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/gxp110x_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 calls with 1 SIP account, 4 programmable keys, single network port, integrated PoE (GXP1105 only). The GXP1100/1105 delivers superior HD audio quality, leading edge telephony features, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.
Installation EQUIPMENT PACKAGING Table 1: Equipment Packaging GXP1100/1105 Yes Yes Yes Yes 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, PPPoE, TELNET, TFTP, NTP, STUN, TR-069, 802.
Temperature Humidity Compliance Package weight: 1.0KG 32 -104° F/ 0 - 40°C 10% - 90% (non-condensing) FCC Part 15 (CFR 47) Class B EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1 AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS UL 60950 (power adapter) Table 6: GXP1100/1105 Technical Specifications Lines Protocol Support Feature Keys Device Management Audio Features Telephony Features Network and Provisioning Firmware Upgrades Grandstream Networks, Inc.
Advanced Server Features Security Grandstream Networks, Inc. Message waiting indication, support DNS SRV Look up and SIP Server Fail Over User and administrator level passwords, MD5 and MD5-sess based authentication, AES based secure configuration file, SRTP, TLS, 802.1x media access control GXP1100/1105 User Manual Firmware version: 1.0.1.
Using the GXP1100/1105 GETTING FAMILIAR WITH THE KEYPAD Table 7: GXP1100/1105 Keypad Definitions Key Definitions Place active call on hold Call waiting: - Bring up a new line, or - Answer the second incoming call 3-way Conference: -Establish 3-way conference Note: For 3-way Conference, before using the Flash key, “Enable Flash key as CONF” option has to be set to “Yes” under web GUI->Advanced Settings Transfer an active call to another number Enter to retrieve messages Programmable hard key for multiple pu
During the call, users can press the FLASH key to hold the current call and make/answer another call. If they are 2 calls established, users can switch the two lines by pressing the FLASH key. Completing Calls The GXP1100/1105 allows you to make phone calls by picking up the handset. There are four ways to complete calls. 1. DIAL: To make a phone call. Take handset off hook The line will have a dial tone Enter the phone number Press “#” or SEND key to send 2.
Enter “47” for Direct IP Call. After hearing “Direct IP Calling”, the dial tone will be heard again Enter the target IP address. (Please see example below) Wait for about 4 seconds and the phone will initiate the call For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input the following: 192*168*1*60#5062. The “*” key represents the dot “.”; the “#” key represents colon “:”. Wait for about 4 seconds and the phone will initiate the call.
NOTE: If users hang up the current call while there is call hold on the other line, there will be an audible ring tone indicating a call is on hold. Pick up the handset so users can resume with the call on hold. Mute Press the MUTE key to mute/unmute the microphone. Call Transfer GXP1100/1105 supports Blind transfer, Attended transfer and Auto-Attended transfer: 1. Blind Transfer: During the call, press TRAN key and dial the number Press “#” or SEND key to complete transfer of active call 2.
1. Initiate a Conference Call: Establish two active calls with two parties Press the Multi Purpose Key which is configured as “3-way Conference” in the web GUI 3-way conference will be established 2.
3. Users could select one of the ways above (MPK option or FLASH key) to establish conference call. When using “Enable FLASH key as CONF”, FLASH key will not be available to switch lines multiple times. 4. When using “FLASH” key to establish conference, GXP110x must be the initiator to establish the 2way conversation in both calls. Voice Messages (Message Waiting Indicator) A blinking red MWI (Message Waiting Indicator) indicates a message is waiting.
- The call will hang up automatically *91 Cancel Busy Call Forward - Pick up the handset - Dial “*91”. A short tone will be heard - Wait for the call to hang up *92 Delayed Call Forward - Pick up the handset - Dial “*92” followed by forwarding number - Press # or SEND key - The call will hang up automatically *93 Cancel Delayed Call Forward - Pick up the handset - Dial “*91”. A short tone will be heard - Wait for the call to hang up Grandstream Networks, Inc.
Configuration Guide The GXP1100/1105 can be configured in two ways: - IVR MENU from the keypad - Web GUI embedded on the phone CONFIGURATION VIA IVR MENU GXP1100/1105 has a built-in voice prompt menu for simple device configuration. Pick up the handset and dial “***” to use the IVR menu. Table 9: GXP1100/1105 IVR Menu Definitions Menu Voice Prompt Options Main Menu “Enter a Menu Option” Press “*” for the next menu option. Press “#” to return to the main menu.
86 99 Others “Voice Mail” “RESET” Announces number of voice mails. “Invalid Entry” Automatically returns to Main Menu Enter MAC address to restore factory default setting. (See Restore Factory Default Setting section) Press “9” to reboot the device. 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.
Advanced Settings: To set advanced network settings, codec settings, language settings and etc. Account: To configure the SIP account. Table 10: Device Configuration - Status MAC Address The device ID, in HEXADECIMAL format. This will be used for provisioning and is written on the label in the original box as well as on the label located on the back panel of the device. IP Address This field shows IP address of GXP1100/1105. Product Model This field contains the product model information.
IP Address The GXP110x operates in three modes: 1. DHCP mode: The GXP110x acquires IP address from the first DHCP server it discovers on the LAN. The DHCP option is reserved for NAT router mode. In DHCP mode, all the field values for the Static IP mode are not used. 2. PPPoE mode: Set PPPoE account ID, PPPoE password and PPPoE service name for the GXP110x to establish PPPoE sesstion. 3.
Admin Password Administrator password. Only the administrator can access the “Advanced Settings” and “Account Settings” page. Password field is purposely blank for security reasons after clicking update and saved. The maximum password length is 25 characters. Confirm Password Enter the end user password again as above to confirm new password. Layer 3 QoS This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or Diff-Serv or MPLS. Default value is 12.
Upgrade Via This field allows the user to choose the firmware upgrade/config server path method: TFTP, HTTP or HTTPS. • TFTP: GXP110x retrieves the new firmware files or new configuration file from the specified TFTP server path at boot time. If there is no new firmware file or configuration file, the system will start the boot process using the existing firmware or config file. If a TFTP server is configured and new firmware files are retrieved, the new downloaded image is saved into the Flash memory.
Automatic Upgrade 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. Note: This function is used by ITSP. End user should NOT touch these parameters. Authenticate Conf File Default is “No”.
Syslog Level Select the syslog level for GXP110x to report. 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.
Secondary SIP Server This field contains the URL or the IP address of a second SIP server. When this field is configured, GXP110x will send out Registration requests and Subscribe messages (except for message waiting) to the “SIP Server” and “Secondary SIP Server” for the same account. When making a call, GXP110x will use the registered primary “SIP server” first. If this primary “SIP Server” is not available, the registered “Secondary SIP Server” will be used.
Register Expiration This parameter allows user to specify the time frequency (in minutes) that GXP1100/1105 refreshes its registration with the specified registrar. The default interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days). Reregister Before Expiration This parameter allows user to specify the time frequency (in seconds) that GXP1100/1105 sends out a re-registration request before the Register Expiration. By default is 0 second.
SUBSCRIBE for Registration Default is “No”. When set to “Yes” a SUBSCRIBE for Registration will be sent periodically. Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. Voice Mail UserID When configured, user can access messages by pressing “MSG” button. This userID is usually the VM portal access number.
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.
Check SIP User ID for Incoming INVITE Check the SIP User ID in Request URI. If they don’t match, the call will be rejected. 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”.
G723 Rate Encoding rate for G723 codec. By default, 6.3kbps rate is set. G726-32 Packing Mode Select “ITU” or “IETF” for G726-32 packing mode. iLBC Frame Size iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required. iLBC Payload Type Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. Jitter Buffer Type Jitter buffer type: Fixed or Adaptive. Default value is Adaptive. Jitter Buffer Length Jitter buffer length. Default value is 300ms.
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.
