TABLE OF CONTENTS GXP USER MANUAL WELCOME.................................................................................................................................................... 4 INSTALLATION............................................................................................................................................ 5 EQUIPMENT PACKAGING ...............................................................................................................................
Table 8: LCD Buttons.................................................................................................................... 12 Table 9: LCD Icons ....................................................................................................................... 12 Table 10: GXP Keypad Buttons .................................................................................................... 14 Table 11: GXP Call Features .....................................................................
Welcome Your Grandstream GXP Series IP phone features a new sophisticated design and is very easy to use. The GXP combines advanced feature functionality with the latest technology to offer excellent audio quality, ease of use, expandability, and broad interoperability with 3rd party SIP platforms. It is ideal for the enterprise customer. The GXP Series supports a broad range of codecs, security protection, PoE (except on GXP-280), dual 10/100mbps Ethernet ports and are very easy to manage.
Installation EQUIPMENT PACKAGING Table 1: Equipment Packaging Main Case Handset Phone Cord Power Adaptor Ethernet Cable High Phone Stand Low Phone Stand Wall Mount Spacers (2) GXP-280 Yes Yes Yes Yes Yes No Yes No GXP-1200 Yes Yes Yes Yes Yes Yes No Yes GXP-2000 Yes Yes Yes Yes Yes No No No GXP-2010 Yes Yes Yes Yes Yes Yes Yes Yes GXP-2020 Yes Yes Yes Yes Yes Yes Yes Yes CONNECTING YOUR PHONE The connectors of the GXP1200/2010/2020 are located on the bottom of the device while they are located on the
1) 2) 3) 4) One GXP Extension unit One PS2 cable One connection plate One Universal Power Adaptor FIGURE 1: CONNECTING THE GXP–2000 AND THE GXP–EXTENSION GXP–2000 w/GXP–Extension GXP Extension Connecting the GXP–2000 w/GXP–Extension Reverse side of connection w/connection plate Connect the first GXP –EXT to the GXP–2000 using the PS2 cable found in the GXP Extension package. The first GXP–Ext draws power directly from the phone.
Figure 2: GXP–2000 Internal Headset Wiring Schema NOTE: For GXP-2000 HW REV. 0.3 and 0.4, a 3.5mm to 2.5mm plug converter is required to use a 2.5mm headset. The converter can be purchased at any electronics store. SAFETY COMPLIANCES The GXP phone complies with FCC/CE and various safety standards. The GXP power adaptor is compliant with the UL standard. Only use the universal power adaptor provided with the GXP package.
Product Overview Table 3: GXP Product Models Model Overview Picture GXP280 is an entry-level SIP phone. It features: y y Single line Three soft keys GXP-280 GXP1200 is an entry-level SIP phone. It features: y y Two lines Three soft keys GXP-1200 GXP2000 is a mainstream SIP phone. It features: GXP-2000 y y Four lines Seven programmable hard keys GXP2010 is a key system SIP phone.
Table 4: GXP Comparison Guide Features GXP-280 GXP-1200 GXP-2000 GXP-2010 GXP-2020 LCD Display 128x32 pixel 128x32 pixel 130x64 pixel 240x120 pixel 320x160 pixel Number of Lines 1 2 4 4 6 Programmable Hard Keys No No 7 18 7 Soft Keys 3 3 No 3 4 Extension Module No No Yes, up to 2 Expansion Modules, 56 nodes each Yes, up to 2 Expansion Modules, 56 nodes each Yes, up to 2 Expansion Modules, 56 nodes each Table 5: GXP Key Features in a Glance Features Benefits Open Standar
Call Appearance LED GXP-280 GXP-1200 GXP-2000 GXP-2010 GXP-2020 2.5mm and RJ22 RJ22 2.5mm 2.5mm and RJ22 2.5mm and RJ22 Dual color (green/red) GXP-280 GXP-1200 GXP-2000 GXP-2010 GXP-2020 No 3 11 22 13 Built-in auto-sensing: Cisco and IEEE 802.
Device Management Audio Features Telephony Features Network and Provisioning Firmware Upgrades Advanced Server Features Security SPEAKERPHONE Yes Yes Yes Yes Yes SEND Yes Yes Yes Yes Yes TRANSFER Yes Yes Yes Yes Yes CONF Yes Yes Yes Yes Yes MUTE Yes Yes Yes Yes Yes DND Yes Yes Yes Yes Yes HEADSET Yes Yes Yes Yes Yes INTERCOM No No No Yes Yes PHONEBOOK No No No Yes Yes MSG Yes Yes Yes Yes Yes MENU Yes Yes Yes Yes Yes NAVIGATION (4) Yes (3) Yes Yes Yes Yes NAT-friendly remote software upgrade (via TFTP/H
Using the GXP SIP Enterprise Phone GETTING FAMILIAR WITH THE LCD GXP-2xxx has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active screen). Table 8: LCD Buttons Key Button Key Button Definitions LINE SELECTORS Selects the phone line printed on its right-hand side. SIP PHONE LINES Displays the available phone lines. Choose a phone line by pressing the corresponding line selector on the left-hand side.
Speaker Phone Status Icon: FLASH when phone rings or a call is pending OFF when the speakerphone is off ON when the speakerphone is on DND Icon: ON when the “do not disturb” is activated Activate by pressing MUTE/DEL button once Calls Forwarded Icon: INDICATES calls are forwarded Follow ‘call forwarding’ procedures Handset, Speakerphone and Ring Volume Icon: Each icon appears next to the volume icon To adjust volume, use the up/down button AM PM Real–time Clock: Synchronized to Internet time server Time z
TABLE 10: GXP KEYPAD BUTTONS Key Button Key Button Definitions LINE BUTTONS Line keys with LED, can be configured to different SIP profiles TRANSFER TRANSFER key: Transfer an ACTIVE call to another number CONF Press CONF button to connect Calling/Called party into conference MUTE Mute an active call; or Delete a key entry Also used to ‘REJECT’ incoming call.
MAKING PHONE CALLS Handset, Speakerphone and Headset Mode Handset can be toggled between Speaker and Headset. To switch between Handset and Speaker/Headset, press the Hook Flash in the handset cradle or press the SPEAKER button. Multiple SIP Accounts and Lines GXP can support up to six independent SIP accounts depending on the product model. Each account is capable of independent SIP server, user and NAT settings. Each of the line buttons is “virtually” mapped to an individual SIP account.
3. CALL RETURN: To call the last phone number that called your phone. When returning a call, the phone will use the same SIP account as the call was made to. Thus, when returning a call made to the third SIP account, the phone will use the third SIP account return the call. i. Hand-free option 1. Press the CALL RETURN soft-key ii. Hand-set option 1. Take the Handset off-hook 2. Press the CALL RETURN soft-key To call a phone number in the phone’s history 4.
Note: The multi-functional buttons will function as LINE keys when all LINEs are busy. The LED will flash in red to indicate an incoming call. Press the button to pick up the call. If any one of the Multi Purpose Keys is associated with a call, the button’s speed dial/BLF function will not work. 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.
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. Incoming multiple calls: When another call comes in while having an active call, the phone will produce a Call Waiting tone (stutter tone). Next available lines will flash red (as described in section 4.3.2).
2. Attended (or Supervised) Transfer: Press “LINEx” button to make a call and automatically place the ACTIVE LINE on HOLD. Once the call is established, press “TRANSFER (or TRNF)” key to transfer the call and hang up. NOTE: To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains. Blind transfer will usually use the primary account SIP profile. 5-Way Conferencing GXP can host conference calls and supports up to 5-way conference calling. 1.
CALL FEATURES The GXP supports traditional and advanced telephony features including caller ID, caller ID w/name, call forward/transfer/park/hold as well as intercom/paging and BLF.
Configuration Guide The GXP 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.
Configure Press Menu button to display the configuration selections: • Network. To enable/disable DHCP. To setup IP-address, Net mask and Gateway address • SIP To change SIP-server settings for primary account. • Upgrade In this menu setting regarding the firmware server and Config server can be changed. It also enables the user to make the phone attempt to download new firmware. • Factory Reset Key in the physical/MAC address on back of the phone. Press Menu button to reset FACTORY DEFAULT setting.
FIGURE 3: KEYPAD GUI FLOW Call History MENU Any of previous menus Answered Calls Dialed Calls Missed Calls Back Back Clear All New Entry Phone Book New Entry Download Phonebook XML Back Name: Number: Acct: Confirm Add: Cancel & Return: LDAP Directory Call History Status View Directory Download Directory Search Configuration Back Search Configuration Select Filter Filter Value Back Instant Message Do Not Disturb Phone Book LDAP Directory Instant Message Direct IP Call Preference Config Fac
CONFIGURATION VIA WEB BROWSER The GXP 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 or Mozilla Firefox.
Table 13: Device Configuration - Status MAC Address The device ID, in HEXADECIMAL format. IP Address This field shows IP address of GXP 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 software (firmware) release number, always used to identify the software (firmware) system of the phone.
LCD Backlight Always On Turn on LC backlight at all times. Default is No. This option applies to GXP1200/GXP-2000 only. Time Display Format LCD time display in 12 hour or 24 hour format Date Display Format Choose one of the following formats: • Year-Month-Day • Month-Day-Year • Day-Month-Year This option applies to GXP280/GXP-1200/GXP-2000 only. Display Clock instead of Choose to display clock or date on LCD. This option applies to GXP-280/GXP1200/GXP-2000 only.
Disable Missed Call Backlight Default is No. By default, LCD backlight will lit whenever there is a missed call. Not for GXP280. HEADSET Key Mode Set Default mode or choose Toggle Headset/Speaker. Not for GXP2000 Advanced User configuration includes not only the end user configuration, but also advanced configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration. Table 15: Advanced Settings Admin Password Administrator password.
Voice Frames per TX This field contains the number of voice frames to be transmitted in a single Ethernet packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500 byte (or 120kbps)). When setting this value, be aware of the requested packet time (ptime, used in SDP message) is a result of configuring this parameter.
STUN Server IP address or Domain name of the STUN server. STUN resolution result will display in the STATUS page of the Web UI. Firmware Upgrade and Provisioning Default method is HTTP. Firmware upgrade may take up to 10 minutes depending on network environment. Do not interrupt the firmware upgrading process. Via TFTP Server This is the IP address of the configured TFTP server.
Idle Screen XML Enable XML Idle Screen download via TFTP or HTTP. Define XML server path. Download XML Application Enter server path for XML application. This option applies to GXP-2020 only. Offhook Auto Dial To configure a User ID/extension to dial autpomatically when the phone is taken offhook. DTMF Payload Type This parameter sets the payload type for DTMF using RFC2833. Default is 101. Onhook Threshhold It determines the time handset has to be down to be recognized it’s onhook. Default is 800ms.
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.
Headset RX gain (dB) Increases the selected headset’s (2.5mm or RJ22) RX gain by + or – 6dB. Default is 0dB Display Language Allows user to choose preferred display language in web UI and key pad UI. The user can only load one secondary language GXP has up to six line appearances, each with an independent SIP account. Each SIP account requires its own configuration page. Their configurations are identical. Table 16: SIP Account Settings Account Active This field indicates whether the account is active.
Local SIP Port This parameter defines the local SIP port used to listen and transmit. The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and Account 4 respectively. SIP Registration Failure Retry registration if the process failed. Default is 20 seconds. Retry Wait Time SIP T1 Timeout RFC 3261 SIP T1 timer. Default is 1 second. SIP T2 Interval RFC 3261 SIP T2 timer. Default is 0.5 seconds. SIP Transport Choose SIP Transport between UDP and TCP. Default is UDP.
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). Turn off speaker on remote disconnect When BYE is received, the phone will turn off its speaker automatically. 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. Refer-To Use Target Contact Default is NO.
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/1.1.6.
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 GXP 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. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
