[asterisk-commits] russell: tag 1.4.20-rc2 r115531 - /tags/1.4.20-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed May 7 14:05:45 CDT 2008


Author: russell
Date: Wed May  7 14:05:44 2008
New Revision: 115531

URL: http://svn.digium.com/view/asterisk?view=rev&rev=115531
Log:
Importing files for 1.4.20-rc2 release

Added:
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    tags/1.4.20-rc2/.version   (with props)
    tags/1.4.20-rc2/ChangeLog   (with props)

Added: tags/1.4.20-rc2/.lastclean
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==============================================================================
--- tags/1.4.20-rc2/ChangeLog (added)
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@@ -1,0 +1,17493 @@
+2008-05-07  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.20-rc2 released.
+
+2008-05-07 18:17 +0000 [r115512-115517]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Track peer references when stored in the
+	  sip_pvt struct as the peer related to a qualify ping or a
+	  subscription. This fixes some realtime related crashes. (closes
+	  issue #12588) (closes issue #12555)
+
+2008-05-06 19:55 +0000 [r115418-115422]  Jason Parker <jparker at digium.com>
+
+	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
+	  7 lines read requires an argument on some non-bash shells (closes
+	  issue #12593) Reported by: bkruse Patches:
+	  getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
+	  ........
+
+	* res/res_musiconhold.c: Switch to using ast_random() rather than
+	  just rand(). This does not fix the bug reported, but I believe it
+	  is correct. (from issue #12446) Patches: bug_12446.diff uploaded
+	  by snuffy (license 35)
+
+2008-05-06 19:31 +0000 [r115415]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c: Don't print the terminating NUL. (Closes issue
+	  #12589)
+
+2008-05-06 13:54 +0000 [r115341]  Joshua Colp <jcolp at digium.com>
+
+	* configure, configure.ac: Add in missing argument.
+
+2008-05-05 22:50 +0000 [r115333]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, main/logger.c: Separate verbose output from CLI
+	  output, by using a preamble. (closes issue #12402) Reported by:
+	  Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
+	  uploaded by Corydon76 (license 14)
+	  20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
+	  Corydon76 (license 14)
+
+2008-05-05 22:10 +0000 [r115327]  Joshua Colp <jcolp at digium.com>
+
+	* build_tools/menuselect-deps.in, configure,
+	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+	  configure.ac: Make sure that either the main speex library
+	  contains preprocess functions or that speexdsp does. If both fail
+	  then speex stuff can not be built.
+
+2008-05-05 21:41 +0000 [r115320]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Don't consider a caller "handled" until the
+	  caller is bridged with a queue member. There was too much of an
+	  opportunity for the member to hang up (either during a delay,
+	  announcement, or overly long agi) between the time that he
+	  answered the phone and the time when he actually was bridged with
+	  the caller. The consequence of this was that if the member hung
+	  up in that interval, then proper abandonment details would not be
+	  noted in the queue log if the caller were to hang up at any point
+	  after the member hangup. (closes issue #12561) Reported by:
+	  ablackthorn
+
+2008-05-05 20:17 +0000 [r115308-115312]  Tilghman Lesher <tlesher at digium.com>
+
+	* Makefile: Reverse order, such that user configs override default
+	  selections
+
+	* include/asterisk/res_odbc.h: Err, the documentation on the return
+	  value of ast_odbc_backslash_is_escape is exactly backwards.
+
+2008-05-05 19:49 +0000 [r115297-115304]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Avoid putting opaque="" in Digest
+	  authentication. This patch came from switchvox. It fixes
+	  authentication with Primus in Canada, and has been in use for a
+	  very long time without causing problems with any other providers.
+	  (closes issue AST-36)
+
+2008-05-05 03:22 +0000 [r115285]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+	  contrib/init.d/rc.debian.asterisk,
+	  contrib/init.d/rc.mandrake.asterisk,
+	  contrib/init.d/rc.redhat.asterisk,
+	  contrib/init.d/rc.gentoo.asterisk,
+	  contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
+	  out if Asterisk is already running. (closes issue #12525)
+	  Reported by: explidous Patches: 20080428__bug12525.diff.txt
+	  uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-05-04 02:09 +0000 [r115276-115282]  Joshua Colp <jcolp at digium.com>
+
+	* configure, acinclude.m4: Expand the test function for GCC
+	  attributes so that more complex attributes are properly
+	  recognized.
+
+	* include/asterisk/compiler.h: For my next trick I will make these
+	  work with what our autoconf header file gives us.
+
+	* configure, acinclude.m4: Treat warnings as errors when checking
+	  if a GCC attribute exists. We have to do this as GCC will just
+	  ignore the attribute and pop up a warning, it won't actually fail
+	  to compile.
+
+2008-05-02 20:25 +0000 [r115257]  Brett Bryant <bbryant at digium.com>
+
+	* channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac, CHANGES: Add new "pri show version" command to show
+	  the libpri version for support reasons.
+
+2008-05-02 14:28 +0000 [r115196]  Mark Michelson <mmichelson at digium.com>
+
+	* include/asterisk/sched.h: Clarify a comment that was, well, just
+	  wrong. It turns out that ignoring the way that macros expand.
+	  Instead, I have clarified in the comment why the macro will work
+	  even if the scheduler id for the task to be deleted changes
+	  during the execution of the macro.
+
+2008-05-01 23:20 +0000 [r115017-115102]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/res_odbc.h: Change the comment of deprecated to
+	  an actual compiler deprecation
+
+	* main/utils.c: '#' is another reserved character for URIs that
+	  also needs to be escaped. (closes issue #10543) Reported by:
+	  blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
+	  Corydon76 (license 14)
+
+2008-05-01  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.20-rc1 released.
+
+2008-04-30 16:30 +0000 [r114891]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
+	  Merge changes from team/russell/iax2_find_callno and
+	  iax2_find_callno_1.4 These changes address a critical performance
+	  issue introduced in the latest release. The fix for the latest
+	  security issue included a change that made Asterisk randomly
+	  choose call numbers to make them more difficult to guess by
+	  attackers. However, due to some inefficient (this is by far, an
+	  understatement) code, when Asterisk chose high call numbers,
+	  chan_iax2 became unusable after just a small number of calls. On
+	  a small embedded platform, it would not be able to handle a
+	  single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
+	  more than about 16 IAX2 channels. Ouch. These changes address
+	  some performance issues of the find_callno() function that have
+	  bothered me for a very long time. On every incoming media frame,
+	  it iterated through every possible call number trying to find a
+	  matching active call. This involved a mutex lock and unlock for
+	  each call number checked. So, if the random call number chosen
+	  was 20000, then every media frame would cause 20000 locks and
+	  unlocks. Previously, this problem was not as obvious since
+	  Asterisk always chose the lowest call number it could. A second
+	  container for IAX2 pvt structs has been added. It is an astobj2
+	  hash table. When we know the remote side's call number, the pvt
+	  goes into the hash table with a hash value of the remote side's
+	  call number. Then, lookups for incoming media frames are a very
+	  fast hash lookup instead of an absolutely insane array traversal.
+	  In a quick test, I was able to get more than 3600% more IAX2
+	  channels on my machine with these changes.
+
+2008-04-30 16:23 +0000 [r114890]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
+	  in Huntsville together with Mark M (putnopvut) (closes issue
+	  #12363) Reported by: jvandal Tested by: putnopvut, oej
+
+2008-04-30 14:46 +0000 [r114875-114880]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
+	  for indexing through the iaxs/iaxsl arrays so that the size of
+	  the arrays can be adjusted in one place, and change the size of
+	  the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
+	  defined
+
+	* Makefile.rules: pay attention to *all* header files for
+	  dependency tracking, not just the local ones (inspired by r578 of
+	  asterisk-addons by tilghman)
+
+2008-04-29 19:40 +0000 [r114848]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
+	  variables instead of the channel's macrocontext and macroexten
+	  fields. This is needed because if macros are daisy-chained, the
+	  incorrect context and extension are placed on the new channel. I
+	  also added locking to the channel prior to accessing these
+	  variables as noted in trunk's janitor project file. (closes issue
+	  #12549) Reported by: darren1713 Patches:
+	  app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
+	  (with modifications from me) Tested by: putnopvut
+
+2008-04-29 17:08 +0000 [r114829]  Jason Parker <jparker at digium.com>
+
+	* res/res_config_pgsql.c: Change warning message to debug, since
+	  there are cases where 0 results is perfectly fine.
+
+2008-04-29 12:53 +0000 [r114823]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
+	  2008) | 2 lines stop script from appending source code if run
+	  multiple times ........
+
+2008-04-28 04:47 +0000 [r114708]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
+	  embedded, they take on a different name, without the ".so"
+	  extension. Specifically check for this name, when we're checking
+	  if a module is loaded. (Closes issue #12534)
+
+2008-04-27 01:26 +0000 [r114695]  Sean Bright <sean.bright at gmail.com>
+
+	* configure, configure.ac: When we don't explicitly pass a path to
+	  the --with-tds configure option, we may end up finding tds.h in
+	  /usr/local/include instead of /usr/include. If this happens, the
+	  grep that looks for the version (from tdsver.h) will fail and
+	  we'll have some problems during the build.
+
+2008-04-26 13:15 +0000 [r114689]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/scripts/vmail.cgi: Clicking forward without selecting a
+	  message leaves an errant .lock file. (closes issue #12528)
+	  Reported by: pukepail Patches: patch.diff uploaded by pukepail
+	  (license 431)
+
+2008-04-25 21:54 +0000 [r114673]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Use consistent logic for checking to see if
+	  a call number has been chosen yet. Also, remove some redundant
+	  logic I recently added in a fix.
+
+2008-04-25 19:32 +0000 [r114662]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_chanspy.c: Move the unlock of the spyee channel to
+	  outside the start_spying() function so that the channel is not
+	  unlocked twice when using whisper mode.
+
+2008-04-25 15:53 +0000 [r114649]  Tilghman Lesher <tlesher at digium.com>
+
+	* configs/zapata.conf.sample, configs/iax.conf.sample,
+	  configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
+	  documentation files that actually exist. (closes issue #12516)
+	  Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
+	  by linuxmaniac (license 472)
+
+2008-04-24 21:35 +0000 [r114624-114632]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Re-invite RTP during a masquerade so that,
+	  for instance, an AMI redirect of two channels which are natively
+	  bridged will preserve audio on both channels. This prevents a
+	  problem with Asterisk not re-inviting due to one of the channels
+	  having being a zombie. (closes issue #12513) Reported by:
+	  mneuhauser Patches:
+	  asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
+	  mneuhauser (license 425)
+
+	* apps/app_queue.c: Output of channel variables when
+	  eventwhencalled=vars was set was being truncated two characters.
+	  This patch corrects the problem. (closes issue #12493) Reported
+	  by: davidw
+
+	* channels/chan_local.c: Resolve a deadlock in chan_local by
+	  releasing the channel lock temporarily. (closes issue #11712)
+	  Reported by: callguy Patches: 11712.patch uploaded by putnopvut
+	  (license 60) Tested by: acunningham
+
+2008-04-24 19:53 +0000 [r114621]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_local.c: Ensure that when we set the accountcode,
+	  it actually shows up in the CDR. (Fix for AMI Originate) (Closes
+	  issue #12007)
+
+2008-04-24 15:55 +0000 [r114608]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix a silly mistake in a change I made
+	  yesterday that caused chan_iax2 to blow up very quickly. (issue
+	  #12515)
+
+2008-04-24 14:55 +0000 [r114603]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Only have one max-forwards header in
+	  outbound REFERs. Discovered in the Asterisk SIP Masterclass in
+	  Orlando. Thanks Joe!
+
+2008-04-23 22:18 +0000 [r114597-114600]  Russell Bryant <russell at digium.com>
+
+	* main/http.c: Improve some broken cookie parsing code. Previously,
+	  manager login over HTTP would only work if the mansession_id
+	  cookie was first. Now, the code builds a list of all of the
+	  cookies in the Cookie header. This fixes a problem observed by
+	  users of the Asterisk GUI. (closes AST-20)
+
+	* apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
+	  the correct next channel to not always work. Also, remove setting
+	  the amount of time to wait for a digit from 5 seconds back down
+	  to 1/10 of a second. I believe this was so the beep didn't get
+	  played over and over really fast, but a while back I put in
+	  another fix for that issue. (closes issue #12498) Reported by:
+	  jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
+	  jsmith (license 15)
+
+2008-04-23 18:28 +0000 [r114594]  Jason Parker <jparker at digium.com>
+
+	* res/res_musiconhold.c: Fix reload/unload for res_musiconhold
+	  module. (closes issue #11575) Reported by: sunder Patches:
+	  M11575_14_rev3.diff uploaded by junky (license 177)
+	  bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
+
+2008-04-23 17:55 +0000 [r114587-114591]  Russell Bryant <russell at digium.com>
+
+	* main/manager.c, include/asterisk/manager.h: Store the manager
+	  session ID explicitly as 4 byte ID instead of a ulong. The
+	  mansession_id cookie is coded to be limited to 8 characters of
+	  hex, and this could break logins from 64-bit machines in some
+	  cases. (inspired by AST-20)
+
+	* channels/chan_iax2.c: Fix find_callno_locked() to actually return
+	  the callno locked in some more cases.
+
+2008-04-23 16:51 +0000 [r114584]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Add 502 support for both directions, not
+	  only one... (see r114571)
+
+2008-04-23 14:54 +0000 [r114579]  Joshua Colp <jcolp at digium.com>
+
+	* main/pbx.c: Instead of stopping dialplan execution when SayNumber
+	  attempts to say a large number that it can not print out a
+	  message informing the user and continue on. (closes issue #12502)
+	  Reported by: bcnit
+
+2008-04-22 23:51 +0000 [r114571]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Treat a 502 just like a 503, when it comes
+	  to processing a response code
+
+2008-04-22 22:15 +0000 [r114522-114558]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: When we receive a full frame that is
+	  supposed to contain our call number, ensure that it has the
+	  correct one. (closes issue #10078) (AST-2008-006)
+
+	* main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
+	  thought I was going to be able to leave 1.4 alone, but that was
+	  not the case. I ran into some problems with G.722 in 1.4, so I
+	  have merged in all of the fixes in this area that I have made in
+	  trunk/1.6.0, and things are happy again.
+
+	* res/res_musiconhold.c: Trivial change to read the number of
+	  samples from a frame before calling ast_write()
+
+	* res/res_features.c: After a parked call times out, allow the call
+	  back to the parker to time out. (closes issue #10890)
+
+	* channels/chan_iax2.c: If the dial string passed to the call
+	  channel callback does not indicate an extension, then consider
+	  the extension on the channel before falling back to the default.
+	  (closes issue #12479) Reported by: darren1713 Patches:
+	  exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
+	  116)
+
+	* channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
+	  team/russell/issue_9520 These changes make sure that the
+	  reference count for sip_peer objects properly reflects the fact
+	  that the peer is sitting in the scheduler for a scheduled
+	  callback for qualifying peers or for expiring registrations.
+	  Without this, it was possible for these callbacks to happen at
+	  the same time that the peer was being destroyed. This was
+	  especially likely to happen with realtime peers, and for people
+	  making use of the realtime prune CLI command. (closes issue
+	  #9520) Reported by: kryptolus Committed patch by me
+
+2008-04-21 14:39 +0000 [r114322]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Only drop audio if we receive it without a
+	  progress indication. We allow other frames through such as DTMF
+	  because they may be needed to complete the call. (closes issue
+	  #12440) Reported by: aragon
+
+2008-04-19 13:57 +0000 [r114297-114299]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_playback.c: Ensure that help text terminates with a
+	  newline
+
+	* res/res_musiconhold.c: MOH usage information needs a terminating
+	  newline, or else "asterisk -rx 'help moh reload'" will hang.
+	  Reported via -dev list, fixed by me.
+
+2008-04-18 21:48 +0000 [r114275-114284]  Russell Bryant <russell at digium.com>
+
+	* main/manager.c: Don't destroy a manager session if poll() returns
+	  an error of EAGAIN.
+
+	* Makefile: ensure directories are created before we try to install
+	  stuff into them
+
+	* Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
+	  bininstall
+
+2008-04-18 17:44 +0000 [r114257]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_zap.c, main/callerid.c: Clearing up error messages
+	  so they make a bit more sense. Also removing a redundant error
+	  message. Issue AST-15
+
+2008-04-18 15:24 +0000 [r114248]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
+	  (closes issue #12476) Reported by: davidw Patch by me
+
+2008-04-18 13:33 +0000 [r114245]  Sean Bright <sean.bright at gmail.com>
+
+	* channels/chan_sip.c: Only complete the SIP channel name once for
+	  'sip show channel <channel>'
+
+2008-04-18 06:49 +0000 [r114242]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_setcallerid.c: For consistency sake, ensure that the
+	  values that ${CALLINGPRES} returns are valid as an input to
+	  SetCallingPres. (Closes issue #12472)
+
+2008-04-17 22:15 +0000 [r114230]  Russell Bryant <russell at digium.com>
+
+	* main/autoservice.c: Remove redundant safety net. The check for
+	  the autoservice channel list state accomplishes the same goal in
+	  a better way. (issue #12470) Reported By: atis
+
+2008-04-17 21:03 +0000 [r114207-114226]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_chanspy.c: Declaration of the peer channel in this scope
+	  was making it so the peer variable defined in the outer scope was
+	  never set properly, therefore making iterating through the
+	  channel list always restart from the beginning. This bug would
+	  have affected anyone who called chanspy without specifying a
+	  first argument. (closes issue #12461) Reported by: stever28
+
+	* main/frame.c, include/asterisk/dsp.h: Add prototype for
+	  ast_dsp_frame_freed. I'm not sure how this was compiling
+	  before...
+
+	* main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
+	  possible for a reference to a frame which was part of a freed DSP
+	  to still be referenced, leading to memory corruption and eventual
+	  crashes. This code change ensures that the dsp is freed when we
+	  are finished with the frame. This change is very similar to a
+	  change Russell made with translators back a month or so ago.
+	  (closes issue #11999) Reported by: destiny6628 Patches:
+	  11999.patch uploaded by putnopvut (license 60) Tested by:
+	  destiny6628, victoryure
+
+2008-04-17 16:23 +0000 [r114204]  Russell Bryant <russell at digium.com>
+
+	* Makefile: Fix the bininstall target to install from subdirs, as
+	  well. (closes issue AST-8, patch from bmd at switchvox)
+
+2008-04-17 13:42 +0000 [r114198]  Philippe Sultan <philippe.sultan at gmail.com>
+
+	* res/res_jabber.c: Use keepalives effectively in order diagnose
+	  bug #12432.
+
+2008-04-17 12:56 +0000 [r114195]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_agi.c: Add special case for when the agi cannot be
+	  executed, to comply with the documentation that we return failure
+	  in that case. (closes issue #12462) Reported by: fmueller
+	  Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
+	  (license 14) Tested by: fmueller
+
+2008-04-17 10:51 +0000 [r114191]  Sean Bright <sean.bright at gmail.com>
+
+	* apps/app_chanspy.c: Make sure we have enough room for the
+	  recording's filename.
+
+2008-04-16 20:46 +0000 [r114184]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
+	  initializing the structure to all zeroes in case it contains
+	  fields that we don't write values into (which it does as of
+	  Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
+
+2008-04-16 19:59 +0000 [r114180]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_vpb.cc: Backport revisions for latest vpb drivers
+	  to 1.4 (Closes issue #12457)
+
+2008-04-16 17:30 +0000 [r114173]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_zap.c: Fix "fallthrough" behavior here, so config
+	  options in a previously configured user don't override settings
+	  in general. (closes issue #12458) Reported by: tzafrir Patches:
+	  chanzap_users_sections.diff uploaded by tzafrir (license 46)
+
+2008-04-16 14:10 +0000 [r114167]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_meetme.c: Include the proper headers for using mkdir on
+	  FreeBSD. (closes issue #12430) Reported by: ys Patches:
+	  app_meetme.c.diff uploaded by ys (license 281)
+
+2008-04-15 20:26 +0000 [r114148]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Handle subscribe queues in all situations...
+	  Thanks to festr_ on irc for telling me about this bug.
+
+2008-04-15 17:17 +0000 [r114120-114138]  Jason Parker <jparker at digium.com>
+
+	* contrib/scripts/autosupport: Update Digium autosupport script,
+	  for more useful information. (closes issue #12452) Reported by:
+	  angler Patches: autosupport.diff uploaded by angler (license 106)
+
+	* apps/app_queue.c: Allow autofill to work in the general section
+	  of queues.conf. Additionally, don't try to (re)set options when
+	  they have empty values in realtime (all unset columns would have
+	  an empty value). (closes issue #12445) Reported by: atis Patches:
+	  12445-autofill.diff uploaded by qwell (license 4)
+
+	* channels/chan_h323.c: The call_token on the pvt can occasionally
+	  be NULL, causing a crash. If it is NULL, we can skip this
+	  channel, since it can't the one we're looking for. (closes issue
+	  #9299) Reported by: vazir
+
+2008-04-14 17:41 +0000 [r114106-114117]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c: Increase the retry count when attempting to show
+	  channels. This apparently cleared an issue someone was seeing
+	  when attempting to show channels when the load was high. (closes
+	  issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
+	  by russell (license 2) Tested by: falves11
+
+	* apps/app_dial.c, apps/app_queue.c: If the datastore has been
+	  moved to another channel due to a masquerade, then freeing the
+	  datastore here causes an eventual double free when the new
+	  channel hangs up. We should only free the datastore if we were
+	  able to successfully remove it from the channel we are
+	  referencing (i.e. the datastore was not moved). (closes issue
+	  #12359) Reported by: pguido
+
+	* main/channel.c: Save a local copy of the generate callback prior
+	  to unlocking the channel in case the generate callback goes NULL
+	  on us after the channel is unlocked. Thanks to Russell for
+	  pointing this need out to me.
+
+2008-04-14 14:52 +0000 [r114100-114103]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: It is possible for the remote side to say
+	  they want T38 but not give any capabilities. (closes issue
+	  #12414) Reported by: MVF
+
+	* main/rtp.c: Don't change the SSRC when a new source comes into
+	  play, this might happen quite often and depending on the remote
+	  side... they might not like this. (closes issue #12353) Reported
+	  by: dimas
+
+2008-04-11 22:32 +0000 [r114083]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_iax2.c: Several places in the code called
+	  find_callno() (which releases the lock on the pvt structure) and
+	  then immediately locked the call and did things with it.
+	  Unfortunately, the call can disappear between the find_callno and
+	  the lock, causing Bad Stuff(tm) to happen. Added
+	  find_callno_locked() function to return the callno withtout
+	  unlocking for instances that it is needed. (issue #12400)
+	  Reported by: ztel
+
+2008-04-11 21:35 +0000 [r114072]  Jason Parker <jparker at digium.com>
+
+	* main/pbx.c: It's possible that a channel can have an async goto
+	  on the successful execution of an application as well. Closes
+	  issue #12172.
+
+2008-04-11 15:44 +0000 [r114045-114063]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_features.c: Fix a race condition that may happen between
+	  a sip hangup and a "core show channel" command. This patch adds
+	  locking to prevent the resulting crash. (closes issue #12155)
+	  Reported by: tsearle Patches: show_channels_crash2.patch uploaded
+	  by tsearle (license 373) Tested by: tsearle
+
+	* main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
+	  LOW_MEMORY is enabled.
+
+	* channels/chan_sip.c: Be sure that we're not about to set
+	  bridgepvt NULL prior to dereferencing it. (closes issue #11775)
+	  Reported by: fujin
+
+2008-04-10 17:26 +0000 [r114035]  Jason Parker <jparker at digium.com>
+
+	* main/file.c: Only try to prefix language if we are not using an
+	  absolute path (suffix it otherwise).
+	  en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
+	  issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
+	  uploaded by qwell (license 4) Tested by: kuj, qwell
+
+2008-04-10 15:58 +0000 [r114021-114032]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_voicemail.c: Forgot the 1.4 branch for russian language
+	  fix. (closes issue #12404) Reported by: IgorG Patches:
+	  voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
+
+	* apps/app_meetme.c: Create the directory where name recordings
+	  will go if it does not exist. (closes issue #12311) Reported by:
+	  rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
+
+	* channels/chan_sip.c: Don't add custom URI options if they don't
+	  exist OR they are empty. (closes issue #12407) Reported by:
+	  homesick Patches: uri_options-1.4.diff uploaded by homesick
+	  (license 91)
+
+2008-04-09 20:54 +0000 [r113927]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: We need to set the persistant_route [sic]
+	  parameter for the sip_pvt during the initial INVITE, no matter if
+	  we're building the route set from an INVITE request or response.
+	  (closes issue #12391) Reported by: benjaminbohlmann Tested by:
+	  benjaminbohlmann
+
+2008-04-09 18:57 +0000 [r113874]  Tilghman Lesher <tlesher at digium.com>
+
+	* cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
+	  not exist in cdr.conf, then an unload/load sequence is needed to
+	  correct the problem. Track whether the load succeeded with a
+	  variable, so we can fix this with a simple reload event, instead.
+
+2008-04-09 16:50 +0000 [r113784]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_iax2.c: If we receive an AUTHREQ from the remote
+	  server and we are unable to reply (for example they have a secret
+	  configured, but we do not) then queue a hangup frame on the
+	  Asterisk channel. This will cause the channel to hangup and a
+	  HANGUP to be sent via IAX2 to the remote side which is the proper
+	  thing to do in this scenario. (closes issue #12385) Reported by:
+	  viraptor
+
+2008-04-09 14:40 +0000 [r113681]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
+	  a reinvite), then we should not send a BYE. (closes issue #12392)
+	  Reported by: fnordian Patches: chan_sip.patch uploaded by
+	  fnordian (license 110) with small modification from me
+
+2008-04-09 01:34 +0000 [r113596]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
+	  happy
+
+2008-04-08 19:07 +0000 [r113507]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_parkandannounce.c: Fix potential buffer overflow that
+	  could happen if more than 100 announce files were specified when
+	  calling ParkAndAnnounce. This overflow is not exploitable
+	  remotely and so there is no need for a security advisory. (closes
+	  issue #12386) Reported by: davidw
+
+2008-04-08 18:48 +0000 [r113402-113504]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c: Add a little more that is required for
+	  previously added devices.
+
+	* channels/chan_skinny.c: Add support for several new(ish) devices
+	  - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
+	  for providing me the required information.
+
+	* main/asterisk.c: Work around some silliness caused by
+	  sys/capability.h - this should fix compile errors a number of
+	  users have been experiencing.
+
+2008-04-08 16:51 +0000 [r113348-113399]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/scripts/astgenkey.8: Add security note on astgenkey's
+	  manpage. (closes issue #12373) Reported by: lmamane Patches:
+	  20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
+
+	* channels/chan_sip.c: Move check for still-bridged channels out a
+	  little further, to avoid possible deadlocks. (Closes issue
+	  #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
+	  uploaded by Corydon76 (license 14) Tested by: callguy
+
+2008-04-08 15:03 +0000 [r113296]  Joshua Colp <jcolp at digium.com>
+
+	* include/asterisk/slinfactory.h, main/slinfactory.c,
+	  main/audiohook.c: If audio suddenly gets fed into one side of a
+	  channel after a lapse of frames flush the other factory so that
+	  old audio does not remain in the factory causing the sync code to
+	  not execute. (closes issue #12296) Reported by: jvandal
+
+2008-04-07 21:34 +0000 [r113240]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
+	  fixes a for loop (in realtime_peer) to check all the
+	  ast_variables the loop was intending to test rather than just the
+	  first one. The change exposed the problem of calling memcpy on a
+	  NULL pointer, in this case the passed in sockaddr_in struct which
+	  is now checked.
+
+2008-04-07 18:00 +0000 [r113118]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c, configs/skinny.conf.sample: Allow
+	  playback with noanswer (and add earlyrtp option). (closes issue
+	  #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
+	  (license 30) Tested by: pj, qwell, DEA, wedhorn
+
+2008-04-07 17:51 +0000 [r113117]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_strings.c: Force ast_mktime() to check for DST, since
+	  strptime(3) does not. (Closes issue #12374)
+
+2008-04-07 16:08 +0000 [r113065]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c: This fix prevents a deadlock that was experienced
+	  in chan_local. There was deadlock prevention in place in
+	  chan_local, but it would not work in a specific case because the
+	  channel was recursively locked. By unlocking the channel prior to
+	  calling the generator's generate callback in
+	  ast_read_generator_actions(), we prevent the recursive locking,
+	  and therefore the deadlock. (closes issue #12307) Reported by:
+	  callguy Patches: 12307.patch uploaded by putnopvut (license 60)
+	  Tested by: callguy
+
+2008-04-07 15:16 +0000 [r113012]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
+	  Reported by: vinsik Tested by: tecnoxarxa This one line change
+	  makes an if inside a for loop (in realtime_peer) check all the
+	  ast_variables the loop was intending to test rather than just the
+	  first one.
+
+2008-04-04 19:26 +0000 [r112766-112820]  Philippe Sultan <philippe.sultan at gmail.com>
+
+	* channels/chan_gtalk.c: Free newly allocated channel before
+	  returning
+
+	* channels/chan_gtalk.c: Prevent call connections when codecs don't
+	  match. (closes issue #10604) Reported by: keepitcool Patches:
+	  branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
+	  by: phsultan
+
+2008-04-04 00:52 +0000 [r112709-112711]  Joshua Colp <jcolp at digium.com>
+
+	* main/Makefile: Pass in the path to Zaptel for systems that
+	  install Zaptel headers in a separate location.
+
+	* main/asterisk.c: One thing at a time... let's get 1.4 building.
+
+2008-04-03 23:57 +0000 [r112689]  Dwayne M. Hubbard <dhubbard at digium.com>
+
+	* main/asterisk.c: add a Zaptel timer check to verify the timer is
+	  responding when Zaptel support is compiled into Asterisk and
+	  Zaptel drivers are loaded. This will help people not waste their
+	  valuable time debugging side effects.
+
+2008-04-03 14:32 +0000 [r112393-112599]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_zap.c: Fix the testing of the "res" variable so
+	  that it is more logically correct and makes the correct warning
+	  and debug messages print. (closes issue #12361) Reported by:
+	  one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
+	  (license 23)
+
+	* main/manager.c: Fix a race condition in the manager. It is
+	  possible that a new manager event could be appended during a
+	  brief time when the manager is not waiting for input. If an event
+	  comes during this period, we need to set an indicator that there
+	  is an event pending so that the manager doesn't attempt to wait
+	  forever for an event that already happened. (closes issue #12354)
+	  Reported by: bamby Patches: manager_race_condition.diff uploaded
+	  by bamby (license 430) (comments added by me)
+
+	* apps/app_queue.c: Ensure that there is no timeout if none is
+	  specified. (closes issue #12349) Reported by: johnlange
+
+2008-04-01  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.19 released.
+
+2008-03-28  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.19-rc4 released.
+
+2008-03-28 16:19 +0000 [r111658]  Jason Parker <jparker at digium.com>
+
+	* formats/format_wav_gsm.c: The file size of WAV49 does not need to
+	  be an even number. (closes issue #12128) Reported by: mdu113
+	  Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
+	  Tested by: qwell, mdu113
+
+2008-03-28 14:35 +0000 [r111442-111605]  Tilghman Lesher <tlesher at digium.com>
+
+	* doc/valgrind.txt: Update debugging text, since Valgrind
+	  eliminated the --log-file-exactly option. (Closes issue #12320)
+
+	* main/acl.c: For FreeBSD, at least, the ifa_addr element could be
+	  NULL. (closes issue #12300) Reported by: festr Patches:
+	  acl.c.patch uploaded by festr (license 443)
+
+2008-03-27 13:03 +0000 [r111341-111391]  Steve Murphy <murf at digium.com>
+
+	* apps/app_playback.c, main/pbx.c: These small documentation
+	  updates made in response to a query in asterisk-users, where a
+	  user was using Playback, but needed the features of Background,
+	  and had no idea that Background existed, or that it might provide
+	  the features he needed. I thought the best way to avert these
+	  kinds of queries was to provide "See Also" references in all
+	  three of "Background", "Playback", "WaitExten". Perhaps a project
+	  to do this with all related apps is in order.
+
+	* pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
+	  #12302) Reported by: pj Tested by: murf These changes will set a
+	  channel variable ~~EXTEN~~ just before generating code for a
+	  switch, with the value of ${EXTEN}. The exten is marked as having
+	  a switch, and ever after that, till the end of the exten, we
+	  substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
+	  arguments; (and the ${EXTEN: also). The reason for this, is that
+	  because switches are coded using separate extensions to provide
+	  pattern matching, and jumping to/from these switch extensions
+	  messes up the ${EXTEN} value, which blows the minds of users.
+
+2008-03-27 00:25 +0000 [r111245-111280]  Jason Parker <jparker at digium.com>
+
+	* main/frame.c: Put this flag back so we don't change the API.
+
+	* main/frame.c: Remove excessive smoother optimization that was
+	  causing audio glitches (small "pops") after (about 200ms later)
+	  an "incorrectly" sized frame was received. While it would be very
+	  nice to keep this as optimized as possible, it makes no sense for
+	  the smoother to be dropping random bits of audio like this. Isn't

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