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Grandstream GXP1100/1105 IP Phone User Manual

1. wm wm wm em wm wm wm wm e e e e p ee we we ew ew we ew wm ew wm ew ew ew ew ew ew wm ew wm wm ew wm wm ew ew wm wm ew wm wm wm wm wm ew wm wm ew ew FCC Part 15 CFR 47 Class B EN55022 Class B EN55024 EN61000 3 2 EN61000 3 3 EN 60950 1 AS NZS CISPR 22 Class B AS NZS CISPR 24 RoHS UL 60950 power adapter Compliance Table 6 GXP1100 1105 Technical Specifications Audio Features Telephony Features Network and Provisioning Firmware Upgrades 2 calls with 1 SIP account am gt mee ew ee ew ew ew ew ww ew we gt ew gt GE ew ew ew gt Bw gt OD Bw ww ew gt we Pew ew ew ww ew ww em www ew wwe ww ew ew ww ew ew ww www ew wwe ew mw ew mw wm gt gt ew mew gt mw eww ew gt gt wm ew ew gt ew gt wm ew gt we ew gt mw ew gt ew ew gt ew we gt ww gt ew gt gt ew ow gt Support SIP3261 TCP IP UDP RTP SRTP by SDES HTTP HTTPS ARP RARP ICMP DNS A record SRV and NAPTR DHCP PPPoE TELNET TFTP NTP STUN TR 069 802 1x ss 8s we nnmnnn HOLD FLASH TRANSFER MUTE VOLUME SEND and MESSAGE with LED indicator 4 Programmable Hard Keys 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 10
2. cccccccsesccccsesseceeeeeeeseceesceeseeeeeeeeeseeesaeaeeeess 5 Table 6 GXP1100 1105 Technical Specifications rrrrrrrrnnnnrorrrrnnnnnvrrrrnnnnnornnrnnnnnvnnnnnnnnneennnnenee 6 Table 7 GXP1100 1105 Keypad Definitions rrnnrrrrnnnnnrnnannrrrnnnnrrrnnnnnenrnnnnrnnnnnnrnnnnnnrennnnnnennnnnsen 7 Table 8 GXP1100 1105 Call Features rrrrrrrrnnrrorrrrnnnrnvvvrrnnnnnvennrnnnnnsennrnnnnnsennnnnnnneennnnnnssennnnne 13 Table 9 GXP1100 1105 IVR Menu Definitions rrrrrnnrrrrnnnnrrrnnnnnenrnnnnrnnnnnnrennnnnennnnnnrnnnnnnnennnnnne 15 Table 10 Device Configuration Status cccccccccccssssecceeceesseceeeeeeeeeeeeceeesceeeseaeaeeessaeaeeeeeeaaaeess 17 Table 11 Device Configuration Settings Basic Settings ccccccesscccecsssscceeseeeseceeeeeeeeeeeeseeeees 17 Table 12 Device Configuration Settings Advanced Settings ccccccseeeeeeessseeseeeeeeseeeeeeeeeeees 18 Table 13 SIP Account Settings rrrrrrrerrrnnnrnorrrrnnnnnvonrrnnnnnrnrnnnnnsennrnnnnnsnnnnnnnsnsennnnnnsssennnnnnssennennnn 24 Grandstream Networks Inc GXP1100 1105 User Manual Page 1 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnevative IP Voice amp Video GUI INTERFACE EXAMPLES GXP1100 1105 USER MANUAL http www grandstream com products gxp_series general documents gxp1 10x_gui Zip Screenshot of Con
3. nnovalive 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 VolP 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 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 ah Se Example if the MAC address is 00068200e895 it should be key in as 0002228200333395 NOTE If there are digits like 22 in the MAG 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 34 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299
4. Ae Innovative IP Voice amp Video Grandstream Networks Inc GXP1100 1105 Small Business IP Phone VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 TABLE OF CONTENTS GXP1100 1105 USER MANUAL MELC OMI os sca cec cose csetsecnnecacseccassevecesssvewavacsaesaevocuswesevenascececeswovecsuoneeteaus svanancececuswoceseunssctdeseevswaecsaunscececusesessesescscececesecss 3 INSTALLATION sass vic cecesccesesevavecacesscceesavesecnccsenccetenccesscusesscuesconacssacaedencveseczenaceateuseacesdeesseaccsaveisssacsaedeeaccssteapaseaseceneneses 4 EQUIPMENT PACKAGING enccnsevencenporiensovsvonesentoeneicusseusteutuamtlons seston EN 4 CONNECTING Y OUR PHONE Danae nen nee ne ne ee ee ee 4 SAFETY GONE ne re ee eer 4 VNR 4 FRODON ENG 5 USING THE GXP1100 1105 esecsesesecsesesccseseosesesccseseoccseseoseseososesesecseseoscseseosesesecsesesecseseoseseseosesesscsescoseseseesesesseseseo 8 GETTING FTV 8 ND PENN ape een nen en en ee ee ee ee ee ee 8 NVE TVEN 10 PHONE FUNCTIONS DURING A PHONE CALL s esnevevnvnevnvnennvnennnnensnnensnnensnnenennensnnnnennnnennnnensnnensnnensnnenennenennenennnnenunnensenene 10 Oe TRB E E EAEI ic eure nse aise scene nig ee fk ous asda ares aaa A E ETE E E A EET 13 CONFIGURATION GUIDE RE 15 CONFIGURATION VA IVR MENU coerce E E N E E E 15 CONFIGURATION VIA WEB BROWSER sccccscsccscsccececcecsccscscoscecescscscscascscnsescsseecssoecssescssescssescessescasescasescesencesecs 16
5. No If set to Yes 3 way conference will be disabled Enable MPK sending Default is No If set to Yes Multi Purpose keys can be sent as DTMF during the call DTMF Enable FLASH key as Default is No If set to Yes FLASH key can be used to establish 3 way conference CONF Disable Transfer Default is No If set to Yes transfer will be disabled Auto Attended Default is No If set to Yes the phone will use attended transfer by default Transfer Grandstream Networks Inc GXP1100 1105 User Manual Page 23 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Display Language Allows user to choose preferred display language in web Ul and key pad Ul Currently the phone supports these languages English Simplified Chinese Traditional Chinese Korean Japanese Italian Spanish French German Portuguese Russian Croatian Hungarian Polish Slovenian Arabic Hebrew 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
6. SAVING THE CONFIGURATION CHANGES sesesesessesesesesesesreresescsesesresesesesesesresesesesesessesescsosesoesesesosesesossesesosesesresesese 31 REBOOTING THE PHONE REMOTELY ccccsceccsceccsceccsceccsceccscscascscascscsseecssoscscescecscscascscasescssescssescesescessceseseesasenees 31 SOFTWARE UPGRADE amp CUSTOMIZATION 1 ccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccccecees 32 FIRMWARE UPGRADE THROUGH TFETP HT TP 1 2 0 0 ccc ccc ccc ceccecscecceccsccsceccecceccscescscecescescscscecescescescssescescescesceseeses 32 CONFIGURATION FILE DOWNLOAD rernenevnvnevnenevnenennenennenennnnennnnennnnennnnennnnenennenennenennenennenennnnennnnennnnenunnenennensnnenenneneenen 33 RESTORE FACTORY DEFAULT SETTING 2 ccccccccscccccccccsccccccccccccccccccccccccccccccccccscccccccccccccccscccccees 34 TABLE OF TABLES GXP1100 1105 USER MANUAL Table 1 Equipment Packaging cccccccccccssscceceeeececeeececeeeceeceeeeesseseceseeaeceessuueeeseasecesseaeeessaeeeeeeas 4 Table 2 GXP1100 1105 Connectors rrrrrrnnnnrnnrnnnnnovvrrnnnnnvrnrrnnnnnennnnnnnsnnrnnnnnnsennnnnnnneennnnnnnennnnnnenene 4 Table 3 GXP1100 1105 Feature Guide rrrerrrnnnnnoorrnnnnnnrnvnnnnnnnnnnnnnnnsnnnnnnnnnsrnnnnnnnnsennnnnnnnennnnnenee 5 Table 4 GXP1100 1105 Key Features in a Glance rrrrrrnnnnnornnnnnnnnvrnrnnnnnonnnnnnnnrennnnnnnsrennnnnenee 5 Table 5 GXP1100 1105 Hardware Specifications
7. VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS 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 7 Adaptive jitter buffer control patent pending and packet delay and loss concealment HD audio handset with HD wideband audio codecs for excellent double talk performance see eee eee ee ee ee ee ee ee eee ee eee ee ee ew ee pS ew eee ee ew eee ew eee ee ew eee ew ee we ee 8 ee ew wee ew ee ew ew ew eee 8 ee 8 ee ow ee 8 8 ee ew ee ew ee es ee es ew ee ew ee ew ee es 8 ee ew ee ee oe 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 Dial plan prefix dial plan support off hook auto dial and speed dial Via Keypad IVR Web browser or secure AES encrypted central configuration file manual or dynamic host configuration protocol DHCP network setup Support NAT traversal using IETF STUN and Symmetric RTP Support for IEEE 802 1p Q tagging VLAN
8. amp Video Secondary SIP Server This field contains the URL or the IP address of a second SIP server When this field is configured GXP110x will send out Registration requests and Subscribe messages except for message waiting to the SIP Server and Secondary SIP Server for the same account When making a call GXP110x will use the registered primary SIP server first If this primary SIP Server is not available the registered Secondary SIP Server will be used If the primary SIP Server is not registered but Secondary SIP Server can be registered GXP110x will use the Secondary SIP Server directly when making a call Note Please do not configure duplicate SIP Server address in SIP server and Secondary SIP Server Outbound Proxy 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 SIP User ID User account information provided by VoIP service provider ITSP either an actual phone number or formatted like one Authenticate ID SIP service subscribers Authenticate ID used for authentication It can be identical to or different from SIP User ID Authenticate Password SIP service subscriber s account password for GXP1100 1105 to register to SIP servers of
9. Forward Pick up the handset Dial 92 followed by forwarding number Press or SEND key The call will hang up automatically 93 Cancel Delayed Call Forward Pick up the handset Dial 91 A short tone will be heard Wait for the call to hang up Grandstream Networks Inc GXP1100 1105 User Manual Page 14 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Configuration Guide The GXP1100 1105 can be configured in two ways IVR MENU from the keypad Web GUI embedded on the phone 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 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 GXP1100 1105 User Manual Firmware version 1 0 1 110 Gs Innovative IP Voice amp Video P
10. be established 2 Split in Conference During the 3 way conference press HOLD key to end the conference call Press HOLD key again and GXP110x will go back to the 2 way conversation with the line when establishing the conference call Press FLASH to toggle between 2 lines Users could re establish conference call by pressing the 3 way Conference Multi Purpose Key again 3 End Conference Press HOLD key to end the conference call Users could resume call with one of the party or re establish conference Or users could simply hang up the call to terminate the conference call To use FLASH Key establish 3 way conference call uses need to login the GXP110x web GUI gt Settings gt Advanced Settings set Enable FLASH key as CONF to Yes Update and reboot the phone from the web GUI Then follow the steps below to host the 3 way conference 1 Initiate a Conference Call Initiate and establish two active calls with two parties from GXP110x Press the FLASH key 3 way conference will be established 2 Split in Conference During the 3 way conference press HOLD key to end the conference call Press HOLD key again and GXP110x will go back to the 2 way conversation with the line when establishing the conference call Users could re establish conference call by pressing the FLASH key again 3 End Conference Press HOLD key to end the conference call Users could resume call with one of the party or re
11. 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 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video Product Overview Table 3 GXP1100 1105 Feature Guide Features GXP1100 1105 LCD Display NA Number of Lines n Programmable Keys Extension Module NA gt Table 4 GXP1100 1105 Key Features in a Glance Features o Benefits ee Open Standards SIP RFC3261 TCP IP UDP RTP HTTP HTTPS ARP RARP ICMP Compatibility DNS A record SRV and NAPTR DHCP PPPoE TELNET TFTP NTP TENE EE eee Superb Audio Quality Advanced Digital Signal Processing DSP Silence Suppression VAD eee nent ING A eee Network Interfaces Single 10 100 Mbps Ethernet port integrated PoE GXP1105 only Feature Rich Traditional voice features including call waiting hold transfer forward block off hook auto dial click to dial flexible dial plan and etc Advanced Features 2 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 it pm af int inc
12. establish conference Or users could simply hang up the call to terminate the conference call NOTE 1 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 2 The option Disable Conference has to be set to No to establish conference on GXP110x Grandstream Networks Inc GXP1100 1105 User Manual Page 12 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 C isten Innovative IP Voice amp Video 3 Users could select one of the ways above MPK option or FLASH key to establish conference call When using Enable FLASH key as CONF FLASH key will not be available to switch lines multiple times 4 When using FLASH key to establish conference GXP110x must be the initiator to establish the 2 way conversation in both calls 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 key Voice Mail User ID has to be properly configured as the voice mail number under web GUI gt Account 1 An IVR will prompt the user through the process of mes
13. is 20 x10ms frames for G 723 it is 32 x30ms frames for G 729 G 728 64 x10ms and 64 x2 5ms frames respectively Please be careful when editing these parameters Adjusting these parameters will also change the dynamic jitter buffer The 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 No Key Entry Timeout Default is 4 seconds Use as Dial Key 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 Grandstream Networks Inc GXP1100 1105 User Manual Page 30 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video G723 Rate Encoding rate for G723 codec By default 6 3kbps rate is set G726 32 Packing Mode Select ITU or IETF for G726 32 packing mode 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
14. 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 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 GE siren nnovative IP Voice Video Installation EQUIPMENT PACKAGING Table 1 Equipment Packaging 0 GXPM00M10 gt Handset Yes 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 RJ9 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 the product directly from Grandstream
15. 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 off8 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 anda pause of OFF ms and then repeat the pattern Up to three cadences are supported Disable Call Waiting Default is No If set to Yes the call waiting feature will be disabled Disable Call Default is No If set to Yes the call waiting tone will be disabled Waiting Tone Disable Direct IP Calls Default is No If set to Yes direct IP calls will be disabled Use Quick IP Call Dial an IP address under the same LAN VPN segment by entering the last octet in the Mode IP address In the Advanced Settings page there is an option Use Quick IP call mode Default setting is No When set to Yes and XXX Is dialed where X is 0 9 and XXX lt 255 phone will make direct IP call to aaa bbb ccc XXX where aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask XX or X are also valid so leading O is not required but OK See Quick IP Call Mode for details Disable Conference Default is
16. the firmware server It can be different from the Configuration server which is used for provisioning For example firmware mycompany com 6688 Grandstream 1 0 0 6 Config Server Path Defines the server path for provisioning it can be different from the firmware server For example provisioning mycompany com 6688 Grandstream gxp1 100 Firmware File Default is blank If configured GXP110x will request the firmware file with the Prefix Postfix prefix postfix and only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone Note This setting is useful for ITSPs End user should keep it blank Config File Default is blank If configured GXP110x will request the config file with the Prefix Postfix prefix postfix and only the file with the matching encrypted prefix will be downloaded and flashed into the phone Note This setting is useful for ITSPs End user should keep it blank Allow DHCP Option 43 Default is Yes This allows device gets provisioned from the server automatically and Option 66 to override server Grandstream Networks Inc GXP1100 1105 User Manual Page 20 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 C isten Innovative IP Voice amp Video Automatic Upgrade Default is No Choose Yes to enable automatic HTTP upgrade and provisioning In Check for upgrade every
17. 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 Download Device Download the current device configuration txt file In the txt file all the P values will be Configuration displayed except for the password fields Table 13 SIP Account Settings Account Name The name associated with each account SIP Server SIP Server s IP address or Domain name provided by VolP service provider Grandstream Networks Inc GXP1100 1105 User Manual Page 24 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice
18. them under the root directory of the TFTP server 2 The 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 httod apache org or use Microsoft IIS web server NOTE When GXP1100 1105 phone boots up it will send TFTP or HTTP request to download configuration file cf9000b82xxxxxx where 000b82xxxxxx is the MAC address of the GXP1100 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 cf9000b82023dd4 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
19. 245 808299 C isten Innovative IP Voice amp Video 86 Voice Mail Announces number of voice mails 99 RESET Enter MAC address to restore factory default setting See Restore Factory Default Setting section Press 9 to reboot the device Others 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 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 be easily 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 automatic
20. ITSP Name SIP service subscriber s name that is used for Caller ID display DNS Mode The default is set to A Record If user wishes to locate the server by DNS SRV the user may select SRV or NATPTR SRV When Use Configured IP option is selected if SIP server is configured as domain name phone 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 Primary IP This option applies only if Use Configured IP is selected the phone will send DNS query to the Primary IP Insert IP address here Backup IP 1 Insert the first back up IP here Backup IP 2 Insert the second back up IP here TEL URL Default is Disabled If the phone has an assigned PSTN telephone number this field should be turned on and a User Phone parameter will be attached to the From header in SIP request SIP Registration This parameter controls sending REGISTER messages to the proxy server The default setting is Yes Unregister on Reboot Default is No If set to Yes the SIP user s registration information will be cleared on reboot Grandstream Networks Inc GXP1100 1105 User Manual Page 25 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Register Expiration This parameter allows user to specify the time frequency in min
21. Layer 3 QoS Support firmware upgrade via TFTP or HTTP Support for Authenticating configuration file before accepting changes User specific URL for configuration file and firmware files Mass provisioning using TR 069 or encrypted XML configuration file Grandstream Networks Inc GXP1100 1105 User Manual Page 6 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovalive IP Voice amp Video Advanced Server Features Message waiting indication support DNS SRV Look up and SIP Server Fail Over 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 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 C istan Innovative IP Voice amp Video Using the GXP1100 1105 GETTING FAMILIAR WITH THE KEYPAD Table 7 GXP1100 1105 Keypad Definitions Key Place active call on hold Call waiting Bring up a new line or Answer the second incoming call 3 way Conference Establish 3 way conference fe Note For 3 way Conference before using the Flash key Enable Flash key as CONF option has to be set to Yes under web GUI
22. P 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 menu Grandstream Networks Inc GXP1100 1105 User Manual Page 9 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovalive IP Voice amp Video 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 lf the target IP address is 192 168 1 60 and the port is 5062 e g 192 168 1 60 5062 input the following 192 168 1 60 5062 The key represents the dot the 4 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 rec
23. Press or SEND key to send 2 REDIAL To redial the last dialed phone number e Take handset off hook e Press the SEND key 3 VIA SPEED DIAL On the GXP1100 1105 the Multiple Purpose Key programmable hard key has 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 To dial the last answered call 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 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 SEND or key to override the 4 second delay 2 If there are digits pressed the SEND key will work as SEND instead of REDIAL 3 By default can be used as SEND to dial the number out Users could disable it by setting User as Dial Key to No from web GUI gt Accounts 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 I
24. TP port ranges from 1024 to 65400 and must be even The default value is 5004 This parameter when set to Yes will force random generation of both the local SIP and 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 When GXP110x is upgrading the MWI on the top right corner will be blinking in red Please do not interrupt the firmware upgrading process Note Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade Please DO NOT interrupt the upgrade process especially the power supply as this will damage the device The password used for encrypting the XML configuration file using OpenSSL This is required for the phone to decrypt the encrypted XML configuration file The user name for the HTTP HTTPS server The pass
25. Table 11 Device Configuration Settings Basic Settings End User Password This contains the password for end user to access the Web Configuration Menu Users can enter new password here This field is case sensitive with a maximum length of 25 characters Confirm Password Enter the end user password again as above to confirm new password GXP1100 1105 User Manual Firmware version 1 0 1 110 Grandstream Networks Inc Page 17 of 34 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video IP Address The GXP110x operates in three modes 1 DHCP mode The GXP110x acquires IP address from the first DHCP server it discovers on the LAN The DHCP option is reserved for NAT router mode In DHCP mode all the field values for the Static IP mode are not used 2 PPPoE mode Set PPPoE account ID PPPoE password and PPPoE service name for the GXP110x to establish PPPoE sesstion 3 Static IP mode Configure IP address Subnet Mask Gateway DNS Server 1 DNS Server 2 and Preferred DNS Server if Static IP mode is used 802 1x Mode This option allows the user to enable disable 802 1x mode on the phone The default value is disabled To enable 802 1x mode this field should be set to EAP MD5 Once enabled the user would be required to enter the following information below to be authenticated on the network 1 Identity 2 MD5 Password Multi P
26. Video e Advanced Settings To set advanced network settings codec settings language settings and etc e Account To configure the SIP account 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 will be used for provisioning and is written on the label in the original box as well as on the label located on the back panel of the device This field shows IP address of GXP1100 1105 This field contains the product model information This field contains the product part number e Program This is the main firmware release number which is always used for identifying the software or firmware system of the phone e Boot Booting code version number e Core Core code version number e Base Base code version number e DSP DSP code version number Aux Aux code version number By default it s unknown for GXP110x This field shows system up time since the last reboot This field shows the current time on the phone system Indicates whether account is registered or not 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 e Phone shows the phone status running or stopped Download core dump file for troubleshooting when necessary
27. ally 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 Oo The default administrator password is admin the default end user password is 123 NOTE When changing any settings always SUBMIT them by pressing UPDATE button on the bottom of the page Reboot the phone to have the changes take effect If after having submitted some changes more settings have to be changed press the menu option needed Definitions This section will describe the options in the web configuration user interface As mentioned a user can log in as an administrator or end user Functions available for the end user are e Status Displays the network status account status software version and MAC address of the phone and service status e Basic Settings Basic preferences such as network settings time settings multi purpose keys and etc can be set up here Additional functions available to administrators are Grandstream Networks Inc GXP1100 1105 User Manual Page 16 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 C isten Innovative IP Voice amp
28. ansfer e Enable Auto Attended Transfer under web GUI gt Advanced Settings Page and Click Update in the web page Established call During the call press TRAN key A new line will be brought up and the first call will be automatically placed on HOLD Enter the number and press SEND key to establish the second call When the second call is established press TRAN key again and the call will be transferred NOTE To transfer calls across SIP domains SIP service providers must support transfer across SIP domains 3 Way Conferencing GXP110x can host 3 way conference call by using Multi Purpose Key or FLASH key To use Multi Purpose Key establish 3 way conference call uses need to login the GXP110x web GUI gt Settings gt Basic Settings configure 3 way Conference to the Multi Purpose Key see Figure below Multi Purpose Keys Key Mode Account CallRtn 3 way Conference Name UserlD Update from the web GUI and then follow the following steps to host the 3 way conference Grandstream Networks Inc GXP1100 1105 User Manual Page 11 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video 1 Initiate a Conference Call Establish two active calls with two parties Press the Multi Purpose Key which is configured as 3 way Conference in the web GUI 3 way conference will
29. blocking the Caller ID from displaying Anonymous Call Default is No If set to Yes anonymous call will be rejected Rejection Refer To Use Target Default is No If set to Yes then for Attended Transfer the Refer To header Contact uses the transferred target s Contact header information Transfer on Conference Defines whether or not the call is transferred to the other party if the initiator of the Hangup conference hangs up Default setting is set to No Grandstream Networks Inc GXP1100 1105 User Manual Page 29 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Check SIP User ID for Check the SIP User ID in Request URI If they don t match the call will be rejected Incoming INVITE Preferred Vocoder 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 SRTP Mode Enable SRTP mode based on selection Default is No Symmetric RTP Selects whether or not symmetric RTP
30. e 18 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Admin Password Confirm Password Layer 3 QoS Layer 2 QoS Local RTP port Use 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 Grandstream Networks Inc C isten Innovative IP Voice amp Video 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 Enter the end user password again as above to confirm new password This field defines the layer 3 QoS parameter It is the value used for IP Precedence or Diff Serv or MPLS Default value is 12 This contains the value used for layer 2 802 1Q VLAN Tag and 802 1p priority value Default setting is 0 The valid value for 802 1Q VLAN Tag is from 0 to 4095 The valid value for 802 1p priority value is from 0 to 7 Note VLAN supported equipment is required when configuring these settings This parameter defines the local RTP port pair used to listen and transmit It is the base RIP 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 R
31. e ID is supported NAT Traversal This parameter activates the NAT traversal mechanism It has options No STUN Keep Alive UPnP Auto VPN If selecting STUN and a STUN server is also specified the phone performs according to the STUN client specification Using this mode the embedded STUN client detects if and what type of NAT Firewall configuration is used If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the phone will use its mapped public IP address and port in all of its SIP and SDP messages lf selecting Keep Alive with no specified STUN server the GXP1100 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 SUBSCRIBE for MWI Default is No When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically Grandstream Networks Inc GXP1100 1105 User Manual Page 26 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video SUBSCRIBE for Default is No When set to Yes a SUBSCRIBE for Registration will be sent Registration periodically Proxy Require SIP Extension to notify SIP server that the unit is behind the NAT Firewall Voice Mail UserID When configured user can access messages by pressing MSG button This userlD is us
32. e 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 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 bli
33. 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 link below to download the corresponding configuration template of the firmware http www grandstream com support tools 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 cf9000b820102ab The configuration file name should be in lower cases For more details on XML provisioning please refer to http Awww grandstream com support Managing Firmware and Configuration File Download When Automatic Upgrade is set to Yes a Service Provider can use P193 Auto Check Interval in minutes default and minimum is 60 minutes to have the devices periodically check for upgrades at pre scheduled time intervals By defining different intervals in P193 for different devices a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time Grandstream Networks Inc GXP1100 1105 User Manual Page 33 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin
34. 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 Note This function is used by ITSP End user should NOT touch these parameters Authenticate Conf File Default is No If set to Yes configuration file would be authenticated before acceptance End user should use default setting Enable TR 069 Default is No ACS URL URL for TR 069 Auto Configuration Servers ACS TR 069 Username Enter ACS username for TR 069 TR 069 Password Enter ACS password for TR 069 Periodic Inform Enable periodic inform Default is No If set to YES device will send inform packets Enable to the ACS Periodic Inform When enabling periodic inform set up the periodic inform interval to send the inform Interval packets to the ACS Connection Request Enter the connection request username This is the user name for the ACS to connect Username to this device Connection Request Enter the connection request password This is the password for the ACS to connect Password to this device Authentication Method Select the authentication method among No authentication Basic or Digest Connection Request Enter the connection request port This is the port for the ACS to connect to this Port device Offhook Auto Dial To configure a User ID extension to dial automatically
35. figuration 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 OD Grandstream Networks Inc GXP1100 1105 User Manual Page 2 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 GE siren nnovative IP Voice 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 different power adaptor with the GXP1100 1105 as it may cause damage to the products and void the
36. gt Advanced Settings Transfer an active call to another number J j i E Enter to retrieve messages Programmable hard key for multiple purposes Speed dial Dial DTMF Call Return 3 way Conference Mute Unmute an active call Send dial out the number SEND Redial redial when there is a previously dialed call Adjust volume by pressing or Jo 6 Standard phone keypad 0 9 Note Press key 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 Grandstream Networks Inc GXP1100 1105 User Manual Page 8 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video 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 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
37. inbound calls with session timer request Force Timer 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 UAC Specify Refresher As a Caller select UAC to use the phone as the refresher or UAS to use the Callee or proxy server as the refresher UAS Specify Refresher As a Callee select UAC to use caller or proxy server as the refresher or UAS to use the phone as the refresher Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method Select Yes to use INVITE method to refresh the session timer Enable 100rel PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is required to support PSTN inter networking Account Ring Tone There are 4 uniquely defined ring tones e One 1 System Ring Tone when selected all calls will ring with system ring tone e Three 3 Customer Ring Tones when selected incoming calls from designated account will play selected ring tone Ring Timeout Defines how long ring will ring when receiving a call Default is 60 seconds Send Anonymous If this parameter is set to Yes the From header in outgoing INVITE message will be set to anonymous essentially
38. is 97 The valid range is between 96 and 127 Jitter Buffer Type Jitter buffer type Fixed or Adaptive Default value is Adaptive Jitter Buffer Length Jitter buffer length Default value is 300ms The valid range is between 100ms and 800ms Conference URI Configure the conference URI when using Broadsoft N way calling feature 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 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 31 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative 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 upgrad
39. is supported Silence Suppression This controls the silence suppression VAD feature of the audio codec G 723 and G 729 If set to Yes when silence is detected a small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled Voice Frames per TX 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
40. nk 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 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 Awww 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 32 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video Instructions for Local TFTP Upgrade 1 Unzip the file and put all of
41. 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 Enable disable Call Log and select type of calls to log Log All Calls Log Incoming Outgoing only Missed calls NOT recorded Disable Call Log GXP1100 1105 User Manual Page 28 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Session Expiration The SIP Session Timer extension enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE Once the session interval expires if there is no refresh via a UPDATE or re INVITE message the session is terminated Session Expiration is the time in seconds at which the session is considered timed out provided no successful session refresh transaction occurs beforehand The default value is 180 seconds Min SE Defines the minimum session expiration in seconds Default is 90 seconds Caller Request Timer If set to Yes the phone will use session timer when it makes outbound calls if remote party supports session timer Callee Request Timer If selecting Yes the phone will use session timer when it receives
42. ommended 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 xxx where x is 0 9 and xxx lt 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 numbers xx or x are also valid The leading 0 is not required but OK For example 192 168 0 2 calling 192 168 0 3 dial 3 followed by or SEND 192 168 0 2 calling 192 168 0 23 dial 23 followed by SEND 192 168 0 2 calling 192 168 0 123 dial 123 followed by SEND 192 168 0 2 dial 43 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 If you have a SIP Server configured a Direct IP IP still works If you are using STUN the Direct IP IP call will also use STUN Configure the Use Random Port to No when completing Direct IP calls ANSWERING PHONE CALLS Receiving Calls 1 Incoming single call Phone rings with selected ring tone 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 b
43. or any 10 digit numbers with leading digits 1617 e Example 2 1900x lt 1617 gt xxxxxxx Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers e Example 3 1xxx 2 9 xxxxxx lt 2 011 gt x Allows any number with leading digit 1 followed by a 3 digit number followed by any number between 2 and 9 followed by any 7 digit number OR Allows any length of numbers with leading digit 2 replacing the 2 with 011 when dialed 3 Default Outgoing x Allow any length of numbers Example of a simple dial plan used in a Home Office in the US 1900x lt 1617 gt 2 9po000 1 2 9 xx 2 9 xxxxxx 011 2 9 x 3469 11 Explanation of example rule reading from left to right e 1900x prevents dialing any number started with 1900 e lt 161 7 gt 2 9 xxxxxx allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically e 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length e 011 2 9 x allows international calls starting with 011 e 3469 11 allow dialing special and emergency numbers 311 411 611 and 911 Note In some cases where the user wishes to dial strings such as 123 to activate voice mail or other applications provided by their service provider the should be predefined inside the dial plan feature An example dial plan will be x which allows the user to dial followed by any length
44. r Ym af mi eae mum eee ee ee ae cme me ume me a ee me ee ee ee SEE ae ee cee eee ume ame geet ee ee eee ee ame ees eae oes ees eee as ees ees os gees me ae ea eas as os ees ees ae ea ees eae ae es ee a ae as aes es es gee a ae ee ee es ae sae Advanced Functionality Customized downloadable ring tones SRIP SIP over TLS multi language support wall mountable AES encryption and etc Table 5 GXP1100 1105 Hardware Specifications GXP1100 1105 LAN Interface 10 100 Mbps Full Half Duplex Ethernet port with auto detection Integrated PoE GXP1105 only Graphic LCD Display NA Expansion Module NA O Lines 2 calls via FLASH key Universal Switching Input 100 240VAC 50 60 Hz Power Adaptor Output 5VDC 800mA 4 0 W UL certified Dimension 201mm W x154mm L x78mm D ss ti itsti OW Weight j Unitweight 0 6KG WW Grandstream Networks Inc GXP1100 1105 User Manual Page 5 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovalive IP Voice amp Video Package weight 1 0KG hv
45. ress 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 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 1 PCMU 2 PCMA 3 iLBC 4 G 726 S CTA 6 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 Page 15 of 34 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1
46. s device gets provisioned for DHCP Option 42 from the to override NTP server server automatically SSL Certificate The user specified SSL certificate used for SIP over TLS SSL Private Key The user specified SSL private key used for SIP over TLS SSL Private Key User specified password to protect the private key above Password Distinctive Ring Tone Select a Distinctive Ring Tone 1 through 3 for the caller ID The Caller ID can be configured as a particular caller ID or Alert Info text When a particular call ID is configured in the Caller ID the selected ring tone will be used when the incoming call is from the Caller ID System Ring Tone will be used for all other calls If server supports Alert Info the Alert Info text can be mapped to the 3 customized ring tones For example if you configure the custom ring tone 1 user ID to priority that ring tone will be used if we receive INVITE with Alert Info header in the following format Alert Info info priority System Ring Tone System ring tone Default is North American standard Users could adjust system ring tone frequencies and cadences based on local telecom standard Grandstream Networks Inc GXP1100 1105 User Manual Page 22 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Call Progress Tones Using these settings users can configure ring
47. sage 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 Pick up the handset Dial 72 A dial tone will be heard Enter the forwarding number Press or SEND key The call will hang up automatically 73 Cancel Unconditional Call Forward Pick up the handset Dial 73 A short tone will be heard Wait for the call to hang up 90 Busy Call Forward Pick up the handset Dial 90 followed by forwarding number Press or SEND key Grandstream Networks Inc GXP1100 1105 User Manual Page 13 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video The call will hang up automatically 91 Cancel Busy Call Forward Pick up the handset Dial 91 A short tone will be heard Wait for the call to hang up 92 Delayed Call
48. ually the VM portal access number Note If server side feature code is used for VM portal access number please also configure Enable Call Features to No under account setting page to use non local feature code Send DTMF This parameter specifies the mechanism to transmit DTMF digit There are 3 supported modes In audio DIMF is combined in audio signal not very reliable with low bit rate codec via RTP RFC2833 via SIP INFO DTMF Payload Type Sends DTMF using RFC2833 The default is 101 Early Dial Default is No Use only if proxy supports 484 responses Dial Plan Prefix Sets the prefix added to each dialed number Grandstream Networks Inc GXP1100 1105 User Manual Page 27 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Dial Plan Delayed Call Forward Wait Time Enable Call Features Call Log Grandstream Networks Inc VolPon www voipon co uk Gs Innovative IP Voice amp Video Dial Plan Rules 1 Accepted Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 a xx at least 2 digit numbers b xx only 2 digit numbers c exclude d 3 5 any digit of 3 4 or 5 e 147 any digit of 1 4 or 7 f lt 2 011 gt replace digit 2 with 011 when dialing g the OR operand e Example 1 369 11 161 7xxxxxxx Allow 311 611 and 911
49. urpose Key X 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 4 3 way Conference Note No UserlD has to be configured in the MPK when using Call Return or 3 way Conference Time Zone 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 Self Defined Time This parameter allows the users to define their own time zone 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 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 Grandstream Networks Inc GXP1100 1105 User Manual Pag
50. utes 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 Reregister Before This parameter allows user to specify the time frequency in seconds that Expiration GXP1100 1105 sends out a re registration request before the Register Expiration By default is 0 second Local SIP Port This parameter defines the local SIP port used to listen and transmit The default value for Account 1 is 5060 SIP Registration Failure Retry registration if the process failed Default is 20 seconds Retry Wait Time SIP T1 Timeout RFC 3261 SIP T1 timer Default is 0 5 second SIP T2 Interval RFC 3261 SIP T2 timer Default is 4 seconds SIP Transport Choose SIP Transport between UDP TCP and TLS TCP Default is UDP SIP URI Scheme when When TLS TCP is used in SIP Transport select sip or sips in this mode using TLS Default is sips Use Actual Ephemeral Enable to use actual ephemeral port in contact with TCP TLS Default is No Port in Contact with TCP TLS Check Domain Enable to check the domain certificate Default is No Certificate Remove OBP from The SIP Extension notifies the SIP server that it is behind a NAT firewall Route Validate Incoming This configuration selects whether or not the incoming messages should be Messages validated Support SIP Instance ID Selects whether or not SIP Instanc
51. when the phone is taken offhook Syslog Server The IP address or URL of System log server This feature is especially useful for ITSPs Grandstream Networks Inc GXP1100 1105 User Manual Page 21 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Syslog Level Select the syslog level for GXP110x to report 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 GXP_1100 device MAC address firmware version error message Send SIP Log When setting the Yes phone will send out SIP Log to syslog server Default setting is No NTP server This parameter defines the URI or IP address of the NIP Network Time Protocol serve It is used to display the current date time Allow DHCP Option 42 Default is Yes This allow
52. word for the HTTP HTTPS server GXP1100 1105 User Manual Page 19 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gs Innovative IP Voice amp Video Upgrade Via This field allows the user to choose the firmware upgrade config server path method TFTP HTTP or HTTPS TFTP GXP110x retrieves the new firmware files or new configuration file from the specified TFTP server path at boot time If there is no new firmware file or configuration file the system will start the boot process using the existing firmware or config file Ifa TFTP server is configured and new firmware files are retrieved the new downloaded image is saved into the Flash memory Please do NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device e HTTP GXP110x retrieves the new firmware files or new configuration file from the specified URL or IP for the HTTP server For example provisioning mycompany com 6688 Grandstream 1 0 0 6 Note 6688 is the specific TCP port where the HTTP server is listening Omit if using default port 80 e HTTPS GXP110x retrieves the new firmware files or new configuration file from the specified URL or IP for the HTTP server via a secured HTTP connection For example provisioning mycompany com Note HTTPS default port is 443 Firmware Server Path Defines the server path for
53. y pressing the FLASH key 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 key 2 Resume Press the HOLD key again to resume 3 Multiple Calls Automatically place ACTIVE call on hold or switch between two calls by pressing the FLASH key Call waiting tone stutter tone will be audible when line is in use Grandstream Networks Inc GXP1100 1105 User Manual Page 10 of 34 Firmware version 1 0 1 110 Last Updated 01 2012 VolPon www voipon co uk sales voipon co uk Tel 44 0 1245 808195 Fax 44 0 1245 808299 Gin nnovative IP Voice amp Video NOTE If users hang up the current call while there is call hold on the other line there will be an audible ring tone indicating a call is on hold Pick up the handset so users can resume with the call on hold Mute Press the MUTE key to mute unmute the microphone Call Transfer GXP1100 1105 supports Blind transfer Attended transfer and Auto Attended transfer 1 Blind Transfer e During the call press TRAN key and dial the number e Press or SEND key to complete transfer of active call 2 Attended Transfer During the call press FLASH key to make another call The ACTIVE LINE will be put on HOLD automatically Establish the second call Press TRAN key Press FLASH key to transfer the call Hang up the phone after the call is transferred 3 Auto Attended Tr

Grandstream GXP1100/1105 IP Phone User Manual

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