[svn-commits] lmadsen: tag 1.6.2.17-rc1 r301943 - /tags/1.6.2.17-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Fri Jan 14 14:59:56 CST 2011


Author: lmadsen
Date: Fri Jan 14 14:59:51 2011
New Revision: 301943

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=301943
Log:
Importing files for 1.6.2.17-rc1 release.

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    tags/1.6.2.17-rc1/.version   (with props)
    tags/1.6.2.17-rc1/ChangeLog   (with props)

Added: tags/1.6.2.17-rc1/.lastclean
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Added: tags/1.6.2.17-rc1/ChangeLog
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--- tags/1.6.2.17-rc1/ChangeLog (added)
+++ tags/1.6.2.17-rc1/ChangeLog Fri Jan 14 14:59:51 2011
@@ -1,0 +1,29078 @@
+2011-01-14  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.17-rc1 Released.
+
+2011-01-14 20:03 +0000 [r301842-301848]  lathama <lathama at localhost>:
+
+	* funcs/func_base64.c, funcs/func_aes.c: Add relationships to
+	  function documentation. Fix amatuer type mistake
+
+	* funcs/func_base64.c, funcs/func_aes.c: Add relationships to
+	  function documentation.
+
+2011-01-13 17:01 +0000 [r301730]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/phoneprov.conf.sample: Add static entry for split Polycom
+	  332 firmware. (closes issue #18607) Reported by: cjacobsen
+	  Patches: polycom_331.diff uploaded by cjacobsen (license 1029)
+	  Tested by: lathama
+
+2011-01-12 21:05 +0000 [r301682]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Don't reject all SUBSCRIBE auth requests
+	  When merging another SUBSCRIBE fix from 1.4, some braces were put
+	  in the wrong place. This patch fixes that. (closes issue #18597)
+	  Reported by: thsgmbh
+
+2011-01-12 18:50 +0000 [r301594]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/manager.c, /: Removed a usleep(1) that shouldn't be
+	  necessary in session_do, and removed the ms_t member from the
+	  mansession_session structure. Merged revisions 301591 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan
+	  2011) | 5 lines Don't store the thread id for the manager session
+	  in the structure we pass to the thread for the manager session.
+	  ABE-2543 ........
+
+2011-01-12 18:11 +0000 [r301503]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, /: Merged revisions 301502 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011)
+	  | 12 lines Fix CPU spike when pressing DTMF after agent login.
+	  The problem here is that DTMF was being continuously deferred and
+	  requeued since ast_safe_sleep is called in a loop. There are
+	  serveral other places in the code that sleeps and then loops in a
+	  similar fashion. Because of this fact I opted to not defer DTMF
+	  any more, which will not affect the original fix:
+	  https://reviewboard.asterisk.org/r/674 (closes issue #18130)
+	  Reported by: rgj ........
+
+2011-01-11 19:14 +0000 [r301310]  Paul Belanger <pabelanger at digium.com>
+
+	* configs/extensions.conf.sample: Fix a logic issue when passing
+	  context ARG
+
+2011-01-11 18:42 +0000 [r301307]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, main/utils.c: Merged revisions 301305 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan
+	  2011) | 4 lines Prevent buffer overflows in ast_uri_encode()
+	  ABE-2705 ........
+
+2011-01-09 21:38 +0000 [r301176-301220]  Paul Belanger <pabelanger at digium.com>
+
+	* autoconf/ast_ext_lib.m4, configure, configure.ac: SOUND_CACHE_DIR
+	  now defaults to empty Sounds files included in the Asterisk
+	  tarball were being ignored and re-downloaded. Users wanting to
+	  cache the files can still override the setting using the
+	  --with-sounds-cache option. (closes issue #18589) Reported by:
+	  pabelanger Patches: issue18589.patch uploaded by pabelanger
+	  (license 224) Tested by: pabelanger Review:
+	  https://reviewboard.asterisk.org/r/1074/
+
+	* apps/app_verbose.c: Indicate log level argument for Log() is not
+	  optional (closes issue #18586) Reported by: kshumard Patches:
+	  app_verbose.c.patch uploaded by kshumard (license 92)
+
+2011-01-07 20:52 +0000 [r301089]  Jason Parker <jparker at digium.com>
+
+	* apps/app_meetme.c: Initialize useropts/adminopts in case there is
+	  no column in the realtime DB. (closes issue #18182) Reported by:
+	  dimas Patches: v1-18182.patch uploaded by dimas (license 88)
+	  Tested by: dimas
+
+2011-01-07 19:57 +0000 [r300951-301046]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c: Fix regression causing forwarding
+	  voicemails to not work with file storage. I had actually already
+	  fixed this in 295200 in 1.4 and thought it wasn't missing in the
+	  other branches for some reason. (closes issue #18358) Reported
+	  by: cabal95
+
+	* apps/app_voicemail.c, /: Merged revisions 300918 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07
+	  Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played
+	  at the correct time. Specifically in the case of timing out but
+	  not leaving voicemail nothing should be heard. And when leaving
+	  voicemail it should be heard. ABE-2647 ........
+
+2011-01-05 18:54 +0000 [r300622]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* res/res_odbc.c, /: Merged revisions 300621 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011)
+	  | 10 lines Use the sanity check in place of the
+	  disconnect/connect cycle. The disconnect/connect cycle has the
+	  potential to cause random crashes. (closes issue #18243) Reported
+	  by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147)
+	  Tested by: ks3 ........
+
+2011-01-05 16:28 +0000 [r300574]  Paul Belanger <pabelanger at digium.com>
+
+	* cdr/cdr_sqlite.c: Change deprecated message to LOG_WARNING Also
+	  removed latter part of message Discussed on #asterisk-dev
+
+2011-01-04 21:52 +0000 [r300431-300520]  Leif Madsen <lmadsen at digium.com>
+
+	* channels/chan_iax2.c, main/xmldoc.c, channels/chan_sip.c,
+	  channels/chan_agent.c: Fix backwards and broken XML
+	  documentation. (closes issue #18547) Reported by: jcovert
+	  Patches: xmldoc.c.patch uploaded by jcovert (license 551)
+	  chan_iax2.c.doc.patch uploaded by jcovert (license 551)
+	  chan_sip.c.patch uploaded by jcovert (license 551)
+	  chan_agent.c.patch uploaded by jcovert (license 551)
+
+	* configs/users.conf.sample: Add some documentation to
+	  users.conf.sample. (closes issue #18531) Reported by: lathama
+	  Patches: users.conf.sample2.diff uploaded by lathama (license
+	  1028) Tested by: lathama
+
+2011-01-04 20:59 +0000 [r300429]  Russell Bryant <russell at digium.com>
+
+	* contrib/scripts/autosupport, /, contrib/scripts/autosupport.8:
+	  Merged revisions 300428 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011)
+	  | 4 lines Update the autosupport script from Digium support.
+	  (closes AST-395) ........
+
+2011-01-04 17:37 +0000 [r300298]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 300216 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011)
+	  | 15 lines Don't authenticate SUBSCRIBE re-transmissions This
+	  only skips authentication on retransmissions that are already
+	  authenticated. A similar method is already used for INVITES. This
+	  is the kind of thing we end up having to do when we don't have a
+	  transaction layer... (closes issue #18075) Reported by: mdu113
+	  Patches: diff.txt uploaded by twilson (license 396) Tested by:
+	  twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/
+	  ........
+
+2011-01-03 23:02 +0000 [r300165]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.c: Use correct variable for atxfercallbackretries
+	  config option. * Misc formatting changes.
+
+2010-12-28 18:51 +0000 [r299864]  Paul Belanger <pabelanger at digium.com>
+
+	* apps/app_chanspy.c: Documentation typo
+
+2010-12-25 10:05 +0000 [r299625]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* channels/chan_local.c, /: Merged revisions 299624 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25
+	  Dec 2010) | 5 lines Move check for extension existence below
+	  variable inheritance, due to the possible use of an eswitch.
+	  (closes issue #16228) Reported by: jlaguilar ........
+
+2010-12-23 03:02 +0000 [r299530-299533]  Moises Silva <moises.silva at gmail.com>
+
+	* channels/chan_dahdi.c: do not use progress which is for PRI and
+	  SS7, add mfcr2_progress member
+
+	* channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after
+	  AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue
+	  #18438) Reported by: mariner7 Tested by: moy
+
+2010-12-22 20:03 +0000 [r299448]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* pbx/ael/ael-test/ref.ael-test19,
+	  pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c,
+	  pbx/ael/ael-test/ref.ael-vtest25,
+	  pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-test3:
+	  Resolve warnings by disambiguating the "s" extension as used by
+	  chan_dahdi from the "s" extension as used by the AEL macros.
+	  (closes issue #18480) Reported by: nivek Patches:
+	  20101215__issue18480__2.diff.txt uploaded by tilghman (license
+	  14) Tested by: nivek
+
+2010-12-20 21:25 +0000 [r299242]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 299194,299198,299220 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec
+	  2010) | 6 lines Respond as soon as possible with a 202 Accepted
+	  to refer requests. This change also plugs a few memory leaks that
+	  can occur when parking sip calls. ABE-2656 ........ r299198 |
+	  mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2
+	  lines Remove changes to via processing that were not supposed to
+	  go into the last commit. ........ r299220 | mnicholson |
+	  2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use
+	  ast_free() instead of free() ABE-2656 ........
+
+2010-12-20 18:16 +0000 [r299130-299136]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* sample.call: Documentation fix
+
+	* cdr/cdr_pgsql.c: If a call was not answered, then the billsec was
+	  calculated unusually large. Also, due to a copy and paste error,
+	  a request for the answer field would have given the start value,
+	  instead. (closes issue #18460) Reported by: joscas Patches:
+	  20101215__issue18460.diff.txt uploaded by tilghman (license 14)
+	  Tested by: joscas
+
+2010-12-20 16:18 +0000 [r299087]  Leif Madsen <lmadsen at digium.com>
+
+	* main/features.c: Note that Park() timeout is milliseconds.
+	  (closes issue #15758) Reported by: mmurdock Tested by: mmurdock,
+	  seanbright
+
+2010-12-20 09:13 +0000 [r299003]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* channels/chan_sip.c: Typos: recieved => received
+
+2010-12-18 00:08 +0000 [r298817-298962]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* main/say.c: Remove backtrace used for testing merge process
+
+	* main/astobj2.c, utils/conf2ael.c, include/asterisk/logger.h,
+	  configure, build_tools/menuselect-deps.in, main/logger.c,
+	  utils/ael_main.c, utils/hashtest2.c, makeopts.in,
+	  utils/check_expr.c, utils/refcounter.c, include/asterisk/utils.h,
+	  build_tools/cflags-devmode.xml, /, main/Makefile,
+	  include/asterisk/autoconfig.h.in, main/say.c, configure.ac,
+	  utils/hashtest.c, main/utils.c: Merged revisions 298905 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010)
+	  | 6 lines Let Asterisk find better backtrace information with
+	  libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will
+	  use libbfd to search for better symbol information within both
+	  the Asterisk binary, as well as loaded modules, to assist when
+	  using inline backtraces to track down problems. ........
+
+	* configure, configure.ac: Also include PTHREAD_LIBS and
+	  PTHREAD_CFLAGS for SQLite 3, as it's needed on some platforms.
+	  (closes issue #18493) Reported by: pprindeville Patches:
+	  asterisk-1.8-sqlite3.patch uploaded by pprindeville (license 347)
+	  Tested by: pprindeville
+
+2010-12-16 23:30 +0000 [r298597-298684]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 298683 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16
+	  Dec 2010) | 2 lines After recording only silence for a voicemail
+	  prepending, restore backup files. ........
+
+	* apps/app_queue.c, /: Merged revisions 298596 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010)
+	  | 7 lines Fix improper hangup when doing an attended transfer to
+	  queue. Had to indicate ringing in wait_for_answer so the attended
+	  transfer code would not try and hang up the local channel it
+	  created, which would kill the call. ABE-2624 ........
+
+2010-12-16 09:04 +0000 [r298393-298481]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* res/res_config_odbc.c, /: Merged revisions 298480 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16
+	  Dec 2010) | 14 lines Only increment the pointer once per loop,
+	  otherwise we corrupt the value. (closes issue #18251) Reported
+	  by: bcnit Patches: 20101110__issue18251.diff.txt uploaded by
+	  tilghman (license 14) Tested by: trev, jthurman, elguero (closes
+	  issue #18279) Reported by: zerohalo Patches:
+	  20101109__issue18279.diff.txt uploaded by tilghman (license 14)
+	  Tested by: zerohalo ........
+
+	* funcs/func_dialgroup.c: Eliminate duplicates from container.
+	  (closes issue #18091) Reported by: bunny Patches:
+	  20101006__issue18091.diff.txt uploaded by tilghman (license 14)
+	  Tested by: bunny
+
+	* /, cdr/cdr_sqlite.c: Merged revisions 298392 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010)
+	  | 8 lines Unregister before shutting down the connection, to
+	  avoid a race. (closes issue #18481) Reported by: pabelanger
+	  Patches: 20101215__issue18481.diff.txt uploaded by tilghman
+	  (license 14) Tested by: pabelanger ........
+
+2010-12-15 21:31 +0000 [r298346]  Sean Bright <sean at malleable.com>
+
+	* main/astobj2.c, /: Merged revisions 298345 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r298345 | seanbright | 2010-12-15 16:28:29 -0500 (Wed, 15 Dec
+	  2010) | 6 lines Fix reference and container leaks when running
+	  'astobj2 test.' We need to make sure that ao2_iterator_destroy is
+	  called once for each time that ao2_iterator_init is called. Also
+	  make sure to unref a newly allocated object that we've linked
+	  into a container. ........
+
+2010-12-13 17:04 +0000 [r298194]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 298193 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13
+	  Dec 2010) | 19 lines Outgoing PRI/BRI calls cannot do DTMF
+	  triggered transfers. Outgoing PRI/BRI calls cannot do DTMF
+	  triggered transfers if a PROCEEDING message is not received. The
+	  debug output shows that the DTMF begin event is seen, but the
+	  DTMF end event is missing. When the DTMF begin happens, the call
+	  is muted so we now have one way audio (until a DTMF end event is
+	  somehow seen). * Made set the proceeding flag when the
+	  PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin
+	  and DTMF end events if we are overlap dialing and have not seen a
+	  PROCEEDING message. * Added a debug message when absorbing a DTMF
+	  event. JIRA SWP-2690 JIRA ABE-2697 ........
+
+2011-01-12  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.16 Released.
+
+2011-01-12  Leif Madsen <lmadsen at digium.com>
+
+	* Merge in changes for configure script to resolve issue for
+	  Debian package builders.
+
+	  ------------------------------------------------------------------------
+	  r301220 | pabelanger | 2011-01-09 15:38:25 -0600 (Sun, 09 Jan 2011)
+	  | 14 lines
+
+	  SOUND_CACHE_DIR now defaults to empty
+
+	  Sounds files included in the Asterisk tarball were being ignored and
+	  re-downloaded. Users wanting to cache the files can still override
+	  the setting
+	  using the --with-sounds-cache option.
+
+	  (closes issue 0018589)
+	  Reported by: pabelanger
+	  Patches:
+	        issue18589.patch uploaded by pabelanger (license 224)
+	      Tested by: pabelanger
+
+	      Review: https://reviewboard.asterisk.org/r/1074/ [^]
+
+	  ------------------------------------------------------------------------
+
+2010-12-13  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.16-rc1 Released.
+
+2010-12-10 16:24 +0000 [r298050]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/netsock.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: Portability issue on OpenSolaris. Also detect the
+	  required structure element, because OpenSolaris defines
+	  SIOCGIFHWADDR, but without support for IP sockets. (closes issue
+	  #18442) Reported by: ranjtech Patches:
+	  20101209__issue18442.diff.txt uploaded by tilghman (license 14)
+	  Tested by: ranjtech
+
+2010-12-09 22:10 +0000 [r297960]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 297959 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010)
+	  | 14 lines Ignore spurious REGISTER requests If a REGISTER
+	  request with a Call-ID matching an existing transaction is
+	  received it was possible that the REGISTER request would
+	  overwrite the initreq of the private structure. This info is used
+	  to generate messages for other responses in the transaction. This
+	  patch ignores REGISTER requests that match non-REGISTER
+	  transactions. (closes issue #18051) Reported by: eeman Tested by:
+	  twilson Review: https://reviewboard.asterisk.org/r/1050/ ........
+
+2010-12-08 18:04 +0000 [r297908]  Tilghman Lesher <tlesher at digium.com>
+
+	* configs/extensions.conf.sample: Use inheritance to get correct
+	  results for SIPFROMDOMAIN. (from an internal Digium discussion)
+
+2010-12-07 22:58 +0000 [r297824]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, /: Merged revisions 297823 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010)
+	  | 12 lines Revert code that changed SSRC for DTMF. Some previous
+	  behavior was attempted to be restored, but mistakingly I did not
+	  realize that the previous behavior was incorrect. This fixes DTMF
+	  not being detected since DTMF shouldn't cause the SSRC to change.
+	  (related to issue #17404) (closes issue #18189) (closes issue
+	  #18352) Reported by: marcbou Tested by: cmbaker82 ........
+
+2010-12-07 22:40 +0000 [r297713-297819]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/init.d/org.asterisk.muted.plist (added), Makefile,
+	  utils/muted.c, /: Merged revisions 297818 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010)
+	  | 4 lines Use non-deprecated APIs for CoreAudio Review:
+	  https://reviewboard.asterisk.org/r/1040/ ........
+
+	* apps/app_followme.c, /: Merged revisions 297689 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010)
+	  | 8 lines Don't create a Local channel if the target extension
+	  does not exist. (closes issue #18126) Reported by: junky Patches:
+	  followme.diff uploaded by junky (license 177) (partially
+	  restructured by me to avoid a possible memory leak) ........
+
+2010-12-06 22:03 +0000 [r297605]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 297603 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010)
+	  | 12 lines Improve handling of REGISTER requests with multiple
+	  contact headers. The changes here attempt to more strictly follow
+	  RFC 3261 section 10.3. Basically the following will now cause a
+	  400 Bad Response to be returned, if: - multiple Contact headers
+	  are present with one set to expire all bindings ("*") - wildcard
+	  parameter is specified for Contact without Expires header or
+	  Expires header is not set to zero. ABE-2442 ABE-2443 ........
+
+2010-12-03 17:40 +0000 [r297534]  Sean Bright <sean at malleable.com>
+
+	* channels/chan_console.c: The CLI command should not contain
+	  <placeholder>s, these are for descriptions.
+
+2010-12-02 20:06 +0000 [r297405]  Paul Belanger <pabelanger at digium.com>
+
+	* Makefile, /: Merged revisions 297404 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec
+	  2010) | 7 lines Resolve compile error under FreeBSD We now set
+	  _ASTCFLAGS+=-march=i686 for i386 processors, still allowing
+	  ASTCFLAGS to override the setting. Review:
+	  https://reviewboard.asterisk.org/r/1043/ ........
+
+2010-12-02 18:07 +0000 [r297311]  Terry Wilson <twilson at digium.com>
+
+	* /, main/abstract_jb.c: Merged revisions 297310 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010)
+	  | 12 lines Initialize offset for adaptive jitter buffer When the
+	  adaptive jitter buffer is enabled in sip.conf, the first frame
+	  placed in the jitter buffer fails with something like:
+	  jb_warning_output: Resyncing the jb. last_delay 0, this delay
+	  -215886466, threshold 1000, new offset 215886466 This happens
+	  because the offset is not initialized before calling jb_put().
+	  This patch modifies jb_put_first_adaptive() to set the offset to
+	  the frame's timestamp. Review:
+	  https://reviewboard.asterisk.org/r/1041/ ........
+
+2010-12-02 13:16 +0000 [r297229]  Russell Bryant <russell at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 297228 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010)
+	  | 6 lines Add "DAHDI" to a couple of app_meetme error messages.
+	  This is in response to some questions on IRC. To the user, there
+	  was nothing that made it obvious that this error had anything to
+	  do with DAHDI not being loaded. ........
+
+2010-12-02 08:55 +0000 [r297186]  Olle Johansson <oej at edvina.net>
+
+	* /, channels/chan_sip.c: Merged revisions 297185 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5
+	  lines If we get a NOTIFY from a non-existing subscription we
+	  should answer with 481, not bad event. If we answer 481 the
+	  subscription that we don't want will be cancelled. ........
+
+2010-12-01 17:52 +0000 [r297073]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 297072 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010)
+	  | 23 lines Fix not stopping MOH when transfered local channel
+	  queue member is answered. The problem here is only present when
+	  local channels are used with the MOH passthru option as well as
+	  no optimization (/nm). I will describe the slightly bizarre
+	  scenario that was used to test, where phones B and C are queue
+	  members: Phone A dials into a queue with two members using local
+	  channels and the above options. Phone B answers. Phone A blind
+	  transfers phone B into the same queue. Phone A hangs up. Phone C
+	  answers, but phone B didn't stop playing MOH. In this scenario,
+	  the unhold frame that should have gotten to phone B never arrived
+	  due to the masquerade from the blind transfer. This is usually
+	  fine since app_queue manages the starting and stopping of MOH.
+	  However, with the passthrough option enabled when app_queue
+	  attempts to stop MOH it tries to do so on the local channel
+	  rather than the real channel. The easiest solution was to just
+	  make sure to send an unhold frame during the transfer since it
+	  wouldn't make sense to have MOH playing after a transfer anyway.
+	  This only modifies SIP transfers, but the other transfers did not
+	  seem to be a problem. If DTMF based transfers were a problem it
+	  might be okay to add ast_moh_stop to finishup, but I didn't want
+	  to have to add that unless required. ABE-2624 ........
+
+2010-12-01 17:01 +0000 [r296950-296991]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/frame.h, /: Merged revisions 296990 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01
+	  Dec 2010) | 5 lines Clarify documentation on how we store codec
+	  preference lists. (closes issue #18397) Reported by: birgita
+	  ........
+
+	* channels/chan_iax2.c: Missed initializations caused startup
+	  errors on Mac OS X (and possibly others, too).
+
+2010-12-01 00:24 +0000 [r296869]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 296868 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30
+	  Nov 2010) | 4 lines Properly restore backup information file when
+	  hanging up during message prepending. ABE-2654 ........
+
+2010-11-29 22:54 +0000 [r296671]  Paul Belanger <pabelanger at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 296670 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon,
+	  29 Nov 2010) | 5 lines Make sure nothing else is needed before
+	  destroying the scheduler. (closes issue #18398) Reported by:
+	  pabelanger ........
+
+2010-11-29 07:27 +0000 [r296533]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: I love standards. There are so many to choose from.
+	  Except when there isn't one. Linux and *BSD disagree on the
+	  elements within the ucred structure. Detect which one is in use
+	  on the system. (closes issue #18384) Reported by: bjm Patches:
+	  cred-diffs uploaded by bjm (license 473)
+	  20101127__issue18384__1.6.2.diff.txt uploaded by tilghman
+	  (license 14) 20101127__issue18384__1.8.diff.txt uploaded by
+	  tilghman (license 14) Tested by: tilghman, bjm
+
+2010-11-27 10:39 +0000 [r296466]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_meetme.c: 18 characters is too short for most date/times
+	  (20 is the usual, but we add more in case of greater precision).
+	  (closes issue #18369) Reported by: tnakonz
+
+2010-11-26 12:23 +0000 [r296351]  Olle Johansson <oej at edvina.net>
+
+	* /, main/say.c: Merged revisions 296309 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11
+	  lines Fix bugs in saying numbers using the Swedish language
+	  syntax (closes issue #18355) Reported by: oej Patch by: oej Much
+	  help from Peter Lindahl. Testing by the ClearIT team during a
+	  coffee break. Review: https://reviewboard.asterisk.org/r/1033/
+	  ........
+
+2010-11-24 23:28 +0000 [r296221]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 296213 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010)
+	  | 6 lines Make Asterisk less crashy. Since we might not put a new
+	  translation path on the channel, go ahead and set it to NULL
+	  right after destroying the old one to ensure we don't try to free
+	  an invalid translation path later on. ........
+
+2010-11-24 22:42 +0000 [r296166]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 296165 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24
+	  Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port
+	  after FXS port gets a CallWaiting pip. The FXS connected phone
+	  has to have CW/CID support to fail, as it will send back a DTMF
+	  'A' or 'D' when it's ready to receive CallerID. A normal phone
+	  with no CID never fails. Also the SIP phone does not hear MOH
+	  when the CW call is answered. The DTMF end frame is suppressed
+	  when the phone acknowledges the CW signal for CID. The problem is
+	  the DTMF begin frame needs to be suppressed as well. The DTMF
+	  begin frame is causing SIP to start sending the DTMF RTP frames.
+	  Since the DTMF end frame is suppressed, SIP will not stop sending
+	  those DTMF RTP packets. * Suppress the DTMF begin and end frames
+	  when the channel driver is looking for DTMF digits. * Fixed a
+	  couple issues caused by not cleaning up the CID spill if you
+	  answer the CW call while it is sending the CID spill. * Fixed not
+	  sending CW/CID spill to the phone when the call is natively
+	  bridged. (Fixed by not using native bridge if CW/CID is
+	  possible.) * Suppress received audio when sending CW/CID spills.
+	  The other parties involved do not need to hear the CW/CID spills
+	  and may be confused if the CW call is for them. (closes issue
+	  #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch
+	  uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+	  NOTE: * v1.4 does not have the main problem fixed by suppressing
+	  the DTMF start frames. The other three items fixed are relevant.
+	  * If you really must restore native bridging between analog
+	  ports, you need to disable CW/CID either by configuring
+	  chan_dahdi.conf callwaitingcallerid=no or dialing *70 before
+	  dialing the number to temporarily disable CW. ........
+
+2010-11-24 20:23 +0000 [r296001-296083]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 296082 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010)
+	  | 12 lines Fix false reporting of an error by set_format(). In
+	  the case that the native format was able to be changed to match
+	  the new requested format, the code proceeded to attempt to build
+	  a translation path, anyway. The result would be NULL, since no
+	  translation path is necessary and resulted in this function
+	  thinking an error has occurred. This case is now specifically
+	  caught and no attempt to build a translation path is attempted.
+	  Thanks to our automated tests and bamboo.asterisk.org for
+	  catching this problem and making a whole lot of noise when things
+	  started failing. :-) ........
+
+	* apps/app_dial.c, main/channel.c, /: Merged revisions 296000 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010)
+	  | 38 lines Handle failures building translation paths more
+	  effectively. The problem scenario occurred on a heavily loaded
+	  system that was using the codec_dahdi module and exceeded the
+	  hardware transcoding capacity. The failure mode at that point was
+	  not good. The report came in to us as an Asterisk lock-up. The
+	  "core show locks" shows a ton of threads locked up (but no
+	  obvious deadlock). Upon deeper investigation, when the system is
+	  in this state, the CPU was maxed out. The CPU was being consumed
+	  by the Asterisk logger spewing messages on every audio frame for
+	  calls set up after transcoder capacity was reached. The purpose
+	  of this patch is to make Asterisk handle failures to create a
+	  translation path in a more graceful manner. If we can't
+	  translate, then the call just needs to be dropped, as it's not
+	  going to work. These are the changes: 1) In set_format() of
+	  channel.c (which is called by set_read_format() and
+	  set_write_format()), it was ignoring if
+	  ast_translator_build_path() failed and returned NULL. It now pays
+	  attention to that case and returns a result reflecting failure.
+	  With this change in place, the bridging code will immediately
+	  detect a failure and end the bridge instead of proceeding to try
+	  to bridge frames that can't be translated and making channel
+	  drivers freak out by sending them frames in a format they weren't
+	  expecting. 2) In ast_indicate_data() of channel.c, failure of
+	  ast_playtones_start() was ignored. It is now reflected in the
+	  return value of the function. This didn't turn out to have any
+	  affect on the bug, but seemed like a good change to leave in. 3)
+	  In app_dial(), when only sending a call to a single endpoint, it
+	  will attempt to do some bridging of its own of early audio. It
+	  uses make_compatible() when it's going to do this. However, it
+	  ignored failure from make compatible. So, even with the fix from
+	  #1, if there was early audio going through app_dial, there would
+	  still be a period of invalid frames passing through. After
+	  detecting failure here, Dial() exits. ABE-2658 ........
+
+2010-11-23 09:36 +0000 [r295907]  Olle Johansson <oej at edvina.net>
+
+	* /, main/say.c: Merged revisions 295906 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8
+	  lines Fix support of saynumber(1,n) in the Swedish language
+	  (closes issue #18353) Reported by: oej Review:
+	  https://reviewboard.asterisk.org/r/1031/ ........
+
+2010-11-22 20:02 +0000 [r295868]  Sean Bright <sean at malleable.com>
+
+	* configs/chan_dahdi.conf.sample: Change some documentation to
+	  suggest dahdi_monitor instead of ztmonitor.
+
+2010-11-22 19:28 +0000 [r295843]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/frame.h, main/channel.c, main/pbx.c, /,
+	  apps/app_macro.c, include/asterisk/channel.h: Merged revisions
+	  295790 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010)
+	  | 46 lines The channel redirect function (CLI or AMI) hangs up
+	  the call instead of redirecting the call. To recreate the
+	  problem: 1) Party A calls Party B 2) Invoke CLI "channel
+	  redirect" command to redirect channel call leg associated with A.
+	  3) All associated channels are hung up. Note that if the CLI
+	  command were done on the channel call leg associated with B it
+	  works. This regression was a result of the fix for issue #16946
+	  (https://reviewboard.asterisk.org/r/740/). The regression affects
+	  all features that use an async goto to execute the dialplan
+	  because of an external event: Channel redirect, AMI redirect, SIP
+	  REFER, and FAX detection. The struct ast_channel._softhangup code
+	  is a mess. The variable is used for several purposes that do not
+	  necessarily result in the call being hung up. I have added
+	  doxygen comments to describe how the various _softhangup bits are
+	  used. I have corrected all the places where the variable was
+	  tested in a non-bit oriented manner. The primary fix is the new
+	  AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so
+	  the soft hangup requests that do not normally result in a hangup
+	  do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171)
+	  Reported by: SantaFox (closes issue #18185) Reported by:
+	  kwemheuer (closes issue #18211) Reported by: zahir_koradia
+	  (closes issue #18230) Reported by: vmarrone (closes issue #18299)
+	  Reported by: mbrevda (closes issue #18322) Reported by: nerbos
+	  Review: https://reviewboard.asterisk.org/r/1013/ ........
+
+2010-11-20 00:45 +0000 [r295710]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/event.h, main/event.c: Fix cache of device state
+	  changes for multiple servers. This patch addresses a regression
+	  where device states across multiple servers were not being
+	  processing completely correctly. The code works to determine the
+	  overall state by looking at the last known state of a device on
+	  each server. However, there was a regression due to some invasive
+	  rewrites of how the cache works that led to the cache only
+	  storing the last device state change for a device, regardless of
+	  which server it was on. The code is set up to cache device state
+	  change events by ensuring that each event in the cache has a
+	  unique device name + entity ID (server ID). The code that was
+	  responsible for comparing raw information elements (which EID is)
+	  always returned a match due to a memcmp() with a length of 0.
+	  There isn't much code to fix the actual bug. This patch also
+	  introduces a new CLI command that was very useful for debugging
+	  this problem. The command allows you to dump the contents of the
+	  event cache. (closes issue #18284) Reported by: klaus3000
+	  Patches: issue18284.rev1.txt uploaded by russell (license 2)
+	  Tested by: russell, klaus3000 (closes issue #18280) Reported by:
+	  klaus3000 Review: https://reviewboard.asterisk.org/r/1012/
+
+2010-11-19 21:55 +0000 [r295672]  Terry Wilson <twilson at digium.com>
+

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