(closes issue #17040)
Reported by: pprindeville
Patches:
      asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347)
Tested by: pprindeville

Review: https://reviewboard.asterisk.org/r/565/



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  r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines
  
  resolves deadlocks in chan_local
  
  Issue_1.
  In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan,
  and pvt->owner.  Proper deadlock avoidance is done when the channel to hangup
  is the outbound chan_local channel, but when it is not the outbound channel we
  have an issue... We attempt to do deadlock avoidance only on the tech pvt, when
  both the tech pvt and the pvt->owner are locked coming into that loop.  By
  never giving up the pvt->owner channel deadlock avoidance is not entirely possible.
  This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt
  when trying to get the pvt->chan lock.
  
  Issue_2.
  ast_prod() is used in ast_activate_generator() to queue a frame on the channel
  and make the channel's read function get called.  This function is used in
  ast_activate_generator() while the channel is locked, which mean's the channel
  will have a lock both from the generator code and the frame_queue code by the
  time it gets to chan_local.c's local_queue_frame code... local_queue_frame
  contains some of the same crazy deadlock avoidance that local_hangup requires,
  and this recursive lock prevents that deadlock avoidance from happening correctly.
  This patch removes ast_prod() from the channel lock so only one lock is held during
  the local_queue_frame function.
  
  (closes issue #17185)
  Reported by: schmoozecom
  Patches:
        issue_17185_v1.diff uploaded by dvossel (license 671)
        issue_17185_v2.diff uploaded by dvossel (license 671)
  Tested by: schmoozecom, GameGamer43
  
  Review: https://reviewboard.asterisk.org/r/631/
........


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  r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) | 6 lines
  
  Update config.guess.
  
  Updating config.guess because after installing Ubuntu Server 9.10 and
  running all the update scripts, running ./configure would not continue
  because it was unable to determine what kind of system I had. After
  updating config.guess things started working again.
........


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  r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line
  
  Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir.
........


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  r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | 1 line
  
  Missed this when removing $ID
........


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  r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | 7 lines
  
  Remove usage of `id` since it isn't useful and was causing breakge.
  
  Solaris `id` doesn't support the -u argument.  Instead of figuring out how to
  fix this to work on Solaris, I decided to check why it was necessary and where
  else it was used.  It was only used in one place, and it hasn't been needed
  for a very long time (I question whether it was ever needed).
........


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  r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines
  
  Do not play goodbye prompt after timeout of message review.
  
  ABE-2124
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  r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines
  
  DAHDI "WARNING" message is confusing and vague
  
  "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success"
  
  Changed the warning to "Failed to decode CallerID on channel 'name'".  The
  message before it is likely more specific about why the CallerID decode
  failed.
  
  SWP-501
  AST-283
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  r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) | 15 lines
  
  Update sounds files.
  
  * Add additional sounds prompts for say_enumeration
  * Update the English conference sounds prompts so they are better
    quality and all sound more consistent
  * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files to
    include all present sound files
  
  Both core (en, fr, es) and extra (en, fr) sounds files have been updated.
  
  (closes issue #16200)
  Reported by: murf
  
  (closes issue #17137)
  Reported by: lmadsen
........


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  r259441 | qwell | 2010-04-27 16:15:46 -0500 (Tue, 27 Apr 2010) | 1 line
  
  Add gar to the check for AR for those silly OSes (Solaris) that don't have ar.
........


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autoconf2.13 couldn't handle AC_PROG_GREP, so I removed it.  This is fine,
since we don't need to use anything that the configure script doesn't.


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(closes issue #17175)
Reported by: lmadsen
Patches:
      Bug_Tracker_Workflow.v2.txt uploaded by pabelanger (license 224)
Tested by: pabelanger, lmadsen

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The hope is that Solaris won't be as whiny after this change.



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  r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | 5 lines
  
  Support the silly OSes that don't have ar and strip.
  
  Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and
  AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS.
........


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  r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines
  
  hidecalleridname parameter in chan_dahdi.conf
  
  Issue #7321 implements a new chan_dahdi configuration option.  However, a
  change mentioned in the issue was never implemented.  This is the change
  that will allow the feature to work.
  
  I added a note to chan_dahdi.conf.sample about the feature.
  
  (closes issue #17143)
  Reported by: djensen99
  Patches:
        diff.txt uploaded by djensen99 (license NA) (One line change)
  Tested by: djensen99
........


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  r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr 2010) | 3 lines
  
  Let compilation succeed warning-free when DONT_OPTIMIZE is turned off.
........


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  r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr 2010) | 13 lines
  
  Prevent Newchannel manager events for dummy channels.
  
  No Newchannel manager event will be fired for channels that are
  allocated to not match a registered technology type. Thus bogus
  channels allocated solely for variable substitution or CDR
  operations do not result in a Newchannel event.
  
  (closes issue #16957)
  Reported by: atis
  
  Review: https://reviewboard.asterisk.org/r/601
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added a "fun:" prefix line 24


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(closes issue #16988)
Reported by: frawd
Patches: 
      chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
Tested by: russell

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The fax session initilization code for T.38 faxes has been rewritten. T.38 session initialization was removed from generic_fax_exec, and split into two different code paths for receive and send.  Also the 'z' option (to send a T.38 reinvite if we do not receive one) was added to sendfax.

In the output of 'fax show sessions', the 'Type' column has been renamed to 'Tech' and replaced with a new 'Tech' column that will report 'G.711' or 'T.38'.

Control of ECM defaults has been added to res_fax

A 'fax show settings' CLI command has been added.

Support of the new AST_T38_REQUEST_PARMS control method request to handle channels that have already received a T.38 reinvite before the FAX application is start has been added.

Support for the 'fax show settings' command has been added to res_fax_spandsp and handling of the ECM flag has been slightly altered.


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Don't pass zero callerid string to ooh323 stack because it can't encode this properly and
can't generate setup message.

(closes issue #17186)
Reported by: vmikhelson
Patches:
      zero_callerid_num.patch uploaded by may213 (license 454)
Tested by: may213



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  r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines
  
  When StopMonitor is called, ensure that it will not be restarted by a channel event.
  (closes issue #16590)
   Reported by: kkm
   Patches: 
         resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888)
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  r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) | 8 lines
  
  Set the proper disposition on originated calls.
  
  (closes issue #14167)
  Reported by: jpt
  Patches:
        call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
  Tested by: dlotina, rmartinez, mnicholson
........
  r258670 | mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 lines
  
  Fix broken CDR behavior.
  
  This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER.
  
  Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call().  To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call().
  
  (closes issue #16797)
  Reported by: VarnishedOtter
  Tested by: mnicholson
........

(closes issue #16222)
Reported by: telles
Tested by: mnicholson



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This patch introduces another test in test_event.c that exercises most of the
subscription related ast_event API calls.  I made some minor additions to the
existing event allocation test to increase API coverage by the test code.
Finally, I made a list in a comment of API calls not yet touched by the test
module as a to-do list for future test development.

During the development of this test code, I discovered a number of bugs in
the event API.

1) subscriptions to AST_EVENT_ALL were not handled appropriately in a couple
   of different places.  The API allows a subscription to all event types,
   but with IE parameters, just as if it was a subscription to a specific
   event type.  However, the parameters were being ignored.  This affected
   ast_event_check_subscriber() and event distribution to subscribers.

2) Some of the logic in ast_event_check_subscriber() for checking subscriptions
   against query parameters was wrong.

Review: https://reviewboard.asterisk.org/r/617/


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"Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a
loaded module was not (or vice versa).  This also immensely simplifies the
lock code, since there are no longer 2 separate versions of them.

Review: https://reviewboard.asterisk.org/r/508/


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This module implements an abstraction for retrieving and exporting
asterisk data.
Developed by:
	Brett Bryant <brettbryant@gmail.com>
	Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h

Review: https://reviewboard.asterisk.org/r/275/



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  r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines
  
  Fix looping forever when no input received in certain voicemail menu scenarios.
  
  Specifically, prompting for an extension (when leaving or forwarding a message)
  or when prompting for a digit (when saving a message or changing folders).
  
  ABE-2122
  SWP-1268
........


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