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@ -103,15 +103,16 @@ ASTERISK_REGISTER_FILE()
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<option name="a">
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<para>Immediately answer the calling channel when the called channel answers in
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all cases. Normally, the calling channel is answered when the called channel
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answers, but when options such as A() and M() are used, the calling channel is
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answers, but when options such as <literal>A()</literal> and
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<literal>M()</literal> are used, the calling channel is
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not answered until all actions on the called channel (such as playing an
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announcement) are completed. This option can be used to answer the calling
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channel before doing anything on the called channel. You will rarely need to use
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this option, the default behavior is adequate in most cases.</para>
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</option>
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<option name="b" argsep="^">
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<para>Before initiating an outgoing call, Gosub to the specified
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location using the newly created channel. The Gosub will be
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<para>Before initiating an outgoing call, <literal>Gosub</literal> to the specified
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location using the newly created channel. The <literal>Gosub</literal> will be
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executed for each destination channel.</para>
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<argument name="context" required="false" />
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<argument name="exten" required="false" />
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@ -121,8 +122,8 @@ ASTERISK_REGISTER_FILE()
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</argument>
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</option>
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<option name="B" argsep="^">
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<para>Before initiating the outgoing call(s), Gosub to the specified
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location using the current channel.</para>
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<para>Before initiating the outgoing call(s), <literal>Gosub</literal> to the
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specified location using the current channel.</para>
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<argument name="context" required="false" />
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<argument name="exten" required="false" />
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<argument name="priority" required="true" hasparams="optional" argsep="^">
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@ -134,7 +135,8 @@ ASTERISK_REGISTER_FILE()
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<para>Reset the call detail record (CDR) for this call.</para>
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</option>
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<option name="c">
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<para>If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere'</para>
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<para>If the Dial() application cancels this call, always set
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<variable>HANGUPCAUSE</variable> to 'answered elsewhere'</para>
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</option>
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<option name="d">
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<para>Allow the calling user to dial a 1 digit extension while waiting for
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@ -156,8 +158,8 @@ ASTERISK_REGISTER_FILE()
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<replaceable>called</replaceable> DTMF string is sent to the called party, and the
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<replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
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can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
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to the called party immediately after receiving a PROGRESS message.</para>
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<para>See SendDTMF for valid digits.</para>
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to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
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<para>See <literal>SendDTMF</literal> for valid digits.</para>
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</option>
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<option name="e">
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<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
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@ -165,7 +167,7 @@ ASTERISK_REGISTER_FILE()
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<option name="f">
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<argument name="x" required="false" />
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<para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
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deflection to the dialplan extension of this Dial() using a dialplan <literal>hint</literal>.
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deflection to the dialplan extension of this <literal>Dial()</literal> using a dialplan <literal>hint</literal>.
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For example, some PSTNs do not allow CallerID to be set to anything
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other than the numbers assigned to you.
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If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
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@ -318,11 +320,11 @@ ASTERISK_REGISTER_FILE()
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<note>
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<para>You cannot use any additional action post answer options in conjunction
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with this option. Also, pbx services are run on the peer (called) channel,
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so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
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so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this macro.</para>
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</note>
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<warning><para>Be aware of the limitations that macros have, specifically with regards to use of
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the <literal>WaitExten</literal> application. For more information, see the documentation for
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Macro()</para></warning>
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<literal>Macro()</literal>.</para></warning>
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</option>
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<option name="n">
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<argument name="delete">
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@ -339,7 +341,7 @@ ASTERISK_REGISTER_FILE()
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</option>
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<option name="N">
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<para>This option is a modifier for the call screening/privacy mode. It specifies
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that if Caller*ID is present, do not screen the call.</para>
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that if CallerID is present, do not screen the call.</para>
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</option>
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<option name="o">
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<argument name="x" required="false" />
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@ -347,7 +349,7 @@ ASTERISK_REGISTER_FILE()
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<emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
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This was the behavior of Asterisk 1.0 and earlier.
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If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
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Note that o(${CALLERID(all)}) is similar to option o without the parameter.</para>
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Note that <literal>o(${CALLERID(all)})</literal> is similar to option <literal>o</literal> without the parameter.</para>
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</option>
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<option name="O">
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<argument name="mode">
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@ -377,13 +379,13 @@ ASTERISK_REGISTER_FILE()
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<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
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party until the called channel has answered.</para>
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<argument name="tone" required="false">
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<para>Indicate progress to calling party. Send audio 'tone' from the indications.conf tonezone currently in use.</para>
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<para>Indicate progress to calling party. Send audio 'tone' from the <filename>indications.conf</filename> tonezone currently in use.</para>
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</argument>
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</option>
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<option name="R">
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<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing.
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<option name="R">
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<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing.
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Allow interruption of the ringback if early media is received on the channel.</para>
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</option>
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</option>
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<option name="S">
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<argument name="x" required="true" />
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<para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
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@ -391,8 +393,8 @@ ASTERISK_REGISTER_FILE()
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</option>
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<option name="s">
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<argument name="x" required="true" />
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<para>Force the outgoing callerid tag parameter to be set to the string <replaceable>x</replaceable>.</para>
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<para>Works with the f option.</para>
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<para>Force the outgoing CallerID tag parameter to be set to the string <replaceable>x</replaceable>.</para>
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<para>Works with the <literal>f</literal> option.</para>
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</option>
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<option name="t">
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<para>Allow the called party to transfer the calling party by sending the
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@ -406,15 +408,15 @@ ASTERISK_REGISTER_FILE()
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</option>
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<option name="U" argsep="^">
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<argument name="x" required="true">
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<para>Name of the subroutine to execute via Gosub</para>
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<para>Name of the subroutine to execute via <literal>Gosub</literal></para>
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</argument>
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<argument name="arg" multiple="true" required="false">
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<para>Arguments for the Gosub routine</para>
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<para>Arguments for the <literal>Gosub</literal> routine</para>
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</argument>
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<para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
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to the calling channel. Arguments can be specified to the Gosub
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using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
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<variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
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<para>Execute via <literal>Gosub</literal> the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
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to the calling channel. Arguments can be specified to the <literal>Gosub</literal>
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using <literal>^</literal> as a delimiter. The <literal>Gosub</literal> routine can set the variable
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<variable>GOSUB_RESULT</variable> to specify the following actions after the <literal>Gosub</literal> returns.</para>
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<variablelist>
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<variable name="GOSUB_RESULT">
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<value name="ABORT">
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@ -438,7 +440,7 @@ ASTERISK_REGISTER_FILE()
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<note>
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<para>You cannot use any additional action post answer options in conjunction
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with this option. Also, pbx services are run on the peer (called) channel,
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so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
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so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
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</note>
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</option>
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<option name="u">
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@ -455,7 +457,7 @@ ASTERISK_REGISTER_FILE()
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<literal>prohib</literal>
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<literal>unavailable</literal></para>
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</argument>
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<para>Works with the f option.</para>
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<para>Works with the <literal>f</literal> option.</para>
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</option>
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<option name="w">
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<para>Allow the called party to enable recording of the call by sending
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@ -497,11 +499,59 @@ ASTERISK_REGISTER_FILE()
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hangs up, or if the call is bridged and either of the parties in the bridge
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ends the call.</para>
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<para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
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application will be put into that group (as in Set(GROUP()=...).
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application will be put into that group (as in <literal>Set(GROUP()=...</literal>).
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If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
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application will be put into that group (as in Set(GROUP()=...). Unlike <variable>OUTBOUND_GROUP</variable>,
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application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
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however, the variable will be unset after use.</para>
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<example title="Dial with 30 second timeout">
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same => n,Dial(PJSIP/alice,30)
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</example>
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<example title="Parallel dial with 45 second timeout">
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same => n,Dial(PJSIP/alice&PJIP/bob,45)
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</example>
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<example title="Dial with 'g' continuation option">
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same => n,Dial(PJSIP/alice,,g)
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same => n,Log(NOTICE, Alice call result: ${DIALSTATUS})
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</example>
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<example title="Dial with transfer/recording features for calling party">
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same => n,Dial(PJSIP/alice,,TX)
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</example>
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<example title="Dial with call length limit">
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same => n,Dial(PJSIP/alice,,L(60000:30000:10000))
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</example>
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<example title="Dial with pre-dial subroutines">
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[default]
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exten => callee_channel,1,NoOp()
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same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
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same => n,Return()
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exten => called_channel,1,NoOp()
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same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
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same => n,Return()
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exten => _X.,1,NoOp()
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same => n,Dial(PJSIP/alice,,b(default^called_channel^1)B(default^callee_channel^1))
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same => n,Hangup()
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</example>
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<example title="Dial with post-answer subroutine executed on outbound channel">
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[default]
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exten => called_channel,1,NoOp()
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same => n,Playback(hello)
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same => n,Return()
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exten => _X.,1,NoOp()
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same => n,Dial(PJSIP/alice,,U(default^called_channel^1))
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same => n,Hangup()
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</example>
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<example title="Dial into ConfBridge using 'G' option">
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same => n,Dial(PJSIP/alice,,G(jump_to_here))
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same => n(jump_to_here),Goto(confbridge)
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same => n,Goto(confbridge)
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same => n(confbridge),ConfBridge(${EXTEN})
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</example>
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<para>This application sets the following channel variables:</para>
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<variablelist>
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<variable name="DIALEDTIME">
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@ -510,6 +560,15 @@ ASTERISK_REGISTER_FILE()
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<variable name="ANSWEREDTIME">
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<para>This is the amount of time for actual call.</para>
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</variable>
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<variable name="DIALEDPEERNAME">
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<para>The name of the outbound channel that answered the call.</para>
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</variable>
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<variable name="DIALEDPEERNUMBER">
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<para>The number that was dialed for the answered outbound channel.</para>
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</variable>
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<variable name="FORWARDERNAME">
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<para>If a call forward occurred, the name of the forwarded channel.</para>
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</variable>
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<variable name="DIALSTATUS">
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<para>This is the status of the call</para>
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<value name="CHANUNAVAIL" />
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@ -530,6 +589,12 @@ ASTERISK_REGISTER_FILE()
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</variable>
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</variablelist>
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</description>
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<see-also>
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<ref type="application">RetryDial</ref>
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<ref type="application">SendDTMF</ref>
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<ref type="application">Gosub</ref>
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<ref type="application">Macro</ref>
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</see-also>
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</application>
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<application name="RetryDial" language="en_US">
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<synopsis>
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@ -562,6 +627,9 @@ ASTERISK_REGISTER_FILE()
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The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
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to the Dial application.</para>
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</description>
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<see-also>
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<ref type="application">Dial</ref>
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</see-also>
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</application>
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***/
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