Unified End Point SIP peer (SDX6)


This resource allows to connect ESCAUX Unified End Point in SIP.

Release notes

Version 1.3.0 - Early deployment
  • Improvement: Better support provisioning in clusters
  • Dependency:
    • SMP Application >= 5.4.1

Version 1.2.2 - Early deployment
  • Improvement: Do not generate configuration when no extension is assigned to SDX6 (M12360)
  • Dependency:
    • SMP Application >= 5.4.1

Version 1.2.1 - Early deployment
  • Bugfix: Mobile number was ignored in the provisioning
  • Dependency:
    • SMP Application >= 5.4.1

Version 1.2.0 - Deprecated
  • Feature: Add Home and Mobile number to the provisioning file (M11138)
  • Improvement: When Unified End Point Support module is installed, deleted resources are removed from the provisioning directory (M11500)
  • Deprecated: Does not take in account the mobile number

Version 1.1.0 - Early deployment
  • Feature: Add Member and Sync From fields in the provisioning file. (M11737)

Version 1.0.0 - Early deployment
  • Feature: Initial release. Create Resource for UEP to allow Escaux Fusion to discriminate based on the resource name. (M10760)

Resource configuration interface

GUI unavailable.

This resource is used to connect your Escaux PBX to an unified end point.

Resource configuration parameters

  • Use authentication:
    • Yes - Custom password: Use authentication. The password will be the one indicated in the field 'Password'. If the 'Password' field isn't set, the default password '0000' will be used.
    • No: Do not use authentication. No password will be required/allowed.
    • Yes - Pincode: Use authentication. The password will be the pincode of the extension linked to this resource. Note that you should only have one extension which uses this resource.
  • DTMF mode:
    • RFC2833: Use the DTMF transmission described in RFC2833. (DTMF are sent through RTP packets)
    • Inband: Try to detect DTMF in the voice traffic.
  • 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
  • Use Remote Party ID header: Set to 'yes' to handle the Remote-Party-ID SIP header. Set to no to ignore it.
  • Pidfnote: The pidfnote is the text display next to the presence of a contact. Set to 'Default' to keep default behaviour. Set to 'Add Remote Party Presence' to add put the information concerning the caller and called channel
  • Mailbox: Set to 'Enabled' to link the SIP peer with the mailbox of the associated extension. This will be used when sending MWI NOTIFY
  • MWI Notify method: Set to 'Automatic' to send MWI-NOTIFY upon the reception of the REGISTER. Set to 'Upon Subscribe' send the NOTIFY only when the proper SUBSCRIBE is received
  • Outside Line Access Code: The Outside Line Access Code is the prefix to be used when making an external call. This code is accessible in the callflow via the channel variable PEER_INFO_OutgoingPrefix. ALERT! If you want to use this variable on your SOP, add a globale parameter PEER_INFO_OutgoingPrefix with the value NULL.
    • Force specific code: Set to yes if you want to use a different Outside Line Access Code for this device otherwise the default server settings are to be used. In the callflow the channel varibale PEER_INFO_OutgoingPrefix will be set to NULL.
    • Code: Value of the Outside Line Access Code. This value is only taken into account if Force Line Access Code is set to yes. The channel variable will then contain this value.
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