DP720

DP725

DP735

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 (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

Protocols/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, G.722.2/AMR-WB (special order), in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, CNG, PLC, AJB

Telephony Features

Hold, transfer, forward, 3-way conference, push to talk, intercom, downloadable phonebook (XML, LDAP, up to 3000 entries), call waiting, call log (up to 300 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 HTTP/HTTPS or FTP/FTPS, mass provisioning using TR-069 or AES encrypted XML configuration file

Multiple SIP Accounts

Up to twenty (20) distinct SIP accounts per system

Each handset may map to any SIP account(s)

Each SIP account may map to any handsets(s)

Ring Group

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

Humidity: 10% to 90% non-condensing

Compliance

FCC: FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID

CE：EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 60950-1;EN 301 489-1/-6; EN 301 406; EN 50332-2; SAR(EN50360;EN50566;EN 50663;EN62209-1; EN62209-2; EN 62479); RED NB Cert

RCM：AS/NZS CISPR32; AS/NZS 60950.1;AS/CA S004;AS/ACIF S040. ANATEL, EAC, UL(adapter).

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

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 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

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

DP722

DP722

DP730

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

Power and Charging Indication Light

Red: Charging. Green: Charge completed.

Blinking: Missed call(s) or Voice Mail received.

2, 10

Left and right softkeys

Correspond to functions displayed on the LCD. These functions change depending on the current context.

3

4 Arrow key combination

Correspond to functions displayed on the LCD. These functions change depending on the current context.

4

Off-hook / Dial key

Shows call information, handset status icons, prompt messages, etc.

5

Alphanumeric Keypad

Permits navigation of the cursor through the displayed menu options.

6

Hands-free / Speaker key

Selects the option chosen by the cursor. (Enters the main menu from the home screen.)

7

3.5mm Headset jack

Enters dialing mode, or dials number entered.

8

Earphone

Delivers audio output.

11

OK/Selection key

Selects the option chosen by the cursor. (Enters the main menu from the home screen.)

12

On-hook / Power key

Terminates calls, or turns the handset on / off.

13

Mute key

Activates or deactivates the mute feature.

14

Microphone

Picks up audio earpiece and hands-free calls.

Key

Description

1

Earphone

Delivers audio output.

2,12

Left and right softkeys

Correspond to functions displayed on the LCD. These functions change depending on the current context.

3

4 Arrow key combination

Permits navigation of the cursor through the displayed menu options.

4

Off-hook / Dial key

Enters dialing mode, or dials number entered.

5

1 / Voicemail

Provides quick access to your voicemail inbox when pressed.

6

Mute key

Activates or deactivates the mute feature.

8

3.5 mm
headset jack

 Phone connector for the headphones/headsets.

9

LED Indicator

1 dual-color LED indicator indicating power, call, battery, message waiting…

10

Color LCD Screen

1.8 inch (128×160) color TFT LCD

11

Micro USB port

The charging port is used to connect a charger for powering the device and recharging the battery.

13

Menu/OK key

Selects the option chosen by the cursor or enter the main menu from the home screen.

14

Hands-free / Speaker key

Switches between handset and hands-free/speaker modes.

15

On-hook or Power key

Picks up audio earpiece and hands-free calls.

16

 # / Lock key

Locks keypad against unintentional entries when keep pressing #.
 
Press and hold the
 
# key for approximately 2 seconds to lock the keys. Press Unlock Softkey
 
and then # to unlock the keys.

17

Mute

Mute the microphone during the conversation.

18

Microphone

Picks up audio earpiece and hands-free calls.

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

Proximity sensor

The proximity sensor can detect and measure gravitational acceleration, tilt, vibration, altitude changes, and static position.

2

Earphone

Delivers audio output.

3,4

Volume up / Down Keys

Configure the handset and ringtone volume.

5

PTT Key

PTT (Push-to-Talk) button, to initiate a PTT call.

6

Hands-free / Speaker key

Switches between Handset and Hands-free / Speaker modes.

7

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

Allows navigation of the cursor through the displayed menu options.

8

Off-hook / Dial key

Enters dialing mode, or dials number entered.

9

1 / Voicemail Key

Accesses the list of voicemail messages.

10

* / Silent Mode key

Activates or deactivates the silent mode (no ringtone heard during an incoming call) when keep pressing on * in idle screen.

11

Charging port

The charging port is used to connect a charger for powering the device and recharging the battery.

12

3.5 mm headset jack

Phone connector for the headphones/headsets.

13

LED indicator

1 dual-color LED indicator indicating power, call, battery, message waiting…

14

Color LCD Screen

2.4-inch (240×320) TFT color LCD

15

Softkeys

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

16

Mute

 Mute the microphone during the conversation.

17

Menu/OK key

Selects the option chosen by the cursor or enter the main menu from the home screen.

18

On-hook /Power key

Terminates calls or turns the handset on / off.

19

Alphanumeric Keypad

Provides the digits, letters, and special characters in context-sensitive applications. For the + sign, press and hold key 0.

20

# / Lock key

Locks keypad against unintentional entries when keep pressing #.

Press and hold the

# key for approximately 2 seconds to lock the keys. Press Unlock Softkey

and then # to unlock the keys.

21

Microphone

Picks up audio earpiece and hands-free calls.

Key

Description

1

Proximity
Sensor

The proximity sensor can detect and measure gravitational acceleration, tilt, vibration, altitude changes, and static position.

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

Handsfree /Speaker key

Switches between handset and hands-free / speaker modes.

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

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.

9

Alphanumeric keypad

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

10

*/ Silent Mode key

Activates or deactivates the silent mode (no ringtone heard during an incoming call) when keep pressing on * in idle screen.

11

LED indicator

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

12

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.

13

Color LCD Screen

2.4-inch (240×320) TFT color LCD

14

Softkeys

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

15

Mute key

Activates or deactivates the mute feature.

16

Menu/Ok key

Selects the option chosen by the cursor or enter the main menu from the home screen.

17

On-hook or Power key

 
Terminates calls or turns the handset on / off.

18

#/ Lock key

Locks keypad against unintentional entries when keep pressing #.
 
Press and hold the
 
# key for approximately 2 seconds to lock the keys. Press Unlock Softkey
 
and then # to unlock the keys.

19

Microphone

Picks up audio earpiece and hands-free calls.

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

Notes:

Display the call history:

Notes:

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.

Icon

Description

Refresh Button: Allows users to refresh the current page.

Subscribe Button: Allows users to open the subscription.

Paging Button: Allows users to page all the registered DP720/DP722/DP725/DP730/DP735 Handsets.

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.

Ringing 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 DP755_[last 6 digits 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

Displays Global Information about upgrading handsets.

Handset

Displays Handset index.

Name

Displays Handsets names

IPEI

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

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

Page

Sends paging request to corresponding Handsets, 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 Handsets’ firmware version number.

Upgrade

Shows Handsets upgrade status or trigger handset upgrade process.

Line Options

Account

Account index.

SIP User ID

Displays the configured SIP User ID for the account.

DND

DND status of the account. Default No.

Forward

The unconditional forward number.

Busy Forward

The forward number for Call Forward Busy.

Delayed Forward

The no answer delayed forward number.

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 Setting

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

IPv4 Address

Displays assigned IPv4 address.
Example: 192.168.5.110

IPv6 Address

Displays assigned IPv6 address.
Example: 0:0:0:0:0:ffff:c0a8:056e

OpenVPN® IP

Displays OpenVPN® IP address.

Subnet Mask

Displays assigned subnet mask.
Example: 255.255.255.0

Gateway

Displays assigned default gateway.
Example: 192.168.5.1

Primary DNS

Shows assigned Primary DNS server address. Example: 8.8.8.8

Secondary DNS

Shows assigned Secondary DNS server address. Example: 8.8.4.4

PPPoE Link Up

Indicates PPPoE connection status.

NAT Type

Displays the NAT type enabled.

NAT Traversal

Indicates the type of NAT for each account ( 20 accounts in total).

System Info

Product Model

Displays product model info. Default is DP755.

Serial Number

Displays the device’s serial number.

Part Number

Shows product part number. Example: 9610006512A (last 2 digits show HW version, in this example 12A for HW version 1.2A)

Certficate Type

Displays the Certificate type installed on the device.

Security Version

This indicates the security version of the DP755 base station.

Software Version

Boot: Specifies Boot version.
Core: Specifies Core version.
Prog: Specifies Prog version, This is the main firmware release number, which is always used for identifying the software system of the DP755.

CPE: Specifies CPE version (current version is 1.0.4.138)
Handset: Specifies Handset firmware version.

IP Geographic Information

Special Feature

Displays wether the OpenVPN® support is enabled or not.

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.

System Information

Gives the option to download System information

User Space

Shows User sapce used and the database status

Core Dump

Generates core dump by killing programs gs_cmbs and gs_phone.

General Settings

Account Active

Activates or deactivates SIP Account.

Account Name

Determines the name of the account, this account name can also be used in Handsets config provisioning for validation.

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 DP755.

Secondary 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.

Outbound Proxy

Specifies the IP address or domain name of an outbound proxy, a media gateway, or a session border controller. Used by DP755 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work, and only the 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.

SIP User ID

User account information, provided by your VoIP service provider.

Authenticate ID

SIP service subscriber’s Authenticate ID is used for authentication. It can be identical to or different from the SIP User ID.

Authenticate Password

The account password required for the phone to authenticate with the SIP server before the account can be registered.

Name

The SIP server subscriber’s name (optional) that will be used for Caller ID display (e.g., John Doe).

Voice Mail Access Number

Allows users to access voice messages by pressing the MESSAGE button on the phone. This value is usually the VM portal access number.

Network 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.

Maximum Number of SIP Request Retries

Specifies the maximum number of retries on SIP Requests.
Default value is 2.

DNS SRV Failover mode

Configures the preferred method for DNS SRV failover.
If “Default” is selected, the primary SIP server or Outbound Proxy will always be attempted first for all REGISTER and INVITE requests.
If “Use current server until DNS TTL” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until DNS times out.
If “Use current server until no response” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until there is no response.
If “Use current server until failback timer expires” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until the failback timer expires.

Failback Timer

Specifies the time interval (in minutes) that the device should continue to send SIP requests (REGISTER or INVITE) to the failover IP. Once it expires, the SIP requests will be sent to the preferred IP. The default value is 60 minutes, and the max value is 45 days.

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.

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 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.

Proxy-Require

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

Use SBC

Indicates 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

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.

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 the concerned account. The default value is “No”.

Register Expiration

Specifies the frequency (in minutes) when the phone refreshes its registration with the specified registrar. The maximum value is 64800 (about 45 days).
The default value is 60 minutes.

Subscribe Expiration

Specifies the frequency (in minutes) when the phone refreshes its subscription with the specified registrar. The maximum value is 64800 (about 45 days).
The default value is 60 minutes.

Reregister Before Expiration

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

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.

Enable TCP Keep Alive

Ensures continuous monitoring of the connection with connected devices, promptly detecting interruptions and allowing for quick re-establishment of the connection for reliable communication.
Enabled by Default

Local SIP Port

The parameter specifies the local ports that the base station uses for sending and receiving SIP packets. By default, Account 1 uses port 5060, Account 2 uses 5062, Account 3 uses 5064, and Account 4 uses 5066, with increments of two for each subsequent account up to Account 20.

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 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 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 Listening Mode

Determines whether or not to listen to multiple SIP protocols. Dual will listen to TCP when UDP is selected while Dual (Secured) will listen to TLS/TCP when UDP is selected. If TCP or TLS/TCP is selected, UDP will be listened to.
Set to Transport Only by Default.

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”.

Outbound Proxy Mode

In Outbound Proxy mode, SIP messages can include the outbound proxy in the route header, or they can be directly sent to the outbound proxy without the route header.
the options to set are: in route, not in route, and always send to.
Set to in-route by default.

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.

SUBSCRIBE for MWI

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

SUBSCRIBE for Registration

When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically.

Enable 100rel

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

Callee ID Display

When the phone is set to ‘Auto,’ the callee ID in the 180 Ringing will be updated in the order of P-Asserted Identity Header, Remote-Party-ID Header, and To Header. If set to ‘Disabled,’ the callee ID will show as ‘Unavailable.’ Choosing ‘To Header’ will keep the caller ID unchanged and display it as per the To Header

Caller ID Display

When configured as “Auto,” the phone will search for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header, and From Header in the incoming SIP INVITE. If set to “Disabled,” all incoming calls will be shown as “Unavailable.”

Add Auth Header On Initial REGISTER

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

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”.

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.

Custom SIP Headers

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.

Use P-Access-Network-Info Header

Use P-Access-Network-Info header in SIP request.
Enabled by Default.

Use P-Emergency-Info Header

Use P-Emergency-Info header in SIP request.
Enabled by Default.

Use MAC Header

When set to “Only for REGISTER,” the MAC header will only be included in SIP messages for registration and unregistration. If set to “Yes to All SIP,” the MAC header will be included in all outgoing SIP messages. When set to “No,” the MAC header will not be present in any outgoing SIP message.

Add MAC in User-Agent

When set to “Yes except REGISTER,” the phone’s MAC address will be added to the User-Agent header in all outgoing SIP messages, except for REGISTER and UNREGISTER. If set to “Yes to All SIP,” the phone’s MAC address will be included in the User-Agent header of all outgoing SIP messages. When set to “No,” the phone’s MAC address will not appear in the User-Agent header of any outgoing SIP messages.

Advanced Features

PUBLISH for Presence

Enables Presence feature on the phone.
Disabled by Default.

Omit charset=UTF-8 in MESSAGE

Determines whether the base station sends SIP MESSAGE requests without including the “charset=UTF-8” declaration (when enabled) or includes it (when disabled) for specifying the character encoding of the text in the message.
Disabled by Default.

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.

Special Feature

Selects Soft switch vendors’ special mode. Examples of vendors: Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, Huawei IMS, Phonepower, and UCM Call Center.
The default is Standard.

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

Check Domain Certificates

Defines whether the domain certificates will be checked when TLS/TCP is used for SIP Transport.
Disabled by Default

Validate Certificate Chain

Validates certificate chain when TLS/TCP is configured.
Disabled by Default.

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.

Audio Settings

Preferred Vocoder- Choice x

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

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.

Codec Negotiation Priority

Configures the phone to use which codec sequence to negotiate as the callee. When set to “Caller”, the phone negotiates by SDP codec sequence from received SIP Invite; When set to “Callee”, the phone negotiates by audio codec sequence on the phone. The default setting is “Callee”.

Disable Multiple m line in SDP

If enabled, the phone always responds to 1 m line in SDP regardless multiple m lines are offered.
Disabled by Default.

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.

SRTP Key Length

The cipher method / key length to use if SRTP is enabled.

Crypto Life Time

Adds crypto lifetime header to SRTP packets. The default is Yes.

Symmetric RTP

Defines whether symmetric RTP is supported or not. The default setting is “No”.

Silence Suppression

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

Jitter Buffer Type

Selects either Fixed or Adaptive based on network conditions.
Set to Adaptive by Default.

Jitter Buffer Length

Selects Low, Medium, or High based on network conditions.
Set to 300ms by Default

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

Selects encoding rate for G723 codec, the options are:

The default value is 5.3kbps encoding rate

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.

OPUS Payload Type

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

DTMF Payload Type

Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS payload type

Send DTMF

Specifies the mechanism to transmit DTMF digits.
you can choose to do it in-audio, via RTP(RFC2833), or via SIP INFO

Call Settings

Dial Plan Prefix

Configures the prefix added to the 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 starting with 1900
<=1617>[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and the 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, the user wishes to dial strings such as *123 to activate voice mail or other application provided by the service provider. In this case * 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.

Bypass Dial plan

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

On Hold Reminder Tone

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

Call Log

Configure the level of call logs or disable the call log.
By default, it is set to Log all calls.

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.

Refer-To Use Target Contact

If set to “Yes”, the “Refer-To” header uses the transferred target’s Contact header information for attended transfer.

Transfer on Conference Hangup

Defines whether the call is transferred to the other party if the conference initiator hangs up.

Blind Transfer Wait Timeout

Defines the timeout (in seconds) for waiting SIP frag response in blind transfer. Valid range is 30 to 300.

Key As Send

Pressing selected key will immediately dial out.
Pound “#” is the default selection.

RFC2543 Hold

If yes, c=0.0.0.0 will be used in INVITE SDP for hold.
Disabled By Default.

Music On Hold URI

Defines the URI for the server from which the DP75x device will retrive the music on hold file.

Allow Unsolicited REFER

The “Allow Unsolicited REFER” option allows the handset to accept, deny, or require authentification for external REFER requests coming from the SIP server, these requests automatically transfer calls to other parties without user intervention, the available options are:

The default value is Disabled.

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.

Ring Timeout

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

Intercom Settings

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 after a short reminding beep, based on the SIP Call-Info/Alert-Info header sent from the server/proxy.
Enabled by Default.

Allow Barging by Call-Info/Alert-Info

When enabled, the phone will automatically put the current call on hold and answer the incoming calls based on the SIP Call-Info/Alert-Info header sent from the server/proxy. However, if the current call was answered based on the SIP Call-Info/Alert-Info header, then all other incoming calls with SIP Call-Info/Alert-Info headers will be rejected automatically.
Disabled by Default.

Custom Alert-Info for Auto Answer

Used exclusively to match the contents of the Alert-Info header for auto answer. The default auto answer headers will not be matched if this is defined.

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.
Enabled By Default

Account Swap

Swap Account Settings

Swap configurations between two accounts.

General Settings

Base Station Name

Displays the name of the base station. The default is DP755[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 opens a subscription to associate with the available repeaters. This option requires rebooting the base station to take effect. The 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 collapsing of like call logs into a single entry for display on handset. Calls will only be combined if of the same type (missed, incoming accepted, or outgoing) and with the same remote party.

Clear Call Logs

Deletes call history logs of all handsets from base station

Account Assignment

SIP User ID

displays the SIP User ID assigned to the account number.

Ringing Mode

Specifies the ringing mode for the account, the options are:

The default ringing mode is “Parallel”.

HS1-HS10

Handset Settings (1-10)

Handset Name

Displays the handset name.

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 a hook on  Cradle Reposition

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

Enable Mute as DND

Enables users to activate/deactivate Do Not Disturb mode by pressing the Mute Key in idle screen. Default setting is Yes.

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.

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.

Handset Phonebook

Selects the phonebook for the handset , you can select from the 10 XML phonebooks available under Phonebook => Private Phonebook Setting.

Offhook Auto-dial

Enables automatic dialing of a predefined number when a handset goes off-hook (lifted from its cradle) without any user input.

Off-hook Auto Dial Delay

Sets the time delay before initiating automatic dialing when a handset goes off-hook (lifted from its cradle) without user input.

Settings => General Settings

Local RTP Port

This parameter defines the local RTP port used to listen and transmit. The valid range is 1024 to 65400 and it must be even.
Default is 5004