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1. 13 COMPLETING CALLS 0 cccccccscccccccccccccceccccssecseccssecenseusnssencetensustnsaeensecsuseensetensuetssstensersuesensnsseaes 13 MAKING CALLS USING IP ADDRESSES 0 s1cccsccccsccecccccccecenseccnccecsesenseesssstensecsnseesaesensssneseens 14 ANSWERING PHONE CALLS 15 RECEIVING CALIES EE 15 DURING A PHONE 15 CALL WAITING CALL 15 MINH M UA e M MELLE 16 CALL souealuueeieeesvese 16 3 WAY CONFERENCING 17 VOICE MESSAGES MESSAGE WAITING sse enne 19 CALL FEATURES 19 CONFIGURATION GUIDE 2 c0ccccccceccsscescceccesccscceccsscesccsccnssesecnseesees 22 CONFIGURATION IVR 22 FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 2 of 44 Innovative IP Voice amp Video CONFIGURATION VIA WEB BROWSER sssesesseeeeeeenr entem en nennen sit nr entente nes nnn internes 23 Blei Mec ERES 24 STATUS PAGE DEFINITIONS 24 ACCOUNT PA
2. Preferred Vocoder Press for the next menu option Press to return to the main menu Enter 01 05 07 10 17 47 86 or 99 for Menu option Enter 9 to toggle the selection If Static IP Mode is selected users need configure all the IP address information through menu 02 to 05 as below If Dynamic IP Mode is selected 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 e PCMU POMA e e G 726 FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 22 of 44 10 13 14 15 16 17 47 86 99 Others MAC Address Firmware Server IP Address Configuration Server IP Address Upgrade Protocol Firmware Version Firmware Upgrade Direct IP Calling Voice Mail RESET Invalid Entry CONFIGURATION VIA WEB BROWSER E sen e G 723 lt lt 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
3. 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 When set to Yes this parameter will force random generation of both the local SIP and RTP ports This is usually necessary when multiple phones are behind the same full cone NAT The default setting is Yes This parameter must be set to No for Direct IP Calling to work Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep the ping hole on the NAT router to open The default setting is 20 seconds The NAT IP address used in SIP SDP messages This field is blank at the default settings It should ONLY be used if it s required by your ITSP The IP address or Domain name of the STUN server STUN resolution results are displayed in the STATUS page of the Web GUI Only non symmetric NAT routers work with STUN Specifies how firmware upgrading and provisioning request to be sent Always Check for New Firmware Check New Firmware only when F W pre suffix changes Always Skip the Firmware Check The password for encrypting the XML configuration file using OpenSSL This is required for the phone to decrypt the encrypted XML configuration file The user name for the HTTP HTTPS server The password for the HTTP HTTPS server Allows users to choose the firmware upgrade method TFTP HTTP or HTTPS Defines the server path for the firmwar
4. Configured IP The default setting is A Record If the user wishes to locate the server by DNS SRV the user may select SRV or NATPTR SRV If Use Configured IP is selected please fill in the three fields below e Primary IP The primary IP address where the phone sends DNS query to e Backup IP 1 e Backup 2 If the phone has an assigned PSTN telephone number this field should be set to User Phone Then a User Phone parameter will be attached to the Request Line and TO header in the SIP request to indicate the E 164 number If set to Enable Tel will be used instead of SIP in the SIP request The default setting is Disable Selects whether or not the phone will send SIP Register messages to the proxy server The default setting is Yes If set to Yes the SIP user s registration information will be cleared when the phone reboots The SIP Contact header will contain to notify the server to unbind the connection The default setting is Specifies the frequency in minutes in which the phone refreshes its registration with the specified registrar The default value is 60 minutes The maximum value is 64800 minutes about 45 days Specifies the time frequency in seconds that the phone sends re registration request before the Register Expiration The default value is 0 Defines the local SIP port used to listen and transmit The default value is 5060 Specifies the interval to retry registratio
5. Firmware upgrade mode Enter 9 to toggle among the following three options e always check e check when pre suffix changes e 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 Enter MAC address to restore factory default setting See Restore Factory Default Setting section Press 9 to reboot the device Automatically returns to Main Menu The GXP1100 GXP1105 embedded Web server responds to HTTP HTTPS GET POST requests Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft s IE Mozilla Firefox and Google Chrome To access the GXP1100 GXP1105 Web GUI 1 Connect the computer to the same network as the phone 2 Make sure the phone is turned on and wait until the indicator on the top right corner turns from RED to OFF 3 Take the handset off hook Enter and then press 02 to hear the IP address FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 23 of 44 E sen Innovative IP Voice amp Video 4 Open a Web browser on your computer 5 Enter the phone s IP address in the address bar of the browser 6 Enter the administrator s login and password to access the Web Configuration Menu Note e The computer has to be connected to the same sub network as the phone This can be easily done by connecting the computer to the same hub or switch as the pho
6. ITSP The URL or IP address and port of the SIP server This will be used when the primary SIP server fails IP address or Domain name of the Primary Outbound Proxy Media Gateway or Session Border Controller It s used by the phone for Firewall or NAT penetration in different network environments If a symmetric NAT is detected STUN will not work and ONLY an Outbound Proxy can provide a solution User account information provided by your VoIP service provider ITSP It s usually in the form of digits similar to phone number or actually a phone number SIP service subscriber s Authenticate ID used for authentication It can be identical to or different from the SIP User ID The account password required for the phone to authenticate with the FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 25 of 44 Name DNS Mode Tel URI 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 08 Innovative IP Voice amp Video ITSP SIP server before the account can be registered After it is saved this will appear as hidden for security purpose The SIP server subscriber s name optional that will be used for Caller ID display This parameter controls how the Search Appliance looks up IP addresses for hostnames There are four modes A Record SRV NATPTR SRV Use
7. and etc supported in web configuration interface Firmware upgrade via TFTP HTTP HTTPS mass provisioning using TR 069 or AES encrypted XML configuration file Universal power adapter Input 100 240VAC 50 60Hz Output 5VDC 800mA Integrated Power over Ethernet 802 3af GXP1105 only Typical power consumption under 1W power adapter or under 1 5W PoE Unit dimension 201mm W x 154mm H x 78mm D Unit weight 0 6kg Package weight 1 0kg 32 104 F 0 40 C 10 90 non condensing GXP1100 GXP1105 phone handset with cord base stand universal power supply network cable quick start guide FCC Part 15 CFR 47 Class B EN55022 Class B EN55024 EN61000 3 2 EN61000 3 3 EN60950 1 AS NZS CISPR 22 Class B AS NZS CISPR 24 RoHS UL 60950 power adapter FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 9 of 44 E sen Innovative IP Voice amp Video INSTALLATION EQUIPMENT PACKAGING Table 2 GXP1100 GXP1105 EQUIPMENT PACKAGING Main Case Handset Phone Cord Power Adaptor Ethernet Cable Phone Stand Quick Start Guide CONNECTING YOUR PHONE Yes 1 1 1 Yes 1 1 1 1 Power Handset Port LAN Port Figure 1 GXP1100 GXP1105 Ports Table 3 GXP1100 GXP1105 CONNECTORS Handset Port 9 handset connector port LAN Port 10 100Mbps RJ 45 port connecting to Ethernet integrated PoE GXP1105 only Power Jack 5V DC Power connector port FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 U
8. 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 FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 37 of 44 Disable Call Waiting Disable Call Waiting Tone Disable Direct IP Calls Use Quick IP Call mode Disable Conference Enable MPK sending DTMF Enable FLASH key as CONF Disable Transfer Auto Attended Transfer In call dial number on pressing transfer key Offhook timeout Do Not Escape as 23 in SIP URI Disable Telnet Display Language Download Device Configuration A Innovative IP Voice amp Video ON is the period of ringing On time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Up to three cadences are supported Disables the call waiting feature The default setting is No Disables the call waiting tone when call waiting is on The default setting is No Disables Direct IP Call The default setting is No When set to Yes users can dial an IP address under the same LAN VPN segment by entering the last octet in the IP address To dial quick IP call offho
9. ette s Pete tat ete ate aa died E nana 10 Figure 2 GXP1100 GXP1105 Multi Purpose Key 3 way Conference 17 FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 3 of 44 CC sen Innovative IP Voice amp Video GUI Interface Examples GXP1100 GXP1105 User Manual http www grandstream com products gxp series general documents gxp110x qgui zi 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 FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 4 of 44 Innovative IP Voice amp Video GNU GPL INFORMATION GXP1100 GXP1105 firmware contains third party software licensed under the GNU General Public License GPL Grandstream uses software under the specific terms of the GPL Please see the GNU General Public License GPL for the exact terms and conditions of the license Grandstream GNU GPL related source code can be downloaded from Grandstream web site from http www grandstream com support faq gnu_qpl FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 5 of 44 E sen Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous ver
10. hold by pressing the HOLD key e Resume Press the HOLD key again to resume FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 15 of 44 E sen e 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 the line is in use Note If users hang up the current call while there is a call on hold in the other line there will be an audible ring tone indicating a call is on hold while your handset is put on hook Pick up the handset so users can resume with the call on hold MUTE During an active call press the MUTE key to mute unmute the microphone CALL TRANSFER GXP1100 GXP1105 supports Blind Transfer Attended Transfer and Auto Attended Transfer e Blind Transfer During the first active call press TRAN key and dial the number to transfer to gt Press SEND key or to complete transfer of active call e Attended Transfer gt During the first active call press FLASH key The first call will be put on hold gt Enter the number for the second call and establish the call gt Press TRAN key gt Press FLASH key to transfer the call e Auto Attended Transfer gt Set Auto Attended Transfer to Yes under Web GUI gt Advanced Settings page And then click Update on the bottom of the page gt Establish one call first gt During the call press TRAN key A new line will be brought up an
11. no key entry If no key is pressed after the timeout the digits will be sent out The default value is 4 seconds Allows users to configure the key as the Send key If set to Yes the key will immediately dial out the input digits In this case this key FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 31 of 44 G723 Rate G 726 32 Packing Mode iLBC Frame Size iLBC Payload Type Jitter Buffer Type Jitter Buffer Length Special Feature Innovative Voice amp Video is essentially equivalent to the Send key If set to No the key is included as part of the dialing string Selects encoding rate for G723 codec The default value is 5 3kbps Select ITU or IETF for G726 32 packing mode Selects iLBC packet frame size The default value is 30ms Specifies iLBC Payload type The default value is 97 The valid range is between 96 and 127 Selects either Fixed or Adaptive based on network conditions The default setting is Adaptive Selects Low Medium or High based on network conditions The default setting is Medium Different soft switch vendors have special requirements Therefore users may need select special features to meet these requirements Users can choose from Standard Nortel MCS Broadsoft CBCOM RNK Sylantro or Huawei IMS depending on the server type The default setting is Standard SETTINGS BASIC SETTINGS PAGE End User Password Confirm Password
12. the Subnet Mask when static IP is used Enter the Default Gateway when static IP is used Enter the DNS Server 1 when static IP is used Enter the DNS Server 2 when static IP is used Enter the Preferred DNS Server Allows users to configure the appropriate network settings on the phone to obtain IPv6 address Users could select Auto configured or Statically configured Enter the static IPv6 address when Full Static is used in Statically configured IPv6 address type Enter the IPv6 prefix length when Full Static is used in Statically configured IPv6 address type Enter the IPv6 Prefix 64 bits when Prefix Static is used in Statically configured IPv6 address type Enter the DNS Server 1 for IPv6 Enter the DNS Server 2 for IPv6 Enter the Preferred DNS Server for IPv6 Allows the user to 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 Enter the Identity for the 802 1x mode Enter the MD5 Password for the 802 1x mode Specifies the HTTP proxy URL for the phone to send packets to The proxy server will act as an intermediary to route the packets to the destination Specifies the HTTPS proxy URL for the phone to send packets to The proxy server will act as an intermediary to route the packets to the destination Assigns a function to the corresponding multi purpose key The key mode options are e Speed Dial Enter the Speed Dial numbe
13. DTMF Payload Type Early Dial Dial Plan Prefix Dial Plan Gee Innovative IP Voice on the phone This ID is usually the VM portal access number For example in Asterisk server 8500 could be used Specifies the mechanism to transmit DTMF digits There are 3 supported modes in audio which means DTMF is combined in the audio signal not very reliable with low bit rate codecs via RTP RFC2833 or via SIP INFO Configures the payload type for DTMF using RFC2833 The default value is 101 Selects whether or not to enable early dial If it s set to Yes the SIP proxy must support 484 response The default setting is No Sets the prefix added to each dialed number A dial plan establishes the expected number and pattern of digits for a telephone number This parameter configures the allowed dial plan for the phone Dial Plan Rules 1 Accepted Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 a b xx at least 2 digit numbers xx only 2 digit numbers a Oo 3 5 any digit of 3 4 or 5 147 any digit of 1 4 or 7 2 011 replace digit 2 with 011 when dialing exclude g the OR operand e Example 1 369 11 1617xxxxxxx Allow 311 611 and 911 or any 10 digit numbers with leading digits 1617 e Example 2 1900x lt 1617 gt xxxxxxx Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit num
14. FEATURES 30 Block Caller ID for all subsequent calls e Off hook the phone e Dial 30 31 Send Caller ID for all subsequent calls e Off hook the phone e Dial 31 67 Block Caller ID per call e Off hook the phone e Dial 67 and then enter the number to dial out 82 Send Caller ID per call FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 19 of 44 70 71 72 73 90 91 92 Innovative IP Voice amp Video e Off hook the phone e Dial 82 and then enter the number to dial out Disable Call Waiting per Call e Off hook the phone e Dial 70 and then enter the number to dial out Enable Call Waiting per Call e Off hook the phone e Dial 71 and then enter the number to dial out Unconditional Call Forward To set up unconditional call forward e Pick up the handset e Dial 72 A dial tone will be heard e Enter the forwarding number e Press or SEND key e The call will hang up automatically with unconditional call forward set up Cancel Unconditional Call Forward To cancel the unconditional call forward e Pick up the handset e Dial 73 A short tone will be heard e Wait for the call to hang up The unconditional call forward is cancelled Busy Call Forward To set up busy call forward e Pick up the handset e Dial 90 followed by forwarding number e Press or SEND key e The call will hang up automatically with busy call forwa
15. GE DEFINITIONS 25 SETTINGS BASIC SETTINGS PAGE aaae rennen 32 SETTINGS ADVANCED SETTINGS PAGE sienne einn ntn erret nnns 34 UPGRADING AND PROVISIONING nnn 40 UPGRADE VIA IVR MENU A ce EL ee M 40 WPGRAGE VIA WEB GU sei er ext E Rr RA DR DR 40 NO 41 CONFIGURATION FILE DOWNLOAD assena nin ansann unnan naa 42 RESTORE FACTORY DEFAULT SETTINGS 43 EXPERIENCING THE GXP1100 GXP1105 44 Table of Tables GXP1100 GXP1105 User Manual Table 1 GXP1100 GXP1105 TECHNICAL SPECIFICATIONSG sse nnne nene 8 Table 2 GXP1100 GXP1105 EQUIPMENT PACKAGING enne nnne trennen 10 Table 3 GXP1100 GXP1105 CONNECTORS sese 10 Table 4 GXP1100 GXP1105 KEYPAD DEFINITIONS sss enne nnne nene 12 Tabl 5 GALL FEATURES rot b een eoe ta He 19 Table 6 GXP1100 GXP1105 IVR 22 Table of Figures GXP1100 GXP1105 User Manual Figure 1 GXP 1100 GXP 1105 POS iecore ta ta
16. Internet Protocol IPv4 Address Type DHCP Host name Option 12 DHCP Vendor Class ID Option 60 Allow DHCP Option 120 to override SIP Server PPPoE Account ID PPPoE Password Allows the administrator to set the password for user level web GUI access This field is case sensitive with a maximum length of 30 characters Confirms the end user password field to be the same as above Selects Prefer IPv4 or Prefer IPv6 Allows users to configure the appropriate network settings on the phone to obtain IPv4 address Users could select DHCP Static IP or PPPoE By default it is set to DHCP Specifies the name of the client This field is optional but may be required by some Internet Service Providers Used by clients and servers to exchange vendor class ID Enables DHCP Option 120 from local server to override the SIP Server on the phone The default setting is No Enter the PPPoE account ID Enter the PPPoE Password FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 32 of 44 PPPoE Service Name IPv4 Address Subnet Mask Gateway DNS Server 1 DNS Server 2 Preferred DNS Server IPv6 Address Type Static IPv6 Address IPv6 Prefix Length IPv6 Prefix DNS Server 1 DNS Server 2 Preferred DNS server 802 1x mode Identity MD5 Password HTTP Proxy HTTPS Proxy Multi Purpose Key X X 1 4 Gorse Enter the PPPoE Service Name Enter the IP address when static IP is used Enter
17. P1100 GXP1105 Web GUI gt Advanced Setting page set Use Quick IP Call Mode to Yes Then take the handset off hook and dial xxx where x is 0 9 and xxx lt 255 Press 4 or SEND and a direct IP call to aaa bbb ccc XXX will be completed aaa bbb ccc is from the local IP address regardless of subnet mask The number xx or x are also valid The leading 0 is not required but it s OK For example e 192 168 0 2 calling 192 168 0 3 dial 3 followed by or SEND e 192 168 0 2 calling 192 168 0 23 dial 23 followed by SEND e 192 168 0 2 calling 192 168 0 123 dial 123 followed by SEND e 192 168 0 2 dial 3 and 03 and 003 results in the same call call 192 168 0 3 Note e The will represent colon in direct IP call rather than SEND key as in normal phone call e f you have a SIP server configured direct IP call still works If you are using STUN direct IP call will also use STUN e Configure the User Random Port to No when completing direct IP calls ANSWERING PHONE CALLS RECEIVING CALLS e Single incoming call Phone rings with selected ring tone Answer call by taking handset off hook e Multiple incoming calls When another call comes in while having an active call the phone will produce a Call Waiting tone stutter tone Answer the incoming call by pressing the FLASH key The current active call will be put on hold DURING A PHONE CALL CALL WAITING CALL HOLD e Hold Place a call on
18. SER MANUAL Page 10 of 44 Innovative IP Voice amp Video To set up the GXP1100 GXP1105 follow the steps below 1 Attach the phone stand to the back of the phone where there is a slot for the phone stand 2 Connect the handset and main phone case with the phone cord 3 Connect the LAN port of the phone to the RJ 45 socket of a hub switch or a router LAN side of the router using the Ethernet cable 4 Connect the 5V DC output plug to the power jack on the phone plug the power adapter into an electrical outlet If PoE switch is used on GXP1105 in step 3 this step could be skipped 5 The LED on the up right corner will light up in red during the booting up provisioning upgrading process Before continuing please wait for the LED turn off 6 Pick up the handset and the dial tone will be heard Press to use the IVR menu and enter menu options to hear the corresponding voice prompt For example dial 02 in the IVR menu will hear the IP address You can further configure the phone using the web GUI by entering GXP1100 GXP1105 s IP address SAFETY COMPLIANCES The GXP1100 GXP1105 phone complies with FCC CE and various safety standards The GXP1100 GXP1105 power adapter is compliant with the UL standard Use the universal power adapter provided with the GXP1100 GXP1105 package only The manufacturers warranty does not cover damages to the phone caused by unsupported power adapters WARRANTY If the GXP1100 GXP1105 phone w
19. as purchased from a reseller please contact the company where the phone was purchased for replacement repair or refund If the phone was purchased directly from Grandstream contact the Grandstream Sales and Service Representative for a RMA Return Materials Authorization number before the product is returned Grandstream reserves the right to remedy warranty policy without prior notification Warning Use the power adapter provided with the phone Do not use a different power adapter as this may damage the phone This type of damage is not covered under warranty FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 11 of 44 stream Innovative IP Voice amp Video USING THE GXP1100 GXP1105 GETTING FAMILAR WITH THE KEYPAD The following table describes the buttons used on the GXP1100 GXP1105 keypad Table 4 GXP1100 GXP1105 KEYPAD DEFINITIONS Hold Place active call on hold or resume the call on hold Flash Flash key can be used for multiple purposes e Call waiting Bring up a new line or answer the second incoming call e 3 way Conference Establish 3 way conference when FLASH key is configured as CONF Before using the Flash key for 3 way conference Enable Flash key as CONF option has to be set to Yes under web GUI gt Advanced Settings Transfer Transfer an active call to another number Message Retrieve voicemail messages Programmable hard key It can be configured for multiple purposes Speed dial D
20. bers e Example 3 1 2 9 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 FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 28 of 44 Delayed Call Forward Wait Time Enable Call Features Call Log Session Expiration Min SE 8 Innovative IP Voice amp Video Example of a simple dial plan used in a Home Office in the US 1900 lt 1617 gt 2 9 xxxxxx 1 2 9 2 9 011 2 9 x 3469 11 Explanation of example rule reading from left to right e 1900x prevents dialing any number started with 1900 e lt 1617 gt 2 9 allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically e 1 2 9 xx 2 9 xxxxxx allows dialing to any US Canada Number with 11 digits length e 011 2 9 allows international calls starting with 011 e 3469 11 allows dialing special and emergency numbers 311 411 611 and 911 Note In some cases where the user wishes to dial strings such as 123 to activate voice mail or other applications provided by their service provider the should be predefined inside the dial plan feature An example dial plan will be x which allows the user to dial followed b
21. could re establish conference call by pressing the Multi Purpose Key again 3 End Conference gt Press HOLD key to split the conference call The conference call will be ended with both calls on hold Or gt Users could simply hang up the call to terminate the conference call FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 18 of 44 innovative IP Voice amp Video Note e party that starts the conference call has to remain in the conference for its entire duration you can put the party on mute but it must remain in the conversation Also this is not applicable when the feature Transfer on call hangup is turned on e The option Disable Conference has to be set to No to establish conference on GXP110x 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 page 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 GXP1105 supports traditional and advanced telephony features including caller ID caller ID with caller Name call forward and etc Table 5 CALL
22. d the first call will be FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 16 of 44 E sen Innovative IP Voice amp Video automatically placed on hold gt Enter the number and press SEND key to establish the second call gt After the second call is established press TRAN key again The call will be transferred Note e Totransfer calls across SIP domains SIP service providers must support transfer across SIP domains e In auto attended transfer use SEND key to dial out the second call instead of using even when could be used as SEND in normal phone calls 3 WAY CONFERENCING GXP1100 GXP1105 can host 3 way conference call by using Multi Purpose Key or FLASH key e use Multi Purpose Key to establish 3 way conference call go to GXP1100 GXP1105 Web GUI gt Settings gt Basic Settings configure the 3 way conference as the Multi Purpose Key mode Click Update on the bottom of the page Then follow the steps below for 3 way conferencing Multi Purpose Keys Key Mode Account 3 Name 3 way Conference UserID Figure 2 GXP1100 GXP1105 Multi Purpose Key 3 way Conference 1 Initiate a conference call Establish two active calls with two parties respectively gt Press the Multi Purpose Key previously configured as 3 way Conference already from Web GUI gt conference will be established FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 17
23. e 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 The transmission gain of the handset The default value is 0 dB SETTINGS ADVANCED SETTINGS PAGE Admin Password Allows users to change the admin password The password field is purposely hidden after clicking the Update button for security purpose FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 34 of 44 Confirm Password Layer 3 QoS Layer 2 QoS 802 1Q VLAN Tag Layer 2 QoS 802 1p Priority Value Local RTP Port Use Random Port Keep alive Interval Use NAT IP STUN Server Firmware Upgrade Provisioning XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Upgrade Via Firmware Server Path and Gosia Innovative This field is case sensitive with a maximum length of 30 characters Confirms the admin password field to be the same as above Defines the Layer 3 QoS parameter This value is used for IP Precedence Diff Serv or MPLS The default value is 12 Assigns the VLAN Tag of the Layer 2 QoS packets The default value is 0 Assigns the priority value of the Layer2 QoS packets The default value is 0 This parameter defines the local RTP port used to listen and transmit It is the base RTP port for channel 0 When configured
24. e server It could be different from the configuration server for provisioning FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 35 of 44 Config Server Path Firmware File Prefix Firmware File Posttix Config File Prefix Config File Postfix Allow DHCP Option 43 and Option 66 Override Server Automatic Upgrade Authenticate Conf File Enable TR 069 ACS URL TR 069 Username TR 069 Password Periodic Inform Enable Periodic Inform Interval Connection Request Username Connection Request Password Connection Request Port CPE SSL Certificate CPE SSL Private Key Offhook Auto Dial Cin Innovative IP Voice Defines the server path for provisioning It could be different from the firmware server for upgrading Enables your ITSP to lock firmware updates If configured only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone Enables your ITSP to lock firmware updates If configured only the firmware with the matching encrypted postfix will be downloaded and flashed into the phone Enables your ITSP to lock configuration updates If configured only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone Enables your ITSP to lock configuration updates If configured only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone If DHCP option 66 is enabled o
25. ed as Speed Dial on Multi Purpose Key gt Goto GXP1100 GXP1105 Web GUI gt Basic Settings configure the Multi Purpose Key s Key Mode as Speed Dial Enter the Name and User ID the number to be dialed out for the Multi Purpose Key Click on Update at the bottom of the Web GUI page gt handset off hook You shall hear dial tone from the handset gt Press the configured Speed Dial key e Call Return Dial the last answered call gt Goto GXP1100 GXP1105 Web GUI gt Basic Settings configure the Multi Purpose Key s Key Mode as Call Return No Name or User ID has to be set on the Multi Purpose Key for Call Return gt Take handset off hook You shall hear dial tone from the handset gt Press the configured Call Return key to dial out FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 13 of 44 E sen Innovative IP Voice amp Video Note e After entering the number the phone waits for the No Key Entry Timeout Default timeout is 4 seconds configurable via Web GUI before dialing out Press SEND or key to override the No Key Entry Timeout e If digits have been entered after handset is off hook the SEND key will works as SEND instead of REDIAL e 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 Account page MAKING CALLS USING IP ADDRESSES Direct IP Call allows two phones to talk to each other in an ad hoc fashion w
26. ed by a file named cfgxxxxxxxxxxxx where is the MAC address of the device i e cfg000b820102ab The configuration file name should be in lower case letters For more details on XML provisioning please refer to http www grandstream com general gs provisioning guide public pdf FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 42 of 44 E sen Innovative IP Voice amp Video RESTORE FACTORY DEFAULT SETTINGS Warning Restoring the Factory Default Settings will delete all configuration information on the phone Please backup or print all the settings before you restore to the factory default settings Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider Please follow the instructions below to reset the phone Pick up the handset press to access the IVR menu Enter 99 for factory reset Then enter the MAC address printed on the bottom of the sticker Please use the following mapping 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 gt Example if the MAC address is 000582006395 it should be key in as 0002228200333395 NOTE e If there are digits like 22 in the MAC you need to wait for 4 seconds to continue to key in another 2 e Once the MAC address is correctly input the phone wil
27. emote party supports session timers the phone will use a session timer when it receives inbound calls If Force Timer is set to Yes the phone will use the session timer even if the remote party does not support this feature If Force Timer is set to No the phone will enable the session timer only when the remote party supports this feature To turn off the session timer select No As a Caller select UAC to use the phone as the refresher or select UAS to use the Callee or proxy server as the refresher As a Callee select UAC to use caller or proxy server as the refresher or select UAS to use the phone as the refresher The Session Timer can be refreshed using the INVITE method or the UPDATE method Select Yes to use the INVITE method to refresh the session timer The use of the PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is very important in order to support PSTN internetworking To invoke a reliable provisional response the 100rel tag is appended to the value of the required header of the initial signaling messages Allows users to configure the ringtone for the account Users can choose from different ringtones from the dropdown menu Specifies matching rules and selects the distinctive ringtone for the rule When the incoming caller ID matches the rule the phone will ring with the corresponding ringtone Matching rules e Specific caller ID number For e
28. 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 as it may cause damage to the products and void the manufacturer warranty This document is subject to change without notice The latest electronic version of this user manual is available for download here http www grandstream com support Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 7 of 44 E sen innovative IP Voice amp Video PRODUCT OVERVIEW FEATURE HIGHTLIGHTS e Single SIP Account up to 2 calls 4 programmable keys e HD handset with support for wideband audio e Single 10 100Mbps network port integrated PoE GXP1105 only e 7 dedicated function keys for Hold Flash Call Waiting Transfer Message Mute Volume Send Redial e Automated provisioning using TR 069 or AES encrypted XML configuration file SRTP and TLS for advanced security and privacy protection LLDP IPv6 GXP1100 GXP1105 TECHNICAL SPECIFICATIONS Protocols and Standards Network Interfaces Graphic Display Feature Keys Voice Codec Telephony Features HD Audio Headset Jack Base Stand Wal
29. ge 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 upgrading is done you will see the phone restarts again Please do not interrupt or power cycle the phone when the upgrading process is on UPGRAGE VIA WEB GUI Open a web browser on PC and enter the IP address for the GXP1100 GXP1105 Then login with the administrator username and password Go to Settings gt Advanced Settings page enter the IP address or the FQDN for the upgrade server in Firmware Server Path field and choose to upgrade via TFTP or HTTP HTTPS Update the change by clicking the Update button Then Reboot or power cycle the phone to update the new firmware FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 40 of 44 E sen 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 reboot using the existing firmware software Firmware upgrades take around 60 seconds in a controlled LAN or 5 10 minutes o
30. ial DTMF VMsg Call Return 3 way Conference Transfer Intercom Mute Press to mute unmute an active call Send It can be used as Send or Redial e Send Enter the digits and then press Send to dial out the number e Redial Redial when there is a previously dialed call Volume Press or to adjust the volume Standard phone keypad FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 12 of 44 E sen Innovative IP Voice amp Video MAKING PHONE CALLS 2 CALLS WITH 1 SIP ACCOUNT GXP1100 GXP1105 can support up to two lines virtually mapped to one SIP account By picking up the handset the GXP1100 GXP1105 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 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 GXP1105 allows you to make phone calls after picking up the handset There are four ways to complete calls e Dial Enter the number and send out gt handset off hook You shall hear dial tone from the handset gt Enter the number gt Press SEND key or to dial out e Redial Redial the last dialed number Take handset off hook You shall hear dial tone from the handset Press SEND key e Speed Dial Dial the number configur
31. it will send a TFTP or HTTP request to download the configuration file where XXXXXXXXXXxXx is the MAC address of the phone If it is a normal TFTP or HTTP upgrade the following messages TFTP Error from IP ADRESS requesting cfg000b82023dd4 File does not exist Configuration File Download can be ignored in the TFTP HTTP server log FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 41 of 44 E sen Innovative IP Voice amp Video CONFIGURATION FILE DOWNLOAD Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File binary or XML through TFTP or HTTP HTTPS The Config Server Path is the TFTP or HTTP HTTPS 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 3 Could be extended to 4 in the future digit numeric numbers i e P2 is associated with the Admin Password in the Web GUI gt Settings gt Advanced Settings For a detailed parameter list please refer to the corresponding firmware release configuration template When a Grandstream Devices boots up or reboots it will issue a request for a configuration XML file named cfgxxxxxxxxxxxx xml follow
32. ithout 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 the 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 the steps below e Take handset off hook You shall hear dial tone from the handset e Press to enter the GXP1100 GXP1105 IVR 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 to dial Please see example below For example If the target IP address is 192 168 1 60 and the port is 5062 i e 192 168 1 60 5062 input the following 192 168 1 60 5062 The key represents the dot the key represents colon Wait for about 4 seconds and the phone will initiate the call Quick IP Call Mode The GXP1100 GXP1105 also supports Quick IP Call mode This enables the phone to make direct IP calls using only the last few digits last octet of the target phone s IP address This is possible only if both phones are under the same LAN VPN This simulates a PBX function using the CMSA CD without a SIP server Controlled static IP usage is recommended FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 14 of 44 E sen Innovative IP Voice amp Video To enable Quick IP Call Mode go to GX
33. l Mountable QoS Security Table 1 GXP1100 GXP1105 TECHNICAL SPECIFICATIONS SIP RFC3261 TCP IP UDP RTP HTTP HTTPS ARP ICMP DNS A record SRV NAPTR DHCP PPPoE TELNET NTP STUN TR 069 802 1x LLDP IPv6 TLS SRTP Single 10 100Mbps port integrated PoE GXP1105 only N A 4 programmable keys 7 dedicated function keys for HOLD FLASH TRANSFER MUTE VOLUME SEND REDIAL and MESSAGE with LED indicator Support for G 723 1 G 729A B G 711u a G 726 32 G 722 wide band iLBC in band and out of band DTMF in audio RFC2833 SIP INFO Hold transfer forward 3 way conference call waiting off hook auto dial click to dial flexible dial plan personalized music ringtones server redundancy and fail over Yes HD handset with support for wideband audio N A Yes 1 angle position available Yes Layer 2 802 1Q 802 1p and Layer 3 ToS DiffServ MPLS QoS User and administrator level passwords MD5 and MD5 sess based authentication 256 bit AES encrypted configuration file TLS SRTP 802 1x media access control FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 8 of 44 Multi language Upgrade and Provisioning Power and Green Energy Efficiency Physical Operating Temperature and Humidity Package Content Compliance Cin Innovative IP Voice English German ltalian French Spanish Portuguese Russian Croatian Simplified Chinese traditional Chinese Korean Japanese
34. l reboot Otherwise it will announce Invalid Entry and exit to the main menu FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 43 of 44 Innovative IP Voice amp Video EXPERIENCING THE GXP1100 GXP1105 Please visit our website http Awww grandstream com to receive the most up to date updates on firmware releases additional features FAQs documentation and news on new products We encourage you to browse our product related documentation FAQs and User and Developer Forum for answers to your general questions If you have purchased our products through a Grandstream Certified Partner or Reseller please contact them directly for immediate support Our technical support staff is trained and ready to answer all of your questions Contact a technical support member or submit a trouble ticket online to receive in depth support Thank you again for purchasing Grandstream IP phone it will be sure to bring convenience and color to both your business and personal life FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 44 of 44
35. n if the process is failed The default value is 20 seconds SIP T1 Timeout The default setting is 0 5 seconds SIP T2 Interval The default setting is 4 seconds Determines the network protocol used for the SIP transport Users can FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 26 of 44 SIP URI Scheme when using TLS Use Actual Ephemeral Port in Contact with TCP TLS Check Domain Certificates Remove OBP from route Validate Incoming Messages Support SIP Instance ID NAT Traversal SUBSCRIBE for MWI SUBSCRIBE for Registration Feature Key Synchronization Proxy Require Voice Mail UserlD 08 Innovative IP Voice amp Video choose from TCP UDP and TLS Specifies if sip or sips will be used when TLS TCP is selected for SIP Transport The default setting is sips Defines whether the actual ephemeral port in contact with TCP TLS will be used or not This is used when TLS TCP is selected for SIP Transfer The default setting is No Defines whether the domain certificates will be checked or not when TLS TCP is used for SIP Transport The default setting is No Configures to remove outbound proxy from route This is used for the SIP Extension to notify the SIP server that the device is behind a NAT Firewall Defines whether the incoming messages will be validated or not The default setting is No Defines whether SIP Instance ID is supported or not The default setting is Ye
36. n the LAN side the TFTP server can be redirected The default setting is Yes Enables automatic upgrade and provisioning The default setting is No Authenticates configuration file before acceptance The default setting is No Enables TR 069 The default setting is No URL for TR 069 Auto Configuration Servers ACS ACS username for TR 069 ACS password for TR 069 Enables periodic inform If set to Yes device will send inform packets to the ACS The default setting is No Sets up the periodic inform interval to send the inform packets to the ACS The user name for the ACS to connect to the phone The password for the ACS to connect to the phone The port for the ACS to connect to the phone The Certificate File for the phone to connect to the ACS via SSL The Private Key for the phone to connect to the ACS via SSL Configures a User ID extension to dial automatically when the phone is off hook The phone will use the first account to dial out The default setting is No FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 36 of 44 Auto Recover From Abnormal Syslog Server Syslog Level Send SIP Log NTP Server Allow DHCP Option 42 Override NTP Server SSL Certificate SSL Private Key SSL Private Key Password System Ring Tone Call Progresses Tones Dial Tone Message Waiting Ring Back Tone Call Waiting Tone Busy Tone Reorder Tone Gee Innovative IP Voice Configures whe
37. ndstream Innovative IP Voice amp Video Grandstream Networks Inc GXP1100 GXP1105 Small Business IP Phone GXP1100 GXP1105 USER MANUAL E sen Innovative IP Voice amp Video GXP1100 GXP1105 User Manual Index GNU GPL 5 CHANGE LOG 6 FIRMWARE VERSION 1 0 4 9 6 WELECME 7 PRODUCT OVERVIEW 8 FEATURE HIGHT LIGHTS 0 cccccccccssccccccsaccececnoseccesceccuccnsaccesccceneccusseceedctesccadeebeneccuseececccnuuscedsuneneccevnece 8 GXP1100 GXP1105 TECHNICAL SPECIFICATIONS 8 INSTALLAT ION MeMOTMH O t 10 EQUIPMENT PACKAGING 10 CONNECTING YOUR PHONE 10 ky idzupdee unes 11 M 11 USING THE 1100 1105 12 GETTING FAMILAR WITH THE KEYPAD 0 ccccccccecccccecceccceccececccceccccucecaecceceecececaesecueeaueceneceeaass 12 MAKING PHONE 13 2 CALLS WITH 1
38. ne connected to In absence of a hub switch or free ports on the hub switch please connect the computer directly to the PC port on the back of the phone e phone is properly connected to a working Internet connection the IP address of the phone can be obtained from IVR Menu option 02 This address has the format xxx xxx xxx xxx where xxx stands for a number from 0 255 Users will need this number to access the Web GUI For example if the phone has IP address 192 168 40 154 please enter http 192 168 40 154 in the address bar of the browser e The default login name for the administrator is admin The default administrator password is set to admin The default login name for the end user is user while the default user password is set to 123 e When changing any settings always SUBMIT them by pressing the UPDATE button on the bottom of the page After submitting the changes in all the Web GUI pages reboot the phone to have the changes take effect DEFINITIONS This section describes the options in the GXP1100 GXP1105 Web GUI As mentioned you can log in as an administrator or an end user e Status Displays the Account status Network status and System Info of the phone e Account To configure the SIP account e Basic Settings To configure basic network settings time settings multi purpose keys and etc e Advanced Settings To configure advanced network settings upgrading and provisioning language setting
39. of 44 E sen Innovative IP Voice amp Video 2 Split call in conference gt During the 3 way conference press HOLD key The conference call will be split and both calls will be put on hold separately gt Press HOLD key again and it will resume the 2 way conversation with the line when establishing the conference call gt Press FLASH key to toggle between the 2 lines gt Users could re establish conference call by pressing the Multi Purpose Key again 3 End Conference gt Press HOLD key to split the conference call The conference call will be ended with both calls on hold Or gt Users could simply hang up the call to terminate the conference call e To use Flash key to establish 3 way conference call go to GXP1100 GXP1105 Web GUI gt Settings gt Advanced Settings set Enable FLASH key as CONF to Yes Click on Update on the bottom of the Web GUI page and then reboot the phone Follow the steps below to host the 3 way conference 1 Initiate a conference call gt Initiate and establish two active calls with two parties from GXP1100 GXP1105 gt Press the FLASH Key gt 3 way conference will be established 2 Split call in conference gt During the 3 way conference press HOLD key The conference call will be split and both calls will be put on hold separately Press HOLD key again and it will resume the 2 way conversation with the line when establishing the conference call gt Users
40. ok the phone and dial XXX X is 0 9 and XXX lt 255 phone will make direct IP call to aaa bbb ccc XXX where aaa bbb ccc comes from the local IP address REGARDLESS of subnet mask XX or X are also valid so leading 0 is not required but OK No SIP server is required to make quick IP call The default setting is No Disables the Conference function The default setting is No Enables Multi Purpose Key to send DTMF during the call The default setting is No If set to Yes FLASH key can be used to establish 3 way conference The default setting is No Disables the Transfer function The default setting is No If set to Yes the phone will use attended transfer by default The default setting is No If configured the phone will use the TRAN key to dial the number as DTMF during the call If configured when the phone is onhook it will go offhook after the timeout in seconds The default value is 30 seconds Specifies whether to replace by 9523 or not for some special situations The default setting is No Disables Telnet access The default setting is No Selects display language on the phone Click to download the device config file in txt format FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 38 of 44 E sen Innovative IP Voice amp Video SAVING THE CONFIGURATION CHANGES After users makes changes to the configuration press the Update button on the bottom of the Web GUI page We
41. r in UserID field to be dialed e Dial DTMF Enter a series of DTMF digits in UserID field to be dialed during the FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 33 of 44 Time Zone Self Defined Time Zone Handset TX gain Innovative IP Voice amp Video call Enable MPK Sending DTMF under Advanced Setting has to be set to Yes first e VMsg Enter the Voice Mail access number in the UserID field e Call Return The last answered calls can be dialed out by using Call Return The Name and UserlD should be left blank e Transfer Enter the number in the UserlD field to be transferred blind transfer during the call e Intercom Enter the extension number in the UserlD field to do the intercom e 3 way Conference Press to establish 3 way conference The Name and UserlD should be left blank Configures the date time on the phone according to the specified time zone This parameter allows the users to define their own time zone The syntax is std offset dst offset start time end time Default is set to MTZ 6MDT 5 M4 1 0 M11 1 0 MTZ 6MDT 5 This indicates a time zone with 6 hours offset with 1 hour ahead which is U S central time If it is positive if the local time zone is west of the Prime Meridian A K A International or Greenwich Meridian and negative if it is east M4 1 0 M11 1 0 The 1st number indicates Month 1 2 3 12 for Jan Feb Dec The 2nd number indicates th
42. rd set up Cancel Busy Call Forward To cancel the busy call forward e Pick up the handset e Dial 91 A short tone will be heard e Wait for the call to hang up The busy call forward is cancelled Delayed Call Forward To set up delayed call forward e Pick up the handset Dial 92 followed by forwarding number e Press or SEND key FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 20 of 44 Innovative IP Voice amp Video e The call will hang up automatically with delayed call forward set up 93 Cancel Delayed Call Forward To cancel the delayed call forward e Pick up the handset e Dial 93 A short tone will be heard e Wait for the call to hang up The delayed call forward is cancelled FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 21 of 44 andstream Innovative IP Voice amp Video CONFIGURATION GUIDE The GXP1100 GXP1105 can be configured via two ways e IVR Menu using the phone s keypad e Web GUI embedded on the phone using PC s web browser CONFIGURATION VIA IVR MENU GXP1100 GXP1105 has a built in voice prompt menu for simple device configuration Pick up the handset and dial to use the IVR menu Main Menu 01 02 03 04 05 07 Table 6 GXP1100 GXP1105 IVR MENU 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
43. recommend rebooting or powering cycle the IP phone after saving changes REBOOTING FROM REMOTE LOCATIONS Press the Reboot button on the bottom of the web GUI page to reboot the phone remotely The web browser will then display a reboot page with message The device is rebooting now Wait for about 1 minute to log in again FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 39 of 44 E sen Innovative IP Voice amp Video UPGRADING AND PROVISIONING The GXP1100 GXP1105 can be upgraded via TFTP HTTP HTTPS by configuring the URL IP Address for the TFTP HTTP HTTPS server and selecting a download method Configure a valid URL for TFTP or HTTP the server name can be FQDN or IP address Examples of valid URLs firmware grandstream com fw ipvideotalk com gs There are two ways to setup a software upgrade server The IVR Menu or the Web Configuration Interface UPGRADE VIA IVR MENU Follow the steps below to configure the Upgrade Server IP address via IVR e Pick up the handset press to access the IVR Menu e Input menu option 15 for Upgrading Protocol Then press 9 to toggle between different upgrading methods e Press to return to the main menu and input menu option 13 for 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 input 010000050191 Then reboot the phone The LED indicator on the top right corner will turn oran
44. rred to the other party if the initiator of the conference hangs up The default setting is No If set to Yes SIP User ID will be checked in the Request URI of the incoming INVITE If it doesn t match the phone s SIP User ID the call will be rejected The default setting is 7 different vocoder types are supported on the phone including G 711 U law PCMU G 711 A law PCMA G 723 1 G 729A B G 722 wide band iLBC and G72 32 Users can configure vocoders in a preference list that is included with the same preference order in SDP message Enables the SRTP mode based on your selection The default setting is Disabled Defines whether symmetric RTP is supported or not The default setting is No Controls the silence suppression VAD feature of the audio codec G 723 and G 729 If set to Yes when silence is detected a small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled The default setting is No Configures the number of voice frames transmitted per packet When configuring this it should be noted that the ptime value for the SDP will change with different configurations here This value is related to the codec used and the actual frames transmitted during the in payload call For end users it is recommended to use the default setting as incorrect settings may influence the audio quality Defines the timeout in seconds for
45. s This parameter configures whether the NAT traversal mechanism is activated Users could select the mechanism from No STUN Keep Alive UPnP Auto or VPN If set to STUN and STUN server is configured the phone will route according to the STUN server If NAT type is Full Cone Restricted Cone or Port Restricted Cone the phone will try to use public IP addresses and port number in all the SIP amp SDP messages The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be Keep Alive Configure this to be No if an outbound proxy is used STUN cannot be used if the detected NAT is symmetric NAT When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically The phone supports synchronized and non synchronized MWI The default setting is No When set to Yes a SUBSCRIBE for Registration will be sent out periodically The default setting is No This feature is used for Broadsoft call feature synchronization When it s enabled DND and Call Forward features can be synchronized with Broadsoft server The default setting is Disabled A SIP Extension to notify the SIP server that the phone is behind a NAT Firewall Do not configure this parameter unless this feature is supported on the SIP server Allows you to access voice messages by pressing the MESSAGE button FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 27 of 44 Send DTMF
46. s call features and etc STATUS PAGE DEFINITIONS MAC Address Global unique ID of device in HEX format The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 24 of 44 IPv4 Address IPv6 Address Product Model Part Number Software Version System Up Time System Time Registered PPPoE Link Up Service Status Core Dump ACCOUNT PAGE DEFINITIONS Account Name SIP Server Secondary SIP Server Outbound Proxy SIP User ID Authenticate ID Authenticate Password C itean The IPv4 address obtained on the phone The IPv6 address obtained on the phone Product model of the phone Product part number e boot boot version number e core version number e base base version number e prog program version number This is the main firmware release number which is always used for identifying the software system of the phone e dsp DSP version number System up time since the last reboot Current system time on the phone system SIP account registration status PPPoE connection status GUI and Phone service status running or stopped Core dump file that could be downloaded for troubleshooting purpose The name associated with the SIP account The URL or IP address and port of the SIP server This is provided by your VoIP service provider
47. sions of GXP1100 GXP1105 user manuals Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here FIRMWARE VERSION 1 0 4 9 e Added instructions for connecting the phone CONNECTING YOUR PHONE e Added Multi Purpose Key options VMsg Transfer Intercom SETTINGS BASIC SETTINGS PAGE e Added IPv6 configuration options SETTINGS BASIC SETTINGS PAGE e Added Matching Incoming Caller ID function in Account Setting ACCOUNT PAGE DEFINITIONS e Added GNU GPL information GNU GPL INFORMATION e Added Change Log for this user manual CHANGE LOG FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 6 of 44 E sen Innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream GXP1100 GXP1105 Small Business IP Phone GXP1100 GXP1105 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 GXP1105 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
48. ther auto recover or not when the phone is running abnormal The default setting is Yes The URL IP address for the syslog server Selects the level of logging for syslog The default setting is None There are 4 levels DEBUG INFO WARNING AND ERROR Syslog messages are sent based on the following events e product model version on boot up INFO level e NAT related info INFO level e sentor received SIP message DEBUG level e SIP message summary INFO level e inbound and outbound calls INFO level e registration status change INFO level e negotiated codec INFO level e ethernet link up INFO level e chip exception WARNING and ERROR levels e memory exception ERROR level Configures whether the SIP log will be included in the Syslog messages or not The default setting is No Defines the URL or IP address of the NTP server The phone may obtain the date and time from the server Defines whether DHCP Option 42 should override NTP server or not When enabled DHCP Option 42 will override the NTP server if it s set up on the LAN The default setting is Yes SSL Certificate used for SIP Transport in TLS TCP SSL Private key used for SIP Transport in TLS TCP SSL Private key password used for SIP Transport in TLS TCP System ring tone Default is North American standard Users could adjust system ring tone frequencies and cadences based on local telecom standard Using these settings users can
49. ver the Internet We recommend completing firmware upgrades in a controlled LAN environment whenever possible NO LOCAL TFTP HTTP SERVERS For users that would like to use remote upgrading without a local TFTP HTTP server Grandstream offers a NAT friendly HTTP server This enables users to download the latest software upgrades for their phone via this server Please refer to the webpage http www grandstream com support firmware Alternatively users can download a free TFTP or HTTP server and conduct a local firmware upgrade A free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm http tftpd32 jounin net Instructions for local firmware upgrade via TFTP 1 Unzip the firmware files and put all of them in the root directory of the TFTP server 2 Connect the PC running the TFTP server and the phone to the same LAN segment 3 Launch the TFTP server and go to the File menu gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade 4 Start the TFTP server and configure the TFTP server in the phone s web configuration interface Configure the Firmware Server Path to the IP address of the PC 6 Update the changes and reboot the phone End users can also choose to download a free HTTP server from http httod apache org or use Microsoft IIS web server Note When the phone boots up
50. xample 8321123 e defined pattern with certain length using x and to specify where x could be any digit from 0 to 9 Samples at least 2 digit number XX only 2 digit number 345 xx 3 digit number with the leading digit of 3 4 or 5 6 9 xx 3 digit number with the leading digit from 6 to 9 Defines the timeout in seconds for the rings on no answer The default setting is 60 seconds If set to Yes the From header in outgoing INVITE messages will be set to anonymous essentially blocking the Caller ID to be displayed FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 30 of 44 Anonymous Call Rejection Allow Auto Answer by Call Info Refer To Use Target Contact Transfer on Conference Hangup Check SIP User ID for incoming INVITE Preferred Vocoder SRTP Mode Symmetric RTP Silence Suppression Voice Frames Per TX No Key Entry Timeout s Use as Dial Key Innovative IP Voice amp Video If set to Yes anonymous calls will be rejected The default setting is No If set to Yes the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep based on the SIP info header sent from the server proxy The default setting is If set to Yes the Refer To header uses the transferred target s Contact header information for attended transfer The default setting is No Defines whether or not the call is transfe
51. y any length of numbers Defines the timeout in seconds before the call is forwarded on no answer The default value is 20 seconds When enabled call forward and other call features will be supported locally provided ITSP support those features The default setting is Yes Configures Call Log setting on the phone You can log all calls only log incoming outgoing calls or disable call log The default setting is Log All Calls The SIP Session Timer extension that enables SIP sessions to be periodically refreshed via a SIP request UPDATE or re INVITE If there is no refresh via an UPDATE or re INVITE message the session will be terminated once the session interval expires Session Expiration is the time in seconds where the session is considered timed out provided no successful session refresh transaction occurs beforehand The default value is 180 seconds The minimum session expiration in seconds The default value is 90 seconds FIRMWARE VERSION 1 0 4 9 GXP1100 GXP1105 USER MANUAL Page 29 of 44 Caller Request Timer Callee Request Timer Force Timer UAC Specify Refresher UAS Specify Refresher Force INVITE Enable 100rel Account Ring Tone Matching Incoming Caller ID Ring Timeout Send Anonymous Innovative IP Voice 4 Video If set to Yes and the remote party supports session timers the phone will use a session timer when it makes outbound calls If set to Yes and the r

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