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2009-06-11  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.6.2.0-beta3

2009-06-11 12:19 +0000 [r200051]  Leif Madsen <lmadsen@digium.com>

	* build_tools/make_version_h, /, build_tools/make_version_c: Merged
	  revisions 200039 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 |
	  lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
	  Fix path for .flavor and .version (issue #14737) Reported by:
	  davidw Patches: flavor.patch uploaded by davidw (license 780)
	  Tested by: davidw ........

2009-06-10 20:37 +0000 [r199998]  David Brooks <dbrooks@digium.com>

	* main/pbx.c, /: Fixes the argument order in definition of
	  new_find_extension(). In the definition of new_find_extension(),
	  the arguments 'callerid' and 'label' were swapped. The prototype
	  declaration and all calls to the function are ordered 'callerid'
	  then 'label', but the function itself was ordered 'label' then
	  'callerid'. (closes issue #15303) Reported by: JimDickenson

2009-06-10 20:18 +0000 [r199966]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 |
	  mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6
	  lines Only try to use the invite_branch on outgoing INVITEs with
	  auth credentials. I have added a comment to the code to help ease
	  understanding of the logic here as well. ........

2009-06-10 16:13 +0000 [r199860]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400
	  (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed,
	  10 Jun 2009) | 2 lines __WORDSIZE is not available on all
	  platforms, so use sizeof(void *) instead. ........
	  ................

2009-06-09 20:48 +0000 [r199744-199819]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 |
	  dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines
	  CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command
	  only used UDP rather than copying the transport type from the
	  peer. (closes issue #15283) Reported by: jthurman Patches:
	  sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
	  Tested by: jthurman, dvossel ........

	* main/loader.c, /, res/res_timing_pthread.c,
	  include/asterisk/module.h, res/res_timing_dahdi.c,
	  res/res_timing_timerfd.c: Merged revisions 199743 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009)
	  | 11 lines module load priority This patch adds the option to
	  give a module a load priority. The value represents the order in
	  which a module's load() function is initialized. The lower the
	  value, the higher the priority. The value is only checked if the
	  AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER
	  flag is not set, the value will never be read and the module will
	  be given the lowest possible priority on load. Since some modules
	  are reliant on a timing interface, the timing modules have been
	  given a high load priorty. (closes issue #15191) Reported by:
	  alecdavis Tested by: dvossel Review:
	  https://reviewboard.asterisk.org/r/262/ ........

2009-06-08 19:39 +0000 [r199634]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400
	  (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun
	  2009) | 21 lines Increase the size of our thread stack on 64 bit
	  processors. We were setting the stack size for each thread to
	  240KB regardless of architecture, which meant that in some
	  scenarios we actually had less available stack space on 64 bit
	  processors (pointers use 8 bytes instead of 4). So now we
	  calculate the stack size we reserve based on the platform's
	  __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
	  bit -> 1008KB (that's right, we're ready for 128 bit processors)
	  Patch typed by me but written by several members of
	  #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
	  issue #14932) Reported by: jpiszcz Patches:
	  06052009_issue14932.patch uploaded by seanbright (license 71)
	  Tested by: seanbright ........ r199628 | seanbright | 2009-06-08
	  15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the
	  stack size calculation just introduced. ........ ................

2009-06-08 17:42 +0000 [r199591]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Recorded merge of revisions 199588 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon,
	  08 Jun 2009) | 9 lines Fix a deadlock that could occur when
	  setting rtp stats on SIP calls. (closes issue #15143) Reported
	  by: cristiandimache Patches: 15143.patch uploaded by mmichelson
	  (license 60) Tested by: cristiandimache ........

2009-06-06 21:39 +0000 [r199369]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 199368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 |
	  russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines
	  Switch from "echo -n" to printf. On my mac, the -n was just
	  getting printed out. ........

2009-06-05 21:25 +0000 [r199299]  David Vossel <dvossel@digium.com>

	* include/asterisk/devicestate.h, /, main/devicestate.c: Merged
	  revisions 199298 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009)
	  | 21 lines Merged revisions 199297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009)
	  | 14 lines Fixes issue with hints giving unexpected results.
	  Hints with two or more devices that include ONHOLD gave
	  unexpected results. (closes issue #15057) Reported by:
	  p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
	  (license 671) pbx.c.1.4.patch uploaded by p (license 558)
	  devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
	  p_lindheimer, dvossel Review:
	  https://reviewboard.asterisk.org/r/254/ ........ ................

2009-06-05 13:52 +0000 [r199230]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun
	  2009) | 14 lines Correct "dahdi show channels" output when
	  specifying a group. Since a DAHDI channel may belong to multiple
	  groups, we need to use a bitwise and instead of equivalence to
	  determine whether to display the channel information. (closes
	  issue #15248) Reported by: gentian Patches: 15248.patch uploaded
	  by mmichelson (license 60) Tested by: gentian ........

2009-06-04 19:15 +0000 [r199140]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500
	  (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04
	  Jun 2009) | 3 lines Additional updates to AST-2009-001 ........
	  ................

2009-06-04 14:53 +0000 [r199054]  Sean Bright <sean.bright@gmail.com>

	* include/asterisk/_private.h, main/asterisk.c, main/loader.c, /:
	  Merged revisions 199051 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun
	  2009) | 47 lines Merged revisions 199022 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun
	  2009) | 40 lines Safely handle AMI connections/reload requests
	  that occur during startup. During asterisk startup, a lock on the
	  list of modules is obtained by the primary thread while each
	  module is initialized. Issue 13778 pointed out a problem with
	  this approach, however. Because the AMI is loaded before other
	  modules, it is possible for a module reload to be issued by a
	  connected client (via Action: Command), causing a deadlock. The
	  resolution for 13778 was to move initialization of the manager to
	  happen after the other modules had already been lodaded. While
	  this fixed this particular issue, it caused a problem for users
	  (like FreePBX) who call AMI scripts via an #exec in a
	  configuration file (See issue 15189). The solution I have come up
	  with is to defer any reload requests that come in until after the
	  server is fully booted. When a call comes in to ast_module_reload
	  (from wherever) before we are fully booted, the request is added
	  to a queue of pending requests. Once we are done booting up, we
	  then execute these deferred requests in turn. Note that I have
	  tried to make this a bit more intelligent in that it will not
	  queue up more than 1 request for the same module to be reloaded,
	  and if a general reload request comes in ('module reload') the
	  queue is flushed and we only issue a single deferred reload for
	  the entire system. As for how this will impact existing
	  installations - Before 13778, a reload issued before module
	  initialization was completed would result in a deadlock. After
	  13778, you simply couldn't connect to the manager during startup
	  (which causes problems with #exec-that-calls-AMI configuration
	  files). I believe this is a good general purpose solution that
	  won't negatively impact existing installations. (closes issue
	  #15189) (closes issue #13778) Reported by: p_lindheimer Patches:
	  06032009_15189_deferred_reloads.diff uploaded by seanbright
	  (license 71) Tested by: p_lindheimer, seanbright Review:
	  https://reviewboard.asterisk.org/r/272/ ........ ................

2009-06-03 15:24 +0000 [r198827-198886]  David Vossel <dvossel@digium.com>

	* main/channel.c, /, main/features.c, include/asterisk/channel.h:
	  Merged revisions 198856 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 |
	  dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
	  Generic call forward api, ast_call_forward() The function
	  ast_call_forward() forwards a call to an extension specified in
	  an ast_channel's call_forward string. After an ast_channel is
	  called, if the channel's call_forward string is set this function
	  can be used to forward the call to a new channel and terminate
	  the original one. I have included this api call in both
	  channel.c's ast_request_and_dial() and feature.c's
	  feature_request_and_dial(). App_dial and app_queue already
	  contain call forward logic specific for their application and
	  options. (closes issue #13630) Reported by: festr Review:
	  https://reviewboard.asterisk.org/r/271/ ........

	* channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009)
	  | 8 lines fixes issue with channels not going down after transfer
	  Iax2 currently does not support native bridging if the timeoutms
	  value is set. We check for that in iax2_bridge, but then set
	  timeoutms to 0 by default. If the timeoutms is not provided it is
	  set to -1. By setting timeoutms to 0 it is processed causing a
	  bridging retry loop. (closes issue #15216) Reported by: oxymoron
	  Tested by: dvossel ........

2009-06-02 13:51 +0000 [r198794]  Joshua Colp <jcolp@digium.com>

	* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
	  198791 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 |
	  file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
	  Correct documentation for the register line, specifically where
	  the domain should be specified. (closes issue #14367) Reported
	  by: Nick_Lewis ........

2009-06-01 21:04 +0000 [r198730]  Russell Bryant <russell@digium.com>

	* channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009)
	  | 2 lines Tell the IAX2 parser about more control frame types.
	  ........

2009-06-01 18:44 +0000 [r198629]  Tilghman Lesher <tlesher@digium.com>

	* /, contrib/scripts/meetme.sql: Merged revisions 198626 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01
	  Jun 2009) | 2 lines Add information for new meetme realtime
	  fields ........

2009-05-31 17:53 +0000 [r198471]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_strings.c: Merged revisions 198470 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009)
	  | 2 lines Fix documentation for FIELDQTY. ........

2009-05-31 01:48 +0000 [r198440]  Eliel C. Sardanons <eliels@gmail.com>

	* /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) |
	  11 lines Avoid a crash when res_timing_dahdi is unloaded but
	  wasn't properly loaded. if dahdi_test_timer() fails,
	  timing_funcs_handle remains NULL causing a crash when calling
	  ast_unregister_timing_interface() with a NULL pointer. (closes
	  issue #15234) Reported by: eliel Patches: timing_dahdi1.diff
	  uploaded by eliel (license 64) ........

2009-05-31 01:21 +0000 [r198436]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c, /: Merged revisions 198312 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009)
	  | 12 lines Merged revisions 198311 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009)
	  | 5 lines Fix a crash that occurred when MWI SMDI messages
	  expired. (closes issue #14561) Reported by: cmoss28 ........
	  ................

2009-05-30 20:22 +0000 [r198297-198397]  Sean Bright <sean.bright@gmail.com>

	* res/res_jabber.c, /: Merged revisions 198375 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 |
	  seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13
	  lines Properly terminate the receive buffer before sending to
	  iksemel. aji_io_recv takes the maximum number of bytes to read
	  (instead of the total buffer size), so we have to subtract 1 from
	  our buffer size. Without this, when we receive packets that are
	  larger than our buffer, iksemel will choke and things get wonky.
	  (closes issue #15232) Reported by: lp0 Patches:
	  05302009_res_jabber.c.patch uploaded by seanbright (license 71)
	  Tested by: seanbright, lp0 ........

	* res/res_jabber.c, /: Merged revisions 198371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May
	  2009) | 19 lines Merged revisions 198370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May
	  2009) | 12 lines Properly terminate AMI JabberSend response
	  messages. The response message (either Error or Success) needs an
	  extra trailing \r\n after the fields to inform the client that
	  the message is complete. (closes issue #14876) Reported by: srt
	  Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
	  (license 71) asterisk_14876.patch uploaded by srt (license 378)
	  trunk-14876-2.diff uploaded by phsultan (license 73) ........
	  ................

	* apps/app_dial.c, /: Merged revisions 198285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May
	  2009) | 15 lines Merged revisions 198251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May
	  2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we
	  treat a missing one. (closes issue #15056) Reported by:
	  p_lindheimer Patches: 05292009_bug15056.diff uploaded by
	  seanbright (license 71) Tested by: p_lindheimer ........
	  ................

2009-05-30 02:35 +0000 [r198250]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 |
	  file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
	  When removing all packets from a dialog we also need to free the
	  data if present. ........

2009-05-29 23:05 +0000 [r198148-198188]  Russell Bryant <russell@digium.com>

	* /, configs/modules.conf.sample: Merged revisions 198186 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29
	  May 2009) | 2 lines Suggesting that only a single timing module
	  be loaded is no longer necessary. ........

	* /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009)
	  | 2 lines Improve handling of trying to ACK too many timer
	  expirations. ........

	* /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009)
	  | 38 lines Resolve issues with choppy sound when using
	  res_timing_pthread. The situation that caused this problem was
	  when continuous mode was being turned on and off while a rate was
	  set for a timing interface. A very easy way to replicate this bug
	  was to do a Playback() from behind a Local channel. In this
	  scenario, a rate gets set on the channel for doing file playback.
	  At the same time, continuous mode gets turned on and off about
	  every 20 ms as frames get queued on to the PBX side channel from
	  the other side of the Local channel. Essentially, this module
	  treated continuous mode and a set rate as mutually exclusive
	  states for the timer to be in. When I dug deep enough, I observed
	  the following pattern: 1) Set timer to tick every 20 ms. 2) Wait
	  almost 20 ms ... 3) Continuous mode gets turned on for a queued
	  up frame 4) Continuous mode gets turned off 5) The timer goes
	  back to its tick per 20 ms. state but starts counting at 0 ms. 6)
	  Goto step 2. Sometimes, res_timing_pthread would make it 20 ms
	  and produce a timer tick, but not most of the time. This is what
	  produced the choppy sound (or sometimes no sound at all). Now,
	  the module treats continuous mode and a set rate as completely
	  independent timer modes. They can be enabled and disabled
	  independently of each other and things work as expected. (closes
	  issue #14412) Reported by: dome Patches: issue14412.diff.txt
	  uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt
	  uploaded by russell (license 2) Tested by: DennisD, russell
	  ........

2009-05-29 19:26 +0000 [r198111]  Eliel C. Sardanons <eliels@gmail.com>

	* CREDITS, /: Merged revisions 198083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 |
	  eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
	  Apply anti-spam obfuscation to an email address. ........

2009-05-29 19:14 +0000 [r198075]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged
	  revisions 198072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May
	  2009) | 21 lines Merged revisions 198068 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May
	  2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as
	  the default CDR disposition. This change also involves the
	  addition of an AST_CDR_FLAG_ORIGINATED flag that is used on
	  originated channels to distinguish: them from dialed channels.
	  (closes issue #12946) Reported by: meral Patches: null-cdr2.diff
	  uploaded by mnicholson (license 96) Tested by: mnicholson,
	  dbrooks (closes issue #15122) Reported by: sum Tested by: sum
	  ........ ................

2009-05-29 18:40 +0000 [r198066]  Joshua Colp <jcolp@digium.com>

	* /, main/file.c: Merged revisions 198064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 |
	  file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix
	  a memory leak of the write buffer when writing a file. ........

2009-05-29 18:18 +0000 [r198008]  Sean Bright <sean.bright@gmail.com>

	* Makefile, /: Merged revisions 198000 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May
	  2009) | 15 lines Merged revisions 197998 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May
	  2009) | 8 lines Fix 'make config' target for Slackware. There was
	  a missing semi-colon after the echo statement in the Makefile
	  that was causing problems for some users. Fix suggested by
	  reporter. (closes issue #15225) Reported by: pdavis ........
	  ................

2009-05-29 16:29 +0000 [r197994]  Russell Bryant <russell@digium.com>

	* /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009)
	  | 2 lines Trim trailing whitespace so that I can work on this bug
	  without it bothering me. :-) ........

2009-05-28 23:54 +0000 [r197894]  Leif Madsen <lmadsen@digium.com>

	* apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009)
	  | 8 lines Update documentation in MixMonitor. Updated the
	  MixMonitor documentation for the 'b' option so that it is more
	  obvious that you must not optimize away the Local channel when
	  using this option. (closes issue #14829) Reported by: licedey
	  Tested by: mmichelson, licedey, lmadsen ........

2009-05-28 18:50 +0000 [r197703]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2
	  lines Fix a bug where the trunkmtu setting was not set to the
	  default value of 1240 on load but was on reload. ........

2009-05-28 16:15 +0000 [r197625]  Eliel C. Sardanons <eliels@gmail.com>

	* /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) |
	  19 lines Merged revisions 197562 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) |
	  13 lines Use the address we already know when reloading a peer
	  with nat=yes. If we already have an address for a peer, and we
	  are reloading the sip configuration, try to use that address to
	  contact the peer, instead of getting it from the Contact. (closes
	  issue #15194) Reported by: ibc Patches: sip.patch uploaded by
	  eliel (license 64) Tested by: manwe ........ ................

2009-05-28 15:44 +0000 [r197548-197619]  Mark Michelson <mmichelson@digium.com>

	* main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
	  Merged revisions 197606 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May
	  2009) | 22 lines Recorded merge of revisions 197588 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu,
	  28 May 2009) | 16 lines Allow for media to arrive from an
	  alternate source when responding to a reinvite with 491. When we
	  receive a SIP reinvite, it is possible that we may not be able to
	  process the reinvite immediately since we have also sent a
	  reinvite out ourselves. The problem is that whoever sent us the
	  reinvite may have also sent a reinvite out to another party, and
	  that reinvite may have succeeded. As a result, even though we are
	  not going to accept the reinvite we just received, it is
	  important for us to not have problems if we suddenly start
	  receiving RTP from a new source. The fix for this is to grab the
	  media source information from the SDP of the reinvite that we
	  receive. This information is passed to the RTP layer so that it
	  will know about the alternate source for media. Review:
	  https://reviewboard.asterisk.org/r/252 ........ ................

	* main/audiohook.c, apps/app_chanspy.c, /,
	  include/asterisk/audiohook.h: Merged revisions 197543 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500
	  (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May
	  2009) | 21 lines Add flags to chanspy audiohook so that audio
	  stays in sync. There are two flags being added to the chanspy
	  audiohook here. One is the pre-existing
	  AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
	  the read and write slinfactories on the audiohook do not skew
	  beyond a certain tolerance. In addition, there is a new audiohook
	  flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
	  we do not allow for a slinfactory to build up a substantial
	  amount of audio before flushing it. For this particular issue,
	  this means that the person spying on the call will hear the
	  conversations in real time with very little delay in the audio.
	  (closes issue #13745) Reported by: geoffs Patches: 13745.patch
	  uploaded by mmichelson (license 60) Tested by: snblitz ........
	  ................

2009-05-28 14:56 +0000 [r197471-197542]  Joshua Colp <jcolp@digium.com>

	* /, main/utils.c: Merged revisions 197538 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 |
	  file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix
	  a bug in stringfields where it did not actually free the pools of
	  memory. (closes issue #15074) Reported by: pj ........

	* /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) |
	  15 lines Merged revisions 197466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8
	  lines Fix a bug where the flag indicating the presence of rport
	  would get overwritten by the nat setting. The presence of rport
	  is now stored as a separate flag. Once the dialog is setup and
	  authenticated (or it passes through unauthenticated) the proper
	  nat flag is set. (closes issue #13823) Reported by: dimas
	  ........ ................

2009-05-28 11:40 +0000 [r197441]  Gavin Henry <ghenry@suretecsystems.com>

	* contrib/scripts/asterisk.ldap-schema,
	  contrib/scripts/asterisk.ldif, doc/ldap.txt,
	  configs/res_ldap.conf.sample: issue #15155 and issue #15156 from
	  trunk

2009-05-27 23:49 +0000 [r197375]  Tilghman Lesher <tlesher@digium.com>

	* /, main/xml.c: Merged revisions 197374 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 |
	  tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
	  Revert commit 192032. This define is needed on Mac OS X. ........

2009-05-27 22:23 +0000 [r197336]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May
	  2009) | 3 lines Ensure that this header includes xmldoc.h, since
	  it depends on it. ........

2009-05-27 20:11 +0000 [r197263]  Sean Bright <sean.bright@gmail.com>

	* Makefile, /: Merged revisions 197260 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 |
	  seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6
	  lines Use bash explicitly when calling build_tools/mkpkgconfig
	  from the Makefile. Since we use bashisms in
	  build_tools/mkpkgconfig, we should call on bash explicitly when
	  running from the Makefile, otherwise we get errors during a 'make
	  install.' (closes issue #15209) Reported by: seandarcy ........

2009-05-27 19:30 +0000 [r197247]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_cut.c: Recorded merge of revisions 197209 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500
	  (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009)
	  | 5 lines Use a different determinator on whether to print the
	  delimiter, since leading fields may be blank. (closes issue
	  #15208) Reported by: ramonpeek Patch by me, though inspired in
	  part by a patch from ramonpeek ........ ................

2009-05-27 17:28 +0000 [r197176]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, include/asterisk/channel.h: Fix broken attended
	  transfers The bridge was terminating immediately after the
	  attended transfer was completed. The problem was because upon
	  reentering ast_channel_bridge nexteventts was checked to see if
	  it was set and if so could possibly return AST_BRIDGE_COMPLETE.
	  (closes issue #15183) Reported by: andrebarbosa Tested by:
	  andrebarbosa, tootai, loloski

2009-05-27 16:12 +0000 [r196950-197092]  Sean Bright <sean.bright@gmail.com>

	* configs/smdi.conf.sample, configs/extensions.conf.sample,
	  configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /,
	  configs/vpb.conf.sample: Merged revisions 197089 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May
	  2009) | 6 lines Fix references to /etc/dahdi/system.conf and
	  /etc/asterisk/chan_dahdi.conf in the sample configuration files.
	  (closes issue #15207) Reported by: seandarcy ........

	* /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May
	  2009) | 9 lines Display an error message when chan_alsa fails to
	  load due to a missing or inaccessible configuration file. Before
	  this change, when chan_alsa failed to load due to a missing or
	  inaccessible configuration file, no message would be displayed.
	  With this change, when chan_alsa fails to load due to a missing
	  or inaccessible configuration file, a message will be displayed.
	  (closes issue #14760) Reported by: Nick_Lewis Patches:
	  chan_alsa.c-confload.patch uploaded by Nick (license 657)
	  ........

	* main/xmldoc.c, /: Merged revisions 196948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 |
	  seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8
	  lines Reset the terminal to the correct fg/bg after XML
	  documenation is rendered. (closes issue #15200) Reported by:
	  ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright
	  (license 71) Tested by: ajohnson ........

	* main/manager.c, /: Merged revisions 196945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 |
	  seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13
	  lines Add ActionID to CoreShowChannel event. There is
	  inconsistency in how we handle manager responses that are lists
	  of items and, unfortunately, third parties have come to rely on
	  ActionID being on every event within those lists instead of just
	  keeping track of the ActionID for the current response. This
	  change makes CoreShowChannels include the ActionID with each
	  CoreShowChannel event generated as a result of it being called.
	  (closes issue #15001) Reported by: sum Patches:
	  patchactionid2.patch uploaded by sum (license 766) ........

2009-05-26 22:44 +0000 [r196870-196949]  Russell Bryant <russell@digium.com>

	* /, autoconf/ast_check_osptk.m4 (added), configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Merged revisions
	  196946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 |
	  russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines
	  Update configure script to check for OSP toolkit 3.5.0. (closes
	  issue #14988) Reported by: tzafrir Patches: configure.ac.diff
	  uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded
	  by homesick (license 91) ........

	* /, res/res_convert.c: Merged revisions 196843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009)
	  | 16 lines Merged revisions 196826 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009)
	  | 9 lines Resolve a file handle leak. The frames here should have
	  always been freed. However, out of luck, there was never any
	  memory leaked. However, after file streams became reference
	  counted, this code would leak the file stream for the file being
	  read. (closes issue #15181) Reported by: jkroon ........
	  ................

2009-05-26 16:39 +0000 [r196793]  Sean Bright <sean.bright@gmail.com>

	* apps/app_queue.c, /: Merged revisions 196792 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 |
	  seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2
	  lines Add a missing unref for queues in handle_statechange.
	  ........

2009-05-26 13:47 +0000 [r196661-196724]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 |
	  file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix
	  a bug where the sip unregister CLI command did not completely
	  unregister the peer. (closes issue #15118) Reported by: alecdavis
	  Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis
	  (license 585) ........

	* contrib/scripts/safe_asterisk, /: Merged revisions 196658 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue,
	  26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7
	  lines Remove some bash specific stuff from safe_asterisk. (closes
	  issue #10812) Reported by: paravoid Patches:
	  safe_asterisk_bashism.diff uploaded by tzafrir (license 46)
	  ........ ................

2009-05-23 05:29 +0000 [r196487]  Moises Silva <moises.silva@gmail.com>

	* channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1
	  line set MFCR2_CATEGORY just when starting the pbx ........

2009-05-22 21:59 +0000 [r196452]  David Vossel <dvossel@digium.com>

	* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
	  196416 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 |
	  dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
	  SIP set outbound transport type from Registration In sip.conf the
	  transport option allows for the configuration of what transport
	  types (udp, tcp, and tls) a peer will accept, but only the first
	  type listed was used for outbound connections. This patch changes
	  this. Now the default transport type is only used until the peer
	  registers. When registration takes place the transport type is
	  parsed out of the Contact header. If the Contact header's
	  transport type is equal to one that the peer supports, the peer's
	  default transport type for outbound connections is set to match
	  the Contact header's type. If the Contact header's transport type
	  is not present, then the peer's default transport type is set to
	  match the one the peer registered with. When a peer unregisters
	  or the registration expires, the default transport type for that
	  peer is reset. (closes issue #12282) Reported by: rjain Patches:
	  reg_patch_1.diff uploaded by dvossel (license 671) Tested by:
	  dvossel (closes issue #14727) Reported by: pj Patches:
	  reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj,
	  dvossel Review: https://reviewboard.asterisk.org/r/249/ ........

2009-05-22 19:48 +0000 [r196378]  Eliel C. Sardanons <eliels@gmail.com>

	* /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 |
	  eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines
	  Unregister every registered application by MiniVM. The MinivmMWI
	  application was not being unregistered on unload and we were not
	  able to load again the module or reload it. (closes issue #15174)
	  Reported by: junky Patches: unregister_minivm_mwi.diff uploaded
	  by junky (license 177) ........

2009-05-22 13:59 +0000 [r196120]  Joshua Colp <jcolp@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri,
	  22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5
	  lines Fix a bug where using immediate with mISDN caused a cause
	  code of 16 to get sent back instead of 1 if the 's' extension did
	  not exist. (closes issue #12286) Reported by: lmamane ........
	  ................

2009-05-21 19:15 +0000 [r196000]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500
	  (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009)
	  | 14 lines Sign problem calculating timestamp for iax frame leads
	  to no audio on the receiving peer. There are rare cases in which
	  a frame's delivery timestamp is slightly less than the iax2_pvt's
	  offset. This causes the pvt's timestamp to be a small negative
	  number, but since the timestamp value is unsigned it looks like a
	  huge positive number. This patch checks for this negative case
	  and sets the ms to zero. A similar check is already done right
	  below this one in the 'else' statement. (closes issue #15032)
	  Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp
	  uploaded by guillecabeza (license 380) Tested by: guillecabeza
	  (closes issue #14216) Reported by: Andrey Sofronov ........
	  ................

2009-05-21 15:57 +0000 [r195883]  Matthew Nicholson <mnicholson@digium.com>

	* main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500
	  (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May
	  2009) | 13 lines This commit prevents cdr records with
	  AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated
	  in certain cases. This is accomplished by adding two functions to
	  update the answer time and disposition of calls that checks for
	  the proper lock flags. These functions are used in the
	  ast_bridge_call() function so that ForkCDR(A) calls are
	  respected. This patch also modifies the way ast_bridge_call()
	  chooses the cdr record to base the bridged_cdr on. Previously the
	  first unlocked cdr record would be chosen, now instead the first
	  cdr record is chosen and forked cdr records are moved to the
	  bridge_cdr. This allows the original cdr record and any forked
	  cdr records to be properly updated with answer and end times.
	  (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes
	  issue #14744) Reported by: deepesh ........ ................

2009-05-20 23:31 +0000 [r195842]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c, /: Merged revisions 195839 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 |
	  tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines
	  If a variable had a blank value upon the initial setting, then it
	  would do nothing. Identified by Dmitry Andrianov via private
	  email, fixed by me. ........

2009-05-20 17:35 +0000 [r195639-195707]  Joshua Colp <jcolp@digium.com>

	* /, main/features.c: Merged revisions 195698 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) |
	  12 lines Merged revisions 195688 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5
	  lines Fix some code that wrongly assumed a pointer would always
	  be non-NULL when dealing with CDRs after a bridge. (closes issue
	  #15079) Reported by: barryf ........ ................

	* /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) |
	  12 lines Merged revisions 195635 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5
	  lines Fix a bug where the MeetMe option 'D' did not actually
	  prompt for the pin. (closes issue #15050) Reported by: pmhaddad
	  ........ ................

2009-05-19 20:19 +0000 [r195531]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500
	  (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009)
	  | 7 lines Ensure thread keys are initialized before attempting to
	  access them. (closes issue #14889) Reported by: jaroth Patches:
	  app_voicemail.c.patch uploaded by msirota (license 758) Tested
	  by: msirota, BlargMaN ........ ................

2009-05-19 14:49 +0000 [r195452]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) |
	  14 lines Merged revisions 195448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7
	  lines Fix a bug where direct RTP setup would partially occur even
	  when disabled if the calling channel was answered. (issue #13545)
	  Reported by: davidw (issue #14244) Reported by: mbnwa ........
	  ................

2009-05-18 21:25 +0000 [r195405]  Eliel C. Sardanons <eliels@gmail.com>

	* main/manager.c, /: Merged revisions 195369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 |
	  eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines
	  Fix the CLI command 'manager show command' documentation and
	  functionality. The CLI command 'manager show command' supports
	  passing multiple action names in the same line, but it was not
	  allowing that because of a incorrect check in the argumentes
	  counter. Also the documentation was updated to show that this
	  usage of the command is possible. ........

2009-05-18 20:55 +0000 [r195359-195373]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c,
	  apps/app_voicemail.c, res/res_smdi.c, /,
	  include/asterisk/monitor.h: Merged revisions 195370 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500
	  (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009)
	  | 8 lines Add a similar dependency on SMDI for voicemail as
	  already exists for ADSI. (closes issue #14846) Reported by: pj
	  Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman
	  (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by
	  tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt
	  uploaded by tilghman (license 14) ........ ................

	* main/asterisk.c, /: Merged revisions 195320 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 |
	  tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines
	  Move the spawn of astcanary down, until after the call to
	  daemon(3). This avoids possible conflicts with the internal
	  implementation of daemon(3). (closes issue #15093) Reported by:
	  tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by
	  tilghman (license 14) Tested by: tzafrir ........

2009-05-18 19:01 +0000 [r195319]  Mark Michelson <mmichelson@digium.com>

	* apps/app_externalivr.c, /: Merged revisions 195316 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May
	  2009) | 18 lines Fix externalivr's setvariable command so that it
	  properly sets multiple variables. The command had a for loop that
	  was guaranteed to only execute once since the continuation
	  operation of the loop would set the input buffer NULL. I rewrote
	  the loop so that its operation was more obvious, and it would set
	  multiple variables correctly. I also reduced stack space required
	  for the function, constified the input string, and modified the
	  function so that it would not modify the input string while I was
	  at it. (closes issue #15114) Reported by: chris-mac Patches:
	  15114.patch uploaded by mmichelson (license 60) Tested by:
	  chris-mac ........

2009-05-18 15:57 +0000 [r195212]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /: Merged revisions 195207 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) |
	  14 lines Merged revisions 195206 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7
	  lines Fix a typo which caused loss of audio when using G729 in
	  some scenarios with a smoother present. (closes issue #15105)
	  Reported by: bamby Patches: process-vad-correctly.diff uploaded
	  by bamby (license 430) ........ ................

2009-05-18 14:54 +0000 [r195164]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged
	  revisions 195162 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 |
	  eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines
	  Warn about the use of the application WaitExten() within a
	  Macro(). Update applications documentation to warn the user about
	  the use of the WaitExten() application within a Macro().
	  Recommend the use of Read() instead. (closes issue #14444)
	  Reported by: ewieling ........

2009-05-18 14:00 +0000 [r195099]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 195096 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) |
	  12 lines Merged revisions 195095 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5
	  lines Fix a bug where the codecs of the called party leg were not
	  properly sent back to the caller call leg when reinvited. (closes
	  issue #13569) Reported by: bkw918 ........ ................

2009-05-18 13:50 +0000 [r195093-195094]  Eliel C. Sardanons <eliels@gmail.com>

	* /, main/xml.c: Merged revisions 195075 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 |
	  eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do
	  not avoid loading the XML documentation if not XInclude
	  substitution is done. ........

	* doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions
	  194982 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 |
	  eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines
	  Allow to include sections of other parts of the xml
	  documentation. Avoid duplicating xml documentation by allowing to
	  include other parts of the xml documentation using XInclude.
	  Example: <xi:include
	  xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
	  (Insert this line to include the synopsis of the CHANNEL function
	  xml documentation). It is also possible to include documentation
	  from other files in the 'documentation/' directory using the
	  href="" attribute inside a xinclude element. (closes issue
	  #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling
	  ........

2009-05-18 13:39 +0000 [r195092]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 |
	  file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix
	  a bug where specifying an empty outboundproxy would cause packets
	  to get sent to ourself. (closes issue #15106) Reported by:
	  timeshell ........

2009-05-18 13:14 +0000 [r195024]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 195021 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009)
	  | 12 lines Recorded merge of revisions 195020 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009)
	  | 5 lines Don't try to unlock a bogus channel. (closes issue
	  #15144) Reported by: cristiandimache ........ ................

2009-05-16 18:43 +0000 [r194946]  Eliel C. Sardanons <eliels@gmail.com>

	* main/pbx.c, /: Merged revisions 194945 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 |
	  eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines
	  Fix a missing unlock in case of error, and a missing free().
	  Always free the allocated memory for a string field, because we
	  are always using it (not only when xmldocs are enabled). Also if
	  there is an error allocating memory for the string field remember
	  to unlock the list of registered applications, before returning.
	  ........

2009-05-15 22:48 +0000 [r194836-194877]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500
	  (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009)
	  | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to
	  terminate invalid registrations. Instead it sent another REGAUTH
	  if the authentication challenge failed. This caused a loop of
	  REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001)
	  (closes issue #14867) Reported by: aragon Tested by: dvossel
	  (closes issue #14717) Reported by: mobeck Patches:
	  regauth_loop_update_patch.diff uploaded by dvossel (license 671)
	  Tested by: dvossel ........ ................

	* channels/chan_iax2.c, channels/iax2-parser.c,
	  channels/iax2-parser.h, /, channels/iax2.h: Merged revisions
	  194833 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009)
	  | 24 lines Merged revisions 194557,194685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009)
	  | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue
	  where people are reporting "Ghost" channels in their 'iax2 show
	  channels' output. The confusion is caused by channels being
	  listed as "(NONE)" with format "unknown". These are not channels
	  of coarse. They are usually just pending registration or poke
	  requests, but it is confusing output. To help make sense of this
	  I have added two columns to 'iax2 show channels'. One shows the
	  first message which started the transaction, and the second shows
	  the last message sent by either side of the call. This helps
	  diagnose why the entry exists and why it may not go away. (closes
	  issue #14207) Reported by: clive18 Review:
	  https://reviewboard.asterisk.org/r/246/ ........ r194685 |
	  dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
	  Update to previous IAX2 "Ghost" Channels patch. Fixed some
	  comments made on reviewboard for the previous patch. (issue
	  #14207) ........ ................

2009-05-15 18:44 +0000 [r194717-194768]  Russell Bryant <russell@digium.com>

	* configs/logger.conf.sample, /: Merged revisions 194765 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r194765 | russell | 2009-05-15 13:43:42 -0500
	  (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009)
	  | 2 lines Fix some spelling fail. ........ ................

	* /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged
	  revisions 194722 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 |
	  russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines
	  Shuttle some bits around to address some gain issues with G.722.
	  (closes AST-209) ........

	* codecs/Makefile, codecs/g722/Makefile (removed), /: Merged
	  revisions 194718 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 |
	  russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines
	  Further simplify codec_g722 build. ........

	* codecs/Makefile, /: Merged revisions 194714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 |
	  russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines
	  Actually force running make for g722. ........

2009-05-15 13:47 +0000 [r194650]  Michiel van Baak <michiel@vanbaak.info>

	* CREDITS, /: Merged revisions 194649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 |
	  mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines
	  add eliel ........

2009-05-15 13:42 +0000 [r194648]  Eliel C. Sardanons <eliels@gmail.com>

	* doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May
	  2009) | 16 lines Allow to specify an enumlist inside an enum. It
	  was not possible to use an enumlist inside an enum: <enumlist>
	  <enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist>
	  Now we will be able to insert as many levels as we want. (closes
	  issue #15112) Reported by: lmadsen ........

2009-05-14 22:31 +0000 [r194545]  Kevin P. Fleming <kpfleming@digium.com>

	* /: Merged revisions 194520 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May
	  2009) | 9 lines Merged revisions 194509 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May
	  2009) | 1 line Update URL to Reviewboard ........
	  ................

2009-05-14 22:23 +0000 [r194510]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May
	  2009) | 30 lines Merged revisions 194484 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May
	  2009) | 24 lines Fix a race condition where a reinvite could
	  trigger a 482 response. The loop detection/spiral detection code
	  in chan_sip used the owner channel's state as a criterion for
	  determining if the incoming INVITE is a looped request. The
	  problem with this is that the INVITE-handling code happens in a
	  different thread than the thread that marks the owner channel as
	  being up. As a result, if a reinvite were to come in very
	  quickly, say from another Asterisk on the same LAN, it was
	  possible for the reinvite to arrive before the owner channel had
	  been set to the up state. This patch corrects the problem by
	  using the invitestate of the sip_pvt instead, since that can be
	  guaranteed to be set correctly by the time the reinvite arrives.
	  Since there is a switch statement further in the INVITE-handling
	  code, the AST_STATE_RINGING state also checks the invitestate of
	  the sip_pvt in case we should actually be treating the channel as
	  if it were up already. (closes issue #12215) Reported by: jpyle
	  Patches: 12215_confirmed.patch uploaded by mmichelson (license
	  60) Tested by: lmadsen ........ ................

2009-05-14 17:07 +0000 [r194437]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 |
	  file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix
	  a bug where the 'T' option to Meetme did not work. (closes issue
	  #15031) Reported by: Stochastic (closes issue #13801) Reported
	  by: justdave ........

2009-05-14 16:23 +0000 [r194431]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 194430 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 |
	  tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines
	  If the timing ended on a zero, then we would loop forever.
	  (closes issue #14983) Reported by: teox Patches:
	  20090513__issue14983.diff.txt uploaded by tilghman (license 14)
	  Tested by: teox ........

2009-05-13 13:42 +0000 [r194213]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 194209 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) |
	  18 lines Merged revisions 194208 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) |
	  11 lines Fix RFC2833 issues with DTMF getting duplicated and with
	  duration wrapping over. (closes issue #14815) Reported by:
	  geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
	  Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
	  #14460) Reported by: moliveras Tested by: moliveras ........
	  ................

2009-05-13 00:54 +0000 [r194141]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 194138 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009)
	  | 14 lines Merged revisions 194137 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009)
	  | 7 lines Fix logic for how to proceed with a single digit
	  extension. (closes issue #15091) Reported by: andrew Patches:
	  20090512__issue15091.diff.txt uploaded by tilghman (license 14)
	  Tested by: andrew ........ ................

2009-05-12 22:48 +0000 [r194059]  Matthew Nicholson <mnicholson@digium.com>

	* apps/app_queue.c, /: Merged revisions 194057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May
	  2009) | 22 lines Merged revisions 194028 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May
	  2009) | 16 lines This change modifies app_queue to properly
	  generate CDR records in failure situations. This involves setting
	  a proper cdr disposition coresponding to the given failure
	  condition and ensuring the proper information is stored in the
	  cdr record. (closes issue #13691) Reported by: dferrer Tested by:
	  mnicholson (closes issue #13637) Reported by: atis Tested by:
	  atis ........ ................

2009-05-12 20:51 +0000 [r193962]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 |
	  mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18
	  lines Update spiral support in trunk and 1.6.X to match what is
	  in 1.4. In 1.4, a SIP spiral is treated the same way as a call
	  forward. This works much better than what is currently in trunk
	  and 1.6.X. The code in trunk and 1.6.X did not create a new call
	  to the recipient of the spiral, instead trying to continue the
	  same call. In addition to just being plain wrong, this also had
	  the side effect of only being able to spiral calls to other SIP
	  channels. With this in place, as long as call forwards are
	  honored, SIP spirals will work properly. This means that it will
	  work for outbound calls made by the Queue, Dial, and Page
	  applications. For originated calls and spool calls, however, the
	  spiral will not work properly until a generic call forward
	  mechanism is introduced into Asterisk. (relates to issue #13630)
	  ........

2009-05-12 20:42 +0000 [r193823-193959]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500
	  (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009)
	  | 6 lines Avoid initializing routines if the authentication
	  fails. Fixes a crash (RR) issue. (closes issue #14508) Reported
	  by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by
	  tiziano (license 377) ........ ................

	* apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009)
	  | 2 lines Convert a THREADSTORAGE object into a simple malloc'd
	  object (as suggested by Russell on -dev) ........

	* apps/app_voicemail.c, /: Recorded merge of revisions 193756 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500
	  (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009)
	  | 18 lines Move 300 bytes around on the stack, to make more room
	  for an extension buffer. This allows more concurrent extensions
	  to be copied for a single voicemail, without creating a
	  possibility of upsetting existing users, where a dialplan could
	  run out of stack space where it had run fine before.
	  Alternatively, we could have allocated off the heap, but that is
	  a larger change and would have increased the chance for
	  instability introduced by this change. This is really solved
	  starting in 1.6.0.11, as the use of an ast_str buffer allows an
	  unlimited number of extensions (up to available memory). We
	  additionally create a new warning message when the buffer length
	  is exceeded, permitting administrators to see an issue after the
	  fact, whereas previously the list was silently truncated. (closes
	  issue #14739) Reported by: p_lindheimer Patches:
	  20090417__bug14739.diff.txt uploaded by tilghman (license 14)
	  Tested by: p_lindheimer ........ ................

2009-05-11 22:12 +0000 [r193719]  Russell Bryant <russell@digium.com>

	* /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009)
	  | 12 lines Fix some timer state corruption. In res_timer_timerfd,
	  handle the case that set_rate gets called while a timer is still
	  in continuous mode. In this case, we want to remember the
	  configured rate, but not actually set it until continuous mode
	  has been disabled. Thanks to dvossel for finding and helping to
	  debug the problem. (closes issue #15080) Reported by: dvossel
	  Tested by: dvossel ........

2009-05-11 19:17 +0000 [r193617]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500
	  (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009)
	  | 12 lines Sent wrong message to clear a call we started if the
	  other end has not responed yet. In the state MISDN_CALLING (i.e.
	  SETUP was sent but no answer has arrived yet), it is not allowed
	  to clear the call with RELEASE_COMPLETE. It must be cleared with
	  DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a
	  SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches:
	  chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862
	  ........ ................

2009-05-11 18:59 +0000 [r193612]  Leif Madsen <lmadsen@digium.com>

	* /, funcs/func_channel.c: Update CHANNEL(transfercapabilities)
	  documentation. (closes issue #15073) Reported by: pkempgen
	  Patches: 20090511__issue15073__trunk.diff.txt uploaded by
	  tilghman (license 14)

2009-05-10 17:08 +0000 [r193503]  Joshua Colp <jcolp@digium.com>

	* main/bridging.c, /: Merged revisions 193502 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 |
	  file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix
	  a bug where receiving a control frame of subclass -1 would cause
	  certain channels to get hung up. ........

2009-05-09 11:33 +0000 [r193462]  Russell Bryant <russell@digium.com>

	* include/asterisk/event.h, /: Merged revisions 193461 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009)
	  | 2 lines Minor documentation update for ast_event_queue().
	  ........

2009-05-08 20:52 +0000 [r193390]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 |
	  dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
	  TCP not matching valid peer. find_peer() does not find a valid
	  peer when using pvt->recv as the sockaddr_in argument. Because of
	  the way TCP works, the port number in pvt->recv is not what we're
	  looking for at all. There is currently only one place that
	  find_peer searches for a peer using the sockaddr_in argument. If
	  the peer is not found after using pvt->recv (works for UDP since
	  the port number will be correct), a temp sockaddr_in struct is
	  made using the Contact header in the sip_request. This has the
	  correct port number in it. Review:
	  http://reviewboard.digium.com/r/236/ ........

2009-05-08 19:51 +0000 [r193350]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 193349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 |
	  mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12
	  lines Reset the members' call counts when resetting queue
	  statistics. This helps to prevent odd scenarios where a queue
	  will claim to have taken 0 calls, but the members appear to have
	  taken a non-zero amount. (closes issue #15068) Reported by: sum
	  Patches: patchreset.patch uploaded by sum (license 766) Tested
	  by: sum ........

2009-05-08 15:36 +0000 [r193336]  Sean Bright <sean.bright@gmail.com>

	* funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May
	  2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate
	  CLI completion. ........

2009-05-08 14:55 +0000 [r193266]  David Vossel <dvossel@digium.com>

	* channels/misdn_config.c, /: Merged revisions 193263 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500
	  (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009)
	  | 9 lines "misdn show config" segfaults asterisk, if no MSN lists
	  (closes issue #14976) Reported by: alecdavis Patches:
	  misdn_config.diff.txt uploaded by alecdavis (license 585) Tested
	  by: alecdavis, FabienToune ........ ................

2009-05-08 14:12 +0000 [r193197]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/logger.conf.sample, /, main/logger.c: Merged revisions
	  193194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May
	  2009) | 13 lines Merged revisions 193193 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May
	  2009) | 7 lines Make absolute paths for logger channels work
	  properly (Note: This is not a new feature, it was previously
	  undocumented and broken.) The Asterisk logger has a feature to
	  support absolute pathnames for logger channels, but the code
	  implementing the feature was broken. This has been fixed, and the
	  absolute path feature is now documented in the sample
	  logger.conf. ........ ................

2009-05-07 23:44 +0000 [r193123]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 193120 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009)
	  | 26 lines Merged revisions 193119 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009)
	  | 19 lines Fix Background within a Macro for FreePBX. If the
	  single digit DTMF is an extension in the specified context, then
	  go there and signal no DTMF. Otherwise, we should exit with that
	  DTMF. If we're in Macro, we'll exit and seek that DTMF as the
	  beginning of an extension in the Macro's calling context. If
	  we're not in Macro, then we'll simply seek that extension in the
	  calling context. Previously, someone complained about the
	  behavior as it related to the interior of a Gosub routine, and
	  the fix (#14011) inadvertently broke FreePBX (#14940). This
	  change should fix both of these situations, but with the possible
	  incompatibility that if a single digit extension does not exist
	  (but a longer extension COULD have matched), it would have
	  previously gone immediately to the "i" extension, but will now
	  need to wait for a timeout. (closes issue #14940) Reported by:
	  p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
	  tilghman (license 14) Tested by: p_lindheimer ........
	  ................

2009-05-07 22:51 +0000 [r193080]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500
	  (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009)
	  | 5 lines Give a more helpful message when an incoming call's
	  dialed extension does not match. Added the dialed extension and
	  context to the chan_misdn messages warning that the dialed number
	  cannot be matched in the dialplan. ........ ................

2009-05-07 17:53 +0000 [r192936-193008]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 |
	  tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines
	  Second result should not contain data from the first result.
	  (closes issue #15039) Reported by: jims Patches:
	  20090506__issue15039.diff.txt uploaded by tilghman (license 14)
	  Tested by: jims ........

	* channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009)
	  | 6 lines Send DTMF frame before playing back audio. (closes
	  issue #14858) Reported by: barryf Patches:
	  20090507__bug14858.diff.txt uploaded by tilghman (license 14)
	  ........

	* /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009)
	  | 17 lines Merged revisions 192932 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009)
	  | 10 lines Eliminate repetition of fullcontact during
	  reconstruction. If the fullcontact field appears in both the
	  sippeers and the sipregs table, then during reconstruction of the
	  field, it will otherwise be doubled. (closes issue #14754)
	  Reported by: Alexei Gradinari Patches:
	  20090506__bug14754.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen ........ ................

2009-05-07  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.6.2.0-beta2

2009-05-06 22:20 +0000 [r192874]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: Merged revisions 192861 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009)
	  | 17 lines Merged revisions 192858 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009)
	  | 10 lines Make ParkedCall application stop execution of the
	  dialplan after hang up Just changed park_exec to always return
	  non-zero. I really wasn't entirely sure at first if this was a
	  bug. Decided it was since it would be surprising when not using
	  ParkedCall in the dialplan to hang up and have dialplan execution
	  continue. (closes issue #14555) Reported by: francesco_r ........
	  ................

2009-05-06 17:57 +0000 [r192813]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) |
	  1 line Make sure that we do not clear the down flag on the BRI
	  during PTMP link transients. Also refix SS7 audio that the early
	  media patch broke. ........

2009-05-06 17:41 +0000 [r192637-192810]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) |
	  10 lines Fix a bug where a timer would be created but not
	  acknowledged. This scenario crept up if chan_iax2 was loaded with
	  no configuration file present. It would create a timer and tell
	  it to go at an interval but the thread that normally acknowledges
	  it would not be created because no configuration file was
	  present. The timer will now be closed if no configuration file is
	  present. (closes issue #15014) Reported by: madkins ........

	* res/res_clialiases.c, /: Merged revisions 192736 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4
	  lines Make the code that prevents an infinite loop from happening
	  into a case insensitive check. (thanks eliel) ........

	* res/res_clialiases.c, /: Merged revisions 192700 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5
	  lines Fix an infinite loop with tab completion of CLI aliases
	  that reference themselves. (closes issue #15020) Reported by:
	  junky ........

	* /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) |
	  14 lines Merged revisions 192633 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7
	  lines Update some old logic to stop both begin and end DTMF
	  frames from reaching the core if rfc2833 is not enabled. (closes
	  issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded
	  by dimas (license 88) ........ ................

2009-05-05 20:02 +0000 [r192528]  Sean Bright <sean.bright@gmail.com>

	* /, static-http/astman.js: Merged revisions 192525 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400
	  (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May
	  2009) | 11 lines Fix Javascript error when using astman.js in
	  Internet Explorer. Internet Explorer (tested with 7.0) does not
	  like trailing commas on constructs like object initializers, so
	  get rid of them to avoid some errors. (closes issue #15026)
	  Reported by: rajnishgiri Patches: bug15026.patch uploaded by
	  seanbright (license 71) Tested by: seanbright ........
	  ................

2009-05-05 18:27 +0000 [r192402-192480]  Joshua Colp <jcolp@digium.com>

	* /, main/features.c: Merged revisions 192462 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) |
	  15 lines Merged revisions 192454 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8
	  lines Fix an incorrect assumption that certain values on the
	  channel will always exist when they may not. The CDR code
	  involved with bridges wrongly assumed that the currently
	  executing application and data values will always exist. It is
	  possible for this to be false when call forwarding is involved.
	  (closes issue #14984) Reported by: gincantalupo ........
	  ................

	* apps/app_followme.c, /: Merged revisions 192430 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) |
	  12 lines Merged revisions 192429 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5
	  lines Fix a bug where the followme application would continue
	  trying numbers after the caller hung up. (closes issue #13624)
	  Reported by: sgenyuk ........ ................

	* /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 |
	  file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines
	  Fix a bug with setting t38pt_udptl at the user or peer level. If
	  an incoming call authenticated as a user or peer and t38pt_udptl
	  was not set to yes in general then no UDPTL session would be
	  present and any T38 related things would fail. This commit
	  changes it so that if after authenticating T38 is enabled but no
	  UDPTL session is present one will be created. (issue AST-215)
	  ........

2009-05-05 13:43 +0000 [r192298-192360]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c:
	  Merged revisions 192357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 |
	  kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5
	  lines Correct some flaws in the memory accounting code for
	  stringfields and ao2 objects Under some conditions, the memory
	  allocation for stringfields and ao2 objects would not have
	  supplied valid file/function names for MALLOC_DEBUG tracking, so
	  this commit corrects that. ........

	* main/astobj2.c, main/datastore.c, main/channel.c, /,
	  include/asterisk/astobj2.h, include/asterisk/datastore.h,
	  include/asterisk/channel.h: Merged revisions 192318 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May
	  2009) | 5 lines Properly account for memory allocated for
	  channels and datastores As in previous commits, when channels are
	  allocated (with ast_channel_alloc) or datastores are allocated
	  (with ast_datastore_alloc) properly account for the memory being
	  owned by the caller, instead of the allocator function itself.
	  ........

	* include/asterisk/stringfields.h, /, main/utils.c: Merged
	  revisions 192279 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 |
	  kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5
	  lines Ensure that string pools allocated to hold stringfields are
	  properly accounted in MALLOC_DEBUG mode This commit modifies the
	  stringfield pool allocator to remember the 'owner' of the
	  stringfield manager the pool is being allocated for, and ensures
	  that pools allocated in the future when fields are populated are
	  owned by that file/function. ........

2009-05-04 22:48 +0000 [r192217]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500
	  (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009)
	  | 11 lines global mohinterpret setting is ignored mohinterpret
	  and mohsuggest global variables were not copied over during
	  build_users and build_peers. (closes issue #14728) Reported by:
	  dimas Patches: v1-14728.patch uploaded by dimas (license 88)
	  Tested by: dimas, dvossel ........ ................

2009-05-04 19:34 +0000 [r192175]  Kevin P. Fleming <kpfleming@digium.com>

	* main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
	  192059 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 |
	  kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5
	  lines Ensure that astobj2 memory allocations are properly
	  accounted for when MALLOC_DEBUG is used This commit ensures that
	  all astobj2 allocated objects are properly accounted for in
	  MALLOC_DEBUG mode by passing down the file/function/line
	  information from the module/function that actually called the
	  astobj2 allocation function. ........

2009-05-04 19:31 +0000 [r192135-192173]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, res/res_agi.c: Merged revisions 192171 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009)
	  | 8 lines Restore 'asyncagi break' command to 1.6.1 and higher.
	  (closes issue #14985) Reported by: nikkk Patches:
	  20090428__bug14985.diff.txt uploaded by tilghman (license 14)
	  20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license
	  14) Tested by: nikkk ........

	* autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04
	  May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes
	  issue #14671) Reported by: Chainsaw Patches:
	  asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by
	  Chainsaw (license 723) ........

2009-05-04 17:45 +0000 [r192097]  Leif Madsen <lmadsen@digium.com>

	* apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 |
	  lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines
	  Commit documentation changes related to issue #14801. (issue
	  #14801) ........

2009-05-04 15:54 +0000 [r192033]  Eliel C. Sardanons <eliels@gmail.com>

	* /, main/xml.c: Merged revisions 192032 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 |
	  eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do
	  not re-define _POSIX_C_SOURCE if it was already defined. ........

2009-05-04 10:01 +0000 [r191958]  Kevin P. Fleming <kpfleming@digium.com>

	* /, configs/modules.conf.sample: Merged revisions 191955 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04
	  May 2009) | 8 lines Ensure that by default only one console
	  channel driver is loaded This configuration file was changed to
	  ensure that only one console channel driver (chan_oss) is loaded
	  by default, but the change would only work if chan_console was
	  not built. Now it will work as expected; if chan_alsa or
	  chan_console are built and installed, they will not be loaded
	  unless explicity requested. ........

2009-05-03 14:06 +0000 [r191885]  Russell Bryant <russell@digium.com>

	* Makefile, /: Merged revisions 191884 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 |
	  russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines
	  Remove unnecessary compiler flag ........

2009-05-02 18:48 +0000 [r191779]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/logger.c: Merged revisions 191775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 |
	  kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5
	  lines Fix an error in queue_log file rotation optimization code
	  This code was copy-and-pasted without properly changing
	  references to event_rotate into queue_rotate, so under some
	  conditions the log rotation would rotate queue_log even though it
	  was not necessary. ........

2009-05-02 15:52 +0000 [r191703]  Sean Bright <sean.bright@gmail.com>

	* main/asterisk.c, /: Merged revisions 191700 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 |
	  seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1
	  line Update copyright year to 2009 ........

2009-05-01 20:02 +0000 [r191554-191563]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009)
	  | 13 lines Merged revisions 191559 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009)
	  | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1.
	  (closes issue #14993) Reported by: BigJimmy Patches: causepatch
	  uploaded by BigJimmy (license 371) ........ ................

	* channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009)
	  | 4 lines Set debug message back to DEBUG level. (closes issue
	  #15007) Reported by: hulber ........

2009-05-01 18:20 +0000 [r191508]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 191489 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009)
	  | 15 lines Merged revisions 191488 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009)
	  | 9 lines Fix DTMF not being sent to other side after a partial
	  feature match This fixes a regression from commit 176701. The
	  issue was that ast_generic_bridge never exited after the feature
	  digit timeout had elapsed, which prevented the queued DTMF from
	  being sent to the other side. This issue was reported to me
	  directly. ........ ................

2009-04-30 17:46 +0000 [r191224-191370]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Merged revisions 191367 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 |
	  tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines
	  Detect eaccess (or euidaccess) before using it. Reported by
	  Andrew Lindh via the -dev list. ........

	* main/asterisk.c, /: Merged revisions 191283 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 |
	  tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11
	  lines Change working directory to / under certain conditions. If
	  backgrounding and no core will be produced, then changing the
	  directory won't break anything; likewise, if the CWD isn't
	  accessible by the current user, then a core wasn't possible
	  anyway. (closes issue #14831) Reported by: chris-mac Patches:
	  20090428__bug14831.diff.txt uploaded by tilghman (license 14)
	  20090430__bug14831.diff.txt uploaded by tilghman (license 14)
	  Tested by: chris-mac ........

	* /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged
	  revisions 191219 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 |
	  tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines
	  Make H.323 compile with FDLEAK detection code enabled ........

2009-04-29 18:40 +0000 [r191139]  David Brooks <dbrooks@digium.com>

	* pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 |
	  dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines
	  Removing crufty code that is no longer necessary. Code cleanup.
	  ........

2009-04-29 08:59 +0000 [r190994]  Russell Bryant <russell@digium.com>

	* main/indications.c, /: Merged revisions 190993 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 |
	  russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines
	  Log an error message if indications.conf is not found. (closes
	  issue #14990) Reported by: tzafrir Patches: indications_err.diff
	  uploaded by tzafrir (license 46) ........

2009-04-29 06:38 +0000 [r190985]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr
	  2009) | 2 lines Updated for OSP Toolkit 3.5. ........

2009-04-28 17:33 +0000 [r190907]  Tilghman Lesher <tlesher@digium.com>

	* doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009)
	  | 2 lines UniqueID column has a maximum size of 150 ........

2009-04-28 14:17 +0000 [r190732-190869]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile, /: Merged revisions 190865 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 |
	  kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5
	  lines Build XML documention from *only* the source files that
	  have docs in them Change the build process so that
	  doc/core-en_US.xml is dependent solely on the source files that
	  have documentation in them, not on all source files. ........

	* /, Makefile.rules: Merged revisions 190861 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 |
	  kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5
	  lines Remove Makefile rules for bison and flex sources We never,
	  ever want these files to processed automatically, because we
	  store the output files in Subversion and users should never need
	  to rebuild them. ........

	* /, configure, include/asterisk/autoconfig.h.in: Merged revisions
	  190725 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr
	  2009) | 13 lines Merged revisions 190721 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr
	  2009) | 7 lines Fix 'inconsistent line endings' when autoconf
	  2.63 is used Attempt to make configure script regeneration 'safe'
	  using autoconf 2.63, which embeds a bare CR into the script, thus
	  making Subversion complain about inconsistent line endings This
	  commit changes the MIME type of the configure script to be
	  'binary' thus making Subversion no longer inspect line endings,
	  and as a bonus 'svn diff' will no longer try to generate diff
	  output for it, which is not generally useful anyway. ........
	  ................

2009-04-27 19:36 +0000 [r190729]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 190726 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 |
	  tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines
	  Don't warn on pipe in the System call. (closes issue #14979)
	  Reported by: pj ........

2009-04-27 19:15 +0000 [r190666]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c, /: Merged revisions 190663 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009)
	  | 22 lines Merged revisions 190661-190662 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009)
	  | 9 lines Resolve a crash in res_smdi when used with chan_dahdi.
	  When chan_dahdi goes to get an SMDI message, it provides no
	  search criteria. It just grabs the next message that arrives.
	  This code was written with the SMDI dialplan functions in mind,
	  since that is now the preferred method of using SMDI. However,
	  this broke support of it being used from chan_dahdi. (closes
	  AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500
	  (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........
	  ................

2009-04-27 16:28 +0000 [r190625]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 190622 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 |
	  mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3
	  lines Update warning message to not have pipes and contain all
	  options. ........

2009-04-23 21:23 +0000 [r190383]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ ........

2009-04-23 20:44 +0000 [r190355]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 190352 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 |
	  tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines
	  Labels are sometimes (most of the time?) NULL for extensions.
	  (closes issue #14895) Reported by: chris-mac Patches:
	  20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
	  Tested by: lmadsen ........

2009-04-23 19:18 +0000 [r190297]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 190287 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu,
	  23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6
	  lines Fix a bug in chan_local glare hangup detection. If both
	  sides of a Local channel were hung up at around the same time it
	  was possible for one thread to destroy the local private
	  structure and have the other thread immediately try to remove the
	  already freed structure from the local channel list. ........
	  ................

2009-04-23 17:47 +0000 [r190253]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 190250 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 |
	  mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9
	  lines Fix reversed behavior of leavewhenempty option in
	  queues.conf. (closes issue #14650) Reported by: alecdavis
	  Patches: 14650.patch uploaded by mmichelson (license 60) Tested
	  by: mmichelson, lmadsen ........

2009-04-22 21:43 +0000 [r190096]  Tilghman Lesher <tlesher@digium.com>

	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/lock.h: Merged revisions 190093 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500
	  (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009)
	  | 7 lines Detect availability of pthread_rwlock_timedwrlock()
	  before using it. (closes issue #14930) Reported by: tilghman
	  Patches: 20090420__bug14930.diff.txt uploaded by tilghman
	  (license 14) Tested by: mvanbaak, tilghman ........
	  ................

2009-04-22 21:18 +0000 [r189997-190066]  Jeff Peeler <jpeeler@digium.com>

	* main/cli.c, funcs/func_groupcount.c, /, main/app.c,
	  include/asterisk/channel.h: Merged revisions 190057 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009)
	  | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to
	  be a bug with old versions of g++ that doesn't allow a structure
	  member to use the name list. Rename list member to group_list in
	  ast_group_info and change the few places it is used. (closes
	  issue #14790) Reported by: stuarth ........

	* channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx,
	  channels/chan_h323.c: Merged revisions 189993 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 |
	  jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
	  Make chan_h323 respect packetization settings and fix small
	  reload issue. Previously, packetization settings were ignored and
	  now they are not. A new config option 'autoframing' has been
	  added to mirror the way chan_sip handles it. Turning on the
	  autoframing option (available both as a global option or per
	  peer) overrides the local settings with the remote packetization
	  settings. Testing was performed with varying packetization levels
	  with the following codecs: ulaw, alaw, gsm, and g729. Also, an
	  unrelated config reload issue has been fixed in the case of the
	  config file not changing. (closes issue #12415) Reported by: pj
	  Patches: 2009012200_h323packetization.diff.txt uploaded by
	  mvanbaak (license 7), modified by me ........

2009-04-22 18:01 +0000 [r189986]  Russell Bryant <russell@digium.com>

	* /, main/features.c: Merged revisions 189951 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 |
	  russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines
	  Fix call parking callback. Pipes -> Commas. ........

2009-04-22 16:04 +0000 [r189816-189914]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009)
	  | 7 lines Do not continue to receive DTMF, when the channel is
	  hungup and about to be destroyed. (closes issue #14858) Reported
	  by: barryf Patches: 20090421__bug14858.diff.txt uploaded by
	  tilghman (license 14) Tested by: barryf ........

	* /, configure, configure.ac: Merged revisions 189813 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009)
	  | 3 lines Detect liblua on SuSE, and add libm for linking for
	  Fedora. (Reported via the -dev list, Subject: Compiling Asterisk
	  with LUA) ........

2009-04-21 20:45 +0000 [r189775]  David Vossel <dvossel@digium.com>

	* /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 |
	  dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
	  Fixes segfault when switching UDP to TCP in sip.conf after
	  reload. If transport in sip.conf is switched from UDP to TCP,
	  Asterisk segfaults right after issuing a sip reload. The problem
	  is the socket type is changed to TCP but the fd may still be
	  present for UDP. Later, when the TCP session should be created or
	  set using an existing one, it isn't because the old file
	  descriptor is still present. Now every time transport is changed
	  during a sip.conf reload, the file descriptor is set to -1,
	  signifying it must be created or found. (closes issue #14727)
	  Reported by: pj Tested by: dvossel Review:
	  http://reviewboard.digium.com/r/229/ ........

2009-04-20 22:11 +0000 [r189540]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009)
	  | 3 lines Use nanosleep instead of poll. This is not just because
	  mmichelson suggested it, but also because Mac OS X puked on my
	  poll(). ........

2009-04-20 21:41 +0000 [r189536]  Terry Wilson <twilson@digium.com>

	* apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r189495 | twilson | 2009-04-20 16:24:34 -0500
	  (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20
	  Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL
	  ........ ................ r189516 | twilson | 2009-04-20 16:29:29
	  -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009)
	  | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is
	  set ........ ................

2009-04-20 21:36 +0000 [r189533]  Sean Bright <sean.bright@gmail.com>

	* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r189464 | seanbright | 2009-04-20 17:09:59 -0400
	  (Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr
	  2009) | 13 lines Properly handle @s within hints in AEL. AEL was
	  not handling the case of a device hint containing an @ symbol,
	  which caused parking hints (e.g. hint(park:exten@context)) to
	  error out the parser. This patch makes AEL treat the @ the same
	  way it treats colon and ampersand now, meaning the characters are
	  included in verbatim. (closes issue #14941) Reported by: bpgoldsb
	  Patches: bug14941.patch uploaded by seanbright (license 71)
	  Tested by: bpgoldsb ........ ................

2009-04-20 17:11 +0000 [r189353]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 |
	  file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
	  Fix a bug with non-UDP connections that caused dialogs to not get
	  freed. This issue crept up because of a reference count issue on
	  non-UDP based dialogs. The dialog reference count was increased
	  when transmitting a packet reliably but never decreased. This
	  caused the dialog structure to hang around despite being unlinked
	  from the dialogs container. (closes issue #14919) Reported by:
	  vrban ........

2009-04-20 14:07 +0000 [r189281]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 189278 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr
	  2009) | 18 lines Merged revisions 189277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr
	  2009) | 12 lines Move the check for chan->fdno == -1 to after the
	  zombie/hangup check. Many users were finding that their hung up
	  channels were staying up and causing 100% CPU usage. (issue
	  #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
	  uploaded by mmichelson (license 60) Tested by: falves11, bamby
	  ........ ................

2009-04-18 01:42 +0000 [r189207-189208]  David Vossel <dvossel@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500
	  (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009)
	  | 12 lines National prefix inserted even when caller ID not
	  available When the caller ID is restricted, the expected behavior
	  is for the caller id to be blank. In chan_dahdi, the national
	  prefix is placed onto the callers number even if its restricted
	  (empty) causing the caller id to be the national prefix rather
	  than blank. (closes issue #13207) Reported by: shawkris Patches:
	  national_prefix.diff uploaded by dvossel (license 671) Review:
	  http://reviewboard.digium.com/r/220/ ........ ................

	* /, channels/chan_agent.c: Merged revisions 189204 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500
	  (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009)
	  | 12 lines Fixed autologoff in agents.conf not working when agent
	  logs in via AgentLogin app An agent logs in by calling an
	  extension that calls the AgentLogin app. In agents.conf
	  ackcall=always is set, so when they get a call they have the
	  choice to either acknowledge it or ignore it. autologoff=10 is
	  set as well, so if the agent ignores the call over 10sec one may
	  assume that the agent should be logged out (and in this case
	  hungup on as well), but this was not happening. (closes issue
	  #14091) Reported by: evandro Patches: autologoff.diff uploaded by
	  dvossel (license 671) Review:
	  http://reviewboard.digium.com/r/225/ ........ ................

2009-04-17 21:56 +0000 [r189140]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 189137 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009)
	  | 17 lines Merged revisions 188833,189134 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009)
	  | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled.
	  Leave jitter setting alone. JIRA ABE-1835 ........ r189134 |
	  rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines
	  Modifed/added some debug messages. JIRA ABE-1835 ........
	  ................

2009-04-17 20:21 +0000 [r189105]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 |
	  mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13
	  lines Prevent a crash when SIP blonde transferring an unbridged
	  call. If one attempts to use the attended transfer button on a
	  SIP phone to transfer an unbridged call (such as a call to an
	  IVR) but hangs up while the target of the transfer is still
	  ringing, we need to not crash. The problem was that ast_hangup
	  was called from outside the channel thread. AST-211 ........

2009-04-17 19:47 +0000 [r189081]  Sean Bright <sean.bright@gmail.com>

	* main/asterisk.c, /: Merged revisions 189077 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 |
	  seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1
	  line Fix copy/paste error with 'transmit silence' flag. ........

2009-04-17 17:31 +0000 [r189068]  Matthew Nicholson <mnicholson@digium.com>

	* main/pbx.c, /: Merged revisions 189010 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr
	  2009) | 12 lines Merged revisions 189009 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr
	  2009) | 5 lines Make Busy() application set the CDR disposition
	  to BUSY. (closes issue #14306) Reported by: cristiandimache
	  ........ ................

2009-04-17 14:50 +0000 [r188941-188950]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) |
	  22 lines Merged revisions 188946 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) |
	  15 lines Fix a bug where a value used to create the channel name
	  was bogus. This commit fixes the scenario where an incoming call
	  is authenticated using a peer entry. Previously the channel name
	  was created using either the username setting from the sip.conf
	  entry or the IP address that the call came from. Now the channel
	  name will be created using the peer name itself. This commit will
	  not change the way the channel name is generated for users or
	  friends. (closes issue #14256) Reported by: Nick_Lewis Patches:
	  chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by:
	  Nick_Lewis, file ........ ................

	* channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri,
	  17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4
	  lines Fix a situation where the DAHDI channel private structure
	  lock was not unlocked when it should have been. (issue AST-210)
	  ........ ................

2009-04-16 22:05 +0000 [r188777-188839]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009)
	  | 14 lines Merged revisions 188835 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009)
	  | 7 lines Only update realtime, if global option rtupdate !=
	  false (closes issue #14885) Reported by: deepesh Patches:
	  20090413__bug14885.diff.txt uploaded by tilghman (license 14)
	  Tested by: deepesh ........ ................

	* apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500
	  (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009)
	  | 4 lines Umask should not be exported into global namespace.
	  (closes issue #14912) Reported by: jcapp ........
	  ................

2009-04-15 20:20 +0000 [r188474-188598]  Mark Michelson <mmichelson@digium.com>

	* /, main/file.c: Merged revisions 188585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr
	  2009) | 13 lines Merged revisions 188582 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr
	  2009) | 7 lines Update ast_readvideo_callback to match
	  ast_readaudio_callback. This fixes potential refcount errors that
	  may occur on ast_filestreams. AST-208 ........ ................

	* apps/app_queue.c, /: Merged revisions 188470 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 |
	  mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3
	  lines Fix a couple of queue member reference leaks. ........

2009-04-14 17:46 +0000 [r188259-188416]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 188413 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 |
	  file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix
	  an incorrect clock rate when sending T140 text. (closes issue
	  #14029) Reported by: epicac ........

	* /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 |
	  file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix
	  a bug with the change I made yesterday to outbound proxy support.
	  Per discussion with oej on IRC we need the actual IP address, not
	  the outbound proxy IP address, in the sa field. Upon further
	  inspection this should make the behaviour of all other uses of
	  the outbound proxy in the code. ........

2009-04-14 05:47 +0000 [r188209-188213]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: Merged revisions 188210 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 |
	  tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines
	  As suggested by Russell, warn users when their dialplan arguments
	  contain pipes, but not commas. ........

	* /, utils/smsq.c: Merged revisions 188206 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 |
	  tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines
	  Application delimiter is ',', not '|'. (closes issue #14881)
	  Reported by: stegro Patches: smsq.patch uploaded by stegro
	  (license 752) ........

2009-04-13 19:33 +0000 [r188105]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c, /: Merged revisions 188102 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr
	  2009) | 5 lines Fix another crash related to cached realtime
	  music on hold. This was another off-by-one problem caused by
	  moh_register. ........

2009-04-13 16:34 +0000 [r188070]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 |
	  file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
	  Fix a bug where using an outbound proxy would cause the local
	  address to be 127.0.0.1. Copy the outbound proxy IP address into
	  the SIP dialog structure as the IP address we will be sending to.
	  This has to be done because the logic that determines what local
	  IP address to use in the SIP messages is not aware of an outbound
	  proxy being in place. It only knows what IP address we are
	  sending to. (closes issue #12006) Reported by: mnicholson
	  ........

2009-04-13 14:20 +0000 [r188039]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 188032 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 |
	  mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6
	  lines Set all queue variables on both the caller and member
	  channels. This allows for the variables to be accessed if a
	  member macro is run. Thanks to Grigoriy Puzankin for bringing
	  this up on the -dev list. ........

2009-04-10 20:28 +0000 [r187916]  Jeff Peeler <jpeeler@digium.com>

	* channels/Makefile, /: Merged revisions 187906 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 |
	  jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines
	  Fix module embedding for chan_h323. Include libchanh323.a in the
	  modules.link file so that all the symbols can be resolved at link
	  time. (closes issue #11966) Reported by: dome Patches:
	  issue_11966.patch uploaded by kpfleming (license 421) Tested by:
	  jpeeler ........

2009-04-10 17:31 +0000 [r187769]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/sip-friends.sql,
	  contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500
	  (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10
	  Apr 2009) | 2 lines Add lastms column to the contributed table
	  designs ........ ................

2009-04-10 16:54 +0000 [r187724]  Kevin P. Fleming <kpfleming@digium.com>

	* /, build_tools/embed_modules.xml: Merged revisions 187721 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10
	  Apr 2009) | 5 lines clean up some patterns for files to remove
	  add embedding support for bridge and test modules ........

2009-04-10 16:05 +0000 [r187679]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 |
	  tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines
	  Ensure pvt is not NULL before dereferencing it. (closes issue
	  #14784) Reported by: pj ........

2009-04-10 16:01 +0000 [r187677]  Russell Bryant <russell@digium.com>

	* tests/test_sched.c, tests/test_heap.c, /: Merged revisions 187675
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10
	  Apr 2009) | 2 lines Disable test modules by default. ........

2009-04-10 03:57 +0000 [r187601]  Tilghman Lesher <tlesher@digium.com>

	* main/audiohook.c, main/bridging.c, main/channel.c, main/pbx.c,
	  main/manager.c, /, include/asterisk/linkedlists.h,
	  main/features.c, main/http.c, main/app.c,
	  include/asterisk/lock.h: Merged revisions 187599 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009)
	  | 2 lines Modify headers and macros, according to Russell's
	  suggestions on the -dev list ........

2009-04-09 21:09 +0000 [r187564]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merge revision 187488 from trunk.

2009-04-09 19:29 +0000 [r187531-187546]  David Vossel <dvossel@digium.com>

	* main/audiohook.c, /: Merged revisions 186379 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 |
	  dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 6 lines
	  audio_audiohook_write_list() did not correctly update sample size
	  after ast_translate. audio_audiohook_write_list() did not take
	  into account that the sample size may change after translation
	  depending on if the original frame is is 8khz or 16khz. the
	  sample size is now updated after translating to reflect this
	  possibility. This caused the audio on the receiving end to sound
	  terrible. Thanks to jcolp and mmichelson for helping me work this
	  out. (issue AST-197) ........

	* /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009)
	  | 16 lines Merged revisions 185845 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009)
	  | 10 lines Fixes issue with dropped calles due to re-Invite glare
	  and re-Invites never executing after a 491 Acknowledgement for
	  491 responses were never being processed because it didn't match
	  our pending invite's seqno. Since the ACK was never processed,
	  the 491 frame would continue to be retransmitted until eventually
	  the call was dropped due to max retries. Now during a pending
	  invite, if we receive another invite, we send an 491 and hold on
	  to that glare invite's seqno in the "glareinvite" variable for
	  that sip_pvt struct. When ACK's are received, we first check to
	  see if it is in response to our pending invite, if not we check
	  to see if it is in response to a glare invite. In this case, it
	  is in response to the glare invite and must be dealt with or the
	  call is dropped. I've changed the wait time for resending the
	  re-Invite after receving a 491 response to comply with RFC 3261.
	  Before this patch the scheduled re-Invite would only change a
	  flag indicating that the re-Invite should be sent out, now it
	  actually sends it out as well. (closes issue #12013) Reported by:
	  alx Review: http://reviewboard.digium.com/r/213/ ........
	  ................

2009-04-09 19:15 +0000 [r187496]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c, /: Merged revisions 187421,187424 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu,
	  09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using
	  cached realtime moh. The moh_register function links an mohclass
	  and then immediately unrefs the class since the container now has
	  a reference. The problem with using realtime music on hold is
	  that the class is allocated, registered, and started in one fell
	  swoop. The refcounting logic resulted in the count being off by
	  one. The same problem did not happen when using a static config
	  because the allocation and registration of an mohclass is a
	  separate operation from starting moh. This also did not affect
	  non-cached realtime moh because the classes are not registered at
	  all. I also have modified res_musiconhold to use the _t_ variants
	  of the ao2_ functions so that more info can be gleaned when
	  attempting to trace the refcounts. I found this to be incredibly
	  helpful for debugging this issue and there's no good reason to
	  remove it. (closes issue #14661) Reported by: sum ........
	  r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr
	  2009) | 3 lines Use safe macro practices even though they really
	  aren't necessary. ........

2009-04-09 18:55 +0000 [r187051-187487]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /, include/asterisk/linkedlists.h,
	  include/asterisk/lock.h: Merged revisions 187483 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500
	  (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009)
	  | 8 lines Race condition between ast_cli_command() and 'module
	  unload' could cause a deadlock. Add lock timeouts to avoid this
	  potential deadlock. (closes issue #14705) Reported by: jamessan
	  Patches: 20090320__bug14705.diff.txt uploaded by tilghman
	  (license 14) Tested by: jamessan ........ ................

	* /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 |
	  tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines
	  Allow '/' in username portion of register; this is a regression.
	  (closes issue #14668) Reported by: Netview ........

	* /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions
	  187363 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009)
	  | 10 lines Merged revisions 187362 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009)
	  | 3 lines Permit zero-length text messages in SIP. (Related to an
	  issue posted to the -users list, subject "AEL2, BASE64_DECODE and
	  hexadecimal") ........ ................

	* main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
	  utils/Makefile, include/asterisk.h, /, main/Makefile,
	  main/file.c, main/astfd.c (added): Merged revisions 187302 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500
	  (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009)
	  | 3 lines Add debugging mode for diagnosing file descriptor
	  leaks. (Related to issue #14625) ........ r187301 | tilghman |
	  2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops,
	  missed this file in the last commit. ........ ................

	* /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 |
	  tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines
	  If the first column is empty, output a delimiter anyway. (closes
	  issue #14848) Reported by: john8675309 Patches:
	  20090408__bug14848.diff.txt uploaded by tilghman (license 14)
	  Tested by: john8675309 ........

2009-04-08 16:54 +0000 [r186988-187049]  Mark Michelson <mmichelson@digium.com>

	* res/res_musiconhold.c, /: Merged revisions 187046 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500
	  (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr
	  2009) | 10 lines Fix a small logical error when loading moh
	  classes. We were unconditionally incrementing the number of
	  mohclasses registered. However, we should actually only increment
	  if the call to moh_register was successful. While this probably
	  has never caused problems, I noticed it and decided to fix it
	  anyway. ........ ................

	* main/channel.c, /: Merged revisions 186985 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr
	  2009) | 30 lines Merged revisions 186984 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr
	  2009) | 24 lines Make a couple of changes with regards to a new
	  message printed in ast_read(). "ast_read() called with no
	  recorded file descriptor" is a new message added after a bug was
	  discovered. Unfortunately, it seems there are a bunch of places
	  that potentially make such calls to ast_read() and trigger this
	  error message to be displayed. This commit does two things to
	  help to make this message appear less. First, the message has
	  been downgraded to a debug level message if dev mode is not
	  enabled. The message means a lot more to developers than it does
	  to end users, and so developers should take an effort to be sure
	  to call ast_read only when a channel is ready to be read from.
	  However, since this doesn't actually cause an error in operation
	  and is not something a user can easily fix, we should not spam
	  their console with these messages. Second, the message has been
	  moved to after the check for any pending masquerades. ast_read()
	  being called with no recorded file descriptor should not
	  interfere with a masquerade taking place. This could be seen as a
	  simple way of resolving issue #14723. However, I still want to
	  try to clear out the existing ways of triggering this message,
	  since I feel that would be a better resolution for the issue.
	  ........ ................

2009-04-08 12:39 +0000 [r186929]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 |
	  russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines
	  Update some comments and resolve potential memory corruption in
	  chan_sip. While browsing chan_sip the other day, I noticed this
	  dangerous code in dialog_needdestroy(). This function is an
	  ao2_callback. It is absolutely _not_ okay to unlock the container
	  from within this function. It's also not clear why it was useful.
	  Given that it could cause memory corruption, I have removed it.
	  There was also a TODO comment left describing a potential
	  implementation of an improvement to the needdestroy handling. I'm
	  not convinced that what was described is the best choice here, so
	  I have briefly described the way that this function is used today
	  that could be improved. ........

2009-04-08 05:08 +0000 [r186901]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 |
	  tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines
	  Add lastms to the require API call. ........

2009-04-08 00:10 +0000 [r186836-186845]  Mark Michelson <mmichelson@digium.com>

	* formats/format_wav_gsm.c, /, formats/format_wav.c: Merged
	  revisions 186842 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr
	  2009) | 14 lines Merged revisions 186841 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr
	  2009) | 8 lines Fix a few typos of the word "frequency." (closes
	  issue #14842) Reported by: jvandal Patches: frequency-typo.diff
	  uploaded by jvandal (license 413) ........ ................

	* /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 |
	  mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7
	  lines Fix bad merge from fix for issue 13867. (closes issue
	  #14686) Reported by: davidw ........

	* main/channel.c, /: Merged revisions 186833 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr
	  2009) | 15 lines Merged revisions 186832 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr
	  2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a
	  p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
	  warning sounds will not be properly played to either party of the
	  bridge. (closes issue #14845) Reported by: adomjan ........
	  ................

2009-04-07 22:33 +0000 [r186807]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_macro.c: Merged revisions 186799 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009)
	  | 10 lines Merged revisions 186775 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009)
	  | 3 lines Fix Macro documentation to match current (and intended)
	  behavior. (See -dev mailing list) ........ ................

2009-04-07 20:59 +0000 [r186723]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, /: Merged revisions 186720 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr
	  2009) | 12 lines Merged revisions 186719 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr
	  2009) | 6 lines Ensure that \r\n is printed after the ActionID in
	  an OriginateResponse. (closes issue #14847) Reported by: kobaz
	  ........ ................

2009-04-03 20:21 +0000 [r186469]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500
	  (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr
	  2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not
	  properly switch formats when requested Don't offer
	  AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
	  provide a slight performance benefit, the translation core in
	  Asterisk has some flaws when a channel driver offers multiple raw
	  formats. this fix is much simpler than fixing the translation
	  core to solve that issue (although that will be done later).
	  ........ ................

2009-04-03 20:05 +0000 [r186449]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
	  revisions 186444,186447 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009)
	  | 14 lines Merged revisions 186415 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009)
	  | 7 lines Distinguish in a sent email between simple sends and
	  forwards. (closes issue #11678) Reported by: jamessan Patches:
	  20090330__bug11678.diff.txt uploaded by tilghman (license 14)
	  Tested by: tilghman, lmadsen ........ ................ r186447 |
	  tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines
	  Merged revisions 186445 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009)
	  | 2 lines Found a conflict in the last commit, due to multiple
	  targets ........ ................

2009-04-03 15:56 +0000 [r186324]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/crypto.h, /: Merged revisions 186321 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri,
	  03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5
	  lines Fix a problem with the crypto variable definitions not
	  actually being defined properly. (closes issue #14804) Reported
	  by: jvandal ........ ................

2009-04-03 15:19 +0000 [r186302]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009)
	  | 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820)
	  Reported by: phsultan ........

2009-04-03 14:34 +0000 [r186289]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr
	  2009) | 20 lines Fix the ability to retrieve voicemail messages
	  from IMAP. A recent change made interactive vm_states no longer
	  get added to the list of vm_states and instead get stored in
	  thread-local storage. In trunk and all the 1.6.X branches, the
	  problem is that when we search for messages in a voicemail box,
	  we would attempt to update the appropriate vm_state struct by
	  directly searching in the list of vm_states instead of using the
	  get_vm_state_by_imap_user function. This meant we could not find
	  the interactive vm_state that we wanted. (closes issue #14685)
	  Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson
	  (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........

2009-04-03 02:11 +0000 [r186233]  Russell Bryant <russell@digium.com>

	* cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009)
	  | 29 lines Merged revisions 186229 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009)
	  | 21 lines Fix a memory leak in cdr_radius. I came across this
	  while doing some testing of my ast_channel_ao2 branch. After
	  running a test overnight that generated over 5 million calls,
	  Asterisk had taken up about 1 GB of my system memory. So, I
	  re-ran the test with MALLOC_DEBUG turned on. However, it showed
	  no leaks in Asterisk during the test, even though Asterisk was
	  still consuming it somehow. Instead, I turned to valgrind, which
	  when run with --leak-check=full, told me exactly where the leak
	  came from, which was from allocations inside the radiusclient-ng
	  library. This explains why MALLOC_DEBUG did not report it. After
	  a bit of analysis, I found that we were leaking a little bit of
	  memory every time a CDR record was passed to cdr_radius. I don't
	  actually have a radius server set up to receive CDR records.
	  However, I always have my development systems compile and install
	  all modules. In addition to making sure there are not build
	  errors across modules, always loading modules helps find bugs
	  like this, too, so it is strongly recommend for all developers.
	  ........ ................

2009-04-02 22:00 +0000 [r186178]  Mark Michelson <mmichelson@digium.com>

	* configs/features.conf.sample, /: Merged revisions 186175 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500
	  (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr
	  2009) | 5 lines Fix instructions in one-step parking comment to
	  make more sense. Changed a capital K to a lowercase k. ........
	  ................

2009-04-02 17:28 +0000 [r186111]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500
	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02
	  Apr 2009) | 3 lines ensure that the buffer passed to
	  DAHDI_SET_BUFINFO is fully initialized ........ ................

2009-04-02 17:14 +0000 [r186022-186063]  Tilghman Lesher <tlesher@digium.com>

	* configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions
	  186060 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009)
	  | 16 lines Merged revisions 186059 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500
	  (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02
	  Apr 2009) | 2 lines Fix for AST-2009-003 ........
	  ................ ................

	* main/strings.c, /: Merged revisions 186021 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 |
	  tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines
	  Missed a common case for needing to extend the buffer. (closes
	  issue #14716) Reported by: sum Patches:
	  20090402__bug14716.diff.txt uploaded by tilghman (license 14)
	  Tested by: sum ........

2009-04-02 13:54 +0000 [r185957]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500
	  (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr
	  2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and
	  DAHDI_GET_PARAMS ioctls were recently corrected to show that they
	  do, in fact, read data from userspace as part of their work. due
	  to this fix, valgrind now reports a number of cases where
	  chan_dahdi passed an uninitialized (or partially) buffer to these
	  ioctls, which could lead to unexpected behavior. this patch
	  corrects chan_dahdi to ensure that buffers passed to these ioctls
	  are always fully initialized. ........ ................

2009-04-01 22:44 +0000 [r185947]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/pbx.h, include/asterisk/strings.h,
	  main/taskprocessor.c, res/res_odbc.c,
	  include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c,
	  main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h,
	  main/ast_expr2f.c: Merged revisions 185912 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 |
	  tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines
	  Merge changes from str_substitution that are unrelated to that
	  branch. Included is a small bugfix to an ast_str helper, but most
	  of these changes are simply doxygen fixes. ........

2009-04-01 13:51 +0000 [r185775]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 185772 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009)
	  | 14 lines Merged revisions 185771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009)
	  | 6 lines Fix a case where DTMF could bypass audiohooks. This
	  change fixes a situation where an audiohook that wants DTMF would
	  not actually get it. This is in the code path where we end DTMF
	  digit length emulation while handling a NULL frame. ........
	  ................

2009-03-31 22:38 +0000 [r185667]  Kevin P. Fleming <kpfleming@digium.com>

	* utils, /: Merged revisions 185664 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 |
	  kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line
	  ignore copied (generated) file ........

2009-03-31 22:13 +0000 [r185472-185605]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 185604 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 |
	  mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3
	  lines Fix trunk's compilation. ........

	* apps/app_queue.c, /: Merged revisions 185600 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar
	  2009) | 12 lines Merged revisions 185599 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar
	  2009) | 6 lines Fix crash that would occur if an empty member was
	  specified in queues.conf. (closes issue #14796) Reported by: pida
	  ........ ................

	* apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500
	  (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar
	  2009) | 8 lines Fix Russian voicemail intro to say the word
	  "messages" properly. (closes issue #14736) Reported by: chappell
	  Patches: voicemail_no_messages.diff uploaded by chappell (license
	  8) ........ ................

2009-03-31 17:51 +0000 [r185428]  David Brooks <dbrooks@digium.com>

	* channels/chan_gtalk.c, /: Merged revisions 185363 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500
	  (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009)
	  | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains
	  extra whitespaces To drill into the xmpp to find the capabilities
	  between channels, chan_gtalk calls iks_child() and iks_next().
	  iks_child() and iks_next() are functions in the iksemel xml
	  parsing library that traverse xml nodes. The bug here is that
	  both iks_child() and iks_next() will return the next iks_struct
	  node *regardless* of type. chan_gtalk expects the next node to be
	  of type IKS_TAG, which in most cases, it is, but in this case (a
	  call being made from the Empathy IM client), there exists
	  iks_struct nodes which are not IKS_TAG data (they are extraneous
	  whitespaces), and chan_gtalk doesn't handle that case, so
	  capabilities don't match, and a call cannot be made.
	  iks_first_tag() and iks_next_tag(), on the other hand, will not
	  return the very next iks_struct, but will check to see if the
	  next iks_struct is of type IKS_TAG. If it isn't, it will be
	  skipped, and the next struct of type IKS_TAG it finds will be
	  returned. This assures that chan_gtalk will find the iks_struct
	  it is looking for. This fix simply changes all calls to
	  iks_child() and iks_next() to become calls to iks_first_tag() and
	  iks_next_tag(), which resolves the capability matching. The
	  following is a payload listing from Empathy, which, due to the
	  extraneous whitespace, will not be parsed correctly by iksemel:
	  <iq from='dbrooksjab@235-22-24-10/Telepathy'
	  to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
	  <session xmlns='http://www.google.com/session'
	  initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
	  id='1837267342'> <description
	  xmlns='http://www.google.com/session/phone'> <payload-type
	  clockrate='16000' name='speex' id='96'/> <payload-type
	  clockrate='8000' name='PCMA' id='8'/> <payload-type
	  clockrate='8000' name='PCMU' id='0'/> <payload-type
	  clockrate='90000' name='MPA' id='97'/> <payload-type
	  clockrate='16000' name='SIREN' id='98'/> <payload-type
	  clockrate='8000' name='telephone-event' id='99'/> </description>
	  </session> </iq> Review: http://reviewboard.digium.com/r/181/
	  ........ ................

2009-03-31 14:59 +0000 [r185264]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c, /: Merged revisions 185261 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 |
	  russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines
	  Don't free() an astobj2 object. (closes issue #14672) Reported
	  by: makoto ........

2009-03-31 14:11 +0000 [r185200]  Joshua Colp <jcolp@digium.com>

	* main/audiohook.c, /: Merged revisions 185197 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) |
	  15 lines Merged revisions 185196 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8
	  lines Fix crash when moving audiohooks between channels. Handle
	  the scenario where we are called to move audiohooks between
	  channels and the source channel does not actually have any on it.
	  (closes issue #14734) Reported by: corruptor ........
	  ................

2009-03-30 20:52 +0000 [r185128-185129]  Richard Mudgett <rmudgett@digium.com>

	* channels/misdn_config.c, /, configs/misdn.conf.sample: Merged
	  revisions 185123 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009)
	  | 9 lines Merged revisions 185121 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009)
	  | 1 line Update the channel allocation method documentation.
	  ........ ................

	* channels/misdn/isdn_lib.c, /: Merged revisions 185122 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500
	  (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009)
	  | 19 lines Make chan_misdn BRI TE side normally defer channel
	  selection to the NT side. Channel allocation collisions are not
	  handled by chan_misdn very well. This patch simply avoids the
	  problem for BRI only. For PRI, allocation collisions are still
	  possible but less likely since there are simply more channels
	  available and each end could use a different allocation strategy.
	  misdn.conf options available: te_choose_channel - Use to force
	  the TE side to allocate channels. method - Specify the channel
	  allocation strategy. (closes issue #13488) Reported by:
	  Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich
	  Tested by: crich, siepkes, festr ........ ................

2009-03-30 16:52 +0000 [r185089]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 185072 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar
	  2009) | 45 lines Merged revisions 185031 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar
	  2009) | 39 lines Fix queue weight behavior so that calls in
	  low-weight queues are not inappropriately blocked. (This is
	  copied and pasted from the review request I made for this patch)
	  Asterisk has some odd behavior when queue weights are used. The
	  current logic used when potentially calling a queue member is: If
	  the member we are going to call is part of another queue and
	  _that other queue has any callers in it_ and has a higher weight
	  than the queue we are calling from, then don't try to contact
	  that member. The issue here is what I have marked with
	  underscores. If the higher-weighted queue has any callers in it
	  at all, then the queue member will be unreachable from the
	  lower-weighted queue. This has the potential to be really really
	  bad if using a queue strategy, such as leastrecent or
	  fewestcalls, with the potential to call the same member
	  repeatedly. The fix proposed by garychen on issue 13220 is very
	  simple and, as far as I can see, works well for this situation.
	  With this set of changes, the logic used becomes: If the member
	  we are going to call is part of another queue, the other queue
	  has a higher weight than the queue we are calling from, and the
	  higher weight queue has at least as many callers as available
	  members, then do not try to contact the queue member. If the
	  higher weighted queue has fewer callers than available members,
	  then there is no reason to deny the call to this member since the
	  other queue can afford to spare a member. Since the fix involved
	  writing a generic function for determining the number of
	  available members in the queue, I also modified the is_our_turn
	  function to make use of the new num_available_members function to
	  determine if it is our turn to try calling a member. There is one
	  small behavior change. Before writing this patch, if you had
	  autofill disabled, then if you were the head caller in a queue,
	  you would automatically be told that it was your turn to try
	  calling a member. This did not take into account whether there
	  were actually any queue members available to take the call. Now
	  we actually make sure there is at least one member available to
	  take the call if autofill is disabled. (closes issue #13220)
	  Reported by: garychen Review:
	  http://reviewboard.digium.com/r/202/ ........ ................

2009-03-30 14:43 +0000 [r184951]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) |
	  21 lines Merged revisions 184947 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) |
	  14 lines Improve our handling of T38 in the initial INVITE from a
	  device. We now answer with matching media streams to what is
	  requested. If an INVITE is received with both a T38 and RTP media
	  stream this means we answer with both. For any outgoing calls
	  created as a result of this inbound one no T38 is requested in
	  the initial INVITE. Instead if we start receiving udptl packets
	  we trigger a reinvite on the outbound side. (closes issue #12437)
	  Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu
	  Review: http://reviewboard.digium.com/r/208/ ........
	  ................

2009-03-30 13:57 +0000 [r184913]  Russell Bryant <russell@digium.com>

	* channels/h323/Makefile.in, /: Merged revisions 184910 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30
	  Mar 2009) | 4 lines Fix build error when chan_h323 is not being
	  built. (reported by cai1982 in #asterisk-dev) ........

2009-03-29 05:56 +0000 [r184839-184846]  Russell Bryant <russell@digium.com>

	* apps/app_followme.c, /: Merged revisions 184843 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009)
	  | 13 lines Merged revisions 184842 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009)
	  | 5 lines Ensure targs variable is fully initialized. (closes
	  issue #14758) Reported by: tim_ringenbach ........
	  ................

	* channels/Makefile, /: Merged revisions 184838 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 |
	  russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines
	  Simplify chan_h323 build to not require a second run of "make".
	  (closes issue #14715) Reported by: jthurman Patches:
	  h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license
	  614) Tested by: tzafrir, russell ........

2009-03-27 19:21 +0000 [r184779]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c, main/timing.c, main/channel.c, /,
	  bridges/bridge_softmix.c, include/asterisk/timing.h,
	  include/asterisk/channel.h: Merged revisions 184762 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar
	  2009) | 12 lines Improve timing interface to remember which
	  provider provided a timer The ability to load/unload timing
	  interfaces is nice, but it means that when a timer is allocated,
	  it may come from provider A, but later provider B becomes the
	  'preferred' provider. If this happens, all timer API calls on the
	  timer that was provided by provider A will actually be handed to
	  provider B, which will say WTF and return an error. This patch
	  changes the timer API to include a pointer to the provider of the
	  timer handle so that future operations on the timer will be
	  forwarded to the proper provider. (closes issue #14697) Reported
	  by: moy Review: http://reviewboard.digium.com/r/211/ ........

2009-03-27 18:12 +0000 [r184707-184729]  Russell Bryant <russell@digium.com>

	* main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27
	  Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure
	  we use the best RNG available. ........

	* apps/app_queue.c, apps/app_voicemail.c, main/cli.c,
	  include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c:
	  Merged revisions 184693 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 |
	  russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines
	  Change global_app_buf to ast_str_thread_global_buf. ........

2009-03-27 15:58 +0000 [r184650-184678]  Joshua Colp <jcolp@digium.com>

	* /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7
	  lines Fix a potential timer leak in bridge_softmix. It is
	  possible for a bridge to be created without actually being used.
	  In that scenario a timing file descriptor would be opened and not
	  closed. To fix this the timing file descriptor is now closed in
	  the destroy callback, not the thread function. ........

	* /, res/res_agi.c: Merged revisions 184673 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 |
	  file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix
	  speech structure leak in the AGI speech recognition integration.
	  The AGI dialplan applications did not destroy the speech
	  structure automatically if it was not destroyed by the running
	  AGI script. They will now do this. (issue LUMENVOX-15) ........

	* /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2
	  lines Remove a cast that is not needed. ........

2009-03-27 14:09 +0000 [r184632]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /,
	  res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions
	  184630 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 |
	  russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines
	  Change g_eid to ast_eid_default. ........

2009-03-27 13:59 +0000 [r184612-184629]  Joshua Colp <jcolp@digium.com>

	* /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6
	  lines Fix a potential race condition when creating a software
	  based mixing bridge. It was possible for no timer to become
	  available between creating the bridge and starting it. We now
	  open a timer when creating it and keep it open until the bridge
	  is destroyed. ........

	* /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) |
	  16 lines Merged revisions 184565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9
	  lines Fix an issue where nat=yes would not always take effect for
	  the RTP session on outgoing calls. If calls were placed using an
	  IP address or hostname the global nat setting was copied over but
	  was not set on the RTP session itself. This caused the RTP stack
	  to not perform symmetric RTP actions. (closes issue #14546)
	  Reported by: acunningham ........ ................

2009-03-27 02:35 +0000 [r184514-184552]  Russell Bryant <russell@digium.com>

	* /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009)
	  | 20 lines Fix some issues with rwlock corruption that caused
	  deadlock like symptoms. When dvossel and I were doing some load
	  testing last week, we noticed that we could make Asterisk trunk
	  lock up instantly when we started generating a bunch of calls.
	  The backtraces of locked threads were bizarre, and many were
	  stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a
	  number of places where a backtrace would be loaded into an
	  invalid index of the backtrace array. It's an off by one error,
	  which ends up writing over the rwlock itself. 2) Ensure that in
	  the array of held locks, we NULL out an index once it is not
	  being used so that it's not confusing when analyzing its
	  contents. 3) Remove a bunch of logging referring to an rwlock
	  operating being done with "deep reentrancy". It is normal for
	  _many_ threads to hold a read lock on an rwlock. ........

	* /, main/file.c: Merged revisions 184515 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 |
	  russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines
	  Don't act surprised if we get a -1 indication. ........

	* include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26
	  Mar 2009) | 2 lines Pass more useful information through to lock
	  tracking when DEBUG_THREADS is on. ........

2009-03-26 22:19 +0000 [r184454]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile, /: Merged revisions 184448 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar
	  2009) | 9 lines Merged revisions 184447 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar
	  2009) | 3 lines use new, improved 8kHz prompts ........
	  ................

2009-03-25 22:15 +0000 [r184343-184346]  Russell Bryant <russell@digium.com>

	* /, main/event.c: Merged revisions 184344 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 |
	  russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines
	  Remove unneeded AST_LIST_ENTRY() and comment on the purpose of
	  ast_event_ref. ........

	* include/asterisk/_private.h, channels/chan_iax2.c,
	  channels/chan_dahdi.c, include/asterisk/event.h,
	  apps/app_minivm.c, res/ais/evt.c, main/event.c,
	  include/asterisk/strings.h, main/asterisk.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c,
	  channels/chan_unistim.c, include/asterisk/devicestate.h, /,
	  channels/chan_sip.c, main/devicestate.c: Merged revisions 184339
	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25
	  Mar 2009) | 35 lines Improve performance of the ast_event cache
	  functionality. This code comes from
	  svn/asterisk/team/russell/event_performance/. Here is a summary
	  of the changes that have been made, in order of both invasiveness
	  and performance impact, from smallest to largest. 1) Asterisk
	  1.6.1 introduces some additional logic to be able to handle
	  distributed device state. This functionality comes at a cost. One
	  relatively minor change in this patch is that the extra
	  processing required for distributed device state is now
	  completely bypassed if it's not needed. 2) One of the things that
	  I noticed when profiling this code was that a _lot_ of time was
	  spent doing string comparisons. I changed the way strings are
	  represented in an event to include a hash value at the front. So,
	  before doing a string comparison, we do an integer comparison on
	  the hash. 3) Finally, the code that handles the event cache has
	  been re-written. I tried to do this in a such a way that it had
	  minimal impact on the API. I did have to change one API call,
	  though - ast_event_queue_and_cache(). However, the way it works
	  now is nicer, IMO. Each type of event that can be cached (MWI,
	  device state) has its own hash table and rules for hashing and
	  comparing objects. This by far made the biggest impact on
	  performance. For additional details regarding this code and how
	  it was tested, please see the review request. (closes issue
	  #14738) Reported by: russell Review:
	  http://reviewboard.digium.com/r/205/ ........

2009-03-25 19:27 +0000 [r184266-184283]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 |
	  file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix
	  issue with a T38 reinvite being sent even if not configured to do
	  so. If we receive a T38 request negotiate control frame we should
	  only attempt to do so if the option is enabled on the dialog.
	  ........

	* main/bridging.c, /: Merged revisions 183652 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 |
	  file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix
	  a minor logic flaw with the bridge generic thread. We only want
	  to move the channel pointers that are actually present. ........

2009-03-25 15:33 +0000 [r184256]  Eliel C. Sardanons <eliels@gmail.com>

	* main/asterisk.c, /: Merged revisions 184220 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) |
	  19 lines Merged revisions 184188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) |
	  13 lines Avoid destroying the CLI line when moving the cursor
	  backward and trying to autocomplete. When moving the cursor
	  backward and pressing TAB to autocomplete, a NULL is put in the
	  line and we are loosing what we have already wrote after the
	  actual cursor position. (closes issue #14373) Reported by: eliel
	  Patches: asterisk.c.patch uploaded by eliel (license 64) Tested
	  by: lmadsen ........ ................

2009-03-25 14:40 +0000 [r184150-184221]  Russell Bryant <russell@digium.com>

	* main/timing.c, /: Merged revisions 184219 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 |
	  russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines
	  Include poll-compat.h ........

	* main/timing.c, /: Merged revisions 184151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 |
	  russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines
	  Change poll() to ast_poll(). ........

	* utils/Makefile, /, include/asterisk/compat.h: Merged revisions
	  184147 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 |
	  russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
	  Fix build issues on Mac OSX. (closes issue #14714) Reported by:
	  ygor ........

2009-03-24 22:42 +0000 [r184082]  Mark Michelson <mmichelson@digium.com>

	* apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar
	  2009) | 15 lines Merged revisions 184078 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar
	  2009) | 9 lines Change NULL pointer check to be ast_strlen_zero.
	  The 'digit' variable is guaranteed to be non-NULL, so the if
	  statement could never evaluate true. Changing to ast_strlen_zero
	  makes the logic correct. This was found while reviewing
	  ast_channel_ao2 code review. ........ ................

2009-03-24 22:02 +0000 [r184041-184044]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 184043 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 |
	  russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines
	  Put siren7 and siren14 in ast_best_codec() just so they're in
	  there somewhere. ........

	* channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009)
	  | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low
	  and =medium The default codec configuration for chan_iax2 is
	  bandwidth=low. I noticed slin16 being negotiated as the codec in
	  some test calls, but that no longer happens after this change.
	  ........

2009-03-24 15:29 +0000 [r183868-183917]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/voicemail.conf.sample: Merged revisions 183914 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500
	  (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009)
	  | 3 lines Additionally note that the operator option needs an 'o'
	  extension. (Related to issue #14731) ........ ................

	* /, main/http.c: Merged revisions 183865 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 |
	  tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines
	  Allow browsers to cache images and other static content. (This is
	  a regression over 1.4) ........

2009-03-23 19:00 +0000 [r183769]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c, /: Merged revisions 183766 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar
	  2009) | 13 lines Merged revisions 183700 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar
	  2009) | 7 lines Fix a memory leak in res_monitor.c The only way
	  that this leak would occur is if Monitor were started using the
	  Manager interface and no File: header were given. Discovered
	  while reviewing the ast_channel_ao2 review request. ........
	  ................

2009-03-23 18:12 +0000 [r183704]  Leif Madsen <lmadsen@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009)
	  | 7 lines Fixes a documentation error introduced during the CLI
	  cleanup at AstriDevCon 2008. (closes issue #14655) Reported by:
	  ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728)
	  Tested by: lmadsen ........

2009-03-20 17:09 +0000 [r183564]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r183560 | russell | 2009-03-20 12:00:58 -0500
	  (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009)
	  | 2 lines Fix a crash in IAX2 registration handling found during
	  load testing with dvossel. ........ ................

2009-03-20 12:19 +0000 [r183519]  Eliel C. Sardanons <eliels@gmail.com>

	* channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) |
	  2 lines Remove duplicate <description> inside the xml
	  documentation. ........

2009-03-19 19:20 +0000 [r183337]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk
	  ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500
	  (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009)
	  | 8 lines Delay signalling progress until a PRI channel really
	  signals progress. (closes issue #13034) Reported by: klaus3000
	  Patches: 20090316__bug13034.diff.txt uploaded by tilghman
	  (license 14) patch_trunk_183progress_klaus3000.txt uploaded by
	  klaus3000 (license 65) Tested by: klaus3000 ........
	  ................

2009-03-19 18:20 +0000 [r183263]  Russell Bryant <russell@digium.com>

	* main/loader.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Merged revisions 183242 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009)
	  | 10 lines Merged revisions 183241 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009)
	  | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving
	  like expected. ........ ................

2009-03-19 18:12 +0000 [r183247]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c, /: Merged revisions 183244 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 |
	  mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16
	  lines Fix a memory leak associated with queues. For every attempt
	  that app_queue made to place an outbound call to a queue member,
	  we would allocate a queue_end_bridge structure. When the bridge
	  for the call had completed, we would free the structure.
	  Unfortunately not all call attempts actually end up bridged to a
	  member, so we need to be more selective of when to allocate the
	  structure. With this change, the allocation occurs in an area
	  where we can guarantee that the call will be bridged. (closes
	  issue #14680) Reported by: caspy Patches: 14680.patch uploaded by
	  mmichelson (license 60) Tested by: caspy ........

2009-03-19  Leif Madsen <lmadsen@digium.com>

	* Release Asterisk 1.6.2.0-beta1

2009-03-19 16:11 +0000 [r183122]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar
	  2009) | 20 lines Merged revisions 183115 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar
	  2009) | 14 lines Fix an issue where cancelled outgoing SIP calls
	  would erroneously report the device as "in use." A user was
	  having an issue where if an outgoing SIP call was canceled, the
	  SIP device would remain in use if we had not received any
	  response to the initial INVITE we sent out. The SIP device would
	  remain in use until the autocongestion timer was exhausted. I
	  tracked down the cause of this to be the section of code I am
	  removing here. I asked several people what the purpose of this
	  code was meant to be, but no one could give me any sort of answer
	  as to why this was here. The person who was having this issue has
	  been using this patch for several months and it has stopped the
	  problems they have had. AST-196 ........ ................

2009-03-19 15:45 +0000 [r183068-183111]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 |
	  file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines
	  Improve our triggering of a T38 switchover internally when
	  triggered by a received reinvite. Previously we reached across
	  the channel bridge to get the other party's SIP dialog structure
	  in order to trigger an outgoing reinvite. This is extremely
	  dangerous to do and only works if bridged to another SIP channel.
	  This patch changes this to use the T38 control frame method of
	  requesting a switchover. This change also causes the SIP channel
	  driver to propogate back whether the switchover worked or not
	  instead of blindly accepting the incoming T38 reinvite. Review:
	  http://reviewboard.digium.com/r/200/ ........

	* main/channel.c, /: Merged revisions 183057 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 |
	  file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix
	  an issue where a T38 control frame would get dropped. If two
	  channels were bridged together using a generic bridge the T38
	  control frame would get passed up instead of being indicated on
	  the other channel. ........

2009-03-18 21:19 +0000 [r183031]  Jeff Peeler <jpeeler@digium.com>

	* /, channels/h323/ast_h323.cxx: Merged revisions 183028 via
	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
	  ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18
	  Mar 2009) | 4 lines Add some code removed by mistake from commit
	  182722 that works around a file descriptor leak in versions of
	  PWLib prior to 1.12.0. ........

2009-03-18 14:39 +0000 [r182947]  Russell Bryant <russell@digium.com>

	* main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c,
	  configure, apps/app_mp3.c, res/res_agi.c,
	  include/asterisk/poll-compat.h, channels/chan_alsa.c,
	  main/asterisk.c, apps/app_nbscat.c, /, main/Makefile,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/io.h, main/utils.c, include/asterisk/channel.h:
	  Merged revisions 182847 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009)
	  | 52 lines Merged revisions 182810 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009)
	  | 44 lines Fix cases where the internal poll() was not being used
	  when it needed to be. We have seen a number of problems caused by
	  poll() not working properly on Mac OSX. If you search around,
	  you'll find a number of references to using select() instead of
	  poll() to work around these issues. In Asterisk, we've had poll.c
	  which implements poll() using select() internally. However, we
	  were still getting reports of problems. vadim investigated a bit
	  and realized that at least on his system, even though we were
	  compiling in poll.o, the system poll() was still being used. So,
	  the primary purpose of this patch is to ensure that we're using
	  the internal poll() when we want it to be used. The changes are:
	  1) Remove logic for when internal poll should be used from the
	  Makefile. Instead, put it in the configure script. The logic in
	  the configure script is the same as it was in the Makefile.
	  Ideally, we would have a functionality test for the problem, but
	  that's not actually possible, since we would have to be able to
	  run an application on the _target_ system to test poll()
	  behavior. 2) Always include poll.o in the build, but it will be
	  empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
	  throughout the source tree to ast_poll(). I feel that it is good
	  practice to give the API call a new name when we are changing its
	  behavior and not using the system version directly in all cases.
	  So, normally, ast_poll() is just redefined to poll(). On systems
	  where AST_POLL_COMPAT is defined, ast_poll() is redefined to
	  ast_internal_poll(). 4) Change poll() in main/poll.c to be
	  ast_internal_poll(). It's worth noting that any code that still
	  uses poll() directly will work fine (if they worked fine before).
	  So, for example, out of tree modules that are using poll() will
	  not stop working or anything. However, for modules to work
	  properly on Mac OSX, ast_poll() needs to be used. (closes issue
	  #13404) Reported by: agalbraith Tested by: russell, vadim
	  http://reviewboard.digium.com/r/198/ ........ ................

2009-03-17 20:53 +0000 [r182725]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /,
	  channels/h323/ast_h323.cxx, configure,
	  autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h,
	  channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions
	  182722 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 |
	  jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines
	  Allow H.323 Plus library to be used in addition to the OpenH323
	  library Chan_h323 can now be compiled against both the previously
	  supported versions of OpenH323 as well as the current H.323 Plus
	  (version 1.20.2). The configure script has been modified to look
	  in the default install location of h323 to hopefully help avoid
	  using the environment variables OPENH323DIR and PWLIBDIR. Also,
	  the CLI command "h323 show version" has been added which
	  indicates which version of h323 is in use. (closes issue #11261)
	  Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
	  uploaded by jthurman (license 614) ........

2009-03-17 16:46 +0000 [r182592]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 182553 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 |
	  russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines
	  Tweak the handling of the frame list inside of ast_answer(). This
	  does not change any behavior, but moves the frames from the local
	  frame list back to the channel read queue using an O(n) algorithm
	  instead of O(n^2). ........

2009-03-17 15:01 +0000 [r182528-182534]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, /: Merged revisions 182530 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 |
	  kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2
	  lines correct logic flaw in ast_answer() changes in r182525
	  ........

	* main/channel.c, /, main/features.c, include/asterisk/channel.h:
	  Merged revisions 182525 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 |
	  kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11
	  lines Improve behavior of ast_answer() to not lose incoming
	  frames ast_answer(), when supplied a delay before returning to
	  the caller, use ast_safe_sleep() to implement the delay.
	  Unfortunately during this time any incoming frames are discarded,
	  which is problematic for T.38 re-INVITES and other sorts of
	  channel operations. When a delay is not passed to ast_answer(),
	  it still delays for up to 500 milliseconds, waiting for media to
	  arrive. Again, though, it discards any control frames, or
	  non-voice media frames. This patch rectifies this situation, by
	  storing all incoming frames during the delay period on a list,
	  and then requeuing them onto the channel before returning to the
	  caller. http://reviewboard.digium.com/r/196/ ........

2009-03-17 05:54 +0000 [r182453]  Tilghman Lesher <tlesher@digium.com>

	* main/db.c, /: Merged revisions 182450 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ................
	  r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009)
	  | 14 lines Merged revisions 182449 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009)
	  | 7 lines Fix race in astdb The underlying db1 implementation
	  does not fully isolate the pages retrieved from astdb, so the
	  lock protecting accesses needs to be extended until the copy from
	  the shared memory structure is done. (closes issue #14682)
	  Reported by: makoto ........ ................

2009-03-17 02:02 +0000 [r182409]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/trunk ........
	  r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009)
	  | 8 lines OPENR2 uses an incorrect string value if the extension
	  delimiter is not present. * Fixed OPENR2 using an incorrect
	  string value if the extension delimiter is not present in the
	  Dial() function. This was fixed for SS7 and PRI in trunk
	  -r172400. * Made OPENR2 stripmsd behavior the same as the SS7,
	  PRI, and others. * Removed trailing whitespace that appeared with
	  OPENR2. ........

2009-03-16 20:51 +0000 [r182360-182361]  Russell Bryant <russell@digium.com>

	* /: svnmerge init

	* / (added): Create a branch for 1.6.2

2009-03-16 20:35 +0000 [r182355]  Russell Bryant <russell@digium.com>

	* CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
	  configure, include/asterisk/autoconfig.h.in, configure.ac,
	  CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This
	  commit introduces official support for R2 signaling in
	  chan_dahdi. The modifications to chan_dahdi, and the supporting
	  library, LibOpenR2, were both written by Moises Silva. Many users
	  are using this code, or a variant of it, in Asterisk 1.2, 1.4 and
	  1.6 in Brazil, México and Argentina. An unknown number of users
	  (but at least 1) are using it in each of the following countries:
	  Colombia, Nepal, Thailand, Venezuela, Perú, and probably others.
	  To use this code, LibOpenR2 must be installed from
	  http://www.libopenr2.org/. Information about configuration can be
	  found in configs/chan_dahdi.conf.sample. The code committed is
	  the most up to date version, which was being maintained in
	  svn/asterisk/team/moy/mfcr2/. I would also like to include a
	  Thank You to the many others that tested this code beyond those
	  listed in this commit message. These are the names that I could
	  find in the mantis issue. (closes issue #12509) Reported by: moy
	  Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested
	  by: moy, korihor, viniciusfontes, Skarmeth, loloski,
	  asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare,
	  ecarruda, rtorresduque, PTorres, ychen Review:
	  http://reviewboard.digium.com/r/40/

2009-03-16 17:49 +0000 [r182282]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 182281 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16
	  Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32
	  byte random padding at the beginning of an encrypted IAX2 frame
	  turns out to not be all that random at all. This patch calls
	  ast_random to fill the padding buffer with random data. The
	  padding is randomized at the beginning of every encrypted call
	  and for every encrypted retransmit frame. Review:
	  http://reviewboard.digium.com/r/193/ ........

2009-03-16 17:33 +0000 [r182211-182278]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Fix an off-by-one error in the FILE() function,
	  and extend FILE()'s length parameter to work like variable
	  substitution. Previously, FILE() returned one less character than
	  specified, due to the terminating NULL. Both the offset and
	  length parameters now behave identically to the way variable
	  substitution offsets and lengths also work. (closes issue #14670)
	  Reported by: BMC

	* channels/chan_local.c, /: Merged revisions 182208 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16
	  Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak
	  of a local pvt structure. (closes issue #14656) Reported by:
	  caspy Patches: 20090313__bug14656__2.diff.txt uploaded by
	  tilghman (license 14) Tested by: caspy ........

2009-03-16 13:58 +0000 [r182171]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Fix a memory leak in the ast_answer /
	  __ast_answer API call. For a channel that is not yet answered
	  this API call will wait until a voice frame is received on the
	  channel before returning. It does this by waiting for frames on
	  the channel and reading them in. The frames read in were not
	  freed when they should have been.

2009-03-13 21:26 +0000 [r182029-182121]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Change faulty comparison used when announcing
	  average hold minutes and seconds (closes issue #14227) Reported
	  by: caspy

	* main/features.c: Remove ast_ prefix from functions which are not
	  public.

	* /, main/features.c: Merged revisions 181990 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar
	  2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and
	  peer when interpreting DTMF. Dynamic features defined in the
	  applicationmap section of features.conf allow one to specify
	  whether the caller, callee, or both have the ability to use the
	  feature. The documentation in the features.conf.sample file could
	  be interpreted to mean that one only needs to set the
	  DYNAMIC_FEATURES channel variable on the calling channel in order
	  to allow for the callee to be able to use the features which he
	  should have permission to use. However, the DYNAMIC_FEATURES
	  variable would only be read from the channel of the participant
	  that pressed the DTMF sequence to activate the feature. The
	  result of this was that the callee was unable to use dynamic
	  features unless the dialplan writer had taken measures to be sure
	  that the DYNAMIC_FEATURES variable was set on the callee's
	  channel. This commit changes the behavior of
	  ast_feature_interpret to concatenate the values of
	  DYNAMIC_FEATURES from both parties involved in the bridge. The
	  features themselves determine who has permission to use them, so
	  there is no reason to believe that one side of the bridge could
	  gain the ability to perform an action that they should not have
	  the ability to perform. Kevin Fleming pointed out on the
	  asterisk-users list that the typical way that this was worked
	  around in the past was by setting _DYNAMIC_FEATURES on the
	  calling channel so that the value would be inherited by the
	  called channel. While this works, the documentation alone is not
	  enough to figure out why this is necessary for the callee to be
	  able to use dynamic features. In this particular case, changing
	  the code to match the documentation is safe, easy, and will
	  generally make things easier for people for future installations.
	  This bug was originally reported on the asterisk-users list by
	  David Ruggles. (closes issue #14657) Reported by: mmichelson
	  Patches: 14657.patch uploaded by mmichelson (license 60) ........

2009-03-13 17:25 +0000 [r182022]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix an issue with requesting a T38 reinvite
	  before the call is answered. The code responsible for sending the
	  T38 reinvite did not check if an INVITE was already being
	  handled. This caused things to get confused and the call to fail.
	  The code now defers sending the T38 reinvite until the current
	  INVITE is done being handled. (issue AST-191)

2009-03-13 16:55 +0000 [r181985]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: improve a bit of suboptimal code

2009-03-13 01:26 +0000 [r181899]  Richard Mudgett <rmudgett@digium.com>

	* /: Merged revisions 181898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 Just
	  recording the v1.4 change in trunk since it originally came from
	  here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500
	  (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated
	  property when generating the version string. Copied the
	  make_version file from Asterisk trunk. ........

2009-03-12 21:43 +0000 [r181769-181846]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Run the macro on the queue member's channel
	  when he answers, not the caller's channel.

	* /, channels/chan_sip.c: Merged revisions 181768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar
	  2009) | 22 lines Properly send a 487 on an INVITE we have not
	  responded to if we receive a BYE. If we receive an INVITE from an
	  endpoint and then later receive a BYE from that same endpoint
	  before we have sent a final response for the INVITE, then we need
	  to respond to the INVITE with a 487. There was logic in the code
	  prior to this commit which seemed to exist solely to handle this
	  situation, but there was one condition in an if statement which
	  was incorrect. The only way we would send a 487 was if the
	  sip_pvt had no owner channel. This made no sense since we created
	  the owner channel when we received the INVITE, meaning that the
	  majority of the time we would never send the 487. The 487 being
	  sent should not rely on whether we have created a channel. Its
	  delivery should be dependent on the current state of the initial
	  INVITE transaction. With this commit, that logic is now correctly
	  in place. (closes issue #14149) Reported by: legranjl Patches:
	  14149.patch uploaded by mmichelson (license 60) Tested by:
	  legranjl ........

2009-03-12 17:32 +0000 [r181731]  Tilghman Lesher <tlesher@digium.com>

	* main/translate.c: Adjust translation table column widths based
	  upon the translation times. Previously, only 5 columns were
	  displayed, and if a translation time exceeded 99,999 useconds, it
	  would be displayed as 0, instead of its actual time. (closes
	  issue #14532) Reported by: pj Patches:
	  20090311__bug14532.diff.txt uploaded by tilghman (license 14)
	  Tested by: pj

2009-03-12 16:56 +0000 [r181612-181665]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 181664 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar
	  2009) | 2 lines Fix incorrect usage of strncasecmp... I really
	  meant to use strcasecmp. ........

	* /, res/res_musiconhold.c: Merged revisions 181659-181660 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8
	  lines Fix another scenario where depending on configuration the
	  stream would not get read. For custom commands we don't know
	  whether the audio is coming from a stream or not so we are going
	  to have to read the data despite no channels. (closes issue
	  #14416) Reported by: caspy ........ r181660 | file | 2009-03-12
	  13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in
	  previous commit. ........

	* /, res/res_musiconhold.c: Merged revisions 181655 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar
	  2009) | 10 lines Fix issue with streaming MOH failing if nobody
	  is listening. When a music class is setup to actually provide
	  music on hold from a stream we need to constantly read audio from
	  it since it will constantly be providing audio. This is now done
	  despite there being no channels listening to it. (closes issue
	  #14416) Reported by: caspy ........

	* apps/app_dial.c: Fix crash when sleep and retries argument was
	  not given to RetryDial application. (closes issue #14647)
	  Reported by: sherpya

2009-03-12 01:33 +0000 [r181542-181577]  Richard Mudgett <rmudgett@digium.com>

	* build_tools/make_version: Whitespace chages.

	* build_tools/make_version: Use the correct branch integrated
	  property when generating the version string

2009-03-11 23:14 +0000 [r181499]  Michiel van Baak <michiel@vanbaak.info>

	* configs/sip.conf.sample: Provide correct hint to debug SIP
	  trouble in the default config (closes issue #14646) Reported by:
	  strk

2009-03-11 22:25 +0000 [r181465]  Russell Bryant <russell@digium.com>

	* main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable
	  thread-safe.

2009-03-11 22:20 +0000 [r181444]  Jason Parker <jparker@digium.com>

	* /, configure, configure.ac: Merged revisions 181436 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar
	  2009) | 4 lines Allow prefix to set localstatedir (when used and
	  different from the default). This is similar to the /etc change
	  that was made for the non-FreeBSD case. ........

2009-03-11 22:14 +0000 [r181424-181428]  Russell Bryant <russell@digium.com>

	* main/channel.c: Make handling of the BRIDGEPVTCALLID variable
	  thread-safe.

	* main/channel.c, /: Merged revisions 181423 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009)
	  | 9 lines Make code that updates BRIDGEPEER variable thread-safe.
	  It is not safe to read the name field of an ast_channel without
	  the channel locked. This patch fixes some places in channel.c
	  where this was being done, and lead to crashes related to
	  masquerades. (closes issue #14623) Reported by: guillecabeza
	  ........

2009-03-11 17:34 +0000 [r181371]  David Vossel <dvossel@digium.com>

	* channels/iax2-parser.h, /, channels/chan_iax2.c: Merged revisions
	  181340 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009)
	  | 11 lines encrypted IAX2 during packet loss causes decryption to
	  fail on retransmitted frames If an iax channel is encrypted, and
	  a retransmit frame is sent, that packet's iseqno is updated while
	  it is encrypted. This causes the entire frame to be corrupted.
	  When the corrupted frame is sent, the other side decrypts it and
	  sends a VNAK back because the decrypted frame doesn't make any
	  sense. When we get the VNAK, we look through the sent queue and
	  send the same corrupted frame causing a loop. To fix this,
	  encrypted frames requiring retransmission are decrypted, updated,
	  then re-encrypted. Since key-rotation may change the key held by
	  the pvt struct, the keys used for encryption/decryption are held
	  within the iax_frame to guarantee they remain correct. (closes
	  issue #14607) Reported by: stevenla Tested by: dvossel Review:
	  http://reviewboard.digium.com/r/192/ ........

2009-03-11 17:26 +0000 [r181345]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 181328 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) |
	  14 lines Fix issue where an attended transfer could not be
	  completed under a rare scenario. When completing an attended
	  transfer chan_sip does a check to make sure the extension in the
	  URI portion of the Refer-To header is a local valid extension. We
	  don't actually need to check this since we know for sure the
	  other channel is already up and talking to the extension. Some
	  devices do not put the extension in the Refer-To header either,
	  which can cause the extension check to fail. We now no longer do
	  this check if it is an attended transfer. (closes issue #14628)
	  Reported by: sverre Patches: 14628.diff uploaded by file (license
	  11) ........

2009-03-11 17:04 +0000 [r181301]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/astobj2.h: Turn off malloc debugging of astobj2,
	  since it apparently doesn't work too well during startup.

2009-03-11 16:40 +0000 [r181296]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 181295 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9
	  lines Fix a problem with inband DTMF detection on outgoing SIP
	  calls when dtmfmode=auto. When dtmfmode was set to auto the
	  inband DTMF detector was not setup on outgoing SIP calls. This
	  caused inband DTMF detection to fail. The inband DTMF detector is
	  now setup for both dtmfmode inband and auto. (closes issue
	  #13713) Reported by: makoto ........

2009-03-11 16:19 +0000 [r181292]  Russell Bryant <russell@digium.com>

	* doc/google-soc2009-ideas.txt: Replace contents of this doc with a
	  pointer to its new home

2009-03-11 14:28 +0000 [r181244]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix segfault when dialing a typo'd queue If
	  trying to dial a non-existent queue, there would be a segfault
	  when attempting to access q->weight, even though q was NULL. This
	  problem was introduced during the queue-reset merge and thus only
	  affects trunk. (closes issue #14643) Reported by: alecdavis

2009-03-11 13:44 +0000 [r181210]  Joshua Colp <jcolp@digium.com>

	* apps/app_confbridge.c: Don't play the "you are about to be placed
	  into the conference" and "the leader has left the conference"
	  sounds if the quiet option is enabled. (reported by Vadim Lebedev
	  on the asterisk-dev list)

2009-03-11 04:06 +0000 [r181135]  Jeff Peeler <jpeeler@digium.com>

	* utils/Makefile, include/asterisk/utils.h,
	  include/asterisk/astmm.h, channels/chan_sip.c,
	  channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c,
	  pbx/pbx_config.c: Fix malloc debug macros to work properly with
	  h323. The main problem here was that cstdlib was undefining free
	  thereby causing the proper debug macros to not be used.
	  ast_h323.cxx has been changed to call ast_free instead to avoid
	  the issue. A few other issues were addressed: - There were a few
	  instances of functions improperly passing ast_free instead of
	  ast_free_ptr. - Some clean up was done to avoid the debug macros
	  intentionally being redefined. (copied below from Kevin's commit,
	  appreciate the help) - disable astmm.h from doing anything when
	  STANDALONE is defined, which is used by the tools in the utils/
	  directory that use parts of Asterisk header files in hackish
	  ways; also ensure that utils/extconf.c and utils/conf2ael.c are
	  compiled with STANDALONE defined. (closes issue #13593) Reported
	  by: pj

2009-03-11 02:25 +0000 [r181099]  Russell Bryant <russell@digium.com>

	* doc/google-soc2009-ideas.txt: tabs to spaces

2009-03-11 00:49 +0000 [r181032-181033]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add missing comment that quotes RFC 3891

	* /, channels/chan_sip.c: Merged revisions 181029,181031 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar
	  2009) | 9 lines Fix incorrect tag checking on transfers when
	  pedantic=yes is enabled. (closes issue #14611) Reported by:
	  klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt
	  uploaded by klaus3000 (license 65) Tested by: klaus3000 ........
	  r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar
	  2009) | 3 lines Remove unused variables. ........

2009-03-11 00:29 +0000 [r181027-181028]  Tilghman Lesher <tlesher@digium.com>

	* main/strings.c, main/hashtab.c, include/asterisk/astobj2.h,
	  main/heap.c, include/asterisk/strings.h,
	  include/asterisk/hashtab.h, main/astobj2.c,
	  include/asterisk/heap.h: Add MALLOC_DEBUG to various utility
	  APIs, so that memory leaks can be tracked back to their source.
	  (related to issue #14636)

	* main/pbx.c: Spacing changes only

2009-03-10 22:03 +0000 [r180944]  Jason Parker <jparker@digium.com>

	* /, configure, configure.ac, autoconf/ast_prog_sed.m4,
	  autoconf/ast_check_gnu_make.m4: Merged revisions 180941 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) |
	  1 line Make things happier when using autoconf 2.62+ ........

2009-03-10 22:03 +0000 [r180935-180942]  Russell Bryant <russell@digium.com>

	* doc/google-soc2009-ideas.txt: Add some notes on getting in
	  contact with the dev community

	* doc/google-soc2009-ideas.txt: Remove difficulty and language
	  specifiers

	* doc/google-soc2009-ideas.txt: Expand upon documentation of
	  manager event project

2009-03-10 21:15 +0000 [r180898]  Michiel van Baak <michiel@vanbaak.info>

	* CHANGES: list the move of the astvarrundir from /var/run to
	  /var/run/asterisk (actually its $(localstatedir)/run/asterisk
	  Makes setups with asterisk as non-root easier to manage because
	  you can setup permissions on this dir instead of touching a file
	  and setting permissions on that. Files that come to mind are
	  asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell

2009-03-10 19:36 +0000 [r180859-180862]  Russell Bryant <russell@digium.com>

	* doc/google-soc2009-ideas.txt: add more projects

	* doc/google-soc2009-ideas.txt: add more project ideas

2009-03-10 14:40 +0000 [r180800]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Reset the thread local string buffer when
	  handling the UserEvent action. (closes issue #14593) Reported by:
	  JimDickenson

2009-03-09 22:00 +0000 [r180750]  Russell Bryant <russell@digium.com>

	* doc/google-soc2009-ideas.txt: Add current mentors list, and first
	  pass on a project list broken out of "PineMango" I will work on
	  adding projects that have been sent to be via email tomorrow.

2009-03-09 20:58 +0000 [r180719]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/rtp.h, include/asterisk/extconf.h,
	  main/devicestate.c, include/asterisk/tcptls.h, main/enum.c,
	  include/asterisk/callerid.h, include/asterisk/doxyref.h,
	  include/asterisk/event.h, include/asterisk/audiohook.h,
	  include/asterisk/dsp.h, include/asterisk/timing.h,
	  include/asterisk/udptl.h, include/asterisk/dlinkedlists.h,
	  include/asterisk/utils.h, include/asterisk/devicestate.h,
	  include/asterisk/taskprocessor.h, include/asterisk/enum.h,
	  include/asterisk/astobj2.h, include/asterisk/config.h,
	  include/asterisk/channel.h, include/asterisk/manager.h,
	  include/asterisk/heap.h, include/asterisk/logger.h,
	  include/asterisk/http.h, include/asterisk/res_odbc.h,
	  include/asterisk/app.h, main/tcptls.c,
	  include/asterisk/linkedlists.h, include/asterisk/sched.h,
	  include/asterisk/datastore.h, include/asterisk/lock.h,
	  include/asterisk/pbx.h, include/asterisk/dnsmgr.h: Add Doxygen
	  documentation for API changes from 1.6.0 to 1.6.1 Copied from my
	  review board description: This is a continuation of the API
	  changes documentation started for describing changes between
	  releases. Most of the API changes were pretty simple needing only
	  to be brought to attention via the new "Asterisk API Changes"
	  list. However, if you see anything that needs further explanation
	  feel free to supplement what is there. The current method of
	  documenting is to add (in the header file): \version <ver number>
	  <description of changes> and then to add the function to the
	  change list in doxyref.h on the AstAPIChanges page. I also made
	  sure all the functions that were newly added were tagged with
	  \since 1.6.1. I think this is a good habit to start both for the
	  historical aspect as well as for the future ability to easily add
	  a "New Asterisk API" page. Review:
	  http://reviewboard.digium.com/r/190/

2009-03-09 14:14 +0000 [r180684]  Russell Bryant <russell@digium.com>

	* doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas
	  list

2009-03-07 15:36 +0000 [r180641]  Russell Bryant <russell@digium.com>

	* contrib/asterisk-ng-doxygen: Make some minor updates to the
	  doxygen configuration - add bridges directory to be processed -
	  add some res/ subdirs - alphabetize subdirs - use consistent
	  indentation

2009-03-06 18:25 +0000 [r180579]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 180567 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri,
	  06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when
	  IMAP storage is enabled. ........

2009-03-06 17:26 +0000 [r180534]  David Vossel <dvossel@digium.com>

	* /, main/enum.c: Merged revisions 180532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
	  | 9 lines Fix handling of backreferences for ENUM lookups enum.c
	  did not handle regex backtraces correctly. The '\1' in the regex
	  is a backreference that requires a pattern match to be inserted.
	  The way the code used to work is that it would find the
	  backreference and insert the entire input string minus the '+'.
	  This is incorrect. The regexec() function takes in a variable
	  called pmatch which is an array of structs containing the start
	  and end indexes for each backreference substring. The original
	  code actually passed the pmatch array pointer into regexec but
	  never did anything with it. Now when a backtrace is found, the
	  backtrace number is looked up in the pmatch array and the correct
	  substring is inserted. (closes issue #14576) Reported by:
	  chris-mac Review: http://reviewboard.digium.com/r/187/ ........

2009-03-05 23:26 +0000 [r180383-180465]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 180464 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu,
	  05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when
	  identical mailbox names were defined in separate contexts. There
	  was a fix put in a while back so that an X-Asterisk-VM-Context
	  message header was added to stored IMAP voicemails. This would
	  allow for us to differentiate if the same mailbox name was used
	  in multiple contexts. The problem still left was that not all
	  places where messages were retrieved actually attempted to use
	  this header for information when retrieving messages. This commit
	  fixes that so that MWI and message retrieval from VoiceMailMain
	  work as expected. (closes issue #13853) Reported by: vicks1
	  Patches: 13853_v2.patch uploaded by mmichelson (license 60)
	  Tested by: lmadsen ........

	* /, configs/voicemail.conf.sample, apps/app_voicemail.c: Merged
	  revisions 180380 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
	  2009) | 25 lines Fix broken mailbox parsing when searchcontexts
	  option is enabled. When using the searchcontexts option in
	  voicemail.conf, the code made the assumption that all mailbox
	  names defined were unique across all contexts. However, the code
	  did nothing to actually enforce this assumption, nor did it do
	  anything to alert a user that he may have created an ambiguity in
	  his voicemail.conf file by defining the same mailbox name in
	  multiple contexts. With this change, we now will issue a nice
	  long warning if searchcontexts is on and we encounter the same
	  mailbox name in multiple contexts and ignore any duplicates after
	  the first box. Whether searchcontexts is enabled or not, if we
	  come across a duplicate mailbox in the same context, then we will
	  issue a warning and ignore the duplicated mailbox. I have also
	  added a small note to voicemail.conf.sample in the explanation
	  for searchcontexts explaining that you cannot define the same
	  mailbox in multiple contexts if you have enabled the option.
	  (closes issue #14599) Reported by: lmadsen Patches: 14599.patch
	  uploaded by mmichelson (license 60) (with slight modification)
	  Tested by: lmadsen ........

2009-03-05 19:05 +0000 [r180382]  Michiel van Baak <michiel@vanbaak.info>

	* Makefile: Make sure we terminate the first s| command so we can
	  actually produce correct files.

2009-03-05 18:29 +0000 [r180373]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c, /, include/asterisk/frame.h, main/rtp.c: Merged
	  revisions 180372 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
	  2009) | 9 lines Fix problems when RTP packet frame size is
	  changed During some code analysis, I found that calling
	  ast_rtp_codec_setpref() on an ast_rtp session does not work as
	  expected; it does not adjust the smoother that may on the RTP
	  session, in fact it summarily drops it, even if it has data in
	  it, even if the current format's framing size has not changed.
	  This is not good. This patch changes this behavior, so that if
	  the packetization size for the current format changes, any
	  existing smoother is safely updated to use the new size, and if
	  no smoother was present, one is created. A new API call for
	  smoothers, ast_smoother_reconfigure(), was required to implement
	  these changes. Review: http://reviewboard.digium.com/r/184/
	  ........

2009-03-05 18:18 +0000 [r180369]  Joshua Colp <jcolp@digium.com>

	* channels/chan_bridge.c (added), main/Makefile,
	  bridges/bridge_simple.c, bridges/bridge_softmix.c,
	  include/asterisk/channel.h, bridges/bridge_multiplexed.c,
	  CHANGES, Makefile, include/asterisk/bridging_technology.h
	  (added), bridges (added), bridges/bridge_builtin_features.c,
	  include/asterisk/bridging_features.h (added),
	  include/asterisk/bridging.h (added), apps/app_confbridge.c
	  (added), main/bridging.c (added), bridges/Makefile: Merge phase 1
	  support for the new bridging architecture. This commit brings in
	  the bridging core, bridging technologies, and the ConfBridge
	  application. For usage information on the ConfBridge application
	  please see the output of "core show application ConfBridge" from
	  the CLI. For API documentation please see the doxygen page
	  describing the architecture and the documentation for each API
	  call. Review: http://reviewboard.digium.com/r/93/

2009-03-05 06:21 +0000 [r180304-180334]  Tilghman Lesher <tlesher@digium.com>

	* contrib/editors/asterisk.vim: Also highlight the preamble and
	  postamble

	* contrib/editors/ael.vim (added), contrib/editors/asterisk.vim
	  (added), contrib/editors (added), contrib/editors/asteriskvm.vim
	  (added): Add syntax coloring files for Vim, including a new one
	  for AEL

2009-03-04 21:01 +0000 [r180261]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Resolve object matching issues related to
	  the removal of the sip_user object. Previously, chan_sip had both
	  sip_peer and sip_user objects in memory. A patch went in to
	  remove sip_user to simplify the code, since everything could be
	  done with just sip_peer. This patch resolves some regressions
	  found that were introduced by those changes. This code comes from
	  svn/asterisk/team/group/sip-object-matching/. Here is a list of
	  the changes that have been made: 1) When doing a match by name
	  with the find_peer() function, make it much easier to specify
	  which objects should be matched by having a parameter that
	  specifies exactly which object types should be considered. Also,
	  update find_by_name() to handle this parameter. Finally, update
	  all code to use the new option values. 2) When looking up an
	  object for an outbound request by name, consider peers only.
	  (create_addr()) 3) Only match peers on an incoming registration
	  request. 4) When doing authentication (except for SUBSCRIBE),
	  look up users by name, instead of all objects by name. 5) When
	  doing authentication (except for SUBSCRIBE), after looking for a
	  user by name, look for a peer by IP address, instead of all
	  objects by IP address. 6) When handling the SIP qualify CLI
	  command or manager action, look for a peer by name, instead of
	  any object by name. 7) When handling the SIP unregister CLI
	  command, look for a peer by name, instead of any object by name.
	  9) In sip_do_debug_peer(), search for a peer by name, instead of
	  any object by name. 9) When handling the SIPPEER() dialplan
	  function, search for a peer by name, instead of any object by
	  name. 10) In the following session timer related functions,
	  st_get_se(), st_get_refresher(), and st_get_mode(), when looking
	  for an object for a given sip_pvt using pvt->peername, look for a
	  peer by name, instead of any object by name. 11) Fix build_peer()
	  to properly handle the case where separate type=peer and
	  type=user entries were specified in sip.conf. (closes issue
	  #14505) Reported by: lmadsen Review:
	  http://reviewboard.digium.com/r/172/

2009-03-04 20:48 +0000 [r180259]  Tilghman Lesher <tlesher@digium.com>

	* main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c,
	  main/alaw.c: Spacing changes only

2009-03-04 19:24 +0000 [r180195]  Joshua Colp <jcolp@digium.com>

	* /, main/callerid.c: Merged revisions 180194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
	  lines Look for the number in a callerid string starting from the
	  end. This way a value using <> can exist in the name portion.
	  (issue #AST-194) ........

2009-03-04 17:03 +0000 [r180155]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic"
	  pickups to work when we wish to ignore the context When the
	  subscription context for a call pickup subscription differs from
	  the context of the call pickup target, there's not an easy way to
	  divine what context should be used for the pickup. The way to
	  work around this is to use PICKUPMARK as the context for the
	  pickup. This has been documented in the sip.conf.sample file
	  (ABE-1708) closes issue #14567 submitted by: alecdavis

2009-03-04 14:39 +0000 [r180120]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Remove duplicate 'k' and 'K' Dial options.
	  (closes issue #14601) Reported by: alecdavis Patches:
	  app_dial.optionk.diff.txt uploaded by alecdavis (license 585)

2009-03-03 23:35 +0000 [r180079]  Steve Murphy <murf@digium.com>

	* utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by
	  mistake. Already done to 1.6.x

2009-03-03 23:21 +0000 [r180032]  David Vossel <dvossel@digium.com>

	* main/channel.c, include/asterisk/app.h, apps/app_read.c,
	  main/app.c: app_read does not break from prompt loop with user
	  terminated empty string In app.c, ast_app_getdata is called to
	  stream the prompts and receive DTMF input. If ast_app_getdata()
	  receives an empty string caused by the user inputing the end of
	  string character, in this case '#', it should break from the
	  prompt loop and return to app_read, but instead it cycles through
	  all the prompts. I've added a return value for this special case
	  in ast_readstring() which uses an enum I've delcared in apps.h.
	  This enum is now used as a return value for ast_app_getdata().
	  (closes issue #14279) Reported by: Marquis Patches:
	  fix_app_read.patch uploaded by Marquis (license 32)
	  read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested
	  by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/

2009-03-03 22:49 +0000 [r180007]  Mark Michelson <mmichelson@digium.com>

	* /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions
	  180006 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
	  2009) | 17 lines Clarify some documentation of queues.conf.sample
	  It had always been possible to explicitly specify a "blank" value
	  for a sound file in queues.conf and have no sound played back.
	  The problem with this is that it would result in some ugly CLI
	  warnings from file.c. This commit introduces a check when playing
	  a file in app_queue to see if the name of the file is zero-length
	  and return early if that is the case. Also, the ability to
	  specify the blank sound files in queues.conf is now mentioned
	  more clearly in queues.conf.sample (closes issue #14227) Reported
	  by: caspy ........

2009-03-03 22:12 +0000 [r179973]  Steve Murphy <murf@digium.com>

	* utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h,
	  main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl,
	  main/ast_expr2.c: Merged revisions 179807 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
	  work to do to port these changes to trunk; the check_expr stuff
	  hasn't been updated here for quite some time, it appears. I added
	  some more tests to the check_expr2 suite. I had to play around
	  with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
	  ast_expr2.y so as not to conflict structure with aelparse.
	  ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
	  2009) | 19 lines These changes allow AEL to better check ${}
	  constructs within $[...], that are concatenated with text. I
	  modified and added rules in ast_expr2.fl to better handle the
	  concatenations. I added some default routines to ast_expr2.y so
	  the standalone would compile. It also looks like I haven't run
	  this thru bison since 2.1, so it's good to get this updated. The
	  Makefile has comments added now for check_expr2 and check_expr to
	  explain what they are for, and how to run them. The testexpr2s
	  stuff has been removed, in favor of check_expr2. expr2.testinput
	  has been updated to include the two expressions that inspired
	  these changes (from mcnobody on #asterisk this morning) The
	  regression has been run and all looks well. ........

2009-03-03 22:01 +0000 [r179972]  David Vossel <dvossel@digium.com>

	* apps/app_meetme.c: app_meetme not setting filename and fileformat
	  correctly for realtime When app_meetme finds a realtime
	  conference, it doesn't get the filename and fileformat correctly
	  when 'r' is set. Now app_meetme first checks to see if fileformat
	  and filename are declared in the db, if they're not it checks the
	  .conf file, if its not declared there either it then uses
	  defaults. (closes issue #14545) Reported by: dalbaech Patches:
	  app_meetme-realtime5.patch uploaded by dvossel (license 671)
	  Realtime_Conference_Record_workaround.txt uploaded by dalbaech
	  (license 705) Tested by: dvossel, dalbaech Review:
	  http://reviewboard.digium.com/r/180/

2009-03-03 20:59 +0000 [r179937]  Mark Michelson <mmichelson@digium.com>

	* res/res_timing_dahdi.c, doc/timing.txt (added): Add documentation
	  for timing modules used in Asterisk This document specifies the
	  timing modules available in Asterisk beginning with Asterisk
	  1.6.1. The document goes into detail about the differences
	  between each and gives a general overview of what timing is used
	  for in Asterisk. There is also a section which can be used to
	  help customize your setup or to troubleshoot timing issues you
	  may have. I also added messages to the DAHDI timing test used in
	  res_timing_dahdi.c that points to this new documentation if
	  people experience problems. Big thanks to all who contributed
	  comments on this. (closes issue #14490) Reported by: mmichelson
	  Patches: timing.txt uploaded by mmichelson (license 60) Review:
	  http://reviewboard.digium.com/r/164/

2009-03-03 20:02 +0000 [r179903]  Brian Degenhardt <bmd@digium.com>

	* apps/app_directed_pickup.c: fix a leaked channel lock (and future
	  deadlock) when we try to pick up our own channel

2009-03-03 18:28 +0000 [r179841]  Joshua Colp <jcolp@digium.com>

	* /, main/features.c: Merged revisions 179840 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
	  lines Do not assume that the bridge_cdr is still attached to the
	  channel when the 'h' exten is finished executing. It is possible
	  for a masquerade operation to occur when the 'h' exten is
	  operating. This operation moves the CDR records around causing
	  the bridge_cdr to no longer exist on the channel where it is
	  expected to. We can not safely modify it afterwards because of
	  this, so don't even try. (closes issue #14564) Reported by: meric
	  ........

2009-03-03 17:03 +0000 [r179745]  Mark Michelson <mmichelson@digium.com>

	* pbx/pbx_spool.c: Convert pbx_spool to use string fields instead
	  of statically-sized buffers. In tests run after making this
	  conversion, I noticed an approximate 85% reduction in memory
	  usage for call file processing. Review:
	  http://reviewboard.digium.com/r/168/

2009-03-03 16:47 +0000 [r179742]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 179741 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
	  | 6 lines Ensure chan->fdno always gets reset to -1 after
	  handling a channel fd event. Since setting fdno to -1 had to be
	  moved, a couple of other code paths that do process an fd event
	  return early and do not pass through the code path where it was
	  moved to. So, set it to -1 in a few other places, too. ........

2009-03-03 15:13 +0000 [r179675]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Please prefix default values with DEFAULT

2009-03-03 14:40 +0000 [r179672]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 179671 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
	  lines Move where fdno is set to the default value to *after* the
	  read callback of the channel driver is called. We have to do this
	  as the underlying channel driver may need the fdno value to
	  determine what to read. ........

2009-03-03 13:54 +0000 [r179609]  Russell Bryant <russell@digium.com>

	* main/channel.c, /: Merged revisions 179608 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
	  | 9 lines Make it easier to detect an improper call to
	  ast_read(). When you call ast_waitfor() on a channel, the index
	  into the channel fds array that holds the file descriptor that
	  poll() determines has input available is stored in fdno. This
	  patch clears out this value after a call to ast_read() and also
	  reports errors if ast_read() is called without an fdno set. From
	  a discussion on the asterisk-dev list. ........

2009-03-03 00:01 +0000 [r179537]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /: Merged revisions 179536 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
	  | 15 lines Fix bridging regression from commit 176701 This fixes
	  a bad regression where the bridge would exit after an attended
	  transfer was made. The problem was due to nexteventts getting set
	  after the masquerade which caused the bridge to return
	  AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
	  tim_ringenbach ........

2009-03-02 23:36 +0000 [r179533]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 179532 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
	  | 40 lines Move ast_waitfor() down to avoid the results of the
	  API call becoming stale. This call to ast_waitfor() was being
	  done way too soon in this section of code. Specifically, there
	  was code in between the call to waitfor and the code that uses
	  the result that puts the channel in autoservice. By putting the
	  channel in autoservice, the previous results of ast_waitfor()
	  become meaningless, as the autoservice thread will do it's own
	  ast_waitfor() and ast_read() on the channel. So, when we came
	  back out of autoservice and eventually hit the block of code that
	  calls ast_read() on the channel, there may not actually be any
	  input on the channel available. Even though the previous call to
	  ast_waitfor() in app_meetme said there was input, the autoservice
	  thread has since serviced the channel for some period of time.
	  This bug manifested itself while dvossel was doing some testing
	  of MeetMe in Asterisk trunk. He was using the timerfd timing
	  module. When the code hit ast_read() erroneously, it determined
	  that it must have been called because of input on the timer fd,
	  as chan->fdno was set to AST_TIMING_FD, since that was the cause
	  of the last legitimate call to ast_read() done by autoservice. In
	  this test, an IAX2 channel was calling into the MeetMe
	  conference. It was _much_ more likely to be seen with an IAX2
	  channel because of the way audio is handled. Every audio frame
	  that comes in results in a call to ast_queue_frame(), which then
	  uses ast_timer_enable_continuous() to notify the channel thread
	  that a frame is waiting to be handled. So, the chances of
	  ast_waitfor() indicating that a channel needs servicing due to a
	  timer event on an IAX2 event is very high. Finally, it is
	  interesting to note that if a different timing interface was
	  being used, this bug would probably not be noticed. When
	  ast_read() is called and erroneously thinks that there is a timer
	  event to handle, it calls the ast_timer_ack() function. The
	  pthread and dahdi timing modules handle the ack() function being
	  called when there is no event by simply ignoring it. In the case
	  of the timerfd module, it results in a read() on the timer fd
	  that will block forever, as there is no data to read. This caused
	  Asterisk to lock up very quickly. Thanks to dvossel and
	  mmichelson for the fun debugging session. :-) ........

2009-03-02 23:10 +0000 [r179469]  Tilghman Lesher <tlesher@digium.com>

	* /, main/app.c: Merged revisions 179468 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
	  | 10 lines When ending a recording with silence detection,
	  remember to reduce the duration. The end of the recording is
	  correspondingly trimmed, but the duration was not trimmed by the
	  number of seconds trimmed, so the saved duration was necessarily
	  longer than the actual soundfile duration. (closes issue #14406)
	  Reported by: sasargen Patches: 20090226__bug14406.diff.txt
	  uploaded by tilghman (license 14) Tested by: sasargen ........

2009-03-02 23:06 +0000 [r179462-179465]  Russell Bryant <russell@digium.com>

	* res/res_timing_timerfd.c: Fix a reference leak in
	  timerfd_set_rate(). (found during a debugging session with
	  dvossel and mmichelson.)

	* main/channel.c, /: Merged revisions 179461 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
	  | 8 lines Ensure that only one thread is calling ast_settimeout()
	  on a channel at a time. For example, with an IAX2 channel, you
	  can have both the channel thread and the chan_iax2 processing
	  threads calling this function, and doing so twice at the same
	  time is a bad thing. (Found in a debugging session with dvossel
	  and mmichelson) ........

2009-03-02 20:16 +0000 [r179396]  Jason Parker <jparker@digium.com>

	* /, main/editline/configure, main/editline/np/unvis.c,
	  main/editline/sys.h, main/editline/configure.in: 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 ........

2009-03-02 17:18 +0000 [r179361]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe
	  if db is not loaded)

2009-03-02 14:28 +0000 [r179291-179323]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Do not try to remove a registration
	  scheduled item if the scheduler context has already been
	  destroyed. (closes issue #14580) Reported by: alecdavis

	* main/audiohook.c: Fix issue where changing the volume of both
	  directions of audio did not work. (closes issue #14574) Reported
	  by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK
	  (license 545)

2009-03-01 23:25 +0000 [r179219-179254]  Mark Michelson <mmichelson@digium.com>

	* apps/app_speech_utils.c: Swap reversed timevals. This was pointed
	  out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!

	* channels/chan_sip.c: Properly free memory and remove scheduler
	  entries when a transmission failure occurs. Previously, only the
	  "data" field of the sip_pkt created during __sip_reliable_xmit
	  was freed when XMIT_ERROR was returned by __sip_xmit. When
	  retrans_pkt was called, this inevitably resulted in the reading
	  and writing of freed memory. XMIT_ERROR is a condition meaning
	  that we don't want to attempt resending the packet at all. The
	  proper action to take is to remove the scheduler entry we just
	  created, free the packet's data as well as the packet itself, and
	  unlink it from the list of packets on the sip_pvt structure.
	  (closes issue #14455) Reported by: Nick_Lewis Patches:
	  14455.patch uploaded by mmichelson (license 60) Tested by:
	  Nick_Lewis

2009-02-27 21:47 +0000 [r179164]  Russell Bryant <russell@digium.com>

	* res/res_ais.c, doc/distributed_devstate.txt,
	  configs/ais.conf.sample: Mark res_ais as experimental, as the
	  binary event format is subject to change.

2009-02-27 21:32 +0000 [r179161]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_sqlite3_custom.c: If config file is blank, don't load
	  module. (Closes issue #14563)

2009-02-27 21:23 +0000 [r179154]  Russell Bryant <russell@digium.com>

	* UPGRADE.txt: Add a note about the ordering of entries in sip.conf
	  in 1.6.1.

2009-02-27 20:34 +0000 [r179122]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Add reload support to chan_skinny.
	  Special thanks goes to DEA who had to redo this patch twice
	  because we first put unload/load support in and later redid the
	  way we configure devices and lines. (closes issue #10297)
	  Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by
	  wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA
	  (license 3) With mods by me based on feedback from wedhorn and
	  Russell and seanbright Tested by: DEA, mvanbaak, pj Review:
	  http://reviewboard.digium.com/r/130/

2009-02-27 19:04 +0000 [r179057]  Jason Parker <jparker@digium.com>

	* doc/tex/channelvariables.tex: Update documentation for DIALEDTIME
	  and ANSWEREDTIME variables. (closes issue #14566) Reported by:
	  klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
	  klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
	  klaus3000 (license 65)

2009-02-27 15:51 +0000 [r179021]  Russell Bryant <russell@digium.com>

	* sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound
	  packages. In passing, also fix downloading SLIN16 extra sound
	  packages. (closes issue #14565) Reported by: jtodd

2009-02-27 03:45 +0000 [r178986]  Steve Murphy <murf@digium.com>

	* /, main/features.c, configs/features.conf.sample: 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! ........

2009-02-26 18:41 +0000 [r178919]  Tilghman Lesher <tlesher@digium.com>

	* main/features.c, CHANGES, configs/features.conf.sample: Sound
	  confirmation of call pickup success. (closes issue #13826)
	  Reported by: azielke Patches: pickupsound2-trunk.patch uploaded
	  by azielke (license 548) __20081124_bug_13826_updated.patch
	  uploaded by lmadsen (license 10) Tested by: lmadsen

2009-02-26 17:46 +0000 [r178871]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last
	  fix A return statement was missing which caused unexpected cli
	  output. issue #14479

2009-02-26 17:45 +0000 [r178828-178870]  Steve Murphy <murf@digium.com>

	* apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent
	  compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to
	  break my dev-mode build. Not a problem in 1.6.x.

	* /, main/features.c: 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 ........

2009-02-26 16:42 +0000 [r178801]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Fix an issue where the timer for file playback would
	  not be stopped if DAHDI was not installed. (closes issue #14541)
	  Reported by: grant

2009-02-26 15:50 +0000 [r178767]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: IAX2 prune realtime fix Iax2 prune realtime
	  had issues. If "iax2 prune realtime all" was called, it would
	  appear like the command was successful, but in reality nothing
	  happened. This is because the reload that was supposed to take
	  place checks the config files, sees no changes, and does nothing.
	  If there had been a change in the the config file, the realtime
	  users would have been marked for deletion and everything would
	  have been fine. Now prune_users() and prune_peers() are called
	  instead of reload_config() to prune all users/peers that are
	  realtime. These functions remove all users/peers with the
	  rtfriend and delme flags set. iax2_prune_realtime() also lacked
	  the code to properly delete a single friend. For example. if iax2
	  prune realtime <friend> was called, only the peer instance would
	  be removed. The user would still remain. (closes issue #14479)
	  Reported by: mousepad99 Review:
	  http://reviewboard.digium.com/r/176/

2009-02-26 15:40 +0000 [r178764]  Joshua Colp <jcolp@digium.com>

	* main/indications.c: Ensure there is a valid tone part before
	  trying to play tones. (closes issue #14558) Reported by:
	  alecdavis

2009-02-26 15:02 +0000 [r178733]  Olle Johansson <oej@edvina.net>

	* configs/res_snmp.conf.sample: Clarifications on the different
	  models and reference to further docs.

2009-02-26 13:39 +0000 [r178703-178704]  Kevin P. Fleming <kpfleming@digium.com>

	* README: another minor commit to test post-commit script changes
	  (now testing post-revprop-change as well, third try)

	* README: minor commit to test post-commit script changes

2009-02-25 19:49 +0000 [r178573-178607]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Picky, picky buildbots

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/stdtime/localtime.c: Use notification when timezone files
	  change and re-scan then. (closes issue #14300) Reported by:
	  jamessan Patches: 20090127__bug14300.diff.txt uploaded by
	  tilghman (license 14) 20090224__bug14300.diff uploaded by
	  jamessan (license 246) Tested by: jamessan Review:
	  http://reviewboard.digium.com/r/136/

	* res/res_odbc.c: Oops, wrong direction of command

2009-02-25 12:45 +0000 [r178509]  Russell Bryant <russell@digium.com>

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

2009-02-24 23:27 +0000 [r178375-178446]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/extensions.conf.sample: Merged revisions 178445 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
	  | 5 lines Add section about the #exec command in configuration
	  files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
	  with additional notes by tilghman (license 14) ........

	* main/asterisk.c: Apparently, a void cast doesn't override
	  warn_unused_result.

	* main/asterisk.c: The 3 possible errors with pipe(2) are all
	  impossible in this situation.

2009-02-24 20:39 +0000 [r178374]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 178373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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 ........

2009-02-24 20:06 +0000 [r178303-178342]  Tilghman Lesher <tlesher@digium.com>

	* utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the
	  process, instead of depending upon the astcanary process being
	  inherited by init.

	* utils/astcanary.c: Cause astcanary to exit if Asterisk exits
	  abnormally and doesn't kill astcanary. Also, add some
	  documentation supporting the use of astcanary. (closes issue
	  #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff
	  uploaded by KNK (license 545)

2009-02-24 17:42 +0000 [r178300]  David Vossel <dvossel@digium.com>

	* doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows
	  manager command to see if IAX link is trunked and encrypted.
	  Displays what kind of encryption is enabled as well. Manager
	  command "iaxpeers" now shows if a link is trunked and encrypted.
	  Instead of encryption saying simply "yes" or "no", it now
	  displays what type of encryption is enabled and if keyrotation is
	  on or not. (closes issue #14427) Reported by: snuffy Patches:
	  iax_show_trunks.diff uploaded by snuffy (license 35)
	  2009022200_iax2_show_trunkencryption.diff.txt uploaded by
	  mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review:
	  http://reviewboard.digium.com/r/173/

2009-02-24 15:18 +0000 [r178213]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 178205 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
	  lines Skip check for extension when subscribing for MWI. Since
	  the remote side is not actually subscribing to a specific
	  extension when subscribing for MWI just skip the check to see if
	  the extension exists. They can't use it to specify the mailbox
	  either since we require configuration of that in sip.conf (closes
	  issue #14531) Reported by: festr ........

2009-02-23 23:11 +0000 [r178142]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 178141 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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 ........

2009-02-23 21:02 +0000 [r178107]  Tilghman Lesher <tlesher@digium.com>

	* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
	  Permit emailsubject and emailbody to be set per mailbox. (closes
	  issue #14372) Reported by: fhackenberger Patches:
	  voicemail_individual_subject_and_body_1.6.1 uploaded by
	  fhackenberger (license 592) with additional fixes by Corydon76
	  (license 14)

2009-02-23 18:23 +0000 [r178061]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: update the new manager commands in
	  chan_skinny to match chan_sip's headers. requested by oej.

2009-02-23 17:59 +0000 [r178030]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Changes the way keyrotation is enabled by
	  default Key rotation was enabled by default by setting the global
	  encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this
	  is that if encryption is not enabled, and the encryption method
	  is set to anything except 0, the peer appears to have encryption
	  enabled when issuing a "iax2 show peers". Rather than have the
	  key rotation bit always set by default, it is now only set when
	  an encryption method is enabled. (closes issue #14523) Reported
	  by: mvanbaak

2009-02-23 17:48 +0000 [r178027]  Michiel van Baak <michiel@vanbaak.info>

	* CHANGES: list the addition of the SKINNY manager actions in the
	  CHANGES file.

2009-02-23 17:29 +0000 [r178022]  Russell Bryant <russell@digium.com>

	* tests/test_sched.c, main/sched.c: Fix a regression in scheduler
	  entry ordering, and add a regression test for it. (closes issue
	  #14522) Reported by: pj Tested by: russell

2009-02-22 23:04 +0000 [r177988]  Michiel van Baak <michiel@vanbaak.info>

	* doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of
	  manager commands to chan_skinny Added: SKINNYdevices
	  SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521)
	  Reported by: mvanbaak Review:
	  http://reviewboard.digium.com/r/170/

2009-02-21 15:59 +0000 [r177944]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: On update, test against the existence of
	  sipregs.

2009-02-21 14:37 +0000 [r177913]  Michiel van Baak <michiel@vanbaak.info>

	* main/asterisk.c: add extra check for sysinfo/sysctl (closes issue
	  #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff
	  uploaded by snuffy (license 35)

2009-02-21 14:16 +0000 [r177884]  Sean Bright <sean.bright@gmail.com>

	* main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace,
	  minor coding guideline fixes, and start beefing up the hashtab
	  documentation a bit.

2009-02-21 13:17 +0000 [r177855]  Russell Bryant <russell@digium.com>

	* include/asterisk/indications.h: Fix build issues on Solaris and
	  OpenBSD. (closes issue #14512) Reported by: snuffy

2009-02-21 13:13 +0000 [r177849-177852]  Michiel van Baak <michiel@vanbaak.info>

	* Makefile, contrib/init.d/rc.debian.asterisk,
	  contrib/init.d/rc.archlinux.asterisk,
	  contrib/scripts/safe_asterisk: set
	  ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When
	  running asterisk as non-root and without this patch the pidfile
	  wants to go into /var/run/asterisk.pid. This directory is not
	  writable for the non-root user and changing permissions is not an
	  option. Putting it in /var/run/asterisk/asterisk.pid makes it
	  possible to set permissions on the /var/run/asterisk dir so
	  everything works as it should be. Patched committed is based on
	  pabelanger's patch. (closes issue #13153) Reported by: pabelanger
	  Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by
	  mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/

	* channels/chan_sip.c: make chan_sip.c compile on OpenBSD again.

2009-02-20 23:02 +0000 [r177732-177787]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, /: 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 ........

	* /, include/asterisk/threadstorage.h: Merged revisions 177701 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
	  | 3 lines This exception does not appear to still be true for
	  Solaris 10, and OpenSolaris definitely needs it to be removed.
	  Fixed for snuff-home on -dev channel. ........

2009-02-20 20:29 +0000 [r177699]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* apps/app_fax.c: Make app_fax compatible with spandsp-0.0.6pre4
	  Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a
	  pages_transferred integer to indicate the number of pages
	  transferred (so far) during the fax session. The
	  spandsp-0.0.6pre4 release removed the pages_transferred integer
	  and replaced it with two different integers - pages_tx and
	  pages_rx. This revision uses the new integers for
	  spandsp-0.0.6pre4 while maintaining backwards compatibility for
	  previous spandsp releases.

2009-02-20 17:29 +0000 [r177661-177664]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/app.h, main/app.c, apps/app_system.c: 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

	* apps/app_voicemail.c: Oops, merge broke trunk

2009-02-20 00:35 +0000 [r177624]  Jeff Peeler <jpeeler@digium.com>

	* channels/chan_sip.c: Set sip_request ast_str data to NULL so
	  ast_str_copy allocates space properly in copy_request (issue
	  #14478) Reported by: erik_dedecker

2009-02-19 23:56 +0000 [r177595]  Steve Murphy <murf@digium.com>

	* /, main/Makefile, main/ast_expr2f.c: 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. ........ 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 ........

2009-02-19 22:33 +0000 [r177506-177537]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 177536 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19
	  Feb 2009) | 7 lines Fix up potential crashes, by reducing the
	  sharing between interactive and non-interactive threads. (closes
	  issue #14253) Reported by: Skavin Patches:
	  20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Skavin ........

	* doc/database_transactions.txt (added): Document how to use
	  database transactions

2009-02-19 16:45 +0000 [r177387]  Jeff Peeler <jpeeler@digium.com>

	* include/asterisk/channel.h: Fix another merge error from 176708

2009-02-19 16:38 +0000 [r177384]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_speech_utils.c: Merged revisions 177383 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb
	  2009) | 3 lines If we are able to create a speech structure unset
	  the ERROR variable in case it was previously set. (issue
	  #LUMENVOX-13) ........

2009-02-19 15:56 +0000 [r177356]  Jeff Peeler <jpeeler@digium.com>

	* main/features.c: Fix mismerge from revision 176708 pointed out by
	  Kaloyan Kovachev on the asterisk-dev mailing list. Thanks!

2009-02-19 00:26 +0000 [r177320]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES,
	  res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction
	  support

2009-02-19 00:08 +0000 [r177291]  Joshua Colp <jcolp@digium.com>

	* CHANGES: Update CHANGES file to include MWI subscription support
	  that was added some time ago.

2009-02-18 23:51 +0000 [r177287]  Tilghman Lesher <tlesher@digium.com>

	* main/strings.c: Handle negative length and eliminate a condition
	  that is always true.

2009-02-18 23:50 +0000 [r177286]  Steve Murphy <murf@digium.com>

	* /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177225 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) |
	  34 lines This patch fixes a regression of sorts that was
	  introduced in rev 24425. It basically fixes AST-190/ABE-1782.
	  What was wrong: the user has 6000 extensions in one context; and
	  then 6000 contexts, one per extension. The parser could only
	  handle about 4893 of the 6000 extens in the single context. This
	  was due to the regression I mentioned. To get rid of shift/reduce
	  conflicts, Luigi set up right-recursive lists for globals,
	  context elements, switch lists, and statements. Right recursive
	  lists got rid of the warnings, but instead, they use up a
	  tremendous amount of stack space when the lists are long. I saw
	  this a few years back, and resolved not to fix it until someone
	  complained. That day has arrived! After the changes were made, I
	  ran the regression test suite, and there were no problems. I took
	  the test case the user provided, and added 100,000 extensions to
	  the single context, that already had 6,000 extens in it. (I'll
	  see your 6, and raise you 100!) It takes a few minutes to read it
	  all in, check it and generate code for it, but no problems. So, I
	  think I can say that fundamentally, there are no longer any
	  limits on the number of items you can place in contexts,
	  statement blocks, switches, or globals, beyond your virt mem
	  constraints. ........

2009-02-18 23:09 +0000 [r177229]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c: 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.

2009-02-18 22:51 +0000 [r177226]  David Vossel <dvossel@digium.com>

	* main/features.c: 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/

2009-02-18 20:11 +0000 [r177162]  Jeff Peeler <jpeeler@digium.com>

	* channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4,
	  channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx,
	  channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added),
	  configure, channels/h323/compat_h323.h, configure.ac,
	  channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h,
	  channels/h323/cisco-h225.cxx: Modify h323 to build against PTLib
	  as well as the older PWLib Several changes in PTLib have occurred
	  requiring build time detection. Changes accounted for include the
	  library name change, config option change, install location
	  change, and a boolean type change which is handled by
	  ast_ptlib.h. Also, the sed check has been modified to properly
	  work with autoconf >= 2.62. (closes issue #14224) Reported by:
	  bergolth Patches: asterisk-autoconf-sed.patch uploaded by
	  bergolth (license 661) asterisk-pwlib-v3.patch uploaded by
	  bergolth (license 661) Tested by: jpeeler

2009-02-18 19:12 +0000 [r177101]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Re-add 'o' option to MeetMe, reverting rev
	  62297. Enabling this option by default proved to be a bad idea,
	  as the talker detection is not very reliable. So, make it
	  optional again, and off by default. (issue #13801) Reported by:
	  justdave

2009-02-18 19:05 +0000 [r177098]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/config.h: Merged revisions 177096 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009)
	  | 2 lines Document the return value of the update method (as
	  requested on -dev list) ........

2009-02-18 17:24 +0000 [r177035]  Doug Bailey <dbailey@digium.com>

	* main/utils.c: Fixed error where a check for an zero length,
	  terminated string was needed.

2009-02-18 17:11 +0000 [r177005]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix ordering of output for a ChannelUpdate
	  manager event. (closes issue #14497) Reported by: vinsik Patches:
	  chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)

2009-02-18 16:09 +0000 [r176948]  Doug Bailey <dbailey@digium.com>

	* main/utils.c: Need to take into account the \0 terminator of the
	  old string to determine the amount available.

2009-02-18 15:35 +0000 [r176943]  Steve Murphy <murf@digium.com>

	* main/pbx.c: 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

2009-02-18 06:14 +0000 [r176901-176904]  Russell Bryant <russell@digium.com>

	* include/asterisk/heap.h: Add example code for a heap traversal.

	* main/pbx.c: Fix a number of incorrect uses of strncpy(). The big
	  problem here is that the 3rd argument provided in these uses of
	  strncpy() did not reserve a byte for the null terminator, leaving
	  the potential for writing one byte past the end of the buffer.
	  Aside from this, there were coding guidelines violations with
	  regards to spacing, as well as hard coded lengths being used
	  instead of sizeof().

2009-02-18 02:55 +0000 [r176869]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* channels/chan_sip.c: T38 faxdetect should jump to the 'fax'
	  extension for incoming calls only The previous implementation of
	  T38 faxdetect resulted in both sides of the call jumping to a fax
	  extension when both sides had 't38pt_udptl=yes' and
	  'faxdetect=yes' in sip.conf and a 'fax' extension in the current
	  context. This revision will jump to a 'fax' extension on incoming
	  calls only.

2009-02-18 02:02 +0000 [r176841]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp.c: suppress smoothers for Siren codecs as well as Speex
	  and G.723.1

2009-02-17 22:52 +0000 [r176771]  Russell Bryant <russell@digium.com>

	* apps/app_milliwatt.c: Remove a dependency that no longer exists.

2009-02-17 22:28 +0000 [r176760]  Shaun Ruffell <sruffell@digium.com>

	* codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
	  with G723. This commit brings in the changes that were living out
	  on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder
	  branch. codec_dahdi.c now always uses signed linear as the simple
	  codec so that a soft g729 codec will not end up being preferred
	  to the hardware codec. There are also changes to allow
	  codec_dahdi.c to feed packets to the hardware in the native
	  sample size of the codec. This solves problems with choppy audio
	  when using G723.

2009-02-17 22:08 +0000 [r176708]  Jeff Peeler <jpeeler@digium.com>

	* main/channel.c, /, main/features.c, include/asterisk/channel.h:
	  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/ ........

2009-02-17 22:02 +0000 [r176706]  Mark Michelson <mmichelson@digium.com>

	* tests/test_sched.c: Use constants from inttypes.h to clear up
	  32-bit compilation errors

2009-02-17 21:59 +0000 [r176705]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* channels/chan_sip.c: create a UDPTL structure in
	  create_addr_from_peer() if it does not already exist for T38 This
	  is required to create a UDPTL structure in
	  create_addr_from_peer() to handle the scenario where
	  't38pt_udptl=yes' is not defined in the [general] section of
	  sip.conf but is defined the peer's context. I tested this patch
	  by enabling t38pt_udptl in the [general] section on one system
	  and only enabling t38pt_udptl in a peer's context on the system
	  sending a fax. Without the patch, the sending system will fail to
	  initiate T38 negotiation with the warning message, "No way to add
	  SDP without an UDPTL structure". When this patch is applied the
	  sending side will successfully initiate T38 negotiation.

2009-02-17 21:40 +0000 [r176697]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/frame.h: Clear up documentation of
	  AST_FRIENDLY_OFFSET in frame.h

2009-02-17 21:23 +0000 [r176669]  Tilghman Lesher <tlesher@digium.com>

	* /: Recorded merge of revisions 176661 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009)
	  | 9 lines Backport change to 1.4: Prior to masquerade, move the
	  group definitions to the channel performing the masq, so that the
	  group count lingers past the bridge. (closes issue #14275)
	  Reported by: kowalma Patches: 20090216__bug14275.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: kowalma ........

2009-02-17 21:22 +0000 [r176666]  Russell Bryant <russell@digium.com>

	* main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
	  res/res_timing_timerfd.c, include/asterisk/timing.h,
	  main/timing.c: Update the timing API to have better support for
	  multiple timing interfaces. 1) Add module use count handling so
	  that timing modules can be unloaded. 2) Implement unload_module()
	  functions for the timing interface modules. 3) Allow multiple
	  timing modules to be loaded, and use the one with the highest
	  priority value. 4) Report which timing module is being use in the
	  "timing test" CLI command. (closes issue #14489) Reported by:
	  russell Review: http://reviewboard.digium.com/r/162/

2009-02-17 21:14 +0000 [r176642]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_local.c: Prior to masquerade, move the group
	  definitions to the channel performing the masq, so that the group
	  count lingers past the bridge. (closes issue #14275) Reported by:
	  kowalma Patches: 20090216__bug14275.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: kowalma

2009-02-17 21:04 +0000 [r176632-176639]  Russell Bryant <russell@digium.com>

	* tests/test_sched.c (added), main/sched.c: Significantly improve
	  scheduler performance under high load. This patch changes the
	  scheduler to use a max-heap to store pending scheduler entries
	  instead of a fully sorted doubly linked list. When the number of
	  entries in the scheduler gets large, this will perform much
	  better. For much more detailed information on this change, see
	  the review request. Review: http://reviewboard.digium.com/r/160/

	* tests/test_heap.c (added): Add a test module for the heap
	  implementation. Review: http://reviewboard.digium.com/r/160/

	* main/Makefile, main/heap.c (added), include/asterisk/heap.h
	  (added): Add an implementation of the heap data structure. A heap
	  is a convenient data structure for implementing a priority queue.
	  Code from svn/asterisk/team/russell/heap/. Review:
	  http://reviewboard.digium.com/r/160/

2009-02-17 20:50 +0000 [r176631]  Olle Johansson <oej@edvina.net>

	* include/asterisk/config.h: Typo

2009-02-17 20:41 +0000 [r176627]  Russell Bryant <russell@digium.com>

	* channels/chan_unistim.c, main/pbx.c, apps/app_read.c,
	  configs/indications.conf.sample, apps/app_playtones.c (added),
	  include/asterisk/indications.h, apps/app_readexten.c,
	  apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h,
	  include/asterisk/_private.h, main/indications.c, main/loader.c,
	  main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
	  funcs/func_channel.c, res/snmp/agent.c, main/app.c,
	  res/res_indications.c (removed), main/asterisk.c: Merge a large
	  set of updates to the Asterisk indications API. This patch
	  includes a number of changes to the indications API. The primary
	  motivation for this work was to improve stability. The object
	  management in this API was significantly flawed, and a number of
	  trivial situations could cause crashes. The changes included are:
	  1) Remove the module res_indications. This included the critical
	  functionality that actually loaded the indications configuration.
	  I have seen many people have Asterisk problems because they
	  accidentally did not have an indications.conf present and loaded.
	  Now, this code is in the core, and Asterisk will fail to start
	  without indications configuration. There was one part of
	  res_indications, the dialplan applications, which did belong in a
	  module, and have been moved to a new module, app_playtones. 2)
	  Object management has been significantly changed. Tone zones are
	  now managed using astobj2, and it is no longer possible to crash
	  Asterisk by issuing a reload that destroys tone zones while they
	  are in use. 3) The API documentation has been filled out. 4) The
	  API has been updated to follow our naming conventions. 5) Various
	  bits of code throughout the tree have been updated to account for
	  the API update. 6) Configuration parsing has been mostly
	  re-written. 7) "Code cleanup" The code is from
	  svn/asterisk/team/russell/indications/. Review:
	  http://reviewboard.digium.com/r/149/

2009-02-17 18:49 +0000 [r176592]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to
	  track down a refcount leak. (closes issue #14485) Reported by:
	  davevg

2009-02-17 17:33 +0000 [r176557]  Russell Bryant <russell@digium.com>

	* main/pbx.c, apps/app_queue.c: Fix a race condition that caused
	  device states to become incorrect for hints. The problem here is
	  that the hint processing code was subscribed to the wrong event
	  type. So, it started processing state for a hint too soon, before
	  the device state cache had been updated. Also, fix a similar bug
	  in app_queue, as it was also subscribed to the wrong event type.
	  (closes issue #14461) Reported by: alecdavis

2009-02-17 17:28 +0000 [r176513-176556]  Olle Johansson <oej@edvina.net>

	* configs/extconfig.conf.sample: Typo

	* main/config.c: If there are no realtime engines, there's no
	  reason to check for realtime families

2009-02-17 14:39 +0000 [r176360-176501]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: In this version, we can combine the queries,
	  because we support dropping nonexistent columns.

	* /, channels/chan_sip.c: Merged revisions 176426 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009)
	  | 10 lines After a 'sip reload', qualifies for realtime peers
	  weren't immediately restarted, instead waiting until the next
	  registration. We're now caching the qualify across a
	  reload/restart and starting the qualify immediately upon loading
	  the peer. (closes issue #14196) Reported by: pdf Patches:
	  20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
	  Tested by: pdf ........

	* main/strings.c: Might want to update the buffer pointer after a
	  realloc (or we crash) (closes issue #14485) Reported by: davevg

2009-02-16 23:37 +0000 [r176356]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/sounds.xml: add support for Siren7 and Siren14 flavors of
	  prompts and music on hold

2009-02-16 23:33 +0000 [r176355]  David Vossel <dvossel@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 176354 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16
	  Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not
	  being relayed correctly during bridging This should have been
	  committed with rev176247, but I missed it. srcupdate frames no
	  longer break out of the native bridge, but are not being sent to
	  the other call leg either. This fixs that. issue #13749 ........

2009-02-16 23:14 +0000 [r176320]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_skinny.c: Use the correct list macros for deleting
	  an item from the middle of a list. (issue #13777) Reported by: pj
	  Patches: 20090203__bug13777.diff.txt uploaded by Corydon76
	  (license 14) Tested by: pj

2009-02-16 21:45 +0000 [r176255]  Kevin P. Fleming <kpfleming@digium.com>

	* /, main/utils.c, include/asterisk/stringfields.h: 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
	  ........

2009-02-16 21:40 +0000 [r176253]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_meetme.c: Merged revisions 176249,176252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon,
	  16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it
	  to be nonblocking atomically Apparently on FreeBSD, attempting to
	  set the O_NONBLOCKING flag separately from opening the file was
	  causing an "inappropriate ioctl for device" error. While I cannot
	  fathom why this would be happening, I certainly am not opposed to
	  making the code a bit more compact/efficient if it also fixes a
	  bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
	  uploaded by ys (license 281) Tested by: ys ........ r176252 |
	  mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3
	  lines Remove unused variable and make dev-mode compilation happy
	  ........

2009-02-16 21:30 +0000 [r176248]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Merged revisions 175597 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 |
	  dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
	  Fixed iax2 key rotation backwards compatibility Turns key
	  rotation back on by default. Added bit into encryption IE to
	  indicate whether or not key rotation is supported or not. If it
	  is not supported then it is not enabled, which insures backwards
	  compatibility. This eliminates the need for the keyrotate option
	  in iax.conf, so it has been removed. ........

2009-02-16 18:25 +0000 [r176174]  Mark Michelson <mmichelson@digium.com>

	* main/logger.c: 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.

2009-02-16 17:44 +0000 [r176138]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c: Can't set debug level 2 (intense
	  debugging) unless the syntax matches

2009-02-16 17:09 +0000 [r176100]  Russell Bryant <russell@digium.com>

	* channels/chan_features.c (removed): Remove chan_features. Review:
	  http://reviewboard.digium.com/r/161/

2009-02-16 15:36 +0000 [r176030]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 176029 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9
	  lines Don't have the Via header stored as a stringfield as it can
	  change often during the lifetime of a dialog. This issue crept up
	  with subscriptions on the AA50. When an outgoing NOTIFY is sent a
	  new branch value is created and the Via header is changed to
	  reflect it. Since this was a stringfield a new spot in the pool
	  was used for the value while the old was left untouched/unused.
	  If the current pool was full a new pool was created. This would
	  cause memory usage to increase steadily. (issue #AA50-2332)
	  ........

2009-02-16 02:54 +0000 [r175983]  Russell Bryant <russell@digium.com>

	* main/channel.c: Make the causes array static, and remove the type
	  name as it is not needed.

2009-02-16 00:26 +0000 [r175952]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_unistim.c, /, channels/chan_sip.c,
	  include/asterisk/manager.h, doc/unistim.txt: Merged revisions
	  175921 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009)
	  | 3 lines fix mis-spelling of the word registered. Reported by
	  De_Mon on #asterisk-dev. ........

2009-02-15 21:27 +0000 [r175829-175882]  Russell Bryant <russell@digium.com>

	* include/asterisk/sched.h, main/sched.c: Make ast_sched_report()
	  and ast_sched_dump() thread safe.

	* channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: Fix
	  a number of problems with ast_sched_report(). 1) It had numerous
	  coding guidelines violations with regards to formatting. 2) It
	  allocated memory using ast_calloc() that was never freed. 3) It
	  didn't check for failure from the allocation. 4) It used
	  sprintf() and strcat() to build the result, doing zero checking
	  to prevent writing past the end of the provided buffer. The
	  function also lacks API documentation, but that has not been
	  addressed in this commit.

2009-02-15 20:39 +0000 [r175783-175827]  Olle Johansson <oej@edvina.net>

	* formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb
	  2009) | 2 lines format_ilbc does not depend on codec libraries
	  and can therefore always be made. My mistake. Ursäkta! ........

	* formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb
	  2009) | 2 lines Disable format_ilbc.so by default, like
	  codec_ilbc.so ........

	* /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2
	  lines Make sure that the debug line is not printed on debug level
	  0 ........

2009-02-13 20:57 +0000 [r175655-175663]  Mark Michelson <mmichelson@digium.com>

	* doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset
	  branch to Asterisk From a user point-of-view, this adds new CLI
	  commands and Manager Actions to better facilitate the reloading
	  of queues and the resetting of their statistics. The new CLI
	  commands are the "queue reload" and "queue reset stats" commands.
	  The new manager actions are the QueueReload and QueueReset
	  commands. Review: http://reviewboard.digium.com/r/115

	* doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for
	  chanspy starting or stopping (closes issue #14469) Reported by:
	  caio1982 Patches: chanspy_events2.diff uploaded by caio1982
	  (license 22)

2009-02-13 20:26 +0000 [r175623-175636]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: fix a few more XML documentation problems

	* main/pbx.c: add missing </para>

2009-02-13 20:11 +0000 [r175597]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c:
	  Fixed iax2 key rotation backwards compatibility Turns key
	  rotation back on by default. Added bit into encryption IE to
	  indicate whether or not key rotation is supported or not. If it
	  is not supported then it is not enabled, which insures backwards
	  compatibility. This eliminates the need for the keyrotate option
	  in iax.conf, so it has been removed. Review:
	  http://reviewboard.digium.com/r/159/

2009-02-13 19:49 +0000 [r175591]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 175590 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri,
	  13 Feb 2009) | 16 lines Fix a potential crash situation when
	  using IMAP voicemail If calling into VoiceMailMain when using
	  IMAP storage, it was possible to crash Asterisk by hanging up the
	  phone when prompted for a voicemail mailbox. This patch fixes the
	  issue. While it may appear that this patch is superficial, it
	  allows code execution to continue to the failure case just below
	  the IMAP_STORAGE code block where this patch has been applied
	  (closes issue #14473) Reported by: dwpaul Patches:
	  voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
	  689) ........

2009-02-13 16:41 +0000 [r175549]  Joshua Colp <jcolp@digium.com>

	* apps/app_record.c: Add an option to keep the recorded file upon
	  hangup. (closes issue #14341) Reported by: fnordian

2009-02-13 13:41 +0000 [r175508-175512]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: document G.722.1/.1C support

	* main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h,
	  channels/chan_h323.c, include/asterisk/frame.h,
	  formats/format_siren14.c (added), main/rtp.c,
	  formats/format_siren7.c (added): Add basic (passthrough,
	  playback, record) support for ITU G.722.1 and G.722.1C (also
	  known as Siren7 and Siren14) This patch adds passthrough, file
	  recording and file playback support for the codecs listed above,
	  with negotiation over SIP/SDP supported. Due to Asterisk's
	  current limitation of treating a codec/bitrate combination as a
	  unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are
	  supported. Along the way, some related work was done: 1) The
	  rtpPayloadType structure definition, used as a return result for
	  an API call in rtp.h, was moved from rtp.c to rtp.h so that the
	  API call was actually usable. The only previous used of the API
	  all was chan_h323.c, which had a duplicate of the structure
	  definition instead of doing it the right way. 2) The hardcoded
	  SDP sample rates for various codecs in chan_sip.c were removed,
	  in favor of storing these sample rates in rtp.c along with the
	  codec definitions there. A new API call was added to allow
	  retrieval of the sample rate for a given codec. 3) Some basic
	  'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip
	  *must* decline any media streams offered for these codecs that
	  are not at the bitrates that we support (otherwise Bad Things
	  (TM) would result). Review: http://reviewboard.digium.com/r/158/

2009-02-13 04:22 +0000 [r175411-175475]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* CHANGES: add 'faxbuffers' configuration option information to
	  CHANGES

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
	  dynamic fax buffer configuration option to chan_dahdi.conf When
	  the 'faxdetect' configuration option is used, one may also want
	  to use the 'faxbuffers' configuration option in chan_dahdi.conf.
	  This option will dynamically use the configured 'faxbuffers'
	  buffer policy on a channel for the life of the call following the
	  detection of fax tones. The faxbuffers buffer policy will be
	  reverted during call teardown. An example use of 'faxbuffers' is
	  below. This example would switch to using 6 buffers with a full
	  buffer policy. faxbuffers=>6,full

2009-02-12 21:41 +0000 [r175368]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove useless string copy, and make sscanf
	  safe again

2009-02-12 21:27 +0000 [r175344]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
	  force encryption option to iax.conf This patch adds
	  forceencryption=yes as an iax.conf option. When force encryption
	  is enabled, no unencrypted connections are allowed. This insures
	  all connections are encrypted. This is a new feature, so CHANGES
	  and iax.conf.sample are updated as well. (closes issue #13285)
	  Reported by: sgofferj Tested by: russell Review:
	  http://reviewboard.digium.com/r/150/

2009-02-12 21:25 +0000 [r175334]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, /: 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 ........

2009-02-12 20:48 +0000 [r175298]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: 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 ........

2009-02-12 20:45 +0000 [r175255-175295]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Avoid using ast_strdupa() in a loop.

	* build_tools/cflags.xml: Don't enable something by default that
	  has a dependency on something _not_ enabled by default.
	  menuselect was not happy with this.

2009-02-12 18:48 +0000 [r175250]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: correct warning message to not refer
	  specifically to DAHDI

2009-02-12 18:00 +0000 [r175188]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c: 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. ........

2009-02-12 17:07 +0000 [r175127]  David Vossel <dvossel@digium.com>

	* channels/chan_iax2.c: Setting key rotation to be off by default
	  Key rotation breaks compatibility between (trunk/1.6.1) and
	  (1.2/1.4/1.6.0). As a follow up to this, I am investigating
	  possible ways to allow key rotation to be on by default and not
	  affect the other branches, but for now it must be turned off.

2009-02-12 16:57 +0000 [r175125]  Russell Bryant <russell@digium.com>

	* /, main/rtp.c: Merged revisions 175124 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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 ........

2009-02-12 16:28 +0000 [r175121]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/astobj2.h, main/astobj2.c: 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.

2009-02-12 14:25 +0000 [r175058-175089]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Issue a warning message if our candidate's
	  IP is the loopback address. (closes issue #13985) Reported by:
	  jcovert Tested by: phsultan

	* /, channels/chan_gtalk.c: Merged revisions 175029 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12
	  Feb 2009) | 12 lines Set the initiator attribute to lowercase in
	  our replies when receiving calls. This attribute contains a JID
	  that identifies the initiator of the GoogleTalk voice session.
	  The GoogleTalk client discards Asterisk's replies if the
	  initiator attribute contains uppercase characters. (closes issue
	  #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
	  by jcovert (license 551) Tested by: jcovert ........

2009-02-11 23:12 +0000 [r174945-174951]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a bit of odd logic for announcing position.
	  Sync with 1.6.0's logic

	* apps/app_queue.c: Fix odd "thank you" sound playing behavior in
	  app_queue.c If someone has configured the queue to play an
	  position or holdtime announcement, then it is odd and potentially
	  unexpected to hear a "Thank you for your patience" sound when no
	  position or holdtime was actually announced. This fixes the
	  announcement so that the "thanks" sound is only played in the
	  case that a position or holdtime was actually announced. There is
	  a way that the "thank you" sound can be played without a position
	  or holdtime, and that is to set announce-frequency to a value but
	  keep announce-position and announce-holdtime both turned off.
	  (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
	  uploaded by putnopvut (license 60) Tested by: caspy

	* apps/app_dial.c, main/channel.c, main/pbx.c, apps/app_dictate.c,
	  apps/app_waitforsilence.c, include/asterisk/channel.h: 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

2009-02-11 14:44 +0000 [r174844]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Tell the device state core a change happened when
	  a channel is freed but not a specific state. We need to do this
	  because while we know that the freeing of the channel may cause
	  something to become not in use we do not know this for sure.
	  There may be another channel that is still up which would cause
	  it to be in use. (closes issue #13238) Reported by: kowalma
	  Patches: 20090121__bug13238.diff.txt uploaded by Corydon76
	  (license 14) Tested by: alecdavis

2009-02-10 23:17 +0000 [r174764-174805]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c: Fix potential for stack overflows in
	  app_chanspy.c When using the 'g' or 'e' options, the stack
	  allocations that were used could cause a stack overflow if a
	  spyer stayed on the line long enough without actually
	  successfully spying on anyone. The problem has been corrected by
	  using static buffers and copying the contents of the appropriate
	  strings into them instead of using functions like alloca or
	  ast_strdupa

	* main/manager.c: 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/

2009-02-10 20:15 +0000 [r174710]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only decrease inringing count if above zero.
	  (issue #13238) Reported by: kowalma

2009-02-10 19:38 +0000 [r174705]  Kevin P. Fleming <kpfleming@digium.com>

	* main/slinfactory.c, include/asterisk/slinfactory.h: improve
	  slinfactory API to remove implicit sample rate and require
	  explicit sample rate selection by creator of the slinfactory

2009-02-10 18:16 +0000 [r174584]  Matthew Nicholson <mnicholson@digium.com>

	* /, main/jitterbuf.c: 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/ ........

2009-02-10 17:48 +0000 [r174543-174580]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Set the type for the peer structure to be a
	  peer as the default. (closes issue #14447) Reported by: triccyx

	* channels/chan_sip.c: Make the logic for inuse and inringing
	  manipluation match that of 1.4. The old broken logic would reset
	  the values back to 0 during certain scenarios causing the wrong
	  state to be reported. (closes issue #14399) Reported by: caspy
	  (issue #13238) Reported by: kowalma

2009-02-10 07:06 +0000 [r174470-174503]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c, apps/app_voicemail.c: Fix0ring build

	* apps/app_stack.c: Remove the usage of the KeepAlive app, as it no
	  longer exists.

2009-02-10 04:49 +0000 [r174370-174435]  Steve Murphy <murf@digium.com>

	* apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE
	  from app_rpt.c. (closes issue #14435) Reported by: D_McNaul

	* apps/app_rpt.c: More intptr_t work.

	* /, apps/app_rpt.c: Merged revisions 174369 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5
	  lines This patch solves some compiler complaints in both 32 and
	  64-bit environments. ........

2009-02-09 17:27 +0000 [r174327]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fix something I messed up in the merge I
	  just did

2009-02-09 17:26 +0000 [r174325]  David Vossel <dvossel@digium.com>

	* apps/app_externalivr.c: Fixes issue with hangups not being sent
	  and external process never terminating. The ignore_hangup,
	  run_dead, and noanswer flags were never initilized to zero
	  causing hangups to never be issued. If the external script
	  expects to be notified of a hangup and never receives one, it
	  runs indefinitely. (closes issue #14251) Reported by: chris-mac
	  Tested by: dvossel

2009-02-09 17:20 +0000 [r174301]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 174282 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
	  2009) | 12 lines Don't do an SRV lookup if a port is specified
	  RFC 3263 says to do A record lookups on a hostname if a port has
	  been specified, so that's what we're going to do. See section
	  4.2. (closes issue #14419) Reported by: klaus3000 Patches:
	  patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
	  (license 65) ........

2009-02-09 14:49 +0000 [r174219]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 174218 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb
	  2009) | 4 lines Don't overwrite our pointer to the music class
	  when music on hold stops. We will use this if it starts again to
	  see if we can resume the music where it left off. (closes issue
	  #14407) Reported by: mostyn ........

2009-02-07 16:16 +0000 [r174149]  Russell Bryant <russell@digium.com>

	* /, res/snmp/agent.c: Merged revisions 174148 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
	  | 2 lines Fix a race condition that could cause a crash. ........

2009-02-06 23:51 +0000 [r174084]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* /, channels/chan_sip.c: Merged revisions 174082 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
	  | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
	  sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
	  didn't actually upload a properly-formed patch, instead a
	  modified chan_sip.c file was uploaded. I created a patch to
	  determine the changes, then modified the suggested changes to
	  create a proper fix. The summary above is a complete description
	  of the changes. (closes issue #13547) Reported by: tecnoxarxa
	  Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
	  Tested by: tecnoxarxa ........

2009-02-06 20:12 +0000 [r174046]  David Vossel <dvossel@digium.com>

	* configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds
	  immediate yes/no option to iax.conf This is very similar to the
	  DAHDI immediate=yes option. When the phone is picked up, instead
	  of giving a dialtone it connects directly to the "s" extension.
	  Changes where implemented in chan_iax2.c to directly connect to
	  the "s" extension in the appropriate context when this option is
	  enabled. Examples explaining its use are added to
	  iax2.conf.sample. CHANGES has been updated as well. (closes issue
	  #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk
	  uploaded by jcovert (license 551) iax.conf.sample.patch uploaded
	  by jcovert (license 551) Tested by: jcovert, dvossel Review:
	  http://reviewboard.digium.com/r/143/

2009-02-06 19:28 +0000 [r173974-174041]  Joshua Colp <jcolp@digium.com>

	* channels/chan_dahdi.c: Don't subscribe to a mailbox on pseudo
	  channels. It is futile. This solves an issue where duplicated
	  pseudo channels would cause a crash because the first one would
	  unsubscribe and the next one would also try to unsubscribe the
	  same subscription. (closes issue #14322) Reported by: amessina

	* /, channels/chan_sip.c: Merged revisions 173967-173968 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
	  lines Some clients do not put the call-id for replaces at the
	  beginning, so support it being anywhere in the string. (closes
	  issue #14350) Reported by: fhackenberger ........ r173968 | file
	  | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
	  debug message I put in by accident. ........

2009-02-06 16:28 +0000 [r173952]  Matthew Nicholson <mnicholson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 173917 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
	  2009) | 7 lines Limit the addition of the Contact header in SIP
	  responses according to various SIP RFCs. (closes issue #13602)
	  Reported by: hjourdain Tested by: mnicholson ........

2009-02-06 15:59 +0000 [r173902]  Joshua Colp <jcolp@digium.com>

	* main/audiohook.c, apps/app_chanspy.c: Always detach and destroy
	  the whisper and barge audiohooks. Additionally also allow an
	  audiohook to be detached if it has not been attached. (closes
	  issue #14414) Reported by: bluecrow76

2009-02-06 10:55 +0000 [r173848-173858]  Russell Bryant <russell@digium.com>

	* include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add
	  a common implementation of a scheduler context with a dedicated
	  thread. This commit expands the Asterisk scheduler API to include
	  a common implementation of a scheduler context being processed by
	  a dedicated thread. chan_iax2 has been updated to use this new
	  code. Also, as a result, this resolves some race conditions
	  related to the previous chan_iax2 scheduler handling. Related to
	  rev 171452 which resolved the same issues in 1.4. Code from
	  team/russell/sched_thread2 Review:
	  http://reviewboard.digium.com/r/129/

	* main/manager.c: Resolve a memory leak that would occur on an
	  invalid channel given to Action: Status

2009-02-05 23:48 +0000 [r173773-173776]  Mark Michelson <mmichelson@digium.com>

	* configs/extensions.conf.sample: Update extensions.conf.sample to
	  be correct. In trunk, the only necessary change pointed out was
	  that the call to ChanIsAvail uses an option that has been
	  removed. For the 1.6.1 branch, however, it appears that the
	  sample file is badly in need of updating since there are |'s used
	  all over the place there. My tentative plan is just to copy
	  trunk's sample config file to those branches since the info there
	  is most up-to-date and should be correct for use in 1.6.1 Thanks
	  to macli in #asterisk-dev for bringing this up

	* apps/app_voicemail.c: Properly set "seen" and "unseen" flags when
	  moving messages from the new to the old folder when using IMAP
	  for voicemail storage (closes issue #13905) Reported by: jaroth
	  Patches: foldermove_v2.patch uploaded by jaroth (license 50)

2009-02-05 21:00 +0000 [r173697]  Jeff Peeler <jpeeler@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 173696 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05
	  Feb 2009) | 12 lines Add new configuration option to make shared
	  IMAP mailboxes function as expected. The new option is
	  "imapvmshareid" which is an ID to tag multiple mailboxes using
	  the same IMAP storage location to function as one mailbox. This
	  allows all messages to be retrieved for any user in the group.
	  The patch alters the 'X-Asterisk-VM-Extension' header that is
	  responsible for matching voicemails for a given user. (closes
	  issue #13673) Reported by: howardwilkinson ........

2009-02-05 20:30 +0000 [r173693]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 173692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
	  2009) | 12 lines Fix situations where queue members could be
	  autopaused unexpectedly Specifically, this patch prevents us from
	  autopausing members when we receive a busy or congestion frame
	  from them. (closes issue #14376) Reported by: fiddur Patches:
	  14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
	  ........

2009-02-05 19:36 +0000 [r173657]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_sqlite.c: Change the first field, or we don't get
	  the necessary field separation.

2009-02-05 18:48 +0000 [r173507-173593]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_mixmonitor.c: Merged revisions 173592 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu,
	  05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor
	  ........

	* /, apps/app_mixmonitor.c: Merged revisions 173559 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu,
	  05 Feb 2009) | 25 lines Fix a problem where a channel pointer
	  becomes invalid due to masquerading or hanging up. app_mixmonitor
	  runs its own thread to monitor the channel's activity and write
	  the mixed audio to a file. Since this thread runs independently
	  of the channel, it is possible that the mixmonitor thread's
	  channel pointer will point to freed memory when the channel
	  either is masqueraded or hangs up (technically, both cases are
	  hangups, but we need to handle the cases slightly differently).
	  The solution for this is to employ a datastore, which has the
	  nice benefit of allowing us to hook into channel masquerades and
	  hangups and update our pointer as necessary. If this looks
	  familiar, this same technique is employed in app_chanspy.
	  app_chanspy is a bit more involved since it does a lot more
	  operations on the channel that is being spied upon.
	  app_mixmonitor does have an extra touch that app_chanspy doesn't
	  have, though. Since there is a thread race between the channel's
	  thread and the mixmonitor thread on a hangup, we em- ploy a
	  condition-and-boolean combination to ensure that the channel
	  thread finishes with our structure before the mixmonitor thread
	  attempts to free it. No crashes! (closes issue #14374) Reported
	  by: aragon Patches: 14374.patch uploaded by putnopvut (license
	  60) Tested by: aragon, putnopvut ........

	* apps/app_queue.c: Fix some areas where the incorrect interface
	  was passed to ast_device_state I swear it feels like I already
	  did this once... (closes issue #14359) Reported by: francesco_r

2009-02-04 21:26 +0000 [r173503]  Tilghman Lesher <tlesher@digium.com>

	* res/res_jabber.c: Add XML documentation for the applications and
	  functions in res_jabber (closes issue #14405) Reported by: snuffy
	  Patches: xml_jabber.diff uploaded by snuffy (license 35)

2009-02-04 21:25 +0000 [r173502]  David Vossel <dvossel@digium.com>

	* channels/iax2-parser.h, channels/chan_iax2.c: Fixes issue with
	  IAX2 transfer not handing off calls. Reverts changes in 116884
	  Fixes issue with IAX2 transfers not taking place. As it was, a
	  call that was being transfered would never be handed off
	  correctly to the call ends because of how call numbers were
	  stored in a hash table. The hash table, "iax_peercallno_pvt",
	  storing all the current call numbers did not take into account
	  the complications associated with transferring a call, so a
	  separate hash table was required. This second hash table
	  "iax_transfercallno_pvt" handles calls being transfered, once the
	  call transfer is complete the call is removed from the transfer
	  hash table and added to the peer hash table resuming normal
	  operations. Addition functions were created to handle storing,
	  removing, and comparing items in the iax_transfercallno_pvt
	  table. The changes reverted in 116884 caused backwards
	  compatibility issues involving iax2 transfer with 1.6.0, 1.4, and
	  1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel

2009-02-04 21:17 +0000 [r173500]  Jeff Peeler <jpeeler@digium.com>

	* /, main/features.c, include/asterisk/features.h: 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/ ........

2009-02-04 18:48 +0000 [r173458]  Tilghman Lesher <tlesher@digium.com>

	* main/tcptls.c: 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)

2009-02-04 17:45 +0000 [r173354-173397]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 173396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
	  2009) | 3 lines Revert my previous change because it was stupid
	  ........

	* /, apps/app_chanspy.c: Merged revisions 173392 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
	  2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
	  matter, but it's needed. ........

	* main/file.c: 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

2009-02-04 00:43 +0000 [r173311]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, pbx/pbx_config.c: 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

2009-02-03 17:35 +0000 [r173169]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Broke up the large conditional blocks so
	  it is easy to see if a function is compiled.

2009-02-03 00:29 +0000 [r173104-173130]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/xml.c, include/asterisk/compiler.h, apps/app_stack.c,
	  include/asterisk/optional_api.h: 1. Make OS X compile cleanly
	  with app_stack. 2. Use curl to download sound files, as curl is
	  installed natively on OS X, whereas wget and fetch are not.
	  (closes issue #14332) Reported by: oej Tested by: Corydon76

	* /, configs/extensions.conf.sample: Merged revisions 173070 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
	  | 5 lines Add warning to standard config, that globals may be
	  overridden by other dialplan configuration files. (closes issue
	  #14388) Reported by: macli ........

2009-02-02 23:57 +0000 [r173067]  Terry Wilson <twilson@digium.com>

	* /, main/features.c: 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
	  ........

2009-02-02 23:21 +0000 [r173028-173047]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c, CHANGES: Reverting commit number 173028 as there
	  are some potential issues

	* main/manager.c, CHANGES: Add a CLI command to log out a manager
	  user (closes issue #13877) Reported by: eliel Patches:
	  cli_manager_logout.patch.txt uploaded by eliel (license 64)
	  Tested by: eliel, putnopvut

2009-02-02 20:40 +0000 [r172963]  Richard Mudgett <rmudgett@digium.com>

	* /: Recorded merge of revisions 172962 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009)
	  | 11 lines channels/chan_dahdi.c * Added doxygen comments to the
	  major dahdi structures. * Fixed PRI using an incorrect string
	  value if the extension delimiter is not present in the Dial()
	  function. * Fixed some uninitialized string variables on FXS
	  ports. configs/chan_dahdi.conf.sample * Updated some
	  documentation. These changes are already in trunk -r172400
	  ........

2009-02-02 19:02 +0000 [r172929]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/features.c, CHANGES,
	  include/asterisk/features.h: This reverts the changes I made for
	  11583; will reviewboard this before committing again... reopened
	  11583 until all Russell's issues are resolved.

2009-02-02 18:13 +0000 [r172894]  Leif Madsen <lmadsen@digium.com>

	* configs/res_ldap.conf.sample: Update the res_ldap.conf file with
	  a better working example. (closes issue #13861) Reported by:
	  scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by
	  blitzrage (license 10) Tested by: jcovert

2009-02-02 17:37 +0000 [r172890]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/features.c, CHANGES,
	  include/asterisk/features.h: This change allows the disconnect
	  feature (as in "one-touch" in features.c) to be used within the
	  dial app, before a call is bridged. Many thanks to sobomax for
	  submitting this patch. Quoting from bug 11582: "So the goal of
	  the patch was to use the user configured feature code during the
	  call setup phase. The original ast_feature_interpret() function
	  is not well suited for this purpose as it uses much call bridge
	  specific data and doesn't separate a detection of feature from a
	  feature handler call. So a new function ast_feature_detect() has
	  been extracted off the ast_feature_interpret() function but
	  keeping the original logic intact except some insignificant
	  changes to locking. "Having created the ast_feature_detect()
	  function the possibility to use feature detection in almost any
	  place of the asterisk code. So a call to this function has been
	  added to wait_for_answer() function of app_dial.so module. This
	  code doesn't call the feature handler however and uses old call
	  leg disconnect logic to make the changes as small and simple as
	  possible to prevent unexpected problems. A disconnect feature
	  currently is the only one supported during call setup as other
	  features as call parking and call transfer don't make much sense
	  during call setup. However if need in some of the features would
	  arise it is much easier to implement as the infrastructure
	  changes are already in place with this patch." I have cleaned up
	  the patch somewhat, and verified that the existing functionality
	  is not harmed, and that the new functionality works. Terry has
	  committed his stuff, and there were no conflicts (see 14274).
	  (closes issue #11583) Reported by: sobomax Patches:
	  patch-apps__app_dial.c uploaded by sobomax (license 359)
	  patch-include__asterisk__features.h uploaded by sobomax (license
	  359) patch-res__res_features.c uploaded by sobomax (license 359)
	  enable-features-during-call-setup.diff uploaded by sobomax
	  (license 359) 11583.newdiff uploaded by murf (license 17)
	  enable-features-during-call-setup-1.diff uploaded by sobomax
	  (license 359) 11583.latest-patch uploaded by murf (license 17)
	  Tested by: sobomax, murf

2009-02-02 16:42 +0000 [r172855]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix a spelling mistake.

2009-02-02 10:46 +0000 [r172816-172818]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Add a todo. I do need to really check what's
	  going on with this kill-the-user business ;-) Why do we suddenly
	  have two flags to set peer type?

	* channels/chan_sip.c: Small formatting change

	* channels/chan_sip.c: Add some well-needed improvements to the
	  wishlist in the code, so that we can close some bug reports.

2009-02-02 01:41 +0000 [r172778]  Sean Bright <sean.bright@gmail.com>

	* channels/chan_sip.c: The CID lookup feature wasn't actually
	  working properly with dialog-info+xml supporting devices. The
	  devices (snoms, specifically) need to receive a SIP URI instead
	  of just an extension. This adds that functionality.

2009-02-01 02:44 +0000 [r172706-172741]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Blank argument crashes Asterisk (closes
	  issue #14377) Reported by: amorsen

	* funcs/func_strings.c: Don't increment the loop, now that
	  incrementing is taken care of by the decoder function. (closes
	  issue #14363) Reported by: andrew53 Patches:
	  func_strings_filter.patch uploaded by andrew53 (license 519)

2009-01-30 22:22 +0000 [r172598]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/channel.h: Fix redefinition of flag in channel.h

2009-01-30 21:50 +0000 [r172580-172581]  Terry Wilson <twilson@digium.com>

	* configs/features.conf.sample: Remove incorrect line from sample
	  config

	* apps/app_dial.c, main/global_datastores.c, main/features.c,
	  include/asterisk/global_datastores.h, CHANGES,
	  configs/features.conf.sample: 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/ ........

2009-01-30 18:36 +0000 [r172441-172548]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_aes.c: Parameter position reversed in documentation

	* /, autoconf/ast_func_fork.m4, configure, main/app.c,
	  apps/app_rpt.c, main/asterisk.c: 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 ........

2009-01-29 23:15 +0000 [r172370-172440]  Richard Mudgett <rmudgett@digium.com>

	* main/cli.c: Remove tabs from comment

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
	  channels/chan_dahdi.c * Added doxygen comments to the major dahdi
	  structures. * Fixed PRI and SS7 using an incorrect string value
	  if the extension delimiter is not present in the Dial() function.
	  * Fixed SS7 not checking if the dialed extension is at least as
	  long as the stripmsd option. * Fixed PRI not handling unknown
	  TON/NPI prefix letters correctly. * Fixed some uninitialized
	  string variables on FXS ports. configs/chan_dahdi.conf.sample *
	  Updated some documentation.

	* include/asterisk/say.h: Fixed some doxygen comments

2009-01-29 17:10 +0000 [r172318-172319]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c: Revert two lines that was extra, but only
	  on fridays.

	* apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c,
	  include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered
	  elsewhere" through app_queue with members in chan_local. Also,
	  implement a private cause code (as suggested by Tilghman). This
	  works with chan_sip, but doesn't propagate through chan_local.

2009-01-29 16:48 +0000 [r172315]  Tilghman Lesher <tlesher@digium.com>

	* configs/func_odbc.conf.sample: Better document mode=multirow,
	  based upon a conversation with Jared.

2009-01-29 13:47 +0000 [r172271]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/realtime_pgsql.sql: The realtime_pgsql.sql script
	  is missing a couple of fields. closes issue #14339) Reported by:
	  fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur
	  (license 678)

2009-01-29 13:24 +0000 [r172173-172270]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample, CHANGES: Update documentation

	* include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make
	  sure we set setvar= variables on outbound calls too, not only
	  inbound calls. - Also, change a function in app.c to return a
	  userful value instead of always returning 0. Patch by fnordian,
	  changed by Corydon76 and myself. This does not close the bug
	  report, as fnordian had an additional change we're still
	  discussing. (related to issue #14059) Reported by: fnordian
	  Patches: chan_sip_hfield.patch uploaded by fnordian (license 110)
	  20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: fnordian, Corydon76, oej

	* channels/chan_sip.c: Make sure register= line supports both port
	  and expiry at the same time. (closes issue #14185) Reported by:
	  Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by
	  Nick (license 657) Tested by: Nick_Lewis

	* /, channels/chan_sip.c: Merged revisions 172169 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16
	  lines Make sure that we always add the hangupcause headers. In
	  some cases, the owner was disconnected before we checked for the
	  cause. This patch implements a temporary storage in the pvt and
	  use that instead. The code is based on ideas from code from
	  Adomjan in issue #13385 (Add support for Reason: header) Thanks
	  to Klaus Darillion for testing! (closes issue #14294) related to
	  issue #13385 Reported by: klaus3000 and adomjan Patches:
	  bug14294b.diff uploaded by oej (license 306) Based on
	  20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan
	  (license 487) Tested by: oej, klaus3000 ........

2009-01-28 22:52 +0000 [r172132]  Steve Murphy <murf@digium.com>

	* channels/chan_misdn.c: A further correction: cast the sizeof to
	  an int.

2009-01-28 22:48 +0000 [r172131]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_odbc.c: Fix how we skip fields (to avoid fields
	  which don't exist) when doing an UPDATE. (closes issue #14205)
	  Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: blitzrage

2009-01-28 21:48 +0000 [r172063-172099]  Steve Murphy <murf@digium.com>

	* channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2
	  didn't like the \%ld and issued a warning, breaking my dev-mode
	  build. This fixes it.

	* apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
	  main/features.c, include/asterisk/channel.h: 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 ........

2009-01-28 17:27 +0000 [r171964]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 171963 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
	  Jan 2009) | 2 lines Clarify log message (suggested by manxpower
	  on #asterisk-dev) ........

2009-01-28 14:39 +0000 [r171838-171925]  Olle Johansson <oej@edvina.net>

	* CHANGES: Yep. Documentation is important.

	* apps/app_queue.c: Add final part of previously committed work for
	  answered elsewhere in queue - the missing piece that started with
	  app_dial() earlier on. This is to avoid having the list and
	  counter of missed calls being touched by queue calls. Add the C
	  option to queue() and nothing will be logged on phones that
	  support the Reason: header on SIP cancel, like the SNOM phones.

	* configs/sip.conf.sample: Add some more notes about device
	  matching.

	* /, configs/sip.conf.sample: Merged revisions 171837 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan
	  2009) | 2 lines Add a better explanation of the difference
	  between the device namespace and the dialplan for newbies.
	  ........

2009-01-28 00:17 +0000 [r171797]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_aes.c: Fix some signedness problems in func_aes.c

2009-01-27 23:28 +0000 [r171793]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Don't complain about lack of D-channels on
	  PTMP connections

2009-01-27 22:43 +0000 [r171757]  David Vossel <dvossel@digium.com>

	* funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and
	  AES_DECRYPT dialplan functions. (closes issue #14301) Reported
	  by: amorsen review: http://reviewboard.digium.com/r/128/

2009-01-27 21:58 +0000 [r171618-171691]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Merged revisions 171689 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
	  2009) | 39 lines Fix devicestate problems for "always-on" agent
	  channels A revision to chan_agent attempted to "inherit" the
	  device state of the underlying channel in order to report the
	  device state of an agent channel more accurately. The problem
	  with the logic here is that it makes no sense to use this for
	  always-on agents. If the agent is logged in, then to the
	  underlying channel, the agent will always appear to be "in use,"
	  no matter if the agent is on a call or not. The reason is that to
	  the underlying channel, the channel is currently in use on a call
	  to the AgentLogin application. The most common cause that I found
	  for this issue to occur was for a SIP channel to be the
	  underlying channel type for an Agent channel. If the SIP phone
	  re-registers, then the registration will cause the device state
	  core to query the device state of the SIP channel. Since the SIP
	  channel is in use, the Agent channel would also inherit this
	  status. Once the agent channel was set to "in use" there was no
	  way that the device state could change on that channel unless the
	  agent logged out. The solution for this problem is a bit
	  different in 1.4 than it is in the other branches. In 1.4, there
	  will be a one-line fix to make sure that only callback agents
	  will inherit device state from their underlying channel type. For
	  the other branches of Asterisk, since callback support has been
	  removed, there is also no need for device state inheritance in
	  chan_agent, so I will simply be removing it from the code. In
	  addition, the 1.4 source is getting a new comment to help the
	  next person who edits chan_agent.c. I'm adding a comment that a
	  agent_pvt's loginchan field may be used to determine if the agent
	  is a callback agent or not. (closes issue #14173) Reported by:
	  nathan Patches: 14173.patch uploaded by putnopvut (license 60)
	  Tested by: nathan, aramirez ........

	* /, main/slinfactory.c: Merged revisions 171621 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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)
	  ........

	* apps/app_queue.c: Fix queue crashes that would occur after the
	  calling channel was masqueraded. The data passed to the
	  end_bridge_callback was assumed to be data which was still
	  stack'd. The problem was that with some call features, attended
	  transfers in particular, a new bridge thread is started once the
	  feature completes, meaning that when the end_bridge_callback is
	  called, the end_bridge_callback_data was invalid. To fix this
	  problem, there are two measures taken 1. Instead of pointing to
	  stacked data, we now used heap-allocated data for passing to the
	  end_bridge_callback in app_queue 2. Since bridges can end
	  multiple times on a single logical call, we wait until the final
	  bridge is broken to actually set any queue variables. This is
	  accomplished through reference-counting and the use of an
	  end_bridge_callback_data_fixup function in app_queue.c (closes
	  issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
	  by putnopvut (license 60) Tested by: ccesario

2009-01-27 15:23 +0000 [r171558]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there
	  are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis
	  Tested by: dbailey

2009-01-27 15:00 +0000 [r171263-171528]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Solving the same issue, but a bit
	  different in trunk... Merged revisions 171527 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
	  lines Use the same branch tag in CANCEL as in INVITE Originally
	  putnopvut implemented some changes in revision 142079 that
	  according to the bug report seemed to have worked then, but
	  somehow fails now. I guess code, as humans, get old and forget
	  stuff. Anyway, this bug caused CANCEL not to work with picky
	  systems. Thanks Fredrik for pointing out where the bug in the SIP
	  messaging was. (closes issue #14346) Reported by: oej Patches:
	  bug14346.diff uploaded by oej (license 306) Tested by: oej
	  ........

	* channels/chan_sip.c: Moving generic setting to friends

	* channels/chan_sip.c: Continue to move variables into the sip_cfg
	  structure to make them easier to handle in the future as a group
	  of settings for a group of devices. At some point, I want one
	  sip_cfg per domain handled, so we can have "group" settings.

	* channels/chan_sip.c: Just moving around variable declarations so
	  that we have all globals in the same place. Default setting is
	  set before we activate the channel or at reloads, not where we
	  declare the variable.

	* /, channels/chan_sip.c: Merged revisions 171264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
	  lines Don't retransmit 401 on REGISTER requests when
	  alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
	  Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
	  klaus3000 (license 65) Tested by: klaus3000 ........

	* main/channel.c: Add extensions and context on manager event when
	  new channel is created.

2009-01-25 23:58 +0000 [r171188]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_oss.c: Merged revisions 171187 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
	  | 6 lines Correctly track the hookstate (closes issue #13686)
	  Reported by: itiliti Patches: 20081013__bug13686.diff.txt
	  uploaded by Corydon76 (license 14) ........

2009-01-25 16:50 +0000 [r171043-171081]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: dont segfault when a MWI event occurs on
	  a line without a registered device

	* configs/skinny.conf.sample: Make the sample skinny.conf work
	  (closes issue #14325) Reported by: DEA Patches:
	  skinny.conf.sample-trunk.txt uploaded by DEA (license 3)

2009-01-25 13:35 +0000 [r170980]  Sean Bright <sean.bright@gmail.com>

	* /, apps/app_page.c: Merged revisions 170979 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
	  2009) | 9 lines Resolve a logic error that was causing Page() to
	  crash when more than one channel was specified. (closes issue
	  #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
	  uploaded by seanbright (license 71) Tested by: kc0bvu ........

2009-01-25 02:49 +0000 [r170902-170943]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h: Change ARRAY_LEN() to be more C++ safe.
	  When the second part of this macro is written as 0[a] instead of
	  a[0], it will force a failure if the macro is used on a C++
	  object that overloads the [] operator.

	* res/res_agi.c: Add a todo to finish the XML docs in this module

2009-01-24 13:55 +0000 [r170837]  Tilghman Lesher <tlesher@digium.com>

	* /, configs/res_odbc.conf.sample: Merged revisions 170836 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009)
	  | 2 lines Remove superfluous implementation note (closes issue
	  #14319) ........

2009-01-23 23:10 +0000 [r170794]  Richard Mudgett <rmudgett@digium.com>

	* doc/tex/Makefile: Fix asterisk.pdf generation if branch name has
	  an underscore in it.

2009-01-23 22:58 +0000 [r170790]  Russell Bryant <russell@digium.com>

	* doc/tex/Makefile: Don't blow up if a branch name has an
	  underscore in it

2009-01-23 20:56 +0000 [r170677-170720]  Mark Michelson <mmichelson@digium.com>

	* /, configs/res_odbc.conf.sample: Merged revisions 170719 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
	  2009) | 8 lines Add notes to the idlecheck explanation in
	  res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
	  Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
	  klaus3000 (license 65) ........

	* /, contrib/i18n.testsuite.conf: Merged revisions 170671 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
	  2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
	  deprecated syntax * Convert Wait,1 to Wait(1) * Convert
	  SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
	  priorities beyond the first Also added test for Chinese numbers,
	  too. (closes issue #14320) Reported by: dant Patches:
	  i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
	  670) ........

2009-01-23 20:18 +0000 [r170652]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: 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 ........

2009-01-23 19:25 +0000 [r170608]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 170588 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
	  Jan 2009) | 2 lines Additions to AST-2009-001 ........

2009-01-23 19:09 +0000 [r170505-170569]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 170568 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
	  lines When a call is forwarded stop any active indications. The
	  new channel will provide an indication, if need be, itself.
	  (closes issue #14310) Reported by: RadicAlish ........

	* /, channels/chan_sip.c: Merged revisions 170504 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
	  lines Use the on hold flag to see if the call is on hold or not.
	  It is possible that our address for them will still be valid even
	  though they are on hold. (closes issue #14295) Reported by:
	  klaus3000 ........

2009-01-23 17:46 +0000 [r170501]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_h323.c: let's use SENTINEL where needed

2009-01-23 17:32 +0000 [r170498]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Reset the ast_str used for escape
	  substitution. We need to do this since it is a thread local
	  variable that may contain the value of a previous substitution.
	  (closes issue #14312) Reported by: pj

2009-01-23 17:03 +0000 [r170463]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: We should not do restart messages if we're
	  in PTMP mode

2009-01-23 16:57 +0000 [r170460]  Michiel van Baak <michiel@vanbaak.info>

	* channels/chan_skinny.c: Dont clear the display of skinny phones
	  when not needed. (closes issue #13182) Reported by: pj Patches:
	  2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak
	  (license 7) Tested by: mvanbaak, pj

2009-01-23 16:35 +0000 [r170457]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c: MWI messages included in CID spill was not
	  being properly handled and prevented the call from being
	  processed (issue #14313) Reported by: seandarcy Tested by:
	  dbailey

2009-01-23 15:44 +0000 [r170393]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: 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
	  ........

2009-01-22 23:23 +0000 [r170351]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c: Make sure we don't set the channel to be
	  inalarm for a D-channel drop on PTMP connections

2009-01-22 21:25 +0000 [r170307]  Tilghman Lesher <tlesher@digium.com>

	* main/abstract_jb.c: Create logfile safely. (closes issue #14160)
	  Reported by: tzafrir Patches: 20090104__bug14160.diff.txt
	  uploaded by Corydon76 (license 14)

2009-01-22 20:04 +0000 [r170240]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: 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) ........

2009-01-22 17:19 +0000 [r170165]  Tilghman Lesher <tlesher@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 170158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
	  | 6 lines Allow global variables after substitution to be as long
	  as other variables. (closes issue #14263) Reported by: markd
	  Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
	  (license 14) ........

2009-01-22 16:52 +0000 [r170148]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 170147 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
	  lines If we are unable to request a DAHDI pseudo channel and we
	  are using the user introduction without review option make sure
	  it gets unset so other code does not blindly assume a DAHDI
	  pseudo channel exists. (closes issue #14282) Reported by:
	  cheesegrits ........

2009-01-22 15:49 +0000 [r170112]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: change VMWI to
	  use new DAHDI_VMWI ioctl call. Change configure script to detect
	  the new ioctl call data structure. (issue #14104) Reported by:
	  alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded
	  by dbailey (license ) Tested by: alecdavis, dbailey

2009-01-22 15:14 +0000 [r170047-170051]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, /: 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) ........

	* apps/app_parkandannounce.c: Clear the autoloop flag when parsing
	  and setting the context/extension/priority to go back to. When
	  the channel executes a PBX again we want it to start out at the
	  point we explicitly say and at that point it will not yet be
	  doing autoloop. (closes issue #14304) Reported by: jcovert

2009-01-22 02:10 +0000 [r170007]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: * Adjust some conditionals to balance
	  curly braces. * Other minor changes.

2009-01-22 00:44 +0000 [r169944]  Tilghman Lesher <tlesher@digium.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 169943 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
	  | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
	  wanted to ask is whether autoconf detected a static initializer
	  value. This fixes rwlocks on all such platforms (mainly, Mac OS
	  X). (closes issue #13767) Reported by: jcovert Patches:
	  20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: jcovert, Corydon76 ........

2009-01-22 00:23 +0000 [r169910]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_dahdi.c: Whitespace changes only

2009-01-21 23:25 +0000 [r169869]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c, /: Merged revisions 169867 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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 ........

2009-01-21 23:20 +0000 [r169794-169866]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c: Test commit for test issue #14303

	* main/say.c: 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

2009-01-21 22:04 +0000 [r169793]  Michiel van Baak <michiel@vanbaak.info>

	* doc/tex/extensions.tex: remove duplicated sentence.

2009-01-21 21:53 +0000 [r169791]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Further fix some oddities in sip show users
	  and sip show peers logic ccesario on IRC pointed out that his sip
	  peers were not displayed properly when he would issue the command
	  "sip show peers." The problem was that the onlymatchonip field
	  was used to determine if the endpoint was a "peer" or "user." The
	  tricky part is that a "friend" is supposed to be treated as both
	  a "user" and a "peer" but the logic would not allow "friends" to
	  show up as "peers" since onlymatchonip was set to FALSE for
	  friends. I have modified the sip_peer structure to more
	  explicitly keep track of what type endpoint it is so that the
	  various manager and CLI commands will display the expected
	  information Reported by ccesario via IRC Tested by ccesario

2009-01-21 21:03 +0000 [r169723]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: Merged revisions 169722 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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 ........

2009-01-21 17:21 +0000 [r169673]  Steve Murphy <murf@digium.com>

	* utils/refcounter.c: This patch corrects a segfault reported in
	  14289, due to a null ptr being refd. Yes, seanbright is right in
	  the bug comments, that is the fix. Sorry for this oversight; I
	  guess my personal usage didn't have this happen! murf (closes
	  issue #14289) Reported by: jamesgolovich

2009-01-21 10:49 +0000 [r169620-169625]  Russell Bryant <russell@digium.com>

	* /: Remove properties that erroneously got merged into trunk

	* main/tcptls.c: Fix a regression in TCP support. This patch fixes
	  a problem that caused chan_sip to think that every open TCP
	  session was to a remote address of 0.0.0.0:0. (closes issue
	  #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt
	  uploaded by jamesgolovich (license 176)

2009-01-21 00:33 +0000 [r169557-169611]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix device state parsing issues for channel
	  names with multiple slashes The fix being applied is a bit
	  different for trunk and the 1.6.X branches. For trunk, we only
	  wish to strip off the characters beyond the second slash if the
	  channel is a Local channel (i.e. we are removing the /n from the
	  device name). Other channel technologies with multiple slashes
	  (e.g. DAHDI) need the information after the second slash in order
	  to get the proper device state information. In addition to this
	  fix, the 1.6.X branches are receiving a much more important fix
	  as well. The problem in 1.6.X is that the member's device name
	  was being directly changed instead of having a copy changed. This
	  meant that we would strip off the second slash and trailing
	  characters and then leave the member's device name like that
	  permanently thereafter. (closes issue #14014) Reported by:
	  kebl0155 Patches: 14014_number2.patch uploaded by putnopvut
	  (license 60) Tested by: kebl0155

	* apps/app_queue.c: Use the default timeout for a queue instead of
	  -1 (closes issue #14272) Reported by: timking

	* /, channels/chan_sip.c: Convert the character pointers in a
	  sip_request to be pointer offsets When an ast_str expands to hold
	  more data, any pointers that were pointing to the data prior to
	  the expansion will be pointing at invalid memory. This change
	  makes such pointers used in chan_sip.c instead be offsets from
	  the beginning of the string so that the same math may be applied
	  no matter where in memory the string resides. To help ease this
	  transition, a macro called REQ_OFFSET_TO_STR has been added to
	  chan_sip.c so that given a sip_request and an offset, the string
	  at that offset is returned. (closes issue #14220) Reported by:
	  riksta Tested by: putnopvut Review
	  http://reviewboard.digium.com/r/126/

2009-01-20 19:22 +0000 [r169486-169510]  Terry Wilson <twilson@digium.com>

	* main/features.c: 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.

	* /, main/features.c: Merged revisions 169485 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  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 ........

2009-01-19 21:42 +0000 [r169438]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/res_odbc.h, funcs/func_odbc.c,
	  include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is
	  *not* part of the ast_str API, it's part of the ast_odbc API and
	  just happens to use an ast_str as the buffer; move all of it to
	  res_odbc.c and res_odbc.h, renaming appropriately along the way
	  fix some minor coding style issues in strings.h and add some
	  attribute_pure annotations to functions in the ast_str API

2009-01-19 20:14 +0000 [r169367-169369]  Michiel van Baak <michiel@vanbaak.info>

	* main/asterisk.c: fix assignment in swapmode plug. Spotted and fix
	  provided by ys (closes issue #14129) Reported by: ys Tested by:
	  ys

	* channels/chan_skinny.c: Redo the event-based MWI in chan_skinny.
	  Dan saw regular segfaults with the old implementation and rewrote
	  it to make it really eventbased. I altered it to be trunk
	  compatible and wedhorn gave some feedback and ideas how to make
	  it even better. (closes issue #13821) Reported by: DEA Patches:
	  chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by:
	  mvanbaak, DEA "no probs by me" from wedhorn

2009-01-19 20:05 +0000 [r169365]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c, /, apps/app_userevent.c: Merged revisions 169364
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
	  | 4 lines Truncate userevents at the end of a line, when the
	  command exceeds the buffer. (closes issue #14278) Reported by:
	  fnordian ........

2009-01-19 18:36 +0000 [r169327]  Michiel van Baak <michiel@vanbaak.info>

	* main/asterisk.c: Make asterisk compile on non-amd64 versions of
	  OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD
	  and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when
	  HW_PHYSMEM64 is not available. (closes issue #14129) Reported by:
	  ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak
	  (license 7) Tested by: mvanbaak, jtodd

2009-01-19 18:22 +0000 [r169277-169325]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c: Get rid of magic number and replace with
	  DAHDI_VMWI_NUMBER_MASK when determining the number of messages
	  pending for MWI call

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
	  enhanced MWI generation to take advantage of new dahdi line
	  reversal MWI ability. (closes issue #14104) Reported by:
	  alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey
	  (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis
	  (license 585) Tested by: alecdavis, dbailey

2009-01-19 15:54 +0000 [r169211]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c, /: Merged revisions 169210 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon,
	  19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a
	  potential NULL pointer dereference Move the check for if both
	  channels on a local_pvt have generators to below where p->chan is
	  checked for NULLity (NULLness?). This prevents a crash from
	  occurring if p->chan is NULL. (closes issue #14189) Reported by:
	  sascha Patches: 14189.patch uploaded by putnopvut (license 60)
	  Tested by: sascha ........

2009-01-17 18:26 +0000 [r169153]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
	  discriminator for when ring pulse alert signal is used to preface
	  MWI spills This prevents the situation when MWI messages are
	  added to caller ID spills causing the channel to be hung up

2009-01-17 02:52 +0000 [r169116]  Sean Bright <sean.bright@gmail.com>

	* pbx/pbx_dundi.c: Change intializer types. Found while working on
	  asterisk-cpp. I have a new favorite error message from g++:
	  pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated
	  initializers not supported I like it when compilers are
	  apologetic.

2009-01-17 01:56 +0000 [r169044-169080]  Terry Wilson <twilson@digium.com>

	* main/tcptls.c, main/http.c, include/asterisk/tcptls.h: Fix
	  qualify for TCP peer (closes issue #14192) Reported by:
	  pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
	  jamesgolovich (license 176) Tested by: jamesgolovich

	* channels/chan_sip.c: Fix port :0 added to SIP INVITE URI when
	  outboundproxy used (closes issue #14233) Reported by: chris-mac
	  Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich
	  (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy

2009-01-16 22:43 +0000 [r168976]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 168975 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan
	  2009) | 18 lines Account for possible NULL pointer when we
	  receive a 408 in response to a REGISTER It may be that by the
	  time we receive a reply to a REGISTER request, the attempt has
	  timed out and thus the registry structure pointed to by the
	  corresponding sip_pvt has gone away. This situation was handled
	  properly for a 200 OK response, but the 408 case assumed that the
	  sip_registry struct was non-NULL, thus potentially causing a
	  crash This commit fixes this assumption and prints out a message
	  to the console if we should receive a late 408 response to a
	  REGISTER (closes issue #14211) Reported by: aborghi Patches:
	  14211.diff uploaded by putnopvut (license 60) Tested by: aborghi
	  ........

2009-01-16 22:16 +0000 [r168941]  Terry Wilson <twilson@digium.com>

	* /, main/features.c: Merged revisions 168716 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
	  | 12 lines Convert call to park_call_full to
	  masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
	  return value, we need to use masqueraded parking, otherwise we
	  will try to call ast_hangup() in __pbx_run() and in
	  do_parking_thread() and then promptly crash. (closes issue
	  #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
	  issue #14228) Reported by: kobaz Tested by: otherwiseguy ........

2009-01-16 19:54 +0000 [r168898]  Mark Michelson <mmichelson@digium.com>

	* res/res_timing_timerfd.c: Fix a logic error that occur when using
	  the timerfd interface This sequence of events posed a problem
	  timerfd_timer_open timerfd_timer_enable_continuous
	  timerfd_timer_set_rate timerfd_timer_disable_continuous The
	  reason was that the timing module was written under the
	  assumption that timerfd_timer_set_rate would not be called
	  between enabling and disabling continuous mode. What happened in
	  this situation was that timerfd_timer_enable_continuous saved off
	  our previously set timer (in this situation a 0 timer, meaning it
	  never runs out). Then timerfd_timer_disable_continuous would
	  restore this 0 timer, even though it logically should set the
	  timer to be whatever was set in timerfd_timer_set_rate. Now the
	  behavior in timerfd_timer_set_rate is to overwrite the saved
	  timer that may or may not have been set in
	  timerfd_timer_enable_continuous. Even if
	  timerfd_timer_enable_continuous has not been previously called,
	  this will not harm the operation. Thanks to Terry Wilson for
	  discovering the problem and giving me a really great debug
	  capture that pointed out the problem clearly

2009-01-16 18:49 +0000 [r168832]  Tilghman Lesher <tlesher@digium.com>

	* /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c:
	  Merged revisions 168828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009)
	  | 6 lines Fix the conjugation of Russian and Ukrainian languages.
	  (related to issue #12475) Reported by: chappell Patches:
	  vm_multilang.patch uploaded by chappell (license 8) ........

2009-01-16 17:09 +0000 [r168759-168760]  Russell Bryant <russell@digium.com>

	* CHANGES: Fix a spelling mistake.

	* channels/chan_misdn.c: build in dev mode

2009-01-16 00:34 +0000 [r168737-168746]  Steve Murphy <murf@digium.com>

	* res/ael/pval.c, /: Merged revisions 168745 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) |
	  14 lines This patch fixes a problem where a goto (or jump, in
	  this case) fails a consistency check because it can't find a
	  matching extension. The problem was a missing instruction to end
	  the range notation in the code where it converts the pattern into
	  a regex and uses the regex code to determine the match. I tested
	  using the AEL code the user supplied, and now, the consistency
	  check passes. (closes issue #14141) Reported by: dimas ........

	* main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: This patch
	  allows null args in ast_expr2 func calls, and fixes commas being
	  converted to pipes, which was 1.4 type stuff. If the user says
	  count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it
	  won't complain about the empty arg (c,,...) and fabled's patch
	  won't let it swap the commas for pipes. Ran it thru my dialplan
	  and no complaints. (closes issue #14169) Reported by: fabled
	  Patches: function-argument-separator-fix.diff uploaded by fabled
	  (license 448)

2009-01-15 20:18 +0000 [r168734]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c, build_tools/menuselect-deps.in, configure,
	  funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c,
	  cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c,
	  apps/app_voicemail.c: remove the PBX_ODBC logic from the
	  configure script, and add GENERIC_ODCB logic that includes
	  copying the relevant LIB and INCLUDE data from either UnixODBC or
	  iODBC, based on which was found; if both were found, prefer
	  UnixODBC this stops modules from being linked against both sets
	  of libraries on systems that have both installed

2009-01-15 20:00 +0000 [r168725-168732]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Add missing brace

	* channels/chan_sip.c: Fix the compactheaders option in sip.conf

	* channels/chan_sip.c: Remove an unneeded condition for line
	  addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the
	  sip_request structure had a statically allocated buffer to hold
	  the text of the request. There was a check in the add_line
	  function to not attempt to write the line into the buffer if we
	  did not have room for it. In trunk and Asterisk versions starting
	  with 1.6.1, an expandable ast_str structure is used to hold the
	  text. Since it may grow to fit an arbitrarily sized string, this
	  check in add_line is no longer valid. I found this oddity while
	  attempting to fix issue #14220; however, I do not believe that
	  this is the fix for that issue since the output supplied by the
	  reporter did not contain the warning message that would be
	  printed had this condition been satisfied.

2009-01-15 18:47 +0000 [r168722]  Olle Johansson <oej@edvina.net>

	* /, configs/extconfig.conf.sample: Merged revisions 168721 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2
	  lines Meetme actually has realtime but wasn't documented ........

2009-01-15 18:39 +0000 [r168719]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/strings.h: Resolve issue with negative vs
	  non-negative length parameters. (closes issue #14245) Reported
	  by: dveiga

2009-01-15 18:08 +0000 [r168711-168712]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Make sure that we have the same terminology
	  in sip.conf.sample and the source code warning. Thanks Nick Lewis
	  for pointing this out in the bug tracker.

	* configs/sip.conf.sample: Clarify some misunderstandings and make
	  it even more clear that you can refer to a peer in the register=
	  line.

2009-01-15 15:33 +0000 [r168705]  Sean Bright <sean.bright@gmail.com>

	* apps/app_meetme.c: Add a missing unlock and properly handle the
	  'maxusers' setting on MeetMe conferences. We were using the 'user
	  number' field to compare against the maximum allowed users, which
	  works assuming users with lower user numbers didn't leave the
	  conference. (closes issue #14117) Reported by: sergedevorop
	  Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright
	  (license 71) Tested by: sergedevorop

2009-01-15 13:37 +0000 [r168636-168639]  Olle Johansson <oej@edvina.net>

	* CREDITS, CHANGES: Related to issue #14246 Update changes for
	  SIPRemoveHeader()

	* channels/chan_sip.c: Add capability to remove added SIP headers
	  *before* INVITE is generated. (closes issue #14246) Reported by:
	  klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt
	  uploaded by klaus3000 (license 65)

	* apps/app_queue.c: Add support for setting the Reason header when
	  cancelling a call in the queue because someone else answered.
	  Previously, only dial() was supported. EDV-102

2009-01-15 00:14 +0000 [r168629]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 168628 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan
	  2009) | 16 lines Fix some crashes from bad datastore handling in
	  app_queue.c * The queue_transfer_fixup function was searching for
	  and removing the datastore from the incorrect channel, so this
	  was fixed. * Most datastore operations regarding the
	  queue_transfer datastore were being done without the channel
	  locked, so proper channel locking was added, too. (closes issue
	  #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
	  putnopvut (license 60) Tested by: ZX81, festr ........

2009-01-14 23:10 +0000 [r168626]  Sean Bright <sean.bright@gmail.com>

	* main/cli.c: Don't crash when typing 'core set verbose' or 'core
	  set debug' by themselves. (closes issue #14219) Reported by:
	  jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded
	  by jamesgolovich (license 176)

2009-01-14 21:51 +0000 [r168623]  Richard Mudgett <rmudgett@digium.com>

	* /, channels/misdn/isdn_lib.c: Merged revisions 168622 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009)
	  | 4 lines * Fixed create_process() allocation of process ID
	  values. The allocated process IDs could overflow their respective
	  NT and TE fields. Affects outgoing calls. ........

2009-01-14 21:19 +0000 [r168619]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c: This fixes a problem where MWI FSK spills
	  were being injected onto off hook fxs lines. (closes issue
	  #14143) Reported by: alecdavis Patches:
	  chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested
	  by: alecdavis

2009-01-14 20:58 +0000 [r168615]  Sean Bright <sean.bright@gmail.com>

	* /, contrib/scripts/autosupport: Merged revisions 168614 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan
	  2009) | 9 lines Update autosupport script to supply info for both
	  Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x
	  and trunk instead of zttest. (closes issue #14132) Reported by:
	  dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by
	  dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded
	  by dsedivec (license 638) ........

2009-01-14 20:51 +0000 [r168613]  Steve Murphy <murf@digium.com>

	* /, apps/app_page.c: Merged revisions 168608 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1
	  line app_page was failing to compile in dev-mode on my gcc-4.2.4
	  system. This change gets rid of the warning. ........

2009-01-14 20:13 +0000 [r168610]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Restore the "sip show users" and "sip show
	  user" CLI commands (closes issue #14180) Reported by: amorsen
	  Patches: sip_show_users_161v3.diff uploaded by putnopvut (license
	  60) Tested by: blitzrage, amorsen

2009-01-14 19:36 +0000 [r168609]  Michiel van Baak <michiel@vanbaak.info>

	* main/asterisk.c: Fix compilation on FreeBSD and OSX This started
	  as work to fix the 'core show sysinfo' CLI command but while
	  working on it oej pointed out that read_credentials did not
	  compile neither. So while being there, fix that as well. Thanks
	  for all the testing oej! (closes issue #14129) Reported by: ys
	  Tested by: oej, mvanbaak

2009-01-14 19:11 +0000 [r168601-168604]  Tilghman Lesher <tlesher@digium.com>

	* main/udptl.c, /: Merged revisions 168603 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009)
	  | 7 lines Don't read into a buffer without first checking if a
	  value is beyond the end. (closes issue #13600) Reported by: atis
	  Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
	  (license 14) Tested by: atis ........

	* channels/chan_misdn.c: Mostly spacing changes; no functionality
	  change at all.

2009-01-14 02:00 +0000 [r168594]  Terry Wilson <twilson@digium.com>

	* /, apps/app_page.c: Merged revisions 168593 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009)
	  | 20 lines Don't overflow when paging more than 128 extensions
	  The number of available slots for calls in app_page was hardcoded
	  to 128. Proper bounds checking was not in place to enforce this
	  limit, so if more than 128 extensions were passed to the Page()
	  app, Asterisk would crash. This patch instead dynamically
	  allocates memory for the ast_dial structures and removes the
	  (non-functional) arbitrary limit. This issue would have special
	  importance to anyone who is dynamically creating the argument
	  passed to the Page application and allowing more than 128
	  extensions to be added by an outside user via some external
	  interface. The patch posted by a_villacis was slightly modified
	  for some coding guidelines and other cleanups. Thanks,
	  a_villacis! (closes issue #14217) Reported by: a_villacis
	  Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
	  uploaded by a (license 660) Tested by: otherwiseguy ........

2009-01-13 23:57 +0000 [r168591]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_misdn.c: Janitor patch for chan_misdn (make channel
	  variable access safe) (closes issue #12887) Reported by: pputman
	  Patches: chan_misdn_threadsafe.patch uploaded by pputman (license
	  81)

2009-01-13 23:05 +0000 [r168585-168588]  Terry Wilson <twilson@digium.com>

	* res/res_http_post.c: Fully overwrite a same-named file when
	  uploading (closes issue #14190) Reported by: timking

	* Makefile, include/asterisk/options.h, main/asterisk.c: Add option
	  to hide console connect messages (closes issue #14222) Reported
	  by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded
	  by jamesgolovich (license 176) Tested by: otherwiseguy

2009-01-13 22:30 +0000 [r168579]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Clarify a message that app_queue prints and
	  change to a debug-level message The "No one is answering..."
	  verbose message contained 3 numbers that were not explained in
	  any way to whoever was viewing the message. It is more helpful
	  now since the message explains what the numbers mean. Also, the
	  message has been downgraded to "DEBUG" level. (closes issue
	  #14172) Reported by: caio1982 Patches: queue_answering_debug.diff
	  uploaded by caio1982 (license 22)

2009-01-13 22:22 +0000 [r168578]  Terry Wilson <twilson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 168551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009)
	  | 7 lines Don't pass a value with a side effect to a macro
	  (closes issue #14176) Reported by: paraeco Patches:
	  chan_sip.c.diff uploaded by paraeco (license 658) ........

2009-01-13 21:18 +0000 [r168575]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow
	  specifying a port number in the user portion of a register =>
	  line in sip.conf With this commit, a register => line in sip.conf
	  may contain a port number in the "user" section of the line.
	  Please see CHANGES and sip.conf.sample for more details regarding
	  this. (closes issue #14198) Reported by: Nick_Lewis Patches:
	  chan_sip.c-domainport2.patch uploaded by Nick (license 657)
	  Tested by: Nick_Lewis

2009-01-13 19:22 +0000 [r168562]  Russell Bryant <russell@digium.com>

	* channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /,
	  include/asterisk/indications.h, apps/app_readexten.c,
	  apps/app_disa.c, include/asterisk/channel.h, main/indications.c,
	  main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
	  funcs/func_channel.c, main/app.c, res/snmp/agent.c,
	  res/res_indications.c: Merged revisions 168561 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009)
	  | 2 lines Revert unnecessary indications API change from rev
	  122314 ........

2009-01-13 17:51 +0000 [r168547]  Tilghman Lesher <tlesher@digium.com>

	* /, funcs/func_logic.c: Merged revisions 168546 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009)
	  | 6 lines If either conditional is NULL, don't try copying it.
	  (closes issue #14226) Reported by: caspy Patches:
	  20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
	  ........

2009-01-13 16:02 +0000 [r168539]  Dwayne M. Hubbard <dwayne.hubbard@gmail.com>

	* main/taskprocessor.c: correct a CLI description

2009-01-12 23:45 +0000 [r168526]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_alsa.c: Merged revisions 167095 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31
	  Dec 2008) | 5 lines Repeat attempts to write when we receive
	  -EAGAIN from the driver, as detailed in the ALSA sample code (see
	  http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
	  Reported by: Jerry Geis (via the -users list) Fixed by: me
	  (license 14) ........

2009-01-12 23:12 +0000 [r168523]  Mark Michelson <mmichelson@digium.com>

	* main/srv.c: bump the verbosity of a message in srv.c up by one.
	  It used to be at this level prior to a large patch merge which
	  converted ast_verbose calls to ast_verb (closes issue #14221)
	  Reported by: jcovert Patches: srv.c.patch uploaded by jcovert
	  (license 551)

2009-01-12 23:06 +0000 [r168522]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/app.c: Some platforms (notably, the BSDs) have a more
	  efficient implementation called closefrom(3).

2009-01-12 21:51 +0000 [r168508-168517]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_agi.c: Merged revisions 168516 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009)
	  | 5 lines (closes issue #13881) Reported by: hoowa Update the app
	  CDR field for AGI commands that are not executing an application
	  via "exec". ........

	* /, channels/chan_agent.c: Merged revisions 168507 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12
	  Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG
	  Tested by: denisgalvao This gits rid of the notion of an
	  owning_app allowing the request and hangup to be initiated by
	  different threads. Originating from an active agent channel
	  requires this. The implementation primarily changes __login_exec
	  to wait on a condition variable rather than a lock. Review:
	  http://reviewboard.digium.com/r/35/ ........

2009-01-12 16:31 +0000 [r168497]  Olle Johansson <oej@edvina.net>

	* apps/app_minivm.c: Better to use the proper app name

2009-01-12 15:00 +0000 [r168485]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Merged revisions 168482 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan
	  2009) | 5 lines I am reverting the fix made in revision 168128
	  (and its upward merges) after being contacted by Olle Johansson
	  and being shown how this fix is incorrect. Thanks to Olle for
	  clearing this up for me. ........

2009-01-12 14:57 +0000 [r168481]  Russell Bryant <russell@digium.com>

	* /, configs/indications.conf.sample: Merged revisions 168480 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009)
	  | 2 lines s/ringdance/ringcadence/ for Bulgaria ........

2009-01-12 14:35 +0000 [r168479]  Olle Johansson <oej@edvina.net>

	* main/asterisk.c: Don't include swap.h unless we have swapctl

2009-01-10 01:42 +0000 [r168334]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for
	  reconstructing a field value.

2009-01-09 23:16 +0000 [r168270]  Kevin P. Fleming <kpfleming@digium.com>

	* /, sounds/Makefile: Merged revisions 168267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan
	  2009) | 1 line update to use new sound file packages that include
	  license files ........

2009-01-09 23:15 +0000 [r168269]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c: Spacing change

2009-01-09 23:04 +0000 [r168265]  Michiel van Baak <michiel@vanbaak.info>

	* contrib/scripts/sip_nat_settings (added), CHANGES: Add a script
	  to find out the correct settings for Asterisk behind NAT (closes
	  issue #13065) Reported by: tzafrir Patches: sip_nat_settings
	  uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by
	  mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and
	  moi

2009-01-09 22:21 +0000 [r168200]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 168198 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09
	  Jan 2009) | 2 lines Make this compile for mvanbaak ........

2009-01-09 21:53 +0000 [r168193]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 168128 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan
	  2009) | 13 lines Add check_via calls to more request handlers
	  INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
	  checking the topmost Via to determine where to send the response.
	  Adding check_via calls to those request handlers solves this.
	  (closes issue #13071) Reported by: baron Patches: check_via.patch
	  uploaded by baron (license 531) Tested by: baron ........

2009-01-09 21:43 +0000 [r168192]  Richard Mudgett <rmudgett@digium.com>

	* channels/chan_misdn.c, /: Merged revisions 168191 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09
	  Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 *
	  Miscellaneous doxygen comments added. ........

2009-01-09 20:25 +0000 [r168142]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Don't leak memory if phoneprov.conf does not
	  exist (closes issue #14203) Reported by: jamesgolovich Patches:
	  asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich
	  (license 176)

2009-01-09 18:30 +0000 [r168090]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c, include/asterisk/strings.h: When using ast_str
	  with a non-ast_str-enabled API, we need to update the buffer or
	  otherwise, we cannot use ast_str_strlen().

2009-01-09 18:01 +0000 [r168014-168054]  Matthew Nicholson <mnicholson@digium.com>

	* main/logger.c: Added a comment to logger.c about where to put
	  includes

	* main/logger.c: Use ast_safe_system() in logger.c instead of
	  system() (closes issue #14194) Reported by: pabelanger

2009-01-09 01:15 +0000 [r167935-167973]  Terry Wilson <twilson@digium.com>

	* apps/app_originate.c: Set ORIGINATE_STATUS instead of
	  OUTGOING_STATUS to match the documentation

	* apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN
	  and MACRO_CONTEXT will be set

2009-01-08 22:37 +0000 [r167894]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 167840 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009)
	  | 6 lines Don't truncate database results at 255 chars. (closes
	  issue #14069) Reported by: evandro Patches:
	  20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
	  ........

2009-01-08 22:34 +0000 [r167888]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Revert chan_sip changes which were
	  accidentally committed in revision 167792

2009-01-08 21:40 +0000 [r167835-167837]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_minivm.c: Fix variables to comply with documentation
	  changes

	* apps/app_minivm.c: Textual changes, consistency in status
	  variable naming, and other minor bugs. (closes issue #13943)
	  Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded
	  by Marquis (license 32)

2009-01-08 19:48 +0000 [r167792]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average
	  talk time for a queue This patch adds the functionality to
	  app_queue of calculating the average amount of time that channels
	  are bridged for a queue. The algorithm used to calculate the
	  average is the same exponential average currently used to
	  calculate the average holdtime. See the CHANGES file to see the
	  methods you may use to view this information. (closes issue
	  #13960) Reported by: coolmig Patches:
	  app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)

2009-01-08 19:44 +0000 [r167791]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_dahdi.c, CHANGES: Convert dialplan application
	  DAHDISendCallreroutingFacility to use commas. (closes issue
	  #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded
	  by eliel (license 64)

2009-01-08 17:26 +0000 [r167700-167720]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 167714 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan
	  2009) | 1 line remove an unnecessary argument to queue_request()
	  ........

	* channels/chan_sip.c: Merged revisions 167620 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan
	  2009) | 5 lines When a SIP request or response arrives for a
	  dialog with an associated Asterisk channel, and the lock on that
	  channel cannot be obtained because it is held by another thread,
	  instead of dropping the request/response, queue it for later
	  processing when the channel lock becomes available.
	  http://reviewboard.digium.com/r/123/ ........

2009-01-08 14:27 +0000 [r167662]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/sip-friends.sql: Oops... fix the fieldname I
	  changed yesterday to be right.

2009-01-07 22:36 +0000 [r167542-167569]  Russell Bryant <russell@digium.com>

	* /, main/file.c: Merged revisions 167566 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009)
	  | 2 lines Fix the last couple of places where free() was
	  improperly used directly. ........

	* /, main/file.c: Merged revisions 167554 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009)
	  | 2 lines Don't fclose() the file early, the filestream
	  destructor will handle it. ........

	* /, main/file.c: Merged revisions 167545 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009)
	  | 2 lines Only try to close the file if one was actually opened
	  ........

	* /, main/file.c: Merged revisions 167541 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009)
	  | 4 lines Don't use free() directly. This caused a crash since
	  ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
	  #asterisk-dev ........

2009-01-07 18:20 +0000 [r167478]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Answer the channel if it has not already
	  been answered and we've already found a valid profile for
	  followme. (closes issue #14140) Reported by: dimas Patches:
	  14140.patch uploaded by dimas

2009-01-07 18:18 +0000 [r167477]  Leif Madsen <lmadsen@digium.com>

	* configs/queues.conf.sample: Update queues.conf.sample
	  documentation. Update the queues.conf.sample documentation to
	  mention that you need to preload chan_local.so as well if you
	  plan on using Local channels for queue members, and you're
	  preloading pbx_config.so. (closes issue #14179) Reported by:
	  CrashHD Tested by: CrashHD

2009-01-07 17:35 +0000 [r167442]  Russell Bryant <russell@digium.com>

	* /, main/indications.c: Merged revisions 167432 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009)
	  | 4 lines Treat an empty string the same way as a NULL country
	  argument. In passing, simplify the handling of returning a
	  default tone zone. ........

2009-01-07 17:05 +0000 [r167416]  Doug Bailey <dbailey@digium.com>

	* channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected
	  during mwi spill. Correct logic error in handling events when
	  sending mwi spill (closes issue #14143) Reported by: alecdavis
	  Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by
	  dbailey

2009-01-07 14:26 +0000 [r167373]  Leif Madsen <lmadsen@digium.com>

	* contrib/scripts/sip-friends.sql: Update the sip-friends.sql file
	  to use the non-deprecated 'defaultname' instead of 'username' and
	  remove an extra comma that would cause the script to fail as-is

2009-01-06 21:36 +0000 [r167301]  Mark Michelson <mmichelson@digium.com>

	* /, main/db.c: Merged revisions 167299 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan
	  2009) | 8 lines Use the correct variable when creating the format
	  string (closes issue #14177) Reported by: nic_bellamy Patches:
	  asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
	  (license 299) ........

2009-01-06 21:02 +0000 [r167265]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 167260 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600
	  (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
	  Jan 2009) | 2 lines Security fix AST-2009-001. ........
	  ................

2009-01-05 16:59 +0000 [r167180]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 167179 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan
	  2009) | 41 lines A couple of changes to T.38 SDP attribute
	  handling There are some boolean attributes for T.38 such as
	  T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
	  T38FaxTranscodingJBIG. By simply being present, we should treat
	  these as a "true" value. The current code, however, was requiring
	  a 1 or 0 as the value of the attribute in order to parse it. This
	  is due to the fact that there are some T.38 endpoints and
	  gateways that also transmit this information incorrectly. This
	  patch follows the "be liberal in what you accept and strict in
	  what you send" philosophy by accepting both the correctly- and
	  incorrectly-formatted attributes, but only sending information as
	  it is supposed to be sent. It was also discovered that a
	  particular type of T.38 gateway sends some non-standard T.38 SDP
	  attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
	  it used T38MaxDatagram and T38FaxMaxRate respectively. We now
	  will properly accept these attributes as well. Note that there
	  are a lot of patches cited in the below commit message template.
	  This is because the person who submitted these patches is an
	  awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
	  issue #13976) Reported by: linulin Patches:
	  chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
	  chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
	  chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
	  chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
	  (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
	  by arcivanov (license 648)
	  chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
	  (license 648) Tested by: arcivanov ........

2009-01-05 16:44 +0000 [r167176]  Tilghman Lesher <tlesher@digium.com>

	* UPGRADE-1.6.txt: More clearly explain that quote marks are no
	  longer necessary. (closes issue #13718) Reported by: davidw
	  Patches: 20081020__bug13718.diff.txt uploaded by Corydon76
	  (license 14) Tested by: blitzrage

2009-01-03 20:29 +0000 [r167125]  Jeff Peeler <jpeeler@digium.com>

	* main/asterisk.c: When parsing environment variable
	  ASTERISK_PROMPT, make sure to proceed to the next character when
	  a non format specifier is used (no %). Otherwise, the while loop
	  looking for the null byte will never exit.

2008-12-31 23:07 +0000 [r167061]  Sean Bright <sean.bright@gmail.com>

	* doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README:
	  Mostly just whitespace, but also convert 'CVS' to 'SVN' in a
	  couple places and fix a few typos I found in the
	  CODING_GUIDELINES.

2008-12-31 22:53 +0000 [r167057]  Terry Wilson <twilson@digium.com>

	* main/xmldoc.c: Don't forget to free typename

2008-12-31 21:52 +0000 [r167021]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c: Change some incorrect syntax for pri set
	  debug and correct an off-by-one error in ss7 set debug command

2008-12-31 19:39 +0000 [r166954-166958]  Tilghman Lesher <tlesher@digium.com>

	* main/ast_expr2.h, main/ast_expr2.c: That was weird...

	* channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c:
	  Merged revisions 166953 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008)
	  | 5 lines Also inherit the musiconhold class. (Closes #14153)
	  Reported by: Jerry Geis, via the users list. Patch by: me
	  (license 14) ........

2008-12-30 20:50 +0000 [r166908]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c, doc/sip-retransmit.txt,
	  doc/tex/phoneprov.tex, res/res_http_post.c,
	  phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some
	  svn:keywords

2008-12-29 18:04 +0000 [r166861]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, apps/app_queue.c: Update app_queue to deal with
	  the removal of AST_PBX_KEEPALIVE When placing a call to a queue
	  which ran a gosub on the member's channel, Asterisk would crash
	  every time, stemming from the fact that the member's channel was
	  being hung up unexpectedly when the Gosub completed. The
	  necessary change was pretty much copied and pasted from
	  app_dial's similar changes made last week. I also took the
	  opportunity to change a LOG_DEBUG message in app_dial to use
	  ast_debug. I am guessing this was due to a direct merge from 1.4
	  that was not corrected to use trunk's preferred syntax.

2008-12-28 15:36 +0000 [r166823]  Eliel C. Sardanons <eliels@gmail.com>

	* funcs/func_audiohookinherit.c: Fix a typo in the XML
	  documentation of the AUDIOHOOK_INHERIT dialplan function.

2008-12-28 15:15 +0000 [r166773]  Russell Bryant <russell@digium.com>

	* /, channels/misdn_config.c: Merged revisions 166772 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28
	  Dec 2008) | 4 lines Use strncat() instead of an sprintf() in
	  which source and target buffers overlap
	  http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
	  ........

2008-12-24 15:10 +0000 [r166731]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I
	  believe 26.2.2 is what was meant: Note that in the SIPS URI
	  scheme, transport is independent of TLS, and thus
	  "sips:alice@atlanta.com;transport=tcp" and
	  "sips:alice@atlanta.com;transport=sctp" are both valid (although
	  note that UDP is not a valid transport for SIPS). The use of
	  "transport=tls" has consequently been deprecated, partly because
	  it was specific to a single hop of the request. This is a change
	  since RFC 2543.

2008-12-23 20:47 +0000 [r166696]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Allow semicolons and extended characters in
	  user-specified SIP headers. (closes issue #14110) Reported by:
	  gork Patches: 20081222__bug14110__2.diff.txt uploaded by
	  Corydon76 (license 14) Tested by: gork, putnopvut

2008-12-23 18:13 +0000 [r166665]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/pbx.c, /, main/features.c,
	  apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c,
	  include/asterisk/features.h: Merged revisions 166093 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4 In
	  order to merge this 1.4 patch into trunk, I had to resolve some
	  conflicts and wait for Russell to make some changes to res_agi. I
	  re-ran all the tests; 39 calls in all, and made fairly careful
	  notes and comparisons: I don't want this to blow up some aspect
	  of asterisk; I completely removed the KEEPALIVE from the pbx.h
	  decls. The first 3 scenarios involving feature park; feature xfer
	  to 700; hookflash park to Park() app call all behave the same,
	  don't appear to leave hung channels, and no crashes. ........
	  r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) |
	  131 lines This merges the masqpark branch into 1.4 These changes
	  eliminate the need for (and use of) the KEEPALIVE return code in
	  res_features.c; There are other places that use this result code
	  for similar purposes at a higher level, these appear to be left
	  alone in 1.4, but attacked in trunk. The reason these changes are
	  being made in 1.4, is that parking ends a channel's life, in some
	  situations, and the code in the bridge (and some other places),
	  was not checking the result code properly, and dereferencing the
	  channel pointer, which could lead to memory corruption and
	  crashes. Calling the masq_park function eliminates this danger in
	  higher levels. A series of previous commits have replaced some
	  parking calls with masq_park, but this patch puts them ALL to
	  rest, (except one, purposely left alone because a masquerade is
	  done anyway), and gets rid of the code that tests the KEEPALIVE
	  result, and the NOHANGUP_PEER result codes. While bug 13820
	  inspired this work, this patch does not solve all the problems
	  mentioned there. I have tested this patch (again) to make sure I
	  have not introduced regressions. Crashes that occurred when a
	  parked party hung up while the parking party was listening to the
	  numbers of the parking stall being assigned, is eliminated. These
	  are the cases where parking code may be activated: 1. Feature one
	  touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3.
	  Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi
	  hookflash xfer to 700) 4. Run Park via manager. The interesting
	  testing cases for parking are: I. A calls B, A parks B a. B hangs
	  up while A is getting the numbers announced. b. B hangs up after
	  A gets the announcement, but before the parking time expires c. B
	  waits, time expires, A is redialed, A answers, B and A are
	  connected, after which, B hangs up. d. C picks up B while still
	  in parking lot. II. A calls B, B parks A a. A hangs up while B is
	  getting the numbers announced. b. A hangs up after B gets the
	  announcement, but before the parking time expires c. A waits,
	  time expires, B is redialed, B answers, A and B are connected,
	  after which, A hangs up. d. C picks up A while still in parking
	  lot. Testing this throroughly involves acting all the
	  permutations of I and II, in situations 1,2,3, and 4. Since I
	  added a few more changes (ALL references to KEEPALIVE in the
	  bridge code eliimated (I missed one earlier), I retested most of
	  the above cases, and no crashes. H-extension weirdness. Current
	  h-extension execution is not completely correct for several of
	  the cases. For the case where A calls B, and A parks B, the 'h'
	  exten is run on A's channel as soon as the park is accomplished.
	  This is expected behavior. But when A calls B, and B parks A,
	  this will be current behavior: After B parks A, B is hung up by
	  the system, and the 'h' (hangup) exten gets run, but the channel
	  mentioned will be a derivative of A's... Thus, if A is DAHDI/1,
	  and B is DAHDI/2, the h-extension will be run on channel
	  Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be
	  those relating to Channel A. And, in the case where A is
	  reconnected to B after the park time expires, when both parties
	  hang up after the joyful reunion, no h-exten will be run at all.
	  In the case where C picks up A from the parking lot, when either
	  A or C hang up, the h-exten will be run for the C channel. CDR's
	  are a separate issue, and not addressed here. As to WHY this
	  strange behavior occurs, the answer lies in the procedure
	  followed to accomplish handing over the channel to the parking
	  manager thread. This procedure is called masquerading. In the
	  process, a duplicate copy of the channel is created, and most of
	  the active data is given to the new copy. The original channel
	  gets its name changed to XXX<ZOMBIE> and keeps the PBX
	  information for the sake of the original thread (preserving its
	  role as a call originator, if it had this role to begin with),
	  while the new channel is without this info and becomes a call
	  target (a "peer"). In this case, the parking lot manager thread
	  is handed the new (masqueraded) channel. It will not run an
	  h-exten on the channel if it hangs up while in the parking lot.
	  The h exten will be run on the original channel instead, in the
	  original thread, after the bridge completes. See bug 13820 for
	  our intentions as to how to clean up the h exten behavior.
	  Review: http://reviewboard.digium.com/r/29/ ........

2008-12-23 16:04 +0000 [r166625]  Russell Bryant <russell@digium.com>

	* CHANGES: Fix spelling error.

2008-12-23 15:17 +0000 [r166569]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c, /: Merged revisions 166568 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec
	  2008) | 12 lines Fix a crash resulting from a datastore with
	  inheritance but no duplicate callback The fix for this is to
	  simply set the newly created datastore's data pointer to NULL if
	  it is inherited but has no duplicate callback. (closes issue
	  #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
	  putnopvut (license 60) Tested by: francesco_r ........

2008-12-23 04:32 +0000 [r166533]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, /: Merged revisions 166509 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008)
	  | 4 lines Use the integer form of condition for integer
	  comparisons. (closes issue #14127) Reported by: andrew ........

2008-12-22 23:25 +0000 [r166470]  Mark Michelson <mmichelson@digium.com>

	* res/res_agi.c: Always use the value of the AGISIGHUP when running
	  an AGI. Prior to this patch, the value of AGISIGUP was not always
	  honored when set on a channel. (closes issue #13711) Reported by:
	  fmueller Patches: 13711.patch uploaded by putnopvut (license 60)

2008-12-22 21:45 +0000 [r166436]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Cosmetic change - don't mix struct
	  initializer styles.

2008-12-22 21:08 +0000 [r166382]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_dahdi.c, /: Merged revisions 166380 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon,
	  22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks
	  and autoservice It has been discovered that if a channel is
	  locked prior to a call to ast_autoservice_stop, then it is likely
	  that a deadlock will occur. The reason is that the call to
	  ast_autoservice_stop has a check built into it to be sure that
	  the thread running autoservice is not currently trying to
	  manipulate the channel we are about to pull out of autoservice.
	  The autoservice thread, however, cannot advance beyond where it
	  currently is, though, because it is trying to acquire the lock of
	  the channel for which autoservice is attempting to be stopped.
	  The gist of all this is that a channel MUST NOT be locked when
	  attempting to stop autoservice on the channel. In this particular
	  case, the channel was locked by a call to ast_read. A call to
	  ast_exists_extension led to autoservice being started and stopped
	  due to the existence of dialplan switches. It may be that there
	  are future commits which handle the same symptoms but in a
	  different location, but based on my looks through the code, it is
	  very rare to see a construct such as this one. (closes issue
	  #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
	  by putnopvut (license 60) Tested by: rtrauntvein Review:
	  http://reviewboard.digium.com/r/107/ ........

2008-12-22 20:26 +0000 [r166273-166377]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix a bad typo.

	* main/astobj2.c: Remove some error messages. This is the default
	  handler that is valid to use.

	* /, main/utils.c: Merged revisions 166297 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008)
	  | 2 lines Fix up timeout handling in ast_carefulwrite(). ........

	* include/asterisk/utils.h, main/manager.c, main/utils.c: Introduce
	  ast_careful_fwrite() and use in AMI to prevent partial writes.
	  This patch introduces a function to do careful writes on a file
	  stream which will handle timeouts and partial writes. It is
	  currently used in AMI to address the issue that has been
	  reported. However, there are probably a few other places where
	  this could be used. (closes issue #13546) Reported by: srt Tested
	  by: russell http://reviewboard.digium.com/r/104/

	* res/res_musiconhold.c: Re-work ref count handling of MoH classes
	  using astobj2 to resolve crashes. (closes issue #13566) Reported
	  by: igorcarneiro Tested by: russell Review:
	  http://reviewboard.digium.com/r/106/

2008-12-22 16:08 +0000 [r166268]  Joshua Colp <jcolp@digium.com>

	* main/dnsmgr.c: Record the previous port in the temporary address
	  structure so that the comparison does not treat the host as
	  having changed even if it did not. This would have been
	  uninitialized before and would have led to a baddddd port.
	  (closes issue #13628) Reported by: pananix Patches:
	  bug13628.patch uploaded by jpeeler (license 325) Tested by: file,
	  blitzrage

2008-12-22 16:07 +0000 [r166267]  Mark Michelson <mmichelson@digium.com>

	* funcs/func_timeout.c, main/file.c: Fix a file playback crash and
	  explicitly initialize values in func_timeout.c A crash was
	  brought up on the bugtracker. The first run through valgrind was
	  full of legitimate complaints of uninitialized values in
	  func_timeout when setting a response timeout. These were fixed
	  but the crash persisted. A second run through showed the real
	  problem. The reference counting used for filestreams was
	  incorrect because there were some missing increments when a frame
	  was read from a format module. (closes issue #14118) Reported by:
	  blitzrage Patches: 14118v2.patch uploaded by putnopvut (license
	  60) Tested by: blitzrage

2008-12-22 14:16 +0000 [r166258]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Remove AST_PBX_KEEPALIVE usage from res_agi. This
	  patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The
	  only usage was for the AGI command, "asyncagi break". This patch
	  removes this feature. Normally, a feature would not be removed
	  like this. However, this code is broken and usage of it will
	  result in a memory leak. Usage of this feature will make the AGI
	  code return a result of AST_PBX_KEEPALIVE. The PBX handler
	  assumes that another thread has assumed ownership of the channel.
	  The channel thread will exit without destroying the channel.
	  Unfortunately, _no_ thread has ownership of the channel at this
	  point. There are a couple of serious problems here: 1) The only
	  way to recover the caller is to issue a channel redirect. This
	  will work, but this will be done with a masquerade, and the old
	  ast_channel structure will be lost. 2) Until the channel redirect
	  happens, there is no code servicing the channel. That means
	  nothing is reading audio or handling events coming from the
	  channel. This is very bad. The recommended way to get this same
	  "break" functionality is to issue the redirect while the channel
	  is still being handled by the AGI code. That way, there will be
	  no memory leak, and there will be no period of time that the
	  channel is not being serviced.

2008-12-20 01:37 +0000 [r166219]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxyref.h: Make a note about formatting the
	  review URL in commit messages

2008-12-19 23:45 +0000 [r166092-166162]  Mark Michelson <mmichelson@digium.com>

	* main/audiohook.c: Get rid of an extra space. I don't know how
	  this crept back in when I had already fixed it earlier

	* funcs/func_audiohookinherit.c: Remove the verbatim tag from the
	  author line I could have sworn I already did that before,
	  though...

	* main/channel.c, funcs/func_audiohookinherit.c (added),
	  include/asterisk/audiohook.h, main/audiohook.c, CHANGES: Adding a
	  new dialplan function AUDIOHOOK_INHERIT This function is being
	  added as a method to allow for an audiohook to move to a new
	  channel during a channel masquerade. The most obvious use for
	  such a facility is for MixMonitor when a transfer is performed.
	  Prior to the addition of this functionality, if a channel running
	  MixMonitor was transferred by another party, then the recording
	  would stop once the transfer had completed. By using
	  AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the
	  call even after the transfer has completed. It has also been
	  determined that since this is seen by most as a bug fix and is
	  not an invasive change, this functionality will also be
	  backported to 1.4 and merged into the 1.6.0 branches, even though
	  they are feature-frozen. (closes issue #13538) Reported by: mbit
	  Patches: 13538.patch uploaded by putnopvut (license 60) Tested
	  by: putnopvut Review: http://reviewboard.digium.com/r/102/

2008-12-19 21:44 +0000 [r166058]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_dahdi.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Add configuration
	  support for half_full DAHDI buffer policy

2008-12-19 18:20 +0000 [r165954]  Eliel C. Sardanons <eliels@gmail.com>

	* apps/app_record.c: Fix the XML documentation for Record().
	  <value> tags inside <variable> elements must have CDATA and no
	  another XML node.

2008-12-19 15:05 +0000 [r165801-165890]  Russell Bryant <russell@digium.com>

	* /, apps/app_chanspy.c: Merged revisions 165889 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008)
	  | 9 lines Ensure that the chanspy datastore is fully initialized.
	  This patch resolved some random crash issues observed by a user
	  on a BSD system (closes issue #14111) Reported by: ys Patches:
	  app_chanspy.c.diff uploaded by ys (license 281) ........

	* include/asterisk/doxyref.h: Disable some automatic links
	  generated by doxygen.

	* include/asterisk/doxyref.h: Introduce commit message formatting
	  guidelines. This documents the recommended outline to use for
	  commit message. It also covers information on special tags that
	  can be used in commit messages, as well as the template
	  functionality that is available on bugs.digium.com. Review:
	  http://reviewboard.digium.com/r/96/

	* /, main/utils.c: Merged revisions 165796 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008)
	  | 11 lines Make ast_carefulwrite() be more careful. This patch
	  handles some additional cases that could result in partial writes
	  to the file description. This was done to address complaints
	  about partial writes on AMI. (issue #13546) (more changes needed
	  to address potential problems in 1.6) Reported by: srt Tested by:
	  russell Review: http://reviewboard.digium.com/r/99/ ........

2008-12-18 21:43 +0000 [r165798]  Jeff Peeler <jpeeler@digium.com>

	* main/manager.c: (closes issue #13993) Reported by: mika Add
	  ActionID response to ping if sent with request.

2008-12-18 21:41 +0000 [r165797]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 165767 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18
	  Dec 2008) | 8 lines Add mutexes around accesses to the IMAP
	  library interface. This prevents certain crashes, especially when
	  shared mailboxes are used. (closes issue #13653) Reported by:
	  howardwilkinson Patches:
	  asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
	  howardwilkinson (license 590) Tested by: jpeeler ........

2008-12-18 21:21 +0000 [r165792]  Joshua Colp <jcolp@digium.com>

	* channels/chan_dahdi.c, channels/chan_misdn.c,
	  channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c,
	  channels/chan_oss.c: Numerous documentation updates. (closes
	  issue #13970) Reported by: pkempgen Patches:
	  __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage
	  (license 10)

2008-12-18 19:34 +0000 [r165724]  Mark Michelson <mmichelson@digium.com>

	* res/res_odbc.c: Fix crashes in res_odbc. The variable "class" was
	  being set NULL just prior to being dereferenced in an ao2_link
	  call. I have moved the setting of the variable to NULL until
	  after the ao2_link call.

2008-12-18 19:33 +0000 [r165662-165723]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: Remove the
	  need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This
	  is part of an effort to completely remove AST_PBX_KEEPALIVE and
	  other similar return codes from the source. While this usage was
	  perfectly safe, there are others that are problematic. Since we
	  know ahead of time that we do not want to PBX to destroy the
	  channel, the PBX API has been changed so that information can be
	  provided as an argument, instead, thus removing the need for the
	  KEEPALIVE return value. Further changes to get rid of KEEPALIVE
	  and related code is being done by murf. There is a patch up for
	  that on review 29. Review: http://reviewboard.digium.com/r/98/

	* /, res/res_musiconhold.c: Merged revisions 165661 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18
	  Dec 2008) | 7 lines Set the process group ID on the MOH process
	  so that all children will get killed (closes issue #14099)
	  Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2
	  uploaded by caspy (license 645) ........

2008-12-18 18:36 +0000 [r165658]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: Fix 2 resource leaks and fix another
	  pipe-to-comma conversion

2008-12-18 17:13 +0000 [r165599]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 165591 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4
	  lines Only care about a compatible codec for early bridging if we
	  are actually bridging to another channel. If we are not we
	  actually want to bring the audio back to us. (closes issue
	  #13545) Reported by: davidw ........

2008-12-18 16:36 +0000 [r165541]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Fix reference counts of the class and add an
	  assertion to the end.

2008-12-18 15:25 +0000 [r165502]  Eliel C. Sardanons <eliels@gmail.com>

	* main/strings.c, include/asterisk/strings.h: Remove duplicate code
	  from the ast_str API. We now use __AST_STR_* to access 'struct
	  ast_str' members, but this must only be used inside the API
	  implementation. (closes issue #14098) Reported by: eliel Patches:
	  ast_str.patch uploaded by eliel (license 64)

2008-12-18 14:23 +0000 [r165433-165469]  Russell Bryant <russell@digium.com>

	* apps/app_originate.c: Add a \todo note for app_originate. Jared
	  Smith suggested that we add a way to be able to set variables and
	  functions on the outbound channel. I think that it's a great
	  idea, so I have added it as a todo so that it gets done at some
	  point.

	* apps/app_originate.c (added), CHANGES: Add a new application,
	  Originate. (closes issue #14075) Reported by: rcasas Patches:
	  app_originate.c uploaded by rcasas (license 641), heavily
	  modified by me Tested by: russell Review:
	  http://reviewboard.digium.com/r/95/

2008-12-17 23:39 +0000 [r165397]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_record.c: Add RECORD_STATUS variable, as requested on
	  the -users list. Patch by me (license 14)

2008-12-17 21:46 +0000 [r165326-165330]  Mark Michelson <mmichelson@digium.com>

	* res/res_odbc.c: Fix a refcount leak in res_odbc

	* apps/app_meetme.c, res/res_realtime.c: Fix the build

2008-12-17 21:28 +0000 [r165319-165325]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_macro.c: Oops, broke trunk

	* /, apps/app_macro.c: Merged revisions 165317 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008)
	  | 4 lines Reverse the fix from issue #6176 and add proper
	  handling for that issue. (Closes issue #13962, closes issue
	  #13363) Fixed by myself (license 14) ........

2008-12-17 21:17 +0000 [r165318]  Mark Michelson <mmichelson@digium.com>

	* apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c,
	  apps/app_queue.c, apps/app_voicemail.c: Merged revisions 165255
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec
	  2008) | 7 lines Fix some memory leaks found while looking at how
	  realtime configs are handled. Also cleaned up some coding
	  guidelines violations in app_realtime.c, mostly related to
	  spacing ........

2008-12-17 20:50 +0000 [r165254]  Steve Murphy <murf@digium.com>

	* utils/extconf.c: This patch is here committed to satisfy the
	  buildbot, who has a problem with the const.

2008-12-17 19:55 +0000 [r165219]  Terry Wilson <twilson@digium.com>

	* res/res_phoneprov.c: Polycom phones close the connection after
	  reading a little bit of the firmware files, we should stop
	  sending in that case. Also, make that case print out a debug
	  statement instead of a scary WARNING.

2008-12-17 19:52 +0000 [r165216]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Call proxy_update so that the IP address
	  gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue
	  #14055) Reported by: chris-mac

2008-12-17 18:49 +0000 [r165180]  Matthew Nicholson <mnicholson@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: This patch
	  adds a new 'ignoresdpversion' option to sip.conf. When this is
	  enabled (either globally or for a specific peer), chan_sip will
	  treat any SDP data it receives as new data and update the media
	  stream accordingly. By default, Asterisk will only modify the
	  media stream if the SDP session version received is different
	  from the current SDP session version. This option is required to
	  interoperate with devices that have non-standard SDP session
	  version implementations (observed by toc on the bug tracker with
	  Microsoft OCS which always uses 0 as the session version).
	  http://reviewboard.digium.com/r/94/ (closes issue #13958)
	  Reported by: toc Tested by: toc

2008-12-17 17:56 +0000 [r165145]  Russell Bryant <russell@digium.com>

	* doc/appdocsxml.dtd: argsep is used as an attribute for an
	  argument, as well

2008-12-17 17:53 +0000 [r165142-165143]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: And actually assign the function to a
	  pointer...

	* apps/app_voicemail.c: Use the create_vm_state_from_user function
	  in a place where it was not being used before. Also, I've moved
	  the urgent folder check in messagecount() up a bit so that the
	  flow is a bit better. This was something I noticed while taking a
	  look at issue #13973, although I don't think this is the
	  underlying cause of the issue.

2008-12-17 16:41 +0000 [r165108]  Kevin P. Fleming <kpfleming@digium.com>

	* utils: ignore this copied file

2008-12-17 05:04 +0000 [r165039-165071]  Steve Murphy <murf@digium.com>

	* utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c,
	  utils/conf2ael.c, utils/check_expr.c: A possibly "horrible fix"
	  for a "horribly broken" situation. As stuff shifts around in the
	  asterisk code, the miscellaneous inclusions from the standalone
	  stuff gets broken. There's no easy fix for this situation. I made
	  sure that everything in utils builds without problem ***AND***
	  that aelparse runs the regressions correctly with the following
	  make menuselect options both on and off: DONT_OPTIMIZE
	  DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE
	  DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE
	  I think from now on, I'm going to #undef all these features in
	  the various utils native files; I guess I could do the same for
	  the copied-in files, surrounded by STANDALONE ifdef. A standalone
	  isn't going to care about threads, mutexes, etc.

	* pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions

2008-12-16 23:06 +0000 [r164978]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_sip.c: Merged revisions 164977 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec
	  2008) | 7 lines After looking through SIP registration code most
	  of the day, this is one of the few things I could find that was
	  just plain wrong. Even though it probably isn't possible for it
	  to happen, it seems weird to have code that checks if a pointer
	  is NULL and then immediately dereferences that pointer if it was
	  NULL. ........

2008-12-16 22:57 +0000 [r164976]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, doc/api-1.6.2-changes.txt (added),
	  funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c,
	  CHANGES, configs/extensions.conf.sample: Add timezone to the
	  possible fields in a timespec. (closes issue #14028) Reported by:
	  mostyn Patches: timezone-v2.patch uploaded by mostyn (license
	  398) (with additional code guideline fixes and a memory leak fix
	  by me - license 14)

2008-12-16 22:45 +0000 [r164942]  Jeff Peeler <jpeeler@digium.com>

	* apps/app_record.c: (closes issue #13669) Reported by: pj Delete
	  file recording if recording terminated from a hangup.

2008-12-16 22:31 +0000 [r164941]  Terry Wilson <twilson@digium.com>

	* channels/chan_sip.c: Make a note of the feature request in bug
	  #11157 as per the reporter and oej, and suspend the bug since no
	  one seems to be keen on implementing it any time soon.

2008-12-16 21:39 +0000 [r164821-164882]  Russell Bryant <russell@digium.com>

	* /, main/utils.c: Merged revisions 164881 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008)
	  | 9 lines Fix an issue where DEBUG_THREADS may erroneously report
	  that a thread is exiting while holding a lock. If the last lock
	  attempt was a trylock, and it failed, it will still be in the
	  list of locks so that it can be reported. (closes issue #13219)
	  Reported by: pj ........

	* /, apps/app_macro.c: Merged revisions 164876 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008)
	  | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has
	  been returned. This is a bug I noticed while looking at the code
	  for app_macro. This return code means that another thread has
	  assumed ownership of the channel and it can no longer be touched.
	  (I hate this return code with a passion, by the way.) ........

	* main/asterisk.c: Fix build issues on Linux after sysinfo related
	  changes

2008-12-16 20:47 +0000 [r164809-164814]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify
	  trumps poke per lmadsen.

	* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
	  configuration options for finer control over how Asterisk handles
	  having to poke all peers at seemingly the same time. (closes
	  issue #13217) Reported by: cervajs

2008-12-16 20:41 +0000 [r164807]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 164806 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008)
	  | 9 lines Add "restart gracefully" to the AMI blacklist of CLI
	  commands. "module unload" was already identified as a command
	  that can not be used from the AMI. "restart gracefully"
	  effectively unloads all modules, and will run in to the same
	  problems. (closes issue #13894) Reported by: kernelsensei
	  ........

2008-12-16 20:08 +0000 [r164802]  Michiel van Baak <michiel@vanbaak.info>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/asterisk.c: introduce 'core show sysinfo' for systems that
	  dont have the Linux-ish sysinfo stuff but do have sysctl. (closes
	  issue #13433) Reported by: mvanbaak Patches:
	  2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license
	  7) with two free calls replaced with ast_free based on feedback
	  on reviewboard Review: http://reviewboard.digium.com/r/91/

2008-12-16 20:04 +0000 [r164801]  Steve Murphy <murf@digium.com>

	* main/pbx.c: (closes issue #14076) Reported by: toc Tested by:
	  murf OK, Well this issue has had its share of flip-flopping. I
	  found the following: 1. the code in question, in ext_cmp1 in
	  pbx.c, would not allow two extensions that vary only by any
	  dashes contained within them, to be defined in the same context.
	  2. for input dialstrings, dashes are NOT ignored. So, skipping
	  them when sorting patterns seemed a bit silly. Thus, you might
	  declare ext 891 in a context, but if you try dialing 8-9-1, it
	  will NOT match 891. So, I proposed to remove the code from
	  ext_cmp1 to skip the spaces and dashes. Just kept us from
	  declaring 891 and 8-9-1 in the same context, forcing users to
	  generate otherwise uselessly obfuscated dialplan code to get the
	  same effect. Then, I tried out 1.4, and found that: 1. you can
	  declare 891 and 8-9-1 in the same context! 2. You can't define
	  891, and have 8-9-1 match it! Nor can you define 8-9-1, and have
	  891 match it! So, it appears that my proposal simply restores the
	  pbx to behaving as it did in 1.4.

2008-12-16 19:54 +0000 [r164798]  Tilghman Lesher <tlesher@digium.com>

	* contrib/scripts/safe_asterisk: Set up umask as a possible
	  configuration option. (closes issue #13753) Reported by: irroot

2008-12-16 17:14 +0000 [r164737]  Russell Bryant <russell@digium.com>

	* /, main/threadstorage.c, include/asterisk/threadstorage.h: Merged
	  revisions 164736 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008)
	  | 14 lines Fix memory leak and invalid reporting issues with
	  DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was
	  being used within the context of the thread local data
	  destructors. We would go off and allocate more thread local data
	  while the pthread lib was in the middle of destroying it all.
	  This led to a memory leak. Another issue was an invalid argument
	  being provided to the the object_add API call. (closes issue
	  #13678) Reported by: ys Tested by: Russell ........

2008-12-16 16:50 +0000 [r164733]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: Be more detailed about why the include did not
	  get included. (closes issue #14071) Reported by: kshumard
	  Patches: pbx_config.patch.improvederroroutput.txt uploaded by
	  kshumard (license 92)

2008-12-16 16:00 +0000 [r164675]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 164672 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008)
	  | 11 lines Fix a memory leak related to the use of the "setvar"
	  configuration option. The problem was that these variables were
	  being appended to the list of vars on the sip_pvt every time a
	  re-registration or re-subscription came in. Since it's just a
	  waste of memory to put them there unless the request was an
	  INVITE, then the fix is to check the request type before copying
	  the vars. (closes issue #14037) Reported by: marvinek Tested by:
	  russell ........

2008-12-16 15:44 +0000 [r164659]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When using externhost make sure the port
	  gets set to the bindaddr port if one was not specified in the
	  externhost value itself. (closes issue #13634) Reported by:
	  performer

2008-12-16 15:31 +0000 [r164648]  Steve Murphy <murf@digium.com>

	* main/pbx.c, /: Merged revisions 164634 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5
	  lines I added a sentence to clarify why - and ' ' are ignored in
	  patterns as per bug 14076. Leif says he'll put some stuff about
	  it in the extensions.conf sample, etc. ........

2008-12-16 15:00 +0000 [r164602-164623]  Russell Bryant <russell@digium.com>

	* apps/app_minivm.c: Set MINIVM_ACCMESS_STATUS in all cases. Also,
	  remove a variable that was not needed. (closes issue #14081)
	  Reported by: pkempgen

	* /, res/res_musiconhold.c: Merged revisions 164605 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16
	  Dec 2008) | 5 lines Don't try to change working directory if a
	  directory was not configured. (closes issue #14089) Reported by:
	  caspy ........

	* channels/chan_dahdi.c: Fix usage of the DAHDI_VMWI ioctl. (closes
	  issue #14090) Reported by: alecdavis Patches:
	  chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license
	  585)

2008-12-16 01:52 +0000 [r164565]  Sean Bright <sean.bright@gmail.com>

	* doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also
	  change a bit of minor formatting.

2008-12-15 22:25 +0000 [r164519-164525]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Open a timer before loading configuration
	  so that the trunking configuration option will take effect.
	  (closes issue #14082) Reported by: seandarcy

	* channels/chan_iax2.c: Fix log message to refer to the generic
	  timing interface, not DAHDI specifically (inspired by issue
	  #14082)

	* main/frame.c: Make sure we handle a uint32_t payload in
	  ast_frdup() (closes issue #14080) Reported by: fnordian Patches:
	  frame.patch uploaded by fnordian (license 110)

2008-12-15 21:17 +0000 [r164485]  Tilghman Lesher <tlesher@digium.com>

	* configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow
	  disabling pattern match searches within the Realtime dialplan
	  switch. (closes issue #13698) Reported by: fhackenberger Patches:
	  20081211__bug13698.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: fhackenberger

2008-12-15 20:07 +0000 [r164419-164428]  Mark Michelson <mmichelson@digium.com>

	* apps/app_page.c: Add an 'i' option to app_page. This option works
	  the same as the 'i' options for app_dial and app_queue, in that
	  they will ignore any attempts by phones to forward the call.
	  (closes issue #13977) Reported by: putnopvut Patches:
	  page_ignore_forwards.patch uploaded by putnopvut (license 60)
	  Tested by: putnopvut, acunningham

	* /, include/asterisk/pbx.h: Merged revisions 164422 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon,
	  15 Dec 2008) | 3 lines Add the deadlock note to
	  ast_spawn_extension as well ........

	* /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged
	  revisions 164416 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec
	  2008) | 4 lines Add notes to autoservice and pbx doxygen
	  regarding a potential deadlock scenario so that it is avoided in
	  the future ........

2008-12-15 19:48 +0000 [r164417]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str
	  opacity in chan_sip for now, since something wasn't quite right
	  in the merge.

2008-12-15 19:42 +0000 [r164415]  Steve Murphy <murf@digium.com>

	* include/asterisk/strings.h: I was getting this warning during a
	  compile on a 64-bit machine running ubuntu server 8.10, and
	  gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings
	  being treated as errors In file included from
	  /home/murf/asterisk/trunk/include/asterisk/utils.h:671, from
	  chan_vpb.cc:46:
	  /home/murf/asterisk/trunk/include/asterisk/strings.h: In function
	  ‘char* ast_str_truncate(ast_str*, ssize_t)’:
	  /home/murf/asterisk/trunk/include/asterisk/strings.h:479: error:
	  comparison between signed and unsigned integer expressions
	  make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2
	  which this fix silences

2008-12-15 18:12 +0000 [r164351]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 164350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6
	  lines Do not try to unlock a non-existant channel if the transfer
	  fails. (closes issue #13800) Reported by: dwagner Patches:
	  asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license
	  608) ........

2008-12-15 18:09 +0000 [r164349]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: When querying for the structure of the CDR
	  table, remove the schema, if it exists. (Closes issue #14058)

2008-12-15 17:24 +0000 [r164312]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Use ast_seekstream to return the file stream back to
	  the beginning instead of directly seeking to zero. This is
	  because some audio formats have headers at the front that need to
	  be skipped, which will be done by the format module. (closes
	  issue #14079) Reported by: elguero

2008-12-15 17:21 +0000 [r164272-164309]  Russell Bryant <russell@digium.com>

	* channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a
	  couple more build issues related to ast_str_opaque

	* pbx/pbx_dundi.c: When a reload is issued, always process the
	  configuration for dundi.conf. The reason is that a reload can be
	  used to refresh DNS lookups for defined peers. Even if the config
	  file hasn't changed, we want to process it for that purpose.
	  (closes issue #13776) Reported by: kombjuder

2008-12-15 16:16 +0000 [r164268-164270]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a compile warning and a logic error that
	  could have been bad for non-realtime queues

	* apps/app_queue.c: Fix up a few issues with regards to queues *
	  Fix reference counting used in the __queues_show function * Add
	  code to be sure that the "queue show" command does not print
	  information for a realtime queue which has been deleted from the
	  backend * Add a missing unref to the realtime queue loading
	  function for the case where a queue is in the module's container
	  but has been deleted from the realtime backend (closes issue
	  #14033) Reported by: cristiandimache Patches: 14033.patch
	  uploaded by putnopvut (license 60) Tested by: cristiandimache

2008-12-15 15:41 +0000 [r164208-164257]  Joshua Colp <jcolp@digium.com>

	* configure, include/asterisk/autoconfig.h.in, apps/app_fax.c,
	  configure.ac: Make app_fax compatible with newer versions of
	  spandsp. This remains backwards compatible with earlier versions
	  though so do not fret. (closes issue #14073) Reported by:
	  seandarcy

	* main/utils.c: Update to work with new ast_str changes.

2008-12-15 14:40 +0000 [r164202-164203]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, main/features.c: Merged revisions 164201 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008)
	  | 31 lines Handle a case where a call can be bridged to a channel
	  that is still ringing. The issue that was reported was about a
	  case where a RINGING channel got redirected to an extension to
	  pick up a call from parking. Once the parked call got taken out
	  of parking, it heard silence until the other side answered.
	  Ideally, the caller that was parked would get a ringing
	  indication. This patch fixes this case so that the caller
	  receives ringback once it comes out of parking until the other
	  side answers. The fixes are: - Make sure we remember that a
	  channel was an outgoing channel when doing a masquerade. This
	  prevents an erroneous ast_answer() call on the channel, which
	  causes a bogus 200 OK to be sent in the case of SIP. - Add some
	  additional comments to explain related parts of code. - Update
	  the handling of the ast_channel visible_indication field. Storing
	  values that are not stateful is pointless. Control frames that
	  are events or commands should be ignored. - When a bridge first
	  starts, check to see if the peer channel needs to be given
	  ringing indication because the calling side is still ringing. -
	  Rework ast_indicate_data() a bit for the sake of readability.
	  (closes issue #13747) Reported by: davidw Tested by: russell
	  Review: http://reviewboard.digium.com/r/90/ ........

	* apps/app_jack.c: Fix build WRT ast_str_opaque

2008-12-14 18:16 +0000 [r164168]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/strings.h: Don't pass a negative to an unsigned
	  type and expect things to work correctly.

2008-12-14 15:26 +0000 [r164054-164137]  Sean Bright <sean.bright@gmail.com>

	* doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art.

	* res/snmp/agent.c: Use ast_str_strlen() instead of recalculating
	  the string length.

2008-12-13 13:26 +0000 [r164028]  Michiel van Baak <michiel@vanbaak.info>

	* res/snmp/agent.c: nuke another use of the ast_str internals.

2008-12-13 08:36 +0000 [r163991]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_sqlite3_custom.c, apps/app_meetme.c,
	  funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c,
	  main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h,
	  cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c,
	  res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c,
	  channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c,
	  configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c,
	  main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c,
	  res/res_config_ldap.c, include/asterisk/threadstorage.h,
	  cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c,
	  res/res_config_pgsql.c, main/strings.c (added), main/pbx.c,
	  channels/chan_sip.c, main/Makefile, main/translate.c,
	  include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c,
	  funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c,
	  main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c,
	  utils/hashtest2.c, include/asterisk/strings.h,
	  include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c,
	  apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque
	  branch (discontinue usage of ast_str internals)

2008-12-13 03:03 +0000 [r163951-163952]  Sean Bright <sean.bright@gmail.com>

	* doc/tex/asterisk.tex: This shouldn't have gotten commited. We
	  might want to generate this into a separate file instead of the
	  version controlled one.

	* doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead
	  of ASCII art ones.

2008-12-13 00:59 +0000 [r163912]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c: Only detach and destroy the whisper
	  audiohooks if they are actually in use.

2008-12-12 23:48 +0000 [r163873]  Terry Wilson <twilson@digium.com>

	* apps/app_queue.c: When using realtime queues, app_queue wasn't
	  updating the strategy if it was changed in the realtime backend.
	  This patch resolves the issue for almost all situations. It is
	  currently not supported to switch to the linear strategy via
	  realtime since the ao2_container for members will have been set
	  to have multiple buckets and therefore the members would be
	  unordered. (closes issue #14034) Reported by: cristiandimache
	  Tested by: otherwiseguy, cristiandimache

2008-12-12 23:06 +0000 [r163828]  Russell Bryant <russell@digium.com>

	* res/res_clioriginate.c: Add a note to indicate why this only
	  supports one channel for now.

2008-12-12 22:04 +0000 [r163762]  Tilghman Lesher <tlesher@digium.com>

	* main/editline/read.c, /, main/asterisk.c: Merged revisions 163761
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008)
	  | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk,
	  but also add a pointer inside editline to look back to
	  asterisk.c, so others don't spend as much time as I did looking
	  (in the wrong place) for the appropriate function. Reported by:
	  ZX81, via the #asterisk-users channel Fixed by: me (license 14)
	  ........

2008-12-12 20:12 +0000 [r163716]  Russell Bryant <russell@digium.com>

	* CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel
	  redirect", which is similar in operation to AMI Redirect. Review:
	  http://reviewboard.digium.com/r/89/

2008-12-12 19:16 +0000 [r163675]  Steve Murphy <murf@digium.com>

	* channels/chan_dahdi.c: demote always-appearing debug message (for
	  certain boards) to ast_debug lev 3 msg instead

2008-12-12 18:45 +0000 [r163642-163670]  Russell Bryant <russell@digium.com>

	* main/tcptls.c, channels/chan_sip.c: Rename a number of
	  tcptls_session variables. There are no functional changes here.
	  The name "ser" was used in a lot of places. However, it is a
	  relic from when the struct was a server_instance, not a
	  session_instance. It was renamed since it represents both a
	  server or client connection.

	* channels/chan_sip.c: Fix a small race condition in
	  sip_tcp_locate(). We must increase the reference count on the
	  tcptls_session _before_ unlocking the thread list.

	* channels/chan_sip.c: Resolve crashes when using SIP TCP/TLS with
	  qualify. The problem was a reference count error on the
	  tcptls_session structure. (closes issue #13989) Reported by:
	  Nugget

2008-12-12 18:17 +0000 [r163629]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When a device registers we need to unlink
	  them (if linked) from the peers_by_ip container and link them
	  back in since their IP address has changed. This would have
	  manifested itself if you configured a new device (as type=peer),
	  registered, and then tried to place a call from the device. Since
	  the peer was not linked into the peers_by_ip container it would
	  have never been found. (closes issue #13811) Reported by: pj

2008-12-12 17:22 +0000 [r163582-163612]  Michiel van Baak <michiel@vanbaak.info>

	* res/res_monitor.c: Document default Monitor file location.
	  (closes issue #14065) Reported by: kshumard Patches:
	  res_monitor.documentation.patch.txt uploaded by kshumard (license
	  92)

	* channels/chan_skinny.c: Fix codec capability setup in chan_skinny
	  Behaviour now is that general codec config flows to default_line
	  and default_device. [devices] stuff amends default_device and
	  similar for [lines]. These are copied to individual device and
	  line as they are created. Added confcapability and confprefs for
	  the configured stuff which doesn't change as device and so on are
	  connected. prefs are based on line prefs if they exist, else the
	  device prefs are used (prefs identifies codec order). (closes
	  issue #13806) Reported by: pj Patches: codecs.diff uploaded by
	  wedhorn (license 30) Tested by: pj and me

2008-12-12 16:55 +0000 [r163579]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, channels/chan_sip.c: Since chan_sip is callback
	  devicestate driven do not pass in actual states, pass in unknown
	  so we get asked. Additionally do not pass in an actual device
	  state value in ast_setstate since the channel may be callback
	  driven. (closes issue #13525) Reported by: pj

2008-12-12 15:10 +0000 [r163516]  Doug Bailey <dbailey@digium.com>

	* configs/phoneprov.conf.sample: Add internationalization to sample
	  configuration file

2008-12-12 14:44 +0000 [r163449-163512]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 163511 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008)
	  | 5 lines Specify uint32_t for variables storing a CRC32 so that
	  it is actually 32 bits on 64-bit machines, as well. (inspired by
	  issue #13879) ........

	* main/channel.c, main/autoservice.c, /,
	  include/asterisk/channel.h: Merged revisions 163448 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12
	  Dec 2008) | 26 lines Resolve issues that could cause DTMF to be
	  processed out of order. These changes come from
	  team/russell/issue_12658 1) Change autoservice to put digits on
	  the head of the channel's frame readq instead of the tail. If
	  there were frames on the readq that autoservice had not yet read,
	  the previous code would have resulted in out of order processing.
	  This required a new API call to queue a frame to the head of the
	  queue instead of the tail. 2) Change up the processing of DTMF in
	  ast_read(). Some of the problems were the result of having two
	  sources of pending DTMF frames. There was the dtmfq and the more
	  generic readq. Both were used for pending DTMF in various
	  scenarios. Simplifying things to only use the frame readq avoids
	  some of the problems. 3) Fix a bug where a DTMF END frame could
	  get passed through when it shouldn't have. If code set
	  END_DTMF_ONLY in the middle of digit emulation, and a digit
	  arrived before emulation was complete, digits would get processed
	  out of order. (closes issue #12658) Reported by: dimas Tested by:
	  russell, file Review: http://reviewboard.digium.com/r/85/
	  ........

2008-12-11 23:38 +0000 [r163384]  Tilghman Lesher <tlesher@digium.com>

	* /, main/asterisk.c: Merged revisions 163383 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008)
	  | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on
	  certain shells, the terminal is messed up. By intercepting those
	  events with a signal handler in the remote console, we can avoid
	  those issues. (closes issue #13464) Reported by: tzafrir Patches:
	  20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: blitzrage ........

2008-12-11 22:49 +0000 [r163317]  Matthew Nicholson <mnicholson@digium.com>

	* /, pbx/pbx_dundi.c: Merged revisions 163316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec
	  2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes
	  issue #13819) Reported by: adomjan Patches:
	  pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
	  dundi_clearecache3.diff uploaded by mnicholson (license 96)
	  Tested by: adomjan ........

2008-12-11 21:48 +0000 [r163241-163254]  Russell Bryant <russell@digium.com>

	* /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions
	  163253 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008)
	  | 8 lines Fix some observed slowdowns in dialplan processing. The
	  change is to remove autoservice usage from dialplan functions
	  that do not need it because they do not perform operations that
	  potentially block. (closes issue #13940) Reported by: tbelder
	  ........

	* res/res_timing_pthread.c: Fix a problem where continuous mode
	  will get inadvertently get turned off if set_rate() is used while
	  continuous mode was already turned on. (closes issue #13738)
	  Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix
	  (license 547)

2008-12-11 20:57 +0000 [r163198-163213]  Mark Michelson <mmichelson@digium.com>

	* configs/voicemail.conf.sample, apps/app_voicemail.c: Add an
	  option to voicemail.conf to allow urgent messages to be forwarded
	  as not urgent. (closes issue #14063) Reported by: jaroth Patches:
	  urgfwd_v2.patch uploaded by jaroth (license 50)

	* main/features.c: Add an appropriate goto if ast_call fails

2008-12-11 20:07 +0000 [r163171]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix the "failed" extension for outgoing calls.
	  The conversion to use ast_check_hangup() everywhere instead of
	  checking the softhangup flag directly introduced this problem.
	  The issue is that ast_check_hangup() checked for tech_pvt to be
	  NULL. Unfortunately, this will be NULL is some valid
	  circumstances, such as with a dummy channel. The fix is simple.
	  Don't check tech_pvt. It's pointless, because the code path that
	  sets this to NULL is when the channel hangup callback gets
	  called. This happens inside of ast_hangup(), which is the same
	  function responsible for freeing the channel. Any code calling
	  ast_check_hangup() better not be calling it after that point, and
	  if so, we have a bigger problem at hand. (closes issue #14035)
	  Reported by: erogoza

2008-12-11 20:02 +0000 [r163168]  Tilghman Lesher <tlesher@digium.com>

	* configure, configure.ac: Sometimes even Linux needs -lm to link
	  libtonezone, such as when libtonezone is compiled statically.
	  (closes issue #13887) Reported by: tzafrir

2008-12-11 19:40 +0000 [r163166]  Mark Michelson <mmichelson@digium.com>

	* main/features.c: Reduce indentation level of
	  ast_feature_request_and_dial

2008-12-11 17:06 +0000 [r163094]  Russell Bryant <russell@digium.com>

	* /, main/features.c: Merged revisions 163092 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008)
	  | 11 lines Fix an issue that made it so you could only have a
	  single caller executing a custom feature at a time. This was
	  especially problematic when custom features ran for any
	  appreciable amount of time. The fix turned out to be quite
	  simple. The dynamic features are now stored in a read/write list
	  instead of a list using a mutex. (closes issue #13478) Reported
	  by: neutrino88 Fix suggested by file ........

2008-12-11 16:52 +0000 [r163089]  Tilghman Lesher <tlesher@digium.com>

	* /, res/res_agi.c: Merged revisions 163088 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008)
	  | 6 lines Don't wait forever, if there's a specified recording
	  timeout. (closes issue #13885) Reported by: bamby Patches:
	  res_agi.c.patch uploaded by bamby (license 430) ........

2008-12-11 16:47 +0000 [r163081-163085]  Mark Michelson <mmichelson@digium.com>

	* /, apps/app_queue.c: Merged revisions 163084 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec
	  2008) | 4 lines Revert this cast to long. Using time_t here
	  causes build failures on a FreeBSD 32-bit build. ........

	* /, apps/app_queue.c: Merged revisions 163080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec
	  2008) | 14 lines Fix a potential crash due to unsafe datastore
	  handling. This patch also contains a conversion from using long
	  to time_t for representing times for a queue, as well as some
	  whitespace fixes. (closes issue #14060) Reported by: nivek
	  Patches: datastore_fixup.patch.corrected uploaded by nivek
	  (license 636) with slight modification from me Tested by: nivek
	  ........

2008-12-11 15:40 +0000 [r163037]  Sean Bright <sean.bright@gmail.com>

	* doc/tex/qos.tex: Fix some of the grammar issues in
	  doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard
	  Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92)
	  (Slight modifications by seanbright)

2008-12-11 15:05 +0000 [r162997]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When a device registers to use it is
	  entirely possible that they may be in use, so tell the core that
	  we don't know the devstate and have it ask us for it. (closes
	  issue #13525) Reported by: pj

2008-12-10 23:01 +0000 [r162930]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Previously missing line, now the substitution works
	  correctly

2008-12-10 22:53 +0000 [r162927]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 162926 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10
	  Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check.
	  Pointed out by mmichelson, thanks! ........

2008-12-10 22:48 +0000 [r162923]  Joshua Colp <jcolp@digium.com>

	* res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to
	  changes done to turn it into a single memory allocation we can't
	  just use the existing CLI alias structure. We have to destroy all
	  existing ones and then create new ones. (closes issue #14054)
	  Reported by: pj

2008-12-10 22:48 +0000 [r162922]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: Checking global variables here actually overwrote the
	  previous substitution by channel variables, and in any case, was
	  redundant; pbx_substitute_variables_helper ALREADY does
	  substitution for global variables. (closes issue #13327) Reported
	  by: pj

2008-12-10 22:11 +0000 [r162891]  Jeff Peeler <jpeeler@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 162874 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10
	  Dec 2008) | 5 lines (closes issue #13229) Reported by:
	  clegall_proformatique Ensure that moh_generate does not return
	  prematurely before local_ast_moh_stop is called. Also, the sleep
	  in mp3_spawn now only occurs for http locations since it seems to
	  have been added originally only for failing media streams.
	  ........

2008-12-10 19:02 +0000 [r162739-162805]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 162804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6
	  lines Fix subscription based MWI up a bit. We only want to put
	  sip: at the beginning of the URI if it is not already there and
	  revert code to ignore destination check if subscribing for MWI.
	  (closes issue #12560) Reported by: vsauer Patches: patch001.diff
	  uploaded by ramonpeek (license 266) ........

	* /, channels/chan_sip.c: Merged revisions 162738 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6
	  lines When a SIP peer unregisters set the expiry time back to 0
	  so that the 200 OK contains an expires of 0. (closes issue
	  #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded
	  by hjourdain (license 583) ........

2008-12-10 17:09 +0000 [r162687]  Michiel van Baak <michiel@vanbaak.info>

	* include/asterisk.h, main/asterisk.c, main/cli.c: add tab
	  completion for 'core set debug X filename.c' (closes issue
	  #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel

2008-12-10 16:39 +0000 [r162664-162667]  Mark Michelson <mmichelson@digium.com>

	* doc/tex/misdn.tex, /: Merged revisions 162659 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec
	  2008) | 8 lines Add missing documentation to misdn.txt (closes
	  issue #14052) Reported by: festr Patches: misdn.txt.patch
	  uploaded by festr (license 443) ........

	* /, channels/chan_sip.c: Merged revisions 162663 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec
	  2008) | 11 lines Revert fix for issue 13570. It has caused more
	  problems than it helped to fix. (closes issue #13783) Reported
	  by: navkumar (closes issue #14025) Reported by: ffs ........

2008-12-10 16:11 +0000 [r162619-162660]  Joshua Colp <jcolp@digium.com>

	* res/res_http_post.c: FreeBSD also needs libgen.h (closes issue
	  #14051) Reported by: ys Patches: res_http_post.c.diff uploaded by
	  ys (license 281)

	* /, main/rtp.c: Merged revisions 162653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6
	  lines Increment the sequence number on the end packets for
	  RFC2833. After reading the RFC some more and doing some testing I
	  agree with this change. (closes issue #12983) Reported by: vt
	  Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
	  520) ........

	* channels/chan_sip.c: When transmitting a register set the socket
	  port to the local one for the transport being used, not the port
	  for the remote server. (closes issue #13633) Reported by:
	  performer

2008-12-10 11:34 +0000 [r162583]  Michiel van Baak <michiel@vanbaak.info>

	* res/snmp/agent.c: Make res_snmp.so compile on OpenBSD. OpenBSD
	  uses an old version of gcc which throws an error if you use a
	  macro that's not #defined

2008-12-10 01:09 +0000 [r162542]  Joshua Colp <jcolp@digium.com>

	* doc/janitor-projects.txt, channels/iax2-parser.c,
	  apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and
	  remove it as a janitor project. (closes issue #14032) Reported
	  by: bkruse Patches: 14032.patch uploaded by bkruse (license 132)

2008-12-09 23:41 +0000 [r162488]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/stringfields.h: it does help if the compiler
	  attribute syntax is correct

2008-12-09 23:10 +0000 [r162466]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 162463 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09
	  Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........

2008-12-09 22:38 +0000 [r162414-162418]  Russell Bryant <russell@digium.com>

	* include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen,
	  main/asterisk.c: Add some additional Asterisk project developer
	  documentation. After the nightly update of the documentation on
	  asterisk.org, I'll post an update to asterisk-dev with a pointer
	  to the changes. This covers some release branch and commit policy
	  information. None of this should be a surprise, since it's just
	  documenting what we have already been doing.

	* include/asterisk/utils.h, /, main/utils.c, main/asterisk.c:
	  Merged revisions 162413 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008)
	  | 8 lines Remove the test_for_thread_safety() function
	  completely. The test is not valid. Besides, if we actually
	  suspected that recursive mutexes were not working, we would get a
	  ton of LOG_ERROR messages when DEBUG_THREADS is turned on.
	  (inspired by a discussion on the asterisk-dev list) ........

2008-12-09 21:57 +0000 [r162355]  Tilghman Lesher <tlesher@digium.com>

	* /, apps/app_voicemail.c: Merged revisions 162348 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
	  ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09
	  Dec 2008) | 4 lines We appear to have documented tz= in the
	  [general] section of voicemail.conf, without actually having
	  implemented it. Oops. (Reported by Olivier on the -users list)
	  ........

2008-12-09 21:16 +0000 [r162342]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_directed_pickup.c: Merged revisions 162341 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4
	  lines Add 'down' as a valid state for directed call pickup. This
	  creeps up when we receive session progress when dialing a device
	  and not ringing. (closes issue #14005) Reported by: ddl ........

2008-12-09 20:59 +0000 [r162291]  Russell Bryant <russell@digium.com>

	* /, apps/app_meetme.c: Merged revisions 162286 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008)
	  | 9 lines Fix an issue where callers on an incoming call on an
	  SLA trunk would not hear ringback. We need to make sure that we
	  don't start writing audio to the trunk channel until we're
	  actually ready to answer it. Otherwise, the channel driver will
	  treat it as inband progress, even though all they are getting is
	  silence. (closes issue #12471) Reported by: mthomasslo ........

2008-12-09 20:46 +0000 [r162275]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_festival.c: Merged revisions 162273 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4
	  lines Fix double declaration of 'x' on the PPC platform. (closes
	  issue #14038) Reported by: ffloimair ........

2008-12-09 20:40 +0000 [r162271]  Steve Murphy <murf@digium.com>

	* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162264
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1
	  line In discussion with seanbright on #asterisk-dev, I have added
	  a default rule, and an option to suppress the default rule from
	  being generated in the flex output, for the sake of those OS's
	  where they didn't tweak flex's ECHO macro, and the compiler
	  doesn't like it. The regressions are OK with this. ........

2008-12-09 20:30 +0000 [r162266]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c, /: Merged revisions 162265 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec
	  2008) | 6 lines If we fail to start a thread for the pbx to run
	  in, we need to be sure to decrease the number of active calls on
	  the system. This fix may relate to ABE-1713, but it is not
	  certain yet. ........

2008-12-09 19:48 +0000 [r162197-162205]  Joshua Colp <jcolp@digium.com>

	* /, main/rtp.c: Merged revisions 162204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7
	  lines Make sure that the timestamp for DTMF is not the same as
	  the previous voice frame and do not send audio when transmitting
	  DTMF as this confuses some equipment. (closes issue #13209)
	  Reported by: ip-rob Patches: 13209.diff uploaded by file (license
	  11) Tested by: ip-rob, bujones ........

	* /, main/rtp.c: Merged revisions 162188 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4
	  lines Take video into account when early bridging RTP. (closes
	  issue #13535) Reported by: davidw ........

2008-12-09 18:35 +0000 [r162079-162140]  Steve Murphy <murf@digium.com>

	* /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 162136
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1
	  line Previous fix used ast_malloc and ast_copy_string and messed
	  up the standalone stuff. Fixed. ........

	* res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c,
	  res/ael/ael.flex: Merged revisions 162013 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
	  r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) |
	  45 lines (closes issue #14019) Reported by: ckjohnsonme Patches:
	  14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme,
	  murf This crash was the result of a few small errors that would
	  combine in 64-bit land to result in a crash. 32-bit land might
	  have seen these combine to mysteriously drop the args to an
	  application call, in certain circumstances. Also, in trying to
	  find this bug, I spotted a situation in the flex input, where, in
	  passing back a 'word' to the parser, it would allocate a b