GigasetN720IP (SDG2)

Description

This resource allows to connect compatible handsets to the Gigaset N720 IP Pro base station.

Release notes

Show release notes
Version 1.2.0 - Early deployment
  • Feature: Support for pickup on a BLF key (M19406)
  • Dependency:
    • Gigaset Phone Support >= 1.2.1
    • DHCP Server >= 2.5.0 (if used)

Version 1.1.2 - Early deployment
  • Bugfix: Removed parameter "order" (M23057)
  • Dependency:
    • Gigaset Phone Support >= 1.2.1
    • DHCP Server >= 2.5.0 (if used)

Version 1.1.1 - Early deployment
  • Bugfix: Not possible to use server default codecs (M21729)
  • Dependency:
    • Gigaset Phone Support >= 1.1.0
    • DHCP Server >= 2.5.0 (if used)

Version 1.1.0 - Early deployment
  • Improvement: Mailbox activated (M20374)
  • Dependency:
    • Gigaset Phone Support >= 1.1.0
    • DHCP Server >= 2.5.0 (if used)

Version 1.0.1 - Deprecated
  • Improvement: Ability to change the CallerID for basestation compatibility (M19158)
  • Deprecated: CallerID not supported by SIP Trunks for external calls (M0)
  • Dependency:
    • Gigaset Phone Support >= 1.1.1
    • DHCP Server >= 2.5.0 (if used)

Version 1.0.0 - Early deployment
  • Feature: Initial Release (M18188)
  • Dependency:
    • Gigaset Phone Support >= 1.1.0
    • DHCP Server >= 2.5.0 (if used)

Resource configuration interface

GUI unavailable.

Resource configuration parameters

This resource allows and administrator to configure a Polycom IP Phone.

  • Password: password used during the SIP authentication towards the provisioning server.
  • DTMF mode:
    • inband: DTMF signalling through the same channel as voice.
    • rfc2833: DTMF signalling using RTP packets, distinct from the audio packets.
  • Use Remote Party ID header:
    • Default: Keep default settings as defined in the "Communication Server" module configuration.
    • Yes: Allow the server to perform the re-invite procedure.
    • No: Do not allow the server to perform the re-invite procedure.
  • 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 "Communication Server" module configuration.
      • 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 "Communication Server" module configuration.
      • 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 accepted codecs. ALERT! You should set at least one of those 3 codecs in order for the handsets to work properly!
    • Handset ID: Here you have to specify the IPUI (International Portable User Identity) of the handset you're trying to connect. See the AdminGuide on how to retrieve that IPUI number. (This is not a MAC address and thus differs from the base station address that you have to specify in the MAC Address field)
  • Advanced options:
    • Voicemail service extension: Define the extension used for the voicemail.

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