Grandstream Networks, Inc. GXP1450 SIP Enterprise Phone Grandstream Networks, Inc. GXP1450 User Manual Firmware 1.0.1.
TABLE OF CONTENTS GXP1450 USER MANUAL WELCOME .................................................................................................................................................................3 INSTALLATION.........................................................................................................................................................4 EQUIPMENT PACKAGING ....................................................................................................................
Table 10: Table 11: Table 12: Table 13: Table 14: Table 15: GXP1450 Call Features ................................................................................................ 17 Key Pad Configuration Menu ........................................................................................ 19 Device Configuration - Status ....................................................................................... 23 Device Configuration – Settings/Basic Settings ..........................................
Welcome GXP1450 is a next generation enterprise grade IP phone that features 2 lines with 2 SIP accounts, a 180x60 backlit graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports with integrated PoE and 3-way conference.
Installation EQUIPMENT PACKAGING Table 1: Equipment Packaging GXP1450 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 GXP1450 are located on the bottom of the device. Table 2: GXP1450 Connectors PC 10/100Mbps RJ-45 ports for PC (downlink) connection. LAN 10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af).
Product Overview Table 3: GXP1450 Feature Guide Features GXP1450 LCD Display 180x60 pixel Number of Lines 2 Programmable Soft Keys 3 Extension Module N/A Table 4: GXP1450 Key Features in a Glance Features Benefits Open Standards Compatible 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.
Dimension 186mm (W) x 210mm (L) x 81mm (D) Weight 0.8KG ° Temperature ° 32 –104 F/ 0 – 40 C Humidity 10% – 90% (non-condensing) Compliance FCC / CE / C-Tick Table 6: GXP1450 Technical Specifications Lines Protocol Support Display Feature Keys Device Management Audio Features Telephony Features Grandstream Networks, Inc. 2 lines with 2 independent SIP accounts Support SIP 2.
Network and Provisioning Firmware Upgrade Advanced Server Features Security Grandstream Networks, Inc. auto answer, early dial and speed dial Via keypad/LCD, Web browser, or secure (AES encrypted) central configuration file, manual or dynamic host configuration protocol (DHCP) network setup Support NAT traversal using IETF STUN and Symmetric RTP Support for IEEE 802.
Using the GXP1450 SIP Enterprise Phone GETTING FAMILIAR WITH THE LCD GXP1450 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 Item Definitions DATE AND TIME Displays the current date and time. Can be synchronized with Internet time servers. LOGO/NAME Displays company logo/name. This logo/name can be customized via xml screen customization.
DND (talking): ON when “Do Not Disturb” is activated in talking Forward All: INDICATES all incoming calls will be forwarded to the configured number Forward on Busy: INDICATES calls will be forwarded when phone is busy Forward on No Answer: INDICATES calls will be forwarded if the phone does not answer Forward All and No Answer: INDICATES calls will be forwarded if Forward All and Forward on No Answer are enabled Keypad Locked: ON when the keypad is locked Enter Keypad Password: Enter the keypad password to
FIGURE 1: GXP1450 KEYPAD LAYOUT Table 9: GXP1450 Keypad Buttons Key Button Key Button Definitions LINE KEYS 2 Line keys with LED, can be configured to different SIP profiles HOLD TRANSFER CONF Place ACTIVE call on hold Transfer an ACTIVE call to another number Press CONF button to connect Calling/Called party into conference Enter to retrieve voice mails or other messages Brings phonebook on screen Mute an active call Press HEADSET key to answer/hang up phone calls while using headset.
MENU/OK: • Enter Keypad Configuration “MENU” mode when phone is in IDLE mode • Use as ENTER/OK key when in Keypad Configuration.
1. DIAL: To make a phone call. • Take Handset/SPEAKER/Headset off-hook or press an available LINE key (activates speakerphone) or press the NEW CALL soft-key • The line will have a dial tone and the primary line (LINE1) LED is red If you wish, select another LINE key (alternative SIP account) • Enter the phone number • Press the SEND key or press the “DIAL” softkey 2. REDIAL: To redial the last dialed phone number. When redialing the phone will use the same SIP account as was used for the last call.
NOTE: Dialtone and dialed number display occurs after the phone is off-hook and the line key is selected. The phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call. Press the “SEND” or “#” button to override the 4 second delay. Making Calls using IP Addresses Direct IP calling allows two phones to talk to each other in an ad-hoc fashion without a SIP proxy.
NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IPIP call will also use STUN. Configure the “Use Random Port” to “NO” when completing Direct IP calls. ANSWERING PHONE CALLS Receiving Calls 1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER or by pressing the corresponding account LINE button. 2.
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 “Transfer Successful” is displayed in the screen. 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.
Press CONF button again or press the ConfCall softkey to join the new party in the existed conference 3. Hold Conference: During the conference, press HOLD button and the conference will be put on hold - To resume the conference, press the ReConf softkey - To split the conference and resume the call with each party, press the corresponding line key 4.
To enable shared call appearance, the user would need to register the shared line account on one of the accounts on the phone. In addition, they would need to navigate to “Settings”->”Basic Settings” on the web GUI and set the line to “Shared Line” with the corresponding account. If the user requires more shared call appearances, the user can configure multiple line buttons to be “shared line” buttons associated with the account.
CUSTOMIZED LCD SCREEN & XML GXP1450 Enterprise IP phone supports both simple and advanced XML applications: 1) XML Custom Screen and 2) XML Downloadable Phonebook. For more information on how to create a downloadable XML phonebook, creating a custom idle screen and/or reprogramming the softkeys on GXP1450, please visit our website at: http://www.grandstream.com/support Grandstream Networks, Inc. GXP1450 User Manual Firmware 1.0.1.
Configuration Guide The GXP1450 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 button to confirm a menu selection, delete an entry by pressing the MUTE/DEL button.
• 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 LDAP Directory Call History Status View Directory Download Directory Search Configuration Back Instant Message Phone Book LDAP Directory Instant Message Direct IP Call Preference Config Factory Functions Clear All Back Preference Do Not Disturb Ring Tone Ring Volume LCD Contrast LCD Brightness Download SCR XML Erase Custo
CONFIGURATION VIA WEB BROWSER The GXP1450 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, Google Chrome.
Table 12: Device Configuration - Status MAC Address The device ID, in HEXADECIMAL format. IP Address This field shows IP address of GXP1450. Product Model This field contains the product model information. Part Number This field contains the product part number. Software Version • Prog: 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: • • Line Keys x Identity MD5 Password This allows the user to configure the account mapped to each line key, as well as enabling Shared Call Appearance for the line. Options available for Key Mode are : 1. Line 2.
Disable Missed Call Backlight Default is “No”. By default, LCD backlight will light up whenever there is a missed call. 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.
Firmware Upgrade and Provisioning Allows the user to select the following options for firmware upgrade: • Always Check for New Firmware • Check New Firmware only when F/W pre/suffix changes • Always Skip the Firmware Check. Firmware upgrade may take up to 10 minutes depending on network environment. Do not interrupt the firmware upgrading process. Note: Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade.
TR-069 Username Enter username for TR-069. TR-069 Password Enter password for TR-069. Periodic Inform Enable Enable periodic inform. Default is “No”. Periodic Inform Interval When enabling periodic inform, set up the periodic inform interval. Connection Request Username Enter the connection request username. Connection Request Password Enter the connection request password. Authentication Method Select the authentication method among “No authentication”, “Basic” or Digest.
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.
Authenticate ID SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from SIP User ID. Authenticate Password SIP service subscriber’s account password for GXP1450 to register to (SIP) servers of ITSP. 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.
Check Domain Certificate Enable to check the domain certificate. Default is “No”. Remove OBP from Route The SIP Extension notifies the SIP server that it is behind a NAT/firewall. Validate Incoming Messages This configuration selects whether or not the incoming messages should be validated. 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.
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. After the timeout, the phone will send out the dialed number. 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.
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 GXP1450 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.
FCC Warning This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment. FCC 15.
