Commit 96770c32 authored by Emmanuel Lepage's avatar Emmanuel Lepage

Update dbus introspection files

parent 1546b6ce
......@@ -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>
......
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