[asterisk-commits] russell: branch 1.6.2 r296001 - in /branches/1.6.2: ./ apps/ main/

[SVN commits to the Asterisk project]

Author: russell
Date: Wed Nov 24 11:03:16 2010
New Revision: 296001

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=296001
Log:
Merged revisions 296000 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
  
  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
........

Modified:
    branches/1.6.2/   (props changed)
    branches/1.6.2/apps/app_dial.c
    branches/1.6.2/main/channel.c

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: branches/1.6.2/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/apps/app_dial.c?view=diff&rev=296001&r1=296000&r2=296001
==============================================================================
--- branches/1.6.2/apps/app_dial.c (original)
+++ branches/1.6.2/apps/app_dial.c Wed Nov 24 11:03:16 2010
@@ -839,7 +839,16 @@
 		/* 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(pa->status, "CONGESTION");
+			ast_cdr_failed(in->cdr);
+			return NULL;
+		}
 	}
 
 #ifdef HAVE_EPOLL

Modified: branches/1.6.2/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/main/channel.c?view=diff&rev=296001&r1=296000&r2=296001
==============================================================================
--- branches/1.6.2/main/channel.c (original)
+++ branches/1.6.2/main/channel.c Wed Nov 24 11:03:16 2010
@@ -3391,9 +3391,8 @@
 	if (ts) {
 		/* We have a tone to play, yay. */
 		ast_debug(1, "Driver for channel '%s' does not support indication %d, emulating it\n", chan->name, condition);
-		ast_playtones_start(chan, 0, ts->data, 1);
+		res = ast_playtones_start(chan, 0, ts->data, 1);
 		ts = ast_tone_zone_sound_unref(ts);
-		res = 0;
 		chan->visible_indication = condition;
 	}
 
@@ -3982,10 +3981,11 @@
 	else
 		/* writing */
 		*trans = ast_translator_build_path(*rawformat, *format);
+	res = *trans ? 0 : -1;
 	ast_channel_unlock(chan);
 	ast_debug(1, "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)