPv4 Address Subnet Mask Gateway DNS Server 1 DNS Server 2 Preferred DNS Server IPv6 Address Type Static
44. ed into the phone If DHCP option 66 is enabled on the LAN side the TFTP server can be redirected The default setting is Yes Enables automatic upgrade and provisioning The default setting is No When Automatic Upgrade is set to Yes check for upgrade every day configure the hour of the day when the upgrading provisioning starts When Automatic Upgrade is set to Yes check for upgrade every week configure the day of the week when the upgrading provisioning starts Authenticates configuration file before acceptance The default setting is Enables 069 The default setting is URL for TR 069 Auto Configuration Servers ACS ACS username for TR 069 ACS password for TR 069 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 44 of 59 Periodic Inform Enable Periodic Inform Interval Connection Request Username Connection Request Password Connection Request Port CPE SSL Certificate CPE SSL Private Key Phonebook XML Download Phonebook XML Server Path Phonebook Download Interval Remove Manually edited Entries on Download LDAP Directory Server Address LDAP Directory Port LDAP Directory Base LDAP Directory User Name LDAP Directory Password LDAP Number Filter Ae Innovative IP Voice amp Video Enables periodic inform If set to Yes device will send inform packets to the ACS The default setting is No Sets up the periodic inform interva
45. g PC s web browser CONFIGURATION VIA KEYPAD To configure via the LCD configuration menu using phone s keypad follow the instructions below e Enter MENU options When the phone is in idle press the round MENU button to enter the configuration menu Navigate in the menu options Press the arrow keys up down left right to navigate in the menu options e Enter Confirm selection Press the round MENU button to enter the selected option e Exit Press LEFT arrow key to exit to the previous menu e The phone automatically exits MENU mode with an incoming call when the phone is off hook or the MENU mode if left idle for more than 20 seconds The MENU options are listed in the following table Table 8 GXP1160 GXP1165 CONFIGURATION MENU Call History Displays call logs for answered calls dialed calls missed calls transferred calls and forwarded calls Status Displays network status account registration status software version number MAC address hardware version number P N number e Network status Press to enter the sub menu for IP setting information DHCP Static IP PPPoE IPv4 address IPv6 address Subnet Mask Gateway and DNS server Phone Book Displays phonebook Users could add edit search and delete contacts here or download phonebook XML to the phone LDAP Directory Configures LDAP directory options displays LDAP directory by FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 26 of 59
46. ing is No If set to Yes the idle screen XML file will be downloaded when the phone boots up The default setting is No Specifies the custom file for the idle screen XML file to be downloaded Configures the server path to download the idle screen XML file This field could be IP address or URL with up to 256 characters Configures a User ID extension to dial automatically when the phone is off hook The phone will use the first account to dial out The default setting is No Configures whether auto recover or not when the phone is running abnormal The default setting is Yes The URL or IP address of the syslog server for the phone to send syslog to Selects the level of logging for syslog The default setting is None There are 4 levels DEBUG INFO WARNING AND ERROR Syslog messages are sent based on the following events e product model version on boot up INFO level e NAT related info INFO level e sentor 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 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 47 of 59 109 NTP Server Allow DHCP Option 42 Override NTP Server Public Mode SSL Certificate SSL Private Key SSL Private Key Password System Ring Tone Call Progresses Tones Dial Tone Message Waiting Ring Back Ton
47. irmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP Option 43 and Option 66 Override Server Automatic Upgrade Hour of the Day 0 23 Day of the Week 0 6 Authenticate Conf File Enable TR 069 ACS URL TR 069 Username TR 069 Password Cin 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 Allows users to choose the firmware upgrade method TFTP HTTP or HTTPS Defines the server path for the firmware server It could be different from the configuration server for provisioning Defines the server path for provisioning It could be different from the firmware server for upgrading Enables your ITSP to lock firmware updates If configured only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone Enables your ITSP to lock firmware updates If configured only the firmware with the matching encrypted postfix will be downloaded and flashed into the phone Enables your ITSP to lock configuration updates If configured only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone Enables your ITSP to lock configuration updates If configured only the configuration file with the matching encrypted postfix will be downloaded and flash
48. irmware Server Path field and choose to upgrade via TFTP or HTTP HTTPS Update the change by clicking the Update button Then Reboot or power cycle the phone to update the new firmware When upgrading starts the screen will show upgrading progress When done you will see the phone restarts again Please do not interrupt or power cycle the phone when the upgrading process is on FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 55 of 59 Innovative Voice amp Video 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 SERVERS For users that would like to use remote upgrading without a local TFTP HTTP server Grandstream offers a NAT friendly HTTP server This enables users to download the latest software upgrades for their phone via this server Please refer to the webpage http www grandstream com support firmware Alternatively users can download a free TFTP or HTTP server and conduct a local firmware upgrade A free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm http tftpd32 jounin net Instructions for local firmware upgrade via TFTP 1 Unzip the firmware files and put all of them in the root directory of the TFTP server 2 Connect the PC running the TFTP server and th
49. is 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 Disables the call waiting feature The default setting is FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 48 of 59 Disable Call Waiting Tone Disable Direct IP Calls Use Quick IP Call mode Disable Conference Disable Transfer Auto Attended Transfer In call dial number on pressing transfer key Configuration via Keypad Menu Enable STAR key Keypad locking Password to lock unlock Offhook timeout China Telecom Mode Gosia Innovative Disables the call waiting tone when call waiting is on The default setting is No Disables Direct IP Call The default setting is When set to Yes users can dial an IP address under the same LAN VPN segment by entering the last octet in the IP address To dial quick IP call off hook the phone and dial XXX 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
50. l to send the inform packets to the ACS The user name for the ACS to connect to the phone The password for the ACS to connect to the phone The port for the ACS to connect to the phone The Cert File for the phone to connect to the ACS via SSL The Cert Key for the phone to connect to the ACS via SSL Configures to enable phonebook XML download Users could select HTTP HTTPS TFTP to download the phonebook file The default setting is No Configures the server path to download the phonebook XML This field could be IP address or URL with up to 256 characters Configures the phonebook download interval in minutes If it s set to 0 the automatic download will be disabled The default value is 0 The valid range is 5 to 720 minutes If set to Yes when XML phonebook is downloaded the entries added manually will be automatically removed The default setting is Yes Configures the IP address or DNS name of the LDAP server Configures the LDAP server port Configures the LDAP search base This is the location in the directory where the search is requested to begin Example dc grandstream dc com ou Boston dc grandstream dc com Configures the bind Username for querying LDAP servers Some LDAP servers allow anonymous binds in which case the setting can be left blank Configures the bind Password for querying LDAP servers The field can be left blank if the LDAP server allows anonymous binds Configures the fi
51. ll press HEADSET button to toggle between Headset and Speaker Defines the interval in seconds to save the call history to phone s flash The default value is 300 seconds Defines the number of unsaved logs before written to phone s flash The default value is 200 entries Configures the transmission gain of the headset The default value is OdB FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 42 of 59 Headset gain Handset TX gain Gee Configures the receiving gain of the headset The default value is OdB Configures the transmission gain of the handset The default value is 0 SETTINGS ADVANCED SETTINGS PAGE Admin Password Confirm Password Layer 3 QoS Layer 2 QoS 802 1Q VLAN Tag Layer 2 QoS 802 1p Priority Value Local RTP Port Use Random Port Keep alive Interval Use NAT IP STUN Server Firmware Upgrade and Provisioning XML Config File Password Allows users to change the admin password The password field is purposely hidden after clicking the Update button for security purpose This field is case sensitive with a maximum length of 30 characters Confirms the admin password field to be the same as above Defines the Layer 3 QoS parameter This value is used for IP Precedence Diff Serv or MPLS The default value is 12 Assigns the VLAN Tag of the Layer 2 QoS packets The default value is 0 Assigns the priority value of the Layer2 QoS packets The default val
52. lter used for number lookups Examples telephoneNumber Mobile returns all records which has the telephoneNumber or Mobile field starting with the entered prefix FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 45 of 59 LDAP Name Filter LDAP Version LDAP Name Attributes LDAP Number Attributes LDAP Display Name Max Hits Search Timeout Sort Results Innovative IP Voice 8 Video amp telephoneNumber returns all the records with the telephoneNumber field starting with the entered prefix and cn field set Configures the filter used for name lookups Examples returns all records which has the cn or sn field starting with the entered prefix sn returns all the records which do not have the sn field starting with the entered prefix amp cn telephoneNumber returns all the records with the cn field starting with the entered prefix and telephoneNumber field set Selects the protocol version for the phone to send the bind requests The default setting is Version 3 Specify the name attributes of each record which are returned in the LDAP search result This field allows the users to configure multiple space separated name attributes Example gn cn sn description Specifies the number attributes of each record which are returned in the LDAP search result This field allows the users to configure multipl
53. n and transmit The default value is 5060 for Account 1 and 5062 for Account 2 Specifies the interval to retry registration if the process is failed The default value is 20 seconds SIP T1 Timeout The default setting is 0 5 seconds SIP T2 Interval The default setting is 4 seconds Determines the network protocol used for the SIP transport Users can choose from TCP UDP and TLS Specifies if sip or sips will be used when TLS TCP is selected for SIP Transport The default setting is sips Defines whether the actual ephemeral port in contact with TCP TLS will be used or not This is used when TLS TCP is selected for SIP Transfer The default setting is No Defines whether the domain certificates will be checked or not when TLS TCP is used for SIP Transport The default setting is No Configures to remove outbound proxy from route This is used for the SIP Extension to notify the SIP server that the device is behind a NAT Firewall Defines whether the incoming messages will be validated or not The default setting is No Defines whether SIP Instance ID is supported or not The default setting is Yes This parameter configures whether the NAT traversal mechanism is activated Users could select the mechanism from No STUN Keep Alive UPnP Auto or VPN If set to STUN and STUN server is configured the phone will route according to the STUN server If NAT type is Full Cone Restricted Cone or Port Restricted Con
54. ook on screen Navigation Keys Menu Press the 4 navigation keys to move up down left right CES e Press the round button in the center to enter Keypad Configuration MENU when phone is idle 57 e The round button MENU can also be used as ENTER key when in Keypad Configuration Volume Press or to adjust the volume 0 9 Standard phone keypad MAKING PHONE CALLS HANDSET SPEAKER AND HEADSET MODE The GXP1160 GXP1165 allows users to switch among handset speaker or headset when making calls Press the Hook Switch to switch to handset press the Headset softkey to switch to headset or press the Speaker button to switch to speaker 2 CALLS WITH 1 SIP ACCOUNT GXP1160 GXP1165 can support up to two lines virtually mapped to one SIP account By picking up the handset the GXP1160 GXP1165 will be in off hook state and the dial tone will be heard To make a call FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 15 of 59 E sen Innovative IP Voice amp Video dial out the number with the current line During the call users can press the FLASH key to hold the current call and make answer another call If they are 2 calls established users can switch the two lines by pressing the FLASH key COMPLETING CALLS There are several ways to complete a call on GXP1160 GXP1165 e On hook dialing Enter the number when the phone is on hook and then send out When the phone
55. ort those features The default setting is Yes If set to No ForwardAll softkey will be hidden for FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 35 of 59 Call Log 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 Matching Incoming Caller ID Innovative Voice amp Video Account 1 Configures Call Log setting on the phone You can log all calls only log incoming outgoing calls or disable call log The default setting is Log All Calls The SIP Session Timer extension that enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE If there is no refresh via an UPDATE or re INVITE message the session will be terminated once the session interval expires Session Expiration is the time in seconds where the session is considered timed out provided no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds If set to Yes and the remote party supports session timers the phone will use a session timer when it makes outbound calls If set to Yes and the remote party supports session timers the phone will use a session timer when it receives inbound calls If Force Timer is set to Yes the phone will
56. phone has IP address 192 168 40 154 please enter http 192 168 40 154 in the address bar of the browser e The default administrator password is set to admin The default user password is set to 123 e When changing any settings always SUBMIT them by pressing the UPDATE button on the bottom of the page After submitting the changes in all the Web GUI pages reboot the phone to have the changes take effect if necessary All the options under Basic Setting and Account Setting and most of the options under Advanced Settings do not require reboot after submitting the changes Under Advanced Setting the parameters on network configuration require reboot after update DEFINITIONS This section describes the options in the GXP1160 GXP1165 Web GUI As mentioned you can log in as FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 30 of 59 administrator an end user ae Innovative IP Voice amp Video e Status Displays the Account status Network status and System Info of the phone e Account To configure the SIP account e Basic Settings To configure basic network settings time settings Line keys and etc e Advanced Settings To configure advanced network settings upgrading and provisioning language settings call features and etc STATUS PAGE DEFINITIONS MAC Address IPv4 Address IPv6 Address Product Model Part Number Software Version System Up Time System Time Registered
57. rical outlet If PoE switch is used on GXP1165 in step 3 this step could be skipped 5 The LCD will display provisioning or firmware upgrade information Before continuing please wait for the date time display to show up 6 Using the keypad configuration menu or phone s embedded web server Web GUI by entering the IP address in web browser you can further configure the phone SAFETY COMPLIANCES The GXP1160 GXP1165 phone complies with FCC CE and various safety standards The GXP1160 GXP1165 power adapter is compliant with the UL standard Use the universal power adapter provided with the GXP1160 GXP1165 package only The manufacturers warranty does not cover damages to the phone caused by unsupported power adapters WARRANTY If the GXP1160 GXP1165 phone was purchased from a reseller please contact the company where the phone was purchased for replacement repair or refund If the phone was purchased directly from Grandstream contact the Grandstream Sales and Service Representative for a RMA Return Materials FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 11 of 59 Innovative Voice amp Video Authorization number before the product is returned Grandstream reserves the right to remedy warranty policy without prior notification Warning Use the power adapter provided with the phone Do not use a different power adapter as this may damage the phone This type of damage is not covered under warranty F
58. rom selected account FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 52 of 59 E sen Innovative IP Voice amp Video PHONE Account First Available Dial Number LINE2 Figure 4 GXP1160 GXP1165 Click to Dial Additionally users could directly send the command for the phone to dial out by specifying the following URL in PC s web browser or in the field as required in other call modules http ip_address cgi bin api make_call phonenumber 1234 amp account 0 amp password admin In the above link replace the fields with e ip address Phone s IP Address phonenumberz 1234 The number for the phone to dial out e account 0 The account index for the phone to make call The index is 0 for account 1 1 for account 2 2 for account 3 and etc passwordzadmin The admin login password of phone s Web GUI FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 53 of 59 E sen Innovative IP Voice amp Video SAVING THE CONFIGURATION CHANGES After users makes changes to the configuration press the Update button on the bottom of the Web GUI page We recommend rebooting or powering cycle the IP phone after saving changes REBOOTING FROM REMOTE LOCATIONS Press the Reboot button on the bottom of the web GUI page to reboot the phone remotely The web browser will then display a reboot page with message The device is rebooting now Wait for about 1 minute to log in again FI
59. rom 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 3 Could be extended to 4 in the future digit numeric numbers i e P2 is associated with the Admin Password in the Web GUI gt Settings gt Advanced Settings For a detailed parameter list please refer to the corresponding firmware release configuration template When a Grandstream Devices boots up or reboots it will issue a request for a configuration XML file named cfgxxxxxxxxxxxx xml followed by a file named cfgxxxxxxxxxxxx where is the MAC address of the device cfg000b820102ab The configuration file name should be in lower case letters For more details on XML provisioning please refer to http www grandstream com general gs provisioning guide public pdf FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 57 of 59 Innovative IP Voice amp Video RESTORE FACTORY DEFAULT SETTINGS Warning Restoring the Factory Default Settings will delete all configuration information on the phone Please backup or print all the settings before you restore to the factory default settings Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider Please follow the instructions below to reset the phone e Press MENU button to bring
60. sed Specifies the mechanism to transmit DTMF digits There are 3 supported modes in audio which means is combined the audio signal not very reliable with low bit rate codecs via RTP 2833 or via SIP INFO Configures the payload type for using RFC2833 The default value is 101 Selects whether or not to enable early dial If it s set to Yes the SIP proxy must support 484 response The default setting is Sets the prefix added to each dialed number A dial plan establishes the expected number and pattern of digits for a telephone number This parameter configures the allowed dial plan for the phone 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 at least 2 digit numbers only 2 digit numbers exclude 8 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 g the OR operand FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 34 of 59 Delayed Call Forward Wait Time Enable Call Features Gee innovative IP Voice 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 Block any number of leading digits 1900 add prefix 1617 for any dialed 7 digit numbers Example 3
61. t dialed out number Table 5 GXP1160 GXP1165 LCD ICONS Registration Status Registered Registration Status Not Registered Handset Status FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 13 of 59 6 Cin Innovative IP Voice OFF handset on hook ON handset off hook Speaker Status OFF speaker off ON speaker on Headset Status OFF headset off ON headset on DND Status OFF Do Not Disturb disabled ON Do Not Disturb enabled Call Forward Status OFF Call Forward feature disabled ON Call Forward feature enabled MUTE Status OFF The active call is not muted ON The active call is muted SRTP Status OFF SRTP is not used ON SRTP is used GETTING FAMILAR WITH THE KEYPAD The following table describes the buttons used the GXP1160 GXP1165 keypad Sa Table 6 GXP1160 GXP1165 KEYPAD DEFINITIONS Place active call on hold or resume the call on hold Transfer an active call to another number Establish 3 way conference with other 2 parties FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 14 of 59 E sen Innovative IP Voice amp Video Bring up a new line or answer the second incoming call Speaker Send Redial e Send Enter the digits and then press Send to dial out the number e Redial when there is a previously dialed call 9 Voicemail Press to retrieve voice mails Phonebook Brings phoneb
62. tact Transfer on Conference Hangup Check SIP User ID for incoming INVITE Authenticate Incoming INVITE Innovative IP Voice 8 Video with selected distinctive ringtone Matching rules e Specific caller ID number For example 8321123 A defined pattern with certain length using x and to specify where x could be any digit from 0 to 9 Samples at least 2 digit number XX only 2 digit number 345 xx 3 digit number with the leading digit of 3 4 or 5 6 9 xx 3 digit number with the leading digit from 6 to 9 e Alert Info text Users could configure the matching rule as certain text e g priority and select the custom ring tone mapped to it The custom ring tone will be used if the phone receives SIP INVITE with Alert Info header in the following format Alert Info lt http 127 0 0 1 gt info priority Selects the distinctive ring tone for the matching rule When the incoming caller ID or Alert Info matches the rule the phone will ring with the selected ring Defines the timeout in seconds for the rings on no answer The default setting is 60 seconds If set to Yes the From header in outgoing INVITE messages will be set to anonymous essentially blocking the Caller ID to be displayed If set to Yes anonymous calls will be rejected The default setting is No If set to Yes the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep
63. the SIP server This will be used when the primary SIP server fails IP address or Domain name of the Primary Outbound Proxy Media Gateway or Session Border Controller It s used by the phone for Firewall or NAT penetration in different network environments If a symmetric NAT is detected STUN will not work and ONLY an Outbound Proxy can provide a solution User account information provided by your VoIP service provider ITSP It s usually in the form of digits similar to phone number or actually a phone number SIP service subscriber s Authenticate ID used for authentication It can be identical to or different from the SIP User ID The account password required for the phone to authenticate with the ITSP SIP server before the account can be registered After it is saved this will appear as hidden for security purpose The SIP server subscriber s name optional that will be used for Caller ID display This parameter controls how the Search Appliance looks up IP addresses for hostnames There are four modes A Record SRV NATPTR SRV Use Configured IP The default setting is A Record If the user wishes to locate the server by DNS SRV the user may select or NATPTR SRV If Use Configured IP is selected please fill in the three fields below e Primary IP The primary IP address where the phone sends DNS query to e Backup IP 1 e Backup IP 2 If the phone has an assigned PSTN telephone number this fiel
64. ue is 0 This parameter defines the local RTP port 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 channel 1 will use port_value 2 for Local RTP port ranges from 1024 to 65400 and must be even The default value is 5004 When set to Yes this parameter will force random generation of both the local SIP and RTP ports This is usually necessary when multiple phones are behind the same full cone NAT The default setting is Yes This parameter must be set to No for Direct IP Calling to work Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep the ping hole on the NAT router to open The default setting is 20 seconds The NAT IP address used in SIP SDP messages This field is blank at the default settings It should ONLY be used if it s required by your ITSP The IP address or Domain name of the STUN server STUN resolution results are displayed in the STATUS page of the Web GUI Only non symmetric NAT routers work with STUN Specifies how firmware upgrading and provisioning request to be sent Always Check for New Firmware Check New Firmware only when F W pre suffix changes Always Skip the Firmware Check The password for encrypting the XML configuration file using OpenSSL FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 43 of 59 HTTP HTTPS User HTTP HTTPS Password Upgrade Via F
65. use the session timer even if the remote party does not support this feature If Force Timer is set to No the phone will enable the session timer only when the remote party supports this feature To turn off the session timer select No As a Caller select UAC to use the phone as the refresher or select 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 select UAS to use the phone as the refresher The Session Timer can be refreshed using the INVITE method or the UPDATE method Select Yes to use the INVITE method to refresh the session timer The use of the PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is very important in order to support PSTN internetworking To invoke a reliable provisional response the 100rel tag is appended to the value of the required header of the initial signaling messages Allows users to configure the ringtone for the account Users can choose from different ringtones from the dropdown menu Specifies matching rules with number pattern or Alert Info text When the incoming caller ID or Alert Info matches the rule the phone will ring FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 36 of 59 Distinctive Ringtones Ring Timeout Send Anonymous Anonymous Call Rejection Auto Answer Allow Auto Answer by Call Info Refer To Use Target Con
66. usy Call Forward To set up busy call forward e Off hook the phone Dial 90 and then enter the number to forward the call e Press OK softkey or SEND key Cancel Busy Call Forward To cancel the busy call forward e Off hook the phone e Dial 91 FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 24 of 59 E sen innovative IP Voice amp Video e Hang up the call 92 Delayed Call Forward To set up delayed call forward e Off hook the phone e Dial 92 and then enter the number to forward the call e Press OK softkey or SEND key 93 Cancel Delayed Call Forward To cancel the delayed call forward e Off hook the phone e Dial 93 e Hang up the call CUSTOMIZED LCD SCREEN amp XML The GXP1160 GXP1165 IP phone supports the following XML applications Please refer to the corresponding link for documentation and templates XML custom idle screen customize idle screen logo softkey layout and etc http www grandstream com products gxp_series general documents GXP140x XML Screen Custo mization zip e XML downloadable phonebook http www grandstream com products gxp series general documents gxp wp xml phonebook pdf FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 25 of 59 innovative Voice amp Video CONFIGURATION GUIDE The GXP1160 GXP1165 can be configured via two ways e LCD Configuration Menu using the phone s keypad e Web GUI embedded on the phone usin
67. when multiple GXPs are behind the same NAT If using a Public IP address set this parameter to e NAT Traversal under Account Setting page Default setting is No Enable the device to use NAT traversal when it is behind firewall on a private network Select Keep Alive Auto STUN with STUN server path configured too or other option according to the network setting PUBLIC MODE The GXP1160 GXP1165 supports hot desking using public mode Under public mode users could login the phone with the SIP account User ID and password Please follow the steps below to configure the phone for public mode FIRMWARE VERSION 1 0 5 2 GXP1160 GXP1165 USER MANUAL Page 50 of 59 E sen e Under Web GUI gt Account 1 setting page fill up the SIP server address for account 1 Click Update on the bottom of the page e Under Web GUI gt Advanced setting page set Public Mode option to Yes Click Update and reboot the phone e When phone boots up SIP User ID and Password to register to the configured SIP server in account 1 will be required Enter the correct account information to log in to the phone When entering the account information press softkey 123 abc to toggle input method e In login page pressing HOLD button on the phone will show phone s IP address e After using the phone go to LCD MENU gt LogOut to log off the public mode EDITING CONTACTS AND CLICK TO DIAL From GXP1160 GXP1165 Web GUI users could view cont

Download Pdf Manuals

Related Search

Related Contents

DR-40 Owner`s Manual - 6.31 MB  πριονοκορδελα dw738, dw739 - Service après vente  LG LS720 Specification Sheet  Ing. MSc. Leonardo Mata Rojas    Ewent EW1218 mobile device charger    User Manual - Berkeley Nucleonics Corporation  PCS Clinical Audit Tool – User Guide  BT-335 Bluetooth Data Logger User Manual  

Copyright © All rights reserved. Failed to retrieve file