[svn-commits] lmadsen: tag 1.8.0-rc1 r288496 - /tags/1.8.0-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Wed Sep 22 16:17:20 CDT 2010


Author: lmadsen
Date: Wed Sep 22 16:17:16 2010
New Revision: 288496

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=288496
Log:
Importing files for 1.8.0-rc1 release.

Added:
    tags/1.8.0-rc1/.lastclean   (with props)
    tags/1.8.0-rc1/.version   (with props)
    tags/1.8.0-rc1/ChangeLog   (with props)

Added: tags/1.8.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc1/.lastclean?view=auto&rev=288496
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Added: tags/1.8.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc1/ChangeLog?view=auto&rev=288496
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--- tags/1.8.0-rc1/ChangeLog (added)
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+2010-09-22  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-rc1 Released.
+
+2010-09-22 17:49 +0000 [r288345-288418]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
+	  (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
+	  | 5 lines RFC3261 section 12.2 explicitly says out of order
+	  requests are responded with a 500 Server Internal Error response.
+	  ABE-2458 ........ ................
+
+	* /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
+	  (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
+	  Sep 2010) | 2 lines During check_pendings, if the dialog is
+	  terminated with a CANCEL, change the invitestate to INV_CANCEL
+	  like in sip_hangup. ........ ................
+
+2010-09-22 16:45 +0000 [r288341]  Russell Bryant <russell at digium.com>
+
+	* main/asterisk.c, /: Merged revisions 288340 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288340 | russell | 2010-09-22 11:44:13 -0500
+	  (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
+	  | 11 lines Fix a 100% CPU consumption problem when setting
+	  console=yes in asterisk.conf. The handling of -c and console=yes
+	  should be the same, but they were not. When you specify -c, it
+	  sets both a flag for console module and for asterisk not to
+	  fork() off into the background. The handling of console=yes only
+	  set console mode, so you would end up with a background process()
+	  trying to run the Asterisk console and freaking out since it
+	  didn't have anything to read input from. Thanks to beagles for
+	  reporting and helping debug the problem! ........
+	  ................
+
+2010-09-22 15:14 +0000 [r288268]  Tilghman Lesher <tlesher at digium.com>
+
+	* UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
+	  Merged revisions 288267 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
+	  (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
+	  | 9 lines Allow the encoding to be set, in case local charset
+	  does not agree with database. (closes issue #16940) Reported by:
+	  jamicque Patches: 20100827__issue16940.diff.txt uploaded by
+	  tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
+	  uploaded by tilghman (license 14) Tested by: jamicque ........
+	  r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
+	  | 5 lines Document addition of encoding parameter. (issue #16940)
+	  Reported by: jamicque ........ ................
+
+2010-09-22 00:06 +0000 [r288194]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
+	  (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
+	  | 26 lines In chan_iax2.c:schedule_delivery() calls
+	  ast_bridged_channel() on an unlocked channel. Near the beginning
+	  of schedule_delivery(), ast_bridged_channel() is called on
+	  iaxs[fr->callno]->owner. However, the channel is not locked,
+	  which can result in ast_bridged_channel() crashing should
+	  owner->tech change to a technology that doesn't implement
+	  bridged_channel. I also fixed the other calls to
+	  ast_bridged_channel() in chan_iax2.c since the owner lock was not
+	  held there either. Converted the existing channel deadlock
+	  avoidance to use iax2_lock_owner(). Using the new function
+	  simplified some awkward code. In the process of fixing the
+	  locking on ast_bridged_channel(), I also found a memory leak in
+	  socket_process() for v1.6.2 and v1.8. The local struct variable
+	  ies.vars is not freed on early/abnormal function exits. (closes
+	  issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
+	  uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
+	  uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
+	  uploaded by rmudgett (license 664) Review:
+	  https://reviewboard.asterisk.org/r/926/ ........ ................
+
+2010-09-21 22:57 +0000 [r288159]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
+	  (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
+	  | 15 lines Try both the encoded and unencoded subscription URI
+	  for a match in hints. When a phone sends an encoded URI for a
+	  subscription, the URI is not matched with the actual hint that is
+	  in decoded format. For example, if we have an extension with a
+	  hint that is named: "#5601" or "*5601", the subscription will
+	  work fine if the phone subscribes with an already decoded URI,
+	  but when it's decoded like "%255601" or "%2A5601", Asterisk is
+	  unable to match it with the correct hint. (closes issue #17785)
+	  Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
+	  uploaded by tilghman (license 14) Tested by: ramonpeek ........
+	  ................
+
+2010-09-21 22:26 +0000 [r288157]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
+	  21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
+	  issue #18019) Reported by: Netview Patches: issue_0018019.patch
+	  uploaded by pabelanger (license 224) Tested by: Netview ........
+
+2010-09-21 21:03 +0000 [r288079-288082]  Richard Mudgett <rmudgett at digium.com>
+
+	* doc/tex/partymanip.tex: Add note in party manipulation chapter on
+	  interception macros.
+
+	* apps/app_queue.c, apps/app_dial.c: Simplify locking code for
+	  REDIRECTING interception macro when forwarding a call. Simplified
+	  the locking code by using a local copy of the redirecting party
+	  information in app_dial.c:do_forward() and
+	  app_queue.c:wait_for_answer() for launching the REDIRECTING
+	  interception macro when a call is forwarded. Reduced the lock
+	  time of the 'o->chan' and 'in' channels.
+
+	* main/channel.c: Protect channel access in CONNECTED_LINE and
+	  REDIRECTING interception macro launch code.
+
+2010-09-21 19:48 +0000 [r288007]  Brett Bryant <bbryant at digium.com>
+
+	* main/channel.c, /: Merged revisions 288006 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400
+	  (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010)
+	  | 8 lines Add a check to fix a rare segmentation fault you'd get
+	  if ast_frdup couldn't allocate memory on the first frame being
+	  queued in ast_queue_frame. (closes issue #17882) Reported by:
+	  seanbright Tested by: seanbright ........ ................
+
+2010-09-21 19:08 +0000 [r287935]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, /: Merged revisions 287934 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500
+	  (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21
+	  Sep 2010) | 2 lines Less than zero is an error, not any non-zero
+	  value. ........ ................
+
+2010-09-21 19:02 +0000 [r287931]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c: Revert change in favor of a more targeted fix
+
+2010-09-21 18:32 +0000 [r287929]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: Send a "415 Unsupported Media Type" after
+	  failure to process sdp due to unknown Content-Encoding header.
+	  ABE-2258
+
+2010-09-21 15:53 +0000 [r287897]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Cut-n-paste error in builtin_blindtransfer().
+
+2010-09-21 15:43 +0000 [r287895]  Russell Bryant <russell at digium.com>
+
+	* res/res_rtp_asterisk.c, main/dnsmgr.c, channels/chan_sip.c,
+	  main/acl.c: Don't use ast_strdupa() from within the arguments to
+	  a function. (closes issue #17902) Reported by: afried Patches:
+	  issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+	  russell Review: https://reviewboard.asterisk.org/r/927/
+
+2010-09-21 15:24 +0000 [r287893]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Anonymous callerid needs a "sip:" uri
+	  prefix. (closes issue #17981) Reported by: avalentin Patches:
+	  sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
+	  (plus an additional fix by me) Tested by: avalentin
+
+2010-09-21 13:41 +0000 [r287863]  Russell Bryant <russell at digium.com>
+
+	* main/logger.c: Fix a regression in verbose logger processing.
+
+2010-09-21 04:37 +0000 [r287833]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c: Don't generate connected line buffer twice for
+	  comparison
+
+2010-09-21 00:00 +0000 [r287760]  Brett Bryant <bbryant at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 287759 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400
+	  (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010)
+	  | 16 lines Fix misvalidation of meetme pins in conjunction with
+	  the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a
+	  user and admin pin setup for your conference, using the user pin
+	  would gain you admin priviledges. Also, when no user pin was set,
+	  an admin pin was, the 'a' MeetMe flag wasn't used, and the user
+	  tried to enter a conference then they were still prompted for a
+	  pin and forced to hit #. (closes issue #17908) Reported by: kuj
+	  Patches: pins_2.patch uploaded by kuj (license 1111) Tested by:
+	  kuj Review: [full review board URL with trailing slash] ........
+	  ................
+
+2010-09-20 23:51 +0000 [r287757]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c: Avoid infinite loop with certain local channel
+	  connected line updates Compare connected line data before sending
+	  a connected line indication to avoid possible loops. Review:
+	  https://reviewboard.asterisk.org/r/932/
+
+2010-09-20 23:20 +0000 [r287701]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/channel.c, /: Merged revisions 287685 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep
+	  2010) | 18 lines ast_channel_masquerade: Avoid recursive
+	  masquerades. Check all 4 combinations of (original/clonechan) *
+	  (masq/masqr). Initially original->masq and clonechan->masqr were
+	  only checked. It's possible with multiple masq's planned - and
+	  not yet executed, that the 'original' chan could already have
+	  another masq'd into it - thus original->masqr would be set, that
+	  masqr would lost. Likewise for the clonechan->masq. (closes issue
+	  #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches:
+	  based on bug16057.diff4.txt uploaded by alecdavis (license 585)
+	  Tested by: ramonpeek, davidw, alecdavis ........
+
+2010-09-20 23:14 +0000 [r287683]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: The inalarm flag was not set in sig_analog
+	  struct if the port is initially in alarm. Fixed initial inalarm
+	  value for sig_analog ports. Along with -r261007, this gets the
+	  inalarm flag in sync with chan_dahdi for sig_analog ports.
+	  (closes issue #16983)
+
+2010-09-20 22:21 +0000 [r287661]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/channel.c: ast_do_masquerade. Keep channels ao2_container
+	  locked while unlink and linking channels. Previously, Masquerade
+	  would unlock 'original' and 'clonechan' and allow another masq
+	  thread to run. End result would be corrupted memory, and the
+	  frequent report 'Bad Magic Number'. (closes issue #17801,#17710)
+	  Reported by: notthematrix Patches: Based on bug17801.diff1.txt
+	  uploaded by alecdavis (license 585) Tested by: alecdavis Review:
+	  https://reviewboard.asterisk.org/r/928
+
+2010-09-20 22:09 +0000 [r287645-287647]  David Vossel <dvossel at digium.com>
+
+	* include/asterisk/channel.h, CHANGES, include/asterisk/framehook.h
+	  (added), main/channel.c, main/framehook.c (added),
+	  funcs/func_frame_trace.c (added): Addition of the FrameHook API
+	  (AKA AwesomeHooks) So far all our tools for viewing and
+	  manipulating media streams within Asterisk have been entirely
+	  focused on audio. That made sense then, but is not scalable now.
+	  The FrameHook API lets us tap into and manipulate _ANY_ type of
+	  media or signaling passed on a channel present today or in the
+	  future. This tool is a step in the direction of expanding
+	  Asterisk's boundaries and will help generate some rather
+	  interesting applications in the future. In addition to the
+	  FrameHook API, a simple dialplan function exercising the api has
+	  been included as well. This function is called FRAME_TRACE().
+	  FRAME_TRACE() allows for the internal ast_frames read and written
+	  to a channel to be output. Filters can be placed on this function
+	  to debug only certain types of frames. This function could be
+	  thought of as an internal way of doing ast_frame packet captures.
+	  Review: https://reviewboard.asterisk.org/r/925/
+
+	* channels/chan_sip.c: Fixes issue with registrations not working
+	  properly with pedantic=yes. (closes issue #18017) Reported by:
+	  schmidts Patches: issues_18017_v1.diff uploaded by dvossel
+	  (license 671) Tested by: schmidts
+
+2010-09-20 21:29 +0000 [r287643]  Jason Parker <jparker at digium.com>
+
+	* /, channels/chan_skinny.c: Merged revisions 287642 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep
+	  2010) | 8 lines Don't crash when parking a non-bridged call.
+	  (closes issue #17680) Reported by: jmhunter Patches:
+	  chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by:
+	  jmhunter, DEA ........
+
+2010-09-20 21:19 +0000 [r287639]  Brett Bryant <bbryant at digium.com>
+
+	* main/logger.c: Fixes an error with the logger that caused verbose
+	  messages to be spammed to the screen if syslog was configured in
+	  logger.conf (closes issue #17974) Reported by: lmadsen Review:
+	  https://reviewboard.asterisk.org/r/915/
+
+2010-09-20 15:57 +0000 [r287559]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 287558 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500
+	  (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint
+	  state changes Merged revisions 287555 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep
+	  2010) | 5 lines Use ast_dynamic_str when processing hint state
+	  changes (related to issue #17928) Reported by: mdu113 ........
+	  ................
+
+2010-09-19 16:09 +0000 [r287471]  Olle Johansson <oej at edvina.net>
+
+	* main/manager.c, /: Merged revisions 287470 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön,
+	  19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7
+	  lines Make sure we always free variables properly in manager
+	  originate. (closes issue #17891) reported, solved and tested by
+	  oej Review: https://reviewboard.asterisk.org/r/869/ ........
+	  ................
+
+2010-09-17 21:08 +0000 [r287388]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 287387 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500
+	  (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010)
+	  | 7 lines Blank columns should get set on reload, not ignored.
+	  (closes issue #16893) Reported by: haakon Patches:
+	  20100818__issue16893.diff.txt uploaded by tilghman (license 14)
+	  ........ ................
+
+2010-09-17 13:37 +0000 [r287309]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 287308 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500
+	  (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep
+	  2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while
+	  processing in ast_hint_state_changed(). (related to issue #17928)
+	  Reported by: mdu113 ........ ................
+
+2010-09-17 08:44 +0000 [r287269-287271]  Jan Kalab <pitlicek at gmail.com>
+
+	* res/res_calendar_ews.c: Events are visible after they were
+	  removed from EWS calendar Because we must merge calendar even
+	  when it's empty. (closes issue #17786)
+
+	* res/res_calendar_ews.c: Asterisk crashing because of double free
+	  when EWS request fails The free is done later in code. I think
+	  ast_free() should have built in checks for double free. (closes
+	  issue #17782)
+
+	* res/res_calendar_caldav.c, res/res_calendar_ews.c,
+	  res/res_calendar_exchange.c, res/res_calendar_icalendar.c:
+	  Support for HTTP redirects in calendar's URL libneon does not
+	  support HTTP redirects (3xx responses) by default. You must tell
+	  it to follow them. Also, another little unsigned int fix. (closes
+	  issue #17776) Review: https://reviewboard.asterisk.org/r/921/
+
+2010-09-16 22:04 +0000 [r287195]  Jason Parker <jparker at digium.com>
+
+	* contrib/init.d/rc.debian.asterisk: Don't fail when running the
+	  Debian init script directly (as one would normally do). readlink
+	  apparently returns 1 when the arg isn't a symlink, which caused
+	  the script to exit. (closes issue #17910) Reported by: wurstsalat
+
+2010-09-16 21:57 +0000 [r287193]  Russell Bryant <russell at digium.com>
+
+	* UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Set
+	  the default for "autofill" and "shared_lastcall" to "yes" in
+	  queues.conf. Review: https://reviewboard.asterisk.org/r/922/
+
+2010-09-16 20:07 +0000 [r287116-287120]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c, /: Merged revisions 287119 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500
+	  (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep
+	  2010) | 8 lines Don't limit hint processing in
+	  ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
+	  (closes issue #17928) Reported by: mdu113 Patches:
+	  20100831__issue17928.diff.txt uploaded by tilghman (license 14)
+	  Tested by: mdu113 ........ ................
+
+	* main/cdr.c, /: Merged revisions 287115 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500
+	  (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep
+	  2010) | 8 lines Don't stop printing cdr variables if we encounter
+	  one with a blank name or value. (closes issue #17900) Reported
+	  by: under Patches: core-show-channel-cdr-fix1.diff uploaded by
+	  mnicholson (license 96) Tested by: mnicholson ........
+	  ................
+
+2010-09-15 22:17 +0000 [r287056]  Terry Wilson <twilson at digium.com>
+
+	* res/res_srtp.c: Don't hang up a call on an SRTP unprotect failure
+	  Also make it more obvious when there is an issue en/decrypting.
+	  (closes issue #17563) Reported by: Alexcr Patches:
+	  res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by:
+	  twilson
+
+2010-09-15 20:58 +0000 [r287020]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/features.c: fix uninintialized variable
+
+2010-09-15 20:53 +0000 [r287017]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_msg_parser.c, channels/chan_misdn.c: Merged
+	  revision 287014 from
+	  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+	  .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed,
+	  15 Sep 2010) | 58 lines The handling of call transfer signaling
+	  for mISDN PTMP is not fully implemented. The handling of call
+	  transfer signaling for mISDN PTMP is not fully implemented. The
+	  signaling of number updates with ISDN/DSS1 ECT supplementary
+	  services (ETS 300 369-1) comes along with a notification
+	  indicator IE and redirection number IE for PTMP. The
+	  implementation in the current Asterisk mISDN channel
+	  unfortunately can handle these information elements only in a
+	  NOTIFY message. These information elements are also signaled in a
+	  FACILTY message with a RequestSubaddress facility, when the
+	  subscriber is already in the active state (see 9.2.4 and 9.2.5 of
+	  ETS 300 369-1). ********** abe_2526_ast.patch * Added support to
+	  handle the notification indicator IE and redirection number IE
+	  with the RequestSubaddress facility. * Made
+	  misdn_update_connected_line() send a NOTIFY message if Asterisk
+	  originated the call and it is not connected yet. * Made
+	  misdn_update_connected_line() send a FACILITY message if the call
+	  is already connected. This patch requires the presence of the
+	  associated mISDN patches to compile. I had to enhance mISDN to
+	  allow the notification indicator IE and the redirection number IE
+	  to be used with a FACILITY message. Earlier versions of the
+	  Digium enhanced mISDN are no longer going to work. **********
+	  abe_2526_misdn.patch * Made an incoming FACILITY message allow
+	  the presence of the notification indicator IE and the redirection
+	  number IE. ********** abe_2526_misdnuser_v3.patch * Added support
+	  to send and receive a FACILITY message with the notification
+	  indicator IE and the redirection number IE. * Added the ability
+	  to send a NOTIFY message in PTMP/NT mode to all responding
+	  subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches:
+	  abe_2526_ast.patch uploaded by rmudgett (license 664)
+	  abe_2526_misdn.patch uploaded by rmudgett (license 664)
+	  abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
+	  Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526
+	  ..........
+
+2010-09-15 20:32 +0000 [r286931-287015]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 286998 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500
+	  (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010)
+	  | 7 lines Ensure mailbox is not filled to capacity before doing
+	  message forwarding. Specifically, before prompting to record a
+	  prepended message the capacity is checked first. If the mailbox
+	  is full the extension will be reprompted. ABE-2517 ........
+	  ................
+
+	* CHANGES, channels/chan_iax2.c, channels/sip/include/sip.h,
+	  configs/features.conf.sample, channels/chan_mgcp.c,
+	  include/asterisk/features.h, channels/chan_dahdi.c,
+	  channels/sig_analog.c, channels/chan_sip.c, main/features.c: Add
+	  parking extension for non-default parking lots. This is a new
+	  feature that allows for parking to custom parking lots to be
+	  accessed directly, rather than with channel variables or by
+	  changing the default parking lot. The extension is set with the
+	  parkext option just as the default parking lot is done. Also, the
+	  manager action has been updated to optionally allow a specified
+	  parking lot. (closes issue #14882) Reported by: vmikhnevych
+	  Patches: patch_14882.txt uploaded by mnick (license 874) modified
+	  by me Review: https://reviewboard.asterisk.org/r/884/
+
+2010-09-15 18:29 +0000 [r286904-286905]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_analog.c: Simplify some code in sig_analog.
+
+	* channels/sig_analog.c: Unable to originate calls using E&M over
+	  T1. When originating a call from Unit Under Test to Reference
+	  Unit using E&M RBS signaling mode, I get the following warning
+	  message: "Ring/Off-hook in strange state 3 on channel 1". Fixed
+	  the sig_analog outgoing flag. It was never set when sig_analog
+	  was extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408
+
+2010-09-15 13:05 +0000 [r286868]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Set tohost to the domain specified in the
+	  configuration file instead of the IP address of the host we are
+	  calling. This fixes a regression introduced in r274783. (closes
+	  issue #17960) Reported by: adriavidal Patches:
+	  sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested
+	  by: mich, mnicholson, adriavidal (closes issue #17676) Reported
+	  by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson
+	  (license 96) Tested by: mnicholson
+
+2010-09-14 21:57 +0000 [r286834]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: Sets subscribed type for outgoing MWI
+	  subscriptions so correct Event header is used.
+
+2010-09-14 19:28 +0000 [r286682-286758]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 286757 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500
+	  (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep
+	  2010) | 13 lines Don't clear the username from a realtime
+	  database when a registration expires. Non-realtime chan_sip does
+	  not clear the username from memory when a registration expiries
+	  so realtime probably shouldn't either. (closes issue #17551)
+	  Reported by: ricardolandim Patches:
+	  reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license
+	  96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson
+	  (license 96) reg-expiry-username-1.8-fix1.diff uploaded by
+	  mnicholson (license 96) reg-expiry-username-trunk-fix1.diff
+	  uploaded by mnicholson (license 96) Tested by: ricardolandim,
+	  mnicholson ........ ................
+
+	* main/channel.c, /: Merged revisions 286681 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500
+	  (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep
+	  2010) | 7 lines Only drop duplicate answer frames if the channel
+	  is bridged. Back in r3710 ast_read() was modified to drop answer
+	  frames on channels that were in the UP state. This modification
+	  prevented bridges that were up before the answer from being
+	  broken and reestablished by an ANSWER control frame. That change
+	  also prevents pickup of channels called from the ast_dial
+	  framework from working properly. The ast_dial framework expects
+	  to see an ANSWER frame after dialing and the pickup code queues
+	  one but ast_read() drops it. This new change only drops ANSWER
+	  frames when the channel is bridged, allowing the answer queued by
+	  the pickup code to properly pass through ast_read() on to the
+	  ast_dial framework. ABE-2473 (related to issue #2342) ........
+	  ................
+
+2010-09-14 15:30 +0000 [r286647]  Richard Mudgett <rmudgett at digium.com>
+
+	* doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Corrected
+	  documented CONNECTED_LINE and REDIRECTING party manipulation
+	  macro names.
+
+2010-09-14 06:55 +0000 [r286617]  Jan Kalab <pitlicek at gmail.com>
+
+	* res/res_calendar_ews.c: Merging events for Exchange web service
+	  doesn't work as expected, resulting in only one event in calendar
+	  The solution is to use "global" counter of events, since we do
+	  new requests for every event and calendar sync after every
+	  request. So now we do sync only after last request. (closes issue
+	  #17877) Review: https://reviewboard.asterisk.org/r/916/
+
+2010-09-14 05:07 +0000 [r286528-286588]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/realtime/mysql/voicemail_data.sql (added), /,
+	  contrib/realtime/mysql/voicemail_messages.sql (added): Merged
+	  revisions 286587 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010)
+	  | 2 lines Add documentation on missing backend tables for
+	  Voicemail ........
+
+	* /, main/features.c: Merged revisions 286557 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010)
+	  | 2 lines C precedence got me ........
+
+	* /, main/features.c: Merged revisions 286527 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010)
+	  | 2 lines Refactor conversion to ast_poll() to fix callparking
+	  regression. ........
+
+2010-09-13 19:40 +0000 [r286457]  Jason Parker <jparker at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 286456 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) |
+	  5 lines Remove "Internal IP" from sip show settings, as it's not
+	  at all useful to display. (closes issue #17840) Reported by: oej
+	  ........
+
+2010-09-13 15:52 +0000 [r286426]  Richard Mudgett <rmudgett at digium.com>
+
+	* configs/chan_dahdi.conf.sample: Update chan_dahdi.conf.sample to
+	  reflect new libpri T309 default value.
+
+2010-09-11 17:09 +0000 [r286270]  Olle Johansson <oej at edvina.net>
+
+	* /, main/file.c: Merged revisions 286268 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör,
+	  11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4
+	  lines Handle error response when we can't make file compatible
+	  Review: https://reviewboard.asterisk.org/r/911/ ........
+	  ................
+
+2010-09-10 22:04 +0000 [r286189]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/channel.h, include/asterisk/pbx.h,
+	  include/asterisk/frame.h, channels/chan_local.c,
+	  funcs/func_channel.c: Merged revisions 286115 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286115 | twilson | 2010-09-10 15:35:25 -0500
+	  (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010)
+	  | 16 lines Inherit CHANNEL() writes to both sides of a Local
+	  channel Having Local (/n) channels as queue members and setting
+	  the language in the extension with Set(CHANNEL(language)=fr) sets
+	  the language on the Local/...,2 channel. Hold time report
+	  playbacks happen on the Local/...,1 channel and therefor do not
+	  play in the specified language. This patch modifies
+	  func_channel_write to call the setoption callback and pass the
+	  CHANNEL() write info to the callback. chan_local uses this
+	  information to look up the other side of the channel and apply
+	  the same changes to it. (closes issue #17673) Reported by:
+	  Guggemand Review: https://reviewboard.asterisk.org/r/903/
+	  ........ ................
+
+2010-09-10 21:11 +0000 [r286120]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 286117 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400
+	  (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep
+	  2010) | 4 lines Load iax.conf before registering any
+	  functions/applications/actions. Review:
+	  https://reviewboard.asterisk.org/r/914/ ........ ................
+
+2010-09-10 20:55 +0000 [r286118]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, /: Merged revisions 286116 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500
+	  (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010)
+	  | 11 lines An outgoing call may not get hung up if a pre-connect
+	  incoming ISDN call is disconnected. If the ISDN link a
+	  pre-connect incoming call is using fails or is reset, the
+	  outgoing leg may not hang up or be delayed in hanging up.
+	  (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER,
+	  PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
+	  PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the
+	  incoming call leg hangs up before connecting for any reason. It
+	  makes no sense to send a BUSY or CONGESTION control frame to the
+	  outgoing call leg under these circumstances. ........
+	  ................
+
+2010-09-10 20:31 +0000 [r286112]  Russell Bryant <russell at digium.com>
+
+	* main/db.c: Rate limit calls to fsync() to 1 per second after
+	  astdb updates. Astdb was determined to be one of the most
+	  significant bottlenecks in SIP registration processing. This
+	  patch improved the speed of an astdb load test by 50000% (yes,
+	  Fifty-Thousand Percent). On this particular load test setup, this
+	  doubled the number of SIP registrations the server could handle.
+	  Review: https://reviewboard.asterisk.org/r/825/
+
+2010-09-10 18:31 +0000 [r286025]  Tilghman Lesher <tlesher at digium.com>
+
+	* /: Merged revisions 286024 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r286024 | tilghman | 2010-09-10 13:30:21 -0500
+	  (Fri, 10 Sep 2010) | 9 lines Merged revisions 286023 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10
+	  Sep 2010) | 2 lines Missing newline ........ ................
+
+2010-09-10 13:13 +0000 [r285992]  David Ruggles <thedavidfactor at gmail.com>
+
+	* doc/externalivr.txt, CHANGES: Added missing documentation for
+	  ExternalIVR feature added in January 2010
+
+2010-09-10 05:32 +0000 [r285931-285962]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/select.h, /: Merged revisions 285961 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010)
+	  | 6 lines Another fix for Mac OS X. While trying to fix this the
+	  "right" way, I wandered into dependency hell. Two hours later, I
+	  backed out, and just removed the offending code. ast_inline_api

[... 23842 lines stripped ...]



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