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It was possible for no timer to become available between creating the bridge
and starting it. We now open a timer when creating it and keep it open until the
bridge is destroyed.


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  r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines
  
  Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.
  
  If calls were placed using an IP address or hostname the global nat setting was copied over
  but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP
  actions.
  
  (closes issue #14546)
  Reported by: acunningham
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When dvossel and I were doing some load testing last week, we noticed that we
could make Asterisk trunk lock up instantly when we started generating a bunch
of calls.  The backtraces of locked threads were bizarre, and many were stuck
on an _unlock_ of an rwlock.

The changes are:

1) Fix a number of places where a backtrace would be loaded into an invalid
   index of the backtrace array.  It's an off by one error, which ends up
   writing over the rwlock itself.

2) Ensure that in the array of held locks, we NULL out an index once it is
   not being used so that it's not confusing when analyzing its contents.

3) Remove a bunch of logging referring to an rwlock operating being done
   with "deep reentrancy".  It is normal for _many_ threads to hold a
   read lock on an rwlock.


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines
  
  use new, improved 8kHz prompts
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  r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines
  
  pri loop TestClient/TestServer fails: server SEND DTMF 8
  
  app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent.  During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up.
  
  (closes issue #12442)
  Reported by: tzafrir
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This code comes from svn/asterisk/team/russell/event_performance/.

Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.

1) Asterisk 1.6.1 introduces some additional logic to be able to handle
   distributed device state.  This functionality comes at a cost.
   One relatively minor change in this patch is that the extra processing
   required for distributed device state is now completely bypassed if
   it's not needed.

2) One of the things that I noticed when profiling this code was that a
   _lot_ of time was spent doing string comparisons.  I changed the way
   strings are represented in an event to include a hash value at the front.
   So, before doing a string comparison, we do an integer comparison on the
   hash.

3) Finally, the code that handles the event cache has been re-written.
   I tried to do this in a such a way that it had minimal impact on the API.
   I did have to change one API call, though - ast_event_queue_and_cache().
   However, the way it works now is nicer, IMO.  Each type of event that
   can be cached (MWI, device state) has its own hash table and rules for
   hashing and comparing objects.  This by far made the biggest impact on
   performance.

For additional details regarding this code and how it was tested, please see the
review request.

(closes issue #14738)
Reported by: russell

Review: http://reviewboard.digium.com/r/205/


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If we receive a T38 request negotiate control frame we should only attempt to do so
if the option is enabled on the dialog.


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines
  
  Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.
  
  When moving the cursor backward and pressing TAB to autocomplete, a NULL is put
  in the line and we are loosing what we have already wrote after the actual
  cursor position.
  
  (closes issue #14373)
  Reported by: eliel
  Patches:
        asterisk.c.patch uploaded by eliel (license 64)
        Tested by: lmadsen
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(closes issue #14714)
Reported by: ygor


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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines
  
  Change NULL pointer check to be ast_strlen_zero.
  
  The 'digit' variable is guaranteed to be non-NULL, so the if
  statement could never evaluate true. Changing to ast_strlen_zero
  makes the logic correct.
  
  This was found while reviewing ast_channel_ao2 code review.
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The default codec configuration for chan_iax2 is bandwidth=low.  I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added an option to respond to a SIP invite with only the single most preferred joint codec.  This limits the options of what codecs the other side can use.

(closes issue #12485)
Reported by: bamby
Review: http://reviewboard.digium.com/r/206/




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  r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines
  
  Additionally note that the operator option needs an 'o' extension.
  (Related to issue #14731)
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https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines
  
  Fix a memory leak in res_monitor.c
  
  The only way that this leak would occur is if Monitor were started
  using the Manager interface and no File: header were given. Discovered
  while reviewing the ast_channel_ao2 review request.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #14655)
Reported by: ulogic
Patches:
      chan_dahdi.patch uploaded by ulogic (license 728)
Tested by: lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We only want to move the channel pointers that are actually present.


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r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines

Fix a crash in IAX2 registration handling found during load testing with dvossel.

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Hey! chan_sip.so loads now!



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  r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
  
  Cleaning up a few things in detect disconnect patch
  
  Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory.  Cleaned up /param tags in features.h.  No longer send dynamic features in ast_feature_detect. 
  
  issue #11583
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  r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009) | 2 lines
  
  Reordering, to change prior to unlocking
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  r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines
  
  Delay signalling progress until a PRI channel really signals progress.
  (closes issue #13034)
   Reported by: klaus3000
   Patches: 
         20090316__bug13034.diff.txt uploaded by tilghman (license 14)
         patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65)
   Tested by: klaus3000
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  r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar 2009) | 1 line
  
  Export some more required symbols.
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For every attempt that app_queue made to place an outbound call to a queue member,
we would allocate a queue_end_bridge structure. When the bridge for the call had
completed, we would free the structure. Unfortunately not all call attempts actually
end up bridged to a member, so we need to be more selective of when to allocate
the structure. With this change, the allocation occurs in an area where we can
guarantee that the call will be bridged.

(closes issue #14680)
Reported by: caspy
Patches:
      14680.patch uploaded by mmichelson (license 60)
Tested by: caspy



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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines

Remove the use of RTLD_NOLOAD, as it is not behaving like expected.

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r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19 Mar 2009) | 1 line

Allow the AES API to work.
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  r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
  
  Allow disconnect feature before a call is bridged
  
  feature.conf has a disconnect option.  By default this option is set to '*', but it could be anything.  If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else.  This is because features are unavailable until bridging takes place.  The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different.  This patch allows features to be detected from outside of the bridge, but not operated on.  In this case, the disconnect feature can be detected before briding and handled outside of features.c.
  
  (closes issue #11583)
  Reported by: sobomax
  Patches:
  	patch-apps__app_dial.c uploaded by sobomax (license 359)
  	11583.latest-patch uploaded by murf (license 17)
  	detect_disconnect.diff uploaded by dvossel (license 671)
  Tested by: sobomax, dvossel
  Review: http://reviewboard.digium.com/r/195/
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