[asterisk-commits] mvanbaak: branch mvanbaak/appdocsxml r119476 - /team/mvanbaak/appdocsxml/apps/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Jun 1 15:04:28 CDT 2008


Author: mvanbaak
Date: Sun Jun  1 15:04:27 2008
New Revision: 119476

URL: http://svn.digium.com/view/asterisk?view=rev&rev=119476
Log:
first shot at a format to document applications.
I could get all of Dial in it so it must at least a step in the correct direction

Modified:
    team/mvanbaak/appdocsxml/apps/app_dial.c

Modified: team/mvanbaak/appdocsxml/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/mvanbaak/appdocsxml/apps/app_dial.c?view=diff&rev=119476&r1=119475&r2=119476
==============================================================================
--- team/mvanbaak/appdocsxml/apps/app_dial.c (original)
+++ team/mvanbaak/appdocsxml/apps/app_dial.c Sun Jun  1 15:04:27 2008
@@ -65,6 +65,282 @@
 static char *app = "Dial";
 
 static char *synopsis = "Place a call and connect to the current channel";
+
+/*** APP
+	<application name="Dial">
+		<synopsis>
+			Dial(Technology/resourse[&Tech2/resource2...][,timout][,options][,URL])
+		</synopsis>
+		<description>
+			This application will place calls to one or more specified channels. As soon
+			as one of the requested channels answers, the originating channel will be
+			answered, if it has not already been answered. These two channels will then
+			be active in a bridged call. All other channels that were requested will then
+			be hung up.
+			Unless there is a timeout specified, the Dial application will wait
+			indefinitely until one of the called channels answers, the user hangs up, or
+			if all of the called channels are busy or unavailable. Dialplan executing will
+			continue if no requested channels can be called, or if the timeout expires.
+		</description>
+		<option name="A">
+			<argument name="x" required="true">
+				The file to play to the called party
+			</argument>
+			Play an announcement to the called party, using 'x' as the file
+		</option>
+		<option name="C">
+			Reset the CDR for this call.
+		</option>
+		<option name="c">
+			If DIAL cancels this call, always set the flag to tell the channel
+			driver that the call is answered elsewhere.
+		<option name="d">
+			Allow the calling user to dial a 1 digit extension while waiting for
+			a call to be answered. Exit to that extension if it exists in the
+			current context, or the context defined in the EXITCONTEXT variable,
+			if it exists.
+		</option>
+		<option name="D" argsep=":">
+			<argument name="called" />
+			<argument name="calling" />
+			Send the specified DTMF strings *after* the called\n
+			party has answered, but before the call gets bridged. The 'called'
+			DTMF string is sent to the called party, and the 'calling' DTMF
+			string is sent to the calling party. Both parameters can be used
+			alone.
+		</option>
+		<option name="e">
+			execute the 'h' extension for peer after the call ends
+		</option>
+		<option name="f">
+			Force the callerid of the *calling* channel to be set as the
+			extension associated with the channel using a dialplan 'hint'.
+			For example, some PSTNs do not allow CallerID to be set to anything
+			other than the number assigned to the caller.
+		</option>
+		<option name="F" argsep="^">
+			<argument name="context" />
+			<argument name="exten" />
+			<argument name="pri" required="true" />
+			When the caller hangs up, transfer the called party
+			to the specified context and extension and continue execution.
+		</option>
+		<option name="g">
+			Proceed with dialplan execution at the current extension if the
+			destination channel hangs up.
+		</option>
+		<option name="G" argsep="^">
+			<argument name="context" />
+			<argument name="exten" />
+			<argument name="pri" required="true" />
+			If the call is answered, transfer the calling party to
+			the specified priority and the called party to the specified priority+1.
+			Optionally, an extension, or extension and context may be specified.
+			Otherwise, the current extension is used. You cannot use any additional
+			action post answer options in conjunction with this option.
+		</option>
+		<option name="h">
+			Allow the called party to hang up by sending the '*' DTMF digit.
+		</option>
+		<option name="H">
+			Allow the calling party to hang up by hitting the '*' DTMF digit.
+		</option>
+		<option name="i">
+			Asterisk will ignore any forwarding requests it may receive on this
+			dial attempt.
+		</option>
+		<option name="k">
+			Allow the called party to enable parking of the call by sending
+			the DTMF sequence defined for call parking in features.conf.
+		</option>
+		<option name="K">
+			Allow the calling party to enable parking of the call by sending
+			the DTMF sequence defined for call parking in features.conf.
+		</option>
+		<option name="L" args="x,y,z" argsep=":">
+			<argument name="x" required="true">
+				Maximum calltime in miliseconds
+			</argument>
+			<argument name="y" />
+			<argument name="z" />
+			Limit the call to 'x' ms. Play a warning when 'y' ms are
+			left. Repeat the warning every 'z' ms.
+			<variable name="LIMIT_PLAYAUDIO_CALLER">
+				<value name="yes" default="true" />
+				<value name="no" />
+				Play sounds to the caller.
+			</variable>
+			<variable name="LIMIT_PLAYAUDIO_CALLEE">
+				<value name="yes" />
+				<value name="no" />
+				Play sounds to the callee.
+			</variable>
+			<variable name="LIMIT_TIMEOUT_FILE">
+				<value name="filename">
+					If not set, the time remaining will be said.
+				</value>
+				File to play when time is up.
+			</variable>
+			<variable name="LIMIT_CONNECT_FILE">
+				<value name="filename">
+					If not set, the time remaining will be said.
+				</value>
+				File to play when call begins.
+			</variable>
+			<variable name="LIMIT_WARNING_FILE">
+				<value name="filename">
+					If not set, the time remaining will be said.
+				</value>
+				File to play as warning if 'y' is defined.
+			<variable>
+		</option>
+		<option name="m">
+			<argument name="class" />
+			Provide hold music to the calling party until a requested
+			channel answers. A specific MusicOnHold class can be
+			specified.
+		</option>
+		<option name="M" args="x,arg" argsep="^">
+			<argument name="x" required="true">
+				Macro name that should be executed.
+			</argument>
+			<argument name="arg">
+				Macro arguments seperated by ^
+			</argument>
+			Execute the Macro for the *called* channel before connecting
+			to the calling channel. Arguments can be specified to the Macro
+			using '^' as a delimiter. The Macro can set the variable
+			MACRO_RESULT to specify the following actions after the Macro is
+			finished executing.
+			<variable name="MACRO_RESULT">
+				If set, this action will be taken after the macro finished executing.
+				<value name="ABORT">
+					Hangup both legs of the call.
+				</value>
+				<value name="CONGESTION">
+					Behave as if line congestion was encountered.
+				</value>
+				<value name="BUSY">
+					Behave as if a busy signal was encountered.
+				</value>
+				<value name="CONTINUE">
+					Hangup the called party and allow the calling party to continue dialplan execution at the next priority.
+				</value>
+				<value name="GOTO:<context>^<exten>^<priority>">
+					Transfer the call to the specified priority. Optionally, an extension, or extension and priority can be specified.
+				</value>
+			</variable>
+			You cannot use any additional action post answer options in conjunction
+			with this option. Also, pbx services are not run on the peer (called) channel,
+			so you will not be able to set timeouts via the TIMEOUT() function in this macro.
+		</option>
+		<option name="n">
+			This option is a modifier for the screen/privacy mode. It specifies
+			that no introductions are to be saved in the priv-callerintros
+			directory.
+		</option>
+		<option name="N">
+			This option is a modifier for the screen/privacy mode. It specifies
+			that if callerID is present, do not screen the call.
+		</option>
+		<option name="o">
+			Specify that the CallerID that was present on the *calling* channel
+			be set as the CallerID on the *called* channel. This was the
+			behavior of Asterisk 1.0 and earlier.
+		</option>
+		<option name="O">
+			<argument name="x" />
+			Operator Services\" mode (Zaptel channel to Zaptel channel
+			only, if specified on non-Zaptel interface, it will be ignored).
+			When the destination answers (presumably an operator services
+			station), the originator no longer has control of their line.
+			They may hang up, but the switch will not release their line
+			until the destination party hangs up (the operator). Specified
+			without an arg, or with 1 as an arg, the originator hanging up
+			will cause the phone to ring back immediately. With a 2 specified,
+			when the \"operator\" flashes the trunk, it will ring their phone
+			back.
+		</option>
+		<option name="p">
+			This option enables screening mode. This is basically Privacy mode
+			without memory.
+		</option>
+		<option name="P">
+			<argument name="x" />
+			Enable privacy mode. Use 'x' as the family/key in the database if
+			it is provided. The current extension is used if a database
+			family/key is not specified.
+		</option>
+		<option name="r">
+			Indicate ringing to the calling party. Pass no audio to the calling
+			party until the called channel has answered.
+		</option>
+		<option name="S">
+			<argument name="x" required="true" />
+			Hang up the call after 'x' seconds *after* the called party has
+			answered the call.
+		</option>
+		<option name="t">
+			Allow the called party to transfer the calling party by sending the
+			DTMF sequence defined in features.conf.
+		</option>
+		<option name="T">
+			Allow the calling party to transfer the called party by sending the
+			DTMF sequence defined in features.conf.
+		</option>
+		<option name="U" argsep="^">
+			<argument name="x" required="true">
+				routine to execute via Gosub
+			</argument>
+			<argument name="arg">
+				Arguments for the Gosub routine
+			</argument>
+			Execute via Gosub the routine 'x' for the *called* channel before connecting
+			to the calling channel. Arguments can be specified to the Gosub
+			using '^' as a delimiter. The Gosub routine can set the variable
+			GOSUB_RESULT to specify the following actions after the Gosub returns.
+			<variable name="GOSUB_RESULT">
+				<value name="ABORT">
+					Hangup both legs of the call.
+				</value>
+				<value name="CONGESTION">
+					Behave as if line congestion was encountered.
+				</value>
+				<value name="BUSY">
+					Behave as if a busy signal was encountered.
+				</value>
+				<value name="CONTINUE">
+					Hangup the called party and allow the calling party
+					to continue dialplan execution at the next priority.
+				</value>
+				<value name="GOTO:<context>^<exten>^<priority>">
+					Transfer the call to the
+					specified priority. Optionally, an extension, or
+					extension and priority can be specified.
+				</value>
+			</variable>
+			You cannot use any additional action post answer options in conjunction
+			with this option. Also, pbx services are not run on the peer (called) channel,
+			so you will not be able to set timeouts via the TIMEOUT() function in this routine.
+		</option>
+		<option name="w">
+			Allow the called party to enable recording of the call by sending
+			the DTMF sequence defined for one-touch recording in features.conf.
+		</option>
+		<option name="W">
+			Allow the calling party to enable recording of the call by sending
+			the DTMF sequence defined for one-touch recording in features.conf.
+		</option>
+		<option name="x">
+			Allow the called party to enable recording of the call by sending
+			the DTMF sequence defined for one-touch automixmonitor in features.conf
+		</option>
+		<option name="X">
+			Allow the calling party to enable recording of the call by sending
+			the DTMF sequence defined for one-touch automixmonitor in features.conf
+		</option>
+	</application>
+ ***/
 
 static char *descrip =
 "  Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):\n"




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