Genesys Engage on-premises

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  • 1.  Callback "assured-connection"

    Posted 08-17-2022 10:45
    Hello guys,
    I'm implementing a callback project with OCS, applying the Progressive dialing method with the "assured-connection" feature where the dialing flow occurs first for the operator and then to the customer, due to the customer's business needs.


    Problem Occurred: I'm having a problem playing the audio for the client, where an audio of 10 seconds the client hears only the end of the audio.

    The configuration adopted for the execution of the audio is through the routing strategy with Transaction List, due to the attempt to attach by audio, in this case we have the following configurations:

    'agent-greeting' 'music/MSG/Rechamada_ATH.wav' (03 Segundos)
    'customer-greeting' 'music/MSG/Rechamada_Paciente_Axial.wav' (10 segundos)

    Options SIPServer

    greeting-call-type-filter=-internal -outbound -consult
    greeting-notification=-started -complete

    11:10:24.774 Trc 04541 RequestRouteCall received from [1860] (000178af ORS_B
    message RequestRouteCall
    AttributeThisDN '390000'
    AttributeConnID 00720331d982d77b
    AttributeOtherDN '681116'
    AttributeLocation 'SIPServer'
    AttributeExtensions [108] 00 02 00 00..
    'agent-greeting' 'music/MSG/Rechamada_ATH.wav'
    'customer-greeting' 'music/MSG/Rechamada_Paciente_Axial.wav'
    AttributeDNIS '390000'
    AttributeRouteType 1 (RouteTypeDefault)
    AttributeReferenceID 902

    Can anyone help me how do I make the client's audio play only when answering the call? Since it is running already in the execution of the strategy
    routing to call operator extension in this "assured-connection" model.

    The Progressive ASM routing model does not serve this project due to the attach deal in the routing strategy, for this reason I am adopting the "assured-connection" model

    Anderson Oliveira

  • 2.  RE: Callback "assured-connection"

    Posted 09-12-2022 16:50
    Hi Anderson,

    I see that you found a resolution for this via a Product Support case. I am posting the response here in case others have the same questions.

    After careful discussion, we don't think that the current option will work. The greetings can NOT be used for this call flow as SIP Server has no way to know that initial call to the agent will later be merged. If call is merged before the greetings are done then perhaps the called party will hear the end of the in progress greeting or perhaps the greeting that the agent was hearing will be cut off and they hear the called party. If they are merged afterward greeting are complete then called party should not hear any greeting or just the end of it and be connected to the agent direct. There really isn't anyway for SIP Server to start the greetings at the moment of the merge since it is programed to start them as soon as call is answered by the agent when specified in TRouteCall and again SIP Server cannot predict that this call will later be merged.



    Christa Welton
    Genesys - Employees

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