Grandstream Networks, Inc.
GXP1100/GXP1105 User Manual Index GUI INTERFACE EXAMPLES ...................................................................... 5 GNU GPL INFORMATION ............................................................................ 6 CHANGE LOG ............................................................................................... 7 FIRMWARE VERSION 1.0.5.15 .......................................................................................................... 7 FIRMWARE VERSION 1.0.4.23 .............
CALL FEATURES .............................................................................................................................. 21 CONFIGURATION GUIDE .......................................................................... 23 CONFIGURATION VIA IVR MENU .................................................................................................... 23 CONFIGURATION VIA WEB BROWSER .......................................................................................... 24 DEFINITIONS ..
Table of Tables GXP1100/GXP1105 User Manual Table 1: GXP1100/GXP1105 TECHNICAL SPECIFICATIONS ................................................................... 9 Table 2: GXP1100/GXP1105 EQUIPMENT PACKAGING ..........................................................................11 Table 3: GXP1100/GXP1105 CONNECTORS............................................................................................11 Table 4: GXP1100/GXP1105 KEYPAD DEFINITIONS ....................................................
GUI INTERFACE EXAMPLES 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 FIRMWARE VERSION 1.0.5.
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.grandstream.com/support/faq/gnu_gpl. FIRMWARE VERSION 1.0.5.
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] • Added pin-out information. [CONNECTING YOUR PHONE] • Updated Auto Attended Transfer information.
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).
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 • 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 Table
Simplified Chinese, traditional Chinese, Korean, Japanese, and etc supported in web configuration interface Upgrade and Provisioning Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or 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.
INSTALLATION EQUIPMENT PACKAGING Table 2: GXP1100/GXP1105 EQUIPMENT PACKAGING Main Case Yes 1 Handset Yes 1 Phone Cord Yes 1 Power Adaptor Yes 1 Ethernet Cable Yes 1 Phone Stand Yes 1 Quick Start Guide Yes 1 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 FIRMWARE VERSION 1.
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.
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.
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. • 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.
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.
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; • By default, # 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->Account->Call Settings.
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. "aaa.bbb.ccc" is from the local IP address regardless of subnet mask. The number #xx or #x are also valid. The leading 0 is not required (but it's OK). For example: • 192.168.0.
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.
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. • To use Multi-Purpose Key to establish 3-way conference call, go to GXP1100/GXP1105 Web GUI->Settings->Programmable Keys, configure the 3-way conference as the Multi Purpose Key mode.
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 • Users could simply hang up the call to terminate the conference call.
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). An IVR will prompt the user through the process of message retrieval.
• Dial *71 and then enter the number to dial out. Unconditional Call Forward. To set up unconditional call forward: *72 • Pick up the handset; • Dial *72. A dial tone will be heard; • Enter the forwarding number; • 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: *73 • Pick up the handset; • Dial *73. A short tone will be heard; • Wait for the call to hang up.
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. Table 6: GXP1100/GXP1105 IVR MENU Menu Main Menu Voice Prompt Options "Enter a Menu Option" Press * for the next menu option. Press # to return to the main menu.
• G-723 • G-729 10 "MAC Address" Announces the MAC address of the unit. 13 "Firmware Server IP Address" Announces current Firmware Server IP address. Enter 12 digit new IP address. 14 15 "Configuration Server IP Announces current Config Server Path IP address. Address" Enter 12 digit new IP address. "Upgrade Protocol" Upgrade Protocol for firmware and configuration update. Enter 9 to toggle between HTTP, TFTP and HTTPS. 16 "Firmware Version" Firmware version information.
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. IP Setting DHCP, Static IP or PPPoE.
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. SIP Server Secondary SIP Server The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (ITSP). The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails.
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. 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.
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". Remove OBP from route Support SIP Instance ID Configures to remove outbound proxy from route. This is used for the SIP 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".
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).
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. DTMF Payload Type Configures the payload type for DTMF using RFC2833.
Account x -> Call Settings Early Dial Dial Plan Prefix 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 "No". 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.
• 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. Ring Timeout Send Anonymous Anonymous Call Rejection Allow Auto Answer by Call-Info Defines the timeout (in seconds) for the rings on no answer. The default setting is 60 seconds. If set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous, essentially blocking the Caller ID to be displayed. If set to "Yes", anonymous calls will be rejected. The default setting is "No".
Use NAT IP 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 STUN Server results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN. Settings -> Call Features Configures a User ID/extension to dial automatically when the phone is Off-hook Auto Dial off hook.
Max Unsaved Log Enable FLASH Key as CONF default value is 200 entries. If set to "Yes", the FLASH key can be used to establish 3-way conference. The default setting is "No". Settings -> Ring Tone Configures ring or tone frequencies based on parameters from local Call Progresses Tones: System Ring Tone Dial Tone Message Waiting Ring Back Tone Call-Waiting Tone Busy Tone Reorder Tone telecom. The default value is North American standard.
• 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". DHCP Host name Specifies the name of the client. This field is optional but may be required by (Option 12) some Internet Service Providers.
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. 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 Upload 802.
Firmware Upgrade and Provisioning 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. 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. HTTP/HTTPS User Name The user name for the HTTP/HTTPS server.
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.
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. Password Download Device Configuration Click to download the device configuration file in .txt format.
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. http://ip_address/cgi-bin/api-make_call?phonenumber=1234&account=0&password=admin In the above link, replace the fields with • ip_address: Phone's IP Address.
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.
The 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 download is done you will see the phone restarts again. Please do NOT disrupt or power down the unit. If a firmware upgrade fails for any reason (e.g.
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".
RESTORE FACTORY DEFAULT SETTINGS Warning: 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. Please follow the instructions below to reset the phone: Pick up the handset, press *** to access the IVR menu. Enter 99 for factory reset.
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.
