Handle failures building translation paths more effectively.

The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity.  The failure
mode at that point was not good.  The report came in to us as an Asterisk
lock-up.  The "core show locks" shows a ton of threads locked up (but no
obvious deadlock).  Upon deeper investigation, when the system is in this
state, the CPU was maxed out.  The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.

The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner.  If we can't translate, then the
call just needs to be dropped, as it's not going to work.  These are the
changes:

1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL.  It now pays attention to that case and returns a result
reflecting failure.  With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.

2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored.  It is now reflected in the return value of the function.  This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.

3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio.  It uses
make_compatible() when it's going to do this.  However, it ignored failure from
make compatible.  So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through.  After detecting failure here, Dial() exits.

ABE-2658


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296000 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 13 insertions, 4 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index a6e0f3d79..a3a32999a 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -436,7 +436,16 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
 		/* Turn off hold music, etc */
 		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);
+		if (ast_channel_make_compatible(outgoing->chan, in) < 0) {
+			/* If these channels can not be made compatible, 
+			 * there is no point in continuing.  The bridge
+			 * will just fail if it gets that far.
+			 */
+			*to = -1;
+			strcpy(status, "CONGESTION");
+			ast_cdr_failed(in->cdr);
+			return NULL;
+		}
 	}
 	
 	
diff --git a/main/channel.c b/main/channel.c
index 43443287a..8630ded01 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -2827,8 +2827,7 @@ int ast_indicate_data(struct ast_channel *chan, int _condition,
 		if (option_debug) {
 				ast_log(LOG_DEBUG, "Driver for channel '%s' does not support indication %d, emulating it\n", chan->name, condition);
 		}
-		ast_playtones_start(chan, 0, ts->data, 1);
-		res = 0;
+		res = ast_playtones_start(chan, 0, ts->data, 1);
 		chan->visible_indication = condition;
 	}
 
@@ -3410,11 +3409,12 @@ static int set_format(struct ast_channel *chan, int fmt, int *rawformat, int *fo
 	else
 		/* writing */
 		*trans = ast_translator_build_path(*rawformat, *format);
+	res = *trans ? 0 : -1;
 	ast_channel_unlock(chan);
 	if (option_debug)
 		ast_log(LOG_DEBUG, "Set channel %s to %s format %s\n", chan->name,
 			direction ? "write" : "read", ast_getformatname(fmt));
-	return 0;
+	return res;
 }
 
 int ast_set_read_format(struct ast_channel *chan, int fmt)