Commit Graph

21552 Commits

Author SHA1 Message Date
David Vossel
e128ee2567 Merged revisions 330586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330586 | dvossel | 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) | 15 lines
  
  Merged revisions 330581 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines
    
    Fixes crash in chan_iax2.
    
    Fixes crash in chan_iax2 resulting from an edge case in the
    way control frames are queued during calltoken negotiation is complete.
    
    (closes issue ASTERISK-17610)
    Reported by: mgrobecker
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2011-08-02 16:19:32 +00:00
David Vossel
6f112cce0d Merged revisions 330579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330579 | dvossel | 2011-08-02 11:08:57 -0500 (Tue, 02 Aug 2011) | 9 lines
  
  Merged revisions 330578 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) | 2 lines
    
    Optimization to buffer initialization fix.
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2011-08-02 16:09:50 +00:00
David Vossel
d50e68c827 Merged revisions 330576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330576 | dvossel | 2011-08-02 10:55:36 -0500 (Tue, 02 Aug 2011) | 12 lines
  
  Merged revisions 330575 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) | 5 lines
    
    Fixes uninitialized string buffer in log message.
    
    (closes issue ASTERISK-17200)
    Reported by: lmadsen
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2011-08-02 16:04:34 +00:00
Jonathan Rose
4e1b07f243 Blocked revisions 330514 via svnmerge
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  r330514 | jrose | 2011-08-01 16:21:23 -0500 (Mon, 01 Aug 2011) | 14 lines
  
  Blocked revisions 330510 via svnmerge
  
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    r330510 | jrose | 2011-08-01 16:20:40 -0500 (Mon, 01 Aug 2011) | 8 lines
    
    Blocked revisions 330505 via svnmerge
    
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      r330505 | jrose | 2011-08-01 16:19:47 -0500 (Mon, 01 Aug 2011) | 3 lines
      
      fixes reference leak pointed out by rmudgett in https://reviewboard.asterisk.org/r/1337/
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2011-08-01 21:22:08 +00:00
Jonathan Rose
5e1a744a9d Blocked revisions 330492 via svnmerge
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  r330492 | jrose | 2011-08-01 16:10:35 -0500 (Mon, 01 Aug 2011) | 24 lines
  
  Blocked revisions 330491 via svnmerge
  
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    r330491 | jrose | 2011-08-01 16:09:15 -0500 (Mon, 01 Aug 2011) | 18 lines
    
    Blocked revisions 330490 via svnmerge
    
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      r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines
      
      Asterisk 18103 - Fix reload crash caused by destroying default parking lot
      
      Default parking lot was being destroyed in reload and was not being rebuilt properly.
      This patch keeps features.c reload from destroying the default parking lot in 1.6.2.
      Bug was caused by a hasty backport which didn't test reload enough times to catch the
      problem.
      
      (Closes Issue ASTERISK-18103)
      Reported by: 808blogger
      
      Review: https://reviewboard.asterisk.org/r/1337/
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2011-08-01 21:11:24 +00:00
Kinsey Moore
dc8df80e56 Merged revisions 330434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330434 | kmoore | 2011-08-01 10:23:29 -0500 (Mon, 01 Aug 2011) | 16 lines
  
  Merged revisions 330433 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | 9 lines
    
    Incorrect playback for Spanish in some circumstances
    
    When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you
    must use female pronunciation "1F". The function "say_date_with_format_es" does
    not take this in account.
    
    (closes ASTERISK-15016)
    Patch-by: Luis Jimenez
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2011-08-01 15:24:21 +00:00
Richard Mudgett
6cf345e023 Fixed compiler warning and a couple prototype mismatches.
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2011-07-31 00:19:11 +00:00
Richard Mudgett
a5be6a0f85 Merged revisions 330369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330369 | rmudgett | 2011-07-30 18:57:56 -0500 (Sat, 30 Jul 2011) | 11 lines
  
  Merged revisions 330368 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines
    
    Remove some redundant locking code in ast_do_masquerade().
    
    Also updated some comments.
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2011-07-31 00:05:55 +00:00
Gregory Nietsky
1c0078286e Merged revisions 330312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330312 | irroot | 2011-07-30 17:34:41 +0200 (Sat, 30 Jul 2011) | 15 lines
  
  Merged revisions 330311 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines
    
    prevent double masqurading channels when one is been hung up and deadlock avoidance is used.
    
    There is a race condition in ast_do_masquerade / ast_hangup (at least)
    
    Reported by me signed off by schmidts with input from David Vossel
    
    Review: https://reviewboard.asterisk.org/r/1323/
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2011-07-30 15:54:23 +00:00
Russell Bryant
6a15e95a32 astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER.
There is a fairly common pattern making its way through the code base where we
put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER.
The purpose is so that it can be passed into the object hash function.
However, this really seems like a hack and potentially error prone.  This patch
is a first stab at approach to avoid having to do that.

It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these
situations.  Then, the hash function can know whether it was given an object or
some custom data to hash.

The patch also changes some uses of ao2_find() for iax2_user and iax2_peer
objects to reflect how OBJ_KEY would be used.

So long, and thanks for all the fish.

Review: https://reviewboard.asterisk.org/r/1184/


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2011-07-29 19:34:36 +00:00
Sean Bright
d2ed4ed5f7 Merged revisions 330217 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330217 | seanbright | 2011-07-29 13:19:42 -0400 (Fri, 29 Jul 2011) | 9 lines
  
  Merged revisions 330213 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) | 2 lines
    
    Correct the check for O_RDONLY.
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2011-07-29 17:20:20 +00:00
Sean Bright
4e1fe6ccb7 Merged revisions 330204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330204 | seanbright | 2011-07-29 12:58:40 -0400 (Fri, 29 Jul 2011) | 9 lines
  
  Merged revisions 330203 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) | 2 lines
    
    Only write to wav files that were opened to be written to.
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2011-07-29 17:02:11 +00:00
Paul Belanger
b6a9795b9a Merged revisions 330162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul 2011) | 4 lines
  
  Fix typo pointed out on #asterisk
  
  Thanks notten
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2011-07-29 05:27:22 +00:00
Terry Wilson
be38ebe316 Merged revisions 330108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330108 | twilson | 2011-07-28 16:44:31 -0500 (Thu, 28 Jul 2011) | 9 lines
  
  Merged revisions 330107 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011) | 2 lines
    
    Make console colors work for TERM=xterm-256color
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2011-07-28 21:46:27 +00:00
Richard Mudgett
a3ba55e7c7 Merged revisions 330051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r330051 | rmudgett | 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) | 29 lines
  
  Merged revisions 330050 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r330050 | rmudgett | 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) | 22 lines
    
    Merged revisions 330033 from
    https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
    
    ..........
      r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines
    
      Datacalls with B410P fail.
    
      Incoming and outgoing call legs of a data call are using different
      formats: a-law, u-law.  When the call is bridged, the media stream is run
      through translation to convert the media formats.  The translation is bad
      for data calls.
    
      * Make incoming call that does not explicitly specify u-law or a-law use
      the DAHDI channel's default law.  The outgoing call always uses the
      default law from the DAHDI channel.
    
      (closes issue ABE-2800)
      Patches:
    	jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett
    ..........
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2011-07-28 17:16:07 +00:00
Jason Parker
16a32f5030 Merged revisions 329995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329995 | qwell | 2011-07-28 10:45:49 -0500 (Thu, 28 Jul 2011) | 13 lines
  
  Merged revisions 329994 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | 6 lines
    
    Fix a SIP transfer deadlock.
    
    The locking in this function is very scary.  There are like 6 structs involved.
    
    (closes issue AST-470)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-28 15:46:16 +00:00
Matthew Nicholson
b05b37dc53 Merged revisions 329992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329992 | mnicholson | 2011-07-28 10:28:21 -0500 (Thu, 28 Jul 2011) | 13 lines
  
  Merged revisions 329991 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul 2011) | 6 lines
    
    check for CONFIG_STATUS_FILE_INVALID when loading the res_fax config file
    
    Patch by: tzafrir
    Reported by: tzafrir
    (closes issue ASTERISK-18161)
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2011-07-28 15:30:30 +00:00
Sean Bright
f90bb00e29 Merged revisions 329952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu, 28 Jul 2011) | 4 lines
  
  The default conf-usermenu says that '8' can be used to leave the conference, so
  put that in the sample user menu.  '5' is supposed to extend the conference, but
  there doesn't appear to be a concept of that in the menu actions.
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2011-07-28 13:04:33 +00:00
Sean Bright
d4e239fa60 Merged revisions 329950 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul 2011) | 1 line
  
  Correct the spelling of 'conference.'
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2011-07-28 12:44:51 +00:00
Sean Bright
73462b32dd Merged revisions 329896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329896 | seanbright | 2011-07-28 07:35:27 -0400 (Thu, 28 Jul 2011) | 9 lines
  
  Merged revisions 329895 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) | 2 lines
    
    Make the output of Externhost in 'sip show settings' more consistent.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-28 11:36:12 +00:00
Jonathan Rose
d170e5e829 reverting 329840 due to failing tests. Going to change this feature to be purely optional.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-27 21:22:12 +00:00
Jonathan Rose
3ee80d6a90 Adds cdr logging of calls resulting in CONGESTION
Applies a patch made a long time ago by alecdavis which adds a CDR feature for logging
calls that failed due to congestion.

(closes issue #15907)
Reported by: alecdavis
Patches: 
      cdr_congestion.diff.txt uploaded by alecdavis (license #5546)

Review: https://reviewboard.asterisk.org/r/454/


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2011-07-27 20:42:18 +00:00
Sean Bright
1e332468dc Merged revisions 329771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329771 | seanbright | 2011-07-27 15:18:47 -0400 (Wed, 27 Jul 2011) | 15 lines
  
  Merged revisions 329767 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul 2011) | 8 lines
    
    Explicitly sort the module list so that the menuselect lists are sorted.
    
    (closes ASTERISK-18141)
    Reported by: Richard Miller
    Patches:
    		sort-order.diff uploaded by seanbright (License #5060)
    Tested by: leifmadsen
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2011-07-27 19:19:46 +00:00
Jonathan Rose
67acd8cbb1 Merged revisions 329710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329710 | jrose | 2011-07-27 13:11:07 -0500 (Wed, 27 Jul 2011) | 14 lines
  
  Merged revisions 329709 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) | 8 lines
    
    Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf
    
    (closes issue ASTERISK-16263)
    Reported by: richardf
    Patches: 
          nz-indications.patch uploaded by richardf (License #6015)
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2011-07-27 18:12:14 +00:00
Sean Bright
5858e755e4 Merged revisions 329670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul 2011) | 2 lines
  
  Sort the module list so that 'module show' is alphabetical.
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2011-07-27 15:26:31 +00:00
Tilghman Lesher
353bcc99c1 Merged revisions 329614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329614 | tilghman | 2011-07-26 23:25:26 -0500 (Tue, 26 Jul 2011) | 13 lines
  
  Merged revisions 329613 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011) | 6 lines
    
    Duration and billsec are swapped in high resolution time.
    
    Closes ASTERISK-18024
    Patches:
    	20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
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2011-07-27 04:27:19 +00:00
Jonathan Rose
41630b37bc Merged revisions 329538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329538 | jrose | 2011-07-26 09:19:34 -0500 (Tue, 26 Jul 2011) | 11 lines
  
  Merged revisions 329529 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | 5 lines
    
    Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same
    prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more
    appropriate anyway.
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2011-07-26 14:27:31 +00:00
Jonathan Rose
462e0fe530 Merged revisions 329528 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines
  
  Merged revisions 329527 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines
    
    Fixes some voicemail forwarding behavior based around prepend mode.
    
    Formerly, prepend forwarding would have the user record a message with no useful prompt
    and an expectation for the user to push a button on the phone when finished recording.
    If a length of silence was detected instead, the recording would be canceled and the user
    would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording
    would also bug out in the sense that they would write over the original message and get
    sent to the recipient regardless of whether they timed out or were accepted. This patch
    fixes this issue and adds a prompt which will be played after a timeout informing the
    user that they needed to press a button. Currently, the sound files that we have are
    somewhat inadquate for this, so after the call we simply have Allison say "Please try
    again. Then press pound." which actually relies on two separate sound files. Just one
    would be more appropriate.
    
    reporter: Vlad Povorozniuc
    Review: https://reviewboard.asterisk.org/r/1327/ 
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2011-07-26 14:17:13 +00:00
Paul Belanger
06343443e1 Merged revisions 329472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329472 | pabelanger | 2011-07-25 15:55:33 -0400 (Mon, 25 Jul 2011) | 9 lines
  
  Merged revisions 329471 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul 2011) | 2 lines
    
    Decrease verbose messages to debug, to help clean up CLI.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-25 19:57:27 +00:00
Gregory Nietsky
3b1cc6de8d dsp_process was enhanced to work with alaw and ulaw in addition to slin.
noticed that some functions could be refactored here it is.

Reported by: irroot
Tested by: irroot, mnicholson
Review: https://reviewboard.asterisk.org/r/1304/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-25 14:07:01 +00:00
Gregory Nietsky
5c627eba2b Remove lastmsgssent from sip it has not been working since 1.6
Clean up the return values to be consistant not currently used
Add doxygen returns
MWI Event is sent on Register

(closes issue ASTERISK-17866)
Reported by: one47
Tested by: irroot, mvanbaak
Review: https://reviewboard.asterisk.org/r/1172/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-25 09:39:54 +00:00
Richard Mudgett
c0f592df46 Merged revisions 329334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011) | 1 line
  
  Make use less redundant loop construct for iterating over hints.
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2011-07-22 21:15:28 +00:00
Richard Mudgett
a5c65bb939 Merged revisions 329331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r329331 | rmudgett | 2011-07-22 15:43:07 -0500 (Fri, 22 Jul 2011) | 55 lines
  
  Merged revisions 329299 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines
    
    Deadlocks dealing with dialplan hints during reload.
    
    There are two remaining different deadlocks reported dealing with dialplan
    hints.
    
    The deadlock in ASTERISK-17666 is caused by invalid locking order in
    ast_remove_hint().  The hints container must be locked before the hint
    object.
    
    The deadlock in ASTERISK-17760 is caused by a catch-22 situation in
    handle_statechange().  The deadlock is caused by not having the conlock
    before calling the watcher callbacks.  Unfortunately, having that lock
    causes a different deadlock as reported in ASTERISK-16961.
    
    * Fixed ast_remove_hint() locking order.
    
    * Made handle_statechange() no longer call the watcher callbacks holding
    any locks that matter.
    
    * Made hint ao2 destructor do the watcher callbacks for extension
    deactivation to guarantee that they get called.
    
    * Fixed hint reference leak in ast_add_hint() if the callback container
    constructor failed.
    
    * Fixed hint reference leak in complete_core_show_hint() for every hint it
    found for CLI tab completion.
    
    * Adjusted locking in ast_merge_contexts_and_delete() for safety.
    
    * Added context_merge_lock to prevent ast_merge_contexts_and_delete() and
    handle_statechange() from interfering with each other.
    
    * Fixed ast_change_hint() not taking into account that the extension is
    used for the hash key.
    
    (closes issue ASTERISK-17666)
    Reported by: irroot
    Tested by: irroot
    JIRA SWP-3318
    
    (closes issue ASTERISK-17760)
    Reported by: Byron Clark
    Tested by: irroot
    JIRA SWP-3393
    
    Review: https://reviewboard.asterisk.org/r/1313/
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-22 20:46:36 +00:00
Russell Bryant
f243d129c9 Merged revisions 329257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines
  
  s/1.10/10.0/
........


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2011-07-21 20:26:44 +00:00
Richard Mudgett
54f92a68c7 Merged revisions 329204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines
  
  Merged revisions 329203 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines
    
    Document parkinglot in chan_dahdi.conf.sample.
    
    * Document existing feature in chan_dahdi.conf.sample.
    
    * Remove some dead code related to the parkinglot option.
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21 18:06:47 +00:00
Richard Mudgett
a2e30b1908 Merged revisions 329200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines
  
  Merged revisions 329199 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines
    
    Update PickupChan documentation.
    
    The PickupChan uses the ampersand as the argument separator.
    Was documented as:
    PickupChan(channel[,channel2[,...][,options]])
    
    Fixed documentation to:
    PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
    
    This is a continuation of ASTERISK-17494 for v1.8 and later.
    
    (closes issue ASTERISK-18144)
    Reported by: Erik Smith
    Patches:
          pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
    Tested by: Erik Smith
  ........
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2011-07-21 17:33:06 +00:00
Richard Mudgett
3b80737787 Merged revisions 329145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines
  
  Merged revisions 329144 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines
    
    Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked!
    
    This appears to be a leftover from when ast_channel was converted to ao2
    objects.
    
    Simply removed the extraneous unlock.
    
    (closes issue ASTERISK-17772)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21 16:59:38 +00:00
Jason Parker
2c198555fd Fix UPGRADE.txt files for Asterisk 10.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21 16:22:58 +00:00
Jason Parker
8761398f0e Remove another 2.0 property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21 16:14:44 +00:00
Russell Bryant
e6e0fb0d9a Fix merge properties to reflect Asterisk 10 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21 16:05:33 +00:00
Paul Belanger
3a4520214a Merged revisions 329055 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/2.0

................
  r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines
  
  Merged revisions 329027 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines
    
    Asterisk now requires libpri 1.4.11+ for PRI support.
  ........
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2011-07-20 21:31:29 +00:00
Terry Wilson
aada2733d6 Merged revisions 328992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/2.0

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  r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines
  
  Merged revisions 328987 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines
    
    We can't guarantee an eth0 is present
    
    FreeBSD test fails on this case presumably because there is no eth0 on the test
    machine. Better to just remove this test for now.
  ........
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2011-07-20 20:19:16 +00:00
Kinsey Moore
9c232a5470 Merged revisions 328936 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/2.0

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  r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines
  
  Merged revisions 328935 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines
    
    Inband DTMF regression
    
    The functionality of inband DTMF in chan_sip relied upon
    ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
    ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
    documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
    never inband.  This fixes the regression introduced in revision 328823.
  ........
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2011-07-20 19:03:17 +00:00
Kevin P. Fleming
6855bb873f Merged revisions 328879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/2.0

................
  r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines
  
  Merged revisions 328878 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines
    
    Revert partial attempt at handling pathnames with spaces.
    
    Revision 299794 attempted to improve the build system to be able to handle
    pathnames (primarily DESTDIR) with spaces in them, since this is common on
    some platforms (including Mac OSX). Unfortunately, the changes were incomplete
    and did not actually provide the desired behavior, and as a side effect the
    functionality that ensured that stale headers in the Asterisk 'include' directory
    were removed got broken. In addition, the check for stale (and possibly
    incompatible) modules in the Asterisk 'modules' directory also got broken, and
    would never report any stale modules. Users upgrading to this version or later
    versions would then see unexpected module load errors.
    
    Since there are few users who actually want to install Asterisk into paths
    that contain spaces, and a proper fix for the build system would take many hours,
    the best solution for now is to just revert the partial solution.
  ........
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2011-07-19 21:32:54 +00:00
Kevin P. Fleming
248b0f7e53 Edit the merge properties to match their names.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-19 21:32:35 +00:00
Russell Bryant
ce291c116c Fix properties after twilson's change so merges still work
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-19 21:21:53 +00:00
Kinsey Moore
1dc97eb69b Merged revisions 328824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines
  
  Merged revisions 328823 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines
    
    RTP bridge away with inband DTMF and feature detection
    
    When deciding whether Asterisk was allowed to bridge the call away from the
    core, chan_sip did not take into account the usage of features on dialed
    channels that require monitoring of DTMF on channels utilizing inband DTMF.
    This would cause Asterisk to allow the call to be locally or remotely bridged, 
    preventing access to the data required to detect activations of such features.
    
    (closes 17237)
    Review: https://reviewboard.asterisk.org/r/1302/
  ........
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2011-07-19 18:07:22 +00:00
Kinsey Moore
4ea4b7e1ab Merged revisions 328771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines
  
  Merged revisions 328770 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines
    
    MeetMe requests a PIN twice in some circumstances
    
    If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
    options, MeetMe will ask for the PIN two times: once for creating the
    conference and once for entering the conference.  This behavior was introduced
    in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
    controlling PIN entry for joining a conference.
    
    (closes AST-601)
    Review: https://reviewboard.asterisk.org/r/1305/
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2011-07-19 15:49:55 +00:00
Terry Wilson
c26bb50cc3 Merged revisions 328717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines
  
  Merged revisions 328716 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines
    
    Make AST_LIST_REMOVE safer
    
    AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This
    commit also adds linked list unit tests.
    
    Review: https://reviewboard.asterisk.org/r/1321/
  ........
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2011-07-19 02:00:56 +00:00
Mark Murawki
3719ee2d65 Merged revisions 328664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines
  
  Merged revisions 328663 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines
    
    app_dial may double free a channel datastore
    
    When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash.  Make sure to check if the datastore still exists before trying to free it.
    
    (closes issue ASTERISK-17917)
    Reported by: Mark Murawski
    Tested by: Mark Murawski
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2011-07-18 20:51:47 +00:00