Grandstream Networks, Inc. GXP21xx SIP Enterprise Phones Grandstream Networks, Inc. GXP21xx User Manual Firmware version: 1.0.1.
TABLE OF CONTENTS GXP21XX USER MANUAL WELCOME .................................................................................................................................................................3 INSTALLATION.........................................................................................................................................................4 EQUIPMENT PACKAGING ....................................................................................................................
Table 6: GXP21xx Hardware Specifications ................................................................................... 8 Table 7: GXP21xx Technical Specifications ................................................................................... 9 Table 8: LCD Display Definition .................................................................................................... 11 Table 9: LCD Icons ......................................................................................................
Welcome Your Grandstream GXP21xx Enterprise IP phone is feature-enriched, sophisticated, yet simple to use. The GXP21xx delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms.
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) GXP2120 Yes Yes Yes Yes Yes Yes Yes Yes GXP2110 Yes Yes Yes Yes Yes Yes Yes Yes GXP2100 Yes Yes Yes Yes Yes Yes No Wall Mount stand CONNECTING YOUR PHONE The connectors of the GXP21xx are located on the bottom of the device.
Figure 1: Connecting the GXP2120/2110 and the GXP Extension GXP2120 with GXP Extension GXP Extension Connecting the GXP2120/2110 to GXP Extension Reverse side of connection with connection plate Reverse side of connection w/connection plate GXP2120/2110 has a special port on the back. Connect the first GXP EXT to the GXP2120/2110 using the connection cable found in the GXP Extension package. The first GXP EXT draws power directly from the phone.
SAFETY COMPLIANCES The GXP21xx complies with FCC/CE and various safety standards. The GXP21xx power adaptor is compliant with the UL standard. Only use the universal power adaptor provided with the GXP21xx package. The manufacturer’s warranty does not cover damages to the phone caused by unsupported power adaptors. WARRANTY If you purchased your GXP21xx from a reseller, please contact the company where you purchased your phone for replacement, repair or refund.
Product Overview Table 3: GXP21xx Product Models Model Overview Picture GXP2120 is an executive SIP phone. It features: GXP2120 Six lines Seven programmable hard keys Four XML programmable soft keys GXP2110 is an executive SIP phone. It features: GXP2110 Four lines Eighteen programmable hard keys Three XML programmable soft keys GXP2100 is an executive SIP phone.
Table 5: GXP21xx 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.
Humidity 10% – 90% (non-condensing) Compliance FCC / CE / C-Tick Table 7: GXP21xx Technical Specifications Lines Protocol Support Display Feature Keys Device Management Audio Features Grandstream Networks, Inc. Multiple direct lines with independent SIP accounts, programmable speed dial keys, XML programmable soft-keys Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP, SRTP by SDES, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, SIMPLE/PRESENCE protocols, TR-069, 802.
Telephony Features Network and Provisioning Firmware Upgrades Advanced Server Features Security Grandstream Networks, Inc.
Using the GXP21xx SIP Enterprise Phone GETTING FAMILIAR WITH THE LCD GXP21xx 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 Display Definition Item Definitions DATE AND TIME Displays the current date and time. Can be synchronized with Internet time servers. LOGO Displays company logo name. This logo name can be customized via xml screen customization.
Call Parking: FOR GXP2120/2110 ONLY. Please refer to GXE5024/5028 Online User Manual for more information. • CallPark When a GXP2120/2110 dials out, the Call Park softkey will display on screen. To park the call, press the “Call Park” button. • PickUp When another GXP2120/2110 goes off-hook, the Call Pickup softkey will display on screen. To pick up the parked call, press the “Call Pickup” button. SPECIAL SOFTKEYS (Only When Integrated with GXE5024/5028) Call Queue: FOR GXP2120/2110 ONLY.
Enter Keypad Unlock Password Voice Mail: ON when there are new voice messages Network Status: Network is down Missed Call Icon: Indicates missed call(s) Save Call Record: Indicates phone system writing the call records into the flash. It might take 10 to 20 seconds to finish the process Waiting For Response: Please wait for the phone system to response before the keypad entry.
Enable/Disable handset mode ; or used as SEND/REDIAL Navigation keys “Up” “Down” “Left” and “Right” • Press to navigate in menu options • During the call, press “Up” “Down” to adjust volume • When the phone is idle, press “Up” to view missed call; press “Down” to view phonebook 0 – 9, *, # Standard phone keypad; press # key to send call; press * key to for IVR functions Multi Purpose Keys 18 MPKs in GXP2110 and 7 MPKs in GXP2120 used for BLF, Speed dial and etc MAKING PHONE CALLS Handset, Speakerphone a
Completing Calls There are six ways to complete a call: 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” soft-key. 2. REDIAL: To redial the last dialed phone number.
• • • • Select the LINE key associated with account Press OK key to display LCD: LINEx: PAGE. Dial the phone number you want to Page/Intercom Press SEND key. 6. VIA CALL RETURN: On the GXP21xx, the Multiple Purpose Key (programmable hard key) has to be configured as Call Return under Web GUI->Basic Settings configuration. No user name and user ID has to be set on the Multiple Purpose Key for Call Return. After pressing the Call Return key, the last answered number will be dialed out.
To make a quick IP call, please see next section. For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input the following: 192*168*1*60#5062 - The “ * ” key represent the dot“.” ; The “#” key represent colon “:”. Press OK to dial out. Quick IP Call Mode The GXP21xx also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phone’s IP-number.
PHONE FUNCTIONS DURING A PHONE CALL Call Waiting/ Call Hold 1. Hold: Place a call on hold by pressing the “HOLD” button. 2. Resume: Resume call by pressing the corresponding blinking LINE. 3. Multiple Calls: Automatically place ACTIVE call on “HOLD” by selecting another available LINE to place or receive another call. Call Waiting tone (stutter tone) audible when line is in use. Mute 1. Press the MUTE button to enable/disable muting the microphone. 2.
If after pressing the “CONF” button, a user decides not to conference anyone, press CONF again or the original LINE button This will resume two-way conversation 3. End Conference: Press HOLD to end the conference call and put all parties on hold; To speak with an individual party, select the corresponding blinking LINE. GXP21xx also supports Easy Conference mode. In Easy Conference mode, users can initiate conference by calling another number when the current line is in talking or conference.
NOTE: • Each line has a separate voicemail account. Each account requires a voicemail portal number to be configured in the “Voicemail User ID” field. • To check which line account has a message 1) press the message button (this always checks the primary account), 2) check each line for stutter tone or 3) check missed calls using the menu. Busy Lamp Field The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account.
*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. *71 Enable Call Waiting (per Call) Offhook, dial “*71” and then enter the number to dial out. *72 Unconditional Call Forward Offhook, dial “*72”.
Configuration Guide The GXP21xx 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. The phone automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if left idle for 20 seconds.
• 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 Phone Book LDAP Directory Instant Message Direct IP Call Preference View Directory Download Directory Search Configuration Back Instant Message Clear All Back Preference Do Not Disturb Ring Tone Ring Volume LCD Contrast LCD Brightness Download SCR XML Erase Custom SCR Display Language Time
CONFIGURATION VIA WEB BROWSER The GXP21xx 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, Google Chrome.
Table 13: Device Configuration - Status MAC Address The device ID, in HEXADECIMAL format. IP Address This field shows IP address of GXP21xx. 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.
Self-Defined Time Zone This parameter allows the users to define their own time zone. The syntax is: std offset dst [offset], start [/time], end [/time] Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0 MTZ+6MDT+5, This indicates a time zone with 6 hours offset with 1 hour ahead which is U.S central time. If it is positive (+) if the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is east. M4.1.0,M11.1.0 The 1st number indicates Month: 1,2,3..
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.
Do not escape “#” as %23 in SIP URI Default is “No”. By default, # will be replaced as %23 in SIP URI. 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.
SIP User ID User account information provided by VoIP service provider (ITSP); either an actual phone number or formatted like one. 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 GXP21xx 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.
Use Actual Ephemeral Port in Contact with TCP/TLS Enable to use actual ephemeral port in contact with TCP/TLS. Default is “No”. 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.
Early Dial Default is “No”. Use only if proxy supports 484 responses. Dial Plan Prefix Sets the prefix added to each dialed number. 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 GXP21xx 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.