[asterisk-commits] lmadsen: tag 1.4.30 r244867 - in /tags/1.4.30: .lastclean .version ChangeLog

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Feb 4 19:50:05 CST 2010


Author: lmadsen
Date: Thu Feb  4 19:50:01 2010
New Revision: 244867

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=244867
Log:
Importing files for 1.4.30 release.

Added:
    tags/1.4.30/.lastclean   (with props)
    tags/1.4.30/.version   (with props)
    tags/1.4.30/ChangeLog   (with props)

Added: tags/1.4.30/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.30/.lastclean?view=auto&rev=244867
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--- tags/1.4.30/ChangeLog (added)
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+2010-02-04  Leif Madsen
+
+	* Release Asterisk 1.4.30-rc1
+
+2010-02-04 23:20 +0000 [r244785]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_local.c: Change channel state on local channels for
+	  busy,answer,ring. Previously local channels channel state never
+	  changed. This became problematic when the state of the other side
+	  of the local channel was lost, for example during a masquerade.
+	  Changing the state of the local channel allows for the scenario
+	  to be detected when the channel state is set to ringing, but the
+	  peer isn't ringing. The specific problem scenario is described in
+	  164201. Although this was noted on one of the issues, here is the
+	  tested dialplan verified to work: exten =>
+	  9700,1,Dial(Local/*9700 at default&Local/#9700 at default) exten =>
+	  *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+	  exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+	  *9700,n,Dial(SIP/5001) exten => #9700,1,Wait(1) ;1 works, 3 did
+	  not exten =>
+	  #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+	  issue #14992) Reported by: davidw
+
+2010-02-01 23:13 +0000 [r244070-244242]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Backup and restore original textfile, for
+	  prosthesis (gerund of prepend). Also, fix menuselect such that
+	  changing voicemail build options correctly causes rebuild.
+	  (closes issue #16415) Reported by: tomo1657 Patches:
+	  prepention.patch uploaded by tomo1657 (license 484) (with
+	  modifications by me to backport to 1.4)
+
+	* res/res_features.c: When a transferer hangs up during an attended
+	  transfer BEFORE the transfer is answered, don't stop playing MOH.
+	  (closes issue #16513) Reported by: litnimax Patches:
+	  atxfer_moh_16513.patch uploaded by gknispel proformatique
+	  (license 261) Tested by: litnimax
+
+	* main/channel.c, channels/chan_local.c: Revert previous chan_local
+	  fix (r236981) and fix instead by destroying expired frames in the
+	  queue. (closes issue #16525) Reported by: kobaz Patches:
+	  20100126__issue16525.diff.txt uploaded by tilghman (license 14)
+	  20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
+	  (license 14) Tested by: kobaz, atis (closes issue #16581)
+	  Reported by: ZX81 (closes issue #16681) Reported by: alexr1
+
+2010-01-28 18:48 +0000 [r243862-243863]  Leif Madsen <lmadsen at digium.com>
+
+	* BUGS: Oops, correct wrong link (https vs. http) in previous
+	  commit.
+
+	* BUGS: Update location of bug tracker in documentation.
+
+2010-01-28 15:03 +0000 [r243779]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Fix a bogus third argument to
+	  ast_copy_string().
+
+2010-01-27 20:35 +0000 [r243570-243691]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_queue.c: Revert 243570, I should have looked at this
+	  closer. Will reopen the issue, but am leaving the review closed
+	  as the change was pointless. (issue #16488)
+
+	* apps/app_queue.c: Extend announcement URL used with Queue from 80
+	  chars to PATH_MAX. (closes issue #16488) Reported by: syspert
+	  Patches: soundfilelen.pacth-2 uploaded by syspert (license 938)
+	  Review: https://reviewboard.asterisk.org/r/475/
+
+2010-01-27 18:06 +0000 [r243486]  Mark Michelson <mmichelson at digium.com>
+
+	* main/pbx.c: Use a safe list traversal while checking for
+	  duplicate vars in pbx_builtin_setvar_helper.
+
+2010-01-26 23:55 +0000 [r243390]  David Vossel <dvossel at digium.com>
+
+	* res/res_features.c: fixes bug with channel receiving wrong
+	  privileges after call parking (closes issue #16429) Reported by:
+	  Yasuhiro Konishi Patches: features.c.diff uploaded by Yasuhiro
+	  Konishi (license 947) Tested by: dvossel
+
+2010-01-26 18:19 +0000 [r243258]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c: Remove unnecessary code in ast_read as issue
+	  16058 has been fully solved now.
+
+2010-01-25 21:50 +0000 [r242852-242969]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/Makefile, pbx/Makefile: Err, and use the new menuselect
+	  define, too.
+
+	* build_tools/cflags.xml, build_tools/menuselect-deps.in,
+	  configure, configure.ac: Only rebuild parsers by an option in
+	  menuselect
+
+	* configure, main/Makefile, configure.ac, pbx/Makefile: Restore
+	  FreeBSD to able-to-compile-ish-mode
+
+2010-01-25 20:08 +0000 [r242850-242851]  Olle Johansson <oej at edvina.net>
+
+	* main/manager.c: Remove debugging that indeed should have been
+	  gone before commit. Sorry.
+
+	* main/manager.c: Report error when writing to functions returns
+	  error in AMI setvar action
+
+2010-01-25 05:42 +0000 [r242520-242728]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/Makefile, pbx/Makefile: Buildbot pointed out an error
+	  (thanks, buildbot!)
+
+	* main/Makefile, pbx/Makefile: Oops, should have used CMD_PREFIX,
+	  not ECHO_PREFIX, for the commands.
+
+	* main/Makefile: Make the build of the Asterisk expression parser
+	  match that of the AEL parser.
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  pbx/ael/ael_lex.c, pbx/Makefile, makeopts.in: Only rebuild bison
+	  and flex source files on demand, if bison and flex are detected
+	  by the configure script. Changed after discussion on the -dev
+	  list about possible unnecessary build failures, due to
+	  checkouts/untars causing these special source files to possibly
+	  be newer than their resulting C files. This should additionally
+	  ensure that nobody need learn about extra Makefile arguments to
+	  ensure the proper files get rebuilt when changes are made to
+	  these special source files.
+
+2010-01-22 21:44 +0000 [r242423]  Tilghman Lesher <tlesher at digium.com>
+
+	* pbx/Makefile: Rebuild from flex, bison sources when necessary.
+	  (issue #14629) Reported by: Marquis Patches:
+	  20100121__issue14629.diff.txt uploaded by tilghman (license 14)
+
+2010-01-22 09:19 +0000 [r242226]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Initialize notify_types to NULL
+
+2010-01-22 01:48 +0000 [r242142]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/cdr.c: Add Dialed Number Identifier (DNID) field to cdr.
+	  Branch support, retains ABI, if backend CDR collector is adaptive
+	  then database requires 'dnid' field to be added, otherwise no
+	  functional changes. Reported by: alecdavis Tested by: alecdavis
+	  Patch cdr_dnid.diff2.txt uploaded by alecdavis (license 585)
+	  Review: https://reviewboard.asterisk.org/r/455/
+
+2010-01-21 15:25 +0000 [r241932]  Sean Bright <sean at malleable.com>
+
+	* configure, configure.ac: Fix configure check for
+	  PTHREAD_ONCE_INIT when manually adding -Wall to CFLAGS. (closes
+	  issue #16666) Reported by: romain_proformatique
+
+2010-01-21 05:53 +0000 [r241765]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_math.c: Guard against division by zero.
+
+2010-01-20 20:00 +0000 [r241626]  David Vossel <dvossel at digium.com>
+
+	* Makefile: fixes parsing error in Makefile. Some echo lines were
+	  missing "; . Thanks to jparker for pointing out the problem.
+
+2010-01-20 14:12 +0000 [r241543-241544]  Sean Bright <sean at malleable.com>
+
+	* pbx/pbx_spool.c: Modify fix for issue 16554 to be more inline
+	  with what is already in trunk. I should have taken a closer look
+	  at trunk/1.6.x, as this bug has already been fixed in a much more
+	  simple manner, by just settings o->vars to NULL after the
+	  ast_pbx_outgoing_* calls. (issue #16554) Reported by: mav3rick
+
+	* pbx/pbx_spool.c: Fix a memory leak in pbx_spool when using SetVar
+	  in a call file. In pbx_spool, when we are freeing our 'outgoing'
+	  struct, we weren't deallocating the ast_variable list we had
+	  built from SetVars in a call file. Adding a call to
+	  ast_variables_destroy in our deallocation routine works, but only
+	  if the variables have not already been passed into
+	  ast_pbx_outgoing_app() or _exten(), both of which take care of
+	  destroying the variable list for us. (closes issue #16554)
+	  Reported by: mav3rick Patches: issue16554_20100119.patch uploaded
+	  by seanbright (license 71) Tested by: mav3rick
+
+2010-01-20 09:38 +0000 [r241458]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/pbx.c: Update CDR variables as pbx starts Allows CDR
+	  variables added in cdr.c:set_one_cid to become visable during the
+	  call, by executing ast_cdr_update() early in __ast_pbx_run. Based
+	  on cdr_update.diff3.txt (issue #16638) Reported by: alecdavis
+	  Patches: cdr_update.diff3.txt uploaded by alecdavis (license 585)
+	  Tested by: alecdavis
+
+2010-01-19 17:41 +0000 [r241228]  Jason Parker <jparker at digium.com>
+
+	* Makefile: Allow parallel make (-j) to work properly. 1.4 changes
+	  are quite different from the others. (issue #16489) Reported by:
+	  Chainsaw Tested by: qwell
+
+2010-01-19 17:22 +0000 [r241227]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_agent.c: Fix deadlock in agent_read by removing
+	  call to agent_logoff. One must always lock the agents list lock
+	  before the agent private. agent_read locks the private
+	  immediately, so locking the agents list lock is not an option
+	  (which is what agent_logoff requires). Because agent_read already
+	  has access to the agent private all that is necessary is to do
+	  the required hanging up that agent_logoff performed. (closes
+	  issue #16321) Reported by: valon24 Patches: bug16321.patch
+	  uploaded by jpeeler (license 325)
+
+2010-01-18 19:54 +0000 [r241015]  Sean Bright <sean at malleable.com>
+
+	* main/config.c: Plug a memory leak when reading configs with their
+	  comments. While reading through configuration files with the
+	  intent of returning their full contents (comments specifically)
+	  we allocated some memory and then forgot to free it. This doesn't
+	  fix 16554 but clears up a leak I had in the lab. (issue #16554)
+	  Reported by: mav3rick Patches: issue16554_20100118.patch uploaded
+	  by seanbright (license 71) Tested by: seanbright
+
+2010-01-18 16:51 +0000 [r240891]  David Vossel <dvossel at digium.com>
+
+	* Makefile: updated transmit_silence option documentation in
+	  asterisk.conf This patch updates the transmit_silence option to
+	  better document why the option exists, and what it affects.
+	  Thanks to russell for providing the verbage for this update.
+
+2010-01-18 13:27 +0000 [r240768]  Olle Johansson <oej at edvina.net>
+
+	* utils/Makefile: Fix muted compilation in 1.4 only
+
+2010-01-15 23:06 +0000 [r240547]  Russell Bryant <russell at digium.com>
+
+	* Makefile: Fix a spelling error in the asterisk.conf sample.
+
+2010-01-15 20:52 +0000 [r240414]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Disallow leaving more than maxmsg
+	  voicemails. This is a possibility because our previous method
+	  assumed that no messages are left in parallel, which is not a
+	  safe assumption. Due to the vmu structure duplication, it was
+	  necessary to track in-process messages via a separate structure.
+	  If at some point, we switch vmu to an ao2-reference-counted
+	  structure, which would eliminate the prior noted duplication of
+	  structures, then we could incorporate this new in-process
+	  structure directly into vmu. (closes issue #16271) Reported by:
+	  sohosys Patches: 20100108__issue16271.diff.txt uploaded by
+	  tilghman (license 14) 20100108__issue16271__trunk.diff.txt
+	  uploaded by tilghman (license 14)
+	  20100108__issue16271__1.6.0.diff.txt uploaded by tilghman
+	  (license 14) Tested by: jsutton
+
+2010-01-14  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.29
+
+2010-01-08  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.29-rc1
+
+2010-01-07 20:14 +0000 [r238409-238411]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: fixes crash in "scheduled_destroy" in
+	  chan_iax A signed short was used to represent a callnumber. This
+	  is makes it possible to attempt to access the iaxs array with a
+	  negative index. (closes issue #16565) Reported by: jensvb
+
+	* channels/chan_sip.c: Change in sip show channels display format
+	  allowing more digits for CID (closes issue 0016459) Reported by:
+	  Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins
+	  (license 953)
+
+2010-01-06 21:41 +0000 [r238230]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_cdr.c: Revise documentation on disposition values to
+	  the actual values used. (closes issue #16289) Reported by:
+	  wdoekes
+
+2010-01-06 15:18 +0000 [r237697-238009]  Russell Bryant <russell at digium.com>
+
+	* apps/app_mp3.c: Resolve a crash due to an ast_frame not being
+	  fully initialized. (closes issue #16531) Reported by: john8675309
+	  (closes SWP-615)
+
+	* main/utils.c: Change a NOTICE log message to DEBUG where it
+	  belongs. (closes issue #16479) Reported by: alexrecarey (closes
+	  SWP-577)
+
+2010-01-04 21:45 +0000 [r237318-237573]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/say.c: Bounds checking for input string (closes issue
+	  #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt
+	  uploaded by tilghman (license 14)
+
+	* main/pbx.c: Regression in issue #15421 - Pattern matching (closes
+	  issue #16482) Reported by: wdoekes Patches:
+	  astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
+	  20091223__issue16482.diff.txt uploaded by tilghman (license 14)
+	  Tested by: wdoekes, tilghman
+
+	* main/pbx.c, res/res_agi.c, include/asterisk/channel.h: Add a flag
+	  to disable the Background behavior, for AGI users. This is in a
+	  section of code that relates to two other issues, namely issue
+	  #14011 and issue #14940), one of which was the behavior of
+	  Background when called with a context argument that matched the
+	  current context. This fix broke FreePBX, however, in a post-Dial
+	  situation. Needless to say, this is an extremely difficult
+	  collision of several different issues. While the use of an
+	  exception flag is ugly, fixing all of the issues linked is rather
+	  difficult (although if someone would like to propose a better
+	  solution, we're happy to entertain that suggestion). (closes
+	  issue #16434) Reported by: rickead2000 Patches:
+	  20091217__issue16434.diff.txt uploaded by tilghman (license 14)
+	  20091222__issue16434__1.6.1.diff.txt uploaded by tilghman
+	  (license 14) Tested by: rickead2000
+
+	* channels/chan_local.c: It's also possible for the Local channel
+	  to directly execute an Application. Reviewboard:
+	  https://reviewboard.asterisk.org/r/452/
+
+2010-01-02 09:52 +0000 [r237135]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Release memory of the contact acl before
+	  unloading module
+
+2009-12-30 21:57 +0000 [r236981]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_local.c: Don't queue frames to channels that have
+	  no means to process them. (closes issue #15609) Reported by:
+	  aragon Patches:
+	  20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by
+	  tilghman (license 14) Tested by: aragon Review:
+	  https://reviewboard.asterisk.org/r/452/
+
+2009-12-30 20:25 +0000 [r236890]  Jeff Peeler <jpeeler at digium.com>
+
+	* utils/astman.c: Remove conflicting function definitions
+	  (asterisk.h) so LOW_MEMORY compiles.
+
+2009-12-28 15:12 +0000 [r236509-236585]  Sean Bright <sean at malleable.com>
+
+	* include/asterisk/threadstorage.h, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: Try a test
+	  compile to see if PTHREAD_ONCE_INIT requires extra braces. There
+	  was conditional code (based on build platform) to optioinally
+	  wrap PTHREAD_ONCE_INIT in braces that was removed since it is
+	  fixed in newer versions of Solaris/OpenSolaris, but I am still
+	  running into it on Solaris 10 x86 so add a configure-time check
+	  for it.
+
+	* apps/app_meetme.c: Avoid a crash with large numbers of MeetMe
+	  conferences. Similar to changes made to Queue(), when we have
+	  large numbers of conferences in meetme.conf (1000s) and we use
+	  alloca()/strdupa(), we can blow out the stack and crash, so
+	  instead just use a single fixed buffer. (closes issue #16509)
+	  Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded
+	  by seanbright (license 71) Tested by: seanbright
+
+2009-12-27 18:19 +0000 [r236433]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/init.d/rc.debian.asterisk: Turn on colors in the daemon,
+	  since there's many requests for it on Ubuntu.
+
+2009-12-26 15:26 +0000 [r236357]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* sounds/Makefile: update to latest releases with zero uid/gid
+
+2009-12-23 15:21 +0000 [r236261]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Properly set T.38 attributes and don't
+	  return before T.38 ports are configured when T.38 is found but no
+	  audio stream is found. (closes issue #16318) Reported by:
+	  bird_of_Luck Patches: t38-sdp-parsing-fix3.diff uploaded by
+	  mnicholson (license 96), written by vrban and mnicholson Tested
+	  by: vrban, mihaill
+
+2009-12-23 02:55 +0000 [r236184]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_agi.c: If EXEC only gets a single argument, don't crash
+	  when the second is used. (closes issue #16504) Reported by:
+	  bklang
+
+2009-12-22 16:58 +0000 [r236062]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes issue with p->method incorrectly set
+	  to ACK It is possible for a second ACK to come in for a
+	  retransmitted message. If an ack does not match an unacked
+	  message in our queue, restore the previous p->method as this ACK
+	  is completely ignored. (closes issue #16295) Reported by:
+	  omolenkamp Patches: issue16295_v2.diff uploaded by dvossel
+	  (license 671)
+
+2009-12-21 19:43 +0000 [r235940]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_monitor.c: Change Monitor to not assume file to write to
+	  does not contain pathing. 227944 changed the fname_base argument
+	  to always append the configured monitor path. This change was
+	  necessary to properly compare files for uniqueness. If a full
+	  path is given though, nothing needs to be appended and that is
+	  handled correctly now. (closes issue #16377) (closes issue
+	  #16376) Reported by: bcnit Patches:
+	  res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
+
+2009-12-21 16:45 +0000 [r235821]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_features.c: Send parking lot announcement to the channel
+	  which parked the call, not the park-ee. (closes issue #16234)
+	  Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt
+	  uploaded by tilghman (license 14)
+	  20091221__issue16234__1.4.diff.txt uploaded by tilghman (license
+	  14) Tested by: yeshuawatso
+
+2009-12-18 22:39 +0000 [r235652]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, configure.ac: Revise verbiage, per #asterisk-dev
+	  discussion
+
+2009-12-18 22:29 +0000 [r235635]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, include/asterisk/cdr.h: Correct CDR dispositions
+	  for BUSY/FAILED This patch is simple in that it reorders the
+	  disposition defines so that the fix for issue 12946 works
+	  properly (the default CDR disposition was changed to
+	  AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set
+	  in ast_call to ensure all CDR records are written. The side
+	  effects of CDR changes are scary, so I'm documenting the test
+	  cases performed to attempt to catch any regressions. The
+	  following tests were all performed using 1.4 rev 195881 vs head
+	  (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A
+	  (Both SIP and features) A calls B A blind transfers to C Hangup C
+	  (Both SIP and features) A calls B A attended transfers to C
+	  Hangup C A calls B A attended transfers to C (SIP) C blind
+	  transfers to A (features) Hangup A All of the test scenario CDRs
+	  matched. The following tests were performed just with the patch
+	  to ensure proper operation (with unanswered=yes): exten
+	  =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w)
+	  exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue
+	  #16180) Reported by: aatef Patches: bug16180.patch uploaded by
+	  jpeeler (license 325)
+
+2009-12-18 21:18 +0000 [r235572]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, configure.ac: Point to the typical missing package,
+	  not the cryptic "termcap support".
+
+2009-12-17  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.28
+
+2009-12-09  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.28-rc1
+
+2009-12-09 19:58 +0000 [r233782-233879]  Russell Bryant <russell at digium.com>
+
+	* main/loader.c: Fix breakage of the "module load <module>" CLI
+	  command.
+
+	* main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
+	  include/asterisk/module.h, formats/format_pcm.c,
+	  formats/format_h263.c, formats/format_g723.c,
+	  formats/format_h264.c, formats/format_jpeg.c,
+	  formats/format_g726.c, formats/format_gsm.c,
+	  formats/format_g729.c, formats/format_sln.c,
+	  formats/format_wav.c, formats/format_ogg_vorbis.c,
+	  formats/format_wav_gsm.c: Set a module load priority for format
+	  modules. A recent change to app_voicemail made it such that the
+	  module now assumes that all format modules are available while
+	  processing voicemail configuration. However, when autoloading
+	  modules, it was possible that app_voicemail was loaded before the
+	  format modules. Since format modules don't depend on anything,
+	  set a module load priority on them to ensure that they get loaded
+	  first when autoloading. This version of the patch is specific to
+	  Asterisk 1.4 and 1.6.0. These versions did not already support
+	  module load priority in the module API. This adds a trivial
+	  version of this which is just a module flag to include it in a
+	  pass before loading "everything". Thanks to mmichelson for the
+	  review! (closes issue #16412) Reported by: jiddings Tested by:
+	  russell Review: https://reviewboard.asterisk.org/r/445/
+
+2009-12-08 00:02 +0000 [r233618]  Atis Lezdins <atis at iq-labs.net>
+
+	* contrib/valgrind.supp: Merged revisions 233577 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 |
+	  atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix
+	  compatibility with valgrind 3.3 and older. (noticed in issue
+	  #16388) Reported by: parisioa Patches: valgrind.supp uloaded by
+	  atis (license 242) Tested by: atis, parisioa ........
+
+2009-12-07 23:24 +0000 [r233471-233609]  David Vossel <dvossel at digium.com>
+
+	* main/utils.c: hex escape control and non 7-bit clean characters
+	  in uri_encode In ast_uri_encode, non 7-bit clean characters were
+	  being hex escaped correctly, but control characters were not.
+	  (issue #16299)
+
+	* channels/chan_sip.c: fixes missing Contact header angle brackets
+	  (closes issue #16298) Reported by: mgernoth Patches:
+	  reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+	  by: dvossel
+
+2009-12-07 16:11 +0000 [r233392]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Allow SDP packets with only video session
+	  information. (closes issue #16387) Reported by: zalex1953 Tested
+	  by: mnicholson, zalex1953
+
+2009-12-04 21:54 +0000 [r233116-233279]  David Vossel <dvossel at digium.com>
+
+	* configs/iax.conf.sample: clarify requirecalltoken option in
+	  iax.sample.conf (closes issue #16223) Reported by: bklang
+	  Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
+	  (license 919)
+
+	* apps/app_voicemail.c: document and rename strip_control() in
+	  app_voicemail (closes issue #16291) Reported by: wdoekes
+
+2009-12-04 17:12 +0000 [r233092]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: Only do frame payload check for HOLD frames. This
+	  code was added for helping to debug the source of invalid HOLD
+	  frames. However, a side effect of this is that it will
+	  incorrectly report errors for frames that have an integer
+	  payload. Make the check for this block specific to the HOLD frame
+	  case.
+
+2009-12-04 16:59 +0000 [r233014-233091]  Matthias Nick <mnick at digium.com>
+
+	* pbx/pbx_config.c: Parse global variables or expressions in hint
+	  extensions Parse global variables or expressions in hint
+	  extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
+	  (closes issue #16166) Reported by: rmudgett Tested by: mnick,
+	  rmudgett
+
+	* main/dsp.c: Warning message gets displayed only once Added
+	  additional field 'int display_inband_dtmf_warning', which when
+	  set to '1' displays the warning ('Inband DTMF is not supported on
+	  codec %s. Use RFC2833'), and when set to '0' doesn't display the
+	  warning. Otherwise you would get hundreds of warnings every
+	  second. (closes issue #15769) Reported by: falves11 Patches:
+	  patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+	  mnick, falves11
+
+2009-12-03 20:10 +0000 [r232820]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Deprecate "cz" in favor of "cs". Also,
+	  change the use of language codes so that language registers as a
+	  prefix, rather than an exact match. (closes issue #16272)
+	  Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt
+	  uploaded by tilghman (license 14)
+
+2009-12-02 21:57 +0000 [r232581]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/manager.c: Send ack (response/message) after receiving
+	  manager action userevent (closes issue #16264) Reported by: dimas
+	  Patches: event-ack.patch uploaded by dimas (license 88)
+
+2009-12-02 19:03 +0000 [r232444]  David Vossel <dvossel at digium.com>
+
+	* apps/app_queue.c: fixes app_queue ao2 error (closes issue #16369)
+	  Reported by: vrban Patches: queue_issue_1.4.diff uploaded by
+	  dvossel (license 671) Tested by: dvossel
+
+2009-12-02 17:04 +0000 [r232355]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_amd.c: Fix a bug where if you hung up very quickly after
+	  calling AMD it would overwrite the AMDSTATUS of HANGUP with
+	  TOOLONG. (closes issue #16239) Reported by: CGMChris
+
+2009-12-02 16:59 +0000 [r232268-232350]  David Vossel <dvossel at digium.com>
+
+	* main/acl.c: ast_outaddrfor doesn't do htons() on port, looks odd
+	  in strace. (closes issue #16290) Reported by: wdoekes
+
+	* funcs/func_groupcount.c: fixes segfault in func_groupcount closes
+	  issue #16337) Reported by: Parantido Patches: issue_16337.diff
+	  uploaded by dvossel (license 671) Tested by: Parantido, dvossel
+
+2009-12-02 04:05 +0000 [r232165]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c: Fix compiling without devmode (closes issue
+	  #16367) Reported by: falves11
+
+2009-12-02 00:42 +0000 [r232090]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Do not modify the gain settings on data
+	  calls. (The digital flag actually represents a data call.)
+	  (closes issue #15972) Reported by: udosw Patches:
+	  transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
+	  Tested by: alecdavis
+
+2009-12-01 23:25 +0000 [r232007]  Russell Bryant <russell at digium.com>
+
+	* main/file.c: Fix a warning pointed out by buildbot.
+
+2009-12-01 21:52 +0000 [r231911-231926]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c: log channel name in dev mode as well
+
+	* main/channel.c: Fix crash with invalid frame data The crash was
+	  happening as a result of a frame containing an invalid data
+	  pointer, but was set with data length of zero. The few times the
+	  issue was reproduced it _seemed_ that the frame was queued
+	  properly, that is the data pointer was set to NULL. I never could
+	  reproduce the crash so as a last resort the crash has been fixed,
+	  but a check in __ast_read has been added to give as much
+	  information about the source of problematic frames in the future.
+	  (closes issue #16058) Reported by: atis
+
+2009-12-01 21:14 +0000 [r231853]  David Vossel <dvossel at digium.com>
+
+	* main/pbx.c: WaitExten m option with no parameters generates frame
+	  with zero datalen but non-null data ptr
+
+2009-12-01 15:34 +0000 [r231614-231740]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/file.c: Ignore unknown formats in ast_format_str_reduce()
+	  and return an error if no know formats are found.
+
+	* apps/app_voicemail.c, include/asterisk/file.h, main/file.c,
+	  main/app.c: Remove duplicate entries from voicemail format lists.
+	  This prevents app_voicemail from entering an infinite loop when
+	  the same format is specified twice in the format list. (closes
+	  issue #15625) Reported by: Shagg63 Tested by: mnicholson Review:
+	  https://reviewboard.asterisk.org/r/429/
+
+2009-11-30 17:14 +0000 [r231437-231441]  David Vossel <dvossel at digium.com>
+
+	* main/rtp.c: fixes crash caused by RTP comfort noise payload
+	  greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+	  by: amorsen Patches: issue16242.diff uploaded by oej (license
+	  306) Tested by: amorsen, oej, dvossel
+
+	* apps/app_queue.c: app_queue crashes randomly, often during
+	  call-transfers In app_queue, it is possible for a call_queue to
+	  be destroyed while another object still holds a pointer to it.
+	  This patch converts call_queue objects to ao2 objects allowing
+	  them to be ref counted. This makes it safe for the queue_ent
+	  object in queue_exec() to reference it's parent call_queue even
+	  after it has left the queue. (closes issue #15686) Reported by:
+	  Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license
+	  671) Tested by: dvossel, aragon Review:
+	  https://reviewboard.asterisk.org/r/427/
+
+2009-11-25 22:31 +0000 [r231298]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c: After a frame duplication failure, unlock the
+	  channel before returning.
+
+2009-11-25 21:38 +0000 [r231233-231235]  David Vossel <dvossel at digium.com>
+
+	* apps/app_dial.c: fixes solaris segfault on dial with verbosity >=
+	  3 (closes issue #16193) Reported by: asgaroth Patches:
+	  bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
+	  Tested by: asgaroth, snuffy
+
+	* channels/chan_sip.c: fixes conditional jump or move depending on
+	  uninitialised STACK value (closes issue #16261) Reported by:
+	  edguy3 Patches: edguy16261.patch uploaded by edguy3 (license 917)
+
+2009-11-23 15:31 +0000 [r230772-230875]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: When 'sip set debug' is enabled, and the
+	  last line of an incoming SIP message is not properly newline
+	  terminated, ensure that that line is included in the debug
+	  output. (part of issue #16268)
+
+	* main/editline/makelist.in, channels/chan_sip.c,
+	  channels/ring_tone.h, channels/busy_tone.h: Correct fix for issue
+	  #16268... the reporter's original patch was very close to
+	  correct.
+
+	* channels/chan_sip.c: Ensure that SDP parsing does not ignore the
+	  last line of the SDP. (closes issue #16268) Reported by: sgimeno
+
+2009-11-20 20:53 +0000 [r230627]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_features.c: Copy the peer CDR's userfield to the bridge
+	  CDR if it exists. This is necessary for the recordagentcalls
+	  option in chan_agent to store the recorded file name in the
+	  bridge CDR. (closes issue #14590) Reported by: msetim Patches:
+	  queue_agent_userfield.patch uploaded by Laureano (license 265)
+	  Tested by: Laureano, mnicholson
+
+2009-11-19 21:22 +0000 [r230508]  David Vossel <dvossel at digium.com>
+
+	* apps/app_mixmonitor.c: fixes MixMonitor thread not exiting when
+	  StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+	  Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+	  671) Tested by: dvossel, AlexMS Review:
+	  https://reviewboard.asterisk.org/r/424/
+
+2009-11-19 16:09 +0000 [r230469]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/asterisk.c: Update copyright year in visible output. (cli)
+	  Spotted by Stuart Henderson
+
+2009-11-30  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27.1
+
+	* AST-2009-010
+
+	* SDP parser regression fix (issue #16268)
+
+2009-11-18  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27
+
+2009-11-13  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27-rc5
+
+2009-11-12 16:41 +0000 [r229669]  David Vossel <dvossel at digium.com>
+
+	* funcs/func_audiohookinherit.c: fixes merging error, datastore was
+	  being freed in the wrong function. (closes issue #16219) Reported
+	  by: aragon
+
+2009-11-11 19:46 +0000 [r229498]  David Brooks <dbrooks at digium.com>
+
+	* main/pbx.c: Solaris doesn't like NULL going to ast_log Solaris
+	  will crash if NULL is passed to ast_log. This simple patch simply
+	  uses S_OR to get around this. (closes issue #15392) Reported by:
+	  yrashk
+
+2009-11-10 22:09 +0000 [r229360]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c: If two pattern classes start with the same digit and
+	  have the same number of characters, they will compare equal. The
+	  example given in the issue report is that of [234] and [246],
+	  which have these characteristics, yet they are clearly not
+	  equivalent. The code still uses these two characteristics, yet
+	  when the two scores compare equal, an additional check will be
+	  done to compare all characters within the class to verify
+	  equality. (closes issue #15421) Reported by: jsmith Patches:
+	  20091109__issue15421__2.diff.txt uploaded by tilghman (license
+	  14) Tested by: jsmith, thedavidfactor
+
+2009-11-10 21:45 +0000 [r229355]  David Ruggles <thedavidfactor at gmail.com>
+
+	* doc/externalivr.txt: Fix ExternalIVR Documentation Remove
+	  documentation for event that doesn't function (closes issue
+	  #16220) Reported by: thedavidfactor Patches:
+	  externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+	  (license 903)
+
+2009-11-10 20:03 +0000 [r229281]  Joshua Colp <jcolp at digium.com>
+
+	* codecs/codec_g726.c: Remove broken support for direct transcoding
+	  between G.726 RFC3551 and G.726 AAL2. On some systems the
+	  translation core would actually consider g726aal2 -> g726 ->
+	  signed linear to be a quicker path then g726aal2 -> signed linear
+	  which exposed this problem. (closes issue #15504) Reported by:
+	  globalnetinc
+
+2009-11-10 17:23 +0000 [r229191]  David Ruggles <thedavidfactor at gmail.com>
+
+	* doc/externalivr.txt: Document ExternalIVR event tag collision
+	  ExternalIVR uses the D tag for two different event types. This
+	  documents that behavior and how to differentiate between the two
+	  cases. Also includes a minor spelling fix and clarification
+	  (closes issue #16211) Reported by: thedavidfactor Patches:
+	  externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+	  (license 903)
+
+2009-11-10 17:15 +0000 [r229167]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: don't crash on log message in solaris
+	  AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
+	  bklang
+
+2009-11-10 15:22 +0000 [r229091]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Reverted revision 202022. (closes issue
+	  #16175) Reported by: paul-tg
+
+2009-11-09  Leif Madsen <lmadsen at digium.com>
+
+	* Release Astersik 1.4.27-rc4
+
+2009-11-09 15:37 +0000 [r228896]  Leif Madsen <lmadsen at digium.com>
+
+	* main/channel.c: Update WARNING message. Update a WARNING message
+	  to give a suggested fix when encountered. (closes issue #16198)
+	  Reported by: atis Tested by: atis
+
+2009-11-09 14:16 +0000 [r228827]  Matthew Nicholson <mnicholson at digium.com>
+
+	* include/asterisk/lock.h: Perform limited bounds checking when
+	  destroying ast_mutex_t structures to make sure we don't try to
+	  use negative indices. (closes issue #15588) Reported by: zerohalo
+	  Patches: 20090820__issue15588.diff.txt uploaded by tilghman
+	  (license 14) Tested by: zerohalo
+
+2009-11-06 22:33 +0000 [r228692]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c: fixes audiohook write crash occuring in chan_spy
+	  whisper mode. After writing to the audiohook list in ast_write(),
+	  frames were being freed incorrectly. Under certain conditions
+	  this resulted in a double free crash. (closes issue #16133)
+	  Reported by: wetwired (closes issue #16045) Reported by:
+	  bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
+	  671) Tested by: bluecrow76, dvossel, habile
+
+2009-11-06 18:32 +0000 [r228547]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Don't overwrite caller ID name on a trunk
+	  with the configured fullname when using users.conf (issue
+	  ABE-1989)
+
+2009-11-06  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27-rc3
+
+2009-11-06 17:07 +0000 [r228418]  David Vossel <dvossel at digium.com>
+
+	* codecs/codec_ilbc.c: fixes segfault in iLBC For reasons not yet
+	  known, it appears possible for an ast_frame to have a datalen
+	  greater than zero while the actual data is NULL during Packet
+	  Loss Concealment. Most codecs don't support PLC so this doesn't
+	  affect them. This patch catches the malformed frame and prevents
+	  the crash from occuring. Additional efforts to determine why it
+	  is possible for a frame to look like this are still being
+	  investigated. (issue #16979)
+
+2009-11-06 16:41 +0000 [r228409]  Joshua Colp <jcolp at digium.com>
+
+	* main/abstract_jb.c: Fix a bug caused by a partially invalid frame
+	  (from the jitterbuffer) passing through the Asterisk core.
+	  (closes issue #15560) Reported by: jvandal (closes issue #15709)
+	  Reported by: covici
+
+2009-11-06 16:26 +0000 [r228378]  Matthew Nicholson <mnicholson at digium.com>
+
+	* funcs/func_base64.c, main/utils.c: Properly handle '=' while
+	  decoding base64 messages and null terminate strings returned from
+	  BASE64_DECODE. (closes issue #15271) Reported by: chappell
+	  Patches: base64_fix.patch uploaded by chappell (license 8) Tested
+	  by: kobaz
+
+2009-11-06 15:41 +0000 [r228272-228338]  David Vossel <dvossel at digium.com>
+
+	* main/astfd.c: fixes crash in astfd.c (closes issue #15981)

[... 26687 lines stripped ...]



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