Merged revisions 336659 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/10

................
  r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines
  
  Merged revisions 336658 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
    
    Made Dial d and H options no longer immediately auto-answer the calling leg.
    
    The Dial d and H options break DTMF attended transfer atxferdropcall
    option.
    
    1) Party A calls party B.
    2) Party B does a DTMF attended transfer to Party C.
    
    If the dialplan uses the Dial d or H options to call Party C then the Dial
    application answers the call immediately before initiating the call leg to
    Party C.  The premature answer causes the transfer code to not invoke the
    atxferdropcall=no behavior for a blonde transfer since Party C has
    "answered".  The transfer code thinks that Party B has "consulted" with
    Party C when Party B hangs up and completes the transfer to Party A.
    Party A now hears ringback until Party C actually answers.
    
    ASTERISK-13294 Dial d option.
    ASTERISK-11067 Dial H option to disconnect before answer.
    
    The referenced issues made Dial answer with the d and H options because
    many SIP and ISDN phones cannot send DTMF before the call is connected.
    
    * Made require the dialplan to control when or if the call needs to be
    answered to use the Dial application d and H options.  (The call is no
    longer surprise answered when using the Dial d or H options.)
    
    Review: https://reviewboard.asterisk.org/r/1381/
    
    JIRA AST-623
    JIRA AST-666
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336662 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 15 insertions, 6 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 3867ebc76..de8deb682 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -120,6 +120,11 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 					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>
+					<note>
+						<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
+						connected.  If you wish to use this option with these phones, you
+						can use the <literal>Answer</literal> application before dialing.</para>
+					</note>
 				</option>
 				<option name="D" argsep=":">
 					<argument name="called" />
@@ -170,10 +175,18 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 					</note>
 				</option>
 				<option name="h">
-					<para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
+					<para>Allow the called party to hang up by sending the DTMF sequence
+					defined for disconnect in <filename>features.conf</filename>.</para>
 				</option>
 				<option name="H">
-					<para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
+					<para>Allow the calling party to hang up by sending the DTMF sequence
+					defined for disconnect in <filename>features.conf</filename>.</para>
+					<note>
+						<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
+						connected.  If you wish to allow DTMF disconnect before the dialed
+						party answers with these phones, you can use the <literal>Answer</literal>
+						application before dialing.</para>
+					</note>
 				</option>
 				<option name="i">
 					<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
@@ -2070,10 +2083,6 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		res = -1; /* reset default */
 	}
 
-	if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
-		__ast_answer(chan, 0, 0);
-	}
-
 	if (continue_exec)
 		*continue_exec = 0;