Merge changes from team/group/appdocsxml

This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 390 insertions, 182 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index f0fd7463d..089523ef4 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -62,188 +62,396 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/global_datastores.h"
 #include "asterisk/dsp.h"
 
-static char *app = "Dial";
+/*** DOCUMENTATION
+	<application name="Dial" language="en_US">
+		<synopsis>
+			Attempt to connect to another device or endpoint and bridge the call.
+		</synopsis>
+		<syntax>
+			<parameter name="Technology/Resource" required="true" argsep="&amp;">
+				<argument name="Technology/Resource" required="true">
+					<para>Specification of the device(s) to dial.  These must be in the format of
+					<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
+					represents a particular channel driver, and <replaceable>Resource</replaceable>
+					represents a resource available to that particular channel driver.</para>
+				</argument>
+				<argument name="Technology2/Resource2" required="false" multiple="true">
+					<para>Optional extra devices to dial in parallel</para>
+					<para>If you need more then one enter them as
+					Technology2/Resource2&amp;Technology3/Resourse3&amp;.....</para>
+				</argument>
+			</parameter>
+			<parameter name="timeout" required="false">
+				<para>Specifies the number of seconds we attempt to dial the specified devices</para>
+				<para>If not specified, this defaults to 136 years.</para>
+			</parameter>
+			<parameter name="options" required="false">
+			   <optionlist>
+				<option name="A">
+					<argument name="x" required="true">
+						<para>The file to play to the called party</para>
+					</argument>
+					<para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
+				</option>
+				<option name="C">
+					<para>Reset the call detail record (CDR) for this call.</para>
+				</option>
+				<option name="c">
+					<para>If the Dial() application cancels this call, always set the flag to tell the channel
+					driver that the call is answered elsewhere.</para>
+				</option>
+				<option name="d">
+					<para>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 <variable>EXITCONTEXT</variable> variable,
+					if it exists.</para>
+				</option>
+				<option name="D" argsep=":">
+					<argument name="called" />
+					<argument name="calling" />
+					<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
+					party has answered, but before the call gets bridged. The 
+					<replaceable>called</replaceable> DTMF string is sent to the called party, and the 
+					<replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments 
+					can be used alone.</para>
+				</option>
+				<option name="e">
+					<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
+				</option>
+				<option name="f">
+					<para>Force the callerid of the <emphasis>calling</emphasis> channel to be set as the
+					extension associated with the channel using a dialplan <literal>hint</literal>.
+					For example, some PSTNs do not allow CallerID to be set to anything
+					other than the number assigned to the caller.</para>
+				</option>
+				<option name="F" argsep="^">
+					<argument name="context" required="false" />
+					<argument name="exten" required="false" />
+					<argument name="priority" required="true" />
+					<para>When the caller hangs up, transfer the called party
+					to the specified destination and continue execution at that location.</para>
+				</option>
+				<option name="g">
+					<para>Proceed with dialplan execution at the next priority in the current extension if the
+					destination channel hangs up.</para>
+				</option>
+				<option name="G" argsep="^">
+					<argument name="context" required="false" />
+					<argument name="exten" required="false" />
+					<argument name="priority" required="true" />
+					<para>If the call is answered, transfer the calling party to
+					the specified <replaceable>priority</replaceable> and the called party to the specified 
+					<replaceable>priority</replaceable> plus one.</para>
+					<note>
+						<para>You cannot use any additional action post answer options in conjunction with this option.</para>
+					</note>
+				</option>
+				<option name="h">
+					<para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
+				</option>
+				<option name="H">
+					<para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
+				</option>
+				<option name="i">
+					<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
+				</option>
+				<option name="k">
+					<para>Allow the called party to enable parking of the call by sending
+					the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="K">
+					<para>Allow the calling party to enable parking of the call by sending
+					the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="L" argsep=":">
+					<argument name="x" required="true">
+						<para>Maximum call time, in milliseconds</para>
+					</argument>
+					<argument name="y">
+						<para>Warning time, in milliseconds</para>
+					</argument>
+					<argument name="z">
+						<para>Repeat time, in milliseconds</para>
+					</argument>
+					<para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
+					left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
+					<para>This option is affected by the following variables:</para>
+					<variablelist>
+						<variable name="LIMIT_PLAYAUDIO_CALLER">
+							<value name="yes" default="true" />
+							<value name="no" />
+							<para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
+						</variable>
+						<variable name="LIMIT_PLAYAUDIO_CALLEE">
+							<value name="yes" />
+							<value name="no" default="true"/>
+							<para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
+						</variable>
+						<variable name="LIMIT_TIMEOUT_FILE">
+							<value name="filename"/>
+							<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
+							If not set, the time remaining will be announced.</para>
+						</variable>
+						<variable name="LIMIT_CONNECT_FILE">
+							<value name="filename"/>
+							<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
+							If not set, the time remaining will be announced.</para>
+						</variable>
+						<variable name="LIMIT_WARNING_FILE">
+							<value name="filename"/>
+							<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
+							a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
+						</variable>
+					</variablelist>
+				</option>
+				<option name="m">
+					<argument name="class" required="false"/>
+					<para>Provide hold music to the calling party until a requested
+					channel answers. A specific music on hold <replaceable>class</replaceable>
+					(as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
+				</option>
+				<option name="M" argsep="^">
+					<argument name="macro" required="true">
+						<para>Name of the macro that should be executed.</para>
+					</argument>
+					<argument name="arg" multiple="true">
+						<para>Macro arguments</para>
+					</argument>
+					<para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel 
+					before connecting to the calling channel. Arguments can be specified to the Macro
+					using <literal>^</literal> as a delimiter. The macro can set the variable
+					<variable>MACRO_RESULT</variable> to specify the following actions after the macro is
+					finished executing:</para>
+					<variablelist>
+						<variable name="MACRO_RESULT">
+							<para>If set, this action will be taken after the macro finished executing.</para>
+							<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>
+							<!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
+							<value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
+								Transfer the call to the specified destination.
+							</value>
+						</variable>
+					</variablelist>
+					<note>
+						<para>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.</para>
+					</note>
+				</option>
+				<option name="n">
+					<para>This option is a modifier for the call screening/privacy mode. (See the 
+					<literal>p</literal> and <literal>P</literal> options.) It specifies
+					that no introductions are to be saved in the <directory>priv-callerintros</directory>
+					directory.</para>
+				</option>
+				<option name="N">
+					<para>This option is a modifier for the call screening/privacy mode. It specifies
+					that if Caller*ID is present, do not screen the call.</para>
+				</option>
+				<option name="o">
+					<para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
+					be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
+					behavior of Asterisk 1.0 and earlier.</para>
+				</option>
+				<option name="O">
+					<argument name="mode">
+						<para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
+						the originator hanging up will cause the phone to ring back immediately.</para>
+						<para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator 
+						flashes the trunk, it will ring their phone back.</para>
+					</argument>
+					<para>Enables <emphasis>operator services</emphasis> mode.  This option only
+					works when bridging a DAHDI channel to another DAHDI channel
+					only. if specified on non-DAHDI interfaces, 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 (the operator) hangs up.</para>
+				</option>
+				<option name="p">
+					<para>This option enables screening mode. This is basically Privacy mode
+					without memory.</para>
+				</option>
+				<option name="P">
+					<argument name="x" />
+					<para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
+					it is provided. The current extension is used if a database family/key is not specified.</para>
+				</option>
+				<option name="r">
+					<para>Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
+					party until the called channel has answered.</para>
+				</option>
+				<option name="S">
+					<argument name="x" required="true" />
+					<para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
+					answered the call.</para>
+				</option>
+				<option name="t">
+					<para>Allow the called party to transfer the calling party by sending the
+					DTMF sequence defined in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="T">
+					<para>Allow the calling party to transfer the called party by sending the
+					DTMF sequence defined in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="U" argsep="^">
+					<argument name="x" required="true">
+						<para>Name of the subroutine to execute via Gosub</para>
+					</argument>
+					<argument name="arg" multiple="true" required="false">
+						<para>Arguments for the Gosub routine</para>
+					</argument>
+					<para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
+					to the calling channel. Arguments can be specified to the Gosub
+					using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
+					<variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
+					<variablelist>
+						<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>
+							<!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
+							<value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
+								Transfer the call to the specified priority. Optionally, an extension, or
+								extension and priority can be specified.
+							</value>
+						</variable>
+					</variablelist>
+					<note>
+						<para>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.</para>
+					</note>
+				</option>
+				<option name="w">
+					<para>Allow the called party to enable recording of the call by sending
+					the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="W">
+					<para>Allow the calling party to enable recording of the call by sending
+					the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="x">
+					<para>Allow the called party to enable recording of the call by sending
+					the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
+				</option>
+				<option name="X">
+					<para>Allow the calling party to enable recording of the call by sending
+					the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
+				</option>
+				</optionlist>
+			</parameter>
+			<parameter name="URL">
+				<para>The optional URL will be sent to the called party if the channel driver supports it.</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>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.</para>
+
+			<para>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.
+			This application will report normal termination if the originating channel
+			hangs up, or if the call is bridged and either of the parties in the bridge
+			ends the call.</para>
+
+			<para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
+			application will be put into that group (as in Set(GROUP()=...).
+			If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
+			application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
+			however, the variable will be unset after use.</para>
+
+			<para>This application sets the following channel variables:</para>
+			<variablelist>
+				<variable name="DIALEDTIME">
+					<para>This is the time from dialing a channel until when it is disconnected.</para>
+				</variable>
+				<variable name="ANSWEREDTIME">
+					<para>This is the amount of time for actual call.</para>
+				</variable>
+				<variable name="DIALSTATUS">
+					<para>This is the status of the call</para>
+					<value name="CHANUNAVAIL" />
+					<value name="CONGESTION" />
+					<value name="NOANSWER" />
+					<value name="BUSY" />
+					<value name="ANSWER" />
+					<value name="CANCEL" />
+					<value name="DONTCALL">
+						For the Privacy and Screening Modes.
+						Will be set if the called party chooses to send the calling party to the 'Go Away' script.
+					</value>
+					<value name="TORTURE">
+						For the Privacy and Screening Modes.
+						Will be set if the called party chooses to send the calling party to the 'torture' script.
+					</value>
+					<value name="INVALIDARGS" />
+				</variable>
+			</variablelist>
+		</description>
+	</application>
+	<application name="RetryDial" language="en_US">
+		<synopsis>
+			Place a call, retrying on failure allowing an optional exit extension.
+		</synopsis>
+		<syntax>
+			<parameter name="announce" required="true">
+				<para>Filename of sound that will be played when no channel can be reached</para>
+			</parameter>
+			<parameter name="sleep" required="true">
+				<para>Number of seconds to wait after a dialattempt failed before a new attempt is made</para>
+			</parameter>
+			<parameter name="retries" required="true">
+				<para>Number of retries</para>
+				<para>When this is reached flow will continue at the next priority in the dialplan</para>
+			</parameter>
+			<parameter name="dialargs" required="true">
+				<para>Same format as arguments provided to the Dial application</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>This application will attempt to place a call using the normal Dial application.
+			If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
+			Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
+			After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
+			If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
+			While waiting to retry a call, a 1 digit extension may be dialed. If that
+			extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
+			one, The call will jump to that extension immediately.
+			The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
+			to the Dial application.</para>
+		</description>
+	</application>
+ ***/
 
-static char *synopsis = "Place a call and connect to the current channel";
-
-static char *descrip =
-"  Dial(Technology/resource[&Tech2/resource2...][,timeout][,options][,URL]):\n"
-"This application will place calls to one or more specified channels. As soon\n"
-"as one of the requested channels answers, the originating channel will be\n"
-"answered, if it has not already been answered. These two channels will then\n"
-"be active in a bridged call. All other channels that were requested will then\n"
-"be hung up.\n"
-"  Unless there is a timeout specified, the Dial application will wait\n"
-"indefinitely until one of the called channels answers, the user hangs up, or\n"
-"if all of the called channels are busy or unavailable. Dialplan executing will\n"
-"continue if no requested channels can be called, or if the timeout expires.\n\n"
-"  This application sets the following channel variables upon completion:\n"
-"    DIALEDTIME   - This is the time from dialing a channel until when it\n"
-"                   is disconnected.\n"
-"    ANSWEREDTIME - This is the amount of time for actual call.\n"
-"    DIALSTATUS   - This is the status of the call:\n"
-"                   CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
-"                   DONTCALL | TORTURE | INVALIDARGS\n"
-"  For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
-"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
-"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
-"wants to send the caller to the 'torture' script.\n"
-"  This application will report normal termination if the originating channel\n"
-"hangs up, or if the call is bridged and either of the parties in the bridge\n"
-"ends the call.\n"
-"  The optional URL will be sent to the called party if the channel supports it.\n"
-"  If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
-"application will be put into that group (as in Set(GROUP()=...).\n"
-"  If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this\n"
-"application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,\n"
-"however, the variable will be unset after use.\n\n"
-"  Options:\n"
-"    A(x) - Play an announcement to the called party, using 'x' as the file.\n"
-"    C    - Reset the CDR for this call.\n"
-"    c    - If DIAL cancels this call, always set the flag to tell the channel\n"
-"           driver that the call is answered elsewhere.\n"
-"    d    - Allow the calling user to dial a 1 digit extension while waiting for\n"
-"           a call to be answered. Exit to that extension if it exists in the\n"
-"           current context, or the context defined in the EXITCONTEXT variable,\n"
-"           if it exists.\n"
-"    D([called][:calling]) - Send the specified DTMF strings *after* the called\n"
-"           party has answered, but before the call gets bridged. The 'called'\n"
-"           DTMF string is sent to the called party, and the 'calling' DTMF\n"
-"           string is sent to the calling party. Both parameters can be used\n"
-"           alone.\n"
-"    e    - execute the 'h' extension for peer after the call ends. This\n"
-"           operation will not be performed if the peer was parked\n"
-"    f    - Force the callerid of the *calling* channel to be set as the\n"
-"           extension associated with the channel using a dialplan 'hint'.\n"
-"           For example, some PSTNs do not allow CallerID to be set to anything\n"
-"           other than the number assigned to the caller.\n"
-"    F(context^exten^pri) - When the caller hangs up, transfer the called party\n"
-"           to the specified context and extension and continue execution.\n"
-"    g    - Proceed with dialplan execution at the current extension if the\n"
-"           destination channel hangs up.\n"
-"    G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
-"           the specified priority and the called party to the specified priority+1.\n"
-"           Optionally, an extension, or extension and context may be specified. \n"
-"           Otherwise, the current extension is used. You cannot use any additional\n"
-"           action post answer options in conjunction with this option.\n"
-"    h    - Allow the called party to hang up by sending the '*' DTMF digit, or\n"
-"           whatever sequence was defined in the featuremap section for\n"
-"           'disconnect' in features.conf\n"
-"    H    - Allow the calling party to hang up by hitting the '*' DTMF digit, or\n"
-"           whatever sequence was defined in the featuremap section for\n"
-"           'disconnect' in features.conf\n"
-"    i    - Asterisk will ignore any forwarding requests it may receive on this\n"
-"           dial attempt.\n"
-"    k    - Allow the called party to enable parking of the call by sending\n"
-"           the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
-"    K    - Allow the calling party to enable parking of the call by sending\n"
-"           the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
-"    L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
-"           left. Repeat the warning every 'z' ms. The following special\n"
-"           variables can be used with this option:\n"
-"           * LIMIT_PLAYAUDIO_CALLER   yes|no (default yes)\n"
-"                                      Play sounds to the caller.\n"
-"           * LIMIT_PLAYAUDIO_CALLEE   yes|no\n"
-"                                      Play sounds to the callee.\n"
-"           * LIMIT_TIMEOUT_FILE       File to play when time is up.\n"
-"           * LIMIT_CONNECT_FILE       File to play when call begins.\n"
-"           * LIMIT_WARNING_FILE       File to play as warning if 'y' is defined.\n"
-"                                      The default is to say the time remaining.\n"
-"    m([class]) - Provide hold music to the calling party until a requested\n"
-"           channel answers. A specific MusicOnHold class can be\n"
-"           specified.\n"
-"    M(x[^arg]) - Execute the Macro for the *called* channel before connecting\n"
-"           to the calling channel. Arguments can be specified to the Macro\n"
-"           using '^' as a delimiter. The Macro can set the variable\n"
-"           MACRO_RESULT to specify the following actions after the Macro is\n"
-"           finished executing.\n"
-"           * ABORT        Hangup both legs of the call.\n"
-"           * CONGESTION   Behave as if line congestion was encountered.\n"
-"           * BUSY         Behave as if a busy signal was encountered.\n"
-"           * CONTINUE     Hangup the called party and allow the calling party\n"
-"                          to continue dialplan execution at the next priority.\n"
-"           * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
-"                          specified priority. Optionally, an extension, or\n"
-"                          extension and priority can be specified.\n"
-"           You cannot use any additional action post answer options in conjunction\n"
-"           with this option. Also, pbx services are not run on the peer (called) channel,\n"
-"           so you will not be able to set timeouts via the TIMEOUT() function in this macro.\n"
-"    n    - This option is a modifier for the screen/privacy mode. It specifies\n"
-"           that no introductions are to be saved in the priv-callerintros\n"
-"           directory.\n"
-"    N    - This option is a modifier for the screen/privacy mode. It specifies\n"
-"           that if callerID is present, do not screen the call.\n"
-"    o    - Specify that the CallerID that was present on the *calling* channel\n"
-"           be set as the CallerID on the *called* channel. This was the\n"
-"           behavior of Asterisk 1.0 and earlier.\n"
-"    O([x]) - \"Operator Services\" mode (DAHDI channel to DAHDI channel\n"
-"             only, if specified on non-DAHDI interface, it will be ignored).\n"
-"             When the destination answers (presumably an operator services\n"
-"             station), the originator no longer has control of their line.\n"
-"             They may hang up, but the switch will not release their line\n"
-"             until the destination party hangs up (the operator). Specified\n"
-"             without an arg, or with 1 as an arg, the originator hanging up\n"
-"             will cause the phone to ring back immediately. With a 2 specified,\n"
-"             when the \"operator\" flashes the trunk, it will ring their phone\n"
-"             back.\n"
-"    p    - This option enables screening mode. This is basically Privacy mode\n"
-"           without memory.\n"
-"    P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if\n"
-"           it is provided. The current extension is used if a database\n"
-"           family/key is not specified.\n"
-"    r    - Indicate ringing to the calling party. Pass no audio to the calling\n"
-"           party until the called channel has answered.\n"
-"    S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
-"           answered the call.\n"
-"    t    - Allow the called party to transfer the calling party by sending the\n"
-"           DTMF sequence defined in the blindxfer setting in the featuremap section\n"
-"           of features.conf.\n"
-"    T    - Allow the calling party to transfer the called party by sending the\n"
-"           DTMF sequence defined in the blindxfer setting in the featuremap section\n"
-"           of features.conf.\n"
-"    U(x[^arg]) - Execute via Gosub the routine 'x' for the *called* channel before connecting\n"
-"           to the calling channel. Arguments can be specified to the Gosub\n"
-"           using '^' as a delimiter. The Gosub routine can set the variable\n"
-"           GOSUB_RESULT to specify the following actions after the Gosub returns.\n"
-"           * ABORT        Hangup both legs of the call.\n"
-"           * CONGESTION   Behave as if line congestion was encountered.\n"
-"           * BUSY         Behave as if a busy signal was encountered.\n"
-"           * CONTINUE     Hangup the called party and allow the calling party\n"
-"                          to continue dialplan execution at the next priority.\n"
-"           * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
-"                          specified priority. Optionally, an extension, or\n"
-"                          extension and priority can be specified.\n"
-"           You cannot use any additional action post answer options in conjunction\n"
-"           with this option. Also, pbx services are not run on the peer (called) channel,\n"
-"           so you will not be able to set timeouts via the TIMEOUT() function in this routine.\n"
-"    w    - Allow the called party to enable recording of the call by sending\n"
-"           the DTMF sequence defined in the automon setting in the featuremap section\n"
-"           of features.conf.\n"
-"    W    - Allow the calling party to enable recording of the call by sending\n"
-"           the DTMF sequence defined in the automon setting in the featuremap section\n"
-"           of features.conf.\n"
-"    x    - Allow the called party to enable recording of the call by sending\n"
-"           the DTMF sequence defined in the automixmon setting in the featuremap section\n"
-"           of features.conf.\n"
-"    X    - Allow the calling party to enable recording of the call by sending\n"
-"           the DTMF sequence defined in the automixmon setting in the featuremap section\n"
-"           of features.conf.\n";
-
-/* RetryDial App by Anthony Minessale II <anthmct@yahoo.com> Jan/2005 */
+static char *app = "Dial";
 static char *rapp = "RetryDial";
-static char *rsynopsis = "Place a call, retrying on failure allowing optional exit extension.";
-static char *rdescrip =
-"  RetryDial(announce,sleep,retries,dialargs): This application will attempt to\n"
-"place a call using the normal Dial application. If no channel can be reached,\n"
-"the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
-"seconds before retrying the call. After 'retries' number of attempts, the\n"
-"calling channel will continue at the next priority in the dialplan. If the\n"
-"'retries' setting is set to 0, this application will retry endlessly.\n"
-"  While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
-"extension exists in either the context defined in ${EXITCONTEXT} or the current\n"
-"one, The call will jump to that extension immediately.\n"
-"  The 'dialargs' are specified in the same format that arguments are provided\n"
-"to the Dial application.\n";
 
 enum {
 	OPT_ANNOUNCE =          (1 << 0),
@@ -2187,8 +2395,8 @@ static int load_module(void)
 	else
 		ast_add_extension2(con, 1, "s", 1, NULL, NULL, "KeepAlive", ast_strdup(""), ast_free_ptr, "app_dial");
 
-	res = ast_register_application(app, dial_exec, synopsis, descrip);
-	res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);
+	res = ast_register_application_xml(app, dial_exec);
+	res |= ast_register_application_xml(rapp, retrydial_exec);
 
 	return res;
 }