FMU phone (SDF1)

Description

This resource allows to connect an FMU phone.

Release notes

Version 1.0.1 - General deployment
  • Bugfix: Caller ID of GSM instead of extension for external call (M7541)

Version 1.0.0 - General deployment
  • Feature: Initial release (M0)

Resource configuration interface

GUI unavailable.

Resource configuration parameters

This resource allows and administrator to configure a generic sip phone.

  • Password: Password used in the SIP authentication when 'use authentication' is set to 'yes'. (Default: '0000')
  • 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 transferred 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 SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
      • Default: Keep the default setting defined in the Telephony server module (for example Asterisk 1.2.x)
      • No: Disable Qualify SIP registration for this device
      • Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
      • Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
    • 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 codec.
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