[Asterisk-cvs] asterisk/apps app_dial.c,1.81,1.82

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/apps
In directory mongoose.digium.com:/tmp/cvs-serv30189/apps

Modified Files:
	app_dial.c 
Log Message:
Cleanups to the ordering of events in dial, don't freak out on the wrong codec


Index: app_dial.c
===================================================================
RCS file: /usr/cvsroot/asterisk/apps/app_dial.c,v
retrieving revision 1.81
retrieving revision 1.82
diff -u -d -r1.81 -r1.82
--- app_dial.c	25 Jun 2004 03:59:07 -0000	1.81
+++ app_dial.c	7 Jul 2004 16:02:13 -0000	1.82
@@ -142,7 +142,7 @@
 	
 	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 @@
 			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 @@
 				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;