Grandstream Networks, Inc. GXP1400/1405 Small-Medium Business IP Phone Grandstream Networks, Inc. GXP1400/1405 User Manual Firmware version 1.0.1.
TABLE OF CONTENTS GXP1400/1405 USER MANUAL WELCOME .................................................................................................................................................................3 INSTALLATION.........................................................................................................................................................4 EQUIPMENT PACKAGING ..............................................................................................................
Table 14: Device Configuration – Settings/Basic Settings ............................................................ 20 Table 15: Device Configuration – Settings /Advanced Settings ................................................... 22 Table 16: SIP Account Settings .................................................................................................... 27 GUI INTERFACE EXAMPLES GXP1400/1405 USER MANUAL http://www.grandstream.com/products/gxp_series/general/documents/gxp21xx_gui.zip 1. 2. 3.
Welcome GXP1400/1405 is a next generation small-to-medium business IP phone that features 2 lines with 1 SIP account, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE (GXP1405 only), and 3-way conference.
Installation EQUIPMENT PACKAGING Table 1: Equipment Packaging GXP1400/1405 Yes Yes Yes Yes (GXP1400 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 GXP1400/1405 are located on the bottom of the device.
Product Overview Table 3: GXP1400/1405 Feature Guide Features GXP1400/1405 LCD Display 128 x 40 pixel Number of Lines 2 Programmable Soft Keys 3 Extension Module N/A Table 4: GXP1400/1405 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.
Universal Switching Input: 100-240VAC 50-60 Hz Power Adaptor Output: +5VDC, 800mA, 4.0 W, UL certified Dimension Weight 186mm (W) x 210mm (L) x 81mm (D) Unit weight: 0.7KG Package weight: 1.1KG (GXP1400), 1.
Network and Provisioning Firmware Upgrades Advanced Server Features Security Grandstream Networks, Inc.
Using the GXP1400/1405 GETTING FAMILIAR WITH THE LCD GXP1400/1405 has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active screen). Table 7: LCD Display Definition Display Item Definitions DATE AND TIME Displays the current date and time. It can be synchronized with Internet time servers LOGO NAME Displays company logo name. This logo name can be customized via xml screen customization.
DND Icon: OFF - “Do Not Disturb” disabled ON - “Do Not Disturb” enabled Calls Forwarded Icon: INDICATES calls are forwarded.
Dual Lines with SIP Account GXP1400/1405 can support up to two lines “virtually” mapped to a SIP account. In off-hook state, select an idle line and the dial tone will be heard. To make a call, select the line you wish to use. The user can switch lines before dialing any number by pressing the LINE button. Completing Calls There are FIVE ways to complete a call: 1. DIAL: To make a phone call.
• • Dial the number to Page/Intercom Press “SEND” button to dial out NOTE: • Dial-tone and dialed number display occurs after the handset is off-hook, or handset button is pressed, or speaker button is pressed, or the line key is selected. 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.
ANSWERING PHONE CALLS Receiving Calls 1. Incoming single call: Phone rings with selected ring-tone. The corresponding LINE flashes in red. Answer call by taking Handset off hook or pressing SPEAKER or HEADSET or by pressing the corresponding account LINE button. 2. Incoming multiple calls: When another call comes in while having an active call, the phone will produce a Call Waiting tone (stutter tone). Answer the incoming call by pressing its corresponding LINE button.
2. Attended Transfer: Press “LINEx” button to make a call and automatically place the ACTIVE LINE on HOLD. Once the call is established, press “TRANSFER” key then the LINE button of the waiting line to transfer the call. Hang up the phone call after the call is transferred. 3. Auto-Attended Transfer: Users could enable Auto-Attended Transfer under Web GUI->Advanced Setting Page. During the first call, press “TRANSFER” hard button and it will bring up another line. The first call will be on hold.
If the users decide not to conference after establishing the second call, press EndCall softkey instead of ConfCall softkey/CONF button. It will end the second call and the screen will show the first call is on hold. During the conference, press EndCall softkey or hang up to end the conference 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.
Key Call Features *30 Block Caller ID (for all subsequent calls) Offhook and dial “*30”. *31 Send Caller ID (for all subsequent calls) Offhook and dial “*31”. *67 Block Caller ID (per call) Offhook, dial “*67” and then enter the number to dial out. *82 Send Caller ID (per call) Offhook, dial “*82” and then enter the number to dial out. *70 Disable Call Waiting (per Call) Offhook, dial “*70” and then enter the number to dial out.
Configuration Guide The GXP1400/1405 can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly, through embedded web-configuration menu. CONFIGURATION VIA KEYPAD To enter the MENU, press the round button. Navigate the menu by using the arrow keys: up/down and left/right. Press the OK softkey to confirm a menu selection. Press left arrow key can exit to the previous menu.
Dutch which are built in the phone. Users could select Automatic for local language based on IP location if available. Also, the phone will download secondary language if available. • Time Settings Users can set the date and time on the phone.
Call History Answered Calls Dialed Calls Missed Calls Transferred Calls Forwarded Calls Clear All Back MENU Phone Book New Entry Download Phonebook XML Delete All Entries Back Call History Items Delete All Entries New Entry First Name: Last Name Number: Acct: Confirm Add: Cancel & Return: Search Configuration Select Filter Filter Value Back LDAP Directory Call History Status Phone Book LDAP Directory Instant Message Direct IP Call Preference Config View Directory Download Directory Search Configu
CONFIGURATION VIA WEB BROWSER The GXP1400/1405 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 GXP1400/1405. 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.
XML Config File Password The password used for encrypting the XML configuration file using OpenSSL. This is required for the phone to decrypt the encrypted XML configuration file. HTTP/HTTPS User Name The user name for the HTTP/HTTPS server. HTTP/HTTPS Password The password for the HTTP/HTTPS server. It won’t display for security protection. Upgrade Via This field allows the user to choose the firmware upgrade method: TFTP, HTTP or HTTPS.
Connection Request Password Enter the connection request password. Authentication Method Select the authentication method among “No authentication”, “Basic” or Digest. Connection Request Port Enter the connection request port. Phonebook XML Download Selects the file download mode for the download server. Users can choose from TFTP/HTTP/No. Phonebook XML Server Path The URL/IP address of the phonebook download server.
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 keypad UI. Currently, the phone supports these languages: Arabic, German, English, Spanish, French, Hebrew, Croatian, Hungarian, Italian, Japanese, Korean, Dutch, Polish, Portuguese, Russian, Slovenian, Simplified Chinese and Traditional Chinese.
DNS Mode The default is set to A Record. If users wish to locate the server by DNS SRV, users 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. Primary IP This option applies only if “Use Configured IP” is selected, the phone will send DNS query to the Primary IP. Insert IP address here.
Support SIP Instance ID Selects whether or not SIP Instance ID is supported. 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.
Allow Auto Answer by Call-Info If the Call-Info header contains answer-after=0, the call be answered automatically (so called paging mode). 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 Hangup conference hangs up. Default setting is set to “No”.
No Key Entry Timeout Default is 4 seconds. 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.
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 GXP1400/1405 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.
