Grandstream Networks, Inc.
TABLE OF CONTENTS GXP1100/1105 USER MANUAL WELCOME ................................................................................................................................................................. 3 INSTALLATION......................................................................................................................................................... 4 EQUIPMENT PACKAGING .............................................................................................................
GUI INTERFACE EXAMPLES GXP1100/1105 USER MANUAL http://www.grandstream.com/products/gxp_series/general/documents/gxp21xx_gui.zip 1. Screenshot of Configuration Login Page 2. Screenshot of Status Page 3. Screenshot of Basic Setting Configuration Page 4. Screenshot of Advanced User Configuration Page 5. Screenshot of SIP Account Configuration Page 6. Screenshot of Saved Configuration Changes Page 7. Screenshot of Reboot Page Grandstream Networks, Inc. GXP1100/1105 User Manual Firmware version: 1.0.1.
Welcome GXP1100/1105 is a next generation small business IP phone that features up to 2 call appearances with 1 SIP account, 4 programmable keys, single network port, integrated PoE (GXP1105 only). The GXP1100/1105 delivers superior HD audio quality, leading edge telephony features, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.
Installation EQUIPMENT PACKAGING Table 1: Equipment Packaging GXP1100/1105 Yes Yes Yes Yes (GXP1100 only) Yes Yes Yes Main Case Handset Phone Cord Power Adaptor Ethernet Cable Base Stand Quick Start Guide 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 LCD Display N/A Number of Lines 1 Programmable 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, 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.
Weight Temperature Humidity Compliance Unit weight: 0.6KG Package weight: 1.0KG (GXP1100), 0.
Firmware Upgrades Advanced Server Features Security Grandstream Networks, Inc. Support for IEEE 802.
Using the GXP1100/1105 GETTING FAMILIAR WITH THE KEYPAD Table 7: GXP1100/1105 Keypad Buttons Key Button Key Button Definitions Place active call on hold Call waiting: bring up a new line or answer the second incoming call Transfer an active call to another number Enter to retrieve voice mails or other messages Programmable hard key.
• Press the SEND key. 3. VIA SPEED DIAL: On the GXP1100/1105, the Multiple Purpose Key (programmable hard key) has to be configured as Speed Dial and with the correct name and user ID under Web GUI->Basic Settings configuration. • Take handset off hook • Press the configured Speed Dial key 4. VIA CALL RETURN: On the GXP1100/1105, the Multiple Purpose Key (programmable hard key) has to be configured as Call Return under Web GUI->Basic Settings configuration.
#xxx where x is 0-9 and xxx <=255. A direct IP call to aaa.bbb.ccc.XXX will be completed. “aaa.bbb.ccc” is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK). For example: 192.168.0.2 calling 192.168.0.3 -- dial #3 followed by # 192.168.0.2 calling 192.168.0.23 -- dial #23 followed by # 192.168.0.2 calling 192.168.0.123 -- dial #123 followed by # 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.
NOTE: To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains. Voice Messages (Message Waiting Indicator) A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Dial into the voicemail box to retrieve the message by entering the voice mail number of the server or pressing the MSG button (Voice Mail User ID has to be properly configured as the voice mail number under web GUI->Account 1).
Configuration Guide The GXP1100/1105 can be configured in two ways. Firstly, using the IVR MENU by the keypad on the phone; secondly, through embedded web configuration menu. CONFIGURATION VIA IVR MENU GXP1100/1105 has a built-in voice prompt menu for simple device configuration. Pick up the handset and dial “***” to use the IVR menu. Table 9: GXP1100/1105 IVR Menu Definitions Menu Voice Prompt Options Main Menu “Enter a Menu Option” Press “*” for the next menu option.
Enter MAC address to restore factory default setting. (See Restore Factory Default Setting section) Automatically returns to Main Menu “Invalid Entry” 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 10: 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 Multi Purpose Key X These options are used to assign a function to the corresponding multiple purpose key. Options available are: 1. Speed Dial 2. Dial DTMF 3.
Keep-alive interval This parameter specifies how often the GXP1100/1105 sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds. Use NAT IP NAT IP address used in SIP/SDP message. Default is blank. STUN Server IP address or Domain name of the STUN server. STUN resolution result will display in the STATUS page of the Web UI.
Automatic Upgrade This function is used by ITSP. End user should NOT touch these parameters. Default is “No”. Choose “Yes” to enable automatic HTTP upgrade and provisioning. In “Check for upgrade every” field, enter the number of minutes to check the HTTP server for firmware upgrade or configuration changes. When set to “No”, the phone will only perform HTTP upgrade and configuration check once at boot up. Authenticate Conf File Default is “No”.
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, Slovenian, Arabic, Hebrew and Dutch.
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.
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.722 (wide-band). Configure Vocoders in a preference list that is included with the same preference order in SDP message. Enter the first Vocoder in this list by choosing the appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by choosing the appropriate option in “Choice 8”. SRTP Mode Enable SRTP mode based on selection.
iLBC Frame Size iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required. iLBC Payload Type Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. Special Feature Default is Standard. Choose the selection to meet special requirements from Soft Switch vendors. SAVING THE CONFIGURATION CHANGES After the user makes a change to the configuration, press the “Update” button in the Configuration Menu.
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. 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. INSTRUCTIONS FOR RESTORATION: Step 1: Press “***” to enter the IVR menu. Input “99” to for factory reset. Step 2: Enter the MAC address printed on the bottom of the sticker.
