Cleanups to the ordering of events in dial, don't freak out on the wrong codec

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3385 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 10 insertions, 8 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 4b05ad478..dfb1a3e8e 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -142,7 +142,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
 	
 	if (single) {
 		/* Turn off hold music, etc */
-		ast_indicate(in, -1);
+		ast_deactivate_generator(in);
 		/* If we are calling a single channel, make them compatible for in-band tone purpose */
 		ast_channel_make_compatible(outgoing->chan, in);
 	}
@@ -853,13 +853,6 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
 		if (numsubst)
 			pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", numsubst);
-		/* Make sure channels are compatible */
-		res = ast_channel_make_compatible(chan, peer);
-		if (res < 0) {
-			ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
-			ast_hangup(peer);
-			return -1;
-		}
  		/* JDG: sendurl */
  		if( url && !ast_strlen_zero(url) && ast_channel_supports_html(peer) ) {
  			ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
@@ -913,6 +906,15 @@ static int dial_exec(struct ast_channel *chan, void *data)
 				sentringing = 0;
 				ast_indicate(chan, -1);
 			}
+			/* Be sure no generators are left on it */
+			ast_deactivate_generator(chan);
+			/* Make sure channels are compatible */
+			res = ast_channel_make_compatible(chan, peer);
+			if (res < 0) {
+				ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
+				ast_hangup(peer);
+				return -1;
+			}
 			res = ast_bridge_call(chan,peer,&config);
 		} else 
 			res = -1;