Merged revisions 296002 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r296002 | russell | 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) | 52 lines
  
  Merged revisions 296001 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
    
    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
    ........
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296034 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 10 insertions, 1 deletions

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 73dbcca3e..50d784b1c 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -984,7 +984,16 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in,
 			ast_deactivate_generator(in);
 			/* If we are calling a single channel, and not providing ringback or music, */
 			/* then, 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;
+			}
 		}
 
 		if (!ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE) && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {