Merge branch 'master' into issue-1657

* master:
  Updated RMS build to #43
  Patch to support nullable FinishOnKey for Recordings  This refer to #1886
  Patch for UserAgentManager to store all Contact header parameters in the Registration location This close #1887
  Patch for sequential dialing using Dial Action Url more than once. This close #1885

release/build.xml

@@ -17,7 +17,7 @@
   <property name="restcomm-sipservlets-as7.version" value="3.1.${sipservlets-build}"/>
   <!--<property name="jain-sip-ri.version" value="7.0.3.190" />-->
   <property name="restcomm-media.version" value="5.1.0"/>
-  <property name="restcomm-media.build" value="42"/>
+  <property name="restcomm-media.build" value="43"/>
   <!--property name="mobicents-diameter.version" value="6.1.2.GA" /-->
 
   <!-- mobicents SIP Servlets 7.x on JBoss AS7-->

restcomm/restcomm.docs/sources-asciidoc/src/main/asciidoc/rcml/record-rcml.adoc

@@ -12,7 +12,7 @@ The *<Record>* verb records the caller's voice and returns to you the URL of a f
 |action |relative or absolute URL |current document URL
 |method |GET, POST |POST
 |timeout |positive integer |5
-|finishOnKey |any digit, #, * |1234567890*#
+|finishOnKey |any digit, #, * or -1 to disable|1234567890*#
 |maxLength |integer greater than 1 with the number of seconds to wait |3600 (1 hour)
 |transcribe |true, false |false
 |transcribeCallback |relative or absolute URL |none
@@ -33,7 +33,7 @@ The *<Record>* verb records the caller's voice and returns to you the URL of a f
 
 * *method.* The 'method' attribute takes the value 'GET' or 'POST'. This tells RestComm whether to request the URL via HTTP GET or POST.
 * *timeout.* The 'timeout' attribute tells RestComm to end the recording after a number of seconds of silence has passed.
-* *finishOnKey.* The 'finishOnKey' attribute lets you choose a set of digits that end the recording when entered. For example, if you set 'finishOnKey' to '\#' and the caller presses '#', RestComm will immediately stop recording and submit 'RecordingUrl', 'RecordingDuration', and the '\#' as parameters in a request to the 'action' URL. The allowed values are the digits 0-9, '#' and '*'. Unlike <Gather>, you may specify more than one character as a 'finishOnKey' value.
+* *finishOnKey.* The 'finishOnKey' attribute lets you choose a set of digits that end the recording when entered. For example, if you set 'finishOnKey' to '\#' and the caller presses '#', RestComm will immediately stop recording and submit 'RecordingUrl', 'RecordingDuration', and the '\#' as parameters in a request to the 'action' URL. The allowed values are the digits 0-9, '#', '\*' and '-1'. The default value is '1234567890*#' which means that any key will end the recording. The value '-1' is a special value that can be used to disable finishOnKey and never stop recording on a DTMF. Unlike <Gather>, you may specify more than one character as a 'finishOnKey' value but '-1' must be used alone.
 * *maxLength.* The 'maxLength' attribute lets you set the maximum length for the recording in seconds.
 * *transcribe.* The 'transcribe' attribute tells RestComm that you would like a text representation of the audio of the recording.
 * *transcribeCallback.* The 'transcribeCallback' attribute is used in conjunction with the 'transcribe' attribute. It allows you to specify a URL to which RestComm will make an asynchronous POST request when the transcription is complete. This is not a request for RCML and the response will not change call flow, but the request will contain the standard RCML request parameters as well as 'TranscriptionStatus', 'TranscriptionText', 'TranscriptionUrl' and 'RecordingUrl'. If 'transcribeCallback' is specified, then there is no need to specify 'transcribe=true'. It is implied. If you specify 'transcribe=true' without a 'transcribeCallback', the completed transcription will be stored for you to retrieve later (see the REST API Transcriptions section), but RestComm will not asynchronously notify your application.
@@ -52,11 +52,11 @@ The *<Record>* verb records the caller's voice and returns to you the URL of a f
 * *playBeep.* The 'playBeep' attribute allows you to toggle between playing a sound before the start of a recording.
 * *Nesting.* The <Record> verb can not have any other verbs or nouns nested.
 
-=== Examples 
+=== Examples
 For an example of how to use the <Record> verb see below.
 
 ----
 <Response>
  <Record maxLength="30"/>
 </Response>
-----
+----

restcomm/restcomm.interpreter/src/main/java/org/restcomm/connect/interpreter/BaseVoiceInterpreter.java

@@ -1654,11 +1654,13 @@ public void execute(final Object message) throws Exception {
             if (attribute != null) {
                 finishOnKey = attribute.value();
                 if (finishOnKey != null && !finishOnKey.isEmpty()) {
-                    if (!PATTERN.matcher(finishOnKey).matches()) {
-                        final Notification notification = notification(WARNING_NOTIFICATION, 13613, finishOnKey
-                                + " is not a valid finishOnKey value");
-                        notifications.addNotification(notification);
-                        finishOnKey = "1234567890*#";
+                    if (!finishOnKey.equals("-1")) {
+                        if (!PATTERN.matcher(finishOnKey).matches()) {
+                            final Notification notification = notification(WARNING_NOTIFICATION, 13613, finishOnKey
+                                    + " is not a valid finishOnKey value");
+                            notifications.addNotification(notification);
+                            finishOnKey = "1234567890*#";
+                        }
                     }
                 } else {
                     finishOnKey = "1234567890*#";

restcomm/restcomm.interpreter/src/main/java/org/restcomm/connect/interpreter/VoiceInterpreter.java

@@ -857,6 +857,12 @@ private void onTagMessage(Object message) throws TransitionFailedException, Tran
                 fsm.transition(message, initializingCall);
             }
         } else if (Verbs.dial.equals(verb.name())) {
+            Attribute action = verb.attribute("action");
+            if (action != null && dialActionExecuted) {
+                //We have a new Dial verb that contains Dial Action URL again.
+                //We set dialActionExecuted to false in order to execute Dial Action again
+                dialActionExecuted = false;
+            }
             dialRecordAttribute = verb.attribute("record");
             fsm.transition(message, startDialing);
         } else if (Verbs.fax.equals(verb.name())) {

restcomm/restcomm.mscontrol.mms/src/main/java/org/restcomm/connect/mscontrol/mms/MgcpMediaGroup.java

@@ -376,7 +376,11 @@ protected void record(final Object message) {
         builder.setPreSpeechTimer(request.timeout());
         builder.setPostSpeechTimer(request.timeout());
         builder.setRecordingLength(request.length());
-        builder.setEndInputKey(request.endInputKey());
+        if (!request.endInputKey().equals("-1")) {
+            builder.setEndInputKey(request.endInputKey());
+        } else {
+            builder.setEndInputKey("null");
+        }
         builder.setRecordingId(request.destination());
         this.lastEvent = AUMgcpEvent.aupr;
         stop(lastEvent);

restcomm/restcomm.telephony/src/main/java/org/restcomm/connect/telephony/ua/UserAgentManager.java

@@ -19,34 +19,11 @@
  */
 package org.restcomm.connect.telephony.ua;
 
-import static java.lang.Integer.parseInt;
-import static javax.servlet.sip.SipServlet.OUTBOUND_INTERFACES;
-import static javax.servlet.sip.SipServletResponse.SC_OK;
-import static javax.servlet.sip.SipServletResponse.SC_PROXY_AUTHENTICATION_REQUIRED;
-import static javax.servlet.sip.SipServletResponse.SC_UNAUTHORIZED;
-import static org.restcomm.connect.commons.util.HexadecimalUtils.toHex;
-
-import java.io.IOException;
-import java.net.UnknownHostException;
-import java.util.HashMap;
-import java.util.Iterator;
-import java.util.List;
-import java.util.Map;
-import java.util.UUID;
-
-import javax.servlet.ServletContext;
-import javax.servlet.ServletException;
-import javax.servlet.sip.Address;
-import javax.servlet.sip.Parameterable;
-import javax.servlet.sip.ServletParseException;
-import javax.servlet.sip.SipApplicationSession;
-import javax.servlet.sip.SipFactory;
-import javax.servlet.sip.SipServletMessage;
-import javax.servlet.sip.SipServletRequest;
-import javax.servlet.sip.SipServletResponse;
-import javax.servlet.sip.SipSession;
-import javax.servlet.sip.SipURI;
-
+import akka.actor.ActorRef;
+import akka.actor.ReceiveTimeout;
+import akka.actor.UntypedActor;
+import akka.event.Logging;
+import akka.event.LoggingAdapter;
 import org.apache.commons.configuration.Configuration;
 import org.joda.time.DateTime;
 import org.restcomm.connect.commons.configuration.RestcommConfiguration;
@@ -62,11 +39,32 @@
 import org.restcomm.connect.telephony.api.Hangup;
 import org.restcomm.connect.telephony.api.UserRegistration;
 
-import akka.actor.ActorRef;
-import akka.actor.ReceiveTimeout;
-import akka.actor.UntypedActor;
-import akka.event.Logging;
-import akka.event.LoggingAdapter;
+import javax.servlet.ServletContext;
+import javax.servlet.ServletException;
+import javax.servlet.sip.Address;
+import javax.servlet.sip.Parameterable;
+import javax.servlet.sip.ServletParseException;
+import javax.servlet.sip.SipApplicationSession;
+import javax.servlet.sip.SipFactory;
+import javax.servlet.sip.SipServletMessage;
+import javax.servlet.sip.SipServletRequest;
+import javax.servlet.sip.SipServletResponse;
+import javax.servlet.sip.SipSession;
+import javax.servlet.sip.SipURI;
+import java.io.IOException;
+import java.net.UnknownHostException;
+import java.util.HashMap;
+import java.util.Iterator;
+import java.util.List;
+import java.util.Map;
+import java.util.UUID;
+
+import static java.lang.Integer.parseInt;
+import static javax.servlet.sip.SipServlet.OUTBOUND_INTERFACES;
+import static javax.servlet.sip.SipServletResponse.SC_OK;
+import static javax.servlet.sip.SipServletResponse.SC_PROXY_AUTHENTICATION_REQUIRED;
+import static javax.servlet.sip.SipServletResponse.SC_UNAUTHORIZED;
+import static org.restcomm.connect.commons.util.HexadecimalUtils.toHex;
 
 /**
  * @author quintana.thomas@gmail.com (Thomas Quintana)
@@ -490,11 +488,13 @@ private void register(final Object message) throws Exception {
         final SipURI uri = (SipURI) contact.getURI();
         final String ip = request.getInitialRemoteAddr();
         final int port = request.getInitialRemotePort();
+
         String transport = (uri.getTransportParam()==null?request.getParameter("transport"):uri.getTransportParam()); //Issue #935, take transport of initial request-uri if contact-uri has no transport parameter
         if (transport == null && !request.getInitialTransport().equalsIgnoreCase("udp")) {
             //Issue1068, if Contact header or RURI doesn't specify transport, check InitialTransport from
             transport = request.getInitialTransport();
         }
+
         boolean isLBPresent = false;
         //Issue 306: https://telestax.atlassian.net/browse/RESTCOMM-306
         final String initialIpBeforeLB = request.getHeader("X-Sip-Balancer-InitialRemoteAddr");
@@ -518,6 +518,17 @@ private void register(final Object message) throws Exception {
         if (transport != null) {
             buffer.append(";transport=").append(transport);
         }
+
+        Iterator<String> extraParameterNames = uri.getParameterNames();
+        while (extraParameterNames.hasNext()) {
+            String paramName = extraParameterNames.next();
+            String paramValue = uri.getParameter(paramName);
+            buffer.append(";");
+            buffer.append(paramName);
+            buffer.append("=");
+            buffer.append(paramValue);
+        }
+
         final String address = buffer.toString();
         // Prepare the response.
         final SipServletResponse response = request.createResponse(SC_OK);

restcomm/restcomm.testsuite/src/test/java/org/restcomm/connect/testsuite/telephony/ClientsDialTest.java

@@ -3,6 +3,7 @@
 import com.github.tomakehurst.wiremock.junit.WireMockRule;
 import com.google.gson.JsonArray;
 import com.google.gson.JsonObject;
+import gov.nist.javax.sip.address.SipUri;
 import org.cafesip.sipunit.Credential;
 import org.cafesip.sipunit.SipCall;
 import org.cafesip.sipunit.SipPhone;
@@ -52,7 +53,7 @@
 
 /**
  * Test for clients with or without VoiceURL (Bitbucket issue 115). Clients without VoiceURL can dial anything.
- * 
+ *
  * @author <a href="mailto:gvagenas@gmail.com">gvagenas</a>
  */
 @RunWith(Arquillian.class)
@@ -187,7 +188,7 @@ public class ClientsDialTest {
     private String pstnNumber = "+151261006100";
 
     private String clientPassword = "qwerty1234RT";
-    
+
     // Maria is a Restcomm Client **without** VoiceURL. This Restcomm Client can dial anything.
     private SipStack mariaSipStack;
     private SipPhone mariaPhone;
@@ -412,7 +413,7 @@ public void testClientsCallEachOther() throws ParseException, InterruptedExcepti
         assertTrue(!(mariaCall.getLastReceivedResponse().getStatusCode() >= 400));
 
         assertTrue(dimitriCall.waitForAck(3000));
-        
+
         //Talk time ~ 3sec
         Thread.sleep(3000);
         dimitriCall.listenForDisconnect();
@@ -421,11 +422,11 @@ public void testClientsCallEachOther() throws ParseException, InterruptedExcepti
         assertTrue(dimitriCall.waitForDisconnect(5 * 1000));
         assertTrue(dimitriCall.respondToDisconnect());
         long endTime   = System.currentTimeMillis();
-        
+
         double totalTime = (endTime - startTime)/1000.0;
         assertTrue(3.0 <= totalTime);
         assertTrue(totalTime <= 4.0);
-        
+
         Thread.sleep(3000);
 
         //Check CDR
@@ -711,7 +712,7 @@ public void testClientDialOutPstnWebRTCClientwithSDP() throws ParseException, In
     public void testClientDialToInvalidNumber() throws ParseException, InterruptedException, InvalidArgumentException, SipException {
         String invalidNumber = "+123456789";
         SipPhone outboundProxy = georgeSipStack.createSipPhone("127.0.0.1", SipStack.PROTOCOL_UDP, 5080, "sip:"+invalidNumber+"@127.0.0.1:5070");
-        
+
         assertNotNull(mariaRestcommClientSid);
         assertNotNull(dimitriRestcommClientSid);
 
@@ -732,7 +733,7 @@ public void testClientDialToInvalidNumber() throws ParseException, InterruptedEx
 
         final SipCall georgeCall = outboundProxy.createSipCall();
         georgeCall.listenForIncomingCall();
-        
+
         georgeCall.waitForIncomingCall(5 * 1000);
         georgeCall.sendIncomingCallResponse(Response.NOT_FOUND, "Not-Found George", 3600);
 
@@ -750,7 +751,7 @@ public void testClientDialToInvalidNumber() throws ParseException, InterruptedEx
 
         outboundProxy.dispose();
     }
-    
+
     @Test
     public void testClientDialOutPstnCancelBefore200() throws ParseException, InterruptedException {
 
@@ -859,6 +860,68 @@ public synchronized void testDialClientAlice() throws InterruptedException, Pars
         assertTrue(aliceCall.respondToDisconnect());
     }
 
+    @Test
+    public synchronized void testDialClientAliceWithExtraParamsAtContactHeader() throws InterruptedException, ParseException {
+        stubFor(get(urlPathEqualTo("/1111"))
+                .willReturn(aResponse()
+                        .withStatus(200)
+                        .withHeader("Content-Type", "text/xml")
+                        .withBody(dialClientRcml)));
+
+        String extraParamName1 = "rc-id";
+        String extraParamValue1 = "7616";
+        String extraParamName2 = "my-param";
+        String extraParamValue2 = "test";
+        aliceContact = aliceContact+";"+extraParamName1+"="+extraParamValue1+";"+extraParamName2+"="+extraParamValue2;
+        // Phone2 register as alice
+        SipURI uri = aliceSipStack.getAddressFactory().createSipURI(null, "127.0.0.1:5080");
+        assertTrue(alicePhone.register(uri, "alice", "1234", aliceContact, 3600, 3600));
+
+        // Prepare second phone to receive call
+        SipCall aliceCall = alicePhone.createSipCall();
+        aliceCall.listenForIncomingCall();
+
+        // Create outgoing call with first phone
+        final SipCall georgeCall = georgePhone.createSipCall();
+        georgeCall.initiateOutgoingCall(georgeContact, "sip:1111@127.0.0.1:5080", null, body, "application", "sdp", null, null);
+        assertLastOperationSuccess(georgeCall);
+        assertTrue(georgeCall.waitOutgoingCallResponse(5 * 1000));
+        final int response = georgeCall.getLastReceivedResponse().getStatusCode();
+        assertTrue(response == Response.TRYING || response == Response.RINGING);
+
+        if (response == Response.TRYING) {
+            assertTrue(georgeCall.waitOutgoingCallResponse(5 * 1000));
+            assertEquals(Response.RINGING, georgeCall.getLastReceivedResponse().getStatusCode());
+        }
+
+        assertTrue(georgeCall.waitOutgoingCallResponse(5 * 1000));
+        assertEquals(Response.OK, georgeCall.getLastReceivedResponse().getStatusCode());
+
+        georgeCall.sendInviteOkAck();
+        assertTrue(!(georgeCall.getLastReceivedResponse().getStatusCode() >= 400));
+
+        assertTrue(aliceCall.waitForIncomingCall(30 * 1000));
+
+        SipUri ruri = (SipUri) aliceCall.getLastReceivedRequest().getRequestEvent().getRequest().getRequestURI();
+
+        assertEquals(extraParamValue1, ruri.getParameter(extraParamName1));
+        assertEquals(extraParamValue2, ruri.getParameter(extraParamName2));
+
+        assertTrue(aliceCall.sendIncomingCallResponse(Response.RINGING, "Ringing-Alice", 3600));
+        String receivedBody = new String(aliceCall.getLastReceivedRequest().getRawContent());
+        assertTrue(aliceCall.sendIncomingCallResponse(Response.OK, "OK-Alice", 3600, receivedBody, "application", "sdp", null,
+                null));
+        assertTrue(aliceCall.waitForAck(50 * 1000));
+
+        Thread.sleep(3000);
+
+        // hangup.
+        georgeCall.disconnect();
+
+        aliceCall.listenForDisconnect();
+        assertTrue(aliceCall.waitForDisconnect(30 * 1000));
+        assertTrue(aliceCall.respondToDisconnect());
+    }
 
     private String dialWebRTCClientRcml = "<Response><Dial timeLimit=\"10\" timeout=\"10\"><Client>bob</Client></Dial></Response>";
     @Test