net.Buzz (SDX4)

Description

Release notes

Version 1.2.0 - General deployment
  • Feature: Presence and IM in net.Buzz (M8035)
  • Dependency:
    • net.Buzz module v1.4.0+

Version 1.1.0 - General deployment
  • Improvement: Added g722 audio codec support (M6601)
  • Improvement: Added video support (M6601)
  • Dependency:
    • net.Buzz module v1.3.0+

Version 1.0.2 - Deprecated
  • Deprecated: Version no longer supported
  • Dependency:
    • net.Buzz module

Version 1.0.1 - General deployment
  • Feature: Added custom SIP servers, username and password
  • Feature: Added SIP server features: qualify, reinvite, codec choice (M6301)
  • Dependency:
    • net.Buzz module

Version 1.0.0 - Deprecated
  • Deprecated: Version no longer supported
  • Dependency:
    • net.Buzz module

Resource configuration interface

GUI unavailable.

This resource allows an administrator to configure a net.Buzz phone usable by the net.Buzz module.

Resource configuration parameters

Useful parameters are described below, please leave the other ones untouched.
  • Description: Description of this net.Buzz phone (E.g. John Doe net.Buzz)
  • SOP1 and SOP2: on which SOP(s) do you want net.Buzz to register to.
  • Restriction Group: the restriction group in which net.Buzz will be set.
  • License: the net.Buzz phone's license. If an individual license is not provided, it will try to use the group license configured in the net.Buzz module.
  • Custom SIP server: by default, net.Buzz will try to register and use the SOP as a SIP server. If you want to use another server, fill in the fields.
    • Use DNS SRV Record: Set to yes if you do have configured a DNS SRV Record. Put in 'SIP server 1' the DNS name.
  • SIP server feature:
    • 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 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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