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GXP1780
Administration Manual
63 pgs1.39 Mb0
Quick Installation Manual
29 pgs8.99 Mb0
Quick User Manual
1 pgs501.96 Kb0
User Manual
72 pgs2.68 Mb0
User Manual
33 pgs1.24 Mb0
Table of contents
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Grandstream Networks GXP1780, GXP1782 User Manual

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Grandstream Networks, Inc.
GXP1760 & GXP1780/82 - Small Business IP Phones Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
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GXP17xx Administration Guide
GXP1760
6 lines 200*80-pixel backlit LCD display 4 XML programmable soft keys 9 Dedicated function Keys 4-way conference
GXP1780 GXP1782
8 lines 200*80-pixel backlit LCD display 4 XML programmable soft keys 9 Dedicated function Keys 5-way conference
Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record,
SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP,
TR-069, 802.1x, TLS, SRTP, IPV6
Network Interface
Dual switched auto-sensing 10/100 Mbps Ethernet ports with integrated PoE
Graphic Display
200*80 pixel backlit LCD display
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Features Keys
6 line keys with up to 3 SIP accounts, 4 XML programmable context sensitive
softkeys, 5 navigation/menu keys, 11 dedicated function keys for : MESSAGE(with
LED indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-
Voice Codecs
Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723, ILBC, in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO)
Auxiliary Ports
RJ9 headset jack (allowing EHS with Plantronics headsets),
Telephony Features
Hold, transfer, forward, 4-way conference, call park, call pickup, shared-call-appearance(SCA)/bridged-line-appearance(BLA), downloadable
phonebook(XML, LDAP, up to 2000 items), call waiting, call log(up to 500 records),
XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible
dial plan, hot desking, personalized music ringtones and music on hold, server
redundancy and fail-over
HD Audio
Yes, HD handset and speakerphone with support for wideband audio
Base Stand
Yes, 2 angle positions available, Wall Mountable(optional)
QoS
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control
Multi-Language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
Chinese, Korean, Japanese
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS, mass provisioning using TR-069 or AES encrypted XML configuration file.
Power & Green
Energy Efficiency
Universal power adapter included: Input:100-240V; Output: +5V, 1A ; Integrated Power-over-Ethernet(802.3af) Max power consumption : 5W
Physical
Dimension: 231mm(W) x 167mm(L) x 86mm(H) Unit weight: 0.925kg. Package weight: 1.55kg
Temperature and
Humidity
Operation: 0°C to 40°CStorage: -10°C to 60°C Humidity: 10％ to 90% Non-condensing
Package Content
GXP1760 phone, handset with cord, base stand, universal power supply, network
cable, Quick Installation Guide, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B CE: EN55022 Class B; EN55024 Class B; EN61000-3-2; EN61000-3-3;
EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; ASs/NZS 60950.1
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Protocols/Standards
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS(A record,
SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP,
TR-069, 802.1x, TLS, SRTP, IPV6
Network Interface
Dual switched Ethernet ports with integrated PoE. Auto-sensing 10/100Mbps (GXP1780), 10/100/1000Mbps (GXP1782)
Graphic Display
200*80 pixel backlit LCD display
Features Keys
8 line keys with up to 4 SIP accounts, 4 XML programmable context sensitive
softkeys, 5 navigation/menu keys, 11 dedicated function keys for : MESSAGE(with
LED indicator), PHONEBOOK, TRANSFER, CONFERENCE, HOLD, HEADSET,
MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-
Voice Codecs
Support for G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723, ILBC, in-band and out-of-band DTMF(in audio, RFC2833, SIP INFO)
Auxiliary Ports
RJ9 headset jack (allowing EHS with Plantronics headsets),
Telephony Features
Hold, transfer, forward, 5-way conference, call park, call pickup, shared-call-appearance(SCA)/bridged-line-appearance(BLA), downloadable
phonebook(XML, LDAP, up to 2000 items), call waiting, call log(up to 500 records),
XML customization of screen, off-hook auto dial, auto answer, click-to-dial, flexible
dial plan, hot desking, personalized music ringtones and music on hold, server
redundancy and fail-over
HD Audio
Yes, HD handset and speakerphone with support for wideband audio
Base Stand
Yes, 2 angle positions available, Wall Mountable(optional)
QoS
Layer 2 QoS (802.1Q, 802.1P) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security
User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control
Multi-Language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
Chinese, Korean, Japanese
Upgrade/Provisioning
Firmware upgrade via TFTP / HTTP / HTTPS, mass provisioning using TR-069 or AES encrypted XML configuration file.
Power & Green
Energy Efficiency
Universal power adapter included: Input:100-240V; Output: +5V, 1A ; Integrated Power-over-Ethernet(802.3af) Max power consumption : 5W
Physical
Dimension: 231mm(W) x 167mm(L) x 86mm(H) Unit weight: 0.925kg. Package weight: 1.55kg
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Temperature and
Humidity
Operation: 0°C to 40°CStorage: -10°C to 60°C Humidity: 10％ to 90% Non-condensing
Package Content
GXP1780/GXP1782 phone, handset with cord, base stand, universal power
supply, network cable, Quick Installation Guide, GPL license
Compliance
FCC: Part 15 (CFR 47) Class B CE: EN55022 Class B; EN55024 Class B; EN61000-3-2; EN61000-3-3;
EN60950-1
RCM: AS/ACIF S004; AS/NZS CISPR22/24; ASs/NZS 60950.1
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Call History
Displays Local call logs:
All Calls/Answered Calls/Dialed Calls/Missed Calls/Transferred Calls
Status
Displays account status, network status, software version number and Hardware
Account status. Network status.
Press to enter the sub menu for MAC address, IP setting information
(DHCP/Static IP/PPPoE), IPv4 address, IPv6 address, Subnet Mask, Gateway
and DNS server.
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System Information
Press to enter the sub menu for Hardware version, P/N number. Boot, Core,
Base, Prog version and IP Geographic Information.
Contacts
Contacts sub menu includes the following options:
Local Phonebook Local Group LDAP Directory
Contacts sub menu is for Local Phonebook, Local Group, LDAP Directory and
Broadsoft Phonebooks. User could configure phonebooks/groups/LDAP options
here, download phonebook XML to the phone and search phonebook/LDAP
directory.
Messages
Message sub menu include the following options:
Instant Message
Displays received instant messages;
Voice Mails
Displays voicemail message information in the format below: new
messages/all messages (urgent messages/all urgent messages).
Preference
Preference sub menu includes the following options: Do Not Disturb
Enables/disables Do Not Disturb on the phone.
Star Key Lock
Turns on/off keypad lock feature and configures keypad lock password. The
default keypad lock password is null. If user enabled Star Key lock without
configuring password, user can unlock keypad by holding * key 4 seconds and
pressing “OK” button.
Sounds
o Ring Tone
Configures different ring tones for incoming call.
o Ring Volume
Adjusts ring volume by pressing left/right arrow key.
Appearance
o Active LCD Brightness
Adjusts active LCD brightness by pressing left/right arrow key
o Idle LCD Brightness
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Adjusts idle LCD brightness by pressing left/right arrow key
o Screensaver
Enable/Disable Screensaver
o Screensaver Timeout
Configures the minutes of idle before the screensaver activates.
Valid range is 3 to 6.
Language and Input
o Display Language
Selects the language to be displayed on the phone's LCD. Users
could select Automatic for local language based on IP location if
available. By default, it is Auto.
o Default Input Selection
Selects the Input mode from Multi-Tap and Shiftable. By default, it is Multi-Tap.
Multi-Tap: User may tap the same key multiple times to switch to the desired character.
Shiftable: After pressing the number button, user will see the IDs
of the characters that matching to the button. User can select the
desired character by entering the corresponding ID on keypad.
Date Time
o Allow DHCP Option 42 to override NTP server o Allow DHCP Option 2 to override Time Zone setting o Time Settings
It is used to configure date and time on the phone.
Search Mode
Specifies the phonebook search mode to QuickMatch or ExactMatch. By
default, it is QuickMatch.
Phone
Phone sub menu includes the following options: SIP
Configures SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP
Password, SIP Transport and Audio information to register SIP account on the
phone.
Call Features
Configures call forward features for Forward All, Forward Busy, Forward No
Answer and No Answer Timeout.
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System
System sub menu includes the following options: Network
o IP Setting
Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE
account ID and password; Configures static IP address, Netmask,
Gateway, DNS Server 1 and DNS Server 2.
o 802.1X
Enables/Disables 802.1X mode; Configures 802.1x identity and
MD5 password.
o Layer 2 QoS
Configures 802.1Q/VLAN Tag and priority value. Select “Reset
Vlan Config” to reset VLAN configuration.
Upgrade
o Firmware Server
Configures firmware server for upgrading the phone.
o Config Server
Configures config server for provisioning the phone.
o Upgrade Via
Specifies upgrade/provisioning via TFTP/HTTP/HTTPS.
o Start Provison
Starts Provision immediately.
Language Download
o Auto Language Download o Language Download
Factory Functions
o Diagnostic Mode
All LEDs will light up. All keys’ name will display in red on LCD
screen before diagnosing. Press any key on the keypad to
diagnose the key’s function. When done, the key’s name will
display in blue on LCD. Lift and put back the handset to exit
diagnostic mode.
o Audio Loopback
Speak to the phone using speaker/handset/headset. If you can
hear your voice, your audio is working fine. Press “Exit” soft key to
exit audio loopback mode.
o LCD on/off
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Selects this option to turn off LCD. Press any button to turn on
LCD.
o LCD Diagnostic
Enters this option and press Left/Right Navigation key to do LCD
Diagnostic. Press “Exit” soft key to quite.
o Certificate Verification
This is used to validate certificate chain for the server’s certificate.
UCM Detect
Detect/connect UCM server to process auto-provision. Manually input the
IP and port of the UCM server phone wants to bind with; Or select from the
available UCM server in network.
Operations
o Factory Reset
It is used to restore the phone to factory default settings.
Reboot
Reboots the phone.
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Call History
Reboot
Account Status
System Status
SIP Call Features
Network
Factory Reset
Local Phonebook
Broadsoft Phonebook
Do Not Disturb
Star key Lock
Instant Messages Voice Mails
Ring Tone Ring Volume
Enable/Disable
Password
Active LCD Brightness
Screensaver Timeout
Firmware Server
Start Provision
Internet Protocol
Diagnostic Mode
Certificate Verification
Local Call Log Broadsoft Call Log
Display Language Default Input Selection
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User Level
Password
Web Pages Allowed
End User Level
123
Only Status and Maintenance
Administrator Level
admin
All pages
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Status Account Status
Account
Account index. For GXP1760: up to 3 SIP accounts For GXP1780: up to 4 SIP accounts For GXP1782: up to 4 SIP accounts
SIP User ID
Displays the configured SIP User ID for the account.
SIP Server
Displays the configured SIP Server address, URL or IP address, and port of
the SIP server.
SIP Registration
Displays SIP registration status for the SIP account, it will display Yes/No with
Green/Red background.
Status Network Status
MAC Address
Global unique ID of device, in HEX format. The MAC address will be used for
provisioning and can be found on the label coming with original box and on the
label located on the back of the device.
IP Setting
Configured address type: DHCP, Static IP or PPPoE.
IPv4 Address
Displays the IPv4 address obtained on the phone.
IPv6 Address
Displays the IPv6 address obtained on the phone.
Subnet Mask
Displays the subnet mask obtained on the phone.
Gateway
Displays the gateway address obtained on the phone.
DNS Server 1
Displays the DNS server address 1 obtained on the phone.
DNS Server 2
Displays the DNS server address 2 obtained on the phone.
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PPPoE Link Up
PPPoE connection status.
NAT Type
Displays the type of NAT connection used by the phone.
NAT Traversal
Displays the status of NAT connection for each account on the phone.
Status System Info
Product Model
Product model of the phone.
Part Number
Product part number.
Software Version
Boot: boot version number; Core: core version number; Base: base version number; Prog: program version number. This is the main firmware release number,
which is always used for identifying the software system of the phone;
Locale: locale version number; Recovery: recovery version number.
IP Geographic
Information
City: displaying phone location; Language: displaying language; Time Zone: displaying time zone;
System Up Time
System up time since the last reboot.
System Time
Current system time on the phone system.
Service Status
GUI and Phone service status.
Core Dump
Core dump file that could be downloaded for troubleshooting purpose.
Account x General Settings
Account Active
Indicates whether the account is active. The default setting is "Yes".
Account Name
Configures the name associated with each account to be displayed on the
LCD.
SIP Server
Specifies the URL or IP address, and port of the SIP server. This should be
provided by VoIP service providers (ITSP).
Secondary SIP Server
Specifies the URL or IP address, and port of the SIP server. This will be used
when the primary SIP server fails.
Outbound Proxy
Configures the IP address or the domain name of the primary outbound
proxy, media gateway or session border controller. It's used by the phone for
firewall or NAT penetration in different network environments. If a symmetric
NAT is detected, STUN will not work and only an outbound proxy can provide
a solution
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BLF Server
Configures the optional server used for SUBSCRIBE requests to indicate other extensions status on the SIP server.
SIP User ID
Configures user account information provided by your VoIP service provider (ITSP). It's usually in the form of digits similar to phone number or actually a phone number.
Authenticate ID
Configures the SIP service subscriber's Authenticate ID used for authentication. It can be identical to or different from the SIP User ID.
Authenticate Password
Configures the account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered. After it is saved, this will appear as hidden for security purpose.
Name
Specifies SIP server subscriber's name (optional) that will be used for Caller ID display.
Voice Mail User ID
Sets if the phone system allows users to access the voice messages by pressing the MESSAGE key on the phone. This ID is usually the VM portal access number. For example, in UCM6100 IPPBX, *97 could be used.
Show Account Name
Only
Enables to display the account name; if it is checked, the user ID will be hidden and the account name will be displayed only.
Account x Network Settings
DNS Mode
Defines which parameter will control how the Search Appliance looks up IP addresses for hostnames. There are four modes: A Record, SRV, NATPTR/SRV, Use Configured IP. The default setting is "A Record". If the user wishes to locate the server by DNS SRV, the user may select "SRV" or "NATPTR/SRV". If "Use Configured IP" is selected, please fill in the three fields below:
Primary IP:  Backup IP 1;  Backup IP 2.
If SIP server is configured as domain name, phone will not send DNS query,
but use “Primary IP” or “Backup IP x” to send SIP message if at least one of
them are not empty. Phone will try to use “Primary IP” first. After 3 tries
without any response, it will switch to “Backup IP x”, and then it will switch
back to “Primary IP” after 3 re-tries. If SIP server is already an IP address, phone will use it directly even “User Configured IP” is selected.
NAT Traversal
Specifies which NAT traversal mechanism will be enabled on the phone system. Users could select the mechanism from No, STUN, Keep-alive, UPnP, Auto or VPN. The default setting is "No".
If set to "STUN" and STUN server is configured, the phone will route
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according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages. The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be "Keep-alive". Configure this to be "No" if an outbound proxy is used. "STUN" cannot be used if the detected NAT is symmetric NAT. Set this to "VPN" if OpenVPN is used.
Proxy-Require
Determines a SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall. Do not configure this parameter unless this feature is supported on the SIP server.
Account x SIP Settings Basic Settings
TEL URI
Determines if the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is "Disable".
SIP Registration
Selects whether or not the phone will send SIP Register messages to the proxy/server. The default setting is "Yes".
Unregister On Reboot
If set to "Yes", the SIP user's registration information will be cleared when the phone reboots. The SIP Contact header will contain "*" to notify the server to unbind the connection. The default setting is "No".
Register Expiration
Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar. The default value is 60 minutes. The maximum value is 64800 minutes (about 45 days).
Reregister Before
Expiration
Specifies the time frequency (in seconds) that the phone sends re-registration request before the Register Expiration. The default value is 0.
Local SIP Port
Defines the local SIP port used to listen and transmit. The default value is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066 for Account 4, 5068 for Account 5, 5070 for Account 6. The valid range is from 1 to 65535.
SIP Registration Failure
Retry Wait Time
Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600. The default value is 20 seconds.
SIP T1 Timeout
SIP T1 Timeout is an estimate of the round trip time of transactions between a client and server. If no response is received the timeout is increased, and request re-transmit retries would continue until a maximum amount of time define by T2. The default setting is 0.5 seconds.
SIP T2 Timeout
SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. The default setting is 4 seconds.
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SIP Transport
Determines the network protocol used for the SIP transport. Users can choose from TCP, UDP and TLS. The default setting is “UDP”.
SIP URI Scheme when
using TLS
Specifies if "sip" or "sips" will be used when TLS/TCP is selected for SIP Transport. The default setting is "sips".
Use Actual Ephemeral
Port in Contact with
TCP/TLS
This option is used to control the port information in the Via header and
Contact header. If set to No, these port numbers will use the permanent
listening port on the phone. Otherwise, they will use the ephemeral port for the
particular connection. The default setting is “No”.
Remove OBP from route
Configures to remove outbound proxy from route. This is used for the SIP Extension to notify the SIP server that the device is behind a NAT/Firewall. The default setting is "No".
Support SIP Instance ID
Defines whether SIP Instance ID is supported or not. The default setting is "Yes".
SUBSCRIBE for MWI
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be sent periodically. The phone supports synchronized and non-synchronized MWI. The default setting is "No".
SUBSCRIBE for
Registration
When set to "Yes", a SUBSCRIBE for Registration will be sent out periodically. The default setting is "No".
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is very important in order to support PSTN internetworking. To invoke a reliable provisional response, the 100rel tag is appended to the value of the required header of the initial signaling messages. The default setting is "No".
Caller ID Display
When set to "Auto", the phone will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE. When set to "Disabled", all incoming calls are displayed with "Unavailable". When set to "From Header", the phone will display the caller ID based on the From Header in the incoming SIP INVITE. The default setting is "Auto".
Use Privacy Header
Controls whether the Privacy header will present in the SIP INVITE message or not, whether the header contains the caller info. When set to "Default", the Privacy Header will show in INVITE only when "Huawei IMS" special feature is on. If set to "Yes", the Privacy Header will always show in INVITE. If set to "No", the Privacy Header will not show in INVITE. The default setting is "Default".
Use P-Preferred-Identity Header
Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message. The default setting is "default": the P-Preferred-Identity Header will show in INVITE unless "Huawei IMS" special feature is on. If set to "Yes", the P-Preferred-Identity Header will always show in INVITE. If set to
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"No", the P-Preferred-Identity Header will not show in INVITE.
Add Auth Header on
Initial REGISTER
To define whether authorization Header will be added on initial REGISTER from the first REGISTER. The default setting is “No”.
Allow SIP Reset
This is used to perform a factory reset through SIP NOTIFY. When the phone receives the NOTIFY with event: RESET, the phone should perform a factory reset after the authentication. The default setting is “No”.
Account x SIP Settings Advanced Features
Line Seize Timeout
For Shared Call Appearance, phone must send a SUBSCRIBE-request for the line-seize event package whenever a user attempts to take the shared line off hook. “Line Seize Timeout” is the line-seize event expiration timer. The default value is 15 seconds. The valid range is from 15 to 60.
Eventlist BLF URI
Configures the eventlist BLF URI on the phone to monitor the extensions in the list with Multi-Purpose Key. If the server supports this feature, users need to configure an eventlist BLF URI on the service side first (i.e., BLF1006@myserver.com) with a list of extension included. On the phone, in this "eventlist BLF URI" field, fill in the URI without the domain (i.e., BLF1006). To monitor the extensions in the list, under Web GUI->Settings->Programmable Keys page, please select "eventlist BLF" in the key mode, choose account, enter the value of each extension in the list.
Auto Provision Eventlist
BLFs
When option is enabled, empty multi-purpose keys will be automatically provisioned to the monitored extensions in the Eventlist BLF. The default setting is “Disabled”.
Conference URI
Configures Conference URI for N-way conference (Broadsoft Standard).
Music On Hold URI
Configures Music On Hold URI to call when a call is on hold. This feature has to be supported on the server side.
Force BLF Call-pickup by prefix
Configures to always use the prefix for BLF Call-pickup. The default setting is “No”.
BLF Call-pickup Prefix
Configures the prefix prepended to the BLF extension when the phone picks up a call with BLF key. The default setting is **.
PUBLISH for Presence
Enables presence feature on the phone. The default setting is "No".
Omit charset=UTF-8 in MESSAGE
Omit charset=UTF-8 in MESSAGE content-type
Special Feature
Different soft switch vendors have special requirements. Therefore, users
may need select special features to meet these requirements. Users can
choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro,
Huawei IMS and PhonePower depending on the server type. The default
setting is "Standard".
Broadsoft Call Center
Default setting is "No". When set to "Yes", a soft key “BSCCenter” is
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displayed on LCD. User can access different Broadsoft Call Center agent
features via this softkey. Please note that “Feature Key Synchronization” will
be enabled regardless of this setting.
Hoteling Event
Broadsoft Hoteling event feature. Default setting is "No". With “Hoteling
Event” enabled, user can access the Hoteling feature option by pressing the
“BSCCenter” sof tkey.
Call Center Status
When set to "Yes", the phone will send SUBSCRIBE to the server to obtain
call center status. The default setting is "No".
Broadsoft Executive
Assistant
When enabled, Feature Key Synchronization will be enabled regardless of
web settings.
Feature Key
Synchronization
This feature is used for Broadsoft call feature synchronization. When it's
enabled, DND, Call Forward features and Call Center Agent status can be
synchronized between Broadsoft server and phone. The default setting is
"Disabled".
Broadsoft Call Park
When enabled, it will send SUBSCRIBE to Broadsoft server to obtain Call
Park notifications. The default setting is “Disabled”.
Account x SIP Settings Session Timer
Session Expiration
The SIP Session Timer extension (in seconds) that enables SIP sessions to be periodically "refreshed" via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800.
Min-SE
The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from 90 to 64800.
Caller Request Timer
If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. The default setting is "No".
Callee Request Timer
If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it receives inbound calls. The default setting is "No".
Force Timer
If Force Timer is set to "Yes", the phone will use the session timer even if the remote party does not support this feature. If Force Timer is set to "No", the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select "No". The default setting is "No".
UAC Specify Refresher
As a Caller, select UAC to use the phone as the refresher; or select UAS to use the Callee or proxy server as the refresher. The default setting is "Omit".
UAS Specify Refresher
As a Callee, select UAC to use caller or proxy server as the refresher; or
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select UAS to use the phone as the refresher. The default setting is "UAC".
Force INVITE
The Session Timer can be refreshed using the INVITE method or the UPDATE method. Select "Yes" to use the INVITE method to refresh the session timer. The default setting is "No".
Account x SIP Settings Security Settings
Check Domain
Certificates
Choose whether the domain certificates will be checked or not when TLS/TCP is used for SIP Transport. The default setting is "No".
Validate Certificate
Chain
Validate certification chain when TCP/TLS is configured. The default setting is “No”.
Validate Incoming
Messages
Choose whether the incoming messages will be validated or not. The default setting is "No".
Check SIP User ID for
incoming INVITE
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
Accept Incoming SIP
from Proxy Only
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
Authenticate Incoming
INVITE
If set to "Yes", the phone will challenge the incoming INVITE for
authentication with SIP 401 Unauthorized response. The default setting is
"No".
Account x Audio Settings
Send DTMF
This parameter specifies the mechanism to transmit DTMF digits. There are 3 supported modes: in audio which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs), via RTP (RFC2833), or via SIP INFO.
• In audio, which means DTMF is combined in the audio signal (not very
reliable with low-bit-rate codecs);
• RFC2833, which means to specify DTMF with RTP packet. Users could know the packet is DTMF in the RTP header as well as the type of DTMF;
• SIP INFO, which use SIP info to carry DTMF. The defect of this mode is that it's easily to cause desynchronized of DTMF and media packet for the reason the SIP and RTP are transmitted respectively. The default setting is "RFC2833".
DTMF Payload Type
This parameter sets the payload type for DTMF using RFC2833. Default is
101. The valid range is from 96 to 127.
Preferred Vocoder
Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference. Users can configure vocoders in a preference list that is included with the same preference order in SDP message.
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Use First Matching
Vocoder in 200OK SDP
When it is set to "Yes", the device will use the first matching vocoder in the received 200OK SDP as the codec. The default setting is "No".
Disable Multiple m line
in SDP
When it is set to “No”, the device will reply with multiple m lines; Otherwise, it will reply 1 m line. The default setting is “No”.
SRTP Mode
Enable SRTP mode based on your selection from the drop-down menu. The default setting is "Disabled".
Crypto Life Time
Enable or disable the crypto life time when using SRTP. If users set to disable this option, phone does not add the crypto life time to SRTP header. The default setting is “Yes”.
Symmetric RTP
Defines whether symmetric RTP is supported or not. The default setting is "No".
Silence Suppression
Controls the silence suppression/VAD feature of the audio codec 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. The default setting is "No".
Voice Frames Per TX
Configures the number of voice frames transmitted per packet. When configuring this, it should be noted that the "ptime" value for the SDP will change with different configurations here. This value is related to the codec used and the actual frames transmitted during the in payload call. For end users, it is recommended to use the default setting, as incorrect settings may influence the audio quality. The default setting is 2.
G.726-32 Packing Mode
Selects "ITU" or "IETF" for G726-32 packing mode. The default setting is “ITU”.
Jitter Buffer Type
Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting is "Adaptive".
Jitter Buffer Length
Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default setting is "300ms".
Account x Call Settings
Early Dial
Selects whether or not to enable early dial. If it's set to "Yes", the SIP proxy must support 484 responses. Early Dial means that the phone sends for each pressed digit a SIP INVITE message to SIP server. SIP server looks into its extensions and, if no match happened yet, it sends back a "484 Address Incomplete" message. Otherwise, it executes the action. The default setting is "No".
Dial Plan Prefix
Configures the prefix to be added to each dialed number.
Dial Plan
A dial plan establishes the expected number and pattern of digits for a telephone number. This parameter configures the allowed dial plan for the phone.
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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.
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 - allows 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.
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Call Log
Configures Call Log setting on the phone. You can log all calls, only log
incoming/outgoing calls (missed calls will not be logged), or disable call log.
The default setting is "Log All Calls".
Account Ring Tone
Allows users to configure the ringtone for the account. Users can choose from
different ringtones from the dropdown menu.
Match Incoming Caller
ID
Specifies matching rules with number, pattern or Alert Info text. When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone. Matching rules:
Specific caller ID number. For example, 8321123;  A defined pattern with certain length using x and + to specify, where x
could be any digit from 0 to 9. Samples:
xx+ : at least 2-digit number; xx : only 2-digit number;
[345]xx: 3-digit number with the leading digit of 3, 4 or 5; [6-9]xx: 3-digit number with the leading digit from 6 to 9.
Alert Info text
Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: <http://127.0.0.1>; info=priority
Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring.
Ring Timeout
Defines the timeout (in seconds) for the rings on no answer. The default
setting is 60. The valid range is from 10 to 300.
Send Anonymous
If set to "Yes", the "From" header in outgoing INVITE messages will be set to
anonymous, essentially blocking the Caller ID to be displayed. The default
setting is "No".
Anonymous Call
Rejection
If set to "Yes", anonymous calls will be rejected. The default setting is "No".
Auto Answer
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep. Default setting is “No”.
Allow Auto Answer by
Call-Info
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls, based on the SIP info header sent from the
server/proxy. The default setting is "No".
Allow Barging by
Call-Info
When it is enabled, the phone will put the current call on hold automatically
and answer the incoming calls based on the SIP Call-Info header sent from
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the server/proxy. However, if the current call was answered based on the SIP
Call-Info header, then all other incoming calls with SIP Call-Info headers will be rejected automatically. The default setting is “No”.
Custom Call-Info for Auto Answer
Used in addition to match the contents of the info parameter in the Call-Info header for auto answer.
Refer-To Use Target Contact
If set to "Yes", the "Refer-To" header uses the transferred target's Contact header information for attended transfer. The default setting is "No".
Transfer on Conference
Hangup
If set to "Yes", when the phone hangs up as the conference initiator, the
conference call will be transferred to the other parties so that other parties will
remain in the conference call. The default setting is "No".
Disable Recovery on
Blind Transfer
Disable recovery to the call to the transferee on failing blind transfer to the
target. The default setting is “No”.
No Key Entry Timeout
(s)
Defines the timeout (in seconds) for no key entry. If no key is pressed after
the timeout, the digits will be sent out. The default value is 4 seconds. The
valid range is from 1 to 15.
Use # as Dial Key
Allows users to configure the "#" key as the "Send" key. If set to "Yes", the "#"
key will immediately dial out the input digits. In this case, this key is
essentially equivalent to the "Send" key. If set to "No", the "#" key is included
as part of the dialing string and please make sure the dial plan is properly
configured to allow dialing # out. The default setting is “Yes”.
On Hold Reminder Tone
If set to “Enabled”, phone will play a reminder tone when it has a call on hold.
The default setting is “Disabled”.
Account x Feature Codes
Enable Local Call
Features
When enabled, Do No Disturb, Call Forwarding and other call features can be
used via the local feature codes on the phone. Otherwise, the provisioned
feature codes from the server will be used. User configured feature codes will
be used only if server provisioned feature codes are not provided. And once
feature codes are configured, either via server provisioning or local setting, a
softkey named “Features” will show on the LCD screen.
Do Not Disturb
(DND)--On
Configures DND feature code to turn on DND.
Do Not Disturb
(DND)--Off
Configures DND feature code to turn off DND.
Call Forward
Unconditionally
(All)--On
Configures Call Forward All feature code to activate unconditional call
forwarding.
Call Forward
Unconditionally
Configures Call Forward All feature code to deactivate unconditional call
forwarding
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(All)--Off
Call Forward Busy--On
Configures Call Forward Busy feature code to activate busy call forwarding.
Call Forward Busy--Off
Configures Call Forward Busy feature code to deactivate busy call
forwarding.
Call Forward Delayed
(No Answer)--On
Configures Call Forward Delayed feature code to activate no answer call
forwarding.
Call Forward Delayed
(No Answer)--Off
Configures Call Forward Delayed feature code to activate no answer call
forwarding.
Delayed Call Forward
Wait Time
Defines the timeout (in seconds) before the call is forwarded on no answer.
The default value is 20 seconds. The valid range is 1 to 120.
Settings General Settings
Local RTP Port
This parameter defines the local RTP port 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
When set to "Yes", this parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is "Yes"
Note: This parameter must be set to "No" for Direct IP Calling to work.
Keep-alive Interval
Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep the "ping hole" on the NAT router to open. The default setting is 20 seconds. The valid range is from 10 to 160.
Use NAT IP
The NAT IP address used in SIP/SDP messages. This field is blank at the default settings. It should ONLY be used if it's required by your ITSP.
STUN Server
The IP address or Domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN.
Public Mode
Configures to turn on/off the public mode for hot desking feature. The default setting is “No”.
Settings Call Features
Off-hook Auto Dial
Configures a User ID/extension to dial automatically when the phone is
off hook. The phone will use the first account to dial out. The default
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setting is "No".
Off-hook Timeout
If configured, when the phone is on hook, it will go off hook after the
timeout (in seconds). The default value is 30 seconds. The valid range is
from 10 to 60.
Bypass Dial Plan Through
Call History and Directories
Enable/Disable the dial plan check while dialing through the call history
and any phonebook directories. The default setting is “No”.
Disable Call Waiting
Disables the call waiting feature. The default setting is "No".
Disable Call Waiting Tone
Disables the call waiting tone when call waiting is on. The default setting
is "No".
Disable Busy Tone on
Remote Disconnect
Disable the busy tone heard in the handset when call is disconnected
remotely. The default setting is “No”.
Disable Direct IP Call
Disables Direct IP Call. The default setting is "No".
Use Quick IP Call mode
When set to "Yes", users can dial an IP address under the same
LAN/VPN segment by entering the last octet in the IP address. To dial
quick IP call, off hook the phone and dial #XXX (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). No
SIP server is required to make quick IP call. The default setting is "No".
Disable Conference
Disables the Conference function. The default setting is "No".
Disable in-call DTMF Display
When it's set to "Yes", the DTMF digits entered during the call will not be displayed on phone LCD. The default setting is "No".
Enable sending DTMF via
specific MPKs
Enables certain MPKs to send DTMF during the call. This option does
not affect Dial DTMF. The default setting is "No".
Disable Active MPK Page
When option is enabled, Active MPK Page on the extension boards will
be disabled. The default setting is "No".
Mute Key Functions While
Idle
Specifies the function of mute key in idle. Default setting is “DND”. When
select “Idle Mute” and press Mute key while idle, the future incoming call
will be answered with mute. When select “Disabled”, Mute key will not
take effect while idle. The default setting is “No”.
Disable Transfer
Disables the Transfer function. The default setting is "No".
In-call dial number on pressing transfer key
Configures the number for the phone to dial as DTMF during the call
using TRAN button.
Auto-Attended Transfer
If set to "Yes", the phone will use attended transfer by default. The default
setting is "No".
Do Not Escape # as %23 in
SIP URI
Specifies whether to replace # by %23 or not for some special situations.
The default setting is "No".
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Click-To-Dial Feature
Enables Click-To-Dial feature. If this feature is enabled, user could click
the green dial button on left top corner of phone’s Web GUI, then choose
the account and dial to the target number. The default setting is
"Disabled".
Call History Flash Writing: Write Timeout
Defines the interval (in seconds) to save the call history to phone's flash. 0 means this option is disabled. The default value is 300 seconds.
Call History Flash Writing: Max Unsaved Log
Defines the number of unsaved logs before written to phone's flash. 0 means this option is disabled. The default value is 200 entries.
Default call log type
This option is used for users to set the default call log list after select MENUCALL HISTORY. Broadsoft Call Log or Local Call Log option will only show its own list. Default option will keep both call log lists.
Local Call Recording
Feature
Enables/Disables the ability to record calls locally while on the call screen. The default setting is "Disabled".
Saved Local Call Recording
Location
This option is used for users to set the location where the recordings will be stored.
Download Local Call
Recordings
When there are recordings presented, you may download them here.
Settings Multicast paging
Paging Barge
During active call if incoming multicast page is higher priority (1 being the highest) than this value the call will be held and multicast page will be played. The default setting is “Disabled”.
Paging Priority Active
If enabled, during a multicast page if another multicast is received with higher priority (1 being the highest) that one will be played instead. The default setting is “Disabled”.
Multicast Paging Codec
The codec for sending multicast pages, there are 5 codecs could be used: PCMU, PCMA, G.726-32, G.729A/B, G.722 (wide band). The default setting is “PCMU”.
Multicast Listening
Defines multicast listening addresses and labels. For example:
“Listening Address” should match the sender’s Value such as “237.11.10.11:6767” “Label” could be the description you want to use.
For details, please check the “Multicast Paging User Guide” on our Website.
Settings Ring Tone
Call Progresses Tones: System Ring Tone Dial Tone Second Dial Tone
Configures ring or tone frequencies based on parameters from local telecom. The default value is North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.
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Message Waiting Ring Back Tone Call-Waiting Tone Busy Tone Reorder Tone
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.
Call Waiting Tone Gain
This adjusts the call waiting tone volume. Users can select "Low",
"Medium" or "High". The default setting is "Low".
Settings Audio Control
Headset Key Mode
When headset is connected to the phone, users could use the HEADSET button in "Default Mode" or "Toggle Headset/Speaker". Default Mode:
When the phone is in idle, press HEADSET button to off hook
the phone and make calls by using headset. Headset icon will display on the screen in dialing/talking status.
When there is an incoming call, press HEADSET button to pick
up the call using headset.
When there is an active call using headset, press HEADSET
button to hang up the call.
When Speaker/Handset is being used in dialing/talking status,
press EADSET button to switch to headset. Press it again to hang up the call. Or press speaker/Handset to switch back to the previous mode.
Toggle Headset/Speaker:
When the phone is in idle, press HEADSET button to switch to
Headset mode. The headset icon will display on the left side of the screen. In this mode, if pressing Speaker button or Line key to off hook the phone, headset will be used.
When there is an active call, press HEADSET button to toggle
between Headset and Speaker.
Headset Type
Selects whether the connected headset is normal RJ11 headset, Plantronics EHS headset.
Always Ring Speaker
Configures to enable or disable the speaker to ring when headset is used on "Toggle Headset/Speaker" mode. If set to "Yes", when the phone is in Headset "Toggle Headset/Speaker" mode, both headset and speaker will ring on incoming call. The default setting is "No".
Headset TX gain
Configures the transmission gain of the headset. The default value is 0dB.
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Headset RX gain
Configures the receiving gain of the headset. The default value is 0dB.
Handset TX gain
Configures the transmission gain of the handset. The default value is 0
dB.
Settings LCD Display
Backlight Brightness: Active
Configures the LCD brightness when the phone is active. Valid range is 10 to 100 where 100 is the brightest. Default value is 100.
Backlight Brightness: Idle
Configures the LCD brightness when the phone is idle. Valid range is 10 to 100 where 0 is off and 100 is the brightest. Default value is 60.
Disable Missed Call
Backlight
If set to "Yes", the screen will turn off the LCD backlight when there is a missed call on the phone. The default setting is "No".
Wallpaper Source
Specifies the wallpaper source mode: Default, Download, USB and Uploaded. User could upload a wallpaper source into your phone, or download it from file server with the server path, or plug your USB drive with wallpaper source into ph to upload the wallpaper.
Wallpaper Server Path
Specifies the wallpaper server path. This option will take effect when wallpaper source is “Download”.
Upload Wallpaper
Click on the “Upload” button to browse and upload the desired wallpaper file. This option will take effect when wallpaper source is “Uploaded”.
Screensaver
Disable/Enable Screensaver. By default, screensaver is enabled.
Screensaver Source
This option is used for users to set the location where screensaver is loaded from. If from USB, please have a folder named “screensavers” containing your pictures.
Screensaver Timeout
Configures the minutes of idle before the screensaver activates. Valid range is 3 to 6. The default time is 3 minutes.
Settings LCD Control
BLF LED Pattern
This is used to configure the color and pattern of the LED based on status updates. The default setting is “Default”. The BLF LED Patterns are listed in the following Table 13.
Settings Date and Time
NTP Server
Defines the URL or IP address of the NTP server. The phone may obtain the date and time from the server. The default setting is us.pool.ntp.org.
NTP Update Interval
Time interval for updating time from the NTP server. Valid time value is in between 5 to 1440 minutes. The default setting is “1440” minutes.
Allow DHCP Option 42
Override NTP Server
Defines whether DHCP Option 42 should override NTP server or not. When enabled, DHCP Option 42 will override the NTP server if it's set up on the LAN. The default setting is "Yes".
Time Zone
Configures the date/time used on the phone according to the specified
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time zone.
Self-Defined Time Zone
This parameter allows the users to define their own time zone. The syntax is: std offset dst [offset], start [/time], end [/time] Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0 MTZ+6MDT+5 This indicates a time zone with 6 hours offset with 1 hour ahead (when daylight saving) 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)
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3
rd
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.
Date Display Format
Configures the date display format on the LCD. The following formats are supported:
yyyy-mm-dd: 2012-07-02  mm-dd-yyyy: 07-02-2012  dd-mm-yyyy: 02-07-2012  dddd, MMMM dd: Friday, October 12  MMMM dd, dddd: October 12, Friday
The default setting is yyyy-mm-dd.
Time Display Format
Configures the time display in 12-hour or 24-hour format on the LCD. The default setting is in 12-hour format.
Settings Web Service
Enable Weather Update
Configures to enable or disable weather update on the phone. The default setting is "Yes". If set to "No", the weather information screen will not show.
City Code
Configures weather city code for the phone to look up the weather information. The default setting is "Automatic" and the weather information will be obtained based on the IP location of the phone if available. Otherwise, specify the self-defined city code. For example, USCA0638 is the city code for Los Angeles, CA, United States.
Update Interval
Specifies the weather update interval (in minutes). The default value is 15 minutes.
Degree Unit
Specifies the degree unit for the weather information to display on the
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phone. User could chose Fahrenheit, Celsius, or Auto to display the degree unit. The default setting is “Auto”.
Enable Currency Update
Configures to enable or disable currency update on the phone. The default setting is "Yes". If set to "No", the currency information screen will not show.
Currency Code
Configures currency code for the phone to look up and display the currency information. The default setting is: “EUR/USD;GBP/USD;CAD/USD;AUD/USD;CNY/USD;JPY/USD”
Settings XML Application
Server Path
Configures the server path to download the idle screen XML file. This field could be IP address or URL, with up to 256 characters.
Softkey Label
Specifies the softkey name displayed on the idle screen for the users to enter XML application. The default Softkey Label is “XMLApp”.
Settings Programmable Keys
Line Key X
Assigns a function to the corresponding line key. The key mode options are: Line
Regular line key to open up a line and switch line. The Value field can be left blank.
Shared Line
Share line for Shared Line Appearance feature. Select the Account registered as Shared line for the line key. The Value field can be left blank.
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed, or enter the IP address to set the Direct IP call as Speed Dial.
Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension number in the Value field to be monitored.
Presence Watcher
This option has to be supported by a presence server and it is tied to the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message. PBX server has to support this feature.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active
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account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the Value field to be dialed during the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value field.
Call Return
The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.
Transfer
Select Account, and enter the number in the Value field to be transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the Value field to park/pick up the call.
Monitored Call Park
Select account from Account field, and enter the call park extension in the Value field to park/pick up the call, and also monitor the parked call via Line Key’s light.
Intercom
Select Account, and enter the extension number in the Value field to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li
Since the Base for LDAP server configuration is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Multicast Paging
This option is for multicast sending. Enter Line key description in
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Description field and multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in Description filed and the recorded extension number in Value field. Please make sure whether your VOIP provider supports this feature before using it.
Call Log
Select Account and enter account number in the Value field to allow configuration of call log for other extension.
Softkeys
Assigns a function to the corresponding softkeys.
The key mode options are: Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2.
Voice Mail
Select Account and enter the Voice Mail access number in the Value field.
Call Return
The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.
Intercom
Select Account, and enter the extension number in the Value field to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li
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Since the Base for LDAP server configuration is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Call Log
Select Account and enter account number in the Value field to access to the Call Log of that selected account.
Multi-Purpose Key
Assigns a function to the corresponding Multi-Purpose Key. The key mode options are: Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed.
BLF (Busy Lamp Field)
This option has to be supported on the PBX and it indicates the status of the extension. The three possible states are idle (green), busy (red), ringing (blinking red).
Presence Watcher
This option has to be supported by a presence server and it is tied to the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message. PBX server has to support this feature.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the Value field to be dialed during the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value field.
Call Return
The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.
Transfer
Select Account, and enter the number in the Value field to be transferred (blind transfer) during the call.
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Call Park
Select Account, and enter the call park extension in the Value field to park/pick up the call.
Monitored Call Park
Select account from Account field, and enter the call park extension in the Value field to park/pick up the call, and also monitor the parked call via MPK button’s light.
Intercom
Select Account, and enter the extension number in the Value field to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li
Since the Base for LDAP server configuration is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Multicast Paging
This option is for multicast sending. Enter Line key description in Description filed and multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in Description filed and the recorded extension number in Value field. Please make sure whether your VOIP provider supports this feature before using it.
Call Log
Select Account and enter account number in the Value field to access to the Call Log of that selected account.
Settings Programmable Keys Programmable Keys
Softkeys
Assigns a function to the corresponding softkeys.
The key mode options are: Speed Dial
Select the Account to dial from. And enter the Speed Dial number in
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the Value field to be dialed.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2.
Voice Mail
Select Account and enter the Voice Mail access number in the Value field.
Call Return
The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.
Intercom
Select Account, and enter the extension number in the Value field to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li
Since the Base for LDAP server configuration is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Call Log
Select Account and enter account number in the Value field to access to the Call Log of that selected account.
Menu
This option is to take users to the main menu screen directly.
Settings Programmable Keys Virtual Multi-Purpose Keys (VPK)
VPK
Assigns a function to the corresponding VPK. The key mode options are: Line
Regular line key to open up a line and switch line. The Value field can be left blank.
Shared Line
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Share line for Shared Line Appearance feature. Select the Account registered as Shared line for the line key. The Value field can be left blank.
Speed Dial
Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed, or enter the IP address to set the Direct IP call as Speed Dial.
Busy Lamp Field (BLF)
Select the Account to monitor the BLF status. Enter the extension number in the Value field to be monitored.
Presence Watcher
This option has to be supported by a presence server and it is tied to the "Do Not Disturb" status of the phone's extension.
Eventlist BLF
This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message. PBX server has to support this feature.
Speed Dial via active account
Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2.
Dial DTMF
Enter a series of DTMF digits in the Value field to be dialed during the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.
Voice Mail
Select Account and enter the Voice Mail access number in the Value field.
Call Return
The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.
Transfer
Select Account, and enter the number in the Value field to be transferred (blind transfer) during the call.
Call Park
Select Account, and enter the call park extension in the Value field to park/pick up the call.
Monitored Call Park
Select account from Account field, and enter the call park extension in the Value field to park/pick up the call, and also monitor the parked
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call via Line Key’s light.
Intercom
Select Account, and enter the extension number in the Value field to do the intercom.
LDAP Search
This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example: If users set MPK 1 as “LDAP Search” for “Account 1”, and set filters:
Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li
Since the Base for LDAP server configuration is
“dc=grandstream,dc=com”, “ou=video,ou=SZ” is added to narrow the
LDAP search scope. “sn=Li” is the example to filter the last name.
Multicast Paging
This option is for multicast sending. Enter Line key description in Description field and multicast sending address in Value field.
Record
This option is for Recording calls. Enter Line key description in Description filed and the recorded extension number in Value field. Please make sure whether your VOIP provider supports this feature before using it.
Call Log
Select Account and enter account number in the Value field to allow configuration of call log for other extension.
Menu
This option is to take users to the main menu screen directly.
Settings Broadsoft Broadsoft Directories
XSI
Configures XSI Directory.
Server
Configure the BroadWorks Xsi server URI. If the server uses HTTPS, please add the header “HTTPS” ahead of the Server URI. For instance, “https://SERVER_URI”.
Port.
Configure the BroadWorks Xsi server port. The default port is 80. If the server uses HTTPS, please configure 443.
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User name.
Configure the Username for the BroadWorks Xsi feature.
Password.
Configure the password for the BroadWorks Xsi feature.
Network Directories
Enable/Disable Broadsoft Network directories and defines the directory name. The directory types are:
Group Directory
Enable/Disable and rename the BroadWorks Xsi Group Direcory features on the phone. If keep the Name box blank, the phone will use the default name “Group” for it.
Enterprise Directory
Enable/Disable and rename the BroadWorks Xsi Enterprise Directory features on the phone. If keep the Name box blank, the phone will use the default name “Enterprise” for it.
Group Common
Enable/Disable and rename the BroadWorks Xsi Group Common Directory features on the phone. If keep the Name box blank, the phone will use the default name “Group Common” for it.
Enterprise Common
Enable/Disable and rename the BroadWorks Xsi Enterprise Common Directory features on the phone. If keep the Name box
blank, the phone will use the default name “Enterprise Common”
for it.
Personal Directory
Enable/Disable and rename the BroadWorks Xsi Personal Directory features on the phone. If keep the Name box blank, the phone will use the default name “Personal” for it.
Missed Call Log
Enable/Disable and rename the BroadWorks Xsi Missed Call Log features on the phone. If keep the Name box blank, the phone will use the default name “Missed” for it.
Placed Call Log
Enable/Disable and rename the BroadWorks Xsi Placed Call Log features on the phone. If keep the Name box blank, the phone will use the default name “Outgoing” for it.
Received Call Log
Enable/Disable and rename the BroadWorks Xsi Placed Call Log features on the phone. If keep the Name box blank, the phone
will use the default name “Incoming” for it.
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Settings Broadsoft Broadsoft IM&P
LogIn Credentials
Server
Broadsoft IM&P server address. Usually not necessary to
configure and can already be found in the Broadsoft IM&P
username.
Port
Port for the Broadsoft IM&P server. Default port is 5222.
Username
Broadsoft IM&P username, not the Broadsoft account username.
Password
Broadsoft IM&P password, not the Broadsoft account password.
Broadsoft IM&P
Enables Broadsoft Instant Message and Presence feature. The default
setting is “Disabled”.
Associated Broadsoft
Account
Specifies the associated account. User could choose each account on
the phone.
Auto Login
Choose to whether or not login to the Broadsoft IM&P account at boot-up. The default setting is “No”.
Display Non XMPP Contacts
Choose whether or not to display non-xmpp contacts associated with the Broadsoft IM&P user. Non-xmpp contacts will not display a presence or status message. The default setting is “No”.
Network Basic Settings
Internet Protocol
Selects Prefer IPv4 or Prefer IPv6. The default setting is “Prefer IPv4”.
IPv4 Address Type
Allows users to configure the appropriate network settings on the phone to obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE". By default, it is set to "DHCP".
DHCP Host name (Option 12)
Specifies the name of the client. This field is optional but may be required by some Internet Service Providers.
DHCP Vendor Class ID (Option 60)
Used by clients and servers to exchange vendor class ID. The default setting is “Grandstream GXP1760” for GXP1760, ”Grandstream GXP1780” for GXP1780, “Grandstream GXP1782” for GXP1780.
PPPoE Account ID
Enter the PPPoE account ID.
PPPoE Password
Enter the PPPoE Password.
PPPoE Service Name
Enter the PPPoE Service Name.
IPv4 Address
Enter the IP address when static IP is used.
Subnet Mask
Enter the Subnet Mask when static IP is used for IPv4.
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Gateway
Enter the Default Gateway when static IP is used for IPv4.
DNS Server 1
Enter the DNS Server 1 when static IP is used for IPv4.
DNS Server 2
Enter the DNS Server 2 when static IP is used for IPv4.
Preferred DNS Server
Enter the Preferred DNS Server for IPv4.
IPv6 Address Type
Allows users to configure the appropriate network settings on the phone to obtain IPv6 address. Users could select "Auto-configured" or "Statically configured" for the IPv6 address type.
Static IPv6 Address
Enter the static IPv6 address when Full Static is used in "Statically configured" IPv6 address type.
IPv6 Prefix Length
Enter the IPv6 prefix length when Full Static is used in "Statically configured" IPv6 address type.
IPv6 Prefix
Enter the IPv6 Prefix (64 bits) when Prefix Static is used in "Statically configured" IPv6 address type.
DNS Server 1
Enter the DNS Server 1 for IPv6.
DNS Server 2
Enter the DNS Server 2 for IPv6.
Preferred DNS server
Enter the Preferred DNS Server for IPv6.
Network Advanced Settings
802.1X mode
Allows the user to enable/disable 802.1X mode on the phone. The default value is disabled. To enable 802.1X mode, this field should be set to EAP-MD5, users may also choose EAP-TLS, or EAP-PEAP.
802.1X Identity
Enter the Identity information for the 802.1x mode.
MD5 Password
Enter the MD5 Password for the 802.1X mode.
802.1X CA Certificate
Upload 802.1X CA certificate to the phone; or delete existed 802.1X CA certificate from the phone.
802.1X Client
Certificate
Upload 802.1X Client certificate to the phone; or delete existed 802.1X Client certificate from the phone.
HTTP Proxy
Specifies the HTTP proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.
HTTPS Proxy
Specifies the HTTPS proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.
Layer 3 QoS for SIP
Defines the Layer 3 QoS parameter for SIP. This value is used for IP Precedence, Diff-Serv or MPLS. The default value is 26.
Layer 3 QoS for RTP
Defines the Layer 3 QoS parameter for RTP. This value is used for IP Precedence, Diff-Serv or MPLS. The default value is 46.
Layer 2 QoS
802.1Q/VLAN Tag
Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is 0.
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Layer 2 QoS 802.1p Priority Value
Assigns the priority value of the Layer2 QoS packets. The default value is 0.
PC Port Mode
Configures the PC port mode. When set to "Mirrored", the traffic in the LAN port will go through PC port as well and packets can be captured by connecting a PC to the PC port. The default setting is "Enabled".
PC Port VLAN Tag
Assigns the VLAN Tag of the PC port. The default value is “0”.
PC Port Priority Value
Assigns the priority value of the PC port. The default value is “0”.
Enable LLDP
Control the LLDP (Link Layer Discovery Protocol) service. The default setting is “Enabled”.
Maintenance Web Access
User Password
Set new password for web GUI access as User. This field is case sensitive.
Confirm Password
Enter the new User password again to confirm.
Current Admin
Password
The current admin password is required for setting a new admin password.
New Password
Set new password for web GUI access as Admin. This field is case sensitive.
Confirm Password
Enter the new Admin password again to confirm.
Maintenance Upgrade and Provisioning
Firmware Upgrade and Provisioning
Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, Always Skip the Firmware Check. The default setting is “Always Check for New Firmware”.
XML Config File
Password
The password 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.
Always Authenticate
Before Challenge
Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before being challenged by the server. The default setting is “No”.
Upgrade Via
Allows users to choose the firmware upgrade method: TFTP, HTTP or HTTPS. The default setting is “HTTP”.
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Firmware Server Path
Defines the server path for the firmware server. It could be different from the configuration server for provisioning.
Config Server Path
Defines the server path for provisioning. It could be different from the firmware server for upgrading.
Firmware File Prefix
Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone.
Firmware File Postfix
Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the phone.
Config File Prefix
Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the phone.
Config File Postfix
Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.
Allow DHCP Option 43
and Option 66 Override
Server
Default setting is “Yes”. DHCP option 66 originally was only designed for
TFTP server. Later on it was extended to support an HTTP URL. GXP
phones support both TFTP and HTTP server via option 66. Users can also
use DHCP option 43 vendor specific option to do this. DHCP option 43
approach has priorities.
Allow DHCP Option
120 to override SIP
Server
Enables DHCP Option 120 from local server to override the SIP Server on the phone. The default setting is "No".
3CX Auto Provision
Enables automatic provision feature on the phone when 3CX is used as the SIP server. The default setting is "Yes".
Automatic Upgrade
Enables automatic upgrade and provisioning. The default setting is "No".
Hour of the Day (0-23)
Defines the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes. The default value is 1.
Day of the Week (0-6)
Defines the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes. The default value is 1.
Disable SIP NOTIFY
Authentication
Device will not challenge NOTIFY with 401 when set to “Yes”. The default setting is “No”.
Authenticate Conf File
Authenticates configuration file before acceptance. The default setting is "No".
Download Device Configuration
Click to download phone’s configuration file in .txt format.
Upload Device
Upload configuration file to phone.
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Configuration
Maintenance Syslog
Syslog Server
The URL or IP address of the syslog server for the phone to send syslog to.
Syslog Level
Selects the level of logging for syslog. The default setting is "None". There are 4 levels: DEBUG, INFO, WARNING and 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).
Send SIP Log
Configures whether the SIP log will be included in the syslog messages. The default setting is "No".
Auto Recover From Abnormal
If set to “Yes”, the phone will automatically recover when running abnormal. The default setting is “Yes”.
Maintenance Language
Display Language
Selects display language on the phone. There are 21 languages can be set
as display language, user could also choose “Auto” or “Downloaded
Language” as display language. The default setting is “Auto”.
Default Input Selection
Configure the default input selection. Multi-Tap: multi-tap to switch character; Shiftable: select input from available characters. The default setting is “Multi-Tap”.
Auto language
download
This is used to configure the device to download language files
automatically from server. The default setting is “No”.
Maintenance TR-069
ACS URL
URL for TR-069 Auto Configuration Servers (ACS).
TR-069 Username
ACS username for TR-069.
TR-069 Password
ACS password for TR-069.
Periodic Inform Enable
Enables periodic inform. If set to "Yes", device will send inform packets to the ACS. The default setting is "No".
Periodic Inform
Sets up the periodic inform interval to send the inform packets to the ACS.
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Interval
Connection Request
Username
The user name for the ACS to connect to the phone.
Connection Request
Password
The password for the ACS to connect to the phone.
Connection Request
Port
The port for the ACS to connect to the phone.
CPE SSL Certificate
The Cert File for the phone to connect to the ACS via SSL.
CPE SSL Private Key
The Cert Key for the phone to connect to the ACS via SSL.
Maintenance Security Settings Security
Configuration via
Keypad Menu
Configures the access control for the users to configure from keypad Menu.
There are three different options:
Unrestricted. All the options can be accessed in keypad Menu. Basic settings only. The SIP option under Phone submenu, and
Network, Upgrade, UCM Detect and Factory Reset options under
System submenu will not be available in LCD Menu.
Constraint Mode. The phone will require administration password to
change the Network, Upgrade and Factory Reset options under System
submenu, and SIP option under Phone submenu as well.
The default setting is “Unrestricted”.
Enable STAR key
Keypad Locking
If set to "Yes", the keypad can be locked by pressing and holding the STAR
* key for about 4 seconds. A lock icon will show indicating the keypad is
locked. The default setting is "Yes".
Note:
When the keypad is locked, users would need press and hold the STAR *
key for about 4 seconds again and then enter the password to unlock it.
Password to
Lock/Unlock
Configures the password to lock/unlock the keypad.
SIP TLS Certificate
SSL Certificate used for SIP Transport in TLS/TCP.
SIP TLS Private Key
SSL Private key used for SIP Transport in TLS/TCP.
SIP TLS Private Key
Password
SSL Private key password used for SIP Transport in TLS/TCP.
Web Access Mode
Sets the protocol for web interface. The default setting is “HTTP”.
Disable SSH
Disables SSH access. The default setting is “No”.
Web/Keypad/Restrict
mode Lockout
Duration
Specifies the time in minutes that the web or LCD login interface will be
locked out to user after five login failures. This lockout time is used for web
login, STAR keypad unlock and LCD restrict mode admin login. Range is
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0-60 minutes.
Maintenance Security Settings Trusted CA Certificates
Trusted CA Certificates
Upload CA Certificate file to phone.
Maintenance Packet Capture
Status
Displays packet capture status. When user starts to capture trace file, it will
show “RUNNING” status, otherwise, it will show “STOPPED”.
Capture Location
Selects the location where capture file will be stored: internal storage or
USB. The default setting is “Internal Storage”.
With RTP Packets
Defines whether the packet capture file contains RTP or not. The default
setting is “No”.
USB Filename
Defines the filename of the capture. Only required when capture location is
USB.
Phonebook Contacts
Add Contact
Specifies Contact’s First Name, Last Name, Phone Number, Accounts and
Groups to add one new contact in phonebook.
Edit Contact
Edits selected contact.
Delete All Contacts
Deletes all contacts from phonebook.
Phonebook Group Management
Add Group
Specifies Group’s name to add new group.
Edit Group
Edits selected group.
Phonebook Phonebook Management
Enable Phonebook
XML Download
Configures to enable phonebook XML download. Users could select HTTP/HTTPS/TFTP to download the phonebook file. The default setting is "Disabled".
HTTP/HTTPS User
Name
The user name for the HTTP/HTTPS server.
HTTP/HTTPS Password
The password for the HTTP/HTTPS server.
Phonebook XML Server
Path
Configures the server path to download the phonebook XML. This field could be IP address or URL, with up to 256 characters.
Phonebook Download
Interval
Configures the phonebook download interval (in minutes). If it's set to 0, the automatic download will be disabled. The default value is 0. The valid range is 5 to 720 minutes.
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Remove
Manually-edited Entries on Download
If set to "Yes", when XML phonebook is downloaded, the entries added manually will be automatically removed. The default setting is "Yes".
Sort Phonebook by
Sort phonebook based on the selection of first name or last name. The default setting is “Last Name”.
Download XML
Phonebook
Click on "Download" to download the XML phonebook file to local PC.
Upload XML
Phonebook
Click on "Upload" to upload local XML phonebook file to the phone.
Phonebook Key
Function
Control the behavior of phonebook key. There are five options: Default, LDAP Search, Local Phonebook, Local Group, and Broadsoft Phonebook. The default setting is “Default”, when user presses it, phone LCD will show the five options.
Default search mode
Configures the default phonebook search mode. The default setting is “Qick match”.
Phonebook LDAP
LDAP Protocol
Configures the LDAP protocol to LDAP or LDAPS. The default setting is “LDAP”. LDAPS is a feature to support LDAP over TLS.
Server Address
Configures the IP address or DNS name of the LDAP server.
Port
Configures the LDAP server port. The default port number is “389”.
Base
Configures the LDAP search base. This is the location in the directory where the search is requested to begin. Example: dc=grandstream, dc=com ou=Boston, dc=grandstream, dc=com
User Name
Configures the bind "Username" for querying LDAP servers. Some LDAP servers allow anonymous binds in which case the setting can be left blank.
Password
Configures the bind "Password" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.
LDAP Number Filter
Configures the filter used for number lookups. Examples: (|(telephoneNumber=%)(Mobile=%) returns all records which has the "telephoneNumber" or "Mobile" field starting with the entered prefix; (&(telephoneNumber=%) (cn=*)) returns all the records with the "telephoneNumber" field starting with the entered prefix and "cn" field set.
LDAP Name Filter
Configures the filter used for name lookups. Examples: (|(cn=%)(sn=%)) returns all records which has the "cn" or "sn" field starting
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with the entered prefix; (!(sn=%)) returns all the records which do not have the "sn" field starting with the entered prefix; (&(cn=%) (telephoneNumber=*)) returns all the records with the "cn" field starting with the entered prefix and "telephoneNumber" field set.
LDAP Version
Selects the protocol version for the phone to send the bind requests. The default setting is "Version 3".
LDAP Name Attributes
Specify the "name" attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated name attributes. Example: gn cn sn description
LDAP Number
Attributes
Specifies the "number" attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated number attributes. Example: telephoneNumber telephoneNumber Mobile
LDAP Display Name
Configures the entry information to be shown on phone's LCD. Up to 3 fields can be displayed. Example: %cn %sn %telephoneNumber
Max. Hits
Specifies the maximum number of results to be returned by the LDAP server. If set to 0, server will return all search results. The default setting is 50.
Search Timeout
Specifies the interval (in seconds) for the server to process the request and client waits for server to return. The default setting is 30 seconds.
Sort Results
Specifies whether the searching result is sorted or not. Default setting is "No".
LDAP Lookup
Configures to enable LDAP number searching when dialing and receiving calls.
Lookup Display Name
Configures the display name when LDAP looks up the name for incoming call or outgoing call. This field must be a subset of the LDAP Name Attributes. Example: gn cn sn description
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Pattern: Default
Pattern: Analog
Call’s state
Light Indication
Call’s state
Light Indication
Offline
Off
Offline
Off
Idle
Solid Green
Idle
Solid Green
Trying
Solid Red
Trying
Solid Red
Talking
Solid Red
Talking
Solid Red
Proceeding
Flashing Red
Proceeding
Solid Red
Incoming call
Flashing Red
Incoming call
Flashing Red
Pattern: Directional
Mode: Inverse
Call’s state
Light Indication
Call’s state
Light Indication
Offline
Off
Offline
Off
Idle
Solid Green
Idle
Solid Red
Trying
Flashing Green
Trying
Solid Green
Talking
Solid Red
Talking
Solid Green
Proceeding
(Initiator)
Flashing Green
Proceeding
Flashing Green
Proceeding
(Receiver)
Flashing Red
Incoming call
Flashing Green
Incoming call
Flashing Red
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Mode: Reversed (Red)
Mode: Reversed (Green)
Call’s state
Light Indication
Call’s state
Light Indication
Offline
Off (Extension Board Icon:
Off)
Offline
Off (Extension Board Icon:
Off)
Idle
Off (Extension Board Icon:
Idle)
Idle
Off (Extension Board Icon:
Idle)
Trying
Solid Red
Trying
Solid Green
Talking
Solid Red
Talking
Solid Green
Proceeding
Solid Red
Proceeding
Solid Green
Incoming call
Flashing Red
Incoming call
Flashing Green
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Add contacts.
Edit contact.
Click to call this contact from the phone.
Click to dial from
available lines.
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Page 72

EXPERIENCING GXP1760/GXP1780/GXP1782

Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream IP phone, it will be sure to bring convenience and color to
both your business and personal life.
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