# Phone and terminal settings

The parameters with which a phone/terminal is defined on the communication server can be found here. In the following parameter-related explanations, for reasons of simplicity, we always talk about phones although at times we may talk about other terminals.

Table 1. Quick links to the settings and instructions
Phone type | Settings | Commissioning
Mitel SIP phones (Mitel SIP) | Mitel SIP phones (Mitel SIP) | Registering Mitel SIP corded phones
IP system phones (IP) | IP system phones (IP) | Registering IP system phones
Digital system phones (DSI-AD2) | Digital system phones (DSI-AD2) | Putting a digital system phone into operation
Digital system phones (DASL) | Digital system phones (DASL) |
DECT wireless phones (DECT) | DECT wireless phones (DECT) | Registering DECT cordless phones
MiCollab Client (MiCollab Softphone) | MiCollab Client (MiCollab Softphone) |
Mitel One | Mitel One |
Analogue phones and terminals (Analogue) | Analogue phones and terminals (Analogue) |
Mobile or external phone (Mobile/external) | Mobile or external phone (Mobile/external) |
Mobile phones with MMC (Mitel Mobile Client) | Mobile phones with MMC (Mitel Mobile Client) |
SIP phones and SIP terminals (Standard SIP) | SIP phones and SIP terminals (Standard SIP) | Registering standard SIP phones
ISDN phones and terminals (BRI S-bus) | ISDN phones and terminals (BRI S-bus) |
Virtual phones (Virtual) | Virtual phones (Virtual) |

## Mitel SIP phones

Table 2. Settings to terminal interface Mitel SIP
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).

Table 3. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: This value is not connected to the number of line keys available on the SIP phone.
Conference circuit | It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server. Note: This parameter can only be configured when at least two lines are set up (Multi lines setting).
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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). Note: The value is only significant for the busy lamp field keys for which the parameter Ring with delay is enabled.
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. Note: The value is only significant for the busy lamp field keys for which the parameter Periodic ring is enabled.
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) Note: The value is only significant for the busy lamp field keys for which the parameter Lowest volume is enabled. (Indicated only for Call type = Individual call.)
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.

Table 4. Connection settings
Parameter | Explanation
State | Indicates whether the phone is registered on the communication server and is thus available. Note: If IP-DECT base station is configured for IP-DECT terminals, then the State is changed to Registered. See IP-DECT Base Station Installation and Operational Manual to configure IP-DECT base station on MiVoice Office 400.
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. Note: The Persistent TLS setting is only available if you synchronise the time and date via an NTP time server (this setting is found under System / General).
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. Note: Call forwarding with Response 302 is not possible in every case.
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
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).

Table 5. Display settings (only configurable as of Mitel 6867 SIP and Mitel 6900 SIP)
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. Note: There must be at least 30 minutes difference between the day and night start times.
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 (= 3 minutes).

Table 6. Actions
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. Tip: The idle text can be given the individual user number or user name as placeholder. The placeholders are <No.> and <name>.
Change display language globally | This function can be used to change the display language on all system phones with display at the same time.

"Registering Mitel SIP corded phones" Commissioning

Mitel SIP phones are platform-independent phones with a wide range of features. They can also be perfectly integrated into one of the Mitel Platforms and used as a system phone. Mitel SIP Phones on MiVoice Office 400 first support MiVoice Office 400 features and have a separate user's guide. Many of the device-specific functions are less significant or are not used at all. Please read the Mitel SIP administration instructions if you wish to carry use device-specific functions or carry out device-specific settings. You can find a direct link in the section See also....

## IP system phones (IP)

Table 7. Settings to terminal interface IP
Parameter 1 | 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.

Table 8. Further settings
Parameter 2 | 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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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. Note: The symbol for unanswered calls in the display is not concerned by this setting.

Table 9. Connection settings
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. Note: The Persistent TLS setting is only available if you synchronise the time and date via an NTP time server (this setting is found under System / General).
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
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).

Table 10. Display settings
Parameter 3 | 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. Note: The backlight intensity can be reduced if the phone is not powered via a plug-in power supply unit.
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. Note: An expansion key module M535 must always be powered via a plug-in power supply unit.

Table 11. Actions
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. Tip: The idle text can be given the individual user number or user name as placeholder. The placeholders are <No.> and <name>.
Change display language globally | This function can be used to change the display language on all system phones with display at the same time.

"Registering Mitel SIP corded phones" Commissioning

## Digital system phones (DSI-AD2)

Table 12. Settings to terminal interface DSI-AD2
Parameter 4 | 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.

Table 13. Further settings
Parameter 5 | 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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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. Note: The symbol for unanswered calls in the display is not concerned by this setting.

Table 14. Connection settings
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).

Table 15. Display settings
Parameter 6 | 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. Note: The backlight intensity can be reduced if the phone is not powered via a plug-in power supply unit.
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. Note: An expansion key module M535 must always be powered via a plug-in power supply unit.

Table 16. Actions
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. Tip: The idle text can be given the individual user number or user name as placeholder. The placeholders are <No.> and <name>.
Change display language globally | This function can be used to change the display language on all system phones with display at the same time.

"Registering Mitel SIP corded phones" Commissioning

## Digital system phones (DASL)

Table 17. Settings to terminal interface DASL
Parameter 7 | 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.

Table 18. Further settings
Parameter 8 | 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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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.

Table 19. Actions
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. Tip: The idle text can be given the individual user number or user name as placeholder. The placeholders are <No.> and <name>.
Change display language globally | This function can be used to change the display language on all system phones with display at the same time.

"Registering Mitel SIP corded phones" Commissioning

## DECT cordless phones (DECT)

Table 20. Settings to DECT terminal interface
Parameter 9 | 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)).

Table 21. Further settings
Parameter 10 | 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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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. Note: The symbol for unanswered calls in the display is not concerned by this setting.

Table 22. Display settings
Parameter 11 | 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.

Table 23. DECT settings
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.

Table 24. Actions
Button | Explanation
Login 12 | 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. Tip: The idle text can be given the individual user number or user name as placeholder. The placeholders are <No.> and <name>.
Change display language globally | This function can be used to change the display language on all system phones with display at the same time.

"Registering Mitel SIP corded phones" Commissioning

## Mitel BluStar 8000i and Mitel BluStar Softphones (BluStar)

Table 25. Settings to terminal interface BluStar
Parameter 13 | 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.

Table 26. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: This value is not connected to the number of line keys available on the SIP phone.
Conference circuit | It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server. Note: This parameter can only be configured when at least two lines are set up (Multi lines setting).
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

Table 27. Connection settings
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. Note: The Persistent TLS setting is only available if you synchronise the time and date via an NTP time server (this setting is found under System / General).
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. Note: Call forwarding with Response 302 is not possible in every case.
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
Bandwidth area | Assign the phone here to an already defined bandwidth area.

Table 28. Actions
Button | Explanation
Restart phone | The phone can be restarted with this button.

"Registering Mitel SIP corded phones" Commissioning

## MiCollab Client (MiCollab Softphone)

Table 29. Settings to terminal interface Standard SIP
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.

Table 30. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: This value is not connected to the number of line keys available on the SIP phone.
Conference circuit | It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server. Note: This parameter can only be configured when at least two lines are set up (Multi lines setting).
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

Table 31. Connection settings
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. Note: Call forwarding with Response 302 is not possible in every case.
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
Bandwidth area | Assign the phone here to an already defined bandwidth area.

"Registering Mitel SIP corded phones" Commissioning

## Mitel One

Table 32. Settings to terminal interface Analogue
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.

Table 33. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: This value is not connected to the number of line keys available on the SIP phone.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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. Note: If the analogue port is used as a door intercom system, (FXS mode = 2-wire door), then Force call waiting is always activated.
Special ringing tone | The called user will hear a special ringing tone (changed ring pattern). Note: If the analogue port is used as a door intercom system, (FXS mode = 2-wire door), then Special ringing tone is always activated.
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

Table 34. Connection settings
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
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.

Table 35. Display settings
Parameter | Explanation
Screensaver on time | Enable to keep the phone idle.

## Analogue phones and terminals (Analogue)

Table 36. Settings to terminal interface Analogue
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). Tip: For the setting Polarity reversal, set the switch of the phone (e.g. 6730 Analogue) to the symbol "-". If the MWI LED is blinking (message available) and off (no message available) the switch is set correctly. If the MWI LED is on (message available) and blinking (no message available) the switch is set wrongly.
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)).

Table 37. Further settings
Parameter 14 | 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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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. Note: If the analogue port is used as a door intercom system, (FXS mode = 2-wire door), then Force call waiting is always activated.
Special ringing tone | The called user will hear a special ringing tone (changed ring pattern). Note: If the analogue port is used as a door intercom system, (FXS mode = 2-wire door), then Special ringing tone is always activated.
Transform *7 into *739 | If this analogue terminal sends the function *7, this is converted into *739 in the communication server. Note: This option is meant for the alarm signal solution with special, analogue terminals, which are upgraded to MiVoice Office 400. All other function codes that start with *7 (e.g. *74 Switch control output) can longer be used on these interfaces.
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

Table 38. Connection settings
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. Note: This setting can also be made for analogue interface configuration.

## Mobile or external phone (Mobile/external)

Table 39. Settings for terminal interface Mobile/external
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. Note: For the enhanced functionality GSM voice channels must be configured in the System / DSP.
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.

Table 40. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

## Mobile phones with MMC (Mitel Mobile Client)

Table 41. Settings to terminal interface Mitel Mobile Client
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 |

Table 42. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

Table 43. Connection settings
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. Note: Call forwarding with Response 302 is not possible in every case.
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
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.

## SIP phones and SIP terminals (Standard SIP)

Table 44. Settings to terminal interface Standard SIP
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.

Table 45. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: This value is not connected to the number of line keys available on the SIP phone.
Conference circuit | It can be specified here whether a three-party conference should be executed in the phone itself or in the communication server. Note: This parameter can only be configured when at least two lines are set up (Multi lines setting).
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

Table 46. Connection settings
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. Note: Call forwarding with Response 302 is not possible in every case.
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. Note: Indirect switching requires two additional VoIP channels on the communication server.
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). Note: A licence is required to use MSRP for third-party applications (e.g. SIP clients).
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).

"Registering Mitel SIP corded phones" Commissioning

## ISDN phones and terminals (BRI S-bus)

Table 47. Settings to terminal interface BRI-S-Bus
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.

Table 48. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

Table 49. Connection settings
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.

## Virtual phones (Virtual)

Table 50. Settings to terminal interface Virtual
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.

Table 51. Further settings
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. Note: Alternatively, you can assign the user a preconfigured hotline destination (Configuration / User view). The configuration here (on the phone) takes precedence over the user configuration.
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. Note: If the phone is not assigned an emergency destination, one of the three call numbers of the emergency destination assigned to the node is called. The configuration here (on the phone) takes precedence.
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). Note: This works only if the terminal of the called user does supports this feature (MiVoice 5300 phones or analogue phones, but not for Mitel SIP phones, for example).
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

Parent topic: Terminal administration
1 Dependent on model
2 Dependent on model
3 Dependent on model
4 Dependent on model
5 Dependent on model
6 Dependent on model
7 Dependent on model
8 Dependent on model
9 Dependent on model
10 Dependent on model
11 Dependent on model
12 Dependent on model
13 Dependent on model
14 Dependent on model
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