• Handle crossover of incoming PRACK and outgoing 200 (PRX-219)

  • Handle crossover of incoming 200/BYE and outgoing BYE within Rhino more cleanly (PRX-101)


  • Updated installer to support SIS-2.5.3 in addition to 2.5.2 (PRX-166)

  • Fixed some cases where an incoming BYE request during user interaction may not have received a response.


* Migrated to CGIN 1.5.2.


  • Behaviour on call abandon has been modified. If the IN call is abandoned, as well as a CANCEL being sent on the outgoing SIP dialog, a 487 error response will also be sent on any incoming SIP dialogs for call legs. This ensures that all SIP dialogs are terminated properly.

  • An EventReportBCSM event for the tAbandon EDP was accidentally being ignored. This is now fixed.


* Fixed behaviour on IN dialog abort.  Incoming SIP dialogs will no longer
  simply be invalidated.  Incoming INVITEs that have not yet had a final
  response will be responded to with a 500 Internal Server Error.  For
  confirmed dialogs, a BYE will be sent.


  • Fixed a state machine issue dealing with ACK and BYE when call monitoring completes when the call is answered.

  • Fixed an issue dealing with IN dialog abort while waiting for an ACK after answer.


* Incoming INVITEs with a From header containing the anonymous URI will now be
  handled correctly.


  • Added a new option to the SIP-AS configuration for a service key that determines if the R-IM-SSF will do any number normalisation when mapping protocol messages between the IN and SIP.

  • Added new profile table, management MBean, and configuration commands to configure options for network initiated call. Added configuration option that determines in the R-IM-SSF will do any number normalisation when mapping protocol messages from SIP to IN for network initiated calls.

  • Fixed an issue setting the ResetTimer timeout configuration parameter to 0.

  • Fixed potential security exceptions when accessing persistent state.


* The R-IM-SSF will no longer accept and try to process IN dialogs for which
  it does not recognise the application context.


  • Updated to cgin-connectivity 1.5.0-protocol-patches


* The main R-IM-SSF configuration profile has a new option that specifies if
  the R-IM-SSF will always send a Connect operation to route a call (the
  previous behaviour), or will send a Connect or Continue depending on whether
  or not the SIP-AS changes the A or B party numbers when forwarding the

* The prefix in a prefix announcement mapping is now stored in its original
  form as a string rather than being converted to a number.  This means that
  any leading zeros are preserved, and also that the digits '#' and '*' can
  be used if necessary.

* User interaction is now supported using an MRF if the MSC responds to an
  Establish Temporary Connection operation request with an MGCP INVITE.

* Fixed issues with some R-IM-SSF management commands not working properly.

* Fixed a NullPointerException when the R-IM-SSF is invoked if the Management
  Resource Adaptor is not active.

* Fixed NullPointerException when attempting to raise an alarm after a CDR
  write failure.

R-IM-SSF SIS Module (2013-02-02 23:20:26 +1300)

  • Updated to cgin-connectivity 1.5.*

  • The value used for the applicationTimer dp-specific criteria when arming the o/tNoAnswer EDP is now a configurable option.

  • The o/tAbandon EDP armed by the RequestReportBCSMEvent operation when using CAPv2 will now correctly use the Notify & Continue arming mode, as Interrupted mode is not supported for this EDP in CAPv2.

  • Fixed an issue where the CalledPartyBCDNumber in CAP originating BCSM triggers was not being recognised.

  • Revised usage parameters collected by the R-IM-SSF. In particular, the "EndedByASCallsCounter" has been removed as it’s no longer easily possible in some cases to determine a specific reason for call termination.

R-IM-SSF SIS Module (2011-09-26 10:22:41 +1300)

* Updated to cgin-connectivity 1.4.*

R-IM-SSF SIS Module (2011-04-20 13:05:10 +1200)

  • Updated to cgin-connectivity 1.3.*

  • Updated to use configured CountryCode to create SIP URIs from Calling and Called Party Numbers.

  • Updated to use original Calling and Called Party Numbers settings where possible to create the forwarding and forwarded party numbers from incoming SIP URIs.

R-IM-SSF SIS Module 1.1.0 (2010-08-30 17:44:44 +1200)

* Initial release.
Previous page