This will save a considerable amount of CPU on the BSDs, including Mac OS X,
as it eliminates several places in the code that we previously used a busy
loop.  Additionally, this adds a res_timing interface, using kqueue timers.

Review: https://reviewboard.asterisk.org/r/543/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17308)
Reported by: pabelanger
Patches:
      cli.modules.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, russell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262800 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Marquis42 suggested a better method of doing what I wanted because I ended up
removing the WARNING message for all instances when really I just wanted to
remove it for the 'return' keyword, not everything.

(issue #17145)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262798 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17145)
Reported by: okrief

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262796 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines
  
  fixes app_meetme dsp error
  
  We attempted to detect silence after translating a frame
  from signed linear.  This caused a flooding of errors.  To
  resolve this the code to detect silence was moved before the
  translation.
  
  (closes issue #17133)
  Reported by: jsdyer
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262744 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Must do a deep copy of the cc_params in duplicate_pseudo().  Otherwise,
when the duplicate pseudo channel is destroyed, it frees the original
pseudo channel cc_params.  The original pseudo channel is then left with a
dangling pointer for when the next duplicated pseudo channel is created.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

..........
  r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines

  Forgot some conditionals around the callrerouting facility help text.

  JIRA ABE-2223
..........
  r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines

  Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line)

  In the case of ISDN point-to-multipoint (multidevice) you can use the
  mISDN "facility calldeflect" application for call diversions from external
  (PSTN) to external (PSTN).  In that case this is the only way to get rid
  of the two call legs to the PBX and let the calling number at the C party
  become the number of the A party.  In the case of ISDN point-to-point
  (exchange line) the call deflection facility may not be used.  Instead a
  call rerouting facility has to be used.

  This patch for chan_misdn.c is an extension to realize this service
  (facility rerouting application).  It can accept either spelling:
  "callrerouting" or "callrerouteing".

  The patch is tested towards Deutsche Telekom and requires a modified
  version of mISDN from Digium, Inc.

  Patches:
        misdn_rerouteing_corrected.patch (Slightly modified.)

  JIRA ABE-2223


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16576)
 Reported by: uxbod
 Patches: 
       20100505__issue16576.diff.txt uploaded by tilghman (license 14)
 Tested by: uxbod


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clean up chan_sip.c to use new AST_CLI functions

(closes issue #17287)
Reported by: pabelanger
Patches:
      issue17287.patch uploaded by pabelanger (license 224)
Tested by: russell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931
Section 5.3.2.  However, this resulted in an unexpected behaviour change
to the upper layer (Asterisk).

This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2
call hangup better if the version of libpri supports it.

(issue #17104)
Reported by: shawkris
Tested by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also cleanup the formatting and add a few more that seem like good candidates.

(closes issue #16157)
 Reported by: wimpy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262513 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines
  
  Use a less silly method for modifying a flex-generated file.
  
  The sed syntax that was used wasn't actually valid, causing some versions to
  choke.  This is the method that is used in 1.6.x+ for similar changes.
  
  (closes issue #16696)
  Reported by: bklang
  Patches: 
        16696-sedfix.diff uploaded by qwell (license 4)
  Tested by: qwell
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16303)
Reported by: dant
Patches:
      issue16303.patch.v2 uploaded by pabelanger (license 224)
Tested by: dant, lmadsen, pabelanger


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17238)
Reported by: pprindeville
Patches:
      chan_sip-bug17238.patch uploaded by pprindeville (license 347)
Tested by: pprindeville


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines
  
  Fix issue #17302 a slightly different way (mad props to Qwell)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262330 65c4cc65-6c06-0410-ace0-fbb531ad65f3

As usual, the way they do things is different, so we need to account for that.
automake is versioned ala BSD/Linux, but autoconf is not.  We don't actually
need to specify a version there, since AC_PREREQ will cover it for us.  Things
will fail pretty loudly if AC_PREREQ isn't met.

(closes issue #16341)
Reported by: bklang
Patches: 
      opensolaris_bootstrap.sh uploaded by bklang (license 919)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16863)
Reported by: schern
Patches:
      app_directed_pickup.c.patch uploaded by schern (license 995)
      for_trunk.diff uploaded by cjacobsen (license 1029)
Tested by: Graber, cjacobsen, lathama, rickead2000, dvossel


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262240 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There is a race condition between console_hangup()
and start_stream().  It is possible for console_hangup()
to be called and then the stream thread to begin after the hangup.
To avoid this a check in start_stream() to make sure the pvt-owner
still exists while the pvt lock is held is made.  If the owner
is gone that means the channel hung up and start_stream should
be aborted.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) | 10 lines
  
  Allow compilation on Mac OS X 10.4 (Tiger)
  
  (closes issue #17297)
   Reported by: jcovert
   Patches: 
         20100506__issue17297.diff.txt uploaded by tilghman (license 14)
  
  (closes issue #17302)
   Reported by: jcovert
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262152 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS.

(closes issue #17309)
 Reported by: stuarth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear
their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber.

This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape.

If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour.

  Reported by: alecdavis
  Tested by: alecdavis
  Patch
	 vm_a_extension.diff2.txt uploaded by alecdavis (license 585)

Review: https://reviewboard.asterisk.org/r/489/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261964 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17245)
 Reported by: thedavidfactor
 Patches: 
       20100426__issue17245.diff.txt uploaded by tilghman (license 14)
 Tested by: murraytm


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261917 65c4cc65-6c06-0410-ace0-fbb531ad65f3

We nicely detect the right flags on each system for building Asterisk with
pthreads, then ignore it for every other build option that requires us to
build with pthreads.  This caused some items to return a false negative.
Also cleanup some minor naming issues that caused "library library" redundancy
in the output.

(closes issue #17303)
 Reported by: stuarth
 Patches: 
       20100507__issue17303.diff.txt uploaded by tilghman (license 14)
 Tested by: stuarth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #17282)
Reported by: stuarth
Tested by: stuarth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The pri_dchannel thread currently violates locking order by locking the private
and then attempting to queue a frame, which needs to lock the channel. Queueing
a frame is unneccesary though and is actually a regression since sig_pri.
All the places that currently use ast_softhangup_nolock now will just set the
softhangup value directly as before.

(closes issue #17216)
Reported by: lmsteffan
Patches: 
      bug17216.patch uploaded by jpeeler (license 325)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made more places use pri_queue_control() instead of pri_queue_frame()
and a local frame variable.

* Made pri_queue_frame() use sig_pri_lock_owner().  pri_queue_frame() no
longer releases the libpri access lock unless it is required.

* Made the pri_queue_frame() and pri_queue_control() parameter list
similar to sig_pri_lock_owner().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines
  
  Only allow the operator key to be accepted after leaving a voicemail.
  
  Or rather disallow the operator key from being accepted when not offered,
  such as after finishing a recording from within the mailbox options menu.
  
  ABE-2121
  SWP-1267
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261736 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | 4 lines
  
  Use the versioned MOH tarballs, now that we have them.
  
  This makes for more reproducibility.  Prompted by a discussion in #asterisk-dev
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261609 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The example given within the related issue showed 120 lines, which was mostly
a result of the body being XML.

(closes issue #17179)
 Reported by: khw


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I modified the original patch for trunk to use the unit test API.

(issue #17277)
Reported by: cappucinoking
Patches:
      test_heap.diff uploaded by cappucinoking (license 1036)
Tested by: cappucinoking, russell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261500 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This bug surfaced in 1.6.2 and does not affect code in any other released
version of Asterisk.  It manifested itself as SIP qualify not happening when
it should, causing peers to go unreachable.  This was debugged down to scheduler
entries sometimes not getting executed when they were supposed to, which was in
turn caused by an error in the heap code.

The problem only sometimes occurs, and it is due to the logic for removing an entry
in the heap from an arbitrary location (not just popping off the top).  The scheduler
performs this operation frequently when entries are removed before they run (when
ast_sched_del() is used).

In a normal pop off of the top of the heap, a node is taken off the bottom,
placed at the top, and then bubbled down until the max heap property is restored
(see max_heapify()).  This same logic was used for removing an arbitrary node
from the middle of the heap.  Unfortunately, that logic is full of fail.  This
patch fixes that by fully restoring the max heap property when a node is thrown
into the middle of the heap.  Instead of just pushing it down as appropriate, it
first pushes it up as high as it will go, and _then_ pushes it down.

Lastly, fix a minor problem in ast_heap_verify(), which is only used for
debugging.  If a parent and child node have the same value, that is not an
error.  The only error is if a parent's value is less than its children.

A huge thanks goes out to cappucinoking for debugging this down to the scheduler,
and then producing an ast_heap test case that demonstrated the breakage.  That
made it very easy for me to focus on the heap logic and produce a fix.  Open source
projects are awesome.

(closes issue #16936)
Reported by: ib2
Tested by: cappucinoking, crjw

(closes issue #17277)
Reported by: cappucinoking
Patches:
      heap-fix.rev2.diff uploaded by russell (license 2)
Tested by: cappucinoking, russell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261496 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes a crash when some config section had an incorrect channel config.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261451 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261405 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes some breakage in the test suite, that uses the contents of asterisk.conf
to discover the install layout on the system.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261364 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Refer-To header field containing the Replaces header in the URI
was not being decoded properly.  This caused invalid parsing between
the caller id field and the domain resulting in a failed transfer.

(closes issue #17284)
Reported by: dvossel



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........
  r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines
  
  Registration fix for SIP realtime.
  
  Make sure realtime fields are not empty.
  
  (closes issue #17266)
  Reported by: Nick_Lewis
  Patches:
        chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657)
  Tested by: Nick_Lewis, sberney
  
  Review: https://reviewboard.asterisk.org/r/643/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a recognized media type was present, but the media type was not
enabled for the channel, then a warning would be emitted. For instance,
attempting to get CHANNEL(rtpsource,video) on a call with no video would
cause a warning message to appear.

With this change, the warning will only appear if the stream argument
is not recognized as being a media type that can be specified.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261313 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Resets each member's lastcall to 0 now.

(closes issue #17262)
Reported by: rain
Patches:
      wrapuptime_reset_fix.diff uploaded by rain (license 327)
Tested by: rain


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261232 65c4cc65-6c06-0410-ace0-fbb531ad65f3