11/09/2015: my jsSIP script

function registerUA() {

    console.log("Iniciando creación del UA SIP....");

    //Configuración de conexión
    var configuration = {
        uri: "sip:1001@"+webserver,
        password: xxxx,
        ws_servers: "ws://"+webserver+":8088/asterisk/ws",
        display_name: "UA WebRTC",
        authorization_user: null,
        register: null,
        register_expires: null,
        no_answer_timeout: null,
        trace_sip: true,
        stun_servers: null,
        turn_servers: null,
        use_preloaded_route: null,
        connection_recovery_min_interval: null,
        connection_recovery_max_interval: null,
        hack_via_tcp: null,
        hack_ip_in_contact: true
    };

    //Registrar el softphone
    ua = new JsSIP.UA(configuration);
    ua.start();
}

////////////////////////////////////////////////////
//Hace una llamada al número en el cuadro de texto//
////////////////////////////////////////////////////
function callAsterisk() {
    console.log("Intentando hacer una llamada ....");

    var numTel = document.getElementById('txtNumero').value;
    console.log(numTel);

    //DOM de contenedores de video
    var myMultimedia = document.getElementById('myMultimedia');

    var theirMultimedia = document.getElementById('theirMultimedia');

    // Register callbacks to desired call events
    var eventH = {
        'progress': function (e) {
            console.log('LLAMADA EN PROGRESO ...........');
        },
        'failed': function (e) {
            console.log('LA LLAMADA FALLÓ DEBIDO A: ' + e.data.cause);
        },
        'ended': function (e) {
            console.log('YA TERMINÓ LA LLAMADA: ');
        },
        'confirmed': function (e) {
            local_stream = session.connection.getLocalStreams()[0];
            console.log('LLAMADA CONFIRMADA');
            console.log(local_stream);
            // Attach local stream to selfView
            //myMultimedia = JsSIP.rtcninja.attachMediaStream(myMultimedia, local_stream);
            console.log(local_stream);
        },
        'addstream': function (e) {
            remote_stream = e.stream;
            console.log('FLUJO REMOTO RECIBIDO');
            console.log(remote_stream);
            // Attach remote stream to remoteView
            theirMultimedia = JsSIP.rtcninja.attachMediaStream(theirMultimedia, remote_stream);
            console.log(remote_stream);
        }
    };


    //parámetros de la llamada
    var options = {
        'eventHandlers': eventH,
        'mediaConstraints': {
            'audio': true,
            'video': false
        }
    };

    //Llamando
    session = ua.call('sip:' + numTel + '@'+webserver, options);
    console.log("Marcando ....");
}