Grandstream GXP1450 Manual de Usario

Grandstream telefono GXP1450

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Grandstream Networks, Inc. GXP1450 User Manual Page 1 of 1
Firmware 1.0.1.66 Last Updated: 05/2011
Grandstream Networks, Inc.
GXP1450 SIP Enterprise Phone
Grandstream Networks, Inc. GXP1450 User Manual Page 1 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
TABLE OF CONTENTS
GXP1450 USER MANUAL
WELCOME .......................................................................................................................................................... 3
INSTALLATION ................................................................................................................................................. 4
EQUIPMENT PACKAGING ...................................................................................................................................... 4
CONNECTING YOUR PHONE .................................................................................................................................. 4
SAFETY COMPLIANCES ......................................................................................................................................... 4
WARRANTY ......................................................................................................................................................... 4
PRODUCT OVERVIEW ..................................................................................................................................... 5
USING THE GXP1450 SIP ENTERPRISE PHONE .......................................................................................... 8
GETTING FAMILIAR WITH THE LCD ...................................................................................................................... 8
MAKING PHONE CALLS ...................................................................................................................................... 10
ANSWERING PHONE CALLS ................................................................................................................................ 13
PHONE FUNCTIONS DURING A PHONE CALL ........................................................................................................ 13
CALL FEATURES ................................................................................................................................................ 15
CUSTOMIZED LCD SCREEN & XML ................................................................................................................... 16
CONFIGURATION GUIDE ................................................................................................................................ 17
CONFIGURATION VIA KEYPAD............................................................................................................................ 17
CONFIGURATION VIA WEB BROWSER ................................................................................................................ 21
SAVING THE CONFIGURATION CHANGES ............................................................................................................. 35
REBOOTING THE PHONE REMOTELY.................................................................................................................... 35
SOFTWARE UPGRADE & CUSTOMIZATION ............................................................................................. 36
FIRMWARE UPGRADE THROUGH TFTP/HTTP ..................................................................................................... 36
CONFIGURATION FILE DOWNLOAD ..................................................................................................................... 37
RESTORE FACTORY DEFAULT SETTING .................................................................................................. 38
TABLE OF FIGURES
GXP1450 USER MANUAL
Figure 1: GXP1450 Keypad Layout……………………………………………………………………. 10
Figure 2: Keypad GUI Flow ........................................................................................................ 18
TABLE OF TABLES
GXP1450 USER MANUAL
Table 1: Equipment Packaging ................................................................................................... 4
Table 2: GXP1450 Connectors ................................................................................................... 4
Table 3: GXP1450 Feature Guide ............................................................................................... 5
Table 4: GXP1450 Key Features in a Glance .............................................................................. 5
Table 5: GXP1450 Hardware Specifications ................................................................................ 5
Table 6: GXP1450 Technical Specifications ................................................................................ 6
Table 7: LCD Buttons ................................................................................................................. 8
Table 8: LCD Icons ..................................................................................................................... 8
Table 9: GXP1450 Keypad Buttons............................................................................................. 9
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Table 10: GXP1450 Call Features ............................................................................................ 15
Table 11: Key Pad Configuration Menu ..................................................................................... 17
Table 12: Device Configuration - Status .................................................................................... 22
Table 13: Device Configuration – Settings/Basic Settings.......................................................... 22
Table 14: Device Configuration – Settings /Advanced Settings .................................................. 24
Table 15: SIP Account Settings................................................................................................. 29
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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.
The GXP1450 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. It is a perfect choice for enterprise users looking for a high quality, feature rich IP
phone with affordable cost.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation
of this product in any way other than as detailed by this User Manual, could void your manufacturer
warranty.
Warning: Please do not use a different power adaptor with the GXP1450 as it may cause damage to the
products and void the manufacturer warranty.
• This document is subject to change without notice.
• Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission is not permitted.
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Installation
EQUIPMENT PACKAGING
Table 1: Equipment Packaging
GXP1450
Main Case
Yes
Handset
Yes
Phone Cord
Yes
Power Adaptor
Yes
Ethernet Cable
Yes
Base Stand
Yes
Quick Start Guide
Yes
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).
Power Jack
5V DC power port; UL Certified
Headset Jack
RJ9
Handset Jack
RJ9
SAFETY COMPLIANCES
The GXP1450 complies with FCC/CE and various safety standards. The GXP1450 power adaptor is
compliant with the UL standard. Only use the universal power adaptor provided with the GXP1450 package.
The manufacturer’s warranty does not cover damages to the phone caused by unsupported power adaptors.
WARRANTY
If you purchased your GXP1450 from a reseller, please contact the company where you purchased your
phone for replacement, repair or refund. If you purchased the product directly from Grandstream, contact
your Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number
before you return the product. Grandstream reserves the right to remedy warranty policy without prior
notification.
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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/RTCP, 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,
802.1x, TR-069
Superb Audio Quality Advanced Digital Signal Processing (DSP), Silence Suppression, VAD,
CNG, AGC
Network Interfaces Dual 10/100mbps Ethernet ports with integrated PoE
Feature Rich Traditional voice features including caller ID, call waiting, hold, transfer,
forward, block, auto-dial, off-hook dial
Advanced Features 2 line keys with dual-color LED and 2 SIP accounts, 3 way
conferencing, backlit graphic 180x60 LCD,
3 XML programmable
context sensitive soft keys, 5 navigation keys, 10 dedicated buttons for
HOLD, TRANSFER, CONFERENCE, VOLUME, HEADSET, MUTE,
SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE
Advanced Functionality Customized downloadable ring-tones, SRTP, SIP over TLS, multi-
language support and XML enabled, adjustable positioning angles, wall
mountable, AES encryption, automatic multimedia service (eg., weather
information)
Table 5: GXP1450 Hardware Specifications
GXP1450
LAN Interface (Ethernet ports) Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and
PC port with auto detection
Graphic LCD Display 180x60 pixel
Expansion Module Support No
Headset Jack RJ9
Call Appearance LED 2 Dual color (green/red)
Power over Ethernet Built-in auto-sensing: Cisco and IEEE 802.3af standard
Universal Switching Input: 100-240VAC 50-60 Hz
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Power Adaptor Output: +5VDC, 800mA, UL certified
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
2 lines with 2 independent SIP accounts, XML programmable soft-keys.
Protocol Support
Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP, SRTP by SDES,
HTTP, ARP/RARP, ICMP, DNS, DHCP, NT
P, TFTP,
SIMPLE/PRESENCE protocols, TR-069, 802.1x
Support multiple SIP accounts and up to 11 media channels
concurrently
Support SIP PUBLISH method (RFC 3903), SIP Presence package
(RFC 3856, 3863) for use of MFKs, SIP Dialog package (RFC 4235)
Support for SIP MESSAGE method (RFC 3428)
Backlit graphic LCD display, up to 4 level grayscale
Feature Keys HOLD, TRANSFER, CONF, VOLUME, HEADSET, MUTE,
SPEAKERPHONE, SEND/REDIAL, PHONEBOOK, MESSAGE, 3 XML
Programmable Softkeys, 5 Navigation keys,
Device
Management NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed
devices including behind firewall/NAT
Auto/manual provisioning system, Web GUI Interface
Address Book
Full-duplex hands-free speakerphone
Advanced Digital Signal Processing (DSP)
Dynamic negotiation of codec and voice payload length
Support for G.723,1 (5.3/6.3K), G.729A/B, G.711 a/µ-law, G.726-32,
G.722 (wide-band), and iLBC codecs
In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Silence Suppression, VAD (voice activity detection), CNG (comfort noise
generation), ANG (automatic gain control)
Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for
speakerphone mode, Support side tone
Adaptive jitter buffer control (patent-pending) and packet delay and loss
concealment
HD audio handset with HD wideband audio codecs for excellent double-
talk performance
Telephony Features Intuitive graphic user interface (GUI), downloadable phone book (XML,
LDAP), support for anonymous call using privacy header, MLS (multi
language support)
Voice mail indicator, downloadable custom ring-tones, call hold, call
transfer (attended/blind), call forward, call waiting, caller ID, mute, redial,
call log, caller ID display or block, Do-Not-Disturb (DND) and volume
control
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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 Buttons
LCD Button
LCD Button 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.
NETWORK STATUS
Displays the status of the phone and network. It will indicate whether the network is
down, starting or running (IP address). “## MISSED CALLS” is shown here too.
STATUS BAR
Shows the status of the phone, using icons as shown in the next table.
LINE STATUS INDICATOR
Displays the name of the account that is in use. Select another account by pressing
the LINE key on the left side.
SOFTKEYS
The softkeys are context sensitive and will change depending on the status of the
phone. Typical functions assigned to soft-buttons are:
• FORWARD ALL Unconditionally forwards the phone line to another phone
• MISSED CALL This option shows up there were unanswered calls to this
phone. The Missed Calls option shows a list of the missed
calls
• NEXTSCR Press this button to toggle between idle screen, weather
and IP Address.
• REDIAL Redials the last number
• END CALL Hangs up phone
Table 8: LCD Icons
Icon
LCD Icon Definitions
DND Icon: ON when the “Do Not Disturb” is activated
Calls Forwarded Icon: INDICATES calls are forwarded
Key pad lock Icon: ON when using STAR key to lock the keypad
Enter Password to unlock the keypad
Voice Mail / Message Waiting Indicator: ON when there is new voice mail
/ message
Network Status: Network is down
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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
FIGURE 1: GXP1450 KEYPAD LAYOUT
Table 9: GXP1450 Keypad Buttons
Key Button
Key Button Definitions
LINE BUTTONS
2 Line keys with LED, can be configured to different SIP profiles
HOLD
Place ACTIVE call on hold
TRANSFER
Transfer an ACTIVE call to another number
CONF
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. It
also allows user to toggle between headset and speaker
Enable/Disable hands-free speaker
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Enable/Disable handset mode ; or used as SEND/REDIAL
Press “–” or “+” to adjust the volume for handset/speakerphone/headset
Enter Keypad Configuration “MENU” mode when phone is in IDLE mode.
Use as ENTER key when in Keypad Configuration.
0 - 9, *, #
Standard phone keypad; press # key to send call; press * key to for IVR
functions
MAKING PHONE CALLS
Handset, Speakerphone and Headset Mode
The GXP1450 allows you to make phone calls via handset, speakerphone, or headset mode. During the
active calls the user can switch between the handset and the speaker by pressing the speaker key. For
headsets to operate, the user must plug the headset to an RJ9 port on the phone, which allows the user to
pick-up, speak, or hang-up calls.
Multiple SIP Accounts and Lines
GXP1450 can support up to two independent SIP accounts. 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.
The name of each account is conveniently printed next to its corresponding button. In off-hook state, select
an idle line and the name of the account (as configured in the web interface) is displayed on the LCD and a
dial tone is heard.
For example: Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as “VoIP 1”, “VoIP 2”,
respectively and ensure that they are active and registered. When LINE1 is pressed, you will hear a dial tone
and see “VoIP 1” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see “VoIP 2” on
the LCD display.
To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. User can
switch lines before dialing any number by pressing the same LINE button one or more times. If you continue
to press a LINE button, the selected account will circulate among the registered accounts.
For example: when LINE1 is pressed, the LCD displays “VoIP 1”; If LINE1 is pressed twice, the LCD
displays “VoIP 2” and the subsequent call will be made through SIP account 2.
Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When the
“virtually” mapped line is in use, the GXP1450 will flash the next available LINE in red. A line is ACTIVE
when it is in use and the corresponding LED is red.
Completing Calls
There are five ways to complete a call:
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NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IP-
IP 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. 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). Answer the incoming call by pressing its corresponding LINE button. The current active call
will be put on hold.
3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via SPEAKER.
(PBX or Server must also supports this feature)
Do Not Disturb
1. Press the menu button, and scroll down to “Preference”.
2. Select “Do Not Disturb” by pressing menu button.
3. Use arrow keys to either enable or disable “Do Not Disturb” feature.
4. When enabled, there will be a special “Do Not Disturb” icon appearing on the display. This will send
the incoming caller directly to voicemail.
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. The “Line Status Indicator” will show “LINEx: SPEAKING” or “LINEx: MUTE” to indicate whether the
microphone is muted.
Call Transfer
GXP1450 supports both Blind and Attended transfer:
1. Blind Transfer: Press “TRANSFER” button, then dial the number and press the “SEND” button to
complete transfer of active call.
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will be notified of an incoming call and will be able to answer the call from the phone with the SCA extension
registered.
All the users that belong to the same SCA group will be notified by visual indicator when a user seizes the
line and places an outgoing call, and all the users of this group will not be able to seize the line until the line
goes back to an idle state or when the call is placed on hold. (With the exception of when multiple call
appearances are enabled on the server side)
In the middle of the conversation, there are two types of hold: Public Hold and Private Hold. When a member
of the group places the call on public hold, the other users of the SCA group will be notified of this by the red-
flashing button and they will be able to resume the call from their phone by pressing the line button. However,
if this call is placed on private-hold, no other member of the SCA group will be able to resume that call.
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.
CALL FEATURES
The GXP1450 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.
Table 10: GXP1450 Call Features
Key Call Features
*30 Block Caller ID (for all subsequent calls)
*31 Send Caller ID (for all subsequent calls)
*67 Block Caller ID (per call)
*82 Send Caller ID (per call)
*70 Disable Call Waiting (per Call)
*71 Enable Call Waiting (per Call)
*72 Unconditional Call Forward
Dial “*72” for a dial tone. Dial the forwarding number followed by “#”. Wait for dial
tone. LCD will display “Call FWD Activated”.
*73 Cancel Unconditional Call Forward: dial “*73” and get the dial tone, then hang up.
LCD will display “Call FWD Activated”.
*90 Busy Call Forward
Dial “*90” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial
tone. Hang up.
*91 Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.
*92 Delayed Call Forward
Dial “*92” for a dial tone. Dial the forwarding number followed by “#”. Wait for a dial
tone. Hang up. LCD will display “Call FWD Activated”.
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• Erase Custom SCR
Custom idle screen will be erased and will be replaced with default
logo.
• Display Language
You can choose English, Simplified Chinese, Traditional Chinese,
Korean, Japanese, Italian, Spanish, French, German, Portuguese,
Russian, Croatian, Hungarian, Polish, Slovenian 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
Press Menu button to choose the menu item
Press “←” or follow the soft keys to return to the main menu
Config Press Menu button to display the configuration selections:
• SIP
To change SIP server settings for SIP accounts.
• 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. Do not use
Factory Reset unless you want to restore factory settings.
• Layer 2 QoS
Configure 802.1Q/VLAN Tag and priority value.
Factory Functions
Press Menu to display the factory function items including
• Audio Loopback
Speak into the handset. If you hear your voice in the handset, your audio
works fine. Press Menu button to exit the mode
• Diagnostic Mode
All LEDs will light up.
Press any key on the keypad, to display the button name in the LCD. Lift
and put back the handset or press Menu button to exit the diagnostic
mode.
Press “←” to return the main menu
Network To enable/disable DHCP; to setup IP-address, Net mask and Gateway address
Reboot Press Menu button to reboot the device
Exit Exit from this menu
FIGURE 2: KEYPAD GUI FLOW
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Call History
Status
Phone Book
LDAP Directory
Instant
Message
Direct IP Call
Preference
Config
Factory
Functions
Network
Reboot
Exit
MENU
Answered Calls
Dialed Calls
Missed Calls
Transferred Calls
Forwarded Calls
Back
Clear All
New Entry
Download Phonebook XML
Back
Phone Book
Name:
Number:
Acct:
Confirm Add:
Cancel and Return:
New Entry
View Directory
Download Directory
Search Configuration
Back
LDAP Directory
Select Filter
Filter Value
Back
Search Configuration
Clear All
Back
Preference
Do Not Disturb
Ring Tone
LCD Contrast
LCD Brightness
Download SCR XML
Erase Custom SCR
Display Language
Back
SIP
Upgrade
Factory Reset
Layer 2 QoS
Back
Audio Loopback
Diagnostic Mode
Back
Factory Function
Config
Enable DND
Disable DND
Back
Default Ring
Ring1
Ring2
Ring 3
Back
Active
Idle
Back
English
Chinese
French
Spanish
German
Italian
Secondary Language
Language File Postfix
Back
Do Not Disturb
Ring Tone
LCD Brightness
Display Language
Account
SIP Proxy
Outbound
Proxy
SIP User ID
SIP Auth ID
SIP Password
SIP Transport
Audio
Save
Firmware
Server
Config Server
Upgrade Via
802.1Q/VLAN Tag
Priority value
Reset Vlan Config
Back
Upgrade
Layer 2 QoS
Instant Message
Keypad/LED Diagnostic
Diagnostic Mode
SIP
IP Setting
PPPoE Settings
IP
Netmask
Gateway
DNS Server 1
DNS Server 2
Back
Network
Call History Any of previous menus
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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.
Access the Web Configuration Menu
To access the phone’s Web Configuration Menu
• Connect the computer to the same network as the phone1
• Make sure the phone is turned on and shows its IP address
• Start a Web browser on your computer
• Enter the phone’s IP address in the address bar of the browser2
• Enter the administrator’s password to access the Web Configuration Menu3
1 The Web-enabled computer has to be connected to the same sub-network as the phone. This can easily
be done by connecting the computer to the same hub or switch as the phone is connected to. In absence
of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone using
the PC port on the phone.
2 If the phone is properly connected to a working Internet connection, the phone will display its IP address.
This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0 to 255. You will need
this number to access the Web Configuration Menu. For example, if the phone shows 192.168.0.60,
please use “http://192.168.0.60” in the address bar of your browser.
3 The default administrator password is “admin”; the default end-user password is “123”.
NOTE: When changing any settings, always SUBMIT them by pressing “UPDATE” button on the bottom of
the page. Reboot the phone to have the changes take effect. If, after having submitted some changes, more
settings have to be changed, press the menu option needed.
Definitions
This section will describe the options in the Web configuration user interface. As mentioned, a user can log in
as an administrator or end-user.
Functions available for the end-user are:
• Status: Displays the network status, account status, software version and MAC address of the
phone, and service status.
• Basic Settings: Basic preferences such as date and time settings, multi-purpose keys and LCD
settings can be set here.
Additional functions available to administrators are:
• Advanced Settings: To set advanced network settings, codec settings and XML configuration
settings and etc.
• Account: To configure the SIP account.
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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
• Program: This is the main firmware release number, which is always used for
identifying the software (or firmware) system of the phone.
• Boot: Booting code version number
• Core: Core code version number
• Base: Base code version number
• DSP: DSP code version number
• Aux: Aux code version number
System Up Time
This field shows system up time since the last reboot.
System Time
This field shows the current time on the phone system.
Registered
Indicates whether accounts are registered to the related SIP server.
PPPoE Link Up
Indicates whether the PPPoE connection is enabled (connected to a modem).
Service Status
• GUI: shows the GUI status: running or stopped
• Phone: shows the phone status: running or stopped
Core Dump
Download core dump file for troubleshooting when necessary.
Table 13: Device Configuration – Settings/Basic Settings
End User Password
This contains the password to access the Web Configuration Menu. This field is case
sensitive with a maximum length of 25 characters.
IP Address
The GXP1450 operates in two modes:
1. DHCP mode: all the field values for the Static IP mode are not used (even
though they are still saved in the Flash memory.) The GXP1450 acquires its
IP address from the first DHCP server it discovers on its LAN. The DHCP
option is reserved for NAT router mode. To use the PPPoE feature, set the
PPPoE account settings. The GXP1450 establishes a PPPoE session if any
of the PPPoE fields is set.
2. PPPoE mode: configure all of the following fields: PPPoE account ID,
PPPoE password and PPPoE service name.
3. Static IP mode: configure all of the following fields: IP address, Subnet
Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2
(secondary). These fields are set to zero by default.
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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.
Table 14: Device Configuration – Settings /Advanced Settings
Admin
Password
Administrator password. Only the administrator can access the “Advanced Settings”
and “Account Settings” page. Password field is purposely blank for security reasons
after clicking update and saved. The maximum password length is 25 characters.
Layer 3 QoS
This field defines the layer 3 QoS parameter. It is the value used for IP Precedence
or Diff-Serv or MPLS. Default value is 12.
Layer 2 QoS
This contains the value used for layer 2 802.1Q/VLAN tag and 802.1p priority value.
Default setting is 0.
Local RTP port
This parameter defines the local RTP-RTCP port pair used to listen and transmit. It
is the base RTP port for channel 0. When configured, channel 0 will use this port
_value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2
for RTP and port_value+3 for its RTCP. Local RTP port ranges from 1024 to 65400
and must be even. The default value is 5004.
Use Random Port
This parameter, when set to “Yes”, will force random generation of both the local
SIP and RTP ports. This is usually necessary when multiple GXP
s are behind the
same NAT. Default is “No”.
Keep-alive interval
This parameter specifies how often the GXP1450 sends a blank UDP packet to the
SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
STUN Server
IP address or Domain name of the STUN server. STUN resolution result will display
in the STATUS page of the Web UI.
GXP1450 User Manual Page 25 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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. Please DO NOT
interrupt the upgrade process (especially the power supply) as this will
damage the device.
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.
Upgrade Via
This field allows the user to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
Firmware Server Path
Defines the server path for the firmware server. It can be different from the
Configuration server which is used for provisioning.
Config Server Path
Defines the server path for provisioning; it can be different from the firmware server.
Firmware File
Prefix/Postfix
Default is blank. If configured, GXP1450 will request the firmware file with the
prefix/postfix and only the firmware with the matching encrypted prefix will be
downloaded and flashed into the phone.
This setting is useful for ITSPs. End user should keep it blank.
Config File
Prefix/Postfix
Default is blank. If configured, GXP1450 will request the config file with the
prefix/postfix and only the file with the matching encrypted prefix will be downloaded
and flashed into the phone.
This setting is useful for ITSPs. End user should keep it blank.
Allow DHCP Option 43
and Option 66 to
override server
Default is “Yes”. This allows device gets provisioned from the server automatically.
Automatic Upgrade
This function is used by ITSP. End user should NOT touch these parameters.
Default is “No”. Choose “Yes” to enable automatic HTTP upgrade and provisioning.
In “Check for upgrade every” field, enter the number of minutes to check the HTTP
server for firmware upgrade or configuration changes.
When set to “No”, the phone
will only perform HTTP upgrade and configuration check once at boot up.
Authenticate Conf File
Default is “No”. If set to “Yes”, configuration file would be authenticated before
acceptance. End user should use default setting.
Enable TR-069
Default is “No”.
ACS URL
URL for TR-069 Auto Configuration Servers (ACS).
GXP1450 User Manual Page 26 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
TR-069 Username
Enter username for TR-069.
TR-069 Password
Enter password for TR-069.
Save Credentials
Save TR-069 credentials. Default is “No”.
Auto Login
Auto Login TR-069 account. Default is “No”.
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.
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.
Phonebook Download
Interval
The interval at which the phonebook will be downloaded from the download server
(in Minutes). The default setting is 0.
Remove Manually-edited
entries on Downloads
If set to “Yes”, the phone will remove the manually-edited entries in the old
phonebook list before downloading the new file. The default setting is set to “Yes”.
LDAP Directory
IP address or domain name of LDAP script server.
Idle Screen XML
Download
Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use
Custom Filename” or not, and define the “XML server path”.
Download Screen XML
At Boot-up:
The phone will download the idle screen xml file if set to “Yes”. The default setting
is “No”.
Use custom filename:
The phone will use custom filename specified in XML server path if set to “Yes”.
The default setting is “No”.
Idle Screen XML Server
Path:
Specify the idle screen XML server path.
XML Application
Server path: enter server path for XML application.
Softkey Label: define the softkey label for the XML application.
Offhook Auto Dial
To configure a User ID/extension to dial automatically when the phone is taken
offhook.
Syslog Server
The IP address or URL of System log server. This feature is especially useful for
ITSPs.
GXP1450 User Manual Page 27 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
Syslog Level
Select the ATA to report the log level. Default is NONE. The level is one of DEBUG,
INFO, WARNING or ERROR. Syslog messages are sent based on the following
events:
• product model/version on boot up (INFO level)
• NAT related info (INFO level)
• sent or received SIP message (DEBUG level)
• SIP message summary (INFO level)
• inbound and outbound calls (INFO level)
• registration status change (INFO level)
• negotiated codec (INFO level)
• Ethernet link up (INFO level)
• SLIC chip exception (WARNING and ERROR levels)
• memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains
the following components: GS_LOG: [device MAC address][error code]
error
message.
For example: May 19 02:40:38 192.168.1.14 GS
_LOG: [00:0b:82:00:a1:be][000].
Ethernet link is up.
Send SIP Log
When setting the “Yes”, phone will send out SIP Log to syslog server. Default
setting is “No”.
NTP server
This parameter defines the URI or IP address of the NTP (Network Time Protocol)
serve. It is used to display the current date/time.
Allow DHCP Option 42
to override NTP server
Default is “Yes”. This allows device gets provisioned for DHCP Option 42 from the
server automatically.
SSL Certificate
This defines the SSL certificate needed to access certain websites.
SSL Private Key
This defines the SSL Private key.
SSL Private Key
Password
This defines the SSL private key password.
Distinctive Ring Tone
Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a
particular Caller ID. The GXP1450 will ONLY use selected ring to
nes for particular
Caller IDs. For all other calls, the GXP1450 will use System Ring Tone.
When
selected and no Caller ID is configured, the se
lected ring tone will be used for all
incoming calls.
System Ring Tone
System ring tone. Default is North American standard.
Adjust system ring tone frequencies and cadences based on local telecom
standard.
GXP1450 User Manual Page 28 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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. In
order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms
and a pause of OFF ms and then repeat the pattern. Up to three cadences are
supported.
Intercom User ID
Configure intercom user ID when intercom is used.
Disable Call Waiting
Default is “No”. If set to “Yes”, the call waiting feature will be disabled.
Disable Call
Waiting Tone
Default is “No”. If set to “Yes”, the call waiting tone will be disabled.
Disable Direct IP Calls
Default is “No”. If set to “Yes”, direct IP calls will be disabled.
Use Quick IP Call Mode
Dial an IP address under the same LAN/VPN segment by entering the last octet in
the IP address.
In the Advanced Settings page there is an option “Use Quick IP-call mode”.
Default
setting is “No”. When set to “Yes”, and #XXX is dialed, where X is 0-
9 and XXX
<=255, phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc
comes from the local IP address REGARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK). See
Quick IP Call
Mode for details.
Disable Conference
Default is “No”. If set to “Yes”, conference will be disabled.
Disable DND Button
Default is “No”. If set to “Yes”, the “DND” button on keypad will be disabled.
Disable Transfer
Default is “No”. If set to “Yes”, transfer will be disabled.
Auto-Attended Transfer
Default is “No”. If set to “Yes”, the phone will use attended transfer by default.
Configuration via
Keypad Menu
Configures the access control of configurations via the phone keypad menu. There
are three modes:
• Unrestricted
• Basic Settings Only
•
Constraint Mode
Enable STAR key
Keypad locking
Default is No. If set to “Yes”, when pressing STAR key for 4-5 seconds, there will be
a lock icon shown in the right side of the screen indication the keypad is locked.
To unlock, pressing STAR key for 4-
5 seconds and there will be a window
prompted asking for password.
Password to lock/unlock
Enter the password to lock the keypad in web GUI.
To unlock the keypad, enter the password in the prompted window in the phone’s
LCD screen.
GXP1450 User Manual Page 30 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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. 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.
Backup IP 1
Insert the first back up IP here.
Backup IP 2
Insert the second back up IP here.
SIP Registration
This parameter controls sending REGISTER messages to the proxy server. The
default setting is “Yes”.
Unregister on Reboot
Default is “No”. If set to “Yes”, the SIP user’s registration information will be cleared
on reboot.
Register Expiration
This parameter allows user to specify the time frequency (in minutes) that GXP1450
refreshes its registration with the specified registrar. The default interval is 60
minutes. The maximum interval is 65,535 minutes (about 45 days).
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 Wait Time
Retry registration if the process failed. Default is 20 seconds.
SIP T1 Timeout
RFC 3261 SIP T1 timer. Default is 0.5 second.
SIP T2 Interval
RFC 3261 SIP T2 timer. Default is 4 seconds.
SIP Transport
Choose SIP Transport between UDP and TCP. Default is UDP.
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.
GXP1450 User Manual Page 31 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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. If the detected
NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use
its mapped public IP address and port in all of its SIP and SDP messages.
If selecting Keep-Alive with no specified STUN server, the GXP1450 will periodically
(every 20 seconds or so) send a blank UDP packet (with no payload data) to the
SIP server to keep the “hole” on the NAT open.
SUBSCRIBE for MWI
Default is “No”. When set to “Yes” a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
PUBLISH for Presence
Enable Presence feature.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail UserID
When configured, user can access messages by pressing “MSG” button. This ID is
usually the VM portal access number.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes: in audio which means DTMF is combined in audio signal (not
very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
DTMF Payload Type
Sends DTMF using RFC2833. The default is 101.
Early Dial
Default is “No”. Use only if proxy supports 484 responses.
Dial Plan Prefix
Sets the prefix added to each dialed number.
GXP1450 User Manual Page 32 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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. - only 2 digit numbers
c) ^ - exclude
d) [3-5] - any digit of 3, 4, or 5
e) [147] - any digit of 1, 4, or 7
f) <2=011> - replace digit 2 with 011 when dialing
g) | - the OR operand
• Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617
• Example 2: {^1900x+ | <=1617>xxxxxxx}
Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit
numbers
• Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}
Allows any number with leading digit 1 followed by a 3 digit number, followed by any
number between 2 and 9, followed by any 7 digit number OR Allows any length of
numbers with leading digit 2, replacing the 2 with 011 when dialed.
3. Default: Outgoing – {x+}
Allow any length of numbers.
Example of a simple dial plan used in a Home/Office in the US:
{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):
• ^1900x. - prevents dialing any number started with 1900
• <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing
7 numbers and 1617 area code will be added automatically
• 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 - allow 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. An example dial plan will be: { *x+ } which
allows the user to dial * followed by any length of numbers.
Delayed Call Forward
Wait Time
Time waited before the call is forward to a number or VM.
Default is 20 seconds.
Enable Call Features
Default is “Yes”. If set to “No”, Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided ITSP support those features. In addition, “ForwardAll”
softkey will be hidden if call feature code is disabled for Account 1.
Call Log
User can choose to disable Call Log and what kind of calls to log.
GXP1450 User Manual Page 33 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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.
Min-SE
Defines the minimum session expiration (in seconds). Default is 90 seconds.
Caller Request Timer
If set to “Yes”, the phone will use session timer when it makes outbound calls if
remote party supports session timer.
Callee Request Timer
If selecting “Yes”, the phone will use session timer when it receives inbound calls
with session timer request.
Force Timer
If set to “Yes”, the phone will use session timer even if the remote party does not
support this feature. If set to “No”, the session timer is enabled only when the
remote party supports this feature. To turn off Session Timer, select “No” for Caller
Request Timer, Callee Request Timer, and Force Timer.
UAC Specify Refresher
As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee
or proxy server as the refresher.
UAS Specify Refresher
As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
Force INVITE
Session Timer can be refreshed using INVITE method or UPDATE method. Select
“Yes” to use INVITE method to refresh the session timer.
Enable 100rel
PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional
responses (1xx series). This is required to support PSTN inter-networking.
Account Ring Tone
There are 4 uniquely defined ring tones:
• One (1) System Ring Tone: when selected, all calls will ring with system
ring tone.
• Three (3) Customer Ring Tones: when selected, incoming calls from
designated account will play selected ring tone.
Ring Timeout
Defines how long ring will ring when receiving a call. Default is 60 seconds.
Send Anonymous
If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will
be set to anonymous, essentially blocking the Caller ID from displaying.
Anonymous Call
Rejection
Default is “No”. If set to “Yes”, anonymous call will be rejected.
Auto Answer
Default is “No”. If set to “Yes”, GXP1450 will automatically switch on speaker to
answer the incoming call. Set to Intercom/Paging mode, it will answer the call based
on the SIP info header from the server.
Allow Auto Answer by
Call-Info
If the Call-Info header contains answer-after=0, the call be answered automatically
(so called paging mode).
GXP1450 User Manual Page 34 of 38
Firmware 1.0.1.66 Last Updated: 05/2011
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
Hangup
Defines whether or not the call is transferred to the other party if the initiator of the
conference hangs up.
Default setting is set to “No”.
Preferred Vocoder
GXP1450 supports up to 7 different Vocoder types including G.711(a/µ) (also
known as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).
Configure Vocoders in a preference list that is included with the same preference
order in SDP message. Enter the first Vocoder in this list by choosing the
appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by
choosing the appropriate option in “Choice 8”.
SRTP Mode
Enable SRTP mode based on selection. Default is “No”.
Symmetric RTP
Selects whether or not symmetric RTP is supported.
Silence Suppression
This controls the silence suppression/VAD feature of the audio codec G.723 and
G.729. If set to “Yes”, when silence is detected, a small quantity of VAD packets
(instead of audio packets) will be sent during the period of no talking. If set to “No”,
this feature is disabled.
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. This parameter is associated
with the first
codec in the above codec Preference List or the actual used payload
type negotiated between the 2 conversation parties at run time. E.g.
, if the first
codec is confi
gured as G.723 and the “Voice Frames per TX” is set to 2, then the
“ptime” value in the SDP message of an INVITE request will be 60ms
because each
G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the
first codec is G.729 o
r G.711 or G.726, then the “ptime” value in the SDP message
of an INVITE request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the IP
phone will use and save the maximum allowed value for the corresponding first
codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20
(x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms)
and 64 (x2.5ms) frames respectively.
Please be careful when editing these parameters. Adjustin
g these parameters will
also change the dynamic jitter buffer. The GXP1450
has a patent dynamic jitter
buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.
We recommend using the default settings provided. We do not recommend
adjusting these parameters if you are an average user. Incorrect settings will affect
the voice quality.
No Key Entry Timeout
Default is 4 seconds. After the timeout, the phone will send out the dialed number.

Especificaciones del producto

Marca: Grandstream
Categoría: telefono
Modelo: GXP1450

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