Phone and terminal settings

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Cordless phone type

Displays the cordless phone type if it is registered (parameter is displayed for Mitel SIP-DECT terminal type only).

Assigned user/pool

Here the phone is assigned to a user or a Free Seating Pool or the assignment is deleted . If you assign the phone to a free seating pool, it is automatically moved to the Free seating terminal view. The phone can only be assigned to one user or one Free seating pool. A new assignment overwrites the old one.

Registration user name, Registration password

Used for registering the phone with the communication server.

Display language

Phone language user interface.

Idle text

Select a predefined value or enter the text to be displayed on the phone display in the idle state. Default: Assigned user name, provided a name is defined during the assignment. Otherwise, the call number is entered.

Idle text 2

Select a predefined value or enter the additional text to be displayed on the phone display in the idle state.

Default: Number of the assigned user, if a name has been defined for this latter at the time of assignment, otherwise the field remains empty.

Phone lock: Set current state

You can see the current phone lock state and can change it here.

Free: The phone is not locked or partially locked (depending on the parameter State when phone is unlocked).

Lock phone: The phone is locked and your private data are protected from viewing. You can configure the call numbers that can be dialled for outgoing calls via special, internal and external digit barring (Users / Permission set view, Settings Internal digit barring (used by phone lock) and External digit barring (used by phone lock)).

State when phone is unlocked

Here you can define whether the lock state of the phone should be free or still partially locked when the phone is unlocked.

Free: The phone is not locked.

Lock phone partially: The system menu on the phone is reduced and some function keys will not work. These partially phone lock is useful for room phones in hospitality environments or for phones in public places. It locks all menus and settings, except call lists, voice mail input, system events and local phone book. Additionally some function keys are locked as well. This means, although key labels are still displayed, pressing on the keys has no effect.

Expansion key module

Depending on phone type, you can add or remove up to three expansion key modules here.

Resource type

Select here the type of TWP expansion module (parameter is displayed for Mitel SIP TWP terminal type only).

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hot­line call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Multi lines

The maximum number of personal lines that can be operated by the SIP phone is specified here. The number of KT-lines and operator lines is calculated in the background and added to this value.

Conference circuit

It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server.

Backup communication server

Select a communication server from the list if you want to operate the Mitel SIP phone on a backup communication server should the primary communication server crash (Dual Homing). If the list is empty you first have to define a backup communication server.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protect­ed against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

Automatic hands-free

Off: The function is switched off.

On: On an internal call, the handsfree device is activated automatically after one ring.

Announcement: The handsfree device is activated automatically for an announcement.

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Busy lamp field: Ring delay (s)

Acoustic ring signal delay on busy lamp field keys (1...30 seconds).

Busy lamp field: Ringing cycle(s)

Ringing cycle for a periodic call (1...30 seconds). If, for instance, the value is set to 8 seconds, a ring tone is emitted on the busy lamp field keys every 8 seconds.

Busy lamp field: Ring attenuation

Attenuation of acoustic ring signals on busy lamp field keys compared to the ring volume currently set on the phone.

1 = lowest attenuation (highest volume)

9 = highest attenuation (lowest volume)

Call list type

(For Mitel 6867 SIP, Mitel 6869 SIP and Mitel 6873 SIP only)

Picture ID: The device's local call list is used with some slight modifications and synchronised with the central call list in the communication server. This also allows contact pictures to be displayed, if they are correctly stored on the connected picture server. You can find out how to store the pictures on the picture server in the Mitel 6800 SIP phone administration instructions. You can find the direct link to the instructions in the section See also....

Enhanced: The central call list of the communication server is used.

Parameter

Explanation

State

Indicates whether the phone is registered on the communication server and is thus available.

Re-register phone button

If a phone has already been registered at another location, old invisible registration credential residues may prevent re-registration.

To successfully register the phone, click Re-register phone then restart the phone.

IP address

Shows the phone IP address, if it is registered on the communication server.

SIP port

The port for the SIP signalling data is displayed here.

RTP port

RTP port used to transmit voice data. Default value is 1024. Must not be changed, as a rule.

MAC address

The MAC address is a unique phone identification and is used by the system to assign the phone to a configuration profile.

MBG Controller

Choose a MBG Controller from the list, if the phone is used as a teleworker through a Mitel Border Gateway.

SIP user name

String of characters generated by WebAdmin. It can be edited, but must be specific. If possible, WebAdmin uses the user's call number assigned to the phone.

SIP password

Random character sequence generated by WebAdmin. It can be edited, but must be specific.

MBG SIP user name

If the phone is used as a teleworker through a Mitel Border Gateway this additional string of characters is generated by WebAdmin, so the phone can register towards the MBG. It can be edited, but must be specific. If possible, WebAdmin uses the user's call number assigned to the phone.

MBG SIP password

If the phone is used as a teleworker through a Mitel Border Gateway this additional random character sequence is generated by WebAdmin, so the phone can register towards the MBG. It can be edited, but must be specific.

Transport protocol

Select Persistent TLS and restart the phone if the connection to the phone should be encrypted.

Terminal is behind NAT

Set up this parameter , if the phone is on another subnet and is only accessible via NAT router.

Enable keep alive

The communication server periodically sends messages to the SIP phone (OPTIONS) in order to maintain the NAT connection. This is necessary for example if the SIP phone is connected to the communication server via the public IP network.

Send redirecting information

Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP provider as well as for each SIP phone.

No: No redirection/redirecting information is displayed.

Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server.

Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection information on its own display.

Relay RTP data via communication server

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Direct switching: During connection set-up to another IP end­point, the voice data is forwarded directly.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

Active line supervision (using session refresh)

Checks, with the session refresh method at regular intervals, whether connection to the remote station is still active. If the remote station does not react within the defined session refresh period, the connection is cut.

VLAN

The phone's tagged VLAN allocation in accordance with IEEE 802.1/Q. Assign a VLAN ID.

The phone is not assigned any VLAN ID or the assignment is port-based on the used switch.

VLAN ID

ID of the VLAN to which the phone is to be allocated (values between 1 and 4094). The chosen VLAN ID must match the communication server VLAN ID.

VLAN-PC

Tagged VLAN allocation of the PC interface on the phone in accordance with IEEE 802.1/Q. Assign a VLAN ID.

The PC interface on the phone is not assigned any VLAN.

VLAN-PC ID

ID of the VLAN to which the PC interface on the phone is to be allocated (values between 1 and 4094).

Parameter

Explanation

Active backlight level

Define the backlight level when the phone is in use. The default value is five.

Backlight on time (s)

After the phone switches to idle, the backlight level should stay on active level for a certain time. This time duration can be configured here. The default value is 30 seconds.

Idle backlight level for day

The idle backlight level for the day can be configured here. The default value is one.

Idle backlight level for night

The idle backlight level for the night can be configured here. The default value is one.

Backlight day-night

The phone has different idle backlight settings for day and night.

Backlight day start

Defines when the day starts (only used for backlight settings). The default value is 07:00.

Backlight night start

Defines when the nights starts (only used for backlight settings). The default value is 22:00.

Screensaver on time (s)

Set the duration for how long a phone is idle before the screen saver comes on. The default value is 1800 seconds (= 30 minutes).

Button

Explanation

Restart phone

The changes made to some parameters only take effect once the Mitel SIP phone has been restarted.

Restart all Mitel SIP phones.

Instead of restarting each Mitel SIP phone individually, you can restart all registered Mitel SIP phones via this function.

Set idle text globally

Instead of entering the idle text individually for each phone, this function can be used to set the idle text globally for all system phones with display.

Change display language globally

This function can be used to change the display language on all system phones with display at the same time.

Parameter¹

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here the phone is assigned to a user or a Free Seating Pool or the assignment is deleted . If you assign the phone to a free seating pool, it is automatically moved to the Free seating terminal view. The phone can only be assigned to one user or one Free seating pool. A new assignment overwrites the old one.

Registration code

You can use this to register the phone on the communication server and to assign the configuration profile you want.

Enter this number on the phone when prompted to do so during the registration process. Default value is the call number of the allocated user or a blank entry. Alternatively you can also make the allocation by entering the phone’s MAC address (see MAC address setting in the connection settings).

Display language

Phone language user interface.

Idle text

Select a predefined value or enter the text to be displayed on the phone display in the idle state. Default: Call number and name of the assigned user, provided the user is entered at the time of the assignment; if not, only the call number is entered.

Phone lock: Set current state

Free: The phone is not locked.

Lock settings: The configuration menu is locked.

Lock phone partially: The configuration menu is locked and some menu points are hidden.

Lock phone: The phone is locked and your private data are protected from viewing. You can configure the call numbers that can be dialled for outgoing calls via special, internal and external digit barring (Users / Permission set view, Settings Internal digit barring (used by phone lock) and External digit barring (used by phone lock)).

Bluetooth module

The MiVoice 5380 / 5380 IP system phone fitted with an EZURIO BISM2 type Bluetooth module. Here you can see whether the phone has a Bluetooth module and in which state it is.

Expansion key module

Depending on phone type, you can add or remove up to three expansion key modules here.

Parameter²

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

Headset

Headset operation on/off

Automatic hands-free

Off: The function is switched off.

On: On an internal call, the handsfree device is activated automatically after one ring.

Announcement: The handsfree device is activated automatically for an announcement.

Discreet ring

The phone rings only once. However, you can also still take the call after the first ring.

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

DTMF automatic

On: DTMF is switched on as standard and can be deactivated for each individual call.

Off: DTMF is switched off as standard and can be activated for each individual call.

Pop-up window for unanswered calls

Enable Unanswered calls view for system phones.

Parameter

Explanation

State

Indicates whether the phone is registered on the communication server and is thus available.

IP address

Shows the phone IP address, if it is registered on the communication server.

RTP port

RTP port used to transmit voice data. Default value is 30000. Must not be changed, as a rule.

MAC address

MAC address of the IP system phone. Read automatically during registration.

The configuration profile is assigned to the phone using this MAC address. Delete it if you wish to cancel the allocation of the terminal to the terminal data.

As an alternative to registering the phone using the registration code you can enter the phone's MAC address manually at this point.

Transport protocol

Select Persistent TLS and restart the phone if the connection to the phone should be encrypted.

Relay RTP data via communication server

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Direct switching: During connection set-up to another IP end­point, the voice data is forwarded directly.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

VLAN

The phone's tagged VLAN allocation in accordance with IEEE 802.1/Q. Assign a VLAN ID.

The phone is not assigned any VLAN ID or the assignment is port-based on the used switch.

VLAN ID

ID of the VLAN to which the phone is to be allocated (values between 1 and 4094). The chosen VLAN ID must match the communication server VLAN ID.

VLAN-PC

Tagged VLAN allocation of the PC interface on the phone in accordance with IEEE 802.1/Q. Assign a VLAN ID.

The PC interface on the phone is not assigned any VLAN.

VLAN-PC ID

ID of the VLAN to which the PC interface on the phone is to be allocated (values between 1 and 4094).

Calling party info E.164 compliant

The Standard SIP phone requires the E.164 format (i.e. canonical format) in the “from”, “contact” and “PAI” header of the appropriate SIP message (e.g. INVITE).

Parameter³

Explanation

Display contrast

You can set the screen display contrast here. The optimum setting varies according to view angle.

Backlighting

You can set the screen backlighting here. Different lighting modules are available, depending on the models. The setting is applied for connected expansion key module M535.

Backlight intensity

You can set the screen backlight intensity here.

Screensaver

On MiVoice 5380 IP the screensaver can be activated and is displayed on screen a few minutes after idle state. You have a choice between a right-angle and a round clock.

M535: Display contrast

You can set the display contrast of connected expansion key modules M535 here. The optimum setting varies according to view angle.

M535: Backlight intensity

You can set the backlight intensity of connected expansion key modules M535 here.

Button

Explanation

Restart phone

The phone can be restarted with this button.

Set idle text globally

Instead of entering the idle text individually for each phone, this function can be used to set the idle text globally for all system phones with display.

Change display language globally

This function can be used to change the display language on all system phones with display at the same time.

Parameter⁴

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned port

Here the phone is assigned to a user or a physical connector or the assignment is deleted . The phone can be assigned to one port only. A new assignment overwrites the old one. Up to two phones can be connected to one DSI-AD2 interface.

Assigned user/pool

Here the phone is assigned to a user or a Free Seating Pool or the assignment is deleted . If you assign the phone to a free seating pool, it is automatically moved to the Free seating terminal view. The phone can only be assigned to one user or one Free seating pool. A new assignment overwrites the old one.

Display language

Phone language user interface.

Idle text

Select a predefined value or enter the text to be displayed on the phone display in the idle state.

Default: Call number and name of the assigned user, provided the user is entered at the time of the assignment; if not, only the call number is entered.

Phone lock: Set current state

Free: The phone is not locked.

Lock settings: The configuration menu is locked.

Lock phone partially: The configuration menu is locked and some menu points are hidden.

Lock phone: The phone is locked and your private data are protected from viewing. You can configure the call numbers that can be dialled for outgoing calls via special, internal and external digit barring (Users / Permission set view, Settings Internal digit barring (used by phone lock) and External digit barring (used by phone lock)).

Bluetooth module

The MiVoice 5380 / 5380 IP system phone fitted with an EZURIO BISM2 type Bluetooth module. Here you can see whether the phone has a Bluetooth module and in which state it is.

Expansion key module

Depending on phone type, you can add or remove up to three expansion key modules here.

Parameter⁵

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

Headset

Headset operation on/off

Automatic hands-free

Off: The function is switched off.

On: On an internal call, the handsfree device is activated automatically after one ring.

Announcement: The handsfree device is activated automatically for an announcement.

Discreet ring

The phone rings only once. However, you can also still take the call after the first ring.

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

DTMF automatic

On: DTMF is switched on as standard and can be deactivated for each individual call.

Off: DTMF is switched off as standard and can be activated for each individual call.

Pop-up window for unanswered calls

Enable Unanswered calls view for system phones.

Button

Explanation

Calling party info E.164 compliant

The Standard SIP phone requires the E.164 format (i.e. canonical format) in the “from”, “contact” and “PAI” header of the appropriate SIP message (e.g. INVITE).

Parameter⁶

Explanation

Display contrast

You can set the screen display contrast here. The optimum setting varies according to view angle.

Backlighting

You can set the screen backlighting here. Different lighting modules are available, depending on the models. The setting is applied for connected expansion key module M535.

Backlight intensity

You can set the screen backlight intensity here.

Screensaver

On MiVoice 5380 the screensaver can be activated and is displayed on screen a few minutes after idle state. You have a choice between a right-angle and a round clock.

M535: Display contrast

You can set the display contrast of connected expansion key modules M535 here. The optimum setting varies according to view angle.

M535: Backlight intensity

You can set the backlight intensity of connected expansion key modules M535 here.

Button

Explanation

Set idle text globally

Instead of entering the idle text individually for each phone, this function can be used to set the idle text globally for all system phones with display.

Change display language globally

This function can be used to change the display language on all system phones with display at the same time.

Parameter⁷

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned port

Here the phone is assigned to a user or a physical connector or the assignment is deleted . The phone can be assigned to one port only. A new assignment overwrites the old one. Only one phone can be connected to a DASL interface.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

Display language

Phone language user interface.

Idle text

Select a predefined value or enter the text to be displayed on the phone display in the idle state. Default: Call number and name of the assigned user, provided the user is entered at the time of the assignment; if not, only the call number is entered.

Phone lock: Set current state

Free: The phone is not locked.

Lock settings: The configuration menu is locked.

Lock phone partially: The configuration menu is locked and some menu points are hidden.

Lock phone: The phone is locked and your private data are protected from viewing. You can configure the call numbers that can be dialled for outgoing calls via special, internal and external digit barring (Users / Permission set view, Settings Internal digit barring (used by phone lock) and External digit barring (used by phone lock)).

Expansion key module

Depending on phone type, you can add or remove up to three expansion key modules here.

Parameter⁸

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

Automatic hands-free

Off: The function is switched off.

On: On an internal call, the handsfree device is activated automatically after one ring.

Announcement: The handsfree device is activated automatically for an announcement.

Discreet ring

The phone rings only once. However, you can also still take the call after the first ring.

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

DTMF automatic

On: DTMF is switched on as standard and can be deactivated for each individual call.

Off: DTMF is switched off as standard and can be activated for each individual call.

Button

Explanation

Set idle text globally

Instead of entering the idle text individually for each phone, this function can be used to set the idle text globally for all system phones with display.

Change display language globally

This function can be used to change the display language on all system phones with display at the same time.

Parameter⁹

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here the phone is assigned to a user or a Free Seating Pool or the assignment is deleted . If you assign the phone to a free seating pool, it is automatically moved to the Free seating terminal view. The phone can only be assigned to one user or one Free seating pool. A new assignment overwrites the old one.

Display language

Phone language user interface.

Idle text

Select a predefined value or enter the text to be displayed on the phone display in the idle state. Default: Call number and name of the assigned user, provided the user is entered at the time of the assignment; if not, only the call number is entered.

Phone lock: Set current state

Free: The phone is not locked.

Lock settings: The configuration menu is locked.

Lock phone partially: The configuration menu is locked and some menu points are hidden.

Lock phone: The phone is locked and your private data are protected from viewing. You can configure the call numbers that can be dialled for outgoing calls via special, internal and external digit barring (Users / Permission set view, Settings Internal digit barring (used by phone lock) and External digit barring (used by phone lock)).

Parameter¹⁰

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

One hotkey only

Only one hotkey is available, that is you can store only one call number or function on the key. This configuration is suitable, in particular, if you wish to trigger a function by pressing a key (e.g. an alarm).

Six hotkeys are available, that is you can store six call numbers or functions on the key.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When a emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protect­ed against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

Discreet ring

The phone rings only once. However, you can also still take the call after the first ring.

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Key confirmation tone

Each time a key is pressed, the phone acknowledges it with a tone.

Direct answer

If the phone is inside a holster when a call comes in, answer the call simply by removing the phone from the holster without pressing the call key.

DTMF automatic

On: DTMF is switched on as standard and can be deactivated for each individual call.

Off: DTMF is switched off as standard and can be activated for each individual call.

Pop-up window for unanswered calls

Enable Unanswered calls view for system phones.

Parameter¹¹

Explanation

Display contrast

You can set the display contrast here. The optimum setting varies according to view angle.

Backlighting

You can set the screen backlighting here.

Parameter

Explanation

State

Subscribed / Not subscribed / Ready to subscribe

Status display. Indicates whether or not a cordless phone has subscribed to the cordless phone.

Diverse version details

Shows the HW and software version information of the DECT cordless phone (not available in all models).

ID cordless phone

This ID is issued when the cordless phone is opened and is used for specific assignment.

Button

Explanation

Login¹²

Starts the login procedure for cordless phones.

Set idle text globally

Instead of entering the idle text individually for each phone, this function can be used to set the idle text globally for all system phones with display.

Change display language globally

This function can be used to change the display language on all system phones with display at the same time.

Parameter¹³

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Assigned user display.

• An Mitel BluStar 8000i is not permanently assigned to a user. It is only assigned if a user logs on to a phone with a user name and password or PIN.
• A BluStar Softphone must be assigned in the user view in the Multimedia table.

Display language

Phone language user interface.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Multi lines

The maximum number of personal lines that can be operated by the phone is specified here. The number of KT-lines and operator lines is calculated in the background and added to this value.

Conference circuit

It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server.

Emergency destination

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

IP address

Shows the phone IP address, if it is registered on the communication server.

SIP port

The port for the SIP signalling data is displayed here.

RTP port

RTP port used to transmit voice data. Must not be changed, as a rule.

MAC address

The MAC address is a unique phone identification and is used by the system to assign the phone to a configuration profile.

Transport protocol

Select Persistent TLS and restart the phone if the connection to the phone should be encrypted.

Terminal is behind NAT

Set up this parameter , if the phone is on another subnet and is only accessible via NAT router.

Enable keep alive

The communication server periodically sends messages to the SIP phone (OPTIONS) in order to maintain the NAT connection. This is necessary for example if the SIP phone is connected to the communication server via the public IP network.

Send redirecting information

Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP provider as well as for each SIP phone.

No: No redirection/redirecting information is displayed.

Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server.

Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection information on its own display.

Relay RTP data via communication server

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Direct switching: During connection set-up to another IP end­point, the voice data is forwarded directly.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

Button

Explanation

Restart phone

The phone can be restarted with this button.

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here you see the assigned user. The phone can be assigned to one user only.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Multi lines

The maximum number of personal lines that can be operated by the SIP phone is specified here. The number of KT-lines and operator lines is calculated in the background and added to this value.

Conference circuit

It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

State

Indicates whether the phone is registered on the communication server and is thus available.

Overwrite registration

Configured when multiple clients are supported for a user. The configuration in this field indicates if the user should always remain logged in to the MiCollab client (default for mobile phones, tablets) or should be logged out of the client (default for PC and WebRTC client). When Overwrite registration is set to “Always”, when a user logs into another instance of the MiCollab client, the first instance automatically gets logged out.

MiCollab client type

Indicates the type of client configured for the terminal interface.

User agent string

User agent header received in the SIP REGISTER message. Includes user information such as the UC endpoint, its version, device, OS version and other such depending on the client device user.

IP address

Shows the phone IP address, if it is registered on the communication server.

SIP port

The port for the SIP signalling data is displayed here.

RTP port

The port for the RTP data is displayed here.

MAC address

MAC address associated with the user agent logged in on the terminal. It is a read-only field, provided for information purposes only.

SIP user name

String of characters generated by WebAdmin. It can be edited, but must be specific. If possible, WebAdmin uses the user's call number assigned to the phone.

SIP password

Random character sequence generated by WebAdmin. It can be edited, but must be specific.

Transport protocol

Transport protocol used to establish the connection with the communication server.

Terminal behind NAT

Indicates that the terminal is behind a Network Address Translation (NAT) server.

Enable keep alive

The communication server periodically sends messages to the SIP phone (OPTIONS) in order to maintain the NAT connection. This is necessary for example if the SIP phone is connected to the communication server via the public IP network.

Send redirecting information

Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP provider as well as for each SIP phone.

No: No redirection/redirecting information is displayed.

Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server.

Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection information on its own display.

Relay RTP data via communication server

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Direct switching: During connection set-up to another IP end­point, the voice data is forwarded directly.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

The phone can be assigned to one user only.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Multi lines

The maximum number of personal lines that can be operated by the SIP phone is specified here. The number of KT-lines and operator lines is calculated in the background and added to this value.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

State

Indicates whether the phone is registered on the communication server and is thus available.

IP address

Shows the phone IP address, if it is registered on the communication server.

SIP user name

String of characters generated by WebAdmin. If possible, WebAdmin uses the user's call number assigned to the phone.

SIP password

Random character sequence generated by WebAdmin.

Transport protocol

TCP (not configurable)

Enable keep alive

The communication server periodically sends messages to the SIP phone (OPTIONS) in order to maintain the NAT connection. This is necessary for example if the SIP phone is connected to the communication server via the public IP network.

Relay RTP data via communication server (indirect switching)

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

Active line supervision (using session refresh)

Checks, with the session refresh method at regular intervals, whether connection to the remote station is still active. If the remote station does not react within the defined session refresh period, the connection is cut.

Parameter

Explanation

Screensaver on time

Enable to keep the phone idle.

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned port

Here the phone is assigned to a user or a physical connector or the assignment is deleted . The phone can be assigned to one port only. A new assignment overwrites the old one. Only one phone can be connected to an analogue interface.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

Display language

Phone language user interface.

MWI mode

The setting needed to display the notification depends on the type of communication server and the connected, analogue phone. The MiVoice Office 400 communication servers support frequency shift keying (FSK) and Low voltage (Low voltage is mainly used for phones in USA/Canada). Mitel 470 and Mitel SMBC also supports Polarity reversal. Additionally Mitel SMBC supports High Voltage. Some analogue phones also have an MWI switch (e.g. Mitel 6730 Analogue).

Phone lock: Set current state

Free: The phone is not locked.

Lock settings: The configuration menu is locked.

Lock phone: The phone is locked and your private data are protected from viewing. You can configure the call numbers that can be dialled for outgoing calls via special, internal and external digit barring (Users / Permission set view, Settings Internal digit barring (used by phone lock) and External digit barring (used by phone lock)).

Parameter¹⁴

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pat­tern).

Transform *7 into *739

If this analogue terminal sends the function *7, this is converted into *739 in the communication server.

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

Fax device

This parameter lets you configure the type of device on the analogue interface:

No fax device: Terminal is not a fax device. Voice connection is established.

Fax device (T.38): Fax device without voice and voice mail system. For connections via IP a T.38 connection is set up whenever possible.

Combo device (voice/T.38): Fax device with voice and/or voice mail system. A voice connection is established first. When transmitting fax data, it is best to switch over to a T.38 connection whenever possible in the case of connections over IP.

Fax over VoIP (G.711): Transmitting fax data as voice data on the IP network. The G.711 protocol is always used.

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

Route

This route is used if the integrated mobile phone’s internal call number is dialled and an external call is then made to the stored mobile call number.

Mobile/external call number

Enter the phone's external call number here.

Use CLIP for authentication

The call number and password entered to authenticate the integrated mobile phone is not necessary.

CLIP selection

Normal: If the direct dialling number of the calling user is used as the CLIP, regardless of his settings. If there is no corresponding direct dialling number, the internal call number is used in its place.

CLIP from user: The CLIP number will soon be created, like for a call to the public network. In this case the calling user’s CLIP settings are decisive.

Enhanced functionality

The extended function reinforces the integration of the mobile phone and allows additional suffix dialling functions, e.g. enquiry call or conference set-up.

MWI route

Route for MWI signalling (new voice mail voice message) to the integrated mobile phone.

MWI CLIP

The CLIP for MWI signalling (new voice mail voice message) to the integrated mobile phone is entered here. (input format the same as an external call number e.g. 00326553827). The CLIP is transmitted in accordance with the Transit CLIP format parameter for the trunk group settings.

Enquiry call using DTMF-A

Depending on its configuration, an enquiry call on a mobile or external phone is triggered with the DTMF character A or with the DTMF characters ***. The communication server interprets both of them as an enquiry call (default setting). However, the user's voice may be interpreted as DTMF signal A and involuntarily trigger an enquiry call. To avoid this, configure the mobile or external phone in such a way that it sends the DTMF signal *** for the enquiry call then deactivate the parameter Enquiry call using DTMF-A.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

MMC Controller

Assign the mobile phone an open MMC controller here. MMC controllers are opened in the Services / MMC controller view.

Mobile call number

Enter the external call number of the mobile phone here.

Static roaming

Call waiting

Call reverse

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

IP address

Shows the mobile phone IP address

SIP port

The port for the SIP signalling data is displayed here.

SIP user name

String of characters generated by WebAdmin. It can be edited, but must be specific. If possible, WebAdmin uses the user's call number assigned to the phone.

SIP password

Random character sequence generated by WebAdmin. It can be edited, but must be specific.

Transport protocol

UDP or TCP (not configurable)

Terminal is behind NAT

Set up this parameter, if the phone is on another subnet and is only accessible via NAT router.

Enable keep alive

The communication server periodically sends messages to the SIP phone (OPTIONS) in order to maintain the NAT connection. This is necessary for example if the SIP phone is connected to the communication server via the public IP network.

Send redirecting information

Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP provider as well as for each SIP phone.

No: No redirection/redirecting information is displayed.

Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server.

Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection information on its own display.

Relay RTP data via communication server

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Direct switching: During connection set-up to another IP end­point, the voice data is forwarded directly.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

Active line supervision (using session refresh)

Checks, with the session refresh method at regular intervals, whether connection to the remote station is still active. If the remote station does not react within the defined session refresh period, the connection is cut.

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

Force UDP usage

Prevents ring delays with Standard SIP phones which only handle UDP but not TCP.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Multi lines

The maximum number of personal lines that can be operated by the SIP phone is specified here. The number of KT-lines and operator lines is calculated in the background and added to this value.

Conference circuit

It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

State

Indicates whether the phone is registered on the communication server and is thus available.

IP address

Shows the phone IP address, if it is registered on the communication server.

SIP port

The port for the SIP signalling data is displayed here.

MBG Controller

Choose a MBG Controller from the list, if the phone is used as a teleworker through a Mitel Border Gateway.

SIP user name

String of characters generated by WebAdmin. It can be edited, but must be specific. If possible, WebAdmin uses the user's call number assigned to the phone.

SIP password

Random character sequence generated by WebAdmin. It can be edited, but must be specific.

MBG SIP user name

If the phone is used as a teleworker through a Mitel Border Gateway this additional string of characters is generated by WebAdmin, so the phone can register towards the MBG. It can be edited, but must be specific. If possible, WebAdmin uses the user's call number assigned to the phone.

MBG SIP password

If the phone is used as a teleworker through a Mitel Border Gateway this additional random character sequence is generated by WebAdmin, so the phone can register towards the MBG. It can be edited, but must be specific.

Used transport protocol

UDP or TCP (not configurable)

Enable keep alive

The communication server periodically sends messages to the SIP phone (OPTIONS) in order to maintain the NAT connection. This is necessary for example if the SIP phone is connected to the communication server via the public IP network.

Send redirecting information

Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP provider as well as for each SIP phone.

No: No redirection/redirecting information is displayed.

Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server.

Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection information on its own display.

Relay RTP data via communication server

Indirect switching: During connection set-up to another IP end­point, the voice data is forwarded via the communication server and not directly. This can help in solving NAT and firewall problems.

Direct switching: During connection set-up to another IP end­point, the voice data is forwarded directly.

Fax device

If the SIP terminal is a fax machine or a combined device (fax/phone), select the type of device. The phone application always takes priority over the fax application.

Bandwidth area

Assign the phone here to an already defined bandwidth area.

Instant messages supported (MS­RP)

MSRP is a protocol used to exchange data between SIP clients.

If the remote station also supports MSRP, online news, for instance, can be exchanged (chat).

Terminal supports session replacement

Active line supervision (using session refresh)

Checks, with the session refresh method at regular intervals, whether connection to the remote station is still active. If the remote station does not react within the defined session refresh period, the connection is cut.

Calling party info E.164 compliant

The Standard SIP phone requires the E.164 format (i.e. canonical format) in the “from”, “contact” and "PAI" header of the appropriate SIP message (e.g. INVITE).

Allows MWI notification without subscription

The SIP phone does gets a message waiting indication (MWI) via the “SIP notify message” without prior sending a “SIP Subscribe for MWI message”.

Use SAVP for SRTP

The Standard SIP phone supports Secure Audio Video Profile (SAVP). This is specified in the media line (m-line) of the session description protocol (SDP).

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned port

Here the phone is assigned to a user or a physical connector or the assignment is deleted . The phone can be assigned to one port only. A new assignment overwrites the old one. Up to eight phones can be connected to one BRI-S interface.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN

Parameter

Explanation

Fax device

This parameter lets you configure the type of device on the BRI-S interface:

No fax device: Terminal is not a fax device. Voice connection is established.

Fax device (T.38): Fax device without voice and voice mail system. For connections via IP a T.38 connection is set up whenever possible.

Combo device (voice/T.38): Fax device with voice and/or voice mail system. A voice connection is established first. When transmitting fax data, it is best to switch over to a T.38 connection whenever possible in the case of connections over IP.

Fax over VoIP (G.711): Transmitting fax data as voice data on the IP network. The G.711 protocol is always used.

Parameter

Explanation

Terminal type

Phone model designation.

Description

You can type in a help text here to designate the phone according to your criteria. This text is only displayed here.

Assigned user/pool

Here the phone is assigned to a user or the assignment is deleted . The phone can be assigned to one user only. A new assignment overwrites the old one.

Parameter

Explanation

Hotline call number

If you enter a call number here, it is dialled as soon as the user seizes a line and the hotline delay has elapsed. You can also enter just part of the call number. The user can then enter the rest of the call number himself.

Hotline delay(s)

Delay between the configuration and automatic dialling of the hotline call number. If the user dials a call number manually during this time (or a part thereof), then the hotline call number is not called.

Emergency destinations

Here you can assign any of the defined emergency destinations to the phone. When an emergency number of the internal numbering plan is dialled, one of the three call numbers of the emergency destination is dialled, based on the switch position of the assigned switch group.

Emergency location

Here you can assign any of the defined emergency location data sets to the phone. When an emergency number out of the public emergency number list is dialled, the system reacts with specific actions: The location of the caller is sent to the provider, an emergency response team is informed, alarms are issued and logs are updated. You can find more information in the focus topic "Emergency service support".

Force call waiting

The called user is notified via call waiting, even if they are protected against it.

Special ringing tone

The called user will hear a special ringing tone (changed ring pattern).

PSTN overflow

No: No PSTN overflow can be made using this phone.

If necessary: On this phone, the connection is set up via PSTN when the connection cannot be set up as normal via the IP network (the PSTN overflow must be set up)

Permanently: On this phone, all connections are set up via PSTN and not via the IP network (the PSTN overflow must be set up).

Region

Region of the phone in the AIN