app_dial: Document DIALSTATUS return values.

Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.

ASTERISK-25716

Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 7d7a4594f94b31e7d25d6ccddc7b421cec5a613c..ee77551143ba7dae70cd9f8e5e0b3a667ba3b007 100644 (file)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -613,12 +613,31 @@
                                </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="CHANUNAVAIL">
+                                               Either the dialed peer exists but is not currently reachable, e.g.
+                                               endpoint is not registered, or an attempt was made to call a
+                                               nonexistent location, e.g. nonexistent DNS hostname.
+                                       </value>
+                                       <value name="CONGESTION">
+                                               Channel or switching congestion occured when routing the call.
+                                               This can occur if there is a slow or no response from the remote end.
+                                       </value>
+                                       <value name="NOANSWER">
+                                               Called party did not answer.
+                                       </value>
+                                       <value name="BUSY">
+                                               The called party was busy or indicated a busy status.
+                                               Note that some SIP devices will respond with 486 Busy if their Do Not Disturb
+                                               modes are active. In this case, you can use DEVICE_STATUS to check if the
+                                               endpoint is actually in use, if needed.
+                                       </value>
+                                       <value name="ANSWER">
+                                               The call was answered.
+                                               Any other result implicitly indicates the call was not answered.
+                                       </value>
+                                       <value name="CANCEL">
+                                               Dial was cancelled before call was answered or reached some other terminating event.
+                                       </value>
                                        <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.
@@ -627,7 +646,9 @@
                                                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" />
+                                       <value name="INVALIDARGS">
+                                               Dial failed due to invalid syntax.
+                                       </value>
                                </variable>
                        </variablelist>
                </description>
@@ -3569,4 +3590,4 @@ AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Dialing Application",
        .load = load_module,
        .unload = unload_module,
        .requires = "ccss",
-);
\ No newline at end of file
+);