Grandstream GXP1400
Grandstream GXP1400
Grandstream GXP1400
WELCOME .................................................................................................................................................................3
INSTALLATION.........................................................................................................................................................4
EQUIPMENT PACKAGING .............................................................................................................................................4
CONNECTING YOUR PHONE ........................................................................................................................................4
SAFETY COMPLIANCES................................................................................................................................................4
WARRANTY .................................................................................................................................................................4
PRODUCT OVERVIEW ............................................................................................................................................5
USING THE GXP1400/1405.......................................................................................................................................8
GETTING FAMILIAR WITH THE LCD ............................................................................................................................8
MAKING PHONE CALLS ...............................................................................................................................................9
ANSWERING PHONE CALLS ....................................................................................................................................... 12
PHONE FUNCTIONS DURING A PHONE CALL ............................................................................................................. 12
CALL FEATURES ........................................................................................................................................................ 14
CUSTOMIZED LCD SCREEN & XML ......................................................................................................................... 15
CONFIGURATION GUIDE ...................................................................................................................................... 16
CONFIGURATION VIA KEYPAD .................................................................................................................................. 16
CONFIGURATION VIA WEB BROWSER ...................................................................................................................... 19
SAVING THE CONFIGURATION CHANGES ................................................................................................................... 33
REBOOTING THE PHONE REMOTELY.......................................................................................................................... 33
SOFTWARE UPGRADE & CUSTOMIZATION .................................................................................................. 34
FIRMWARE UPGRADE THROUGH TFTP/HTTP .......................................................................................................... 34
CONFIGURATION FILE DOWNLOAD ........................................................................................................................... 35
RESTORE FACTORY DEFAULT SETTING ....................................................................................................... 36
TABLE OF TABLES
GXP1400/1405 USER MANUAL
http://www.grandstream.com/products/gxp_series/general/documents/gxp21xx_gui.zip
GXP1400/1405 is a next generation small-to-medium business IP phone that features 2 lines with 1 SIP
account, a 128x40 graphical LCD, 3 XML programmable context-sensitive soft keys, dual network ports
with integrated PoE (GXP1405 only), and 3-way conference. The GXP1400/1405 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
rd
privacy, and broad interoperability with most 3 party SIP devices and leading SIP/NGN/IMS platforms. It
is a perfect choice for small-to-medium businesses 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 GXP1400/1405 as it may cause damage
to the products and void the manufacturer warranty.
Note:
• 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.
The connectors of the GXP1400/1405 are located on the bottom of the device.
SAFETY COMPLIANCES
The GXP1400/1405 phone complies with FCC/CE and various safety standards. The GXP1400/1405 power
adaptor is compliant with the UL standard. Please use the universal power adaptor provided with the
GXP1400/1405 package only. The manufacturer’s warranty does not cover damages to the phone caused by
unsupported power adaptors.
WARRANTY
If you purchased your GXP1400/1405 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.
Features GXP1400/1405
LCD Display 128 x 40 pixel
Number of Lines 2
Programmable Soft Keys 3
Extension Module N/A
Features Benefits
Open Standards SIP RFC3261, TCP/IP/UDP, RTP, HTTP/HTTPS, ARP/RARP, ICMP,
Compatibility 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 10/100 Mbps Ethernet port, integrated PoE (GXP1405 only)
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 1 SIP account, 3 way conference,
graphic LCD, 3 XML programmable context sensitive soft keys, 5
navigation keys, 8 dedicated buttons for HOLD, TRANSFER,
CONFERENCE, VOLUME, HEADSET, MUTE/DND, SPEAKERPHONE,
SEND/REDIAL
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)
GXP1400/1405
LAN Interface 10/100 Mbps Full/Half Duplex Ethernet port with auto detection
Integrated PoE (GXP1405 only)
Graphic LCD Display 128 x 40 pixel
Expansion Module N/A
Call Appearance LED 2 Dual color (green/red) line keys
GXP1400/1405 has a dynamic and customizable screen. The screen displays differently depending on
whether the phone is idle or in use (active screen).
DATE AND TIME Displays the current date and time. It can be synchronized with Internet time
servers
LOGO NAME Displays company logo name. This logo name can be customized via xml screen
customization. The maximum size for logo name is 22 characters in English
NETWORK Shows the status of network in the middle of the screen. It will indicate whether
STATUS the network is down or starting
STATUS BAR Shows the status of the phone, using icons as shown in the next table
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 unanswered calls to this phone.
• NEXTSCR Press this button to toggle between idle screen, weather
and IP Address.
• REDIAL Redials the last dialed-out number
• END CALL Hangs up the call
MUTE Icon:
INDICATES call is on MUTE during the call
SRTP Icon:
INDICATES SRTP is enabled for the call
Button Descriptions
Mute an active call; or use as DND button when the phone is in idle state.
Press HEADSET key to answer/hang up phone calls when using headset. It also
allows user to toggle between headset and speaker
Standard phone keypad; press # key to send call; press * key to for IVR
0 - 9, *, #
functions
The GXP1400/1405 allows you to make phone calls via handset, headset or speakerphone. During the
active calls the user can switch between the handset, headset and the speakerphone by pressing the
corresponding keys on the phone.
Grandstream Networks, Inc. GXP1400/1405 User Manual Page 9 of 36
Firmware version: 1.0.1.83 Last Updated: 08/2011
GXP1400/1405 can support up to two lines “virtually” mapped to a SIP account. In off-hook state, select an
idle line and the dial tone will be heard. To make a call, select the line you wish to use. The user can switch
lines before dialing any number by pressing the LINE button.
Completing Calls
5. VIA PAGE/INTERCOM: Server/PBX has to support Page/Intercom. Also, GXP1400/1405 and PBX have
to be configured correctly.
• Take Handset off hook
or press SPEAKER button
or press HEADSET button
or press an available LINE key to activate speakerphone
• Press OK and the screen will display “LINEx: PAGE”
NOTE:
• Dial-tone and dialed number display occurs after the handset is off-hook, or handset button is
pressed, or speaker button is pressed, or the line key is selected. After dialing the number, the
phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call.
Press “#” button to override the 4 second delay.
Direct IP Call allows two phones to talk to each other in an ad-hoc fashion without a SIP proxy. VoIP calls
can be made between two phones if:
• Both phones have public IP addresses, or
• Both phones are on a same LAN/VPN using private or public IP addresses, or
• Both phones can be connected through a router using public or private IP addresses (with necessary
port forwarding or DMZ)
To make a direct IP call, please follow these steps:
• Press MENU button to bring up MAIN MENU
• Select “Direct IP Call” using the arrow-keys
• Press OK to select
• Input the 12-digit target IP address. (Please see example below)
• Press OK key to initiate call.
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 represents the dot “.”; the “#” key represents colon “:”. Press
OK to dial out.
The GXP1400/1405 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. This is possible only if both phones
are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server.
Controlled static IP usage is recommended.
To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the
“Advanced Settings” page, set the "Use Quick IP-call mode” to “Yes”. When #xxx is dialed, where x is 0-9
and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. “aaa.bbb.ccc” is from the local IP address
regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK).
For example:
192.168.0.2 calling 192.168.0.3 -- dial #3 followed by #
192.168.0.2 calling 192.168.0.23 -- dial #23 followed by #
192.168.0.2 calling 192.168.0.123 -- dial #123 followed by #
192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3
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.
Receiving Calls
1. Incoming single call: Phone rings with selected ring-tone. The corresponding LINE flashes in red.
Answer call by taking Handset off hook or pressing SPEAKER or HEADSET or by pressing the
corresponding account LINE button.
2. Incoming multiple calls: When another call comes in while having an active call, the phone will
produce a Call Waiting tone (stutter tone). Answer the incoming call by pressing its corresponding
LINE button. The current active call will be put on hold.
Do Not Disturb
Do Not Disturb can be enabled/disabled by pressing the MUTE/DND button on the phone. Or users
could set it from the MENU following the steps below.
Mute
1. During the call, press the MUTE button to enable/disable muting the microphone.
2. The “Line Status Indicator” will show “LINEx: TALKING” or “LINEx: MUTE” to indicate whether the
microphone is muted.
Call Transfer
GXP1400/1405 supports both Blind and Attended transfer. Also, users could make auto-attended transfer
when this feature is enabled from web GUI.
1. Blind Transfer: Press “TRANSFER” button, then dial the number and press the # button to
complete transfer of active call.
3-Way Conferencing
GXP1400/1405 can host conference calls and supports up to 3-way conference calling.
2. Cancel Conference:
If after pressing the “CONF” button, a user decides not to conference anyone, press HOLD
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 LINE key
GXP1400/1405 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. Also the conference
can be re-established by pressing the ReConf softkey when the conference is on hold. Easy Conference
mode can be used combined with the traditional ways to establish 3-way conference.
2. 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
-
3. End Conference:
NOTE:
• The party that starts the conference call has to remain in the conference for its entire duration, you
can put the party on mute but it must remain in the conversation. Also, this is not applicable when the
feature “Transfer on call hangup” is turned on.
• When using Easy Conference mode, press SEND button to establish the second call after entering
the number instead of using “#”.
A blinking red MWI (Message Waiting Indicator) on the top right corner of the GXP1400/1405 indicates a
message is waiting. Dial into the voicemail box to retrieve the message. An IVR will prompt the user through
the process of message retrieval.
The GXP1400/1405 phone supports shared call appearance by Broadsoft standard. This feature allows
members of the SCA group to shared SIP lines and provides status monitoring (idle, active, progressing,
hold) of the shared line. When there is an incoming call designated for the SCA group, all of the members of
the group 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 the phone. In
addition, they would need to navigate to “Settings”->”Basic Settings” on the web UI and set the line to
“Shared Line”. 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 GXP1400/1405 supports traditional and advanced telephony features including caller ID, caller ID
w/name, call forward/transfer/park/hold as well as intercom/paging.
GXP1400/1405 IP phone support 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 soft-keys on GXP1400/1405, please visit our website at
http://www.grandstream.com/support.
Item Description
Call History Displays histories of answered, dialed, missed, and transferred and forwarded
calls. Select “Clear All” to clear all the call history entries.
Status Displays the network status, account status, software version and hardware
version of the phone.
Press network status to enter the sub menu for IP setting information
(DHCP/Static IP/PPPoE), Subnet Mask, Gateway and DNS server.
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
is working 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.
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 in
Menu->Status. 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.
• All the options under Basic Setting and Account Setting, and most of the options under Advanced
Setting do not require reboot after submitting the changes. Under Advanced Setting, the parameters
on network configuration require reboot after update.
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.
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) and the
NAT type.
Core Dump Download core dump file for troubleshooting when necessary.
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.
Line Keys x This allows the user to configure the account mapped to each line key, as well as
enabling SCA (Shared Call Appearance) for the line.
Options available for Key Mode are :
1. Line
2. Shared Line
Time Zone This parameter controls the date/time display according to the specified time zone.
If “Allow DHCP Option 2 to override Time Zone setting” is checked, the time zone will
be overridden by the DHCP server.
Self-Defined Time This parameter allows the users to define their own time zone.
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.., 12 (for Jan, Feb, .., Dec)
rd
The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3
Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, … ,Sat)
Therefore, this example is the DST which starts from the first Sunday of April to the
1st Sunday of November.
Weather Update By default, “Enable Weather Update:” is set to “Yes”. If set to “No”, weather
information will not display on the phone.
This is displayed when “Enable Weather Update” is set to “Yes” and pressing the
‘SwitchSCR’ soft-key once.
Disable in-call DTMF Default is “No”. This field is used to hide the keypad input during a call.
display
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.
Admin Administrator password. Only the administrator can access the “Advanced Settings”
Password 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 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; channel 1 will use port_value+2 for RTP. 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 GXPs are behind the
same NAT. Default is “No”.
Keep-alive interval This parameter specifies how often the GXP1400/1405 sends a blank UDP packet
to the SIP server in order to keep the “hole” on the NAT open. Default is 20
seconds.
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 Allows the user to select the following options for firmware upgrade:
Provisioning • Always Check for New Firmware
• Check New Firmware only when F/W pre/suffix changes
• Always Skip the Firmware Check.
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.
HTTP/HTTPS User Name The user name for the HTTP/HTTPS server.
HTTP/HTTPS Password The password for the HTTP/HTTPS server. It won’t display for security protection.
Upgrade Via This field allows the user to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
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 config server path for provisioning; it can be different from the Firmware
server.
Firmware File Default is blank. If configured, GXP1400/1405 will request the firmware file with the
Prefix/Postfix 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 Default is blank. If configured, GXP1400/1405 will request the config file with the
Prefix/Postfix 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 Default is “Yes”. This allows device to get provisioned from the server automatically.
and Option 66 to
override server
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.
Periodic Inform Interval When enabling periodic inform, set up the periodic inform interval.
Authentication Method Select the authentication method among “No authentication”, “Basic” or Digest.
Phonebook XML Selects the file download mode for the download server. Users can choose from
Download TFTP/HTTP/No.
Phonebook XML Server The URL/IP address of the phonebook download server.
Path
Phonebook Download The interval at which the phonebook will be downloaded from the download server
Interval (in Minutes). The default setting is 0.
Remove Manually-edited If set to “Yes”, the phone will remove the manually-edited entries in the old
entries on Downloads phonebook list before downloading the new file. The default setting is set to “Yes”.
Idle Screen XML Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use
Download Custom Filename” or not, and define the “XML server path”.
Download Screen XML The phone will download the idle screen xml file if set to “Yes”. The default setting
At Boot-up 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 Specify the idle screen XML server path.
Path
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.
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 Default is “Yes”. This allows device gets provisioned for DHCP Option 42 from the
to override NTP server server automatically.
SSL Certificate This defines the SSL certificate needed to access certain websites.
SSL Private Key This defines the SSL private key password.
Password
Distinctive Ring Tone Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a
particular Caller ID. The GXP1400/1405 will ONLY use selected ring tones for
particular Caller IDs. For all other calls, the GXP1400/1405 will use System Ring
Tone. When selected and no Caller ID is configured, the selected 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.
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.
Disable Call Waiting Default is “No”. If set to “Yes”, the call waiting feature will be disabled.
Disable Call Default is “No”. If set to “Yes”, the call waiting tone will be disabled.
Waiting Tone
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 DND Button Default is “No”. If set to “Yes”, the “DND” button on keypad will be disabled.
Auto-Attended Transfer Default is “No”. If set to “Yes”, the phone will use attended transfer by default.
Configuration via Configures the access control of configurations via the phone keypad menu. There
Keypad Menu are three modes:
• Unrestricted
• Basic Settings Only:
CONFIG option will not display in keypad MENU
• Constraint Mode:
CONFIG, FACTORY FUNCTIONS and NETWORK options will not display
in keypad MENU
Enable STAR key If enabled, when the phone is in idle screen, press and hold STAR key for 4
Keypad locking seconds and the keypad will be locked. The password to lock/unlock can be
configured.
Do not escape “#” as Default is “No”. By default, # will be replaced as %23 in SIP URI.
%23 in SIP URI
Language file postfix allows the language file to have different postfixes so the
phone can request a particular file. It will append an underscore "_" plus the string
in the language file postfix.
The default language file name is "gxp.txt". If the field “Language File postfix “has
"NL" string in it, then the phone will request "gxp_NL.txt" instead of "gxp.txt".
1. Get the language file gxp.txt ready. Make sure the file is using UTF-8 encoding.
2. Copy gxp.txt to the firmware server directory using your local TFTP or HTTP
server.
3. Access the advanced settings of the Web GUI, set “Display Language” to
“Download Language” and enter the server path in Firmware Server Path. Select
TFTP or HTTP for firmware upgrade.
4. Update and reboot the phone.
Account Name The name associated with each account - displayed on LCD.
SIP Server SIP Server’s IP address or Domain name provided by VoIP service provider.
Secondary SIP Server This field allows administrator to configure a backup SIP Server.
Outbound Proxy IP address or Domain name of Outbound Proxy, Media Gateway, or Session Border
Controller. Used for firewall or NAT penetration in different network environment. If
the system detects symmetric NAT, STUN will not work. ONLY outbound proxy can
provide solution for symmetric NAT.
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 GXP1400/1405 to register to (SIP)
servers of ITSP.
Name SIP service subscriber’s name that is used for Caller ID display.
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.
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
GXP1400/1405 refreshes its registration with the specified registrar. The default
interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days).
Reregister Before This parameter allows user to specify the time frequency (in seconds) that
Expiration GXP1400/1405 sends out a re-registration request before the Register Expiration.
By default is 0 second.
Local SIP Port This parameter defines the local SIP port used to listen and transmit. The default
value is 5060.
SIP Registration Failure Retry registration if the process failed. Default is 20 seconds.
Retry Wait Time
SIP Transport Choose SIP Transport between UDP and TCP. Default is UDP.
Use Actual Ephemeral Enable to use actual ephemeral port in contact with TCP/TLS. Default is “No”.
Port in Contact with
TCP/TLS
Remove OBP from The SIP Extension notifies the SIP server that it is behind a NAT/firewall.
Route
Validate Incoming This configuration selects whether or not the incoming messages should be
Messages validated.
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.
SUBSCRIBE for MWI Default is “No”. When set to “Yes”, a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
SUBSCRIBE for Default is “No”. When set to “Yes” a SUBSCRIBE for Registration will be sent
Registration periodically.
Feature Key Default is “No”. This option is to synchronize DND/Call Forward features with
Synchronization Broadsoft. When set to “Yes”, a SUBSCRIBE will be sent out periodically to the
server. Then when DND/Call Forward features (Call Forward No Answer,
Unconditional Call Forward and Call Forward on Busy) are configured or changed
on the phone and the Broadsoft server side, those features will be synchronized on
the phone side and the Broadsoft server side.
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.
Delayed Call Forward Time waited before the call is forward to a number or VM. Default is 20 seconds.
Wait Time
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.
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.
Ring Timeout Defines how long ring will ring when receiving a call. Default is 60 seconds.
Line-seize Timeout Defines how long before the line can be seized when Share Line is used. Default is
15 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 Default is “No”. If set to “Yes”, anonymous call will be rejected.
Rejection
Auto Answer Default is “No”. If set to “Yes”, GXP1400/1405 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.
Refer-To Use Target Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header
Contact 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”.
Preferred Vocoder GXP1400/1405 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”.
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 configured 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 or 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. Adjusting these parameters will
also change the dynamic jitter buffer. The GXP1400/1405 has a patent dynamic
jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.
Use # as Dial Key This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If
set to “Yes”, the “#” key will immediately send the call. In this case, this key is
essentially equivalent to the “(Re)Dial” key. If set to “No”, the “#” key is included as
part of the dial string.
G723 Rate Encoding rate for G723 codec. By default, 6.3kbps rate is set.
G726-32 Packing Mode Select “ITU” or “IETF” for G726-32 packing mode.
ilbc Frame Size ilbc packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required.
ilbc Payload Type Payload type for Ilbc. Default value is 97. The valid range is between 96 and 127.
Conference URI Configure the conference URI when using Broadsoft N-way calling feature.
Special Feature Default is Standard. Choose the selection to meet special requirements from Soft
Switch vendors.
• firmware.mycompany.com:6688/Grandstream/1.2.3.5
• 72.172.83.110
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web
Configuration Interface.
To configure the Upgrade Server via Key Pad Menu options, select “Config” from the Main Menu, then select
“Upgrade”. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN
format. Choose “Save and use TFTP” or “Save and use HTTP” to select upgrade method. Select “Reboot”
from the Main Menu to reboot the phone.
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the
GXP1400/1405 IP address. Enter the admin password to access the web configuration interface. In the
ADVANCED SETTINGS page, enter the Upgrade Server’s IP address or FQDN in the “Firmware Server
Path” field. Select TFTP or HTTP upgrade method. Update the change by clicking the “Update” button.
“Reboot” or power cycle the phone to update the new firmware.
During this stage, the LCD will display the firmware file downloading process. Please do NOT disrupt or
power down the unit. If a firmware upgrade fails for any reason (e.g., TFTP/HTTP server is not responding,
there are no code image files available for upgrade, or checksum test fails, etc), the phone will stop the
upgrading process and re-boot using the existing firmware/software.
Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We
recommend completing firmware upgrades in a controlled LAN environment whenever possible.
For users who do not have a local TFTP/HTTP server, we provide a HTTP server on the public Internet for
users to download the latest firmware upgrade automatically. Please check the Support/Download section of
our website to obtain this HTTP server IP address: http://www.grandstream.com/support/firmware.
Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades. A
free Windows version TFTP server is available:
http://support.solarwinds.net/updates/New-customerFree.cfm.
1. Unzip the file and put all of them under the root directory of the TFTP server.
2. The PC running the TFTP server and the GXP1400/1405 should be in the same LAN
segment.
3. Go to File -> Configure -> Security to change the TFTP server's default setting from
"Receive Only" to "Transmit Only" for the firmware upgrade.
4. Start the TFTP server, in the phone’s web configuration page
5. Configure the Firmware Server Path with the IP address of the PC
6. Update the change and reboot the unit
User can also choose to download the free HTTP server from http://httpd.apache.org/ or use Microsoft IIS
web server.
NOTE:
• When GXP1400/1405 phone boots up, it will send TFTP or HTTP request to download configuration
file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP1400/1405 phone.
This file is for provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following
error messages in a TFTP or HTTP server log can be ignored: “TFTP Error from [IP ADRESS]
requesting cfg000b82023dd4 : File does not exist. Configuration File Download”
The GXP1400/1405 can be configured via Web Interface as well as via Configuration File (binary or XML)
through TFTP or HTTP/HTTPS. The “Config Server Path” is the TFTP or HTTP server path for the
configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server
Path” can be the same or different from the “Firmware Server Path”.
A configuration parameter is associated with each particular field in the web configuration page. A parameter
consists of a Capital letter P and 2 to 4 digit numeric numbers. i.e., P2 is associated with “Admin Password”
in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding
configuration template of the firmware.
Once the GXP1400/1405 boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx”
followed by a request for configuration XML file named “cfgxxxxxxxxxxxx.xml”, where “xxxxxxxxxxxx” is the
MAC address of the device, i.e., “cfg000b820102ab”. The configuration file name should be in lower cases.
When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at pre-
scheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider can
manage and reduce the Firmware or Provisioning Server load at any given time.
Step 1: Press “OK” button to bring up the keypad configuration menu, select “Config”, press “OK” to
enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)
Step 2: Enter the MAC address printed on the bottom of the sticker. Please use the following mapping:
0-9: 0-9
A: 22 (press the “2” key twice, “A” will show on the LCD)
B: 222
C: 2222
D: 33 (press the “3” key twice, “D” will show on the LCD)
E: 333
F: 3333
NOTE:
• If there are digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to
move the cursor or wait for 4 seconds to continue to key in another “2”.
Step 3: Press the “OK” button to move the cursor to “OK”. Press “OK” button again to confirm. If the
MAC address is correct, the phone will reboot. Otherwise, it will exit to previous keypad menu interface.