PureEngage On-Premises

We are trying to record work from home agent calls using Media duplication and forwarding of RTP packets to the recorder.

When we get the required call event that we wish to record actively, we send a private service request as follows -

                    KeyValueCollection reasons = new KeyValueCollection();
                    KeyValueCollection extensions = new KeyValueCollection();
                    extensions.Add("record", "source");
                    extensions.Add("id", ev.CallUuid);
                    extensions.Add("dest", "sip:" + sipSocketListenerIP + ":" + sipSocketListenerPort);
                    RequestPrivateService requestPrivateService = RequestPrivateService.Create(3013, ev.thisDN, ev.connID, ev.userData, reasons, extensions);
                    tServerProtocol.Send(requestPrivateService);


We have a listener open on udp.5060, which is the sipSocketListenerPort.  We should get a sip invite on the listener socket.  

We get an 'EventACK' to our request followed by an 'EventAttachedDataChanged'.  This event contains the following attributes -

      'GSIP_REC_FN' [str] = "CE937U6TP90HF509O6NAINAN3O004O6H"
      'GSIP_RECORD' [str] = "ON"

We do not receive an invite on our listener socket.  The above attributes indicates that the media server is creating a recording file and not duplicating and sending us the RTP packets.

How do we make the media server forward the packets to us ?  Is that a change in the Genesys configuration ?  Will the RTP packets be received on the same port as the invite ?

Thanks for your help.
[Modify message]

Dave,

I recommend you open a case for this issue.  Please provide Sip Server and wireshark logs if you have them.

Thanks,
Roger Farr
Genesys Customer Care

Thanks Roger.  I have opened a case for this.

Where can I get any sample code which would implement the client sip negotiation and opening a connection for receiving the media ?

Thanks
Dave