Grandstream Networks GXP1100 1lines Wired handset Black
Contents
1. 9 Table 2 GXP1100 GXP1105 EQUIPMENT 11 Table 3 GXP1100 GXP1105 11 Table 4 GXP1100 GXP1105 KEYPAD 14 Table EON EMIZugliccc Y 21 Table 6 GXP1100 GXP1105 MENU 23 Table of Figures GXP1100 GXP1105 User Manual Figure 1 GXP1100 GXP1105 11 Figure 2 GXP1100 GXP1105 12 Figure 3 GXP1100 GXP1105 Multi Purpose Key way Conference 19 Figure 4 Click to lal oreet RP ee XR aan AR d eR aR 42 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 4 of 47 sen Innovative IP Voice amp Video GUI INTERFACE EXAMPLES http www grandstream com products gxp series general documents gxp21xx qui 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 Screens2. 0 The account index for the phone to make call The index is 0 for account 1 1 for account 2 2 for account 3 and etc passwordzadmin The admin login password of phone s Web GUI SAVING THE CONFIGURATION CHANGES After users makes changes to the configuration press the Save button will save but not apply the changes until the Apply button on the top of web GUI page is clicked Or users could directly press Save and Apply button We recommend rebooting or powering cycle the phone after applying all the changes REBOOTING FROM REMOTE LOCATIONS Press the Reboot button on the top right corner of the web GUI page to reboot the phone remotely The web browser will then display a reboot message Wait for about 1 minute to log in again FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 42 of 47 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
3. SIP Settings gt Security Settings UAS Specify Refresher Force INVITE Check Domain Defines whether the domain certificates will be checked or not when TLS TCP Certificates is used for SIP Transport The default setting is No Validate Incoming Defines whether the incoming messages will be validated or not The default Messages 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 No Check SIP User ID for incoming INVITE When set to Yes the SIP address of the Request URL in the incoming SIP message will be checked If it doesn t match the SIP server address of the account the call will be rejected The default setting is No Accept Incoming SIP from Proxy Only FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 30 of 47 Authenticate Incoming INVITE E sen innovative IP Voice amp Video If set to Yes the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response The default setting is No Account x gt Audio Settings Send DTMF DTMF Payload Type Preferred Vocoder Use First Matching Vocoder in 2000K SDP SRTP Mode Symmetric RTP Silence Suppression Voice Frames Per TX G723 Rate G 726 32 Packing Mode iLBC Frame Size iLBC Payload Type Jitter Buffer Type Jitter Buffer Length Spec
4. users it is recommended to use the default setting as incorrect settings may influence the audio quality Selects encoding rate for G723 codec The default value is 5 3kbps Selects ITU or IETF for G726 32 packing mode Selects iLBC packet frame size The default value is 30ms Specifies 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 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 31 of 47 am innovative IP Voice amp Video Account x gt Call Settings Early Dial Dial Plan Prefix Dial Plan Selects whether or not to enable early dial If it s set to Yes the SIP proxy must support 484 response The default setting is Sets the prefix added to each dialed number A dial plan establishes the expected number and pattern of digits for a telephone number This parameter configures the allowed dial plan for the phone Dial Plan Rules 1 Accepted Digits 1 2 3 4 5 6 7 8 9 0 A a B b C c D d 2 Grammar x any digit from 0 9 a at least 2 digit numbers b xx only 2 digit numbers exclude 8 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 g the OR operand gt 0 Qa Oo e
5. Example 1 869 11 1617xxxxxxx Allow 311 611 and 911 or any 10 digit numbers with leading digits 1617 e Example 2 1900 lt 1617 gt 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 Example of a simple dial plan used in a Home Office in the US 1900x lt 1617 gt 2 9 xxxxxx 1 2 9 2 9 011 2 9 x 3469111 Explanation of example rule reading from left to right e 1900 prevents dialing any number started with 1900 1617 2 9 xxxxxx allows dialing to local area code 617 numbers by dialing 7 numbers and 1617 area code will be added automatically FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 32 of 47 Delayed Forward Wait Time Enable Call Features Call Log Account Ring Tone Matching Incoming Caller ID Gosia Innovative I 1 2 9 2 9 allows dialing to any US Canada Number with 11 digits length 011 2 9 allows international calls starting with 011 1346911 allows dialing special and emergency numbers 311 411 611 and 911 Note In some cases where the user wishes to
6. certificate to the phone or delete existed 802 1X Client certificate from the phone 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 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 Configures the PC port mode When set to Mirrored the traffic in the LAN port will go through PC port as well and packets can be captured by connecting a PC to the PC port The default setting is Enable MAINTENANCE PAGE DEFINITIONS Maintenance gt Web Telnet Access Disable Telnet End User Password Confirm Password Admin Password Confirm Password Disables Telnet access The default setting is No 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 Allows users to change the admin password The password field is purposely hidden for security purpose This field is case sensitive with a maximum length of 30 character
7. dial strings such as 123 to activate voice mail or other applications provided by their service provider the should be predefined inside the dial plan feature An example dial plan will be x which allows the user to dial followed by any length of numbers Defines the timeout in seconds before the call is forwarded on no answer The default value is 20 seconds When enabled Do No Disturb Call Forward and other call features will be supported locally provided ITSP 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 Allows users to configure the ringtone for the account Users can choose from different ringtones from the dropdown menu Specifies matching rules with number pattern or Alert Info text When the incoming caller ID or Alert Info matches the rule the phone will ring with selected distinctive ringtone Matching rules e Specific caller ID number For example 8321123 A defined pattern with certain length using x and to specify where x could be any digit from 0 to 9 Samples xx at least 2 digit number xx only 2 digit number 345 xx 3 digit number with the leading digit of 3 4 or 5 6 9 xx 3 digit number with the leading digit from 6 to 9 e Alert Info text Users could configure the matching rule as certain text e g priority and select
8. in process When download is done you will see the phone restarts again Please 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 over the Internet We recommend completing firmware upgrades in a controlled LAN environment whenever possible NO LOCAL TFTP HTTP SERVERS For users that would like to use remote upgrading without a local TFTP HTTP server Grandstream offers a NAT friendly HTTP server This enables users to download the latest software upgrades for their phone via this server Please refer to the webpage http www grandstream com support firmware Alternatively users can download a free TFTP or HTTP server and conduct a local firmware upgrade A free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm http tftpd32 jounin net Instructions for local firmware upgrade via TFTP 1 Unzip the firmware files and put all of them in the root directory of the TFTP server 2 Connect the PC running the TFTP server and 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 TF
9. sen Innovative IP Voice amp Video 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 Settings gt Call Features Off hook Auto Dial Off hook Timeout Disable Call Waiting Disable Call Waiting Tone Disable Direct IP Call Use Quick IP Call mode Disable Conference Enable MPK sending DTMF Disable Transfer In call dial number on pressing transfer key Auto Attended Transfer Do Not Escape as 23 in SIP URI Click To Dial Feature Call History Flash Writing Write Timeout Call History Flash Writing 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 If configured when the phone is on hook it will go off hook after the timeout in seconds The default value is 30 seconds 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 off hook the phone and d
10. set up busy call forward e Pick up the handset e Dial 90 followed by forwarding number e Press or SEND key The call will hang up automatically with busy call forward set up Cancel Busy Call Forward To cancel the busy call forward Pick up the handset e Dial 91 Ashort 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 Pickup the handset e Dial 92 followed by forwarding number e Press or SEND key The call will hang up automatically with delayed call forward set up 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 5 15 GXP1100 GXP1105 USER MANUAL Page 22 of 47 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 Ga
11. setting is No j Defines the timeout in seconds for no key entry If no key is pressed after the No Key Entry Timeout s t timeout the digits will be sent out The default value is 4 seconds Allows users to configure the key as the Send key If set to Yes the key will immediately dial out the input digits In this case this key is essentially equivalent to the Send key If set to No the key is included as part of the dialing string Use as Dial Key SETTINGS PAGE DEFINITIONS Settings gt General Settings This parameter defines the local RTP port used to listen and transmit It is the base RTP port for channel 0 When configured channel 0 will use Local RTP Port 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 for Direct IP Calling to work Use Random Port Specifies how often the phone sends a blank UDP packet to the SIP Keep alive Interval server in order to keep the ping hole on the NAT router to open The default setting is 20 seconds FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 34 of 47 Use NAT IP STUN Server
12. syslog to Selects the level of logging for syslog The default setting is None There are 4 levels DEBUG INFO WARNING AND ERROR Syslog messages are sent based on the following events product model version on boot up INFO level e NAT related info INFO level sent or received SIP message DEBUG level Syslog 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 chip exception WARNING and ERROR levels memory exception ERROR level Configures whether the SIP log will be included in the syslog messages or not The default setting is No Auto Recover From Configures whether auto recover or not when the phone is running abnormal Abnormal The default setting is Yes Display Language Selects display language on the phone Language File Postfix Specifies the language file postfix for downloaded language Enable TR 069 Enables TR 069 The default setting is No ACS URL URL for TR 069 Auto Configuration Servers ACS TR 069 Username ACS username for TR 069 TR 069 Password ACS password for TR 069 Enables periodic inform If set to Yes device will send inform packets to the Periodic Inform Enable ACS The default setting is No Periodic Inform Interval Sets up the periodic inform interval to send the inform packets to the ACS Connectio
13. the custom ring tone mapped to it The custom ring tone will be used if the phone receives SIP INVITE with Alert Info header in the following format Alert Info lt http 127 0 0 1 gt info priority Selects the distinctive ring tone for the matching rule When the incoming FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 33 of 47 sen Innovative IP Voice amp Video caller ID or Alert Info matches the rule the phone will ring with the selected ring Defines the timeout in seconds for the rings on no answer The default setting Ring Timeout 9 is 60 seconds If set to Yes the From header in outgoing INVITE messages will be set to Send Anonymous anonymous essentially blocking the Caller ID to be displayed Anonymous Call neos If set to Yes anonymous calls will be rejected The default setting is No Rejection If set to Yes the phone will automatically turn on the speaker phone to Allow Auto Answer by answer incoming calls after a short reminding based on the SIP info Call Info UN header sent from the server proxy The default setting is No Refer To Use Target If set to Yes the Refer To header uses the transferred target s Contact Contact header information for attended transfer The default setting is No Transfer on Conference Defines whether or not the call is transferred to the other party if the initiator of Hangup the conference hangs up The default
14. Ae Innovative IP Voice amp Video Grandstream Networks Inc GXP1100 GXP1105 Small Business IP Phone GXP1100 GXP1105 USER MANUAL sen Innovative IP Voice 4 Video GXP1100 GXP1105 User Manual Index GUI INTERFACE 5 GNU GPL INFORMATION eren rune nnn n nnn nnn nn ar a saa Ra caua 6 CHANGE O EEO Sa TEENIE NEEGER 7 FIRMWARE VERSION 1 0 5 15 cccccccccccccccecccccccececcccceucccecsceuucccaceaueeeuceaceeaeeeaueuacecaeececeueecaeensceaeeeees 7 FIRMWARE VERSION 1 0 4 23 c ccccccccccccesccceccececcccueuccacecseceucececeuseeeueeaueeaeecaueuaeeeaeecaueuaeeeaueuseeaeeenes 7 FIRMWARE VERSION 1 0 4 9 ccccccccecccccecccccccccececcceeccucecseccucecaeeueeeaeeaeeeaeeccecaueeaeeceueaaeecaeenseeaeeeees 7 LU CHE S l CENE PTT 8 PRODUCT 9 FEATURE HIGHTLIGHTS 9 GXP1100 GXP1105 TECHNICAL SPECIFICATIONS 0ccccccceccceccececceccccccceaccceuecacccauecesenaeeeaeees 9 NS TAL E qe 11 EQUIPMENT PACKAGING esses e e mnm senem rre nr serie resi srii rper ssi srii pss serrer se resp resi serena 11 CONNECTING YOUR PHONE ccccccccccccececcccceccccccacccecccucecacecececuceceeeuceeaueecceeauecececae
15. Enter the number gt Press SEND key or to dial out e Redial Redial the last dialed number gt Take handset off hook You shall hear dial tone from the handset Press SEND key Speed Dial Dial the number configured as Speed Dial on Multi Purpose Key gt Goto GXP1100 GXP1105 Web GUI Settings Programmable Keys configure the Multi Purpose Key s Key Mode as Speed Dial Enter the Description and Value the number to be dialed out for the Multi Purpose Key Click on Save and Apply at the bottom of the Web GUI page Take 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 Go to GXP1100 GXP1105 Web GUI Settings Programmable Keysp configure the Multi Purpose Key s Key Mode as Call Return No Value has to be set on the Multi Purpose Key for Call Return Take handset off hook You shall hear dial tone from the handset Press the configured Call Return key to dial out FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 15 of 47 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 f digits have been entered after handset is off hook the SEND key will works as SEND instead of REDIAL e def
16. Header options in web GUI ACCOUNT PAGE DEFINITIONS e gt Added NAT Settings information NAT SETTINGS e Added Click to Dial feature CLICK TO DIAL 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 PAGE e Added IPv6 configuration options 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 5 15 GXP1100 GXP1105 USER MANUAL Page 7 of 47 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 expressly approved by Grandstream or
17. TP 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 5 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 hitp httpd apache org or use Microsoft IIS web server Note When the phone boots up it will send a TFTP or HTTP request to download the configuration file where is the MAC address of the phone If itis normal TFTP or HTTP upgrade the following messages TFTP Error from IP ADRESS requesting cfg000b82023dq4 File does not exist Configuration File Download can be ignored in the TFTP HTTP server log FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 44 of 47 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 config
18. VPN This simulates a PBX function using the CMSA CD without a SIP FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 16 of 47 sen Innovative IP Voice amp Video server Controlled static IP usage is recommended To enable Quick IP Call Mode go to GXP1100 GXP1105 Web GUI Settings Call Features 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 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 or x are also valid The leading 0 is not required but it s 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 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 Whe
19. XP1100 GXP1105 TECHNICAL SPECIFICATIONS SIP RFC3261 TCP IP UDP RTP RTCP HTTP HTTPS ARP ICMP DNS A record SRV NAPTR DHCP PPPoE TELNET TFTP NTP STUN TR 069 802 1x LLDP LLDP MED 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 G726 32 G722 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 MDs5 sess based authentication 256 bit AES encrypted configuration file TLS SRTP 802 1x media access control English German Italian French Spanish Portuguese Russian Croatian FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 9 of 47 Upgrade and Provisioning Power and Green Energy Efficiency Physical Operating Temperature Humidity Package Content Compliance Gosia Innovative I Simplified Chinese traditional Chinese Korean Japanese and etc supported in web configuration interfac
20. ameter to Traversal Default setting is No Enable the device to use NAT traversal when it is behind firewall on a private network Select Keep Alive Auto STUN with STUN server path configured too or other option according to the network setting CLICK TO DIAL From GXP1100 GXP1105 Web GUI users could dial out with Click to Dial feature on the top menu of the Web GUI when the account is registered Before using the Click To Dial feature make sure the option Click To Dial Feature under web GUI Settings Call Features is turned on By default it s disabled and the dialing icon in web GUI is in grey 4 After clicking on the icon a new dialing window will show as the figure below Enter number and click FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 41 of 47 sen Innovative IP Voice amp Video on Dial the phone will go off hook and dial out the number from account 1 Click to Dial Account Dial Number 1085 Figure 4 Click to Dial Additionally users could directly send the command for the phone to dial out by specifying the following URL in PC s web browser or in the field as required in other call modules hitp ip_address cgi bin api make_call phonenumber 1234 amp account 0 amp password admin In the above link replace the fields with address Phone s IP Address phonenumber 1234 The number for the phone to dial out
21. ault 4 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 gt Call Settings MAKING CALLS USING IP ADDRESSES Direct IP Call allows two phones to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both phones have public IP addresses or e Both phones are on the same LAN VPN using private or public IP addresses 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
22. ce call gt Establish two active calls with two parties respectively Press the Multi Purpose Key previously configured as 3 way Conference in Web GUI 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 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 19 of 47 sen establishing the conference call Press FLASH key to toggle between the 2 lines Users could re establish conference call by pressing the Multi Purpose Key again 3 End Conference 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 Call Features set Enable FLASH key as CONF to Yes Click on Save and Apply on the bottom of the Web GUI page 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 Press the FLASH Key gt conference will be established 2 Split call in conference gt During the 3 way conference press HOLD key The conferen
23. ce 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 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 Note e 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 Conference Hangup is turned on e option Disable Conference has to be set to No to establish conference on GXP110x FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 20 of 47 VOICE MESSAGES MESSAGE WAITING INDICATOR Ghin Innovative I 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 gt General Settings 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 th
24. computer to the same hub or switch as the phone connected to In absence of a hub switch or free ports on the hub switch please connect the computer directly to the PC port on the back of the phone If the 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 The default administrator password is set to admin The default end user password is set to 123 When changing any settings always SUBMIT them by pressing the Save or Save and Apply button on the bottom of the page If the change is saved only but not applied after making all the changes click on the APPLY button on top of the page to submit After submitting the changes in all the Web GUI pages reboot the phone to have the changes take effect if necessary All the options under Accounts page and Phonebook page do not require reboot Most of the options under Settings page do not require reboot 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 Status Displays the Account status Network status and System Info of the phone Account To co
25. e Firmware upgrade via TFTP HTTP HTTPS mass provisioning using TR 069 or AES encrypted XML configuration file Universal power adapter Input 100 240 VAC 50 60Hz Output 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 0 40 10 9096 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 5 15 GXP1100 GXP1105 USER MANUAL Page 10 of 47 Innovative IP Voice 4 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 Yes Yes Yes Yes Yes Yes Power Handset Port LAN Port Figure 1 GXP1100 GXP1105 Ports Table 3 GXP1100 GXP1105 CONNECTORS Handset Port RJ9 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 5 15 GXP1100 GXP1105 USER MANUAL Page 11 of 47 sen Innovat
26. e IP Voice amp Video 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 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 Account x SIP Settings gt Basic Settings 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 SIP URI Scheme when using TLS Use Actual Ephemeral 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
27. e Transfer Enter the number in the value field to be transferred blind transfer during the call e Intercom Enter the extension number in the value field to do the intercom FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 36 of 47 sen Innovative IP Voice amp Video 3 way Conference To establish 3 way conference NETWORK PAGE DEFINITIONS Network gt Basic Settings Internet Protocol IPv4 Address Type DHCP Host name Option 12 DHCP Vendor Class ID Option 60 PPPoE Account ID PPPoE Password PPPoE Service Name IPv4 Address Subnet Mask Gateway DNS Server 1 DNS Server 2 Preferred DNS Server IPv6 Address Type Static Address IPv6 Prefix Length IPv6 Prefix DNS Server 1 DNS Server 2 Preferred DNS server 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 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 Enter the PPPoE account ID Enter the PPPoE Password Enter the PPPoE Service Name Enter the IP address when static IP is used Enter the Subnet Mask when static IP is used for IPv4 Enter the Default Gateway when static IP is used for IPv4 Enter the DNS Server 1 when stat
28. e in depth support Thank you again for purchasing Grandstream IP phone it will be sure to bring convenience and color to both your business and personal life FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 47 of 47
29. e 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 FEATURES Block Caller ID for all subsequent calls 30 e Off hook the phone e Dial 30 Send Caller ID for all subsequent calls 31 e Off hook the phone e Dial 31 Block Caller ID per call 67 e Off hook the phone e Dial 67 and then enter the number to dial out Send Caller ID per call 82 Off hook the phone e Dial 82 and then enter the number to dial out Disable Call Waiting per Call 70 Off hook the phone e Dial 70 and then enter the number to dial out cx Enable Call Waiting per Call Off hook the phone FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 21 of 47 72 73 90 91 92 93 Innovative I 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 The call will hang up automatically with unconditional call forward set up Cancel Unconditional Call Forward To cancel the unconditional call forward Pickup the handset e Dial 73 A short tone will be heard Wait for the call to hang up The unconditional call forward is cancelled Busy Call Forward To
30. efault value is North American standard Frequencies should be configured with known values to avoid uncomfortable high pitch sounds Call Progresses Tones System Ring Tone Dial Tone Message Waiting Ring Back Tone Call Waiting Tone Busy Tone Reorder Tone Syntax f1 val f2 val c on1 off1 on2 off2 on3 off3 Frequencies are in Hz and cadence on and off are in 10ms ON is the period of ringing On time in ms while OFF is the period of silence In order to set a continuous ring OFF should be zero Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern Up to three cadences are supported Configures the call waiting tone gain to adjust call waiting tone volume Call Waiting Tone Gain ee The default setting is Low Settings gt Audio Control Configures the transmission gain of the handset The default value is 0 dB Settings gt Programmable Keys Assigns a function to the corresponding Multi Purpose Key The key mode options are Handset TX gain e Speed Dial Enter the Speed Dial number in Value field to be dialed e Dial DTMF Enter a series of DIMF digits in the Value field to be dialed during the call Enable MPK Sending has to be set to Yes first e Voice Mail Multi Purpose Keys X Enter the Voice Mail access number in the Value field e Call Return The last answered calls can be dialed out by using Call Return The Value field should be left blank
31. ersion 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 Aux Aux version number Dsp DSP version number System Up Time System up time since the last reboot System Time Current system time on the phone system Service Status GUI and Phone service status Core Dump Core dump file that could be downloaded for troubleshooting purpose FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 26 of 47 sen Innovative IP Voice amp Video ACCOUNT PAGE DEFINITIONS Account x gt General Settings Account Active Activates deactivates account The default setting is Yes Account Name The name associated with the SIP account The URL or IP address and port of the SIP server This is provided by your VoIP service provider ITSP SIP Server The URL or IP address and port of the SIP server This will be used when the Secondary SIP Server s 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 Outbound Proxy User account information provided by your VoIP service provider ITSP It s SIP User ID usually in the form of digits similar to phone number or ac
32. esh 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 Session Expiration The minimum session expiration in seconds The default value is 90 seconds Min SE If set to Yes and the remote party supports session timers the phone will use Caller Request Timer B a session timer when it makes outbound calls If set to Yes and the remote party supports session timers the phone will use Callee Request Timer MS 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 Force Timer As a Caller select UAC to use the phone as the refresher or select UAS to UAC Specify Refresher du 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 Account x
33. etauececeaeesenes 11 SAFETY 12 WARRANTY cceccceccecececccucccececccccucccaceuscceuccaeeeaeecaueuaeecauecseeaaeecacedseeeueesueeaeecceuaeeeaeecseeaneeseeeaeeeauees 13 USING THE 1100 1105 14 GETTING FAMILAR WITH THE KEYPAD csseeee m emm m nene n rn 14 MAKING PHONE 15 2 CALLS WITH 1 SIP ACCOUNT ccccccccccccccccececceceeccnececsenseetececeesesenaeesceensetansusesenseransunecenss 15 COMPLETING CALLS u i cccccccceccscccecccscccceccecensecacecuecenserseennsessenseenuusnesenseeaneusesensersetsnseesesnesesaees 15 MAKING CALLS USING 6 5 16 ANSWERING PHONE 17 RECEIVING CAL kL kG E 17 DURING A PHONE CALL 2 ccccccceecccccccccceccceeccececceccececaceceeecuceeeeeueeceuecauecaeeceeeaaueeaeecseeaeeeeeeeaeeeaees 17 CALL WAITING CALL 17 MUE MEE 18 CALL TRANSFER 18 3 WAY CONFERENCING 0ccccccccccececcececccceccccecccsccececnecsesensecaueusceensenseetaeesdeenseenessetense
34. hot of Reboot Page NOOO ff OD FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 5 of 47 sen 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 qrandstream com support fag gnu FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 6 of 47 sen Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions 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 5 15 e Updated Web GUI interface examples with new screenshots for 1 0 5 15 GUI INTERFACE EXAMPLES e Added pin out information CONNECTING YOUR PHONE e Updated Auto Attended Transfer information CALL TRANSFER e Updated Click To Dial feature information CLICK TO DIAL e Updated Web GUI options DEFINITIONS FIRMWARE VERSION 1 0 4 23 e Updated generic config file cfg xml information CONFIGURATION FILE DOWNLOAD e Added Use Privacy Header and Use P Preferred Identity
35. ial X is 0 9 and XXX 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 or are also valid so leading 0 is not required but 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 Disables the Transfer function The default setting is No Configures the number for the phone to dial as DTMF during the call using TRAN button If set to Yes the phone will use attended transfer by default The default setting is No Specifies whether to replace by 9623 or not for some special situations The default setting is No Enables Click To Dial feature The default setting is Disabled Defines the interval in seconds to save the call history to phone s flash The default value is 300 seconds Defines the number of unsaved logs before written to phone s flash The FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 35 of 47 sen Innovative IP Voice 4 Video Max Unsaved Log default value is 200 entries If set to Yes the FLASH key can be used to establish 3 way Enable FLASH Key as CONF conference The default setting is No Settings gt Ring Tone Configures ring or tone frequencies based on parameters from local telecom The d
36. ic IP is used for IPv4 Enter the DNS Server 2 when static IP is used for IPv4 Enter the Preferred DNS Server for IPv4 Allows users to configure the appropriate network settings on the phone to obtain IPv6 address Users could select Auto configured or Statically configured for the IPv6 address type Enter the static IPv6 address when Full Static is used in Statically configured IPv6 address type Enter the IPv6 prefix length when Full Static is used in Statically configured IPv6 address type Enter the IPv6 Prefix 64 bits when Prefix Static is used in Statically configured IPv6 address type Enter the DNS Server 1 for IPv6 Enter the DNS Server 2 for IPv6 Enter the Preferred DNS Server for IPv6 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 37 of 47 A Innovative IP Voice amp Video Network gt Advanced Settings 802 1X mode 802 1X Identity MD5 Password 802 1X CA Certificate 802 1X Client Certificate HTTP Proxy HTTPS Proxy Layer 3 QoS Layer 2 QoS 802 1Q VLAN Tag Layer 2 QoS 802 1p Priority Value PC Port Mode 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 Upload 802 1X CA certificate to the phone or delete existed 802 1X CA certificate from the phone Upload 802 1X Client
37. ifies 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 7 different vocoder types are supported on the phone including G 711 U law PCMU G 711 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 When set to Yes the device will use the first matching vocoder in the received 2000 SDP as the codec The default setting is No 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
38. ion 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 Dial 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 when there is a previously dialed call Volume Press or to adjust the volume Standard phone keypad FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 14 of 47 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 Take handset off hook You shall hear dial tone from the handset
39. is when Huawei IMS special Use Privacy Header feature is on the Privacy Header will not show in INVITE If set to Yes the Privacy Header will always show in INVITE If set to No the Privacy Header will not show in INVITE Controls whether the P Preferred Identity Header will present in the SIP INVITE message or not The default setting is default which is when Use P Preferred Identity Huawei IMS special feature is on the P Preferred Identity Header will not Header show in INVITE If set to Yes the P Preferred Identity Header will always show in INVITE If set to No the P Preferred Identity Header will not show in INVITE Account x gt SIP Settings gt Advanced Features Configures Music On Hold URI to call when a call is on hold This feature has Music On Hold URI to be supported the server side Special Feature Different soft switch vendors have special requirements Therefore users may FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 29 of 47 sen Innovative IP Voice amp Video 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 Account x SIP Settings Session Timer 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 refr
40. ive 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 Please see below the pin out information for GXP1100 GXP1105 GXP1100 GXP1105 Handset Jack GXP1100 GXP1105 Handset Plug p Figure 2 GXP1100 GXP1105 Pin out 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 wa
41. key gt Press FLASH key to transfer the call Auto Attended Transfer gt Set Auto Attended Transfer to Yes under Web GUI gt Settings gt Call Features And then click Save and Apply on the bottom of the page Establish one call first gt During the call press TRAN key A new line will be brought up and the first call will be automatically placed on hold gt Enter the number and press SEND key or to make a second call FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 18 of 47 ndstream Innovative IP Voice amp Video gt Press TRAN key again The call will be transferred Note e transfer calls across SIP domains SIP service providers must support transfer across SIP domains 3 WAY CONFERENCING GXP1100 GXP1105 can host 3 way conference call PCMU PCMA by using Multi Purpose Key or FLASH key e To use Multi Purpose Key to establish 3 way conference call go to GXP1100 GXP1105 Web GUI gt Settings gt Programmable Keys configure the 3 way conference as the Multi Purpose Key mode Click Save and Apply on the bottom of the page Then follow the steps below for 3 way conferencing Programmable Keys Account Description Speed Dial Voice Mail Call Transfer Tees LDAP Search poer 3 way Conference Save Save and Apply Figure 3 GXP1100 GXP1105 Multi Purpose Key 3 way Conference 1 Initiate a conferen
42. n Request The user name for the ACS to connect to the phone Username Connection Request The password for the ACS to connect to the phone Password Connection Request Port The port for the ACS to connect to the phone CPE SSL Certificate The Cert File for the phone to connect to the ACS via SSL FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 40 of 47 A Innovative IP Voice amp Video CPE SSL Private Key The Cert Key for the phone to connect to the ACS via SSL SSL TLS Certificate SSL Certificate used for SIP Transport in TLS TCP SSL TLS Private Key SSL Private key used for SIP Transport in TLS TCP SSL TLS Private Key SSL Private key password used for SIP Transport in TLS TCP Password Download Device Click to download the device configuration file in txt format Configuration NAT SETTINGS If the devices are kept within a private network behind a firewall we recommend using STUN Server The following settings are useful in the STUN Server scenario e STUN Server Enter a STUN Server IP or FQDN that you may have or look up a free public STUN Server on the internet and enter it on this field If using Public IP keep this field blank e Use Random Ports This setting depends on your network settings When set to Yes it will force random generation of both the local SIP and RTP ports This is usually necessary when multiple GXPs are behind the same NAT If using a Public IP address set this par
43. n 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 hold by pressing the HOLD key e Resume Press the HOLD key again to resume e Multiple calls Automatically place active call on hold or switch between two calls by pressing the FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 17 of 47 sen Innovative IP Voice amp Video 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 Blind Transfer gt 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 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
44. nd HTTPS Firmware version information Firmware upgrade mode Enter 9 to toggle among the following three options e always check 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 5 15 GXP1100 GXP1105 USER MANUAL Page 24 of 47 sen Innovative IP Voice amp Video 4 Opena 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 The computer has to be connected to the same sub network as the phone This can be easily done by connecting the
45. nfigure the SIP account Network To configure network settings Settings To configure call features ring tone programmable keys and etc Maintenance To configure web Telnet access upgrading and provisioning language settings TR 069 security and etc FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 25 of 47 sen Innovative IP Voice amp Video STATUS PAGE DEFINITIONS Status gt Account Status SIP User ID Displays the configured SIP User ID SIP Server Displays the configured SIP Server address SIP Registration Displays SIP registration status YES NO Status gt Network Status Global unique ID of device in HEX format The MAC address will be used for MAC Address provisioning and can be found on the label coming with original box and on the label located on the back of the device IP Setting DHCP Static IP or PPPoE IPv4 Address The IPv4 address obtained on the phone IPv6 Address The IPv6 address obtained on the phone Subnet Mask The subnet mask obtained on the phone Gateway The gateway address obtained on the phone DNS Server 1 The DNS server address 1 DNS Server 2 The DNS server address 2 PPPoE Link Up PPPoE connection status NAT Traversal NAT traversal status for each account Status gt System Info Product Model Product model of the phone Part Number Product part number e Boot boot version number e Core core version number e Base base version number Software V
46. one 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 kx 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 Doo moo 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 will reboot Otherwise it will announce Invalid Entry and exit to the main menu FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 46 of 47 sen Innovative IP Voice amp Video EXPERIENCING THE GXP1100 GXP1105 Please visit our website http www 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 receiv
47. 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 5 15 GXP1100 GXP1105 USER MANUAL Page 8 of 47 Innovative I PRODUCT OVERVIEW FEATURE HIGHTLIGHTS e Single SIP Account up to 2 calls 4 programmable keys HD handset with support for wideband audio e Single 10 100Mbps network port integrated PoE GXP1105 only e 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 Wall Mountable QoS Security Multi language Table 1 G
48. rom route This is used for the SIP Remove OBP from route Extension to notify the SIP server that the device is behind a NAT Firewall Defines whether SIP Instance ID is supported or not The default setting is Yes When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent Support SIP Instance ID SUBSCRIBE for MWI periodically The phone supports synchronized and non synchronized MWI The default setting is No SUBSCRIBE for When set to Yes a SUBSCRIBE for Registration will be sent out periodically Registration The default setting is No The use of the PRACK Provisional Acknowledgment method enables reliability to SIP provisional responses 1xx series This is very important in Enable 100rel 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 When set to Auto the phone will look for the caller ID in the order of P Asserted Identity Header Remote Party ID Header and From Header in the incoming SIP INVITE When set to Disabled all incoming calls are displayed with Unavailable When set to From Header the phone will display the caller ID based on the From Header in the incoming SIP INVITE The default setting is Auto Caller ID Display Controls whether the Privacy Header will present in the SIP INVITE message or not The default setting is default which
49. rranty does not cover damages to the phone caused by unsupported power adapters FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 12 of 47 sen Innovative IP Voice amp Video WARRANTY If the GXP1100 GXP1105 phone was purchased from a reseller please contact the company where the phone was purchased for replacement repair or refund If the phone was purchased directly from Grandstream contact the Grandstream Sales and Service Representative for a RMA Return Materials 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 5 15 GXP1100 GXP1105 USER MANUAL Page 13 of 47 ream 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 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 opt
50. s Confirms the admin password field to be the same as above Maintenance gt Upgrade and Provisioning FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 38 of 47 Firmware Upgrade and Provisioning XML Config File Password HTTP HTTPS User Name HTTP HTTPS Password Upgrade Via Firmware Server Path Config Server Path Firmware File Prefix Firmware File Postfix Config File Prefix Config File Postfix Allow DHCP Option 43 and Option 66 Override Server Allow DHCP Option 120 to override SIP Server Automatic Upgrade Hour of the Day 0 23 Day of the Week 0 6 Authenticate Conf File Gosia Innovative I 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 firmware server It could be different from the configuration server for provisioning Defines the server path for provisioning It could be different from the firmware server for upgrading Enables your ITSP to lock firmware updates If configured only the firmware
51. setting is Yes If set to Yes the SIP user s registration information will be cleared when the phone reboots The SIP Contact header will contain to notify the server to unbind the connection The default setting is No Specifies the frequency in minutes in which the phone refreshes its 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 for Account 1 Specifies the interval to retry registration if the process is failed The default value is 20 seconds SIP T1 Timeout The default setting is 0 5 seconds SIP T2 Interval The default setting is 4 seconds Determines the network protocol used for the SIP transport Users can choose from TCP UDP and TLS Specifies if sip or sips will be used when TLS TCP is selected for SIP Transport The default setting is sips Defines whether the actual ephemeral port in contact with TCP TLS will be FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 28 of 47 sen Innovative IP Voice amp Video Port in Contact with used or not This is used when TLS TCP is selected for SIP Transfer The TCP TLS default setting is No Configures to remove outbound proxy f
52. teway IP address DNS Server IP address 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 PCMU POMA e e G 726 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 23 of 47 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 Gosia e G 723 e G 729 Announces the MAC address of the unit Announces current Firmware Server IP address Enter 12 digit new IP address Announces current Config Server Path IP address Enter 12 digit new IP address Upgrade Protocol for firmware and configuration update Enter 9 to toggle between HTTP TFTP a
53. 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 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 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 of the phone Then login with the administrator username and password Go to Maintenance gt Upgrade and Provisioning 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 Save and Apply button Then Reboot or power cycle the phone to update the new firmware FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 43 of 47 sen The indicator on 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
54. tsnssnetenss 19 VOICE MESSAGES MESSAGE WAITING INDICATOR sisse enne 21 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 2 of 47 sen Innovative IP Voice 4 Video CALE FEATURE Sinuaren e dud ue E UI oM NN AMNEM PME E 21 CONFIGURATION GUIDE 23 CONFIGURATION VIA IVR radi aps qat titius 23 CONFIGURATION VIA WEB BEOWDSERU sa iR ltd 24 25 STATUS PAGE DEEINITIDNS 26 ACCOUNT PAGE DEFINITIONS 27 SETTINGS PAGE DEIN 34 NETWORK PAGE DEFINITIONS 37 MAINTENANCE PAGE 38 NAT SETIUNGS 41 CLICK TODIAE 41 UPGRADING AND PROVISIONING eene 43 UPGRADE VIA IVR 43 Macc cmd ifc 43 NO LOCAL brin t 44 CONFIGURATION FILE DOWNLOAD 45 RESTORE FACTORY DEFAULT SETTINGS 46 EXPERIENCING THE GXP1100 GXP1105 47 FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 3 of 47 sen Innovative IP Voice amp Video Table of Tables GXP1100 GXP1105 User Manual Table 1 GXP1100 GXP1105 TECHNICAL
55. tually a phone number SIP service subscriber s Authenticate ID used for authentication It can be identical to or different from the SIP User ID Authenticate ID The account password required for the phone to authenticate with the ITSP Authenticate Password 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 Name Allows you to access voice messages by pressing the MESSAGE button on Voice Mail User ID the phone This ID is usually the VM portal access number For example in Asterisk server 8500 could be used Account x gt Network Settings This parameter controls how the Search Appliance looks up IP addresses for hostnames There are four modes A Record SRV NATPTR SRV Use Configured IP The default setting is A Record If the user wishes to locate the server by DNS SRV the user may select SRV or NATPTR SRV If Use DNS Mode Configured IP is selected please fill in the three fields below e Primary IP The primary IP address where the phone sends DNS query to e Backup IP 1 e Backup IP 2 This parameter configures whether the NAT traversal mechanism is activated Users could select the mechanism from No STUN Keep Alive UPnP Auto or NAT Traversal FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 27 of 47 Proxy Require sen Innovativ
56. uration 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 the GXP1100 GXP1105 boots up or reboots it will issue a request to download a configuration XML file named cfgxxxxxxxxxxxx xml followed by a file named cfgxxxxxxxxxxxx where XXXXXXXXXXXX is the MAC address of the phone i e cfg000b820102ab xml and cfg000b820102ab If the download of file is not successful the provision program will download a generic cfg xml file The configuration file name should be in lower case letters For more details on XML provisioning please refer to http www grandstream com general gs provisioning guide public pdf FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 45 of 47 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 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 ph
57. 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 on the LAN side the TFTP server can be redirected The default setting is Yes Enables DHCP Option 120 from local server to override the SIP Server on the phone The default setting is No Enables automatic upgrade and provisioning The default setting is No Defines the hour of the day to check the HTTP TFTP server for firmware upgrades or configuration files changes The default value is 1 Defines the day of the week to check HTTP TFTP server for firmware upgrades or configuration files changes The default value is 1 Authenticates configuration file before acceptance The default setting is No FIRMWARE VERSION 1 0 5 15 GXP1100 GXP1105 USER MANUAL Page 39 of 47 Aaa Innovative IP Voice amp Video Maintenance gt Syslog Syslog Server The URL or IP address of the syslog server for the phone to send
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