Grandstream Networks, Inc. GXP1100/1105 Small-Medium Business IP Phone Grandstream Networks, Inc. GXP1100/1105 User Manual Firmware version 1.
TABLE OF CONTENTS GXP1100/1105 USER MANUAL EQUIPMENT PACKAGING .............................................................................................................................................4 CONNECTING YOUR PHONE ........................................................................................................................................4 SAFETY COMPLIANCES................................................................................................................................
Welcome GXP1100/1105 is a next generation small-to-medium business IP phone that features 1 lines with 1 SIP account, 4 XML programmable context-sensitive soft keys, one network ports with integrated PoE (GXP1105 only).
Installation EQUIPMENT PACKAGING Table 1: Equipment Packaging Main Case Handset Phone Cord Power Adaptor Ethernet Cable Base Stand Quick Start Guide GXP1100/1105 Yes Yes Yes Yes (GXP1100 only) Yes Yes Yes CONNECTING YOUR PHONE The connectors of the GXP1100/1105 are located on the bottom of the device.
Product Overview Table 3: GXP1100/1105 Feature Guide Features GXP1100/1105 Number of Lines 1 Programmable Soft Keys 4 Extension Module N/A Table 4: GXP1100/1105 Key Features in a Glance Features Benefits Open Standards Compatibility SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV and NAPTR), DHCP (both client and server), PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, SIP over TLS, 802.
Table 6: GXP1100/1105 Technical Specifications Lines Protocol Support Display Feature Keys Device Management Audio Features Telephony Features Network and Provisioning Firmware Upgrades Advanced Server Features Security Grandstream Networks, Inc. 1 lines with 1 SIP account, 4 XML programmable soft-keys Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols, TR-069, 802.
Using the GXP1100/1105 Table 9: GXP1100/1105 KEYPAD BUTTONS Key Button HOLD TRANSFER FLASH MSG REDIAL Key Button Definitions Place active call on hold Transfer an active call to another number Press FLASH button to answer another coming call while having an active call Press MSG button to receive the voice message。 To redial the last dialed phone number Mute an active call; or use as DND button when the phone is in idle state.
• or press the SPEAKER button Press the REDIAL soft-key NOTE: Dial-tone and dialed number display occurs after the handset is off-hook, or handset button is pressed, or speaker button is pressed. After dialing the number, the phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call. Press “#” button to override the 4 second delay. 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.
Do Not Disturb 1. 2. 3. 4. Press the MUTE button and scroll down to “Preference”. Select “Do Not Disturb” by pressing menu button. Use arrow keys to either enable or disable “Do Not Disturb” feature. When enabled, there will be a special ‘Do Not Disturb” icon appearing on the display. This will send the incoming caller directly to voicemail. PHONE FUNCTIONS DURING A PHONE CALL Call Waiting/Call Hold 1. Hold: Place a call on ‘hold’ by pressing the “HOLD” button. 2.
Press HOLD to end the conference call and put all parties on hold To speak with an individual party, select the corresponding LINE key NOTE: The party that starts the conference call has to remain in the conference for its entire duration, you can put the party on mute but it must remain in the conversation. Also, this is not applicable when the feature “Transfer on call hangup” is turned on.
*71 Enable Call Waiting (per Call) *72 Unconditional Call Forward Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wait for dial tone. LCD will display “Call FWD Activated” *73 Cancel Unconditional Call Forward: dial “*73” and get the dial tone, then hang up LCD will display “Call FWD Activated” *90 Busy Call Forward Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial tone. Hang up *91 Cancel Busy Call Forward: dial “*91”.
Configuration Guide The GXP1100/1105 can be configured through embedded web-configuration menu. CONFIGURATION VIA WEB BROWSER The GXP1100/1105 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsoft’s IE, Mozilla Firefox and Google Chrome.
Table 13: Device Configuration - Status MAC Address The device ID, in HEXADECIMAL format. IP Address This field shows IP address of GXP1100/1105. Product Model This field contains the product model information. Part Number This field contains the product part number. Software Version • Program: This is the main firmware release number, which is always used for identifying the software (or firmware) system of the phone.
802.1x Mode This option allows the user to enable/disable 802.1x mode on the phone. The default value is disabled. To enable 802.1x mode, this field should be set to EAP-MD5. Once enabled, the user would be required to enter the following information below to be authenticated on the network: • Identity • MD5 Password Line Keys x This allows the user to configure the account mapped to each line key, as well as enabling SCA (Shared Call Appearance) for the line. Options available for Key Mode are : 1.
HEADSET Key Mode Default Mode: - Toggle to Headset when using Speaker/Handset - Dial, pick up call or hang up call using Headset Toggle Headset/Speaker: - toggle between using Headset and using Speaker Headset TX gain (dB) Set headset TX gain to -6, 0 or +6. Default is 0 db. Headset RX gain (dB) Set headset RX gain to -6, 0 or +6. Default is 0 db. Table 15: Device Configuration – Settings /Advanced Settings Admin Password Administrator password.
Firmware Upgrade and Provisioning Allows the user to select the following options for firmware upgrade: • Always Check for New Firmware • Check New Firmware only when F/W pre/suffix changes • Always Skip the Firmware Check. Firmware upgrade may take up to 10 minutes depending on network environment. Do not interrupt the firmware upgrading process. Note: Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade.
TR-069 Username Enter username for TR-069. TR-069 Password Enter password for TR-069. Save Credentials Save TR-069 credentials. Default is “No”. Auto Login Auto Login TR-069 account. Default is “No”. Periodic Inform Enable Enable periodic inform. Default is “No”. Periodic Inform Interval When enabling periodic inform, set up the periodic inform interval. Connection Request Username Enter the connection request username. Connection Request Password Enter the connection request password.
Syslog Server The IP address or URL of System log server. This feature is especially useful for ITSPs. Syslog Level Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR.
Call Progress Tones Using these settings, users can configure ring or tone frequencies based on parameters from local telecom. By default, they are set to North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]; (Frequencies are in Hz and cadence on and off are in 10ms) ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence.
Display Language Allows user to choose preferred display language in web UI and key pad UI Currently, the phone supports these languages: English, Simplified Chinese, Traditional Chinese, Korean, Japanese, Italian, Spanish, French, German, Portuguese, Russian, Croatian, Hungarian, Polish and Slovenian.
Name SIP service subscriber’s name that is used for Caller ID display. DNS Mode The default is set to A Record. If user wishes to locate the server by DNS SRV, the user may select SRV or NATPTR/SRV. When "Use Configured IP" option is selected, if SIP server is configured as domain name, phone will not send DNS query, but use "Primary IP" or "Secondary IP" to send sip message if at least one of them are not empty.
NAT Traversal This parameter activates the NAT traversal mechanism. It has options: No, STUN, Keep-Alive, UPnP, Auto, VPN. If selecting STUN and a STUN server is also specified, the phone performs according to the STUN client specification. Using this mode, the embedded STUN client detects if and what type of NAT/Firewall configuration is used.
Dial Plan Dial Plan Rules: 1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d 2. Grammar: x - any digit from 0-9; a) xx+ - at least 2 digit numbers b) xx.
Session Expiration The SIP Session Timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session is terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default value is 180 seconds.
Refer-To Use Target Contact Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header uses the transferred target’s Contact header information. Transfer on Conference Defines whether or not the call is transferred to the other party if the initiator of the conference hangs up. Hangup Default setting is set to “No”. Preferred Vocoder GXP1100/1105 supports up to 7 different Vocoder types including G.711(a/µ) (also known as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.
Use # as Dial Key This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If set to “Yes”, the “#” key will immediately send the call. In this case, this key is essentially equivalent to the “(Re)Dial” key. If set to “No”, the “#” key is included as part of the dial string. G723 Rate Encoding rate for G723 codec. By default, 6.3kbps rate is set. G726-32 Packing Mode Select “ITU” or “IETF” for G726-32 packing mode. iLBC Frame Size iLBC packet frame size. Default is 20ms.
Software Upgrade & Customization Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page. FIRMWARE UPGRADE THROUGH TFTP/HTTP To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples of some valid URLs. • • firmware.mycompany.com:6688/Grandstream/1.2.3.5 72.
INSTRUCTIONS FOR LOCAL TFTP UPGRADE: 1. Unzip the file and put all of them under the root directory of the TFTP server. 2. The PC running the TFTP server and the GXP1100/1105 should be in the same LAN segment. 3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server, in the phone’s web configuration page 5. Configure the Firmware Server Path with the IP address of the PC 6.
Restore Factory Default Setting WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone. Please backup or print all the settings before you restoring factory default settings. We are not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures: —Reorient or relocate the receiving antenna. —Increase the separation between the equipment and receiver.
