Mitel6863 (SDM7)


This resource allows to connect a Mitel 6863 IP Phone.

Release notes

Version 2.0.0 - Early deployment
  • Feature: full sync_code rewriting for source SOP configuration and FMA support (PC-3420)
  • Bugfix: the language of the Mitel phones must be the one of the user (PC-3401)
  • Bugfix: call forward must be disabled on Mitel phones (PC-3400)
  • Bugfix: the name of the screen of the Mitel phones needed to be inline with Fusion Policy (PC-3405)
  • Bugfix: admin password of the Mitel phone should be the same than the one for the Polycom (PC-3402)
  • Dependency:
    • Mitel Phone Support module >= 1.4.0

Version 1.0.1 - Early deployment
  • Feature: Initial version (PC-516)
  • Dependency:
    • Mitel Phone Support module > 1.2.0

Resource configuration interface

GUI unavailable.

Resource standing for a Mitel 6863i SIP Phone.

A noticeable difference with the higher models 6867i/6869i is the absence of softkeys. The LDAP integration to query internal directory is disabled as well because this model only supports loading whole directory at once, which is too dangerous.

Resource configuration parameters

  • Display: Format the line name on the main screen
  • SIP server feature: Note: These technical settings if improperly configured can have impact on the availability of the phone
    • SIP re-invite: allow the voice media to be transfered directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
      • Default: Keep default settings as defined in the Telephony server module (for example Asterisk 1.2.x)
      • Yes: Allow the server to perform the re-invite procedure.
      • No: Do not allow the server to perform the re-invite procedure.
    • Qualify SIP devices registration: In order to monitor the availability of the phone, the SIP server can send SIP OPTIONS message to monitor the SIP stack availability. This parameter configures the frequency of this SIP OPTION message. Possible value are:
      • Default: Keep default settings as defined in the Telephony server module (for example Asterisk 1.2.x)
      • Yes: Put a default value of 2000ms
      • Another value in ms.
    • Force specific codec: If this parameter is set to 'Yes', the SIP server will only accept the 3 codecs defined in this resource with the specified preference.
    • First Codec, Second Codec, Third Codec: Ordered list of preferred codecs.
  • Advanced Options:
    • Timezone: Timezone where the phone is located
    • Automatic Off-hook Call Placement: If set to enabled, the phone will automatically dial the Contact to call when the handset is off hook.
  • Network Settings: ERROR! Important note : The following network parameters can break the connection between the phone and the SOP. If incorrect values are set, a factory reset of the phone might become necessary.
    • Type of service : Diffserv classification for the phone outbound voice traffic: Currently not supported due to a limitation on the phone. The DSCP value will always be 0x2e - Expedited Forwarding.
    • 802.1p priority: IEEE 802.1p voice VLAN priority
    • Voice VLAN id (1 -> 4094) : IEEE 802.1q voice VLAN identifier
    • RTP Port start & RTP Port stop : The port used for RTP communication may be specified. It is recommended to leave this option empty so the default values will be chosen.



It is possible that after a factory reset the phone indicates an erroneous date. A simple reboot after should solve the issue.

Mitel's Doc :
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