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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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- <value name="CONGESTION" />
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- <value name="NOANSWER" />
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- <value name="BUSY" />
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- <value name="ANSWER" />
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- <value name="CANCEL" />
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+ <value name="CHANUNAVAIL">
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+ Either the dialed peer exists but is not currently reachable, e.g.
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+ endpoint is not registered, or an attempt was made to call a
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+ nonexistent location, e.g. nonexistent DNS hostname.
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+ </value>
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+ <value name="CONGESTION">
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+ Channel or switching congestion occured when routing the call.
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+ This can occur if there is a slow or no response from the remote end.
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+ </value>
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+ <value name="NOANSWER">
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+ Called party did not answer.
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+ </value>
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+ <value name="BUSY">
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+ The called party was busy or indicated a busy status.
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+ Note that some SIP devices will respond with 486 Busy if their Do Not Disturb
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+ modes are active. In this case, you can use DEVICE_STATUS to check if the
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+ endpoint is actually in use, if needed.
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+ </value>
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+ <value name="ANSWER">
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+ The call was answered.
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+ Any other result implicitly indicates the call was not answered.
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+ </value>
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+ <value name="CANCEL">
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+ Dial was cancelled before call was answered or reached some other terminating event.
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+ </value>
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<value name="DONTCALL">
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For the Privacy and Screening Modes.
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Will be set if the called party chooses to send the calling party to the 'Go Away' script.
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For the Privacy and Screening Modes.
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Will be set if the called party chooses to send the calling party to the 'torture' script.
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</value>
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- <value name="INVALIDARGS" />
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+ <value name="INVALIDARGS">
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+ Dial failed due to invalid syntax.
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+ </value>
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</variable>
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</variablelist>
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</description>
@@ -3602,4 +3623,4 @@ AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Dialing Application",
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.load = load_module ,
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.unload = unload_module ,
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.requires = "ccss" ,
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- );
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+ );
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