GXP User Manual
Contents
1. 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 GXP1100 1105 User Manual Page 28 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream Innovative IP Voice amp Video Restore Factory Default Setting WARNING Restoring the Factory Default Setting will delete all configuration information of the phone Please backup or print all the settings before you restoring factory default settings We are not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider INSTRUCTIONS FOR RESTORATION Step 1 Press to enter the IVR menu Input 99 to for factory reset 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 E 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 show on the LCD 333 3333 if the MAC address is 000582006395 it should be key in as 0002228200333395 NOTE f there are digi2. network setup i Support NAT traversal using IETF STUN and Symmetric RTP _ Support for IEEE 802 1p Q tagging VLAN Layer3QoS Firmware Support firmware upgrade via TFTP or HTTP Upgrades Support for Authenticating configuration file before accepting changes User specific URL for configuration file and firmware files Grandstream Networks Inc GXP1100 1105 User Manual Page 6 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream Innovative IP Voice amp Video Mass provisioning using TR 069 or encrypted XML configuration file Advanced Server Features Message waiting indication support DNS SRV Look up and SIP Server Fail Over Support customizable idle screen via downloading XML by OOS O LRL Lm 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 GXP1100 1105 User Manual Page 7 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 andstream Innovative IP Voice amp Video Using the GXP1100 1105 GETTING FAMILIAR WITH THE KEYPAD Table 7 GXP1100 1105 Keypad Buttons Key Button Key Button Definitions Place active call on hold Call waiting bring up a new line or answer the second incoming call V A j 0808 Transfer an active call to another number Enter to retrie
3. Otherwise it will ring ON ms anda pause of OFF ms and then repeat the pattern Up to three cadences are supported Default is No If set to Yes the call waiting feature will be disabled Default is No If set to Yes the call waiting tone will be disabled Default is No If set to Yes direct IP calls will be disabled Dial an IP address under the same LAN VPN segment by entering the last octet in the IP address In the Advanced Settings page there is an option Use Quick IP call mode Default setting is No When set to Yes and XXX is dialed where X is 0 9 and XXX 2255 phone will make direct IP call to aaa bbb ccc XXX where aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask XX or X are also valid so leading O is not required but OK See Quick IP Call Mode for details 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 GXP1100 1105 User Manual Page 19 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 Display Language andstream Innovative IP Voice amp Video Allows user to choose preferred display language in web UI and key pad UI Currently the phone supports these languages English Simplified Chinese Traditional Chinese Korean Japanese Italian Spanish French German Portuguese Russian Croatian Hungarian Polish Slovenian Arabic Hebrew
4. Update button in the Configuration Menu 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 indicator on the top right corner will first turn orange and then red Wait until the indicator is off Then log in again Grandstream Networks Inc GXP1100 1105 User Manual Page 26 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 andstream Innovative IP Voice amp Video Software Upgrade amp Customization Software or firmware upgrades are completed via either TFTP or HTTP The corresponding configuration settings are in the ADVANCED SETTINGS configuration page FIRMWARE UPGRADE THROUGH TFTP HTTP To upgrade via TFTP or HTTP select TFTP or HTTP upgrade method Upgrade Server needs to be set to a valid URL of a HTTP server Server name can be in either FQDN or IP address format Here are examples of some valid URLs e firmware mycompany com 6688 Grandstream 1 2 3 5 e 72 172 83 110 There are two ways to set up the Upgrade Server to upgrade firmware via IVR Menu and Web Configuration Interface IVR Menu To configure the Upgrade Server via IVR e Pick up the handset press e Input menu option 15 Upgrade Protocol Press 9 to toggle between different upgrade methods e Press to return to the main menu and input menu option 13
5. Innovative IP Voice amp Video GUI INTERFACE EXAMPLES GXP1100 1105 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 NOOR WD Grandstream Networks Inc GXP1100 1105 User Manual Page 2 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 innovative IP Voice amp Video Welcome GXP1100 1105 is a next generation small business IP phone that features up to 2 calls with 1 SIP account 4 programmable keys single network port integrated PoE GXP1105 only The GXP1100 1105 delivers superior HD audio quality leading edge telephony features 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 business lobby and hotel applications looking for a high quality basic IP phone with attractive 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 diffe
6. Web Configuration Menu e Connect the computer to the same network as the phone e Make sure the phone is turned on and wait until the indicator on the top right corner turns from RED to OFF e Take the handset off hook Enter and then press 02 to hear the IP address e Start a Web browser on your computer e Enter the phone s IP address in the address bar of the browser e Enter the administrator s password to access the Web Configuration Menu The Web enabled computer has to be connected to the same sub network as the phone This can easily be done by connecting the computer to the same hub or switch as the phone is connected to If the phone is properly connected to a working Internet connection and dynamic IP mode is selected under IVR menu option 01 the phone will get an IP address automatically and the IP address will be announced under IVR menu option 02 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 http 192 168 0 60 in the address bar of your browser Co The default administrator password is admin the default end user password is 123 NOTE When changing any settings always SUBMIT them by pressing UPDATE button on the bottom of the page Reboot the phone to have the changes take effect If after having submitted some changes m
7. and Dutch 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 13 SIP Account Settings Account Name SIP Server Secondary SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password Grandstream Net
8. e Base Base code version number e 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 e GUI shows the GUI status running or stopped Phone shows the phone status running or stopped Download core dump file for troubleshooting when necessary Table 11 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 GXP1100 1105 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 GXP1100 1105 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 GXP1100 1105 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 primar
9. is announced Enter 12 digit new IP address if in Static IP Mode Same as Menu option 02 Same as Menu option 02 Same as Menu option 02 Enter 9 to go to the next selection in the list e PCMU e POMA e iLBC e G 726 e G 723 e G 729 Announces the Mac address of the unit Announces current Firmware Server IP address Enter 12 digit new IP address Announces current Config Server Path IP address Enter 12 digit new IP address Upgrade Protocol for firmware and configuration update Enter 9 to toggle between HTTP TFTP and HTTPS Firmware version information Firmware upgrade mode Enter 9 to toggle among the following three options 1 always check 2 check when pre suffix changes 3 never upgrade Enter the target IP address to make a direct IP call after dial tone See Make a Direct IP Call section Announces number of voice mails Press 9 to reboot the device Page 12 of 29 Last Updated 08 201 1 Innovative IF Voice amp Video Enter MAC address to restore factory default setting See Restore Factory Default Setting section Invalid Entry Automatically returns to Main Menu CONFIGURATION VIA WEB BROWSER The GXP1100 1105 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
10. 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 11 161 7xxxxxxxj Allow 311 611 and 911 or any 10 digit numbers with leading digits 1617 e Example 2 1900x 21617 xxxxxxx Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers e Example 3 1xxx 2 9 xxxxxx lt 2 011 gt x Allows any number with leading digit 1 followed by a 3 digit number followed by any number between 2 and 9 followed by any 7 digit number OR Allows any length of numbers with leading digit 2 replacing the 2 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 161 7 gt 2 9 xxxxxx 1 2 9 xx 2 9 xxxxxx 011 2 9 x 8469 11 Explanation of example rule reading from left to right e 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 3469 11 allow dialing special and emergency numbers 31
11. 01 1 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 IP Voice amp Video Select the ATA to report the log level Default is NONE The level is one of DEBUG INFO WARNING or ERROR Syslog 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 address error 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 SS
12. 05 supports Blind transfer Attended transfer and Auto Attended transfer 1 Blind Transfer During the call press TRAN button then dial the number and press the button to complete transfer of active call 2 Attended Transfer During the call press FLASH button to make another call and automatically place the ACTIVE LINE on HOLD Once the second call is established press TRAN key then the FLASH button to transfer the call Hang up the phone call after the call is transferred 3 Auto Attended Transfer Users need to enable Auto Attended Transfer under web GUI Advanced Settings Page first During the call press TRAN button It will bring up another line and automatically place the ACTIVE CALL on HOLD Enter the number and press SEND key to establish the second call Once the second call is established press TRAN button again and the call will be transferred Grandstream Networks Inc GXP1100 1105 User Manual Page 10 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 C sen Innovative IP Voice amp Video NOTE To transfer calls across SIP domains SIP service providers must support transfer across SIP domains 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 MSG button Voice Mail User ID has to be properly configured as the voice ma
13. 1 411 611 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 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 GXP1100 1105 User Manual Page 23 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 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 Refer To Use Target Contact E sen Innovative IP Voice amp Video The SIP Session Timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the session is terminated Session Expiration is the time in seconds at which the session is considered time
14. 100 1105 can be configured in two ways Firstly using the IVR MENU by the keypad on the phone secondly through embedded web configuration menu CONFIGURATION VIA IVR MENU GXP1100 1105 has a built in voice prompt menu for simple device configuration Pick up the handset and dial to use the IVR menu Table 9 GXP1100 1105 IVR Menu Definitions Main Menu 01 02 03 04 05 07 10 13 14 15 16 17 47 86 99 Grandstream Networks Inc Enter a Menu Option DHCP Mode PPPoE Mode Static IP Mode IP Address IP address Subnet IP address Gateway IP address DNS Server IP address Preferred Vocoder MAC Address Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade Direct IP Calling Voice Mail RESET GXP1100 1105 User Manual Firmware version 1 0 1 86 Press for the next menu option Press to return to the main menu Enter 01 05 07 10 17 47 86 or 99 Menu option Enter 9 to toggle the selection If users select Static IP Mode users need configure all the IP address information through menu 02 to 05 as below If users select Dynamic IP Mode the device will retrieve all IP address information from DHCP server automatically after user reboots the device The current WAN IP address
15. 8 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 GXP1100 1105 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 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 GXP1100 1105 User Manual Page 25 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream Innovative IP Voice amp Video iLBC Frame Size iLBC packet frame size Default is 20ms For Asterisk PBX 30ms might be required ILBC Payload Type Payload type for iLBC Default value is 97 The valid range is between 96 and 127 Special Feature Default is Standard Choose the selection to meet special requirements from Soft Switch vendors SAVING THE CONFIGURATION CHANGES After the user makes a change to the configuration press the
16. Am Innovative IP Voice amp Video Grandstream Networks Inc GXP1100 1105 Small Business IP Phone TABLE OF CONTENTS GXP1100 1105 USER MANUAL MELL CV cases nt 3 I E EE LE rc Y 4 BOUIPMENT PACK Ae EINER RR 4 CONINEC TING Y A SIN Mais PIDE RRORUS 4 vin tua EIE INMINENTE NOEREUNEEUS 4 bg IP NR 4 14 88 LL EO EV DOS 5 USING THE GXP1100 1105 esesesesessssoscscsescsesesesesesesessseososcsesesesesesesesesessoscsesesesesesesesesesessoscsesesesesesesesesesesesesesssseso 8 GETING FAMILIAR VITH IHE KEYPAD aah ose eects noes TE duci Etats U e D AR PIDEN LAS AR LI DD EE CAE EE 8 TT AAI EOIN EE TEMMC E 8 ANSWERING PHONE CALLS ri dao evuta ve cv pe ceu e uox egrave tk edet eve sy serus NENE Per eV vag ouk dascisdwvssalancuwadiseguudevsadenddieedevecdlenciwadsvaceuwes 10 PHONE FUNCTIONS DURING A PHONE CALL c cccsceccececcececcsceccscuccscecescscscscesescssescsceecsseecsseecscescscscscusescesesceseecs 10 GU URL WEE AD NR RTT A m B 11 CONFIGURATION GUIDE iiss vckcecesecccevesv
17. Firmware Server IP Address e Input the 12 digit firmware upgrade IP address For example If the firmware upgrade IP address is 10 0 50 191 010000050191 should be the input Web Configuration Interface To configure the Upgrade Server via the Web configuration interface open the web browser Enter the GXP1100 1105 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 The indicator on the top right corner will turn orange and red and then turn off which indicates the phone has restarted After a while the indicator will blink in red meaning the download is in process When download is done you will see the phone restarts again 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
18. L 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 GXP1100 1105 will ONLY use selected ring tones for particular Caller IDs For all other calls the GXP1100 1105 will use System Ring Tone When selected and no Caller ID is configured the selected ring tone will be used for all incoming calls oystem ring tone Default is North American standard Adjust system ring tone frequencies and cadences based on local telecom standard GXP1100 1105 User Manual Page 18 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 Call Progress Tones Disable Call Waiting Disable Call Waiting Tone Disable Direct IP Calls Use Quick IP Call Mode Disable Transfer Auto Attended Transfer Grandstream Networks Inc andstream Innovative IF 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
19. Random Port Keep alive interval Use NAT IP STUN Server Firmware Upgrade and Provisioning XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Upgrade Via Firmware Server Path Config Server Path Firmware File Prefix Postfix Config File Prefix Postfix Allow DHCP Option 43 and Option 66 to override server Grandstream Networks Inc andstream Innovative IP Voice amp Video This parameter when set to Yes will force random generation of both the local SIP and RIP ports This is usually necessary when multiple GXPs are behind the same NAT Default is No This parameter specifies how often the GXP1100 1105 sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open Default is 20 seconds NAT IP address used in SIP SDP message Default is blank IP address or Domain name of the STUN server STUN resolution result will display in the STATUS page of the Web UI 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 especia
20. SAGE i with LED indicator 4 Programmable Hard 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 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 Adaptive jitter buffer control patent pending and packet delay and loss concealment HD audio handset with HD wideband audio codecs for excellent double sss talk performance 7 Telephony Features Web interface GUI support for anonymous call using privacy header MLS multi language support Voicemail indicator downloadable custom ring tones call hold call transfer attended blind call forward call waiting mute redial caller ID display or block and volume control e Dial plan prefix dial plan support off hook auto dial and speed dial Network and Provisioning Via Keypad IVR Web browser or secure AES encrypted central configuration file manual or dynamic host configuration protocol DHCP
21. TbRM EX FUIT ari PEIA PavV SU Aon ER xus 4 Table 3 GXP1100 1 105 Feature Guide scixentsskntoaks khpRR XR tk ETE ERA Ye FER Rs tC TbR MEX FUpEP EEXT Ex Pav ERU AoPKPPR C Ix 5 Table 4 GXP1100 1105 Key Features in a Glance eesssseeeeseseseeeennennnene nennen 5 Table 5 GXP1100 1105 Hardware Specifications cccccccccccssscccceeseceeceeeeeeseeeceeseeeeeeseeeeesaaeees 5 Table 6 GXP1100 1105 Technical Specifications cc cc cecccccseecccceeeeceeceeeeeeseeecesseeeeeeseeeeessaees 6 Table 7 GXP1100 1105 Keypad Buttons 2 0 0 ccccccccccccceeecceceeeeeeeeeeeceeeeseeeeseeeeeesseeeeeeseeeeeesseeees 7 Table 8 GXP1100 1105 Call Features seeeeessseessssseessseseeennennnn nennen nnne nnns 11 Table 9 GXP1100 1105 IVR Menu Detfinitions lseeeeeseseeeseeeeeennnennnnnnn nnn 12 Table 10 Device Configuration Status 2 0 0 0 ccccccceecccceeeececeeeeeeeseeeeeseeeeeesseaeceeseeeeeesaaeeeessaaeees 14 Table 11 Device Configuration Settings Basic Settings ccccceccccceesececeeeeceeceeeeeeseeeeeeseeeeees 14 Table 12 Device Configuration Settings Advanced Settings eeseseeeeessssess 15 Table 13 SIP Account Settings seeeeesssesssssessssssesseseeee nennen nennen nennen annes n annis 20 Grandstream Networks Inc GXP1100 1105 User Manual Page 1 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream
22. XP1100 1105 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 GXP1100 1105 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 GXP1100 1105 boots up or re booted it will request a configuration file named cfgxxxxxxxxXxxx followed by a request for configuration XML file named cfgxxxxxxxxxxxx xml where xxxxxxxxxxxx is the MAC address of the device i e cfg000b820102ab The configuration file name should be in lower cases For more details on XML provisioning please refer to http www grandstream com support
23. ccount 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 then for Attended Transfer the Refer To header uses the transferred target s Contact header information Grandstream Networks Inc GXP1100 1105 User Manual Page 24o0f29 Grandstream Networks Inc Page 24 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 Preferred Vocoder SRTP Mode Symmetric RTP Silence Suppression Voice Frames per TX No Key Entry Timeout Use as Dial Key G723 Rate G726 32 Packing Mode Grandstream Networks Inc andstream Innovative IP Voice amp Video GXP1100 1105 supports up to 7 different Vocoder types including G 711 a u also known as PCMU PCMA G 723 1 G 729A B G 726 32 ILBC G 722 wide band Configure Vocoders in a preference list that is included with the same preference order in SDP message Enter the first Vocoder in this list by choosing the appropriate option in Choice 1 Similarly enter the last Vocoder in this list by choosing the appropriate option in Choice 8 Enable SRTP mode based on selection Default is No Selects whether or not symmetric RTP is supported Thi
24. d out provided no successful session refresh transaction occurs beforehand The default value is 180 seconds Defines the minimum session expiration in seconds Default is 90 seconds If set to Yes the phone will use session timer when it makes outbound calls if remote party supports session timer If selecting Yes the phone will use session timer when it receives inbound calls with session timer request If set to Yes the phone will use session timer even if the remote party does not support this feature If set to No the session timer is enabled only when the remote party supports this feature To turn off Session Timer select No for Caller Request Timer Callee Request Timer and Force Timer As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy server as the refresher As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is required to support PSTN inter networking There are 4 uniquely defined ring tones e One 1 System Ring Tone when selected all calls will ring with system ring tone e Three 3 Customer Ring Tones when selected incoming calls from designated a
25. dvanced Functionality Customized downloadable ring tones SRTP SIP over TLS multi language support wall mountable AES encryption and etc Table 5 GXP1100 1105 Hardware Specifications GXP1 100 1105 LAN Interface 10 100 Mbps Full Half Duplex Ethernet port with auto detection ass t S Integrated PoE GXP1105 only Graphic LCD Display NA Expansion Module i N A Lines 2 calls via FLASH key Universal Switching Input 1 100 M40VAC 50 60 H7 ee ss s lt iC Power Adaptor Output 5 VDC 800mA 4 0 W UL certified Dimension gt gt n 201mm W x 154mm L x 78mm D Grandstream Networks Inc GXP1100 1105 User Manual Page 5 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 ndstream Innovative IP Voice amp Video Weight Unit weight 0 6KG Package weight 1 0KG GXP1100 0 9KG GXP1105 Temperature 32 104 F 0 40C MEM Humidty d 10 90 non condensing On Compliance 3 FCCPat15 CFR47 ClasB 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 GXP1100 1105 Technical Specifications Lines 3 2 Calls with 1 SIP account Protocol Support Support SIP 2 0 TCP UDP IP PPPoE RTP SRTP by SDES HTTP ARP RARP ICMP DNS DHCP NTP TFTP SIMPLE PRESENCE i protocols TR 069 802 1x Feature Keys HOLD FLASH TRANSFER MUTE VOLUME REDIAL and MES
26. ecceneciccestcancscccedeuscedacectceisaieeseveuectcecaivsvociesvescasesiseesdeeneveeseiewsoodacxcesessesseseaks 12 CONFIGURATION VIAIVR MENU oes otic ccacemceice Sagan se letras Sene EE AE E EANA 12 CONFIGURATION VIA WEB BROWSER ssecesdcesincsacsndedecedehonsaceetnenwoiwideeenceeeasincseaeedodecnestedsarescaswdeaedaaseauadodeeatebedsaxescaueeees 13 SAVING THE CONFIGURATION CHANGES cccsceccscescsceccsceccscuccscescececescscescscescscscscesescnsescesescesescsseecesscssscscesescnsesces 26 REBOOTING THE PHONE REMOTELY ee eoe pode ue ave ERUL Pu QE Cue FUCE Vue bei guae ue C Toa LEUR Su Ce abet CENERE ENE NEA EIE ENEE 26 SOFTWARE UPGRADE amp CUSTOMIZATIONN ee eee eee eee ee eee eere eee seee sese eee sese sees esee esee esee seseeo sees soe 27 FIRMWARE UPGRADE THROUGH TFTP HT TP eese cececcscnccececcccececscscscesescecescesesceceececescscscscesescesesceseses 27 CONFIGURATION FILE DOWNLOAD ccceccsceccsceccscecceceecececcecscscccecuccecucesceceecesacceceecscescsctscecssescecescecesceceecssesetcess 28 RESTORE FACTORY DEFAULT SETTING eee eee eee eee eee eroe eee ee eese eeeee eese see ees esses eese see esses ese essesee oos 29 TABLE OF TABLES GXP1100 1105 USER MANUAL Table 1 Equipment Packaging cccccccccssececceseceeceeseceeceeeceeceeecesseeeceeseaeeeesseecesseueeeesaaeeeesseeeeees 4 Table 2 X3XP1100 1105 CONNECTS csssaxsksiuti thbX Ene E i CEURR S RIE PESE ERA xe FER RU ct
27. il number under web GUI gt Account 1 An IVR will prompt the user through the process of message retrieval NOTE Users can press to the IVR menu and then enter 86 to hear the number of new voice messages CALL FEATURES The GXP1100 1105 supports traditional and advanced telephony features including caller ID caller ID w name call forward transfer and etc Table 8 GXP1100 1105 Call Features Key Call Features 30 Block Caller ID for all subsequent calls 31 Send Caller ID for all subsequent calls 67 Block Caller ID per call 82 Send Caller ID per call 70 Disable Call Waiting per Call 71 Enable Call Waiting per Call 72 Unconditional Call Forward Dial 72 and the forwarding number followed by Wait for the call to hang up 73 Cancel Unconditional Call Forward Dial 73 and wait for the call to hang up 90 Busy Call Forward Dial 90 and the forwarding number followed by Wait for the call to hang up 91 Cancel Busy Call Forward Dial 91 and wait for the call to hang up 92 Delayed Call Forward Dial 92 and the forwarding number followed by Wait for the call to hang up 93 Cancel Delayed Call Forward Dial 93 and wait for the call to hang up Grandstream Networks Inc GXP1100 1105 User Manual Page 11 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 Configuration Guide E sen Innovative IP Voice amp Video The GXP1
28. ing 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 23 and 03 and 003 results in the same call call 192 168 0 3 NOTE 1 The will represent colon in Direct IP Call instead of Send key as in normal phone call 2 f you have a SIP Server configured a Direct IP IP still works If you are using STUN the Direct IP IP call will also use STUN Configure the Use Random Port to No when completing Direct IP calls ANSWERING PHONE CALLS Receiving Calls 1 Incoming single call Phone rings with selected ring tone Answer call by taking Handset off hook 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 the FLASH button The current active call will be put on hold PHONE FUNCTIONS DURING A PHONE CALL Call Waiting Call Hold 1 Hold Place a call on hold by pressing the HOLD button 2 Resume Press the HOLD button again to resume 3 Multiple Calls Automatically place ACTIVE call on hold or switch between two calls by pressing the FLASH button Call Waiting tone stutter tone will be audible when line is in use Mute Press the MUTE button to mute unmute the microphone Call Transfer GXP1100 11
29. ller ID display The default is set to A Record If user wishes to locate the server by DNS SRV the user may select SRV or NATPTR SRV When Use Configured IP option is selected if SIP server is configured as domain name phone will not send DNS query but use Primary IP or Secondary IP to send sip message if at least one of them are not empty This option applies only if Use Configured IP is selected the phone will send DNS query to the Primary IP Insert IP address here Insert the first back up IP here Insert the second back up IP here Select Tel URI as Disabled User Phone or Enabled Default is Disabled This parameter controls sending REGISTER messages to the proxy server The default setting is Yes Default is No If set to Yes the SIP user s registration information will be cleared on reboot This parameter allows user to specify the time frequency in minutes that GXP1100 1105 refreshes its registration with the specified registrar The default interval is 60 minutes The maximum interval is 65 535 minutes about 45 days This parameter allows user to specify the time frequency in seconds that GXP1100 1105 sends out a re registration request before the Register Expiration By default is 0 second 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 respecti
30. lly the power supply as this will damage the device The password used for encrypting the XML configuration file using OpenSSL This is required for the phone to decrypt the encrypted XML configuration file The user name for the HTTP HTTPS server The password for the HTTP HTTPS server This field allows the user to choose the firmware upgrade method TFTP HTTP or HTTPS Defines the server path for the firmware server It can be different from the Configuration server which is used for provisioning Defines the server path for provisioning it can be different from the firmware server Default is blank If configured GXP1100 1105 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 GXP1100 1105 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 GXP1100 1105 User Manual Firmware version 1 0 1 86 Page 16 of 29 Last Updated 08 201 1 Automatic Upgrade Authenticate Conf File Enable TR 069 ACS URL TR 069 Username TR 069 Password Periodic Inform Enable Periodic Info
31. mapped public IP address and port in all of its SIP and SDP messages If selecting Keep Alive with no specified STUN server the GXP1 100 1105 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 Default is No When set to Yes a SUBSCRIBE for Registration will be sent periodically Enable Presence feature SIP Extension to notify SIP server that the unit is behind the NAT Firewall When configured user can access messages by pressing MSG button This ID is usually the VM portal access number This parameter specifies the mechanism to transmit DTMF digit There are 3 supported modes in audio which means 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 GXP1100 1105 User Manual Grandstream Networks Inc Page 22 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 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
32. menu e Enter 47 for Direct IP Call After hearing Direct IP Calling the dial tone will be heard again e Enter the target IP address Please see example below e Wait for about 4 seconds and the phone will initiate the call For example If the target IP address is 192 168 1 60 and the port is 5062 e g 192 168 1 60 5062 input the following 192 168 1 60 5062 The key represents the dot the key represents colon Wait for about 4 seconds and the phone will initiate the call The GXP1100 1105 also supports Quick IP Call mode This enables the phone to make direct IP calls using only the last few digits last octet of the target phone s IP number This is possible only if both phones are in under the same LAN VPN This simulates a PBX function using the CMSA CD without a SIP server Controlled static IP usage is recommended 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 Then take the handset offhook and dial Grandstream Networks Inc GXP1100 1105 User Manual Page 9 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream Innovative IP Voice amp Video xxx where x is 0 9 and xxx lt 255 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 numbers xx or x are also valid The lead
33. not have a local TFTP HTTP server we provide a HTTP server on the public Internet for users to download the latest firmware upgrade automatically Please check the Support Download section of our website to obtain this HTTP server IP address http www grandstream com support firmware Alternatively download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades A free Windows version TFTP server is available http support solarwinds net updates New customerFree cfm Grandstream Networks Inc GXP1100 1105 User Manual Page 27 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 andstream Innovative IP Voice amp Video Instructions for Local TFTP Upgrade 1 Unzip the file and put all of them under the root directory of the TFTP server 2 he PC running the TFTP server and the GXP1100 1105 should be in the same LAN segment 3 Go to File gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade 4 Start the TFTP server 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 GXP1100 1105 phone boots up it will send TFTP or HTTP request to download configuration file cfgOO0b82xxxxxx where 000b82xxxxxx is the MAC address of the G
34. ore 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 network settings time settings multi purpose keys and etc can be set up here Additional functions available to administrators are e Advanced Settings To set advanced network settings codec settings language settings and etc e Account To configure the SIP account Grandstream Networks Inc GXP1100 1105 User Manual Page 13 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream Innovative IP Voice amp Video Table 10 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 GXP1100 1105 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
35. rent power adaptor with the GXP1100 1105 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 GXP1100 1105 User Manual Page 3 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 andstream Innovative IP Voice amp Video Installation EQUIPMENT PACKAGING Table 1 Equipment Packaging QGXPf 100110 Handset Yes Yes pes CONNECTING YOUR PHONE The connectors of the GXP1100 1105 are located on the bottom of the device Table 2 GXP1100 1105 Connectors LAN 10 100Mbps RJ 45 port for LAN uplink connection integrated PoE GXP1105 only Power Jack 5V DC power port UL Certified Handset Jack HJ9 SAFETY COMPLIANCES The GXP1100 1105 phone complies with FCC CE and various safety standards The GXP1100 1105 power adaptor is compliant with the UL standard Please use the universal power adaptor provided with the GXP1100 1105 package only The manufacturer s warranty does not cover damages to the phone caused by unsupported power adaptors WARRANTY If you purchased your GXP1100 1105 from a reseller please contact the company where you purchased your phone for replacement repair or refund If you purchased
36. rm Interval Connection Request Username Connection Request Password Authentication Method Connection Request Port Offhook Auto Dial Syslog Server Grandstream Networks Inc andstream Innovative IF Voice amp Video This function is used by ITSP End user should NOT touch these parameters Default is No Choose Yes to enable automatic HTTP upgrade and provisioning In Check for upgrade every field enter the number of minutes to check the HTTP server for firmware upgrade or configuration changes When set to No the phone will only perform HTTP upgrade and configuration check once at boot up Default is No If set to Yes configuration file would be authenticated before acceptance End user should use default setting Default is No URL for TR 069 Auto Configuration Servers ACS Enter username for TR 069 Enter password for TR 069 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 To configure a User ID extension to dial automatically when the phone is taken offhook The IP address or URL of System log server This feature is especially useful for ITSPs GXP1100 1105 User Manual Page 17 of 29 Firmware version 1 0 1 86 Last Updated 08 2
37. s 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 x10ms frames for G 723 it is 32 x30ms frames for G 729 G 72
38. s to be configured as Speed Dial and with the correct name and user ID under Web GUI gt Basic Settings configuration e Take handset off hook e Press the configured Speed Dial key 4 VIA CALL RETURN On the GXP1100 1105 the Multiple Purpose Key programmable hard key has to be configured as Call Return under Web GUI gt Basic Settings configuration No user name and user ID has to be set on the Multiple Purpose Key for Call Return After pressing the Call Return key the last answered number will be dialed out e Take handset off hook e Press the configured Call Return key NOTE 1 Dial tone occurs after the handset is off hook 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 2 If there are digits pressed the SEND key will work as SEND instead of REDIAL 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 Take handset off e After hearing the dial tone press to enter the IVR
39. 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 GXP1100 1105 User Manual Page 4 of 29 Firmware version 1 0 1 86 Last Updated 08 2011 ndstream Innovative IP Voice amp Video Product Overview Table 3 GXP1100 1105 Feature Guide Features GXP1100 1105 LCDDispay jo NA Number of Lines 1 Programmable Keys 4 Extension Module NA gt Table 4 GXP1100 1105 Key Features in a Glance Features _ Benefits Open Standards SIP RFC3261 TCP IP UDP RTP HTTP HTTPS ARP RARP ICMP Compatibility DNS A record SRV and NAPTR DHCP both client and server PPPoE TELNET TFTP NTP STUN SIMPLE SIP over TLS 802 1x EN uu E TR 069 BS I Superb Audio Quality Advanced Digital Signal Processing DSP Silence Suppression VAD BI CNG AGC MEN Network Interfaces E Single 10 100 Mbps Ethernet port integrated PoE GXP1105 only Feature Rich Traditional voice features including call waiting hold transfer forward MCN block off hook auto dial click to dial flexible dial plan and etc B Advanced Features L2 calls with 1 SIP account 4 programmable hard keys 7 dedicated function keys for HOLD FLASH TRANSFER MUTE VOLUME REDIAL and MESSAGE with LED indicator A
40. tional 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 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 Table 12 Device Configuration Settings Advanced Settings Admin Password Layer 3 QoS Layer 2 QoS Local RTP port 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 O This parameter defines the local RTP 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 channel 1 will use port value 2 for RTP Local RTP port ranges from 1024 to 65400 and must be even The default value is 5004 GXP1100 1105 User Manual Page 15 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 Use
41. ts like 22 in the MAC you need to wait for 4 seconds to continue to key in another 2 Step 3 Once the MAC address is correctly input the phone will reboot Otherwise it will announce Invalid Entry and exit to the main menu Grandstream Networks Inc GXP1100 1105 User Manual Page 29 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1
42. ve voice mails or other messages Programmable hard key It can be used for Multiple Purpose Key Speed dial Dial DTMF or Call Return Mute an active call Dial out the number and can be used as SEND or REDIAL Adjust volume by pressing or 0 9 Standard phone keypad press key to send call press key to for IVR functions MAKING PHONE CALLS 2 Calls with 1 SIP Account GXP1100 1105 can support up to two lines virtually mapped to a SIP account By picking up the handset the GXP1100 1105 will be in off hook state and the dial tone will be heard To make a call dial out the number with the current line During the call users can press the FLASH button to hold the current call and make answer another call Users can switch the two lines by pressing the FLASH button Completing Calls The GXP1100 1105 allows you to make phone calls by picking up the handset There are four ways to complete calls 1 DIAL To make a phone call e Take handset off hook e The line will have a dial tone e Enter the phone number e Press or SEND button to send 2 REDIAL To redial the last dialed phone number e Take handset off hook Grandstream Networks Inc GXP1100 1105 User Manual Page 8 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 andstream Innovative IP Voice amp Video e Press the SEND key 3 VIA SPEED DIAL On the GXP1100 1105 the Multiple Purpose Key programmable hard key ha
43. vely Retry registration if the process failed Default is 20 seconds RFC 3261 SIP T1 timer Default is 0 5 second HFC 3261 SIP T2 timer Default is 4 seconds Choose SIP Transport between UDP and TCP Default is UDP select sip or sips Default is sips Enable to use actual ephemeral port in contact with TCP TLS Default is No Enable to check the domain certificate Default is No The SIP Extension notifies the SIP server that it is behind a NAT firewall GXP1100 1105 User Manual Page 21 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 Validate Incoming Messages Support SIP Instance ID NAT Traversal SUBSCRIBE for MWI SUBSCRIBE for Registration PUBLISH for Presence Proxy Require Voice Mail UserlD Send DTMF DTMF Payload Type Early Dial Dial Plan Prefix andstream Innovative IF Voice amp Video This configuration selects whether or not the incoming messages should be validated Selects whether or not SIP Instance ID is supported 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
44. works Inc 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 GXP1100 1105 to register to SIP servers of ITSP GXP1100 1105 User Manual Page 20 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 Name DNS Mode Primary IP Backup IP 1 Backup IP 2 TEL URL SIP Registration Unregister on Reboot Register Expiration 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 Certificate Remove OBP from Route Grandstream Networks Inc andstream Innovative IP Voice amp Video SIP service subscriber s name that is used for Ca
45. y DNS Server 2 secondary and Preferred DNS Server These fields are set to zero by default Inc GXP1100 1105 User Manual Page 14 of 29 Firmware version 1 0 1 86 Last Updated 08 201 1 802 1x Mode Multi Purpose Key X Time Zone Self Defined Time Zone andstream Innovative IP Voice amp Video This option allows the user to enable disable 802 1x mode on the phone The default value is disabled To enable 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 These options are used to assign a function to the corresponding multiple purpose key Options available are 1 Speed Dial 2 Dial DTMF 3 Call Return Call the last answered number No UserID has to be configured in the MPK when using Call Return 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 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 Interna
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