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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259439 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259438 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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



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

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

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

........
  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.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  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
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

added a "fun:" prefix line 24


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258934 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258855 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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

........
  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



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258387 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue #17220)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258383 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These changes add the ability to run 'make asterisk.txt' just like the existing
'make asterisk.pdf' commands to generate a text document from the TeX files we
have in the doc/tex/ directory. I've also updated a few of the .tex files because
they weren't properly escaping certain characters so they would show up as Unicode
characters (like [U+021C]). Made changes to the configure scripts so it would
detect the catdvi program which is required to convert the .dvi file generated
by latex.

I've also added a few lines to the build_tools/prep_tarball script so that the
text documentation gets generated and added to future tarballs of Asterisk
releases.

(closes issue #17220)
Reported by: lmadsen
Patches: 
      asterisk.txt.patch uploaded by lmadsen (license 10)
      asterisk.txt.patch-v4 uploaded by pabelanger (license 224)
Tested by: lmadsen, pabelanger

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This directs users to documents which can help explain the
concepts and configuration options settable with the function.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258345 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17100)
Reported by: secesh
Tested by: pabelanger

Review: https://reviewboard.asterisk.org/r/594/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is a clear mistake in logic.  Future discussions
about how to avoid having to handle uri's like this
should take place in the future, but this fix needs
to go in for now.

(closes issue #15847)
Reported by: ebroad
Patches:
      doublesip.patch uploaded by ebroad (license 878)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258256 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added State and Direction variables for new MixMonitorMute AMI command.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258228 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a new manager command to mute/unmute MixMonitor audio on a channel. 
Added a new feature to audiohooks so that you can mute either read / write
(or both) types of frames - this allows for MixMonitor to mute either side
of the conversation without affecting the conversation itself.

(closes issue #16740)
Reported by: jmls

Review: https://reviewboard.asterisk.org/r/487/



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Minor tweaks and documentation added by me.

(closes issue #17058)
Reported by: pprindeville
Patches: 
      freenum.patch#5 uploaded by pprindeville (license 347)
Tested by: lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258147 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17171)
Reported by: thehar
Patches: 
      sip-friends.patch uploaded by thehar (license 831)
Tested by: pabelanger, thehar

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