hi all ...  i'm using sip.js for my IVR project ... it could connect to freeswitch correctly but there is no audio between call legs ... first of all i have to say i really cannot use Verto . this is my function of when call button pressed 

  phoneCallButtonPressed : function(sessionid) {

            var s      = ctxSip.Sessions[sessionid],

                target = $("#numDisplay").val();

            if (!s) {

                $("#numDisplay").val("");

                ctxSip.sipCall(target);

            } else if (s.accept && !s.startTime) {

                s.accept({

                    media : {

                        stream      : ctxSip.Stream,

                        constraints : { audio : true, video : false },

                        render      : {

                            remote : { audio: $('#audioRemote').get()[0] }

                        },

                        RTCConstraints : { "optional": [{ 'DtlsSrtpKeyAgreement': 'true'} ]}

                    }

                });

            }

        },

-----------------------------------------------------------------------------------------

also i have to say that i tested it between my phone and x-lite ... X-lite could receive the voice but my phone could'nt ... i was using Zoiper in my phone.

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