# This file contains the configuration for Sentinel VoLTE that is not already in a shared file.

deployment-config:sentinel-volte:

  # Whether session replication is enabled.
  session-replication-enabled: true

  # SCC configuration.
  scc:

    # Whether this deployment is 'gsm' or 'cdma'.
    scc-mobile-core-type: cdma

    # Whether to retrieve the MSISDN from 'MSISDN' or 'EXTENDED_MSISDN'.
    # If set to 'EXTENDED_MSISDN', udr-included-identities MUST be set to 'IMPU_AND_IMPI'.
    fetch-cmsisdn-source: MSISDN

    # Defines which IMS user identities to include in outgoing user data requests.
    # Can be either 'IMPU' or 'IMPU_AND_IMPI'.
    # Must be set to 'IMPU_AND_IMPI' if fetch-cmsisdn-source is set to 'EXTENDED_MSISDN'.
    udr-included-identities: IMPU_AND_IMPI

    # Service continuity configuration.
    service-continuity:

      # Time (in milliseconds) to wait before we consider the ATCF update has failed.
      atcf-update-timeout-milliseconds: 2000

      # STN-SR.
      stn-sr: "6421999999"

    # Service Centralisation configuration.
    service-centralisation:

      # The SCCP address of the Sentinel VoLTE AS.
      # Note that using this also requires setting 'sccp-variant: ANSI' in sgc-config.yaml.
      inbound-ss7-address: type=A7,ri=gt,ssn=146,tt=8,digits=987654,national=true

      # Add the I-CSCF to the route header of the reoriginated invite.
      use-direct-icscf-routing: true

      # A template string for the P-Visited-Network-Information header generated in the reorigination,
      # where {mnc} and {mcc} are replaced with the MNC and MCC respectively.
      # IR 65 versions 12 or earlier define this to be epc.ims.mnc{mnc}.mcc{mcc}.3gppnetwork.org,
      # while later versions define this as ims.mnc{mnc}.mcc{mcc}.3gppnetwork.org.
      generated-pvni-template: "epc.ims.mnc{mnc}.mcc{mcc}.3gppnetwork.org"

      # Police incoming originating requests, and reject attempts to hijack the call. Enabled by default.
      police-originating-requests: true

      # Simple IMRN pool config for mainline case.
      simple-imrn-pool:
        # The minimum correlation ID value used in the cluster. 0 to maximum-correlation-id.
        minimum-correlation-id: "0"

        # The maximum correlation ID value used in the cluster. 0 to (10^18-1).
        maximum-correlation-id: "999"

        # The number of digits the correlation ID should have.
        # Minimum of number of digits in maximum-correlation-id to 18 maximum.
        number-of-digits-in-correlation-id: 10

      # CDMA specific configuration.
      scc-cdma-service-centralisation:
        # CDMA actions to be taken.
        scc-cdma-actions:
          # Action to take when an unexpected trigger is received.
          action-on-unsupported-trigger: accessDeniedReason_terminationDenied

          # Action to take when there is a failure generating a routing number.
          action-on-failed-to-allocate-routing-number: accessDeniedReason_busy

          # Default action to take on error.
          default-failure-action: accessDeniedReason_serviceDenied

        # Config for IMRN formation.
        cdma-imrn-formation:
          # The type of digits field of the IMRN generated.
          imrn-type-of-digits: DIALED_OR_CALLED_PARTY_NUMBER

          # The nature field of the IMRN generated.
          imrn-nature-of-number: INTERNATIONAL

          # The numbering plan field of the IMRN generated.
          imrn-numbering-plan: TELEPHONY

        # If true, reorigination is skipped if the subscriber is not registered in the IMS network.
        bypass-forwarding-if-served-user-not-ims-registered: true

    # TADS configuration.
    tads:
      # (Optional) prefix to append to the CMSISDN, MSRN or TLDN when forming a CSRN.
      # csrn-prefix:

      # By default SCC TADS Routing uses the 'CMSISDN' from the HSS,
      # but it can also be configured to use the 'TLDN' from the HLR.
      # Valid values are 'CMSISDN' and 'TLDN'.
      address-source-for-scc-tads: TLDN

      # If true, the HSS must be queried for voice over PS support as part of
      # the decision to attempt to route via PS.
      voice-over-ps-support:
        # Specifies which identities will be used for the voice over PS support
        # request to the HSS.
        # One of 'IMPU', 'MSISDN', 'IMPU_IMPI' and 'MSISDN_IMPI'.
        request-user-identity-type: IMPU

      # Allow WLAN access.
      wlan-allowed: false

      # The identity of the terminating device that TADS will send the request to.
      # One of 'IMS_PUBLIC_IDENTITY', 'SIP_INSTANCE', and 'PATH_FROM_SIP_INSTANCE'
      tads-identity-for-terminating-device: IMS_PUBLIC_IDENTITY

      # The SIP response code that is returned when a session is ended due to an error.
      end-session-error-code: 480

      # When enabled INVITE requests destined for the CS network will be sent directly via
      # the I-CSCF, bypassing the S-CSCF.
      cs-routing-via-icscf: true

      # Configuration for TADS sequential routing
      on-sequential-routing:
        # Time to wait (in milliseconds) for a potentially better forked response.
        tads-timer-max-wait-milliseconds: 5000

        # List of SIP response codes that will trigger attempts of more routes after a PS attempt.
        ps-fallback-response-codes: []

      # Configuration for TADS parallel routing
      on-parallel-routing:
        # Time to wait (in milliseconds) for a final response.
        parallel-timer-max-wait-milliseconds: 20000

        # When enabled TADS will end all parallel forks on the first busy response (486).
        release-all-legs-on-busy: false

      # When present, requests destined to the CS domain will contain a Diversion header to
      # suppress call diversion in the CS domain side of the call.
#      suppress-cs-domain-call-diversion:
#
#        # When true, use diversion counter parameter, otherwise use number of headers.
#        use-diversion-counter-parameter: true
#
#        # The configured diversion limit in the CS network to suppress further call diversion.
#        cs-domain-diversion-limit: 1

  # Configuration for MMTel services.
  mmtel:

    announcement:

      # Media server URI, used when playing announcements
      # This is distinct from mrf-uri for Conferencing
      announcements-media-server-uri: sip:annc-audio@localhost:5260;lr;transport=tcp

      # Each announcement needs to be configured with the following:
      #   id - The announcement ID that is used to correlate between services and the
      #     announcement that will be played.
      #   description - A human readable string that indicates what the announcement is for.
      #   announcement-url - The URL to the file on the MRF that will be played.
      #   duration-milliseconds - The maximum duration of the announcement (in milliseconds).
      #   repeat - The number of times the announcement will be repeated.
      #   delay-milliseconds - The delay between repeated announcements (in milliseconds).
      #   mimetype - The MimeType of the announcement to be played.
      #   interruptable - Whether or not the announcement can be interrupt by user actions.
      #   end-session-on-failure - Whether or not the current session should be ended if there
      #     is an error while playing the announcement.
      #   enforce-one-way-media - Whether or not the media stream should be forced to be one
      #     way between the MRF and the user during the announcement.
      #   suspend-charging - Whether or not charging should be suspended while the
      #     announcement is being played.
      announcements:
        - id: 20
          description: "MMTel - Outgoing Call Barring"
          announcement-url: "file://mmtel_ocb.3gp"
          duration-milliseconds: 13000
          repeat: 2
          delay-milliseconds: 1000
          mimetype: "audio/3gpp"
          interruptable: false
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 21
          description: "MMTel - Incoming Call Barring"
          announcement-url: "file://mmtel_icb.3gp"
          duration-milliseconds: 13000
          repeat: 2
          delay-milliseconds: 1000
          mimetype: "audio/3gpp"
          interruptable: false
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 22
          description: "MMTel - Anonymous Call Rejection"
          announcement-url: "file://mmtel_acr.3gp"
          duration-milliseconds: 21000
          repeat: 2
          delay-milliseconds: 1000
          mimetype: "audio/3gpp"
          interruptable: false
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 23
          description: "MMTel - Call Diversion"
          announcement-url: "file://mmtel_cdiv.3gp"
          duration-milliseconds: 5000
          repeat: 1
          delay-milliseconds: 0
          mimetype: "audio/3gpp"
          interruptable: false
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 24
          description: "MMTel - Call Waiting"
          announcement-url: "file://mmtel_cw.3gp"
          duration-milliseconds: 600000
          repeat: 500
          delay-milliseconds: 10000
          mimetype: "audio/3gpp"
          interruptable: true
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 25
          description: "MMTel - Call Hold (Hold)"
          announcement-url: "file://mmtel_hold_hold.3gp"
          duration-milliseconds: 600000
          repeat: 500
          delay-milliseconds: 10000
          mimetype: "audio/3gpp"
          interruptable: true
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 100
          description: "OCS - Low Balance"
          announcement-url: "file://charging_low.3gp"
          duration-milliseconds: 4000
          repeat: 1
          delay-milliseconds: 0
          mimetype: "audio/3gpp"
          interruptable: false
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 101
          description: "OCS - Insufficient Balance"
          announcement-url: "file://charging_insuf.3gp"
          duration-milliseconds: 13000
          repeat: 2
          delay-milliseconds: 1000
          mimetype: "audio/3gpp"
          interruptable: false
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

        - id: 102
          description: "OCS - Out Of Credit"
          announcement-url: "file://charging_out.3gp"
          duration-milliseconds: 9000
          repeat: 2
          delay-milliseconds: 1000
          mimetype: "audio/3gpp"
          interruptable: true
          end-session-on-failure: false
          enforce-one-way-media: false
          suspend-charging: false

      # Default error code announcement config.
      # Presence determines whether an announcement should be played to the calling party on receipt
      # of any error code not explicitly listed in the error-code-announcements configuration.
#      default-error-code-announcement:
#        # ID of the announcement to be played on an error response.
#        announcement-id: 22
#        # Whether the call should be ended with a 487 response rather than the error code that
#        # triggered this announcement.
#        end-call-with-487-response: false

      # Config for whether or not an announcement should be played to calling party on particular error codes
#      error-code-announcements:
#        # Each announcement is configured with the following:
#        #   error-code - The SIP error response code that should trigger this announcement.
#        #   announcement-id - The announcement ID that is to be used for this error code.
#        #     Cannot be specified when disable-announcement is set to true.
#        #   disable-announcement - Optional. If set to true, no announcement
#        #     will be played for the specified error code, overriding any default error
#        #     code announcement that may have been specified. If set to true, then
#        #     announcement-id must not be specified.
#        #   end-call-with-487-response - Once the announcement is finished should the call be
#        #     ended with a 487 response or the original error that triggered this announcement.
#        - error-code: 486
#          announcement-id: 24
#          end-call-with-487-response: false
#        - error-code: 488
#          announcement-id: 22
#          end-call-with-487-response: true
#        - error-code: 489
#          disable-announcement: true
#          end-call-with-487-response: true

    # Support for enabling or disabling specific HSS queries.
    hss-queries-enabled:

      # Whether to query for Operator Determined Barring Information (IMS-ODB-Information).
      odb: false

      # Whether to query for Metaswitch TAS Services (Metaswitch-TAS-Services).
      metaswitch-tas-services: false

    # Configuration for conferencing.
    conferencing:

      # The URI of the Media Resource Function used for Conferencing.
      # This is distinct from the media-server-uri used for Announcements
      # The hostname part should either be a resolvable name or the IP address of the MRF.
      conference-mrf-uri: sip:mrf@mrfhost.example:5060

      # Should messages to the MRF be routed via the IMS, or are they allowed to be routed
      # direcectly from the TAS to MRF.
      route-to-mrf-via-ims: false

      # The MMTel Conference MSML Schema Vendor Name.
      # Used by the Conf feature to determine mapper selection
      # when creating MSML documents for interaction with the MRF.
      # 'Dialogic' or 'Radisys'.
      msml-vendor: Radisys

      # Whether to use the Re-INVITE based three-party conference flow.
      enable-scc-conf-handling: true

      # Decides whether the 'root' element will be a child of the 'selector' element,
      # otherwise will be a child of 'videolayout'
      root-on-selector: true

      # A list of conference factory PSIs to use in addition to the standard conference
      # factory PSIs, as per TS 23.003. They are as follows:
      # "sip:mmtel@conf-factory.<HOME-DOMAIN>"
      # "sip:mmtel@conf-factory.ims.mnc<MNC>.mcc<MCC>.3gppnetwork.org"
      # "sip:mmtel@conf-factory.ics.mnc<MNC>.mcc<MCC>.3gppnetwork.org"
      conference-factory-psi-aliases: []

      # Maximum number of participants that are allowed in each conference call.
      maximum-participants: 3

      # Should video be allowed during conference calls
      allow-video-conference-calls: false

      # Delay (in milliseconds) after a conference ends before ConferenceView profiles are
      # removed.
      conference-view-removal-delay-milliseconds: 0

      # Conferencing event subscription configuration.
      subscription:
        # Value to be used if the SUBSCRIBE message doesn't contain an Expires header.
        default-subscription-expiry-seconds: 3600

        # SUBSCRIBE requests with an Expires value lower than this are rejected.
        min-subscription-expiry-seconds: 5

        # Frequency of polls for changes to conference view.
        polling-interval-seconds: 5

    # Configuration for determining international and roaming status
    international-and-roaming:

      # Treat non-international format numbers as national.
      non-international-format-number-is-national: false

      # End call if no visited network.
      end-call-if-no-visited-network: false

      # Minimum length of destination address to set international
      # and roaming status for. Destination addresses less than this
      # length will not have international or roaming status set.
      min-length: 0

    # Configuration for the communication hold feature.
    communication-hold:
      # Parameters for Hold response processing. If one is specified, then all should be.
      # Their presence indicates bandwidth should be adjusted when a session is Held and Resumed.
      # Default values taken from 3GPP TS 24.610 Rel 12.6.0 section 4.5.2.4.
      bandwidth-adjustment:
        # The value of the "b=AS:" parameter to use when processing a Hold response.
        b-as-parameter: 0

        # The value of the "b=RR:" parameter to use when processing a Hold response.
        b-rr-parameter: 800

        # The value of the "b=RS:" parameter to use when processing a Hold response.
        b-rs-parameter: 800

      # Should an announcement be played when a session is held.
      announcement:
        # The announcement to be played when a session is held.
        announcement-id: 25

      # Determines how media streams for the holding party are handled while an announcement
      # to the held party is in progress. Can be set to NO_HOLD, BLACK_HOLE_ONLY, or FULL_HOLD.
      holding-party-media-mode: FULL_HOLD


    # Configuration for the communication waiting feature.
    communication-waiting:
      # Should an announcement be played to a calling user when communication waiting is
      # applied.
      announcement:
        # The announcement to be played when communication waiting is applied.
        announcement-id: 24

      # Time (in seconds) for the communication waiting timer.
      timer-seconds: 0

    # Configuration for privacy features.
    privacy:
      # Configuration for the Originating Identification Presentation (OIP) feature.
      originating-identification-presentation:
        # When set to true, the from header of the originating INVITE will be
        # anonymized if the OIP feature is not active for the subscriber.
        #
        # If set to 'true', this typically means that:
        # - OIP is authorized for the large majority of subscribers, and that;
        # - The OIP active flag is either not present in the subscriber's data
        #   (therefore defaulting to true), or the OIP flag is present in the
        #   subscriber's data (typically set to true).
        # - This also implies that the OIR user-policy is set to 'None'.
        #
        # If set to 'false', this typically means that:
        # - OIP is not 'active' or 'authorized' yet the operator desires the
        #   called party to see the calling party.
        # - This implies that the OIR user-policy would be set to
        #   ANONYMIZE_FROM or ADD_USER_PRIVACY.
        anonymize-from-header: true

        # If true, allows History-Info header deletion.
        allow-history-info-header-deletion: false

      # Configuration for the Originating Identification Restriction (OIR) feature.
      originating-identification-restriction:
        # Can be one of two values: ONLY_IDENTITY, and ALL_PRIVATE_INFORMATION.
        # Use of ONLY_IDENTITY means the Privacy header is set to Privacy:id.
        # Use of ALL_PRIVATE_INFORMATION means the Privacy header is set to Privacy:header.
        presentation-restriction-type: ALL_PRIVATE_INFORMATION

        # The user policy for OIR. Must be one of NONE, ANONYMIZE_FROM, and ADD_USER_PRIVACY.
        user-policy: NONE

    # Configuration for PSAP callback functionality.
    psap-callback:
      # Use the contents of the Priority header in the initial INVITE
      # to determine whether the session is a PSAP callback.
      use-priority-header: false

      # Configuration for the SIP MESSAGE mechanism for determining PSAP callbacks.
      # Presence determines whether the mechanism is used or not.
      sip-message-options:
        # For use when use-sip-message is set to true.
        # When a SIP MESSAGE notifying that a PSAP call has taken place, this is the time
        # (in seconds) after receiving that MESSAGE that sessions for the identified user are
        # assumed to be a PSAP callback.
        expiry-time-seconds: 86400

        # For use when use-sip-message is set to true.
        # If set to true, SIP MESSAGEs notifying a PSAP call will be terminated at the MMTel,
        # otherwise they are propagated through the network.
        terminate-message: true

    # Configuration for call diversion (CDIV)
    call-diversion:

      # Play announcement on diversion.
      announcement:
        announcement-id: 23
#        voicemail-announcement-id: 23

      # Standard MMTel call diversion configuration
      mmtel-call-diversion:

        # Maximum number of diversions that may be made while
        # attempting to establish a session.
        max-diversions: 20

        # Action to take when the maximum number of diversions is exceeded.
        max-diversion-action: REJECT

        # Address to divert to when max-diversions limit is reached and
        # max-diversion-action is set to DELIVER_TO_FIXED_DESTINATION.
#        max-diversion-fixed-destination: sip:no-reply@example.com

        # Time to wait (in seconds) for a reply before diverting due to a no reply rule.
        # This value is the network default, and can be overridden in subscriber data.
        no-reply-timeout-seconds: 20

        # Whether to add orig tag when diverting a call.
        add-orig-tag: true

        # URIs allowed to be re-targeted to in case of max-diversions limit being reached.
#        diversion-limit-exempt-uris:
#          - sip:user@example.com

        # Whether diversion should be suppressed if call terminates in CS domain.
        suppress-for-cs-terminating-domain: false

        # Whether subscriber configuration should take precedence over operator
        # configuration.
        prefer-subscriber: false

        # Address to forward to if operator or subscriber forward-to rule has no target
        # specified.
#        default-target-uri: sip:user@example.com

        # Additional response codes that will trigger CDIV Not-Reachable (in addition to
        # those outlined in the MMTel CDIV specification).
#        additional-not-reachable-status-codes:
#          - 488

        # Whether to allow CDIV rules with not-reachable conditions to be triggered after
        # a 180 response has been received from the called-party.
        allow-not-reachable-during-alerting: false

        # Whether to add 'hi-target-param' of type 'mp' to the 'hi-entry' added by a
        # diversion.
        add-mp-param: false

      # Configuration for forwarding to a voicemail server
      # If present, enables forwarding to subscriber's voicemail server if all other
      # connection attempts fail.
#      forward-to-voicemail:
#
#        # URIs of voicemail servers for which a voicemail-specific announcement may be
#        # played (if specified) and for which forwarding to without allocated credit
#        # can be allowed (if enabled).
#        voicemail-uris:
#          - sip:vms1@example.com
#          - sip:vms2@example.com
#
#        # Time to wait (in seconds) for a call to be successfully connected before
#        # forwarding to voicemail (if enabled) or 0 to disable timer.
#        forward-to-voicemail-timeout-seconds: 0
#
#        # When to allow forwarding to voicemail when out of credit.
#        # Only applies when using 'ro' for online charging
#        forward-to-voicemail-without-ocs-credit: NEVER_ALLOW

    # Configuration for communication barring
    communication-barring:

      # Configuration for incoming communication barring
      incoming-communication-barring:

        # Should an announcement be played when an incoming call is barred.
        announcement:
          # The announcement to be played when an incoming call is barred.
          announcement-id: 21

          # (Optional) A different announcement can be played if the call is barred
          # because it is from an anonymous user.
          anonymous-call-rejection-announcement-id: 22

        # If false, incoming call barring will ignore International and International-exHC
        # rules. This is because it is not possible to accurately determine whether the
        # calling party is international in all circumstances.
        international-rules-active: false

#     # Configuration for outgoing communication barring
#     outgoing-communication-barring:
#
#       # Should an announcement be played when an outgoing call is barred.
#       announcement:
#
#         # The announcement to be played when an outgoing call is barred.
#         announcement-id: 20

      # Configuration for operator communication barring
#      operator-communication-barring:
        # The set of operator barring rules that can be applied.
        # (Optional) Only required if '../hss-queries-enabled/odb' is 'true'.
#        operator-barring-rules:
#
#          # Operator barring rule for 'Type1'
#          type1:
#
#            # The barring rule to be applied
#            rule: # Bar domain "example.com"
#              <cp:ruleset xmlns="http://uri.etsi.org/ngn/params/xml/simservs/xcap"
#                    xmlns:cp="urn:ietf:params:xml:ns:common-policy">
#                <cp:rule id="blacklist-domain">
#                  <cp:conditions>
#                    <cp:identity>
#                      <many domain="example.com" />
#                    </cp:identity>
#                  </cp:conditions>
#                  <cp:actions>
#                    <allow>false</allow>
#                  </cp:actions>
#                </cp:rule>
#              </cp:ruleset>
#
#            # Should the call be redirected when barred by this rule.
#            retarget:
#
#              # The URI to redirect the barred call to.
#              retarget-uri: sip:retarget@retargethost.example:5060
#
#              # Should an announcement be played when the call is barred by this rule.
#              announcement:
#
#                # The announcement to be played if the call is barred by this rule.
#                announcement-id: 20
#
#              # If we retarget the call, should online charging be disabled.
#              disable-online-charging-on-retarget: false
#
#          # Operator barring rule for 'Type2'
#          type2:
#
#            # The barring rule to be applied
#            rule: # Bar international calls
#              <cp:ruleset xmlns="http://uri.etsi.org/ngn/params/xml/simservs/xcap"
#                    xmlns:cp="urn:ietf:params:xml:ns:common-policy">
#                <cp:rule id="bar-international">
#                  <cp:conditions>
#                    <international/>
#                  </cp:conditions>
#                  <cp:actions>
#                    <allow>false</allow>
#                  </cp:actions>
#                </cp:rule>
#              </cp:ruleset>
#
#            # Should the call be redirected when barred by this rule.
#            retarget:
#
#              # The URI to redirect the barred call to.
#              retarget-uri: sip:retarget@retargethost.example:5060
#
#              # Should an announcement be played when the call is barred by this rule.
#              announcement:
#
#                # The announcement to be played if the call is barred by this rule.
#                announcement-id: 20
#
#              # If we retarget the call, should online charging be disabled.
#              disable-online-charging-on-retarget: false
#
#          # Operator barring rule for 'Type3'
#          type3:
#
#            # The barring rule to be applied
#            rule: # Bar roaming calls
#              <cp:ruleset xmlns="http://uri.etsi.org/ngn/params/xml/simservs/xcap"
#                    xmlns:cp="urn:ietf:params:xml:ns:common-policy">
#                <cp:rule id="bar-roaming">
#                  <cp:conditions>
#                    <roaming/>
#                  </cp:conditions>
#                  <cp:actions>
#                    <allow>false</allow>
#                  </cp:actions>
#                </cp:rule>
#              </cp:ruleset>
#
#            # Should the call be redirected when barred by this rule.
#            retarget:
#
#              # The URI to redirect the barred call to.
#              retarget-uri: sip:retarget@retargethost.example:5060
#
#              # Should an announcement be played when the call is barred by this rule.
#              announcement:
#
#                # The announcement to be played if the call is barred by this rule.
#                announcement-id: 20
#
#              # If we retarget the call, should online charging be disabled.
#              disable-online-charging-on-retarget: false
#
#          # Operator barring rule for 'Type4'
#          type4:
#
#            # The barring rule to be applied
#            rule: # Allow audio calls
#              <cp:ruleset xmlns="http://uri.etsi.org/ngn/params/xml/simservs/xcap"
#                    xmlns:cp="urn:ietf:params:xml:ns:common-policy">
#                <cp:rule id="allow-audio">
#                  <cp:conditions>
#                    <media>audio</media>
#                  </cp:conditions>
#                  <cp:actions>
#                    <allow>true</allow>
#                  </cp:actions>
#                </cp:rule>
#              </cp:ruleset>
#
#            # Should the call be redirected when barred by this rule.
#            retarget:
#
#              # The URI to redirect the barred call to.
#              retarget-uri: sip:retarget@retargethost.example:5060
#
#              # Should an announcement be played when the call is barred by this rule.
#              announcement:
#
#                # The announcement to be played if the call is barred by this rule.
#                announcement-id: 20
#
#              # If we retarget the call, should online charging be disabled.
#              disable-online-charging-on-retarget: false

        # Outgoing prefix barring configuration
#        outgoing-prefix-barring:
#
#          # The list of prefixes to match against when doing prefix barring
#          prefixes:
#
#            # The prefix barring rule for prefix '87'
#            - prefix: '87'
#              # List of the classifications to apply when the above prefix is matched.
#              # Must refer to the name of a defined classification.
#              classifications:
#                - 'Operator Bar'
#
#          # The list of classifications that can be applied to a prefix match
#          classifications:
#
#            # A barring configuration called 'Operator Bar'.
#            - name: 'Operator Bar'
#
#              # The minimum length of dialled digits to match against
#              minimum-number-length: 5
#
#              # The maximum length of dialled digits to match against
#              maximum-number-length: 8
#
#              # When true, the normalized number must be international and not
#              # within the Home Country Code to match this classification.
#              match-international: false
#
#              # The barring treatment to apply if this condition matches.
#              # Valid values are: 'OSBType1', 'OSBType2', 'OSBType3', 'OSBType4',
#              # 'OperatorAllow', 'OperatorBar', 'PremiumRateInformation',
#              # and 'PremiumRateEntertainment'.
#              barring-treatment: 'OperatorBar'
#
#              # Disables the outgoing-call-barring (OCB) announcement.
#              # Cannot be specified alongside a prefix barring announcement.
#              #disable-ocb-announcement: true
#
#              # Should a different announcement be used rather than the one
#              # from 'outgoing-communication-barring/announcement'
#              announcement:
#
#                # The announcement ID to use instead of the one from
#                # `outgoing-communication-barring/announcement/announcement-id`
#                announcement-id: 20

    # Configuration for Vertical Service Codes
    vertical-service-codes:
      # Configuration for Vertical Service Codes XCAP Data Update feature
      xcap-data-update:
        # Internally-accessible hostname of the XCAP server.
        host: xcap-internal.example

        # Internally-accessible port of the XCAP server.
        port: 8080

        # Whether to use TLS for the HTTP connection.
        use-https: false

        # Base URI of XCAP server when accessed via specified hostname:port.
        base-uri: /rem/sentinel/xcap

        # Application Unique Identifier (AUID) to use in the request URI.
        auid: simservs.ngn.etsi.org

        # XCAP document to use in request URI.
        document: simservs.xml

        # SIP status code to respond with following a successful HTTP response.
        success-response-status-code: 606

        # SIP status code to respond with following an unsuccessful HTTP response.
        failure-response-status-code: 409

        # Whether an announcement should be played on failure.
#        failure-announcement:
#          # ID of the announcement to play on failure.
#          announcement-id: 23

  # Configuration for the SIS.
  sis:

    # RFC3263 unavailable peer list timer (milliseconds).
    unavailable-peer-list-timer-milliseconds: "60000"

    # RFC3263 failover timer (milliseconds).
    failover-timer-milliseconds: "4000"

  # Origin configuration for this application when connecting to the HLR.
  # The actual HLR SCCP address (destination) is in the hlr-configuration.yaml file
  # Uncomment this section if using the TLDN for SCC TADS Routing
  hlr-connectivity-origin:

    # The SCCP address of the Sentinel VoLTE AS.
    originating-address: type=A7,ri=pcssn,pc=0-0-5,ssn=147,national=true

    # The timeout value for opening the MAP dialog with the HLR (in milliseconds).
    map-invoke-timeout-milliseconds: 5000

    # CDMA specific configuration.
    cdma:
      # The market ID value to be used in the MSCID set on outgoing CDMA requests to the HLR
      market-id: 1

      # The switch number value to be used in the MSCID set on outgoing CDMA requests to the HLR
      switch-number: 1

  # Charging configuration
  charging:

    # Whether to enable online charging
    # If disabled, then you must also remove or comment out the ro-charging section below.
    cdma-online-charging-enabled: true

    # When not using Diameter Ro, comment out the contents of this file down to the <<<< END ro-charging
    # Also you must remove the lines from mmt-gsm-vmpool-config.yaml
    ro-charging:

      diameter-ro:

        # The Diameter Ro release to use.
        diameter-ro-release: Vcb0

        # The origin realm to use when sending messages.
        origin-realm: metaswitch.com

        # The value to use as the destination realm.
        destination-realm: metaswitch.com

        # The Diameter Ro destination peers.
        destination-peers:
          - destination-hostname: peer.metaswitch.com
            port: 3868
            protocol-transport: aaa
            metric: 1

        # Whether the session is permitted to continue if there is an OCS failure.
        continue-session-on-ocs-failure: false

      charging-announcements:

        # Config for low balance announcements
        low-credit-announcements:

          # Low balance announcement ID to be used during call setup.
          call-setup-announcement-id: 100

          # Low balance announcement ID to be used after call setup.
          mid-call-announcement-id: 100

          # The delay (in milliseconds) before a another credit check should happen after a
          # low balance announcement has occurred.
          charging-reauth-delay-milliseconds: 30000

        # Config for out of credit announcements.
        out-of-credit-announcements:

          # Out of credit announcement ID to be used during call setup.
          call-setup-announcement-id: 101

          # Out of credit announcement ID to be used after call setup.
          mid-call-announcement-id: 102
# <<<< END ro-charging

    # Whether to enable connections to Diameter Rf peer through the Diameter Rf Control RA.
    # If present, Rf is enabled. Comment out to disable Rf.
    # When not using Diameter Rf, comment out the contents of this file down to the <<<< END rf-charging
    # Also you must remove the lines from mmt-gsm-vmpool-config.yaml
#    rf-charging:
#
#      diameter-rf:
#
#        # The Diameter Rf release to use.
#        diameter-rf-release: Vcb0
#
#        # The origin realm to use when sending messages.
#        origin-realm: metaswitch.com
#
#        # The value to use as the destination realm.
#        destination-realm: metaswitch.com
#
#        # The Diameter Rf destination peers.
#        destination-peers:
#          -   destination-hostname: peer.metaswitch.com
#            port: 3868
#            protocol-transport: sctp
#            metric: 1
# <<<< END rf-charging

    # Configuration for CDRs
    cdr:
      # If present, interim CDRs are enabled. If Diameter Rf has been enabled, this is required.
      interim-cdrs:
        # Enable CDRs to go to the local filesystem
        # Diameter Rf is selected separately
        write-cdrs-in-filesystem: true

        # Indicates whether or not to write CDRs on SDP changes.
        write-cdr-on-sdp-change: true

        # The maximum duration (in seconds) between timer driven interim CDRs.
        # Setting this to zero will disable timer based interim CDRs.
        interim-cdrs-period-seconds: 300

      # Enable session CDRs.
      session-cdrs-enabled: true

  # Configuration for the Session Refresh feature.
  session-refresh:
    # The interval of the periodic timer (in seconds).
    timer-interval-seconds: 30

    # Period of no activity for leg tp refresh (in seconds).
    refresh-period-seconds: 570

    # Whether the session should be refreshed using UPDATE requests,
    # as long as the endpoint allows UPDATE requests.
    refresh-with-update-if-allowed: true

    # Maximum allowed duration of a call (in seconds).
    max-call-duration-seconds: 86400

  # DO NOT ENABLE IN PRODUCTION
  # Enable extensive logging for verification and issue diagnosis during acceptance testing
  debug-logging-enabled: false
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Rhino VoLTE TAS VMs Version 4.0.0