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.
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 Updated Web GUI interface examples with new screenshots for 1.0.5.15. [GUI INTERFACE EXAMPLES] ...
IP phone with attractive cost. Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
PRODUCT OVERVIEW FEATURE HIGHTLIGHTS Single SIP Account, up to 2 calls, 4 programmable keys HD handset with support for wideband audio Single 10/100Mbps network port, integrated PoE (GXP1105 only) 7 dedicated function keys for Hold, Flash/Call Waiting, Transfer, Message, Mute, Volume, Send/Redial ...
Page 10
Simplified Chinese, traditional Chinese, Korean, Japanese, and etc supported in web configuration interface Upgrade Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or Provisioning AES encrypted XML configuration file Universal power adapter: Power and Green Input: 100-240VAC 50-60Hz; Output: 5VDC, 800mA Energy Efficiency Integrated Power-over-Ethernet (802.3af, GXP1105 only) Typical power consumption under 1W (power adapter) or under 1.5W (PoE)
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 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...
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.
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.
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. ...
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.
Note: 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; If digits have been entered after handset is off hook, the SEND key will works as SEND instead of REDIAL;...
phones are under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended. 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<255. Press # or SEND and a direct IP call to aaa.bbb.ccc.XXX will be completed.
Multiple calls. Automatically place active call on hold or switch between two calls by pressing the FLASH key. Call waiting tone (stutter tone) will be audible when the line is in use. Note: If users hang up the current call while there is a call on hold in the other line, there will be an audible ring tone indicating a call is on hold while your handset is put on hook.
Enter the number and press SEND key or # to make a second call; Press TRAN key again. The call will be transferred. Note: To 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.
Page 20
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; Press FLASH key to toggle between the 2 lines; Users could re-establish conference call by pressing the Multi Purpose Key again. 3.
The option "Disable Conference" has to be set to "No" to establish conference on GXP110x. VOICE MESSAGES (MESSAGE WAITING INDICATOR) A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Dial into the voicemail box to retrieve the message by entering the voice mail number of the server or pressing the MSG key (Voice Mail User ID has to be properly configured as the voice mail number under Web GUI->Account->General Settings page).
Page 22
Enable Call Waiting (per Call) Off hook the phone; Dial *71 and then enter the number to dial out. Unconditional Call Forward. To set up unconditional call forward: Pick up the handset; Dial *72. A dial tone will be heard; ...
Page 23
FIRMWARE VERSION 1.0.5.23 GXP1100/GXP1105 USER MANUAL Page 23 of 49...
CONFIGURATION GUIDE The GXP1100/GXP1105 can be configured via two ways: IVR Menu using the phone's keypad; 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.
G-723 G-729 "MAC Address" Announces the MAC address of the unit. "Firmware Server IP Address" Announces current Firmware Server IP address. Enter 12 digit new IP address. "Configuration Server Announces current Config Server Path IP address. Address" Enter 12 digit new IP address. "Upgrade Protocol"...
4. Open a Web browser on your computer; 5. Enter the phone’s IP address in the address bar of the browser; 6. Enter the administrator’s login and password to access the Web Configuration Menu. Note: The computer has to be connected to the same sub-network as the phone. This can be easily done by connecting the computer to the same hub or switch as the phone connected to.
STATUS PAGE DEFINITIONS Status -> 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 -> 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.
ACCOUNT PAGE DEFINITIONS Account x -> 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 SIP Server VoIP service provider (ITSP).
Page 29
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&SDP messages.
Page 30
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 from 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.
Page 31
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"...
Page 32
Authenticate Incoming If set to "Yes", the phone will challenge the incoming INVITE for authentication INVITE with SIP 401 Unauthorized response. The default setting is "No". Account x -> Audio Settings Specifies the mechanism to transmit DTMF digits. There are 3 supported Send DTMF 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.
Page 33
Account x -> Call Settings Selects whether or not to enable early dial. If it's set to "Yes", the SIP proxy Early Dial must support 484 response. The default setting is "No". Dial Plan Prefix Sets the prefix added to each dialed number. A dial plan establishes the expected number and pattern of digits for a telephone number.
Page 34
1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length; 011[2-9]x - allows international calls starting with 011; [3469]11 - allows dialing special and emergency numbers 311, 411, 611 and 911. Note: In some cases where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature.
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 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.
Page 36
The NAT IP address used in SIP/SDP messages. This field is blank at Use NAT IP 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 STUN Server results are displayed in the STATUS page of the Web GUI.
Page 37
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 -> Ring Tone Configures ring or tone frequencies based on parameters from local telecom.
3-way Conference To establish 3-way conference. NETWORK PAGE DEFINITIONS Network -> Basic Settings Internet Protocol Selects Prefer IPv4 or Prefer IPv6. Allows users to configure the appropriate network settings on the phone to IPv4 Address Type obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE". By default, it is set to "DHCP".
Network -> Advanced Settings Allows the user to enable/disable 802.1X mode on the phone. The default 802.1X mode value is disabled. To enable 802.1X mode, this field should be set to EAP-MD5. 802.1X Identity Enter the Identity for the 802.1X mode. MD5 Password Enter the MD5 Password for the 802.1X mode.
Page 40
Specifies how firmware upgrading and provisioning request to be sent: Always Firmware Upgrade and Check for New Firmware, Check New Firmware only when F/W pre/suffix Provisioning changes, Always Skip the Firmware Check. XML Config File The password for encrypting the XML configuration file using OpenSSL. This Password is required for the phone to decrypt the encrypted XML configuration file.
Page 41
Maintenance -> Syslog Syslog Server The URL or IP address of the syslog server for the phone to send 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: ...
CPE SSL Private Key The Cert Key for the phone to connect to the ACS via SSL. Maintenance -> Security 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.
After clicking on the icon, a new dialing window will show as the figure below. Enter number and click on "Dial", the phone will go off hook and dial out the number from account 1. 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.
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 1.0.5.23 GXP1100/GXP1105 USER MANUAL Page 44 of 49...
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.
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.
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.
Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
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 1.0.5.23 GXP1100/GXP1105 USER MANUAL...