From 974652e0e725baad2db2bf9e5c0d00d195f38440 Mon Sep 17 00:00:00 2001
From: markster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>
Date: Wed, 7 Jul 2004 16:02:13 +0000
Subject: Cleanups to the ordering of events in dial, don't freak out on the
 wrong codec

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3385 f38db490-d61c-443f-a65b-d21fe96a405b
---
 apps/app_dial.c     | 18 ++++++++++--------
 channels/chan_sip.c |  4 ++--
 2 files changed, 12 insertions(+), 10 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;
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2cd5b6f5e..b3dff70c1 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1799,7 +1799,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
 		if (!(frame->subclass & ast->nativeformats)) {
 			ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n",
 				frame->subclass, ast->nativeformats, ast->readformat, ast->writeformat);
-			return -1;
+			return 0;
 		}
 		if (p) {
 			ast_mutex_lock(&p->lock);
@@ -7864,7 +7864,7 @@ static struct sip_peer *build_peer(char *name, struct ast_variable *v)
 				peer->promiscredir = ast_true(v->value);
 			else if (!strcasecmp(v->name, "fromuser"))
 				strncpy(peer->fromuser, v->value, sizeof(peer->fromuser)-1);
-                       else if (!strcasecmp(v->name, "dtmfmode")) {
+            else if (!strcasecmp(v->name, "dtmfmode")) {
 				if (!strcasecmp(v->value, "inband"))
 					peer->dtmfmode=SIP_DTMF_INBAND;
 				else if (!strcasecmp(v->value, "rfc2833"))
-- 
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