DP735

DP730

DP725

DP722

Air Interface

Telephony Standards: DECT

Frequency Bands:

1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US),

1910 – 1920 MHz (Brazil), 1786 – 1792 MHz (Korea),

1893 – 1906 MHz (Japan), 1880 – 1895 MHz (Taiwan)

Number of Channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)

Outdoor Range: up to 400 meters (DP730/DP725/DP735) or up to 350 meters (DP722/DP720)

Indoor Range: up to 50 meters

Energy Saving: ECO Mode

Peripherals

3 LED indicators: Power, Network, DECT

Pairing/Paging button

One 10/100 Mbps auto-sensing Ethernet port with integrated PoE

Protocol/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP

Voice Codecs

G.711μ/a-law, G.723.1, G.729A/B, G.726-32, iLBC, G.722, OPUS, in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, CNG, PLC, AJB

Telephony Features

Hold, transfer, forward, 3-way conference, downloadable phonebook (XML, LDAP, up to 3000 entries), call waiting, call log (up to 750 records), auto answer, flexible dial plan, server redundancy and fail-over

QoS

Layer 2 QoS (802.1Q, 802.1P) and Layer 3 QoS (ToS, DiffServ, MPLS)

Security

User and administrator level access control, MD5 and MD5-sess based authentication, 256- bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access control, DECT authentication & encryption

Multi-language

English, Czech, German, Spanish, French, Hebrew, Italian, Dutch, Polish, Portuguese, Russian, Turkish, Arabic, Chinese Simple, Chinese Tradition, Japanese, Korean, Slovakian, Serbian

Upgrade/ Provisioning

Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or AES encrypted XML configuration file

Multiple SIP Accounts

Up to ten (10) distinct SIP accounts per system

Each handset may map to any SIP account(s)

Each SIP account may map to any handset(s)

Ring Group

Flexible options when multiple handsets share the same SIP account:

Circular Mode: all phones ring sequentially from the phone next to the one that answered last

Linear Mode: all phones ring sequentially in the predesignated order

Parallel Mode: all phones ring concurrently and after one phone answers, the remaining available phones can make new calls

DID Call Group: multiple handsets can call out on the same account, but all incoming calls are directed to a single handset

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; Micro-USB connection;

PoE: IEEE802.3af Class 1, 0.44W–3.84W

Package Content

Base Unit, Universal Power Supply; Ethernet cable; Quick Start Guide, GPL statement

Dimensions (H x W x D)

140.31 x 64.98 x 105mm

Weight

Base unit: 140g; Universal power supply: 50g; Package: 370g

Temperature and Humidity

Operation: -10 to 50ºC (14 to 122ºF); Storage: -20º to 60ºC (-4 to 140ºF)

Humidity: 10% to 90% non-condensing

Compliance

FCC, CE, RCM, IC

Air Interface

Telephony standards: DECT
Frequency bands:

1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil), 1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan), 1880 – 1895 MHz (Taiwan)
Number of Channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)
Outdoor Range: up to 400 meters (DP752/DP755) or up to 300 meters (DP750)
Indoor Range: up to 50 meters

Peripherals

2.4 inch (240×320) color TFT LCD
27 keys including 3 soft keys, 5 navigation/ menu keys, 4 dedicated function keys for SPEAKERPHONE, MUTE, SEND, POWER/END, 3 side keys including 2 volume (up and down) and 1 Push-to-Talk key
3-color MWI LED
3.5mm headset jack
Proximity and accelerometer sensors
Removable belt clip
Micro-B USB port for alternative charging and non-battery operation Vibration Ring Mode

Protocol/Standards

Hearing Aid Compatibility (HAC) compliant

Voice Codecs

G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law,G.723.1, G.729A/B, iLBC and OPUS are supported via companion DECT base station), AEC, AGC, Ambient noise reduction on handset mic, advanced noise suppression forincoming audio

Telephony Features

Hold, transfer, forward, 3-way conference, push-to-talk, intercom, downloadable phonebook, call waiting, call log, auto answer, click-to-dial, flexible dial plan

HD Audio

Yes, both on Handset and Speakerphone

Security

DECT authentication & encryption

Multi-language

English, Czech, German, Spanish, French, Hebrew, Italian, Dutch, Polish, Portuguese, Russian, Turkish, Arabic, Chinese Simple, Chinese Tradition, Japanese, Korean, Slovakian, Serbian, Greek, Swedish

Upgrade/ Provisioning

Software Upgrade Over-The-Air (SUOTA), handset provisioning Over-The-Air

Multi-line Access

Each handset may access up to twenty (20) lines

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; Micro-B USB Connector;
Rechargeable 2000mAh Li-ion battery, 650 hours of standby time and 65 hours of talk time (laboratory data)

Package Content

Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Start Guide

Dimensions (H x W x D)

Handset:166.00 x 57.80 x 26.30mm; Charger cradle: 82.00 x 80.00 x 30.60mm

Weight

Handset: 221.5g;
Charger cradle: 56g;
Universal power supply: 55g;
Package: 535g

Weatherproof

IP67 Rated-Water-proof, dust-proof, cleaning chemical resistant, anti-microbial casing and 2.5m drop safe

Temperature and Humidity

Operation: -10º to 45ºC (14 to 113ºF); Charging: 0 to 45ºC (32 to 113ºF)
Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing

Compliance

FCC, CE, RCM, IC

Air Interface

Telephony standards: DECT

Frequency bands:

Number of channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)

Outdoor Range: up to 400 meters (DP752/DP755) or up to 300 meters (DP750)

Indoor Range: up to 50 meters

Peripherals

2.4 inch (240×320) color TFT LCD

27 keys including 3 soft keys, 5 navigation/ menu keys, 4 dedicated function keys for SEND, POWER/END, SPEAKERPHONE, MUTE, 3 side keys including 2 volume (up and down) and 1 Push-to-Talk key

3-color MWI LED

3.5mm headset jack

Proximity and accelerometer sensors

Backlit keypad

Removable belt clip

Micro-USB port for alternative charging and non-battery operation

Protocols/Standards

Hearing Aid Compatibility (HAC) compliant

Voice Codecs

G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via companion DECT base station), AEC, AGC, Ambient noise reduction on handset mic, advanced noise suppression for incoming audio

Telephony Features

Hold, transfer, forward, 3-way conference, push-to-talk, intercom, downloadable phonebook, call waiting, call log, auto answer, click-to-dial, flexible dial plan

HD Audio

Yes, in both Handsets and Speakerphone modes

Security

DECT authentication & encryption

Multi-language

English, Czech, German, Spanish, French, Hebrew, Italian, Dutch, Polish, Portuguese, Russian, Turkish, Arabic, Chinese Simple, Chinese Tradition, Japanese, Korean, Slovakian, Serbian, Greek, Swedish.

Upgrade/ Provisioning

Software Upgrade Over-The-Air (SUOTA), handsets provisioning Over-The-Air

Multiple Line Access

 Each handset may access up to twenty (20) lines

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; Micro-USB connection;

Rechargeable Li-ion battery, 500 hours of standby time and 40 hours of talk time. (laboratory data)

Package Content

Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Installation Guide

Dimensions (H x W x D)

Handset: 168.5 x 52.5 x 21.8mm;

Charger cradle: 76 x 73 x 81mm

Weight

Handset: 180g;

Charger cradle: 78g;

Universal power supply: 50g;

Package: 465g

Temperature and Humidity

Operation: -10º to 50ºC (14 to 122ºF); Charging: 0 to 45ºC (32 to 113ºF); Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing

Compliance

FCC, CE, RCM, IC

Air Interface

Telephony standards: DECT
Frequency bands:
1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US),
1910 – 1920 MHz (Brazil), 1786 – 1792 MHz (Korea),
1893 – 1906 MHz (Japan), 1880 – 1895 MHz (Taiwan)
Number of Channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)
Outdoor Range: up to 400 meters (DP752/DP755) or up to 300 meters (DP750)
Indoor Range: up to 50 meters

Peripherals

1.8 inch (128×160) color TFT LCD
27 keys including 2 soft keys, 5 navigation/ menu keys, 5 dedicated function keys for SPEAKERPHONE, MUTE, SEND, POWER/END, 3 side keys including 2 volume (up and down) and 1 Push-to-Talk key,
3-color MWI LED
3.5mm headset jack
Removable belt clip
Type C USB port for alternative charging and non-battery operation

Protocols/Standards

Hearing Aid Compatibility (HAC) compliant

Voice Codecs

G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via companion DECT base station), AEC, AGC, Ambient noise reduction on handset mic, advanced noise suppression for incoming audio

Telephony Features

Hold, transfer, forward, 3-way conference, push-to-talk, intercom, downloadable phonebook, call waiting, call log, auto answer, click-to-dial, flexible dial plan

HD Audio

Yes, both on Handset and Speakerphone

Security

DECT authentication & encryption

Multi-language

English, Czech, German, Spanish, French, Hebrew, Italian, Dutch, Polish, Portuguese, Russian, Turkish, Arabic, Chinese Simple, Chinese Tradition, Japanese, Korean, Slovakian, Serbian, Greek, Swedish

Upgrade/ Provisioning

Software Upgrade Over-The-Air (SUOTA), handset provisioning Over-The-Air

Multiple Line Access

Each handset may access up to twenty (20) lines

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; USB-C Connector;

Rechargeable 1500mAh Li-ion battery (500 hours of standby time and 50 hours of talk time) (laboratory data)

Package Content

Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Installation Guide

Dimensions (H x W x D)

Handset: 135 x 49 x 15.5mm; Charger cradle: 85 x 85 x 25mm

Weight

Handset: 107g; Charger cradle: 61g; Universal power supply: 55g; Package: 364g

Temperature and Humidity

Operation: -10º to 45ºC (14 to 113ºF); Charging: 0 to 45ºC (32 to 113ºF)
Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing

Compliance

FCC, CE, RCM, IC

Air Interface

Telephony standards: DECT

Frequency bands:

Number of channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)

Outdoor Range: up to 350 meters (with DP755 and DP752) or up to 300 meters (DP750)

Indoor Range: up to 50 meters

Peripherals

1.8 inch (128×160) color TFT LCD

23 keys including 2 softkeys, 5 navigation / menu keys, 4 dedicated function keys for SEND, POWER/END, SPEAKERPHONE, MUTE

3-color MWI LED

3.5mm headset jack

Removable belt clip

Micro-USB port for alternative charging and non-battery operation

Protocols/Standards

Hearing Aid Compatibility (HAC) compliant

Voice Codecs

G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via companion DECT base station), AEC, AGC, Ambient noise reduction on handset mic, advanced noise suppression for incoming audio

Telephony Features

Hold, transfer, forward, 3-way conference, push-to-talk, intercom, downloadable phonebook, call waiting, call log, auto answer, click-to-dial, flexible dial plan

HD Audio

Yes, in both Handsets and Speakerphone modes

Security

DECT authentication & encryption

Multi-language

English, Czech, German, Spanish, French, Hebrew, Italian, Dutch, Polish, Portuguese, Russian, Turkish, Arabic, Chinese Simple, Chinese Tradition, Japanese, Korean, Slovakian, Serbian, Greek, Swedish.

Upgrade/ Provisioning

Software Upgrade Over-The-Air (SUOTA), handsets provisioning Over-The-Air

Multiple Line Access

Each handset may access up to twenty (20) lines

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; Micro-USB connection; Rechargeable 800mAh Ni-MH Low Self-Discharge (LSD) AAA batteries (250 hours of standby time and 20 hours of talk time) (laboratory data)

Package Content

Handset unit, universal power supply, charger cradle, belt clip, 2 batteries, Quick Start Guide

Dimensions (H x W x D)

Handset: 158 x 50 x 28.1mm;

Charger cradle: 81.15 x 75.89 x 36.36mm

Weight

Handset: 110g;

Charger cradle: 44g;

Universal power supply: 50g;

Package: 328g

Temperature and Humidity

Operation: -10º to 50ºC (14 to 122ºF); Charging: 0 to 45ºC (32 to 113ºF); Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing

Compliance

FCC, CE, RCM, IC

DP730

DP722

LED Light

Status

Indicates Power ON/OFF.

Indicates the status of SIP account registration and network.

Indicates the status of the DECT handset registration:

Indicates the status if 3 LEDs are blinking:

Key

Description

1

LED Indicator

Red: Charging
Green: Charge completed
Blinking green: Missed call(s) or Voice Mail received
Blinking red: Low Battery

2

Earphone

Delivers audio output.

3

Volume Up key

Increases the audio volume during calls or while using other audio features.

4

Volume down key

Decreases the audio volume during calls or while using other audio features.

5

PTT key

Used for instant communication similar to a walkie-talkie, allowing the user to press and hold to talk and release to listen.

6

Off-hook /Dial key

Initiates a call or dials the number entered. It can also bring the handset into dialing mode from the home screen.

7

4 Arrow key combination (Up, Down,
Left, Right)

Allows the user to navigate through the menu options and settings on the LCD screen.

8

1 / Voicemail key

Directly accesses the voicemail system, allowing the user to listen to and manage their voice messages.

9

Handsfree / Speaker key

Switches the call to the hands-free speaker mode, enabling the user to speak and listen without holding the handset.

10

Microphone

Captures the user’s voice during calls, ensuring clear communication.

11

Type C USB port

Used for charging the handset.

12

Dedicated function key

Executes a specific function that varies depending on the current context or application being used.

13

Mute key

Toggles the mute feature on or off, allowing the user to mute their microphone during a call.

14

# / Lock key

Locks the keypad to prevent accidental key presses. To lock the keypad, press and hold the # key for about 2 seconds. To unlock, press the Unlock softkey and then the # key.

15

Alphanumeric keypad

Provides numbers, letters, and special characters for dialing phone numbers, entering text, or navigating through the menu.

16

Menu /OK key

Opens the main menu from the home screen and confirms selections or actions within the menu.

17

On-hook /Power key

Ends calls or turns the handset on and off. When the handset is off, holding this key will power it on; pressing it during a call will end the call.

18

Softkeys

Perform functions displayed on the LCD screen directly above them. The functions change depending on the current menu or application.

19

Color LCD Screen

Displays all relevant information such as call details, status icons, prompt messages, and menu options. It is the main interface for interacting with the handset.

20

3.5 mm headset jack

Allows the user to connect a wired headset for audio input and output, providing an alternative to the built-in microphone and speaker.

Battery status

Not equipped with a battery

Battery status

Battery empty

Battery status

Battery low

Battery status

Battery normal

Battery status

Battery full

Battery status

Charging

Signal status

Not subscribed

Signal status

Not in range

Signal status

Signal very low

Signal status

Signal low

Signal status

Signal normal

Signal status

Signal good

Signal status

Signal very good

Microphone MUTE Status

OFF – Not muted

ON – Muted

Speaker status

OFF – The speaker

is inactivated ON – The speaker

is activated

Headset icon

Missed Call icon

Voicemail icon

Ringtone status

OFF – Ringtone off (Silent mode)

ON – Ringtone on

Keypad Lock status

OFF – Keypad unlock

ON – Keypad locked

DND Status.

OFF – Do Not Disturb disabled

ON – Do Not Disturb enabled

Call waiting

Information

Account not registered

Account Registered

Error message

Handset number

Incoming Call notification

Outgoing Call notification

Missed Call notification

Voicemail notification

Contacts

Call History

Registration

Voice Mail

Preferences

Shortcut

Call Features

Status

Settings

Key

Description

1

LED Indicator

Red: Charging
Green: Charge completed
Blinking green: Missed call(s) or Voice Mail received
Blinking red: Low Battery

2

Earphone

Delivers audio output.

3

Volume Up key

Increases the audio volume during calls or while using other audio features.

4

Volume down key

Decreases the audio volume during calls or while using other audio features.

5

PTT key

Used for instant communication similar to a walkie-talkie, allowing the user to press and hold to talk and release to listen.

6

Off-hook /Dial key

Initiates a call or dials the number entered. It can also bring the handset into dialing mode from the home screen.

7

4 Arrow key combination (Up, Down,
Left, Right)

Allows the user to navigate through the menu options and settings on the LCD screen.

8

1 / Voicemail key

Directly accesses the voicemail system, allowing the user to listen to and manage their voice messages.

9

Handsfree / Speaker key

Switches the call to the hands-free speaker mode, enabling the user to speak and listen without holding the handset.

10

Microphone

Captures the user’s voice during calls, ensuring clear communication.

11

Type C USB port

Used for charging the handset.

12

Dedicated function key

Executes a specific function that varies depending on the current context or application being used.

13

Mute key

Toggles the mute feature on or off, allowing the user to mute their microphone during a call.

14

# / Lock key

Locks the keypad to prevent accidental key presses. To lock the keypad, press and hold the # key for about 2 seconds. To unlock, press the Unlock softkey and then the # key.

15

Alphanumeric keypad

Provides numbers, letters, and special characters for dialing phone numbers, entering text, or navigating through the menu.

16

Menu /OK key

Opens the main menu from the home screen and confirms selections or actions within the menu.

17

On-hook /Power key

Ends calls or turns the handset on and off. When the handset is off, holding this key will power it on; pressing it during a call will end the call.

18

Softkeys

Perform functions displayed on the LCD screen directly above them. The functions change depending on the current menu or application.

19

Color LCD Screen

Displays all relevant information such as call details, status icons, prompt messages, and menu options. It is the main interface for interacting with the handset.

20

3.5 mm headset jack

Allows the user to connect a wired headset for audio input and output, providing an alternative to the built-in microphone and speaker.

Key

Description

1

LED Indicator

Red: Charging
Green: Charge completed
Blinking green: Missed call(s) or Voice Mail received
Blinking red: Low Battery

2

Earphone

Delivers audio output.

3

Volume Up key

Increases the audio volume during calls or while using other audio features.

4

Volume down key

Decreases the audio volume during calls or while using other audio features.

5

PTT key

Used for instant communication similar to a walkie-talkie, allowing the user to press and hold to talk and release to listen.

6

Off-hook /Dial key

Initiates a call or dials the number entered. It can also bring the handset into dialing mode from the home screen.

7

4 Arrow key combination (Up, Down,
Left, Right)

Allows the user to navigate through the menu options and settings on the LCD screen.

8

1 / Voicemail key

Directly accesses the voicemail system, allowing the user to listen to and manage their voice messages.

9

Handsfree / Speaker key

Switches the call to the hands-free speaker mode, enabling the user to speak and listen without holding the handset.

10

Microphone

Captures the user’s voice during calls, ensuring clear communication.

11

Type C USB port

Used for charging the handset.

12

Dedicated function key

Executes a specific function that varies depending on the current context or application being used.

13

Mute key

Toggles the mute feature on or off, allowing the user to mute their microphone during a call.

14

# / Lock key

Locks the keypad to prevent accidental key presses. To lock the keypad, press and hold the # key for about 2 seconds. To unlock, press the Unlock softkey and then the # key.

15

Alphanumeric keypad

Provides numbers, letters, and special characters for dialing phone numbers, entering text, or navigating through the menu.

16

Menu /OK key

Opens the main menu from the home screen and confirms selections or actions within the menu.

17

On-hook /Power key

Ends calls or turns the handset on and off. When the handset is off, holding this key will power it on; pressing it during a call will end the call.

18

Softkeys

Perform functions displayed on the LCD screen directly above them. The functions change depending on the current menu or application.

19

Color LCD Screen

Displays all relevant information such as call details, status icons, prompt messages, and menu options. It is the main interface for interacting with the handset.

20

3.5 mm headset jack

Allows the user to connect a wired headset for audio input and output, providing an alternative to the built-in microphone and speaker.

Battery status

Not equipped with a battery

Battery status

Battery empty

Battery status

Battery low

Battery status

Battery normal

Battery status

Battery full

Battery status

Charging

Signal status

Not subscribed

Signal status

Not in range

Signal status

Signal very low

Signal status

Signal low

Signal status

Signal normal

Signal status

Signal good

Signal status

Signal very good

Microphone MUTE Status

OFF – Not muted

ON – Muted

Speaker status

OFF – The speaker

is inactivated ON – The speaker

is activated

Headset icon

Missed Call icon

Voicemail icon

Ringtone status

OFF – Ringtone off (Silent mode)

ON – Ringtone on

Keypad Lock status

OFF – Keypad unlock

ON – Keypad locked

DND Status.

OFF – Do Not Disturb disabled

ON – Do Not Disturb enabled

Call waiting

Information

Account not registered

Account Registered

Error message

Handset number

Incoming Call notification

Outgoing Call notification

Missed Call notification

Voicemail notification

Contacts

Call History

Registration

Voice Mail

Preferences

Shortcut

Call Features

Status

Settings

Contacts

Note: Private/Global Phonebooks will be merged on the handset.

Call History

Display the call history:

Note: You can add contacts to Shared Contacts directly from call logs.

Registration

Voice Mail

Preferences

Customizing keys functions

Customizing keys functions:

Select key and press OK button to configure function. Following functions are available for arrow keys: 1. Disabled, 2. Missed Calls, 3: Accepted Calls, 4: Outgoing Calls, 5: History, 6: Contacts, 7: Status, 8: Line, 9: Voice Mail, 10: Ringer Volume Up, 11: Ringer Volume Down, 12: Audio Volume Up, 13: Audio Volume Down, 14: Intercom.

Select a key [2], [3], [4], [5], [6], [7], [8] or [9] and press OK button. Select “Edit” to manually specify the destination number or select “From Contacts” to select a contact as speed dial destination.

Call Features

Status

Settings

All LEDs will light up, and the LCD will display a table listing the names of all keys in red. Press any key to diagnose; the key’s name will display in blue. After all keys are diagnosed, a prompt message (“PASS”) will display; press “Back” (right softkey) to exit.

Note: User can long press arrow UP key to exit at any time.

User Level

Username

Password

Web Pages Allowed

End User Level

 user

Password defined by the administrator

 Only Status, Settings and Maintenance

Administrator Level

admin

Random password available on the sticker at the back of the unit.

All pages

Account Status

Account

Displays list of configured accounts’ names, from Account 1 to Account 10.

SIP User ID

Displays list of SIP user id registered.

SIP Server

Displays list of SIP Server.

SIP Registration

Shows the status of SIP registration. If the SIP account is successfully registered, it will display “YES” with green background. If the SIP account is not registered, it will display “NO” with red background.

HS Mode

Displays the HS mode configured for each account.

HS status table

Illustrates both handsets and SIP accounts statuses. Each column is dedicated to one HS; each row shows the status of the account on that HS:

For example, if accounts 1, 3 and 4 are assigned to HS3 with account 3 in use, the column for HS3 will have cell 3 with red icon, cells 1 and 4 with green icon, and cells 2 and 5 with gray icon.

DECT Base Status

Base Station Name

Displays name of base station. Default is DP75x_[last 6 digit of MAC address].

Base DECT FW Version

Shows firmware version of base DECT.

Base DECT RF Region

Indicates region of base DECT RF.

Base DECT RFPI Address

Specifies DECT RFPI (Radio Fixed Part Identity) address which is a unique identity for the base.

Global Functions

– Unsubscribe Handsets: Unsubscribes all handsets from DECT base station.

– Upgrade Handsets: Upgrade all handsets from DECT base station.

Handset Status

• Handset Name: Displays index and name (customizable) of each handset.

• IPEI: Indicates IPEI number of each handset; this is the unique identity for the handset. If the handset is in range, the IPEI will be displayed with a green background, otherwise, it will be displayed with a red background.

• Battery icon

: Illustrates battery status for each Handset; it can be either:

Fully charged

Not fully charged

Low, needs to be charged or replaced

Charging

• Page: Sends paging request to corresponding handset, which will receive incoming ring tone and “Paging” will be displayed on their LCD screens; this function helps you locate the handsets.

• Unsubscribe: Unsubscribes corresponding handset from DECT base station.

• HS Firmware: Indicates handset’s firmware version number.

DECT Repeater Status

Paired Devices

Displays paired repeaters showing:

Discovered Devices

Displays discovered repeaters using:

Line Options

Account

Displays list of configured accounts’ names, from Account 1 to Account 10.

SIP User ID

Displays list of SIP user id registered.

DND

Shows DND (Do Not Disturb) feature status (per SIP account).

Forward

Indicates destination to which all incoming calls will be forwarded (per SIP account).

Busy Forward

Indicates destination to which incoming calls will be forwarded when the line is busy (per SIP account).

Delayed Forward

Indicates destination to which incoming calls will be forwarded if the call is not answered within a specified period of time or number of rings (per SIP account).

Network Status

MAC Address

Shows Device ID in hexadecimal format. This is needed by network administrators for troubleshooting. The MAC address will be used for provisioning and can be found on the label on original box and on the label located on the bottom panel of the device.

IP Address Mode

Indicates used IP address mode: DHCP, Static IP or PPPoE.

IP Address

Displays assigned IP address. Example: 192.168.5.110

Subnet Mask

Displays assigned subnet mask. Example: 255.255.255.0

Gateway

Displays assigned default gateway. Example: 192.168.5.1

PPPoE Link Up

Indicates PPPoE connection status.

DNS Server 1

Shows assigned DNS server address 1. Example: 8.8.8.8

DNS Server 2

Shows assigned DNS server address 2. Example: 8.8.4.4

NAT Traversal

Indicates type of NAT for each Profile. (Based on STUN protocol.)

System Info

Product Model

Displays product model info. Default is DP752.

Part Number

Shows product part number. Example: 9610003814A (last 2 digits show HW version, in this example 14A for HW version 1.4A)

Certificate Type

Displays the ceritificate type used for encryption.

Hardware Version

Shows the hardware version of the unit.

Serial Number

Shows the serial number of the unit.

Security Version

This indicates the security version of the DP752 base station.

Software Version

System Up Time

Indicates system uptime since last reboot.

System Time

Shows actual time and date according to your configuration.

Service Status

Reveals status of VoIP applications.

General Settings

Profile Active

Activates or deactivates the SIP profile. The profile is deactivated by default.

Profile Name

Determines the name of the profile, this profile name can also be used in Handsets config provisioning for validation. By default, the values are profile 1 – profile 4.

SIP Server

Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used to send/receive SIP messages from/to DP752/DP750.
Example:  sip.my_company.com

Failover SIP Server

Specifies failover SIP server IP address or domain name provided by VoIP service provider. This server will be used if the primary SIP server becomes unavailable.

Prefer Primary SIP Server

Prefers primary SIP server. The profile will register to primary Server if registration with Failover server expires. Default is No.

Outbound Proxy

Specifies IP address or domain name of outbound proxy, media gateway or session border controller. Used by DP752 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work, and only outbound proxy can correct the problem.

Backup Outbound Proxy

IP address or Domain name of the Secondary Outbound Proxy which will be used when the primary proxy cannot be connected.

Voice Mail Access Number

Defines the voice mail portal access number to allow users accessing their voice messages. Max allow length 30

Network Settings

Layer 3 QoS Settings

SIP Diff-Serv

Defines SIP Diff-Serv value. Default is 24.

RTP Diff-Serv

Defines RTP Diff-Serv value. Default is 46.

DNS Settings

DNS Mode

Selects DNS mode to use for the client to look up server. Default is A Record.
A Record: resolves IP Address of target according to domain name.
SRV: DNS SRV resource records indicate how to find services for various protocols.
NAPTR/SRV: Naming Authority Pointer according to RFC 2915.
Use Configured IP: If selected, please fill in Primary IP, Backup IP 1 and Backup IP 2 to be used for server look up.

DNS SRV Failover mode

This feature is used to configure the preferred IP mode for DNS SRV, the options available are:

The default value is set to “Default”

Register before DNS SRV failover

When the DNS SRV Failover Mode is enabled, you can also choose to “Register before DNS SRV failover” that can waive the 3 failed tries, or still try 3 times then use the failover DNS.

Maximum Number of SIP Request Retries

Defines the number of SIP retries before DNS failover. 
the Default value is 2.

Failback Expiration (m)

Defines the expiration time in minutes when using Broadsoft R23 failback mode. The default value is 60.
The valid range is from 0-64800 minutes.
Note: Setting the Faliback Expiration timer to 0 will be helpful only when using Telstra Special Feature.

Primary IP

Specifies primary IP address where the base sends DNS query to, when “Use Configured IP” is selected for DNS mode.

Backup IP 1

Specifies backup IP 1 address where the base sends DNS query to, when “Primary IP” is not responding.

Backup IP 2

Specifies backup IP 2 address where the base sends DNS query to, when “Backup IP 1” is not responding.

NAT Settings

NAT Traversal

Enables/disables NAT traversal mechanism. If activated (by choosing “STUN”) and a STUN server is also specified (Maintenance 🡪 Network Settings 🡪 STUN Settings); the base performs according to STUN client specification.
Under this mode, embedded STUN client will detect if and what type of firewall/NAT is being used. If detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the base will use its mapped public IP address and port in all of its SIP and SDP messages. If NAT Traversal field is set to “Keep Alive”, the base will periodically (every 20 seconds) send a blank UDP packet (with no payload data) to SIP proxy to keep the “ping hole” on the NAT open.

Use NAT IP

Defines NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP.

Proxy-Require

Determines a SIP Extension to notify the SIP server that the base is behind a NAT/Firewall.

Use SBC

Indicate whether or not a SBC server is used, if users want to work under SBC associated with 3CX, they should enable this feature to have better communication with the server.

SIP Settings

Basic Settings

SIP Transport

Selects transport protocol for SIP packets; UDP or TCP or TLS. Make sure your SIP server or network environment supports SIP over the selected transport method. Default 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 TCP/TLS

Defines whether the actual ephemeral port in contact with TCP/TLS will be used when TLS/TCP is selected for SIP Transport. 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 connection.
The default setting is “No”.

SIP Registration

Controls whether to send SIP REGISTER messages to the proxy server. Device may not be able to make/receives calls if disabled. Default is Yes.

Unregister on Reboot

Controls whether to clear SIP user’s information by sending an un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with the “Contact” header set to * and Expires=0 parameters to the SIP server. This will unregister all SIP accounts under concerned Profile. The default value is “No”.

Add Auth Header on Initial REGISTER

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request. Default is No.

Outgoing Calls Without Registration

Enables the ability to place outgoing calls even when not registered (if allowed by ITSP); the device will not be able to receive incoming calls. Any HS member of a hunting group that is not registered with a SIP account, will be able to place outbound calls using the SIP credentials of the primary hunting group HS.
For example, HS 1, 3, and 5 are the same Hunting Group members. HS 1 is registered with a SIP account. HS 3 and 5 are not reported. HS 3 and 5 will be able to place outbound calls using the SIP account of HS 1, even if Outgoing Call without Registration is set to No. The default is Yes.

Register Expiration

Refreshes registration periodically with specified SIP proxy (in minutes). Maximum interval is 65535 minutes (about 45 days). Default is 60 minutes (or 1 hour).

SIP Registration Failure Retry Wait Time

Sends re-register request after specific time (in seconds) when registration process fails. Maximum interval is 3600 seconds (1 hour). Default is 20 seconds.

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Sends re-register request after specific time (in seconds) when registration process fails due to “403 Forbidden”. Valid range is 0 to 3600 in second. 0 second means stop retry registration. Default is 1200 seconds.

Reregister Before Expiration

Sends re-register request after specific time (in seconds) to renew registration before the previous registration session expires.

Local SIP Port

Defines local port to use by the base for listening and transmitting SIP packets. Default value for Profile 1 is 5060, 6060 for Profile 2, 7060 for Profile 3 and 8060 for Profile 4.

Use Random SIP Port

Controls whether to use configured or random SIP ports. This is usually necessary when multiple base stations are behind the same NAT.

The default setting is “Yes”.

Local RTP Port

Defines local RTP port used to listen and transmit RTP packets. Default is 5004.

Use Random RTP Port

Defines local port to use by the base for listening and transmitting RTP packets. Default value for Profile 1 is 5004, 6004 for Profile 2, 7004 for Profile 3 and 8004 for Profile 4.

SIP T1 Timeout

Defines T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, the base station will attempt to send a request to a SIP server. The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time defined by T2. Default is 0.5 seconds.

SIP T2 Timeout

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling of T1 continues until it reaches T2 value. Default is 4 seconds.

SIP Timer D

Configures SIP timer D defined in RFC3261. Range of values 0-64. Default is 0.

SIP Timer F 

Configures SIP timer F defined in RFC3261. Range of values 0-64. The default is 0.

SIP Timer B 

Configures SIP timer B defined in RFC3261. Range of values 0-64. Default is 0.

Enable OPTIONS Keep Alive

Enables OPTIONS Keep Alive, to check SIP server.
Default is No.

OPTIONS Keep Alive Interval

Time interval for OPTIONS Keep Alive feature in seconds. Range of values is 1–64800. Default is 30.

OPTIONS Keep Alive Max Lost

A maximum number of lost packets for the OPTIONS Keep Alive feature before the phone sends a re-registration. Range of values 3-10.
The default is 3.

Remove OBP from Route

Removes outbound proxy information from “Route” header when sending SIP packets. Default is No.

Support SIP Instance ID

Adds “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft. Default is Yes.
Note: Users can now unregister SIP instance ID to manage SIP sessions.

Hold Target Before Refer

Sends re-INVITE to hold transfer target before sending REFER message to transferee. Default is No.

Refer-To Use Target Contact

Includes target’s “Contact” header information in “Refer-To” header when using attended transfer. Default is No. 

SUBSCRIBE for MWI

Sends periodic “SUBSCRIBE” requests (depends on “Register Expiration” parameter) for message waiting indication service. Default is No.

Enable 100rel

Appends “100rel” attribute to the “required” header of the initial signaling messages. Default is No.

TEL URI

Indicates E.164 number in the “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the base has an assigned PSTN Number.
Disabled: Will use “SIP User ID” information in the Request-Line and “From” header.
Disabled: Will use “SIP User ID” information in the Request-Line and “From” header.
Enabled: “Tel:” will be used instead of “sip:” in the SIP request.
Please consult your carrier before changing this parameter. The default is Disabled.

Do Not Escape ‘#’ as %23 in SIP URI

Replaces “#” by “%23” when sending SIP packets. Default is No.

Disable Multiple m Line in SDP

Sends only one m line in SDP, regardless of how many m fields are in the incoming SDP. Default is No.

Use Privacy Header

Controls whether the Privacy Header will be present in SIP INVITE message. Default is Default.

Use P-Preferred-Identity Header

Controls whether PPI Header will be present in SIP INVITE message. Default is Default.

Ignore Alert-Info Header

This option is used to configure default ringtone. If set to “Yes”, configured default ringtone will be played. 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”.

Allow SIP Reset

Allows to reset the devices directly through SIP Notify. If “Allow SIP Reset” is set to “YES”, then the base receives the NOTIFY from the SIP server with Event: reset, the base should perform a factory reset after the authentication.

The authentication in this case can be either with:
The admin password if no SIP account is configured on the base.
With the credentials of the SIP account if configured on the base.

By default, it is set to “No”.

Session Timer

Enable Session Timer

Enables/Disables the Session Timer Support. Default is Yes.

Session Expiration

Enables periodic refresh of SIP session via a SIP request (UPDATE, or re-INVITE). When the session interval expires and there is no refresh via an UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. Default is 180 seconds.

Min-SE

Defines Minimum session expiration (in seconds).
Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it.
Default is No.

Callee Request Timer

Uses session timer when receiving inbound calls with session timer request.
Default is No.

Force Timer

It uses a session timer even if the remote party does not support this feature. Selecting “No” will enable session timer only when the remote party supports it.
The default is No.
To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls:
UAC: The base station acts as the refresher.
UAS: Callee or proxy server act as the refresher.
Default is Omit.

UAS Specify Refresher

Specifies which end will act as a refresher for incoming calls:
UAS: The base station serves as the refresher.
UAC: Callee or proxy server act as the refresher.
The default is Omit.

Force INVITE

Uses INVITE message to refresh the session timer. Default is No.

Security Settings

Validate Incoming Messages

Defines whether incoming messages will be validated or not. Default is No.

Check SIP User ID for Incoming INVITE

Checks SIP User ID in the Request URI of incoming INVITE; if it doesn’t match the base SIP User ID, the call will be rejected. Direct IP calling will also be disabled. Default is No.

Accept Incoming SIP from Proxy Only

Checks SIP address of the Request URI in the incoming SIP message; if it doesn’t match SIP server address of the account, the call will be rejected. Default is No.

Authenticate Incoming INVITE

Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.

Authenticate Server Certificate Domain

Checks server TLS certificate to ensure that common name matches the configured SIP server. Default is No.

Authenticate Server Certificate Chain

Checks server TLS certificate to ensure that it is authorized by a known certificate authority. Default is No.

Trusted CA Certificate

Treats entered certificate as a valid CA for authenticating the server TLS certificate. The default is No.

Audio Settings

DTMF Settings

Send DTMF

Specifies the mechanism to transmit DTMF digits. Default is via RTP (RFC2833)
In audio: which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs);
Via RTP (RFC2833): permits 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: uses 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.

Disable DTMF Negotiation

Uses above DTMF order without negotiation. Default is No.

DTMF Payload Type

Defines payload type for DTMF using RFC2833.

Vocoder Settings

Preferred Vocoder

Configures vocoders in a preference list (up to 8 preferred vocoders) that will be included with same order in SDP message. Vocoder types are G.711 A-/U-law, G.722, G.726-32, G.723, G.729, iLBC and OPUS. Set to “Disabled“, to disable a preferred vocoder.

Voice Frames per TX

Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for G711/G726/G723/other codecs respectively.

G723 Rate

Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is 6.3kbps.

G726-32 Packing Mode

Defines G726-32 packing mode (“ITU” or “IETF”). Default is ITU.

iLBC Frame Size

Specifies iLBC packet frame size (20ms or 30ms). Default is 20ms.

iLBC Payload type

Determines payload type for iLBC. The valid range is between 96 and 127.
Default is 97.

Disable OPUS stereo in SDP

Disables OPUS stereo attribute in SDP header. Default is No.

OPUS Payload Type

Determines OPUS payload type. The valid range is between 96 and 127.
Default is 123.

Use First Matching Vocoder in 200OK SDP

Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same order received in INVITE. Default is No.

Callee Codec Negotiation Priority

When callee, whose codecs are given priority in Codecs negotiation, local or remote. Set it Remote will use remote codec priority, or local will use local priority. Default is Remote.

SRTP Mode

Selects the SRTP mode to use (“Disabled”, “Enabled but not forced”, “Enabled and forced”,  “Follow SIP Transport”, or “Optional”).

The default is Disabled.

Crypto Life Time

Adds crypto life time header to SRTP packets. Default is Yes.

Silence Suppression (VAD)

Allows detecting the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. Default is No.

Jitter Buffer Type

Selects jitter buffer type (Fixed or Adaptive) based on network conditions.

Jitter Buffer Length

High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.
Medium (initial 100ms, min 20ms, max 200ms).

Voice Monitoring

Enable RTCP

Enables RTCP statistics and control information. Default settings is RTCP.

Call Settings

Early Dial

Sends an early INVITE each time a key is pressed when a user dials a number. Otherwise, only one INVITE is sent after full number is dialed (the user presses Dial Key or after “no key entry timeout” expires). This option should be used only if there is a SIP proxy configured and supporting “484 Incomplete Address” responses. Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).
The default is No.
This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

Adds specified prefix to dialed number.

Dial Plan

Dial Plan Rules:

Accept Digits : +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d ;
Grammar: x – any digit from 0-9;xx+ – at least 2-digit number;
xx – exactly 2-digit number;
^ – exclude;
. – wildcard, matches one or more characters
[3-5] – any digit of 3, 4, or 5;
[147] – any digit 1, 4, or 7;
<2=011> – replace digit 2 with 011 when dialing
<=1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed
| – or
Example 1: {[369]11 | 1617xxxxxxx} –
Allow 311, 611, 911, and any 10-digit

numbers of leading digits 1617

Example 2: {^1900x+ | <=1617>xxxxxxx} –
Block any number with leading digits 1900 and add prefix 1617 for any dialed 7-digit numbers

Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –
Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing.

Default: Outgoing – {x+}
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 | +x+ }

Explanation of example rule (reading from left to right):

^1900x. – prevents dialing any number started with 1900
<=1617>[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically
1[2-9]xx[2-9]xxxxxx – allows dialing to any US/Canada Number with 11 digits length
011[2-9]x. – allows international calls starting with 011
[3469]11 – allow dialing special and emergency numbers 311, 411, 611 and 911
+x+ – allow dialing any digit with leading + sign; example: +16175669300
Note: In some cases, user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.

Dial Plan Bypass: Users can bypass the dial plan when making a call from call history by enabling the Feature using Pvalues:

Profile 1: P2382=6

Profile 2: P2482=6

Profile 3: P2582=6

Profile 4: P2682=6

Bypass Dial plan

Enable/Disable the dial plan check while making outgoing calls.
The default setting is “No”.

Use # as Dial Key

Treats “#” as the “Send” (or “Dial”) key. If set to “No”, this “#” key can be included as part of the dialed number. Default is Yes.

Use # as Redial Key

If set to “Yes”, the “#” key will immediately
redial the number of the last outgoing call, if no digits have been dialed. The default is Yes.

On Hold Reminder Tone

Supports to disable or enable “On Hold Reminder Tone” to play a reminder tone when a call is on hold.

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 ringtone for the matching rule. When the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring.
Note: Alert info can also be input as a string.

Allow Auto Answer by Call-info/Alert-Info

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls, based on the SIP Call-Info/Alert-Info header sent from the server/proxy. The default setting is “No”

Custom Alert-Info for Auto Answer

Allows the device to automatically answer incoming calls based on specific custom values in the Alert-Info or Call-Info headers of a SIP INVITE message.

Allow Barging for Auto Answer by Call-Info/Alert-Info.

Allows to auto answer a call by Call-Info/Alert-Info, even the phone is current in middle of a call, it is disabled by default.

Enable Call Features

Enables do not disturb, call forward and other call features via the local feature codes on the base. Otherwise, ITSP feature codes can be used. Default is Yes.

Disable Call Waiting Caller ID

Disables displaying caller ID when receiving a second incoming call.
Default is No.

Enable Call Waiting in Parallel Mode

Enables call waiting for accounts using this profile which are set to Parallel ring mode. Default is No.

Disable Visual MWI

Disables use of visual message waiting indicator when there is an unread voicemail message. Default is No.

Transfer on Conference Hangup

Transfers the call to the other party if the conference initiator hangs up.
Default is No.

Ring Timeout

Stops ringing when incoming call is not answered within a specific period of time. Default is 60 seconds.

Hunting Group Ring Timeout

Forwards incoming call to the next member of a hunt group if not answered within a specific period of time. Default is 20 seconds.

Send Anonymous

Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, blocking caller ID. Default is No.

Anonymous Call Rejection

Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Default is No.

Special Feature

Selects Soft switch vendors’ special mode. Example of vendors: Broadsoft, CBCOM, RNK, Huawei, ZTE IME, Phone Power, Metaswitch, Bitrix24, Telstra, and Kandy. Default is Standard.

Music On Hold URI

Configure the music on hold URI to call when a call is on hold if server support it.

Allow Unsolicited REFER

Allow unsolicited REFER to accomplish an outgoing call.

Feature Key Synchronization

When enabled, call features like DND, call forward, call waiting will be synchronized between the server and the phone. It will use NOTIFY to send the status in XML content to server and accept the NOTIFY from the server. This feature following the Broadsoft and MetaSwitch standard. Any server following the same standard will be compatible with this feature.

Call Log

Configure the level of call logs or disable the call log.

General Settings

Base Station Name

Displays the name of the base station. Default is DP75x[last 6 digits of MAC
address].

Admin PIN Code

Configures admin PIN code for authentication. Default is 0000

Enable Repeater Mode

Enables the base station repeater mode to associate with available repeaters. Once enabled the base station starts searching for nearby repeaters and open
subscription to associate with the available repeaters. This option requires
rebooting the base station to take effect. Default is No.

Enable Repeater
Management

Enables base station network management of discovered and paired repeaters.
Once enabled, users need first to reboot the base station to take effect, then login
on the web UI and browser to Status page, a new tab “DECT Repeater Status” will
be available to display discovered and paired devices and also allowing users to
associate / dissociate repeaters and also access their web GUI. Default is No.

DECT PTT Silence
Timer

Sets timeout for PTT call (in minutes) if no handset unmutes. If set to 0, this timer
will be disabled.

Collapse Call History

Enables combining similar call logs into a single entry on the handset display. Calls are merged only if they are of the same type (missed, incoming accepted, or outgoing) and from the same remote party.

The default setting is “No”.

Enable Handset Configuration Control

This setting controls whether the handset can modify the “Mute as DND” option provisioned from the base station. If set to “No”, the “Mute as DND” option will be greyed out on the handset and cannot be changed by the user.

The default setting is “Yes”.

Clear Call Logs

Deletes call history logs of all handsets from base station

Handset Settings

SIP Account Settings

Account

Displays list of accounts’ indexes, from account 1 to account 10.

SIP User ID

Enters SIP user ID provided by VoIP service provider (ITSP). Usually in the form of
digit similar to phone number or actually a phone number.

Authenticate ID

Enters account authenticate ID provided by VoIP service provider (ITSP). Can be
identical to or different from “SIP user ID”.

Password

Specifies account password provided by VoIP provider (ITSP) to register to SIP servers

Name

Chooses a name to be associated to user. Supported length is 256.

Profile

Selects the profile ID (1/2/3/4).

HS Mode

Determines HS modes; the base station supports 4 hunting group modes and 1
non-hunting group:

Active

Activates/deactivates the account.

Handset Line Settings

Handset Line Settings

Configures handset line settings; the base station supports up to 10 SIP accounts,
5 handsets; each Handset can be configured with up to 10 accounts.

Please be aware that the handset line settings will be affected by DID settings
(hunting group settings) in “DECT → SIP Account Settings”. For more details about
Hunting Group, please refer to [Hunting Groups].

Handset Settings (1-5)

Enable Auto Answer

Enables / disables auto answer of incoming calls to handset. Default setting is No.

Enable Mute for Auto Answer

Enables/disables auto mute right after the call been answer, this can be configured by either the handset GUI or Web UI.

Enable Off hook on Cradle Pickup

Enables / disables off hook of handset when picked up from cradle. Default setting is No.

Enable on hook on
Cradle Reposition

Enables / disables on hook of handset when repositioned on cradle. Default setting
is No.

Enable Mute as DND

Enables to use Mute Key, in idle screen, as Global DND Key.

Disable Conference

Enables / disables the conference option on this handset. Default setting is No.

Disable Transfer

Enables / disables transfer option on this handset. Default setting is No.

Disable Busy Tone on Remote Disconnect

Enables / disables the busy tone heard in the handset when call is disconnected
remotely.

Disable Call Waiting Tone

Disables playing call waiting tone during active call when receiving a second incoming call. The CWCID will still be displayed. Default is No.

Play warning tone for Auto Answer
Intercom

Allows to play a warning tone for auto answer for protect the privacy. This setting can be configured by either the handset GUI or the Web UI.

No Key Entry Timeout

Initiates the call within this time interval if no additional key entry during dialing stage. Default is 4 seconds.

Blind transfer wait timeout

Defines the timeout (in seconds) for no key entry during a blind transfer. Default is 30 seconds.

Custom Ringtone

Assigns custom ringtone to specific handset from the ringtones available on the
base station. It takes up to 10 ringtone files which have be named as ring1.bin to
ring10.bin, and you can assign one ringtone to each handset. Default is Disabled.

Time Format

Set the displayed Time Format on handsets to 12 hours or 24 hours. Default is 12hr.

Date Format

Set the displayed Date Format on handsets.

Network Settings – Basic Settings

IP Address Mode

Selects IP address mode (DHCP, Static IP or PPPoE) for DP75x Base Station.

Preferred DNS Server

Specifies preferred DNS server to use when DHCP, PPPoE or Static mode is set.

DHCP Settings

Host name (Option 12)

Specifies the name of the client. The name may or may not be qualified with the local domain name. This field is optional but may be required by ISP.

Vendor Class ID (Option 60)

Exchanges vendor class ID by clients and servers to convey particular configuration or other identification information about a client. Default is DP7XX.

PPPoE Settings

Configures PPPoE account ID, password and service name.

Static IP Settings

Sets IP address, subnet mask, gateway, DNS server 1, and DNS server 2.

Network Settings – Advanced Settings

802.1X Mode

Enables/Disables 802.1X mode. To enable this mode, you should select EAP-MD5. Default is Disable.

802.1X Identity

Configures the identity for 802.1X mode.

MD5 Password

Determines the MD5 password for 802.1X mode.

802.1X CA Certificate

Uploads / deletes the 802.1X CA certificates.

802.1X Client Certificate

Uploads / Deletes the 802.1X Client Certificates.

Enable LLDP

Activates LLDP (Link Layer Discovery Protocol). Default is No.

Layer 2 QoS Settings

Enable VLAN

Enables / Disables the VLAN mode. Default is Disabled.

Layer 2 QoS 802.1Q/VLAN Tag

Sets layer 2 QoS 802.1Q/VLAN tag. Default is 0.

Layer 2 QoS 802.1p Priority Value for SIP signaling

Sets layer 2 QoS 802.1p priority value for SIP signaling.

Default is 0.

Layer 2 QoS 802.1p Priority Value for RTP media

Sets layer 2 QoS 802.1p priority value for RTP media.
Default is 0.

STUN Settings

Use STUN

Enables STUN. Default is No.

STUN server

Configures IP address or domain name of STUN server. Only non-symmetric NAT routers work with STUN.

Number of STUN Response Misses Allowed

Specifies number of STUN response misses allowed before restarting DHCP service. The minimum is 3 misses.

Keep-Alive Interval

Sends periodically a blank UDP packet to SIP server to keep “ping hole” on the NAT router open. Default is 20 seconds.

UPnP Discovery Settings

Enable UPnP discovery