Commit Graph

2045 Commits

Author SHA1 Message Date
Jason Parker
98d73b0f65 Merged revisions 179396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines
  
  Merged revisions 179395 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line
    
    Remove several silly warnings in editline.  One about a broken preprocessor directive, and another about strlcpy/strlcat.

    (closes issue #14264)
    Reported by: dimas
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02 20:17:55 +00:00
Steve Murphy
f42c14308d Merged revisions 178986 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines
  
  Merged revisions 178956 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  In this case, it's just a matter of reducing the default timeouts from 2000
  to 1000 msec, as the max def feature digit timeout is no longer halved.
  
  ........
    r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
    
    This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
    
    As per bug 14515, a dev discussion arrived at a "mediated concensus" 
    of a default feature digit timeout of 1.0 sec. Some voted for 1300;
    ctooley thought 1500 for distracted phone users in phone booths; 
    kpfleming put his foot down at 1.0 sec. 
    
    Users who found the previous default max delay of 250 msec perfect,
    are welcome to override the new default. Notice that I said that
    250 msec was the default; wait a minute, you might say, the config
    file said it was 500 msec!; well, because of the bug fix for 14515,
    we found that 500 msec was actually enforcing a max of 250. The bug
    fix would restore 500 msec, but we felt even that was a bit tight
    for most users... 2000 msec was pushed earlier by mmichelson, so
    that reduces to 1000 msec after the bug fix. Enjoy!
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27 03:52:31 +00:00
Steve Murphy
547b76aa6a Merged revisions 178828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines
  
  Merged revisions 178804 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
    
    This patch prevents the feature detection timeout from being cut in half.
    
    Because the ast_channel_bridge() call will return 0 and pass
    a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
    field in hte config struct is getting decremented twice, which 
    effectively cuts the digittimeout in half. I added conditions
    to the if statement to only let DTMF_END frames to flow thru,
    which solved the problem. Also, when the frame pointer is null,
    let control flow thru-- this usually happens on timeouts. I added
    a comment to the code to explain what's going on and why.
    
    Many thanks to sodom for reporting this problem. Personnally, it always seemed
    like something was wrong with the featuredigittimeout, but I never
    could quite decide what... and was too busy to investigate.
    This bug forced the issue, and now we know.
    
    Sodom had other issues in 14515, but I couldn't reproduce them. If
    he still has problems, and wants to get them solved, he is welcome
    to reopen 14515.
    
    
    (closes issue #14515)
    Reported by: sodom
    Patches:
          14515.patch uploaded by murf (license 17)
    Tested by: murf, sodom
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26 17:29:53 +00:00
Russell Bryant
e1eba74ef4 Merged revisions 178509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines

Merged revisions 178508 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines

Update the copyright year for the main page of the doxygen documentation.

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2009-02-25 12:46:06 +00:00
Tilghman Lesher
41451ea689 Merged revisions 178381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
  
  Apparently, a void cast doesn't override warn_unused_result.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 20:53:27 +00:00
Russell Bryant
239ad71be7 Merged revisions 178374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines

Merged revisions 178373 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines

Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly.

(issue #14460)
Reported by: moliveras
Tested by: russell

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2009-02-24 20:43:16 +00:00
Tilghman Lesher
425201ca94 Merged revisions 178375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
  
  The 3 possible errors with pipe(2) are all impossible in this situation.
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2009-02-24 20:40:49 +00:00
Tilghman Lesher
31fd4d3573 Merged revisions 178342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines
  
  Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24 20:07:36 +00:00
Russell Bryant
07b9f97f48 Merged revisions 178142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines

Merged revisions 178141 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines

Fix infinite DTMF when a BEGIN is received without an END.

This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem.  The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.

In passing, I removed the dtmfsamples variable which was completed unused.  I
also removed a redundant setting of the lastrxts variable.

(closes issue #14460)
Reported by: moliveras

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-23 23:17:30 +00:00
Tilghman Lesher
d6c817f2f1 Merged revisions 177787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines
  
  Merged revisions 177786 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines
    
    Don't print the CR-NL combination when we aren't outputting to the manager.
    
    An embedded CR-NL in a CLI command screws up several AMI parsers that don't
    expect to see that combination in the middle of output.
    
    (Closes issue #14305)
    Reported by: martins
    Patch by: tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20 23:04:33 +00:00
Tilghman Lesher
36f41cee56 Merged revisions 177664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines
  
  Allow semicolons to be escaped, when passing arguments to the System command.
  (closes issue #14231)
   Reported by: jcovert
   Patches: 
         20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
         corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551)
   Tested by: jcovert
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20 17:42:16 +00:00
Steve Murphy
7ea292e195 Merged revisions 177595 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, 19 Feb 2009) | 32 lines

Merged revisions 177540 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.

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  r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
  
  This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
  
  The real culprit was the --full argument to flex
  in the Makefile! This causes a 7-bit scanner to be
  generated.
  
  I reviewed the rules and found one rule where I needed
  to specifically include 8-bit chars for a token.
  
  I tested against the text supplied by ibercom, and 
  all looks very well.
  
  This has been there a surprisingly long time!
  
  
  (closes issue #14498)
  Reported by: ibercom
  Patches:
        14498.patch uploaded by murf (license 17)
  Tested by: murf
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2009-02-20 00:16:46 +00:00
Jeff Peeler
676fb48b9b Merged revisions 177356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines
  
  Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the
  asterisk-dev mailing list. Thanks!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-19 15:57:16 +00:00
Kevin P. Fleming
b72c17951f Merged revisions 177229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177229 | kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 lines
  
  fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18 23:15:11 +00:00
David Vossel
48cccb7858 Merged revisions 177226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines
  
  Locking issue in action_bridge and bridge_exec
  
  action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock.  Instead, they should hold the lock until the masquerade is complete.  This will guarantee the channel remains and prevent any other weirdness from occurring.  In action_bridge() some more weirdness comes into play.  Both channels are needlessly locked at the same time and perform the exact same logic.  It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up.  There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it.
  
  issue# 14296
  Review: http://reviewboard.digium.com/r/167/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18 22:55:56 +00:00
Doug Bailey
010ec27cfd Merged revisions 177035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
  
  Fixed error where a check for an zero length, terminated string was needed.
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2009-02-18 17:26:59 +00:00
Doug Bailey
20dea27b4b Merged revisions 176948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176948 | dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines
  
  Need to take into account the \0 terminator of the old string to determine the amount available.
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2009-02-18 16:30:15 +00:00
Steve Murphy
babd94b38e Merged revisions 176943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r176943 | murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines


This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present.

Reason: when I re-engineered the merge_and_delete func to
reduce its lock time, I failed to notice that the 
functions it calls still also do locking as before.
This leads to deadlocks on dialplan reloads, when
there are actually living, subscribed hints registered
in the system.

While the reporter come across this problem while using
AEL, I might note that these deadlocks should also happen
if extensions.conf were used.

Here I added these routines to pbx.c:

ast_add_extension_nolock
add_pri_lockopt
ast_add_extension2_lockopt
find_context
add_hint_nolock

All of the above routines are static and restricted
to be used only within pbx.c, and more specifically
within the merge_contexts_and_delete routine.

They are pretty much the same as their counterparts
except they don't lock contexts or hints.

Most of them now do the real work of their
name-alike, with optional locking via extra arguments,
and are called by their name-alike. The goal was to
have the original functions so they would behave
exactly as before.

Both PJ and I tested these fixes, and the deadlocking
problem is no longer encountered.

(closes issue #14357)
Reported by: pj
Patches:
      14357.diff uploaded by murf (license 17)
Tested by: pj, murf


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2009-02-18 15:49:10 +00:00
Jeff Peeler
46db811169 Merged revisions 176708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines
  
  Merged revisions 176701 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines
    
    Modify bridging to properly evaluate DTMF after first warning is played
    
    The main problem is currently if the Dial flag L is used with a warning sound,
    DTMF is not evaluated after the first warning sound. To fix this, a flag has 
    been added in ast_generic_bridge for playing the warning which ensures that if
    a scheduled warning is missed, multiple warrnings are not played back (due to a
    feature evaluation or waiting for digits). ast_channel_bridge was modified to
    store the nexteventts in the ast_bridge_config structure as that information
    was lost every time ast_channel_bridge was reentered, causing a hangup due to
    incorrect time calculations.
    
    (closes issue #14315)
    Reported by: tim_ringenbach
    
    Reviewed on reviewboard:
    http://reviewboard.digium.com/r/163/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 22:14:38 +00:00
Kevin P. Fleming
f3ae0e00a7 Merged revisions 176255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines
  
  Merged revisions 176216 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines
    
    fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
  ........
    r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines
  
    correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
  ........
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2009-02-16 21:50:47 +00:00
Mark Michelson
1999cdee6c Merged revisions 176174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines
  
  Assist proper thread synchronization when stopping the logger thread.
  
  I was finding that on my dev box, occasionally attempting to "stop now" in
  trunk would cause Asterisk to hang. I traced this to the fact that the logger
  thread was waiting on a condition which had already been signalled. The logger
  thread also need to be sure to check the value of the close_logger_thread variable.
  
  The close_logger_thread variable is only checked when the list of logmessages is empty.
  This allows for the logger thread to print and free any pending messages before exiting.
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2009-02-16 18:33:05 +00:00
Tilghman Lesher
96a87efb7f Merged revisions 175334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines
  
  Merged revisions 175311 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines
    
    Fix crashes when receiving certain T.38 packets.  Also, increase the maximum
    size of T.38 packets and warn users when they try to set the limits above those
    maximums.
    (closes issue #13050)
     Reported by: schern
     Patches: 
           20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
     Tested by: schern
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 21:27:58 +00:00
Jeff Peeler
8008dca29d Fix mistake in merging conflict from 175299.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 20:59:09 +00:00
Jeff Peeler
0a72cfe440 Merged revisions 175298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines
  
  Merged revisions 175294 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines
    
    Fix ParkedCall event information for From field in the case of a blind transfer
    
    If the parker information can not be obtained from the peer, try and see if
    the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
    to the ParkAndAnnounce app would return nothing for the From.
    
    Closes AST-189
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2009-02-12 20:50:30 +00:00
Jeff Peeler
17df1c9d13 Merged revisions 175188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines
  
  Merged revisions 175187 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines
    
    Fix crash in event of failed attempt to transfer to parking
    
    The peer may not necessarily exist, such as in the case of a transfer to 
    ParkAndAnnounce. In this case don't try to play a sound to it.
  ........
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2009-02-12 18:00:49 +00:00
Russell Bryant
d79cc1e799 Merged revisions 175125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines

Merged revisions 175124 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines

Don't send DTMF for infinite time if we do not receive an END event.

I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf, 
dtmftimeout, that was intended to handle this situation.  However, in between 
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.

The default timeout is 3 seconds.  However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:

      Limiting the time period of extending the tone is necessary
      to avoid that a tone "gets stuck". Regardless of the
      algorithm used, the tone SHOULD NOT be extended by more than
      three packet interarrival times. A slight extension of tone
      durations and shortening of pauses is generally harmless.

Three seconds will pretty much _always_ be far more than three packet 
interarrival times.  However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.

Code from svn/asterisk/team/russell/issue_14460

(closes issue #14460)
Reported by: moliveras

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 17:03:21 +00:00
Mark Michelson
90ef4eb33e Merged revisions 175121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines
  
  Make lock information for ao2_trylock be more useful and gnarly
  
  Core show locks information involving an ao2_trylock did not
  show the function that called ao2_trylock, but would instead
  show ao2_trylock as the source of the lock. This is not useful
  when trying to debug locking issues.
  
  One bizarre note is that this logic is already in 1.4 but somehow
  did not get merged to trunk or the 1.6.X branches.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12 16:33:24 +00:00
Mark Michelson
a45ec0c30a Merged revisions 174945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines
  
  Fix 'd' option for app_dial and add new option to Answer application
  
  The 'd' option would not work for channel types which use RTP to transport
  DTMF digits. The only way to allow for this to work was to answer the channel
  if we saw that this option was enabled.
  
  I realized that this may cause issues with CDRs, specifically with giving false
  dispositions and answer times. I therefore modified ast_answer to take another
  parameter which would tell if the CDR should be marked answered.
  
  I also extended this to the Answer application so that the channel may be answered
  but not CDRified if desired.
  
  I also modified app_dictate and app_waitforsilence to only answer the channel if it
  is not already up, to help not allow for faulty CDR answer times.
  
  All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all
  the changes except for the change to the Answer application will go in since we do
  not introduce new features into stable branches
  
  (closes issue #14164)
  Reported by: DennisD
  Patches:
        14164.patch uploaded by putnopvut (license 60)
  Tested by: putnopvut
  
  Review: http://reviewboard.digium.com/r/145
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-11 22:48:11 +00:00
Mark Michelson
0aaff466b7 Merged revisions 174764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r174764 | mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 lines

Fix an fd leak that would occur in HTTP AMI sessions

The explanation behind this fix is a bit complicated, and I've already
typed it up in the code as a huge comment inside of manager.c, so I'll
give the abridged version here.

We needed a way to separate action-specific data from session-specific data.
Unfortunately, the only way to maintain API compatibility and to not have to
change every single manager action was to rename the current mansession structure
and wrap it inside a new mansession structure which actually contains action-
specific data.

(closes issue #14364)
Reported by: awk
Patches:
      14364_better.patch uploaded by putnopvut (license 60)
Tested by: putnopvut

Review: http://reviewboard.digium.com/r/148/


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 21:49:14 +00:00
Matthew Nicholson
bd217cb067 Merged revisions 174584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb 2009) | 25 lines
  
  Merged revisions 174583 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines
    
    Improve behavior of jitterbuffer when maxjitterbuffer is set.
    
    This change improves the way the jitterbuffer handles maxjitterbuffer and
    dramatically reduces the number of frames dropped when maxjitterbuffer is
    exceeded.  In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
    new frames were dropped until the jitterbuffer is empty.  This change modifies
    the code to only drop frames until maxjitterbuffer is no longer exceeded.
    
    Also, previously when maxjitterbuffer was exceeded, dropped frames were not
    tracked causing stats for dropped frames to be incorrect, this change also
    addresses that problem.
    
    (closes issue #14044)
    Patches:
          bug14044-1.diff uploaded by mnicholson (license 96)
    Tested by: mnicholson
    Review: http://reviewboard.digium.com/r/144/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10 18:19:45 +00:00
Jeff Peeler
58d71a9470 Merged revisions 173500 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines
  
  Merged revisions 173211 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines
    
    Parking attempts made to one end of a bridge no longer will hang up due to a
    parking failure.
    
    Parking attempts made using either one-touch, or doing either a blind or 
    assisted transfer to the parking extension now keep up the bridge instead of
    hanging up the attempted parked party. Normal causes for the parking attempt
    to fail includes the specific specified extension (via PARKINGEXTEN) not being 
    available or if all the parking spaces are currently in use. To avoid having
    to reverse a masquerade park_space_reserve was made to provide foresight if
    a parking attempt will succeed and if so reserve the parking space.
    
    (closes issue #13494)
    Reported by: mdu113
    
    Reviewed by Russell: http://reviewboard.digium.com/r/133/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 23:38:54 +00:00
Tilghman Lesher
8c1a726973 Merged revisions 173458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
  r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines
  
  When using a socket as a FILE *, the stdio functions will sometimes try to do
  an fseek() on the stream, which is an invalid operation for a socket.  Turning
  off buffering explicitly lets the stdio functions know they cannot do this,
  thus avoiding a potential error.
  (closes issue #14400)
   Reported by: fnordian
   Patches: 
         tcptls.patch uploaded by fnordian (license 110)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 18:55:32 +00:00
Mark Michelson
a71ca0b0e3 Merged revisions 173354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines

Fix a problem where file playback would cause fds to remain open forever

The problem came from the fact that a frame read from a format interpreter
was not freed. Adding a call to ast_frfree fixed this. The explanation for
why this caused the problem is a bit complex, but here goes:

There was a problem in all versions of Asterisk where the embedded frame
of a filestream structure was referenced after the filestream was freed. This
was fixed by adding reference counting to the filestream structure. The refcount
would increase every time that a filestream's frame pointer was pointing to an
actual frame of data. When the frame was freed, the refcount would decrease. Once
the refcount reached 0, the filestream was freed, and as part of the operation,
the open files were closed as well.

Thus it becomes more clear why a missing ast_frfree would cause a reference leak
and cause the files to not be closed. You may ask then if there was a frame leak
before this patch. The answer to that is actually no! The filestream code was
"smart" enough to know that since the frame we received came from a format interpreter,
the frame had no malloced data and thus didn't need to be freed. Now, however, there
is cleanup that needs to be done when we finish with the frame, so we do need to
call ast_frfree on the frame to be sure that the refcount for the filestream is
decremented appropriately.

(closes issue #14384)
Reported by: fiddur
Patches:
      14384.patch uploaded by putnopvut (license 60)
Tested by: fiddur, putnopvut


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 15:30:54 +00:00
Tilghman Lesher
3d71c38287 Merged revisions 173311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines
  
  Ensure that commas placed in the middle of extension character classes do not
  interfere with correct parsing of the extension.  Also, if an unterminated
  character class DOES make its way into the pbx core (through some other
  method), ensure that it does not crash Asterisk.
  (closes issue #14362)
   Reported by: Nick_Lewis
   Patches: 
         20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
   Tested by: Corydon76
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-04 00:45:32 +00:00
Terry Wilson
6aa5e4f230 Merged revisions 173067 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines
  
  Merged revisions 173066 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines
    
    Fix a feature inheritance bug I added after code review
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02 23:59:46 +00:00
Terry Wilson
af2b34cb56 Merged revisions 172580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines
  
  Merged revisions 172517 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
    
    Fix feature inheritance with builtin features
    
    When using builtin features like parking and transfers, the AST_FEATURE_* flags
    would not be set correctly for all instances when either performing a builtin
    attended transfer, or parking a call and getting the timeout callback.  Also,
    there was no way on a per-call basis to specify what features someone should
    have on picking up a parked call (since that doesn't involve the Dial() command).
    There was a global option for setting whether or not all users who pickup a
    parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
    AUTOMON, or PARKCALL.
    
    This patch:
    1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
    dialplan or with setvar in channels that support it.  This variable can be set
    to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
    equivalent dial options), to set what features should be activated on this
    channel.  The patch moves the setting of the features datastores into the
    bridging code instead of app_dial to help facilitate this.
    
    2) adds global options parkedcallparking, parkedcallhangup, and
    parkedcallrecording to be similar to the parkedcalltransfers option for
    globally setting features.
    
    3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
    extension since tracking everything through multiple masquerades, etc. is
    difficult and error-prone
    
    4) attempts to fix all cases of return calls from parking and completed builtin
    transfers not having the correct permissions
    (closes issue #14274)
    Reported by: aragon
    Patches: 
          fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
    Tested by: aragon, otherwiseguy
    
    Review http://reviewboard.digium.com/r/138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 23:58:31 +00:00
Tilghman Lesher
407d3d8861 Merged revisions 172441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines
  
  Merged revisions 172438 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
    
    Lose the CAP_NET_ADMIN at every fork, instead of at startup.  Otherwise, if
    Asterisk runs as a non-root user and the administrator does a 'restart now',
    Asterisk loses the ability to set QOS on packets.
    (closes issue #14004)
     Reported by: nemo
     Patches: 
           20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Corydon76
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29 23:47:00 +00:00
Steve Murphy
491c4a9c68 Merged revisions 172063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines
  
  Merged revisions 172030 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
    
    This patch fixes h-exten running misbehavior in manager-redirected 
    situations.
    
    What it does:
    1. A new Flag value is defined in include/asterisk/channel.h,
     AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
     bridge hangup exten code not to run the h-exten there (nor
     publish the bridge cdr there). It will done at the pbx-loop
     level instead.
    2. In the manager Redirect code, I set this flag on the channel
     if the channel has a non-null pbx pointer. I did the same for the
     second (chan2) channel, which gets run if name2 is set...
     and the first succeeds.
    3. I restored the ending of the cdr for the pbx loop h-exten
     running code. Don't know why it was removed in the first place.
    4. The first attempt at the fix for this bug was to place code
       directly in the async_goto routine, which was called from a
       large number of places, and could affect a large number of
       cases, so I tested that fix against a fair number of transfer
       scenarios, both with and without the patch. In the process,
       I saw that putting the fix in async_goto seemed not to affect
       any of the blind or attended scenarios, but still, I was
       was highly concerned that some other scenarios I had not tested
       might be negatively impacted, so I refined the patch to 
       its current scope, and jmls tested both. In the process, tho,
       I saw that blind xfers in one situation, when the one-touch
       blind-xfer feature is used by the peer, we got strange 
       h-exten behavior.  So, I  inserted code to swap CDRs and
       to set the HANGUP_DONT field, to get uniform behavior.
    5. I added code to the bridge to obey the HANGUP_DONT flag,
       skipping both publishing the bridge CDR, and running
       the h-exten; they will be done at the pbx-loop (higher)
       level instead.
    6. I removed all the debug logs from the patch before committing.
    7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
       so it's only done if the h-exten is going to be run. A very
       minor performance improvement, but technically correct.
    
    
    (closes issue #14241)
    Reported by: jmls
    Patches:
          14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
    Tested by: murf, jmls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28 20:41:45 +00:00
Mark Michelson
0a9595289a Merged revisions 171622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan 2009) | 26 lines

Merged revisions 171621 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines

Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory

slinfactory used the "samples" field of an ast_frame in order to determine
the amount of data contained within the frame. In certain cases, such as
jitter buffer interpolated frames, the frame would have a non-zero value for
"samples" but have NULL "data"

This caused a problem when a memcpy call in ast_slinfactory_read would attempt
to access invalid memory. The solution in use here is to never feed frames into
the slinfactory if they have NULL "data"

(closes issue #13116)
Reported by: aragon
Patches:
      13116.diff uploaded by putnopvut (license 60)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 20:13:17 +00:00
Matthew Fredrickson
d970de93a6 Revert some changes that shouldn't have made it in
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 16:15:40 +00:00
Matthew Fredrickson
38c977d6d0 Make sure we do not go into alarm on PTMP links with non persistent D-channels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27 16:14:08 +00:00
Joshua Colp
d59c77dc20 Merged revisions 170652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines
  
  Merged revisions 170648 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines
    
    When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them.
    (closes issue #14249)
    Reported by: RadicAlish
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 20:19:37 +00:00
Mark Michelson
d57e63c0f1 Merged revisions 170393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan 2009) | 36 lines

Merged revisions 170392 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines

Fix broken call pickup

There was a subtle change in ast_do_masquerade which
resulted in failed attempts to pickup calls. The problem
was that the value of the AST_FLAG_OUTGOING flag was
copied from the clone to the original channel. In the case
of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting
to execute the pickup.

Because this flag was not set, when ast_read came across
an answer frame, it ignored it. The result of this was that
the calling channel was never properly answered.

This fix changes the behavior in ast_do_masquerade to set
the flags on the original channel to the union of the flags
on the clone channel. This way, if the AST_FLAG_OUTGOING
flag is set on either of the two channels involved in the
masquerade, the resulting channel will have the flag set
as well.

(closes issue #14206)
Reported by: francesco_r
Patches:
      14206.patch uploaded by putnopvut (license 60)
Tested by: francesco_r, aragon, putnopvut


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 15:49:50 +00:00
Joshua Colp
ebdd169c19 Merged revisions 170240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) | 14 lines
  
  Merged revisions 170239 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines
    
    Don't crash if RTCP is not enabled on an RTP structure but statistics are output.
    (closes issue #14234)
    Reported by: jcovert
    Patches:
          rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
          rtp.c.patch-svn-165599 uploaded by jcovert (license 551)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 20:05:24 +00:00
Joshua Colp
1f843fc79d Merged revisions 170051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) | 13 lines
  
  Merged revisions 170050 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines
    
    Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness.
    (closes issue #14011)
    Reported by: dveiga
    Patches:
          pbx.c.patch uploaded by dveiga (license 665)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 15:15:35 +00:00
Joshua Colp
dbf61acd1f Merged revisions 169869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) | 11 lines
  
  Merged revisions 169867 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 lines
    
    Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed.
    (closes issue #13839)
    Reported by: mcallist
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 23:27:19 +00:00
Mark Michelson
06b1a253d6 Merged revisions 169794 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r169794 | mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17 lines

Fix a crash when saying certain numbers in Chinese

This commit fixes a crash that was occurring when attempting to
say a number between 10000 and 100000 due to dividing by 0.

This also removes some places where a "zero" is spoken when it
should not be.


(closes issue #14291)
Reported by: dant
Patches:
      say.c-14291.diff uploaded by dant (license 670)
Tested by: dant



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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 22:11:11 +00:00
Tilghman Lesher
69e39ce8e9 Merged revisions 169723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009) | 15 lines
  
  Merged revisions 169722 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines
    
    Extra NULLs in the output cause some terminal types to abort in the middle of
    a color code, causing terminal weirdness.
    (closes issue #14130)
     Reported by: coolmig
     Patches: 
           20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
     Tested by: Corydon76, coolmig
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 21:04:24 +00:00
Terry Wilson
cc34381691 Merged revisions 169510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r169510 | twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines
  
  Make a proper builtin attended transfer to parking work
  
  This is an ugly hack from 1.4 that allows the timeout callback from a parked
  call to use the right channel name for the callback when the park is done with
  a builtin attended transfer (that isn't completed early).  This hasn't ever
  worked in trunk and no one has complained yet, so eh.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 19:29:24 +00:00
Terry Wilson
5c9043b90b Merged revisions 169486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009) | 13 lines
  
  Merged revisions 169485 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) | 6 lines
    
    Don't play audio to the channel if we've masqueraded
    
    (closes issue #14066)
    Reported by: bluefox
    Tested by: otherwiseguy, bluefox
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 18:58:34 +00:00