Update dbus introspection files

src/dbus/callmanager-introspec.xml

@@ -4,7 +4,8 @@
 	<interface name="org.sflphone.SFLphone.CallManager">
 
 		<tp:docstring xmlns="http://www.w3.org/1999/xhtml">
-			<p>The CallManager interface is used to manage any call related actions.</p>
+			<p>The CallManager interface is used to manage
+			any call and conference related actions.</p>
 			<p>Since SFLphone-daemon support multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. SFLphone-clients is responsible to generate the callID on outgoing call. On the other hand, SFLphone-daemon will generate a unique callID on incoming calls.</p>
 		</tp:docstring>
 		<method name="placeCall" tp:name-for-bindings="placeCall">
@@ -116,7 +117,7 @@
 			</arg>
 		</method>
 
-		<method name="transfert" tp:name-for-bindings="transfert">
+		<method name="transfer" tp:name-for-bindings="transfer">
 			<tp:docstring>
 			  Transfer a call to given phone number.
 			</tp:docstring>
@@ -132,6 +133,22 @@
 			</arg>
 		</method>
 		
+		<method name="attendedTransfer" tp:name-for-bindings="attendedTransfer">
+			<tp:docstring>
+			  Perform an attended transfer on two calls
+			</tp:docstring>
+			<arg type="s" name="transferID" direction="in">
+			  <tp:docstring>
+			    The callID of the call to be transfered.
+			  </tp:docstring>
+			</arg>
+			<arg type="s" name="targetID" direction="in">
+			  <tp:docstring>
+			    The callID of the target call.
+			  </tp:docstring>
+			</arg>
+		</method>
+
 		<method name="playDTMF" tp:name-for-bindings="playDTMF">
 			<tp:docstring>
 			  Dual-Tone multi-frequency. Tell the core to play dial tones. A SIP INFO message is sent to notify the server.
@@ -195,6 +212,15 @@
 			<arg type="s" name="drag_callID" direction="in"/>
 		</method>
 
+                <method name="createConfFromParticipantList" tp:name-for-bindings="createConfFromParticipantList">
+			<tp:added version="0.9.14"/>
+			<tp:docstring>
+				<p>Create a conference from a list of participant</p>
+			<tp:rationale>The signal <tp:member-ref>conferenceCreated</tp:member-ref> is emitted on success.</tp:rationale>
+			</tp:docstring>
+			<arg type="as" name="participants" direction="in"/>
+		</method>
+
 		<method name="addParticipant" tp:name-for-bindings="addParticipant">
 			<tp:added version="0.9.7"/>
 			<tp:docstring>
@@ -254,13 +280,19 @@
 			</tp:docstring>
 			<arg type="s" name="callID" direction="in">
 			  <tp:docstring>
-					The call ID
+			    The conference ID
 			  </tp:docstring>
 			</arg>
 			<annotation name="com.trolltech.QtDBus.QtTypeName.Out0" value="MapStringString"/>
 			<arg type="a{ss}" name="infos" direction="out">
 			  <tp:docstring>
-					A map containing the ID of the conferences and their states.
+			    A map containing the ID of the conferences
+			    and their states:
+			    <ul>
+			      <li>ACTIVE_ATACHED</li>
+			      <li>ACTIVE_DETACHED</li>
+			      <li>HOLD</li>
+			    </ul>
 			  </tp:docstring>
 			</arg>
 		</method>
@@ -268,7 +300,11 @@
 		<method name="getConferenceList" tp:name-for-bindings="getConferenceList">
 			<tp:added version="0.9.7"/>
 			<tp:docstring>
-				Returns a list containing all active conferences.
+			  Returns a list containing all active
+			  conferences.
+			  <tp:rationale>To update client status, one should
+			  use <tp:member-ref>getParticipantList</tp:member-ref>
+			  with provided conference IDs.</tp:rationale> 
 			</tp:docstring>
 			<arg type="as" name="list" direction="out">
 			  <tp:docstring>
@@ -304,6 +340,21 @@
 			</arg>
 		</method>
 
+		<signal name="recordPlaybackFilepath" tp:name-for-bindings="recordPlaybackFilepath">
+			<tp:docstring>
+			  Once after starting recording for the first time, this signal is emited to 
+			  provide the recorded file path to client application.
+			</tp:docstring>
+			<arg type="s" name="callID" />
+			<arg type="s" name="filepath"/>
+		</signal>
+
+		<signal name="recordPlaybackStopped" tp:name-for-bindings="recordPlaybackStopped">
+			<tp:docstring/>
+			<arg type="s" name="filepath" />
+		</signal>
+			
+
 		<method name="getCallDetails" tp:name-for-bindings="getCallDetails">
 			<tp:docstring>
 			  Get all the details about a specific call.
@@ -343,35 +394,59 @@
 
 		<method name="getCurrentCallID" tp:name-for-bindings="getCurrentCallID">
 			<tp:docstring>
-				Get the CURRENT call ID.
+			  Unused
 			</tp:docstring>
-			@param[out] output callID
 			<arg type="s" name="callID" direction="out">
 			  <tp:docstring>	
-					The current call ID.	
 			  </tp:docstring>
 			</arg>
 		</method>
 
-		<method name="getCurrentCodecName" tp:name-for-bindings="getCurrentCodecName">
+		<method name="getCurrentAudioCodecName" tp:name-for-bindings="getCurrentAudioCodecName">
 			<tp:docstring>
+			  Unused
 			</tp:docstring>
 			<arg type="s" name="callID" direction="in"/>
 			<arg type="s" name="codecName" direction="out"/>
 		</method>
 
-		<signal name="currentSelectedCodec" tp:name-for-bindings="currentSelectedCodec">
+		<method name="sendTextMessage" tp:name-for-bindings="sendTextMessage">
 			<tp:docstring>
+				Send a text message to the specified call
 			</tp:docstring>
-			<arg type="s" name="callID" />
-			<arg type="s" name="codecName"/>
+			<arg type="s" name="callID" direction="in"/>
+			<arg type="s" name="message" direction="in"/>
+		</method>
+
+		<signal name="newCallCreated" tp:name-for-bindings="newCallCreated">
+			<tp:docstring>
+			  <p>Notify that a cell have been created.</p>
+			  <p>The callID generated by the daemon must be stored by the clients in order to address other action for
+			    this call. This signal is emitted when call have been created by the daemon itself.</p>
+			  <tp:rationale> The client must subscribe to this signal to handle calls created by other clients </tp:rationale>
+			</tp:docstring>
+			<arg type="s" name="accountID">
+			  <tp:docstring>
+			    The account ID of the calle. Clients must notify teh right account when receiving this signal.
+			  </tp:docstring>
+			</arg>
+			<arg type="s" name="callID">
+			  <tp:docstring>
+			    A new call ID.
+			  </tp:docstring>
+			</arg>
+			<arg type="s" name="to">
+			  <tp:docstring>
+			   The sip uri this call is trying to reach
+			  </tp:docstring>
+			</arg>
 		</signal>
 
 		<signal name="incomingCall" tp:name-for-bindings="incomingCall">
 			<tp:docstring>
 			  <p>Notify an incoming call.</p>
-				<p>The callID generated by the daemon and must be stored by the clients in order to address other action for
-					this call.</p>
+			  <p>The callID generated by the daemon must be stored by the clients in order to address other action for
+			    this call. This signal is emitted when we receive a call from a remote peer</p>
 			  <tp:rationale>The client must subscribe to this signal to handle incoming calls.</tp:rationale>
 			</tp:docstring>
 			<arg type="s" name="accountID">
@@ -393,9 +468,10 @@
 
 		<signal name="incomingMessage" tp:name-for-bindings="incomingMessage">
 			<tp:docstring>
-				Unused
+				Notify clients that a new text message has been received. 
 			</tp:docstring>
-			<arg type="s" name="accountID" />
+			<arg type="s" name="callID" />
+			<arg type="s" name="from" />
 			<arg type="s" name="message" />
 		</signal>
 
@@ -441,9 +517,17 @@
 			  <tp:docstring>
 			    The acceptable states are: 
 			    <ul>
-						<li>ACTIVE_ATTACHED</li>
-						<li>ACTIVE_DETACHED</li>
-						<li>HOLD</li>
+			      <li>ACTIVE_ATTACHED: SFLphone user is
+			      participating to this conference</li>
+			      <li>ACTIVE_DETACHED: This situation can
+			      occur if a call is received while
+			      SFLphone user is participating to a
+			      conference. In this case, one can leave
+			      the conference by answering the
+			      call. Other participants may continue
+			      conferencing normally.</li>
+			      <li>HOLD: Each call in this conference
+			      is on state HOLD</li>
 			    </ul>
 			  </tp:docstring>
 			</arg>
@@ -514,18 +598,43 @@
 			</arg>
 		</method>
  	        
+                <method name="startRecordedFilePlayback" tp:name-for-bindings="startRecordedFilePlayback">
+			<tp:added version="0.9.14"/>
+			<tp:docstring>
+			</tp:docstring>
+			<arg type="s" name="filepath" direction="in"/>
+			<arg type="b" name="result" direction="out"/>
+	        </method>
+
+		<method name="stopRecordedFilePlayback" tp:name-for-bindings="stopRecordedFilePlayback">
+			<tp:added version="0.9.14"/>
+			<tp:docstring/>
+			<arg type="s" name="filepath" direction="in"/>
+		</method>
+
 		<signal name="sipCallStateChanged" tp:name-for-bindings="sipCallStateChanged">
 			<tp:docstring>
+			  <p>Call state changed, SFLphone received a notification
+			    from registrar concerning this call.</p>
 			</tp:docstring>
 			<arg type="s" name="callID"  />
+			  <tp:docstring>
+			    The call ID
+			  </tp:docstring>
 			<arg type="s" name="state"  />
+			  <tp:docstring>
+			    Description string
+			  </tp:docstring>
 			<arg type="i" name="code"  />
+			  <tp:docstring>
+			    The SIP or IAX2 message code
+			  </tp:docstring>
 		</signal>    
 
 		<signal name="registrationStateChanged" tp:name-for-bindings="registrationStateChanged">
 			<tp:docstring>
-				<p>The state of an account has changed.</p>
-				<p>The clients have to subscribe to this signal to use accounts.</p>
+				<p>Account state changed, SFLphone received a notification
+				from registrar.</p>
 			</tp:docstring>
 			<arg type="s" name="accountID" >
 			  <tp:docstring>
@@ -534,12 +643,12 @@
 			</arg>
 			<arg type="s" name="state">
 			  <tp:docstring>
-					The new registration state
+			    Description string
 			  </tp:docstring>
 			</arg>
 			<arg type="i" name="code">            
 			  <tp:docstring>
-					The SIP or IAX2 code
+			    The SIP or IAX2 message code
 			  </tp:docstring>
 			</arg>
 		</signal> 
@@ -562,7 +671,11 @@
 
 		<signal name="volumeChanged" tp:name-for-bindings="volumeChanged">
 		        <tp:docstring>
-				Notify the clients of a volume level change.
+			  <p>Notify clients of a volume level
+			    change.</p> 
+			  <p>This signal occurs only if ALSA is
+			    enabled since Pulseaudio streams are
+			    managed externally. </p>
 			</tp:docstring>
 			<arg type="s" name="device">
 			  <tp:docstring>
@@ -578,19 +691,29 @@
 
 		<signal name="transferSucceded" tp:name-for-bindings="transferSucceded">
 			<tp:docstring>
-				The transfer has been successfully processed.
+			  <p>Transfer has been successfully
+			  processed. Client should remove transfered
+			  call from call list as it is no longer
+			  accessible in SFLphone-daemon.</p>
 			</tp:docstring>
 		</signal>
 
 		<signal name="transferFailed" tp:name-for-bindings="transferFailed">
 			<tp:docstring>
-				The transfer operation has failed.
+			  <p>Transfer operation failed. Corespondin
+			  call is no longer accessible in
+			  SFLphone-daemon.</p>
 			</tp:docstring>
 		</signal>
 
 		<signal name="secureSdesOn" tp:name-for-bindings="secureSdesOn">
 			<tp:added version="0.9.7"/>
 			<tp:docstring>
+			  <p>Signal sent on SDES session success. Media transmission is encripted
+			  for this call only. It does not apply for a
+			  conference.</p> 
+			  <p>A conference can be considered to be secured if and only if each
+			  participant is secured.</p>
 			</tp:docstring>
 			<arg type="s" name="callID"/>
 		</signal>
@@ -598,6 +721,9 @@
 		<signal name="secureSdesOff" tp:name-for-bindings="secureSdesOff">
 			<tp:added version="0.9.7"/>
 			<tp:docstring>
+			  <p>Sinal sent to notify that SDES session
+			  failed.</p> 
+			  <p>Media transmission is not encrypted.</p>
 			</tp:docstring>
 			<arg type="s" name="callID" />
 		</signal>