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2008-03-18  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.18.1 released.

2008-03-18 16:25 +0000 [r109482]  Terry Wilson <twilson@digium.com>

	* include/asterisk/astobj.h: Fix character string being treated ad
	  format string

2008-03-18 15:10 +0000 [r109393]  Jason Parker <jparker@digium.com>

	* /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
	  3 lines Do not return with a successful authentication if the
	  From header ends up empty. (AST-2008-003) ........

2008-03-18 14:58 +0000 [r109386]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
	  number of payloads accepted, and also make sure a given payload
	  does not exceed our maximum value. (AST-2008-002)

2008-01-31  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.18 released.

2008-01-31 23:10 +0000 [r101601]  Russell Bryant <russell@digium.com>

	* main/translate.c, main/file.c: Fix a couple of places where
	  ast_frfree() was not called on a frame that came from a
	  translator. This showed itself by g729 decoders not getting
	  released. Since the flag inside the translator frame never got
	  unset by freeing the frame to indicate it was no longer in use,
	  the translators never got destroyed, and thus the g729 licenses
	  were not released. (closes issue #11892) Reported by: xrg
	  Patches: 11892.diff uploaded by russell (license 2) Tested by:
	  xrg, russell

2008-01-31 21:00 +0000 [r101531]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
	  beginning of an absolute pathname. 2. If ast_monitor_change_fname
	  is called and the new filename is the same as the old, then exit
	  early and don't set the filename_changed field in the monitor
	  structure. Setting it in this case was causing ast_monitor_stop
	  to erroneously delete them. (closes issue #11741) Reported by:
	  garlew Tested by: putnopvut

2008-01-31 19:52 +0000 [r101482]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
	  for struct in_addr funkiness. Issue #11885, patch by snuffy.

2008-01-31 19:30 +0000 [r101480]  Steve Murphy <murf@digium.com>

	* main/pbx.c: closes issue #11845; that's the one where there's a
	  1004 byte cdr leak with every AMI Redirect to a zap channel

2008-01-31 19:17 +0000 [r101413-101433]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Add more missing locking of the agents
	  list ...

	* channels/chan_agent.c: Move the locking from find_agent() into
	  the agent dialplan function handler to ensure that the agent
	  doesn't disappear while we're looking at it.

	* channels/chan_agent.c: Add missing locking to the find_agent()
	  function.

2008-01-30 15:41 +0000 [r101222]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Fix an issue where if a frame of higher
	  sample size preceeded a frame of lower sample size and
	  ast_slinfactory_read was called with a sample size of the
	  combined values or higher a crash would happen. (closes issue
	  #11878) Reported by: stuarth

2008-01-30 15:34 +0000 [r101219]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample: Change default config to use
	  descending channel order of groups, rather than ascending. Fixes
	  a potential source of confusion in glare-type situations. Issue
	  11875, reported by JimVanM.

2008-01-30 15:23 +0000 [r101216]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a logic error with regards to autofill.
	  Prior to this change, it was possible for a caller to go out of
	  turn if autofill were enabled and callers ahead in the queue were
	  attempting to call a member. This change fixes this.

2008-01-30 11:20 +0000 [r101152]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Stop musiconhold on attended transfer.
	  (closes issue #11872) Reported by: gareth Patches:
	  svn-101018.patch uploaded by gareth (license 208)

2008-01-29 23:50 +0000 [r101080]  Dwayne M. Hubbard <dhubbard@digium.com>

	* build_tools/make_version: updated build_tools to handle the
	  autotag directory structure changes; changes related to BE-353.
	  Patch by The Russell and reviewed by The Me.

2008-01-29 23:02 +0000 [r100973-101035]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Remove a memory leak from updating realtime
	  queues

	* apps/app_queue.c: Fixing an erroneous return value returned when
	  attempting to pause or unpause a queue member fails. Fixes
	  BE-366, thanks to John Bigelow for writing the patch.

2008-01-29 17:57 +0000 [r100934]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c: Don't forget to record the channel so we
	  know whether it is bridged or not later. (closes issue #11811)
	  Reported by: slavon

2008-01-29 17:43 +0000 [r100932]  Russell Bryant <russell@digium.com>

	* main/Makefile: Fix the last couple of issues related to building
	  from a path that contains spaces. (closes issue #11834)

2008-01-29 17:41 +0000 [r100930]  Jason Parker <jparker@digium.com>

	* channels/misdn_config.c: Initialize an array to 0s if config
	  option not specified. (closes issue #11860) Patches:
	  misdn_get_config.v1.diff uploaded by IgorG (license 20)

2008-01-29 17:21 +0000 [r100882-100922]  Russell Bryant <russell@digium.com>

	* Makefile: Use GNU make magic instead of shell magic to escape
	  spaces in the working directory. (related to issue #11834)

	* Makefile: Fix building Asterisk when the working path has spaces
	  in it. (closes issue #11834) Reported by: spendergrass Patched
	  by: me

2008-01-29 16:10 +0000 [r100835]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Allow zap groups above 30 to work properly.
	  (closes issue #11590) Reported by: tbsky

2008-01-29 10:36 +0000 [r100793]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed potential segfault in misdn show
	  channels CLI command

2008-01-29 08:26 +0000 [r100740]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: (closes issue #11736) Reported by: MVF
	  Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
	  oej, MVF, revolution (russellb: This was the showstopper for the
	  release.)

2008-01-28 21:02 +0000 [r100675]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
	  (went to 't' instead of 'T')

2008-01-28 20:55 +0000 [r100673]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
	  chan_vpb. It is alive and well.

2008-01-28 20:42 +0000 [r100672]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
	  greeting files into the database like we do with the others.
	  Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
	  by dimas (license 88)

2008-01-28 18:34 +0000 [r100626-100629]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: For some reason, the use of this strdupa()
	  is leading to memory corruption on freebsd sparc64. This trivial
	  workaround fixes it. (closes issue #10300, closes issue #11857,
	  reported by mattias04 and Home-of-the-Brave)

	* res/res_features.c: Fix a crash in ast_masq_park_call() (issue
	  #11342) Reported by: DEA Patches: res_features-park.txt uploaded
	  by DEA (license 3)

2008-01-28 18:23 +0000 [r100624]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Correct a comment which made little/no
	  sense.

2008-01-28 17:15 +0000 [r100581]  Russell Bryant <russell@digium.com>

	* main/channel.c, channels/chan_local.c,
	  include/asterisk/channel.h: Make some deadlock related fixes.
	  These bugs were discovered and reported internally at Digium by
	  Steve Pitts. - Fix up chan_local to ensure that the channel lock
	  is held before the local pvt lock. - Don't hold the channel lock
	  when executing the timing function, as it can cause a deadlock
	  when using chan_local. This actually changes the code back to be
	  how it was before the change for issue #10765. But, I added some
	  other locking that I think will prevent the problem reported
	  there, as well.

2008-01-27 21:59 +0000 [r100465]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
	  channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
	  channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
	  pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
	  the scheduler, ignoring the return value could possibly cause
	  memory to be accessed after it is freed, which causes all sorts
	  of random memory corruption. Instead, if a deletion fails, wait a
	  bit and try again (noting that another thread could change our
	  taskid value). (closes issue #11386) Reported by: flujan Patches:
	  20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
	  Tested by: Corydon76, flujan, stuarth`

2008-01-25 22:32 +0000 [r100418]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
	  now preferred that users of Voicetronix products use chan_zap in
	  combination with their zaptel drivers.

2008-01-25 21:24 +0000 [r100378]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: This would have never been true, since we're
	  passing (sizeof(req.data) - 1) as the len to recvfrom().

2008-01-24 21:57 +0000 [r100264]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/app.h: make these macros not assume that the
	  only other field in the structure is 'argc'... this is true when
	  someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
	  to define your own structure as long as it has the right fields

2008-01-24 17:22 +0000 [r100164]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Update main Asterisk copyright info to 2008

2008-01-24 16:41 +0000 [r100138]  Jason Parker <jparker@digium.com>

	* main/acl.c: Fix compilation on Solaris. (closes issue #11832)
	  Patches: bug-11832.diff uploaded by snuffy (license 35)

2008-01-23 21:07 +0000 [r99977-99978]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Second attempt. Don't change invitestate
	  when receiving 18x messages in CANCEL state. (issue #11736)
	  Reported by: MVF Patch by oej.

	* channels/chan_sip.c: Make sure we don't cancel destruction on
	  calls in CANCEL state, even if we get 183 while waiting for
	  answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
	  bug11736.txt uploaded by oej (license 306) Tested by: MVF

2008-01-23 20:25 +0000 [r99975]  Mark Michelson <mmichelson@digium.com>

	* apps/app_externalivr.c: Fixing a typo.

2008-01-23 17:46 +0000 [r99923]  Russell Bryant <russell@digium.com>

	* apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
	  beginning of a list of channels to potentially spy on. However,
	  if there were no matching channels, it would beep at you over and
	  over, which is pretty annoying. Now, it will only beep once in
	  the case that there are no channels to spy on, but it will still
	  beep again once it reaches the beginning of the channel list
	  again. (closes issue #11738, patched by me)

2008-01-23 16:18 +0000 [r99878]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: These flag tests were illogical. They were
	  testing sip_peer flags on a sip_pvt. Thanks to Russell for
	  helping to get this odd problem figured out.

2008-01-23 04:31 +0000 [r99718-99777]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_voicemail.c: When we reset the password via an external
	  command, we should also reset the password stored in the
	  in-memory list, too (otherwise it doesn't really take effect).
	  (closes issue #11809) Reported by: davetroy Patches:
	  fix_externpass.diff uploaded by davetroy (license 384)

	* res/res_odbc.c: Oops, should have checked for a NULL obj, here,
	  too

	* main/acl.c: Just confirmed that all current platforms need this
	  header file

2008-01-22 20:56 +0000 [r99652]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Thanks to Russell's education I realize that
	  BUFSIZ has changed since I learned the C language over 20 years
	  ago... Resetting chan_sip to the size of BUFSIZ that I expected
	  in my old head to avoid to heavy memory allocations on some
	  systems.

2008-01-22 20:34 +0000 [r99643]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Fix the defines for OS X (and Solaris, too)

2008-01-22 17:41 +0000 [r99592-99594]  Olle Johansson <oej@edvina.net>

	* channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
	  apps/app_followme.c: Add more dependencies on chan_local and add
	  a note to the description of chan_local so that people don't
	  disable it in menuselect just to clean up.

	* apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
	  still runs without chan_local, but will be missing forwarding
	  functionality.

2008-01-22 16:54 +0000 [r99540]  Tilghman Lesher <tlesher@digium.com>

	* main/acl.c: Ensure that we can get an address even when we don't
	  have a default route. (closes issue #9225) Reported by: junky
	  Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
	  (license 14) Tested by: oej, loloski, sergee

2008-01-22 15:08 +0000 [r99501]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Cleaning up some documentation that led to
	  confusion in a bug report

2008-01-21 23:55 +0000 [r99426]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Fixing an issue wherein monitoring local
	  channels was not possible. During a channel masquerade, the
	  monitors on the two channels involved are swapped. In 99% of the
	  cases this results in the desired effect. However, if monitoring
	  a local channel, this caused the monitor which was on the local
	  channel to get moved onto a channel which is immediately hung up
	  after the masquerade has completed. By swapping the monitors
	  prior to the masquerade, we avoid the problem by tricking the
	  masquerade into placing the monitor back onto the channel where
	  we want it. During the investigation of the issue, the channel's
	  monitor was the only thing that was swapped in such a manner
	  which did not make sense to have done. All other variable
	  swapping made sense.

2008-01-21 18:11 +0000 [r99341]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample,
	  include/asterisk/res_odbc.h: Permit the user to specify number of
	  seconds that a connection may remain idle, which fixes a crash on
	  reconnect with the MyODBC driver. (closes issue #11798) Reported
	  by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
	  uploaded by Corydon76 (license 14) Tested by: mvanbaak

2008-01-21 16:01 +0000 [r99301]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Bump the buffer size for Via headers up to
	  512. There are some exceptionally large Via headers out there.
	  (closes issue #11783) Reported by: ofirroval

2008-01-19 10:05 +0000 [r99187]  Russell Bryant <russell@digium.com>

	* main/slinfactory.c: Fix a couple of memory leaks with frame
	  handling. Specifically, ast_frame_free() needed to be called on
	  the frame that came from the translator to signed linear.

2008-01-18 22:57 +0000 [r99127]  Joshua Colp <jcolp@digium.com>

	* include/asterisk/channel.h: Remove the __ in front of the unused
	  variable. This causes some compilers to freak out.

2008-01-18 21:37 +0000 [r99079-99081]  Russell Bryant <russell@digium.com>

	* include/asterisk/translate.h, main/frame.c: Revert adding the
	  packed attribute, as it really doesn't make sense why that would
	  do any good. Fix the real bug, which is to do the check to see if
	  the frame came from a translator at the beginning of
	  ast_frame_free(), instead of at the end. This ensures that it
	  always gets checked, even if none of the parts of the frame are
	  malloc'd, and also ensures that we aren't looking at free'd
	  memory in the case that it is a malloc'd frame. (closes issue
	  #11792, reported by explidous, patched by me)

	* include/asterisk/translate.h: Since we're relying on the offset
	  between the frame and the beginning of the translator pvt struct,
	  set the packed attribute to make sure we get to the right place.
	  (potential fix for issue #11792)

2008-01-18 17:13 +0000 [r99032]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: This should at least temporarily fix a
	  problem where the 't' Dial option is incorrectly passed to the
	  transferee when built-in attended transfers are used. There is
	  still a problem with 'T', but better to fix some problems than no
	  problems while we work on it. (closes issue #7904) Reported by:
	  k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
	  (license 138) Tested by: sergee, otherwiseguy

2008-01-17 23:42 +0000 [r99007-99014]  Pari Nannapaneni <paripurnachand@digium.com>

	* configs/cdr.conf.sample: doh! revert a revert of a revert
	  (changed by mistake in 99010)

	* main/manager.c, configs/cdr.conf.sample: missed that one while
	  reverting

	* main/manager.c: reverting 99001 - We need the Max-Age for
	  extending the life of cookie mansession_id

2008-01-17 22:37 +0000 [r99004]  Russell Bryant <russell@digium.com>

	* main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
	  Have IAX2 optimize the codec translation path just like chan_sip
	  does it. If the caller's codec is in our codec list, move it to
	  the top to avoid transcoding. (closes issue #10500) Reported by:
	  stevedavies Patches: iax-prefer-current-codec.patch uploaded by
	  stevedavies (license 184) iax-prefer-current-codec.1.4.patch
	  uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
	  sheldonh

2008-01-17 21:31 +0000 [r99001]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: we should only send the Set-Cookie header to the
	  browser on the first response after creating a manager session,
	  not on every response (doing so causes the browser to clear any
	  local cookies it may have associated with the session)

2008-01-17 16:19 +0000 [r98991]  Jason Parker <jparker@digium.com>

	* configs/zapata.conf.sample: Add a clarification about the
	  immediate= option of zapata.conf Issue 11784, patch by klaus3000.

2008-01-16 22:36 +0000 [r98982]  Russell Bryant <russell@digium.com>

	* .cleancount, include/asterisk/channel.h: Add an unused pointer to
	  the ast_channel struct. This makes the ast_channel structure
	  retain the same size as it had in previous 1.4 releases. Also,
	  all of the offsets for members in the structure are still the
	  same (except for the two pointers that got replaced for the new
	  spy/whisper architecture.)

2008-01-16 20:34 +0000 [r98966-98973]  Joshua Colp <jcolp@digium.com>

	* .cleancount: Bump up cleancount due to previous commit that
	  changed the channel structure.

	* apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
	  main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
	  (added), main/Makefile, include/asterisk/chanspy.h (removed),
	  include/asterisk/channel.h, main/audiohook.c (added): Replace
	  current spy architecture with backport of audiohooks. This should
	  take care of current known spy issues.

	* channels/chan_iax2.c: Add missing NULLs at end of two
	  ast_load_realtimes. (closes issue #11769) Reported by: tequ
	  Patches: chaniax.patch uploaded by dimas (license 88)

2008-01-16 17:20 +0000 [r98964]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_local.c: Fix a deadlock in chan_local in
	  local_hangup. There was contention because the local_pvt was held
	  and it was attempting to lock a channel, which is the incorrect
	  locking order. (closes issue #11730) Reported by: UDI-Doug
	  Patches: 11730.patch uploaded by putnopvut (license 60) Tested
	  by: UDI-Doug

2008-01-16 15:08 +0000 [r98951-98960]  Joshua Colp <jcolp@digium.com>

	* main/dial.c: Introduce a lock into the dialing API that protects
	  it when destroying the structure. (closes issue #11687) Reported
	  by: callguy Patches: 11687.diff uploaded by file (license 11)

	* main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
	  issue #11777) Reported by: tootai

	* channels/chan_sip.c: Don't drop the old record route information
	  when dealing with packets related to a reinvite. (closes issue
	  #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
	  uploaded by kebl0155 (license 356)

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
	  configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
	  versions use a separate library for some things so we need to use
	  it if present in codec_speex. (closes issue #11693) Reported by:
	  yzg

2008-01-15 23:50 +0000 [r98943-98946]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Change a buffer in check_auth() to be a
	  thread local dynamically allocated buffer, instead of a massive
	  buffer on the stack. This fixes a crash reported by Qwell due to
	  running out of stack space when building with LOW_MEMORY defined.
	  On a very related note, the usage of BUFSIZ in various places in
	  chan_sip is arbitrary and careless. BUFSIZ is a system specific
	  define. On my machine, it is 8192, but by definition (according
	  to google) could be as small as 256. So, this buffer in
	  check_auth was 16 kB. We don't even support SIP messages larger
	  than 4 kB! Further usage of this define should be avoided, unless
	  it is used in the proper context.

	* main/rtp.c, include/asterisk/translate.h, main/frame.c,
	  main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
	  codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
	  some memory access errors pointed out by the valgrind2.txt output
	  on issue #11698. The issue here is that it is possible for an
	  instance of a translator to get destroyed while the frame
	  allocated as a part of the translator is still being processed.
	  Specifically, this is possible anywhere between a call to
	  ast_read() and ast_frame_free(), which is _a lot_ of places in
	  the code. The reason this happens is that the channel might get
	  masqueraded during this time. During a masquerade, existing
	  translation paths get destroyed. So, this patch fixes the issue
	  in an API and ABI compatible way. (This one is for you,
	  paravoid!) It changes an int in ast_frame to be used as flag
	  bits. The 1 bit is still used to indicate that the frame contains
	  timing information. Also, a second flag has been added to
	  indicate that the frame came from a translator. When a frame with
	  this flag gets released and has this flag, a function is called
	  in translate.c to let it know that this frame is doing being
	  processed. At this point, the flag gets cleared. Also, if the
	  translator was requested to be destroyed while its internal frame
	  still had this flag set, its destruction has been deffered until
	  it finds out that the frame is no longer being processed.
	  Admittedly, this feels like a hack. But, it does fix the issue,
	  and I was not able to think of a better solution ...

2008-01-15 20:08 +0000 [r98894-98934]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Based on the boundary found move over the
	  correct amount. (closes issue #11750) Reported by: tasker

	* channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
	  in the multipart mixed content type. (closes issue #11750)
	  Reported by: tasker

2008-01-14 20:59 +0000 [r98849]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Adding in appropriate unlocks for the locks
	  I added. Thanks to joetester on IRC for pointing this out.

2008-01-14 17:38 +0000 [r98774]  Russell Bryant <russell@digium.com>

	* main/translate.c: Revert a change that introduces an unacceptable
	  performance hit and is causing memory leaks ... (from rev 97973)

2008-01-14 16:35 +0000 [r98733-98737]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing another compilation error. I'm a bit off
	  today :(

	* apps/app_queue.c: Oops. Last commit had compilation error.

	* apps/app_queue.c: Adding explicit defaults for missing options to
	  init_queue. This is necessary because if a user either removes or
	  comments one of these options and reloads their queues, the
	  option will not reset to its default, instead maintaining the
	  value from prior to the reload. Thanks to John Bigelow for
	  pointing this error out to me.

2008-01-12 00:05 +0000 [r98467]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Add a connection timeout attribute, as that was
	  what was intended with the login timeout, but ODBC divides it up
	  into 2 different timeouts. (Closes issue #11745)

2008-01-11 22:46 +0000 [r98390]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
	  (closes issue #11646) Reported by: caio1982 Patches:
	  dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
	  dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
	  by: caio1982, mvanbaak

2008-01-11 21:28 +0000 [r98372]  Pari Nannapaneni <paripurnachand@digium.com>

	* main/http.c: Comment explaining how to force browser to always
	  read some html files from server.

2008-01-11 19:51 +0000 [r98317-98325]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If the incoming RTP stream changes codec force the
	  bridge to break if the other side does not support it. (closes
	  issue #11729) Reported by: tsearle Patches:
	  new_codec_patch_udiff.patch uploaded by tsearle (license 373)

	* res/res_agi.c: If the channel is hungup during RECORD FILE send a
	  result code of -1 to be uniform with everything else. (closes
	  issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
	  by davevg (license 209)

2008-01-11 19:10 +0000 [r98315]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Properly report the hangup cause as no answer
	  when someone does not answer (closes issue #10574, reported by
	  boch, patched by moy)

2008-01-11 18:25 +0000 [r98266]  Tilghman Lesher <tlesher@digium.com>

	* codecs/gsm/Makefile: Add another exception (which doesn't work)
	  for -march optimization flag. Reported by: thomasmebes Patch by:
	  tilghman (Closes issue #11563)

2008-01-11 18:25 +0000 [r98265]  Russell Bryant <russell@digium.com>

	* doc/security.txt, main/asterisk.c, configure,
	  include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
	  makeopts.in: Backport the ability to set the ToS bits on Linux
	  when not running as root. Normally, we would not backport
	  features into 1.4, but, I was convinced by the justification
	  supplied by the supplier of this patch. He pointed out that this
	  patch removes a requirement for running as root, thus reducing
	  the potential impacts of security issues. (closes issue #11742)
	  Reported by: paravoid Patches: libcap.diff uploaded by paravoid
	  (license 200)

2008-01-11 17:22 +0000 [r98219]  Joshua Colp <jcolp@digium.com>

	* apps/app_followme.c: Ensure the return value of ast_bridge_call
	  is passed back up as the application return value. This is needed
	  for transfers to function so the PBX core knows to continue
	  execution. (closes issue #10327) Reported by: kkiely

2008-01-11 15:52 +0000 [r98164]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Back out changes from revision 97077, since
	  it wasn't perfect

2008-01-11 03:39 +0000 [r97976-98082]  Russell Bryant <russell@digium.com>

	* main/frame.c: Fix samples vs. length calculations for g722

	* main/translate.c: Simplify this code with a suggestion from Luigi
	  on the asterisk-dev list. Instead of using is16kHz(), implement a
	  format_rate() function.

	* main/translate.c: Fix various timing calculations that made
	  assumptions that the audio being processed was at a sample rate
	  of 8 kHz.

2008-01-10 23:08 +0000 [r97973]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c, main/translate.c: 1) When we get a
	  translated frame out, clone it, because if the translator pvt is
	  freed before we use the frame, bad things happen. 2) Getting a
	  failure from ast_sched_delete means that the schedule ID is
	  currently running. Don't just ignore it. (Closes issue #11698)

2008-01-10 21:57 +0000 [r97925]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Let us leave a voicemail for ourself if we
	  have logged into VoiceMailMain and chosen to leave a message.
	  (closes issue #11735, reported and patched by jamessan)

2008-01-10 21:37 +0000 [r97849-97889]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
	  same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
	  overrode the normally generate yyfree func with our own version
	  that checks the pointer for non-null before passing to free().
	  Also takes care of a little problem with 2.5.33 and the use of
	  the __STDC_VERSION__ macro.

	* main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
	  fix for 2 things: a problem Terry was having in OSX with null
	  pointers, which was my fault, as I probably forgot to run the sed
	  script last time I made mods. So, I moved the fix into the flex
	  input itself. Then, I found when I used flex 2.5.33, that it was
	  using __STDC_VERSION__, and that's not real good; so I added back
	  in a DIFFERENT sed script to fix that little mess. Tested
	  everything, a couple different ways. Hope I did no harm, at the
	  least.

2008-01-10 20:12 +0000 [r97847]  Jason Parker <jparker@digium.com>

	* include/asterisk/frame.h: Fix a comment that is no longer true.

2008-01-10 16:19 +0000 [r97734-97753]  Russell Bryant <russell@digium.com>

	* pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
	  pbx/kdeconsole_main.cc (removed): Remove other remnants of
	  pbx_kdeconsole

	* pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
	  configure, include/asterisk/autoconfig.h.in, configure.ac,
	  makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
	  worked in ages, and nobody has complained. (closes issue #11706,
	  reported by caio1982)

2008-01-10 15:07 +0000 [r97697]  Joshua Colp <jcolp@digium.com>

	* funcs/func_groupcount.c: Don't try to copy the category from the
	  group if no category exists. (closes issue #11724) Reported by:
	  IgorG Patches: group_count.v1.patch uploaded by IgorG (license
	  20)

2008-01-09 23:01 +0000 [r97640-97645]  Russell Bryant <russell@digium.com>

	* pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
	  messages

	* pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
	  actually load on my system still (related to issue #11706)

2008-01-09 20:28 +0000 [r97618-97622]  Jason Parker <jparker@digium.com>

	* main/cli.c: Correctly display a message if a command could not be
	  found. Also fix a comment which may have led to this happening.
	  Issue 11718, reported by kshumard.

	* main/cli.c: Fix some locking and return value funkiness. We
	  really shouldn't be unlocking this lock inside of a function,
	  unless we locked it there too.

2008-01-09 18:48 +0000 [r97575]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
	  smaller functions this time

2008-01-09 18:02 +0000 [r97529]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Fix saying the parking space number to the
	  caller doing the parking ...

2008-01-09 17:21 +0000 [r97491]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: report the same message whether Zaptel does
	  not have transcoder support loaded or no transcoders were found

2008-01-09 16:44 +0000 [r97489]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
	  function. As underlined in issue #10437 by Josh, we need to
	  prevent a possible memory leak. We only set the name part of the
	  caller id, the number part is not relevant when dealing with
	  JIDs. Closes issue #11549.

2008-01-09 16:11 +0000 [r97450]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Don't do conferencing totally in Zaptel if
	  Monitor is running on the channel. (closes issue #11709) Reported
	  by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
	  (license 371)

2008-01-09 15:43 +0000 [r97410-97448]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: pass the right variable to get an error
	  string... oops

	* channels/chan_zap.c: add error number output to ioctl failure
	  messages to help with debugging

2008-01-09 00:44 +0000 [r97350]  Tilghman Lesher <tlesher@digium.com>

	* main/cli.c, main/editline/readline.c: Allow filename completion
	  on zero-length modules, remove a memory leak, remove a file
	  descriptor leak, and make filename completion thread-safe.
	  Patched and tested by tilghman. (Closes issue #11681)

2008-01-09 00:17 +0000 [r97206-97308]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: use the \retval doxygen command properly

	* apps/app_queue.c: Part 1 of N of adding doxygen comments to
	  app_queue. I picked some of the most common functions used (which
	  also happen to be some the biggest/ugliest functions too) to
	  document first. I'm pretty new to doxygen so criticism is
	  welcome.

	* apps/app_queue.c: Some coding guidelines-related cleanup

2008-01-08 20:48 +0000 [r97195]  Joshua Colp <jcolp@digium.com>

	* channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
	  (closes issue #11443) Reported by: eferro Patches:
	  dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
	  337)

2008-01-08 20:47 +0000 [r97194]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c, main/utils.c: Increase constants to where
	  we're less likely to hit them while debugging. (Closes issue
	  #11694)

2008-01-08 20:42 +0000 [r97192]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Making some changes designed to not allow
	  for a corrupted mailstream for a vm_state. 1. Add locking to the
	  vm_state retrieval functions so that no linked list corruption
	  occurs. 2. Make sure to always grab the persistent vm_state when
	  mailstream access is necessary. 3. Correct an incorrect return
	  value in the init_mailstream function. (closes issue #11304,
	  reported by dwhite)

2008-01-08 19:53 +0000 [r97093-97152]  Joshua Colp <jcolp@digium.com>

	* funcs/func_groupcount.c: If no group has been provided to the
	  GROUP_COUNT dialplan function then use the first one specific to
	  the channel. (closes issue #11077) Reported by: m4him

	* apps/app_queue.c: Make app_queue calls work with directed pickup.
	  (closes issue #11700) Reported by: jbauer

2008-01-08 18:02 +0000 [r97077]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
	  found in issue #11386, but not completely closing that issue.

2008-01-07 20:47 +0000 [r96884-96932]  Russell Bryant <russell@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 96931 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
	  2 lines Change misery.digium.com to pbx.digium.com ........

	* res/res_smdi.c: Don't crash if something happens when setting up
	  an SMDI interface and it gets destroyed before the SMDI port
	  handling thread gets created.

2008-01-07 14:34 +0000 [r96797-96815]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Indentation fix, makes the code easier to read

	* res/res_jabber.c: Compute the base64 value over the
	  [authzid]\0authcid\0password string, thus excluding the trailing
	  NULL byte. This change has already been committed to trunk, see
	  #11644.

2008-01-05 02:09 +0000 [r96644]  Russell Bryant <russell@digium.com>

	* main/devicestate.c: Don't pass an empty string as the device
	  name.

2008-01-04 23:03 +0000 [r96575]  Tilghman Lesher <tlesher@digium.com>

	* main/devicestate.c: Fix the problem of notification of a device
	  state change to a device with a '-' in the name. Could probably
	  do with a better fix in trunk, but this bug has been open way too
	  long without a better solution. Reported by: stevedavies Patch
	  by: tilghman (Closes issue #9668)

2008-01-04 22:55 +0000 [r96573]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Properly continue in the dialplan if using
	  PARKINGEXTEN and the slot is full. Issue 11237, patch by me.

2008-01-04 19:27 +0000 [r96525]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: If you change the bindaddr in sip.conf to a
	  non-bound address and reload, sip goes kablooie. Reported and
	  patched by: one47 (Closes issue #11535)

2008-01-04 16:19 +0000 [r96394-96449]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Make use of the temporary channel pointer
	  while the pvt is unlocked. (closes issue #11675) Reported by:
	  flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
	  uploaded by flefoll (license 244)

	* channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
	  been destroyed before we get to this point (closes issue #11672,
	  reported by snuffy, patched by me)

2008-01-03 21:37 +0000 [r96318]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c: Missed initialization caused crash.
	  Reported and fixed by: tiziano (Closes issue #11671)

2008-01-03 12:12 +0000 [r96198-96199]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: make sure frame is completely clean,
	  before we send it to asterisk as DTMF. If we don't make it clean,
	  it happens that one way audio occurs..

	* channels/chan_misdn.c: when overlapdial was used and no number
	  was dialed, the call was dropped, now we just jump into the s
	  extension, which makes a lot more sense.

2008-01-02 23:46 +0000 [r96102]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: We need to reset the membername to NULL on each
	  iteration of this loop, otherwise the result is that multiple
	  members can have the same name, since the variable was not reset
	  on each iteration of the loop.

2008-01-02 22:14 +0000 [r96020-96024]  Russell Bryant <russell@digium.com>

	* pbx/pbx_config.c: Convert locks of the contexts list in
	  pbx_config to the appropriate rdlock or wrlock

	* pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
	  list.

	* apps/app_macro.c: app_macro only needs a rdlock on the contexts
	  list.

2008-01-02  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.17 released.

2008-01-02 20:24 +0000 [r95946]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Allocate a SIP refer structure when
	  performing a transfer using BYE with Also so that the transfer
	  information is properly stored. (AST-2008-028) (closes issue
	  #11637) Reported by: greyvoip

2008-01-02 17:51 +0000 [r95890]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: A change to improve the accuracy of queue
	  logging in the case where a member does not answer during the
	  specified timeout period. Prior to this change, there was a small
	  chance that the member name recorded in this case would be blank.
	  Also prior to this change, if using the ringall strategy, if no
	  one answered the call during the specified timeout, the member
	  name listed in the queue log would randomly be one of the members
	  that was rung. (closes issue #11498, reported and tested by
	  hloubser, patched by me)

2007-12-31 23:43 +0000 [r95577]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Avoiding a potentially bad locking situation.
	  ast_merge_contexts_and_delete writelocks the conlock, then calls
	  ast_hint_extension, which attempts to readlock the same lock.
	  Recursion with read-write locks is dangerous, so the inner lock
	  needs to be removed. I did this by copying the "guts" of
	  ast_hint_extension into ast_merge_contexts_and_delete (sans the
	  extra lock). (this change is inspired by the locking problems
	  seen in issue #11080, but I have no idea if this is the
	  problematic area experienced by the reporters of that issue)

2007-12-31 20:27 +0000 [r95470]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Allow the default "0" to be returned if the
	  STAT fails (Closes issue #11659)

2007-12-28 18:24 +0000 [r95191]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Remove duplicate increment of the header
	  count in the add_header() function. (closes issue #11648)
	  Reported by: makoto Patch provided by sergee, committed patch by
	  me, inspired by comments from putnopvut

2007-12-28 00:16 +0000 [r95095]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: I found a bug while browsing the queue code and
	  managed to reproduce it in a small setup. If a queue uses the
	  ringall strategy, it was possible through unfortunate coincidence
	  for a single member at a given penalty level to make app_queue
	  think that all members at that penalty level were unavailable and
	  cause the members at the next penalty level to be rung. With this
	  patch, we will only move to the next penalty level if ALL the
	  members at a given penalty level are unreachable.

2007-12-27 21:40 +0000 [r95024]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't report a syntax error when an empty string
	  is passed to ast_get_group. Just return 0. (closes issue #11540)
	  Reported by: tzafrir Patches: group_empty.diff uploaded by
	  tzafrir (license 46) -- slightly changed by me

2007-12-27 20:09 +0000 [r94977]  Mark Michelson <mmichelson@digium.com>

	* main/io.c: Fixing a typo in a comment.

2007-12-27 17:32 +0000 [r94905-94924]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Include types.h in chan_h323 as without it
	  it can not be compiled on some operating systems like FreeBSD to
	  name one. (closes issue #11585) Reported by: sobomax Patches:
	  chan_h323.c.diff uploaded by sobomax (license 359)

	* channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
	  set or not on the dialog. It is possible for it to be a pointer
	  to NULL. (closes issue #11557) Reported by: FuriousGeorge

2007-12-27 15:16 +0000 [r94828-94831]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Now that the contexts lock is a read/write lock, it
	  should not be locked here in ast_hint_state_changed(). This makes
	  it get locked recursively which now causes a deadlock. (closes
	  issue #11080, thanks to callguy for the access to a deadlocked
	  machine)

	* include/asterisk/translate.h, main/translate.c: Use the constant
	  that I really meant to use here ...

	* main/translate.c: Change ast_translator_best_choice() to only pay
	  attention to audio formats. This fixes a problem where Asterisk
	  claims that a translation path can not be found for channels
	  involving video. (closes issue #11638) Reported by: cwhuang
	  Tested by: cwhuang Patch suggested by cwhuang, with some
	  additional changes by me.

2007-12-27 01:01 +0000 [r94824]  Kevin P. Fleming <kpfleming@digium.com>

	* main/manager.c: make this comment explain the situation in an
	  even more explicit fashion

2007-12-26 20:43 +0000 [r94808]  Tilghman Lesher <tlesher@digium.com>

	* main/manager.c: Workaround for what is probably a glibc bug (but
	  we'll see this crop up again and again, if we don't add the
	  workaround). Reported by: rolek Patch by: tilghman (Closes issue
	  #11601, closes issue #11426)

2007-12-26 19:04 +0000 [r94789-94801]  Russell Bryant <russell@digium.com>

	* main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
	  was already set before starting autoservice, remember it and
	  ensure that the channel has the same setting when autoservice
	  gets stopped. (pointed out by d1mas, patched up by me)

	* main/autoservice.c: When a channel is in autoservice, mark a flag
	  on the channel that says that we only care about the END of a
	  digit. That way, no magic digit emulation stuff will happen when
	  all we're doing is queueing up END frames.

	* res/res_features.c: Don't try to send a parked call back to
	  itself. (closes issue #11622, reported by djrodman, patched by
	  me)

	* main/autoservice.c: Don't store DTMF BEGIN frames while a channel
	  is in autoservice. It's just going to make ast_read() do a lot of
	  extra work when the channel comes back out of autoservice.
	  (closes issue #11628, patched by me)

	* Makefile: List include/asterisk/version.h as a .PHONY target
	  because we want the commands listed for this target to be
	  executed regardless of whether the file exists or not. This fixes
	  having the version not up to date when running from svn. (closes
	  issue #11619, reported by plack, fixed by me)

2007-12-25 02:27 +0000 [r94769]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: file says... build on the builders.

2007-12-24 19:36 +0000 [r94763-94767]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c: Race: we need to wait to queue a NewChannel event
	  until after the channel is inserted into the channel list. The
	  reason is because some manager users immediately queue requests
	  from the channel when they see that event and are confused when
	  Asterisk reports no such channel. (Closes issue #11632)

	* channels/chan_sip.c: More deadlock avoidance code (this time
	  between sip_monitor and sip_hangup) Reported by: apsaras Patch
	  by: tilghman (Closes issue #11413)

	* channels/chan_sip.c: Another bit of bad logic in realtime_peer
	  Reported by: dimas Patch by: dimas (Closes issue #11631)

2007-12-23 01:21 +0000 [r94660]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: Argh... I suppose third time's the charm.

2007-12-21 20:21 +0000 [r94468-94543]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Bunch of coding guidelines cleanup

	* apps/app_voicemail.c: Better quota support for using IMAP storage
	  voicemail (closes issue #11415, reported by jaroth) (closes issue
	  #11152, reported by selsky) Patch provided by jaroth

	* apps/app_voicemail.c: The mail_copy c-client function does not
	  expect a full imap mailbox string, just the name of the mailbox.
	  (closes issue #11419, reported and patched by jaroth, with
	  additional patchwork from me)

	* main/dial.c: Since we are freeing list elements within a list
	  traversal, we need to use the safe traversal and remove the item
	  from the list before freeing it. (closes issue 11612, reported by
	  dtyoo)

2007-12-21 16:37 +0000 [r94466]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
	  a read/write lock to resolve a deadlock. This has a nice side
	  benefit of improving performance. :) (closes issue #11609)
	  (closes issue #11080)

2007-12-21 16:11 +0000 [r94420-94464]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing a debug message I accidentally just
	  committed

	* main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
	  dates and times. Also some coding guidelines cleanup. (closes
	  issue #11599, reported and patched by caio1982, coding guidelines
	  cleanup by me)

2007-12-21 15:07 +0000 [r94418]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: Fix for restart-as-user problem reported via the
	  -dev list

2007-12-20  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.16.2 released.

2007-12-20 20:22 +0000 [r94215-94256]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
	  5 lines Fix another potential seg fault ... (closes issue #11606)
	  Reported by: dimas ........

	* channels/chan_zap.c: Fix a deadlock in d-channel handling in
	  chan_zap. This deadlock was introduced by the fix to ensure that
	  channels are properly locked when handling channel variables.
	  There were sections of this code where the channel pvt was locked
	  before the channel lock, when in fact it _must_ be the other way
	  around. (closes issue #11582) Reported by: bugi

	* /: Blocked revisions 94214 via svnmerge ........ r94214 | russell
	  | 2007-12-20 11:29:11 -0600 (Thu, 20 Dec 2007) | 2 lines Fix a
	  couple of places where it's possible to dereference a NULL
	  pointer. ........

2007-12-19 23:02 +0000 [r94122]  Mark Michelson <mmichelson@digium.com>

	* res/res_monitor.c: Sox versions 13.0.0 and newer do not have
	  "soxmix" and instead use sox -m. res_monitor needs to use this if
	  the user does not have soxmix. (closes issue #11589, reported by
	  amessina, patch inspired by amessina but with a flourish from me)

2007-12-19 22:48 +0000 [r94077]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Check
	  for the existence of the soxmix application on the target
	  platform and have the result available in autoconfig.h. (part of
	  issue #11589)

2007-12-19  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.16.1 released.

2007-12-19 17:29 +0000 [r93955]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
	  NULL.

2007-12-19 17:04 +0000 [r93949]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
	  defined (Closes issue #11602)

2007-12-18 22:42 +0000 [r93764]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: FreeBSD also does not have byte swap
	  functions. Issue 11586, patch by sobomax.

2007-12-18  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.16 released.

2007-12-18 18:45 +0000 [r93668-93676]  Tilghman Lesher <tlesher@digium.com>

	* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
	  93667 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
	  | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........

2007-12-18 17:02 +0000 [r93625]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Rework deadlock avoidance used in ast_write,
	  since it meant that agent channels which were being monitored had
	  one audio file recorded and one empty audio file saved. (closes
	  issue #11529, reported by atis patched by me)

2007-12-17 22:56 +0000 [r93381-93420]  Jason Parker <jparker@digium.com>

	* main/translate.c: What was I thinking when I wrote this
	  masterpiece? -1 + 1 = 0.. who woulda thunk it?.

2007-12-17 22:28 +0000 [r93377]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Do not try to access information about a lock when
	  printing out a trylock attempt. It is possible for the lock that
	  it references to no longer be valid. This would have caused
	  segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
	  Reported by: callguy (closes issue #11100) Reported by: callguy

2007-12-17 21:12 +0000 [r93336]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/time.h: Today is tomorrow's yesterday, and
	  yesterday's tomorrow is today, and tomorrow's tomorrow is the day
	  after tomorrow, so who cares if you recycle anyway? If this
	  confuses you, that's nothing compared to what this fixes. ;-)

2007-12-17 19:53 +0000 [r93291]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: We need to create the directory for a
	  voicemail user even if they are using IMAP storage since
	  greetings are stored in the filesystem. (closes issue #11388,
	  reported by spditner, patch by me inspired by a patch by
	  spditner)

2007-12-17 18:05 +0000 [r93250]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: If a call is received with a called number
	  IE containing nothing go to the 's' extension. (closes issue
	  #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
	  uploaded by Corydon76 (license 14)

2007-12-17 07:21 +0000 [r93183]  Kevin P. Fleming <kpfleming@digium.com>

	* funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
	  res/Makefile, channels/Makefile, formats/Makefile: fix some
	  copy-and-paste leftovers

2007-12-17 07:15 +0000 [r93182]  Olle Johansson <oej@edvina.net>

	* channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
	  apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
	  dependencies on res_monitor and res_features. I wonder if
	  Asterisk can run at all without res_features. My guess is that
	  there's propably a lot of more modules and the core that depends
	  on it. Reported by: caio1982 (closes issue #11574)

2007-12-17 06:44 +0000 [r93180]  Kevin P. Fleming <kpfleming@digium.com>

	* formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
	  configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
	  formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
	  include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
	  configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
	  (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
	  http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
	  rizzo brought up some issues related to the way that the metadata
	  required for menuselect and the rest of the build system is
	  extracted from the source files. Since I had a few hours to kill
	  on an airplane today, I decided to improve this situation... so
	  now the system caches the extracted metadata and uses it to build
	  the menuselect 'tree' as much as it can. The result of this is
	  that when a single source file is changed, only the metadata for
	  that file needs to be extracted again, and the rest is used from
	  the cache files. I also reduced the number of forked processes
	  required to do the metadata extraction; it was actually possible
	  to do most of what we needed in the Makefiles themselves without
	  using any shell scripts at all! On my laptop, these changes
	  resulted in an 80% decrease in the time required for the
	  'menuselect.makeopts' automatic check to occur after editing a
	  single source file. While doing this work I also cleaned up a few
	  minor things in the Makefiles, adding a check for 'awk' to the
	  configure script and changed all remaining places we use 'grep'
	  or 'awk' to use the ones found by the configure script, and
	  changed the 'prep_tarball' script to build the menuselect
	  metadata so that tarballs of Asterisk will include it and won't
	  require the user to wait while it is extracted after unpacking.

2007-12-14 17:36 +0000 [r93000]  Russell Bryant <russell@digium.com>

	* main/config.c: There are a lot of existing systems that #include
	  non-existent files. So, to make the transition to treating this
	  as an error a bit less painless, just issue a huge error message
	  for now. Then, later, we can reinstate the code that treats it as
	  a failure. (Thanks to philippel for the feedback)

2007-12-14 15:16 +0000 [r92937]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Up the length of the format on the SIP
	  channel since it can now be rather long. (closes issue #11552)
	  Reported by: francesco_r

2007-12-14 15:05 +0000 [r92934]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
	  state setting and overlap_task thread starting.

2007-12-14 15:01 +0000 [r92933]  Tilghman Lesher <tlesher@digium.com>

	* res/res_agi.c: Change help documentation to match actual behavior
	  (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
	  (Closes issue #11548)

2007-12-14 01:24 +0000 [r92875]  Mark Michelson <mmichelson@digium.com>

	* include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
	  don't spam the CLI with messages about possible deadlocks.
	  Instead just print the intended single message every five
	  seconds. (closes issue 11537, reported and patched by dimas)

2007-12-13 21:28 +0000 [r92815]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Properly initialize polarity statuses, so
	  that they are detected properly. Reported by: julianjm Patch by:
	  julianjm (Closes issue #10238)

2007-12-13 20:13 +0000 [r92809]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Make application help text a little more clear about
	  the use of extensions in a filename.

2007-12-13 20:03 +0000 [r92803-92807]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Prevent another potential fd leak

	* apps/app_voicemail.c: Prevent a possible fd leak.

2007-12-13 00:11 +0000 [r92696]  Jason Parker <jparker@digium.com>

	* main/config.c, channels/chan_sip.c, channels/chan_h323.c,
	  channels/chan_iax2.c: If a typo is found in a config file, we
	  previous continued on with what was already loaded. We do not
	  want to do this (see bug below for details). This makes it so
	  that if a [ is found without a ], the entire config will fail,
	  and nothing in it will be loaded. Isue #10690.

2007-12-12 22:00 +0000 [r92656]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: emit a warning message when we drop a G.729B
	  CNG frame destined for the transcoder

2007-12-12 21:15 +0000 [r92617]  Jason Parker <jparker@digium.com>

	* apps/app_meetme.c: Don't increment user count until after name
	  has been recorded (if enabled). Issue 11048, tested by pep.

2007-12-12 19:40 +0000 [r92556]  Russell Bryant <russell@digium.com>

	* res/res_features.c: resolve compiler warning

2007-12-12 17:46 +0000 [r92510]  Mark Michelson <mmichelson@digium.com>

	* res/res_features.c: Correctly detect where a dynamic feature was
	  activated. Before this patch, the channel which initiated the
	  bridge was always assumed to have been the one which activated
	  the dynamic feature. This patch corrects this. (closes issue
	  #11529, reported and patched by nic_bellamy)

2007-12-12 16:52 +0000 [r92463]  Tilghman Lesher <tlesher@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac: Test
	  directly for the API that fixed AST-2007-026, to ensure that
	  older versions of PostgreSQL are no longer acceptable. (Closes
	  issue #11526)

2007-12-12 16:08 +0000 [r92443]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing an unused variable.

2007-12-11 19:51 +0000 [r92363]  Joshua Colp <jcolp@digium.com>

	* main/global_datastores.c: Fix potential memory leak with the
	  dialed interfaces list if another memory allocation fails.
	  (closes issue #11507) Reported by: eliel Patches:
	  global_datastores.c.patch uploaded by eliel (license 64)

2007-12-11 17:42 +0000 [r92323]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing autofill to be more accurate.
	  Specifically, if calls ahead of the current caller were ringing
	  members (but not yet bridged) there could be available members
	  and waiting callers who would not get matched up. The member
	  availability checker was correctly determining the number of
	  available members in this scenario, but the queue itself did not
	  parallelly reflect this status on the pending calls. This commit
	  corrects the issue. (closes issue #11459, reported by
	  equissoftware, patched by me)

2007-12-10 16:36 +0000 [r92204]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Add G729A as another possible payload name for G729.
	  Some devices use this instead of G729, which is perfectly normal
	  since the payload number itself is defined and can't be used by
	  anything else so the name doesn't matter that much. (closes issue
	  #11483) Reported by: revolution Patches: rtp.diff uploaded by
	  revolution (license 346)

2007-12-10 16:29 +0000 [r92202]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: If there are no members in a queue, then the
	  loop where the datastore for detecting duplicate dialed numbers
	  will be skipped, meaning the datastore isn't created. This means
	  that when we try to free it, there's a crash. This stops that
	  crash from occurring. (closes issue #11499, reported by slavon,
	  patched by eliel)

2007-12-10 16:13 +0000 [r92200]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: It is possible for nativeformats to contain
	  more then one codec, so print out multiple ones. (closes issue
	  #11366) Reported by: ovi

2007-12-10 14:04 +0000 [r92158]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Avoid reinvite race situations with two
	  Asterisks trying to reinvite each other in 1.4 and trunk. This
	  patch implements support for the 491 error code that Asterisk 1.4
	  generates on situations where we get an incoming INVITE and
	  already has one in progress. Thanks to mavetju for reporting and
	  to Raj Jain for an excellent explanation of the problem. Patch by
	  myself. Tested with 8 Asterisk servers connected to each other in
	  a training network. Closes issue #10481

2007-12-07 23:29 +0000 [r91890]  Jason Parker <jparker@digium.com>

	* main/dsp.c: We need to make sure we free the input frame if we
	  return a different frame in ast_dsp_process. Issue 11273, pointed
	  out by dimas, with a patch by eliel.

2007-12-07 22:30 +0000 [r91870]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: even though Asterisk explicitly requests that
	  endpoints using G.729 do *not* use Annex B (silence detection and
	  comfort noise generation) some do anyway; the transcoder card
	  interface does not currently work properly with CNG frames, so
	  trim off the CNG before sending the data

2007-12-07 21:24 +0000 [r91777-91830]  Russell Bryant <russell@digium.com>

	* main/utils.c: Make the lock protecting each thread's list of
	  locks it currently holds recursive. I think that this will fix
	  the situation where some people have said that "core show locks"
	  locks up the CLI. (related to issue #11080)

	* include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
	  code. The ast_mutex_init() function had the mutex attribute
	  object marked as static. This means that multiple threads
	  initializing locks at the same time could step on each other and
	  end up with improperly initialized locks. (found when tracking
	  down locking issues related to issue #11080)

	* include/asterisk/lock.h: I love fixing lock related errors in the
	  lock debugging code. That's about as ironic as it gets in
	  Asterisk programming land. Anyway, I spotted this bug while
	  trying to track down why systems are locking up and acting weird
	  in issue #11080. The mutex attribute object was marked as static
	  in this function when it should not have been.

	* apps/app_dial.c: * Add channel locking around datastore
	  operations that expect the channel to be locked. * Document why
	  we don't record Local channels in the dialed interfaces list. *
	  Remove the dialed variable as it isn't needed. * Restructure some
	  code for clarity and coding guidelines stuff

	* apps/app_queue.c: * Add channel locking around datastore
	  operations that expect the channel to be locked. * Document why
	  we don't record Local channels in the dialed interfaces list. *
	  Handle memory allocation failure. * Remove the dialed variable,
	  as it wasn't actually needed. * Tweak some formatting to conform
	  to coding guidelines.

	* main/autoservice.c: * Add a bit more of a verbose comment as to
	  why a hangup frame needs to be queued up if autoservice gets a
	  NULL return from ast_read(). * Make the process of queueing the
	  hangup frame more efficient by putting the frame where it is
	  going to end up and avoiding some locking and extra memory
	  allocations and freeing.

2007-12-07 15:39 +0000 [r91737]  Mark Michelson <mmichelson@digium.com>

	* main/autoservice.c: Hangups that happen during autoservice were
	  not processed appropriately. This is because a hangup actually
	  causes a NULL frame to be received, not a hangup frame. Queueing
	  a hangup if we receive a NULL frame during autoservice corrects
	  this problem (closes issue #11467, reported by jmls, patched by
	  me)

2007-12-07 02:51 +0000 [r91675-91693]  Russell Bryant <russell@digium.com>

	* apps/app_dial.c: Don't unlock the dialed_interfaces list until
	  we're done messing with the iterator.

	* apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
	  from Queue() and Dial() again. There was a slight flaw in the
	  code to prevent call forwards from looping that caused this
	  problem. (related to issue #11486)

	* apps/app_queue.c: Fix in an issue in the call forwarding handling
	  code that was causing crashes on every call into a queue. I'm not
	  entirely sure about the logic in this part of the code, so I want
	  to look at it some more tomorrow. However, this makes it safe and
	  keeps it from crashing. (closes issue #11486, reported by adamg,
	  patched by me)

2007-12-07 00:52 +0000 [r91637]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c: At the end of a call, when we're reporting, RTCP may
	  already be partially torn down, so check for NULL dereference
	  Reported by: blitzrage Patch by: tilghman (Closes issue #11450)

2007-12-06 20:25 +0000 [r91541]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: IMAP storage did not honor the maxmsg
	  setting in voicemail.conf, and it also had the possibility of
	  crashing if a user had more than 256 messages in their voicemail.
	  This patch kills two birds with one stone by adding maxmsg
	  support and also setting a hard limit on the number of messages
	  at 255 so that the crashes cannot happen. (closes issue #11101,
	  reported by Skavin, patched by me)

2007-12-06 19:11 +0000 [r91501]  Russell Bryant <russell@digium.com>

	* main/loader.c, include/asterisk/module.h: Add a new module flag
	  to indicate that a build sum is present. Modules built against
	  older Asterisk 1.4 headers will now load properly with just a
	  warning indicating that they are old and may cause problems.
	  (patch by paravoid)

2007-12-06 16:49 +0000 [r91439-91450]  Joshua Colp <jcolp@digium.com>

	* main/udptl.c: Fix various in the udptl implementation. It could
	  return empty modem frames, have an incorrect sequence number on
	  packets, and display the wrong sequence number in the debug
	  messages. (closes issue #11228) Reported by: Cache Patches:
	  udptl-4.patch uploaded by dimas (license 88)

	* channels/chan_sip.c: Add support for accepting and sending T.38
	  in the initial INVITE. (closes issue #9402) Reported by: thdei

2007-12-06 12:54 +0000 [r91366]  Olle Johansson <oej@edvina.net>

	* main/loader.c, include/asterisk/logger.h, main/logger.c: Make
	  sure logger is reloaded at general reload in the cli. (Discovered
	  during Asterisk training in Portugal)

2007-12-05 22:57 +0000 [r91273-91292]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Reverting extra stuff I didn't mean to
	  commit

	* apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
	  did not properly handle the case where only a label was provided.
	  This was due to the fact that the answering channel did not have
	  an extension set, so ast_parseable_goto would fail. This fix
	  eliminates the call to ast_parseable_goto on the answering
	  channel since it is a wasteful call. The answering channel and
	  the calling channel are both directed to the same extension and
	  context, just different priorities, so we can just copy the
	  values from the calling channel to the answering channel and
	  increment the answering channel's priority. (closes issue #11382,
	  reported by jon, patch by me with correction by jon)

2007-12-05 21:38 +0000 [r91237]  Tilghman Lesher <tlesher@digium.com>

	* sounds/Makefile: Upgrade to the latest version of extra sounds

2007-12-05 17:31 +0000 [r90967-91192]  Russell Bryant <russell@digium.com>

	* main/threadstorage.c: Make the lock in the threadstorage
	  debugging code untracked to avoid a deadlock on thread
	  destruction. (closes issue #11207) Reported by: ys Patches:
	  threadstorage.c.diff uploaded by ys (license 281) Also fixes an
	  open bug report: (closes issue #11446)

	* main/utils.c: When DEBUG_THREADS is enabled, we only have the
	  details about who is holding a lock that we are waiting on for a
	  mutex, not rwlocks. This should fix the problem where people have
	  reported "core show locks" crashing sometimes.

	* include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
	  reported as happening on Mac systems. It turns out that the
	  problem was the Mac version of the ast_atomic_fetchadd_int()
	  function. The Mac atomic add function returns the _new_ value,
	  while this function is supposed to return the old value. So, the
	  crashes happened on unreferencing objects. If the reference count
	  was decreased to 1, ao2_ref() thought that it had been decreased
	  to zero, and called the destructor. However, there was still an
	  outstanding reference around. (closes issue #11176) (closes issue
	  #11289)

	* include/asterisk/file.h, configure,
	  include/asterisk/autoconfig.h.in, configure.ac,
	  include/asterisk/compiler.h: Modify file.h to maintain API
	  compatibility with earlier versions. If a recent compiler is
	  being used, then a warning will show up for any modules still
	  using the old name "private" instead of "_private". (patch
	  suggested by paravoid)

	* main/pbx.c: Make some changes to some additions I made recently
	  for doing channel autoservice when looking up extensions. This
	  code was added to handle the case where a dialplan switch was in
	  use that could block for a long time. However, the way that I
	  added it, it did this for all extension lookups. However, lookups
	  in the in-memory tree of extensions should _not_ take long enough
	  to matter. So, move the autoservice stuff to be only around
	  executing a switch.

2007-12-04 17:28 +0000 [r90876]  Jason Parker <jparker@digium.com>

	* main/channel.c: If we fail to create a channel after allocating a
	  timing fd, we need to make sure to close it. Issue 11454, patch
	  by eliel.

2007-12-04 05:29 +0000 [r90798]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c: Fix build issue on the build cluster.

2007-12-03 23:50 +0000 [r90736-90753]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/compat.h: Solaris requires the inclusion of
	  sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
	  snuffy,tilghman (Closes issue #11430)

	* res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
	  NULL deref could result Reported by: xrg Patch by: tilghman
	  (Closes issue #11387)

2007-12-03 23:12 +0000 [r90735]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c, main/channel.c, main/global_datastores.c
	  (added), channels/chan_local.c, main/Makefile,
	  include/asterisk/channel.h, include/asterisk/global_datastores.h
	  (added), apps/app_queue.c: A big one... This is the merge of the
	  forward-loop branch. The main change here is that call-forwards
	  can no longer loop. This is accomplished by creating a datastore
	  on the calling channel which has a linked list of all devices
	  dialed. If a forward happens, then the local channel which is
	  created inherits the datastore. If, through this progression of
	  forwards and datastore inheritance, a device is attempted to be
	  dialed a second time, it will simply be skipped and a warning
	  message will be printed to the CLI. After the dialing has been
	  completed, the datastore is detached from the channel and
	  destroyed. This change also introduces some side effects to the
	  code which I shall enumerate here: 1. Datastore inheritance has
	  been backported from trunk into 1.4 2. A large chunk of code has
	  been removed from app_dial. This chunk is the section of code
	  which handles the call forward case after the channel has been
	  requested but before it has been called. This was removed because
	  call-forwarding still works fine without it, it makes the code
	  less error-prone should it need changing, and it made this set of
	  changes much less painful to just have the forwarding handled in
	  one place in each module. 3. Two new files, global_datastores.h
	  and .c have been added. These are necessary since the datastore
	  which is attached to the channel may be created and attached in
	  either app_dial or app_queue, so they need a common place to find
	  the datastore info. This approach was taken in case similar
	  datastores are needed in the future, there will be a common place
	  to add them.

2007-12-03 22:06 +0000 [r90696]  Jason Parker <jparker@digium.com>

	* apps/app_meetme.c: Make sure we always close the conference fd if
	  we have an open one. Issue 11383, reported by markmhy, patch by
	  eliel.

2007-12-03 20:59 +0000 [r90639]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_mgcp.c: Changing some bad logic when calculating
	  the interdigit timeout. (closes issue #11402, reported and
	  patched by eferro)

2007-12-03 20:51 +0000 [r90607]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Fix crash in ParkAndAnnounce application.
	  Issue #11436, reported by lytledd, patch by eliel.

2007-12-03 20:05 +0000 [r90548-90588]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Do not create a smoother for G723.1 frames, they need
	  to be left alone to their native 20/24 byte size.

	* .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
	  the indication currently playing on a channel when a masquerade
	  operation happens. (issue #BE-88)

2007-12-03 18:20 +0000 [r90546]  Jason Parker <jparker@digium.com>

	* channels/chan_iax2.c: Only log debug messages if debug is
	  enabled. Closes issue #11416, patch by casper.

2007-12-02 18:18 +0000 [r90470]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: The other day when I went through making
	  changes as a result of the ao2_link() change, I added some code
	  to set pointers to NULL after they were unreferenced. This
	  pointed out that in this place, the object was unreferenced
	  before the code was done using it. So, move the unref down a
	  little bit. (crash reported by jmls on IRC)

2007-12-02 09:34 +0000 [r90432]  Tilghman Lesher <tlesher@digium.com>

	* main/autoservice.c: Clarify the return value on autoservice.
	  Specifically, if you started autoservice and autoservice was
	  already on, it would erroneously return an error. Reported by:
	  adiemus Patch by: dimas (Closes issue #11433)

2007-11-30 19:26 +0000 [r90310-90348]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
	  apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
	  ao2_link(). Previously, in inherited a reference. Now, it
	  automatically increases the reference count to reflect the
	  reference that is now held by the container. This was done to be
	  more consistent with ao2_unlink(), which automatically releases
	  the reference held by the container. It also makes it so it is no
	  longer possible for a pointer to be invalid after ao2_link()
	  returns.

	* include/asterisk/astobj2.h: Add some notes on the behavior of
	  ao2_unlink() after a discussion with Tilghman

2007-11-30 14:43 +0000 [r90269]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix locking issues under one legged replaces
	  scenarios. (closes issue #11420) Reported by: irroot Patches:
	  chan_sip_oneleg.patch uploaded by irroot (license 52)

2007-11-30 00:16 +0000 [r90231]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_mgcp.c: Clear the DTMF buffer if the call times
	  out. (closes issue #11418, reported and patched by eferro)

2007-11-29  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.15 released.

2007-11-29 19:48 +0000 [r90166]  Tilghman Lesher <tlesher@digium.com>

	* cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
	  we use thread-safe escaping (Fixes AST-2007-026)

2007-11-29 19:38 +0000 [r90163]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This patch handles the case where a queue
	  member with a negative penalty is added via the manager. If a
	  negative value is submitted for a member penalty, we set it to 0.
	  (closes issue #11411, reported and patched by Laureano)

2007-11-29 19:24 +0000 [r90154-90160]  Tilghman Lesher <tlesher@digium.com>

	* res/res_config_pgsql.c: Properly escape input buffers (Fixes
	  AST-2007-025)

	* formats/format_g726.c, include/asterisk/file.h,
	  formats/format_wav.c, formats/format_pcm.c,
	  formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
	  formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
	  as a field name in a header file messes with C++ projects
	  Reported by: chewbacca Patch by: casper (Closes issue #11401)

	* sounds/Makefile: Upgrade the core sounds release version

2007-11-29 00:36 +0000 [r90142-90147]  Russell Bryant <russell@digium.com>

	* funcs/func_callerid.c: fix some formatting i accidentally changed

	* funcs/func_callerid.c, main/channel.c,
	  include/asterisk/channel.h: This set of changes is to make some
	  callerID handling thread-safe. The ast_set_callerid() function
	  needed to lock the channel. Also, the handlers for the CALLERID()
	  dialplan function needed to lock the channel when reading or
	  writing callerid values directly on the channel structure.

	* include/asterisk/file.h, main/file.c: Merge a change from
	  team/russell/chan_refcount ... This makes ast_stopstream()
	  thread-safe.

2007-11-28 22:59 +0000 [r90101]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
	  Reported by: eliel Patches: load_realtime.patch uploaded by eliel
	  (license 64)

2007-11-28 22:30 +0000 [r90098]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/users.conf.sample, main/manager.c: it is impossible to
	  set permissions for manager accounts created by users.conf
	  (reported internally, patched by me)

2007-11-28 22:08 +0000 [r89999-90059]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: Removing some seemingly pointless code. This sets a
	  channel variable for every priority executed in the dialplan if
	  you have debug set to anything non-zero. This seems pointless due
	  to the fact that these channel variables are not referenced
	  anywhere else in the code and their names are esoteric enough
	  that they would not be practical to reference in the dialplan.
	  Plus the fact that this behavior isn't documented anywhere means
	  that the change is not likely to cause any disruption. If
	  anything, this may actually cause a slight performance increase
	  if running with debug on. The motivating influence for this code
	  change is the eventwhencalled option for queues. If set to vars,
	  all channel variables will be output to the manager. These
	  unnecessary channel variables make the output a lot more
	  difficult to deal with.

	* apps/app_voicemail.c: Recording greetings when using IMAP storage
	  was causing zero-length files to be stored. Since greetings are
	  not retrieved from IMAP anyway, it is pointless to attempt
	  storing them there. (closes issue #11359, reported by spditner,
	  patched by me)

2007-11-28 00:20 +0000 [r89839-89893]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: - update documentation for
	  some of the goto functions to note that they handle locking the
	  channel as needed - update ast_explicit_goto() to lock the
	  channel as needed

	* main/autoservice.c: Don't do frame processing if ast_read()
	  returned NULL.

	* apps/app_queue.c: Instead of depending on the return value of
	  ast_true(), explicitly set the eventwhencalled variable to 1.

	* main/pbx.c: Don't start/stop autoservice in
	  pbx_extension_helper() unless a channel exists

2007-11-27 23:10 +0000 [r89837]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Two changes with regards to the
	  'eventwhencalled' option of queues.conf 1) Due to some signed vs.
	  unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
	  did exactly the same thing. Thus the sign change of the ast_true
	  call. 2) The vars2manager function overwrote a \n for every
	  channel variable it parsed, resulting in bizarre output for the
	  channel variables. This patch remedies this. (related to issue
	  #11385, however I'm not sure if this will actually be enough to
	  close it)

2007-11-27 21:45 +0000 [r89790]  Russell Bryant <russell@digium.com>

	* main/autoservice.c, main/pbx.c: Merge changes from
	  team/russell/autoservice_1.4 This set of changes fixes an issue
	  that was reported to me on IRC yesterday. The user, d1mas, was
	  using chan_zap for incoming calls and was having DTMF recognition
	  issues in some situations. Specifically, he noticed that the
	  problem occurred when using DISA or WaitExten. He also noticed
	  that when using Read, the problem did not occur. His system also
	  used DUNDi for dialplan lookups. So, he theorized that if the
	  DUNDi lookups blocked for some period of time, that audio from
	  the zap channel could get lost. If the audio got lost, then it
	  wouldn't be run through the DTMF detector, and digits could get
	  lost. He was correct, and the following set of changes fixes the
	  problem. However, the changes go a little bit further than what
	  was necessary to fix this exact problem. 1) I updated
	  pbx_extension_helper() to autoservice the associated channel to
	  handle cases where extension lookups may take a long time. This
	  would normally be a dialplan switch that does some lookup over
	  the network, such as the DUNDi or IAX2 switches. This ensures
	  that even while a DUNDi lookup is blocking, the channel will be
	  continuously serviced. 2) I made a change to the autoservice
	  code. This is actually something that has bothered me for a long
	  time. When a channel is in autoservice, _all_ frames get thrown
	  away. However, some frames really shouldn't be thrown away. The
	  most notable examples are signalling (CONTROL) frames, and DTMF.
	  So, this patch queues up important frames while a channel is in
	  autoservice. When autoservice is stopped on the channel, the
	  queued up frames get stuck back on the channel so that they can
	  get processed instead of thrown away. 3) I made another change to
	  the autoservice code to handle the case where autoservice is
	  started on channels recursively. Previously, you could call
	  ast_autoservice_start() multiple times on a channel, and it would
	  stop the first time ast_autoservice_stop() gets called. Now, it
	  will ensure that autoservice doesn't actually stop until the
	  final call to ast_autoservice_stop().

2007-11-27 20:22 +0000 [r89727]  Mark Michelson <mmichelson@digium.com>

	* res/res_config_pgsql.c: Changing some calls from free() to
	  ast_free() since they were allocated with ast_calloc(). (closes
	  issue #11390, reported and patched by Laureano)

2007-11-27 20:16 +0000 [r89701-89709]  Kevin P. Fleming <kpfleming@digium.com>

	* main/app.c: on second thought... revert all the other changes
	  i've made in app options parsing leaving only one: if an empty
	  argument is supplied for an option, set that argument pointer to
	  point to an empty string rather than NULL, so that the
	  application can do normal checks on it without worrying about it
	  being NULL

	* main/app.c: generate a warning when an application option that
	  requires an argument is ignored due to lack of an argument

2007-11-27 16:12 +0000 [r89634]  Russell Bryant <russell@digium.com>

	* configs/voicemail.conf.sample: Add a note to the sample voicemail
	  config noting that when using IMAP storage, only the first format
	  specified will be attached to the message.

2007-11-27 15:38 +0000 [r89631]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_env.c: Default result of STAT should be "0" not "".
	  Reported via the -users mailing list, fixed by me.

2007-11-27 15:23 +0000 [r89624-89630]  Olle Johansson <oej@edvina.net>

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
	  get a codec offer using a well-known payload type, but using it
	  for another codec that we don't know, Asterisk did not remove
	  that codec from the list. With this patch, we remove the codec
	  from audio and video rtp objects and deny it ever existed. Thanks
	  to lasse for testing. (closes issue #11376) Reported by: lasse
	  Patches: bug11376.txt uploaded by oej (license 306) Tested by:
	  lasse

	* configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
	  #11304) Reported by: pj

2007-11-27 06:24 +0000 [r89622]  Steve Murphy <murf@digium.com>

	* apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
	  include/asterisk/cdr.h: closes issue #11379; OK, this is an
	  attempt to make both sides happy. To the cdr.conf file, I added
	  the option 'unanswered', which defaults to 'no'. In this mode,
	  you will see a cdr for a call, whether it was answered or not.
	  The disposition will be NO ANSWER or ANSWERED, as appropriate.
	  The src is as you'd expect, the destination channel will be one
	  of the channels from the Dial() call, usually the last in the
	  list if more than one chan was specified. With unanswered set to
	  'yes', you will still see this cdr entry in both cases. But in
	  the case where the dial timed out, you will also see a cdr for
	  each line attempted, marked NO ANSWER, with no destination
	  channel name. The new option defaults to 'no', so you don't see
	  the pesky extra cdr's by default, and you will not see the
	  irritating 'not posted' messages.

2007-11-26 23:10 +0000 [r89616-89618]  Mark Michelson <mmichelson@digium.com>

	* apps/app_playback.c: After issuing a "say load new", if a caller
	  hangs up during the middle of playback of a number, app_playback
	  will continue to try to play the remaining files. With this
	  change, no more files will be played back upon hangup. (closes
	  issue #11345, reported and patched by IgorG)

	* apps/app_playback.c: After issuing a "say load new" tons of
	  warning messages are printed out to the CLI every time do_say in
	  app_playback is called. Removing these warnings

2007-11-26 21:10 +0000 [r89599-89610]  Joshua Colp <jcolp@digium.com>

	* main/dial.c: Fix issues with async dialing with an application
	  executing. The application has to be terminated and control
	  returned to the thread before hanging things up. (issue #BE-252)

	* res/res_features.c: Add module counting removal for error
	  conditions. (closes issue #11333) Reported by: Laureano Patches:
	  res_features_v2.c.patch uploaded by Laureano (license 265)

2007-11-26 17:41 +0000 [r89594]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Add channel locking to a function that needed to be
	  doing it. This is just a little something I noticed while working
	  on a completely unrelated issue.

2007-11-26 17:36 +0000 [r89587-89592]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_config.c: Use ast_free to free memory, or else we shall
	  implode if MALLOC_DEBUG is enabled. (closes issue #11347)
	  Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
	  (license 281)

	* apps/app_mixmonitor.c: Close the audio file before sending it to
	  the post processing application. (closes issue #11357) Reported
	  by: reformed Patches: mixmonitor.patch uploaded by reformed
	  (license 330)

2007-11-26 17:20 +0000 [r89586]  Kevin P. Fleming <kpfleming@digium.com>

	* main/app.c: when parsing application options that take arguments,
	  don't indicate that the option was supplied unless a
	  non-zero-length argument was found for it

2007-11-26 15:48 +0000 [r89580]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Revert vmu->email back to an empty string
	  if it was empty when imap_store_file was called. This prevents
	  sending a duplicate e-mail. (closes issue #11204, reported by
	  spditner, patched by me)

2007-11-26 15:34 +0000 [r89571-89577]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: If channel allocation fails because the alert
	  pipe could not be created also free the scheduler context.
	  (closes issue #11355) Reported by: eliel Patches:
	  main.channel.c.patch uploaded by eliel (license 64)

	* apps/app_meetme.c: When unloading app_meetme destroy any auto
	  created contexts created by SLA. (closes issue #11367) Reported
	  by: eliel

2007-11-25 17:17 +0000 [r89559]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample,
	  include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
	  attempted to use the ESCAPE clause to set the escape delimiter to
	  a backslash. Unfortunately, this does not universally work on all
	  databases, since on databases which natively use the backslash as
	  a delimiter, the backslash itself needs to be delimited, but on
	  other databases that have no delimiter, backslashing the
	  backslash causes an error. So the only solution that I can come
	  up with is to create an option in res_odbc that explicitly
	  specifies whether or not backslash is a native delimiter. If it
	  is, we use it natively; if not, we use the ESCAPE clause to make
	  it one. Reported by: elguero Patch by: tilghman (Closes issue
	  #11364)

2007-11-24 16:59 +0000 [r89534-89545]  Tilghman Lesher <tlesher@digium.com>

	* res/res_adsi.c: Free some frames that would otherwise leak on
	  error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
	  issue #11351)

	* apps/app_voicemail.c, main/app.c: Currently, zero-length
	  voicemail messages cause a hangup in VoicemailMain. This change
	  fixes the problem, with a multi-faceted approach. First, we do
	  our best to avoid these messages from being created in the first
	  place, and second, if that fails, we detect when the voicemail
	  message is zero-length and avoid exiting at that point. Reported
	  by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)

	* main/manager.c: Up until this point, the XML output of the
	  manager has been technically invalid, due to the repetition of
	  certain parameters in a single event. This caused various issues
	  for XML parsers, some of which refused to parse at all, given the
	  invalidity of the rendered XML. So this commit fixes the XML
	  output, ensuring that each entity parameter has a unique name,
	  thus ensuring valid XML. Reported by: msetim Patch by: tilghman
	  (Closes issue #10220)

	* res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
	  not just 2nd-Nth parameters. Reported by: apsaras Patch by:
	  tilghman (Closes issue #11353)

2007-11-22 17:29 +0000 [r89527]  Russell Bryant <russell@digium.com>

	* configs/agents.conf.sample: mvanbaak pointed out a spelling error
	  in this sample configuration file. While I was at it, I went
	  ahead and tweaked it a little bit more.

2007-11-21 19:27 +0000 [r89493-89495]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fix a small error I made in my previous commit

	* apps/app_queue.c: Changing an inaccurate debug message to be less
	  inaccurate. Under the circumstances, this message would always
	  report that there were 0 members available, even though that may
	  not be true.

2007-11-21 18:59 +0000 [r89491]  Terry Wilson <twilson@digium.com>

	* res/res_features.c: If a channel gets masqueraded in the middle
	  of a park, don't play the announcement to the masqueraded
	  channel, and dial back to the original channel on timeout.

2007-11-20 19:16 +0000 [r89461-89462]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/module.h: re-doxygen some comments

	* main/loader.c, include/asterisk/module.h,
	  build_tools/make_buildopts_h: bring back compile-option checking
	  when loading modules, only this time use a string-based storage
	  and comparison mechanism because it is easier to support on other
	  platforms

2007-11-20 17:50 +0000 [r89457]  Mark Michelson <mmichelson@digium.com>

	* main/pbx.c: According to comments in main/pbx.c, it is essential
	  that if we are going to lock the conlock as well as the hints
	  lock, it must be locked in that respective order. In order to
	  prevent a potential deadlock, we need to lock the conlock prior
	  to locking the hints lock in ast_hint_state_changed (see the call
	  stack example on issue #11323 for how this can happen). (closes
	  issue #11323, reported by eelcob, suggestion for patch by eelcob,
	  patch by me)

2007-11-20 15:22 +0000 [r89450]  Steve Murphy <murf@digium.com>

	* doc/queues-with-callback-members.txt: closes issue #11324; break
	  statements missing in switch cases.

2007-11-20 13:40 +0000 [r89445]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: added RR patch from iroot #10908, thanks.

2007-11-19 15:53 +0000 [r89416-89419]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Print out the correct filename
	  (features.conf) in the log message when parkpos options are
	  incorrect. (closes issue #11295) Reported by: Laureano Patches:
	  res_features.c.patch uploaded by Laureano (license 265)

	* doc/localchannel.txt: Clarify documentation a bit, include that a
	  frame has to pass through the core in order for the Local channel
	  optimization to happen. (closes issue #11246) Reported by: jon

2007-11-16  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.14 released.

2007-11-16 22:26 +0000 [r89339]  Russell Bryant <russell@digium.com>

	* main/loader.c, include/asterisk/module.h,
	  build_tools/make_buildopts_h: Temporarily revert revision 89325,
	  which added md5 magic for keeping track of what build options
	  were used. We agreed that we should remove this before making a
	  1.4 release, and then we can put it back in. Then, we can take a
	  month or so to play around with it to get it how we want it.

2007-11-16 16:47 +0000 [r89325]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c, include/asterisk/module.h,
	  build_tools/make_buildopts_h: To help combat problems where
	  people build external modules (asterisk-addons or others) and
	  then change the build options of the Asterisk build in a way that
	  makes the incompatible without warning, this commit introduces an
	  MD5 signature of the important build-time options and includes
	  that signature into modules when they are built. When the loader
	  loads one of these modules and notices the problem, it will emit
	  a warning to console and refuse to initialize the module, as
	  doing so could cause the system to be unstable or even crash. If
	  you upgrade to this version of Asterisk, you must rebuild *all*
	  of your modules that came from other sources before trying to run
	  this version. If you are using Digium's G.729 binary codec
	  module, you will need v33 or newer.

2007-11-16 15:28 +0000 [r89323]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Make realtime queues accessible from the
	  QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
	  patched by atis, with small modifications from me)

2007-11-15 18:37 +0000 [r89298-89302]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Start Asterisk in Debian at a more reasonable time
	  (since zaptel is at level 20)

	* channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
	  by valgrind

	* channels/chan_iax2.c: Yet another memory corruption issue.
	  Reported by: atis Patch by: tilghman Fixes issue #10923

2007-11-15 17:19 +0000 [r89296]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Update the SLAStation application to account
	  for the case where the SLA thread has a call out to the station,
	  but the user has pressed a line button to answer the call instead
	  of picking up the handset. If they do, the phone sends out a new
	  INVITE. So, the SLAStation app must check to see if it is picking
	  up a ringing trunk, and ensure that the other stations stop
	  ringing. (reported internally, patched by me, tested by mogorman)

2007-11-15 14:57 +0000 [r89286-89288]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Undoing previous commit since I realize it was
	  wrong

	* main/manager.c: Adding a missing mutex unlock. (closes issue
	  11256, reported and patched by ys)

2007-11-15 11:26 +0000 [r89280-89281]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't send re-invites during pending INVITE
	  transactions. Patch by one47 - thanks! Closes issue #9305

	* channels/chan_sip.c: Improve support for multipart messages. Code
	  by gasparz, changes by me (mostly formatting). Thanks, gasparz!
	  Closes issue #10947

2007-11-14 23:23 +0000 [r89275]  Tilghman Lesher <tlesher@digium.com>

	* main/app.c: When a recording ends with '#', we are improperly
	  trimming an extra 200ms from the recording. Reported by: sim
	  Patch by: tilghman Closes issue #11247

2007-11-14 01:15 +0000 [r89260]  Joshua Colp <jcolp@digium.com>

	* main/srv.c: Return the proper value when the srv_callback
	  function executes properly. (closes issue #11240) Reported by:
	  jtodd

2007-11-13 21:07 +0000 [r89248-89254]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
	  systems which require a third arg to open() when using O_CREAT.
	  Issue 11238, reported by puzzled.

	* res/res_features.c: Revert change from revision 67064. It is
	  documented behavior that if a parking extension already exists
	  while using PARKINGEXTEN, dialplan execution will continue. If
	  blind transferring to a Park with PARKINGEXTEN, you must keep
	  this in mind, and handle the failure yourself. Issue 11237,
	  reported by jon.

2007-11-13 17:34 +0000 [r89246]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_sip.c: If we set a value for qualify, we should
	  actually pay attention to it, instead of overriding the value

2007-11-13 16:02 +0000 [r89241]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: Reverting commit made in revision 89205
	  since it is unnecessary. Thanks to Kevin for pointing this out

2007-11-13 13:51 +0000 [r89239]  Tilghman Lesher <tlesher@digium.com>

	* main/utils.c: Debugging is running into the 16-lock limit.
	  Increase to avoid. (This define is only effective when debugging
	  is turned on, so there's no effect for most installations.)

2007-11-13 00:56 +0000 [r89205]  Mark Michelson <mmichelson@digium.com>

	* apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
	  only 1 argument is given, then the args.options and
	  args.post_process strings are uninitialized and could contain
	  garbage. This change handles this situation properly by only
	  using arguments that we have parsed.

2007-11-12 20:46 +0000 [r89194]  Jason Parker <jparker@digium.com>

	* main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev

2007-11-12 20:16 +0000 [r89184-89191]  Tilghman Lesher <tlesher@digium.com>

	* main/config.c: If two config writes collide, file corruption
	  could result. Use a mkstemp() file, instead. Reported by:
	  paravoid Patch by: tilghman Closes issue #10781

	* main/channel.c, channels/chan_sip.c: Fix two cases of memory
	  corruption caused by background threads. Reported by: atis Patch
	  by: tilghman Fixes issue #10923

2007-11-12 11:26 +0000 [r89169-89173]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
	  no number was dialed and overlapdial is set, we wait for the ISDN
	  timeout instead of starting our own timer. added a comment for
	  the misdn.conf.sample for the overlapdial config option.

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
	  channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
	  restart all interfaces Restart_Indicator, to automatically send a
	  RESTART after the L2 of a PTP Port comes up. Also fixed some
	  places where we have send a RELEASE without need for it.

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
	  state/event issue with overlapdial=yes when no extension matched.
	  removed the general sending of a RELEASE_COMPLETE when we receive
	  a RELEASE, this is done by mISDNuser/mISDN. This makes it
	  possible to use asterisk-1.4 with mISDN trunk, but requires users
	  of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
	  (when using the NT mode at all)

	* channels/misdn/isdn_lib.c: fixed the support for CW and therefore
	  for the reject_cause option.

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
	  aded ntkeepcalls option, to avoid droÃpping calls when the L2
	  goes down on a PTP link. There are some pbx which do turn off the
	  L1 for a very short while and restart it immediately. normally
	  T310 should be started and after 10 seconds or so the calls
	  should be dropped, this is a simple fix wihtout this timer.

2007-11-08 23:52 +0000 [r89125]  Jason Parker <jparker@digium.com>

	* main/say.c: Properly say the seconds here.. Issue 11203, fix
	  described by vma.

2007-11-08 21:00 +0000 [r89119]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Rework of the commit I made yesterday to use
	  the already built-in ast_uri_decode function as opposed to my
	  home-rolled one. Also added comments. Thanks to oej for pointing
	  me in the right direction

2007-11-08 18:45 +0000 [r89115]  Jason Parker <jparker@digium.com>

	* configs/res_odbc.conf.sample: Avoid warnings on load when using
	  sample configuration files. Issue 11195, patch by eliel.

2007-11-08 16:47 +0000 [r89111]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: I made this same adjustment in trunk to fix
	  a bug, and it makes sense to do it in 1.4 as well. If an
	  imapfolder is specified in voicemail.conf, don't ever explicitly
	  connect to INBOX since it may not exist.

2007-11-08 05:26 +0000 [r89105]  Kevin P. Fleming <kpfleming@digium.com>

	* main/srv.c: fix a glaring bug in the new SRV record handling that
	  would cause incorrect weight sorting

2007-11-08 04:55 +0000 [r89103]  Tilghman Lesher <tlesher@digium.com>

	* doc/valgrind.txt: Typo

2007-11-08 02:26 +0000 [r89095-89101]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Do not add a sip: to the beginning of the To
	  URI unless needed. (closes issue #10756) Reported by: goestelecom

	* channels/chan_sip.c: Improve the devicestate logic for multiple
	  devices. If any are available then the extension is considered
	  available. (closes issue #10164) Reported by: nic_bellamy
	  Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
	  (license 299)

	* channels/chan_sip.c: Add support for allowing one outgoing
	  transaction. This means if a response comes back out of order
	  chan_sip will still handle it. I dream of a chan_sip with real
	  transaction support. (closes issue #10946) Reported by: flefoll
	  (closes issue #10915) Reported by: ramonpeek (closes issue #9567)
	  Reported by: atca_pres

	* channels/chan_sip.c: If callerid is configured in sip.conf use
	  that for checking the presence of an extension in the dialplan.
	  (closes issue #11185) Reported by: spditner

2007-11-07 23:39 +0000 [r89093]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: The member refcount must be incremented, to
	  avoid using it after deallocation. A huge thanks go to lvl- for
	  patiently providing the necessary valgrind output that was
	  necessary to finding this problem of memory corruption. Reported
	  by: lvl- Patch by: tilghman Closes issue #11174

2007-11-07 22:40 +0000 [r89090]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: This patch makes it possible for SIP phones
	  to dial extensions defined with '#' characters in extensions.conf
	  AND maintain their escaped characters when forming URI's (closes
	  issue #10681, reported by cahen, patched by me, code review by
	  file)

2007-11-07 21:40 +0000 [r89088]  Steve Murphy <murf@digium.com>

	* cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
	  10578, I just ran 1.4 thru valgrind; some of the config leakage
	  I've already fixed, but it doesn't hurt to double check. I found
	  and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
	  tho.

2007-11-07 15:56 +0000 [r89085]  Mark Michelson <mmichelson@digium.com>

	* main/manager.c: Fixing a segfault in the manager "core show
	  channels concise" command. (closes issue #11183, reported by arnd
	  and patched by ys)

2007-11-07 04:07 +0000 [r89079]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.ael.sample: Suppress AEL warnings on load.
	  Reported by: eliel Patch by: eliel Closes issue #11178

2007-11-06 20:18 +0000 [r89053]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix init_classes() so that classes that
	  actually do have files loaded aren't treated as empty, and
	  immediately destroyed ...

2007-11-06 19:09 +0000 [r89046]  Jason Parker <jparker@digium.com>

	* codecs/codec_zap.c: Correctly set the total number of channels
	  from a zaptel transcoder board. SPD-49, patch by Matthew
	  Nicholson.

2007-11-06 19:09 +0000 [r89045]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: We went to the trouble of creating a
	  method of tracking failed trylocks, then never turned it on
	  (oops).

2007-11-06 18:53 +0000 [r89042]  Olle Johansson <oej@edvina.net>

	* main/tdd.c: Bug fixes to tdd support in zaptel.

2007-11-06 18:20 +0000 [r89037]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: If someone were to delete the files used
	  by an existing MOH class, and then issue a reload, further use of
	  that class could result in a crash due to dividing by zero. This
	  set of changes fixes up some places to prevent this from
	  happening. (closes issue #10948) Reported by: jcomellas Patches:
	  res_musiconhold_division_by_zero.patch uploaded by jcomellas
	  (license 282) Additional changes added by me.

2007-11-06 17:52 +0000 [r89036]  Steve Murphy <murf@digium.com>

	* main/config.c: closes issue #8786 - where the [catname](!) and
	  [catname](othercat1,othercat2,...) notation gets dropped across a
	  ConfigUpdate (or any other thing that would cause a config file
	  to be written). While I was at it, I also cleaned up some of the
	  destroy routines to free up comments, which was not being done.
	  Made sure the new struct I introduced is also cleaned up properly
	  at destruction time. My code handles multiple template
	  inclusions. Many thanks to ssokol for his patch, which, while not
	  literally used in the final merge, served as a foundation for the
	  fix.

2007-11-06 17:08 +0000 [r88994-89032]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make it so that if a peer is determined to
	  be unreachable using qualify their devicestate will report back
	  unavailable. (closes issue #11006) Reported by: pj

	* channels/chan_zap.c: Fix improbable but possible memory leaks in
	  chan_zap. (closes issue #11166) Reported by: eliel Patches:
	  chan_zap.c.patch uploaded by eliel (license 64)

2007-11-06 13:50 +0000 [r88931]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Remove some checks to see if locks are
	  initialized from the non-DEBUG_THREADS versions of the lock
	  routines. These are incorrect for a number of reasons: - It
	  breaks the build on mac. - If there is a problem with locks not
	  getting initialized, then the proper fix is to find that place
	  and fix the code so that it does get initialized. - If additional
	  debug code is needed to help find the problem areas, then this
	  type of things should _only_ be put in the DEBUG_THREADS
	  wrappers.

2007-11-06 02:52 +0000 [r88862]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/srv.h: update comment to match the state of the
	  code

2007-11-05 23:29 +0000 [r88826]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: Reworked deadlock avoidance in __ast_read.
	  Restored audio to callback agents. (closes issue #11071, reported
	  by callguy, patched by me, tested by callguy and Ted Brown)

2007-11-05 22:07 +0000 [r88709-88805]  Russell Bryant <russell@digium.com>

	* main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
	  to channel variables, I went looking around at the ways that
	  channel variables are handled. In general, they were not handled
	  in a thread-safe way. The channel _must_ be locked when reading
	  or writing from/to the channel variable list. What I have done to
	  improve this situation is to make pbx_builtin_setvar_helper() and
	  friends lock the channel when doing their thing. Asterisk API
	  calls almost all lock the channel for you as necessary, but this
	  family of functions did not. (closes issue #10923, reported by
	  atis) (closes issue #11159, reported by 850t)

	* channels/chan_sip.c: When traversing the list of channel
	  variables here in transmit_invite(), the asterisk channel must be
	  locked, as this data may change at any time. (I have seen
	  numerous reports of crashes related to the handling of channel
	  variables. There are a couple of issues on the bug tracker
	  related to it, but it has also been noted on IRC and mailing
	  lists. So, I am finding and fixing some places where channel
	  variables are handled improperly.)

	* channels/chan_sip.c: Fix up some indentation.

	* main/srv.c, include/asterisk/srv.h: Merge changes from
	  asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
	  record support in Asterisk was broken. There was no guarantee on
	  what record Asterisk would choose to actually use. This set of
	  changes improves the situation by ensuring that Asterisk will
	  choose the highest priority record.

	* main/channel.c: Merge the last bit of changes from
	  asterisk/team/russell/readq-1.4 The issue here is that the
	  channel frame readq handling got broken when the code was
	  converted to use the linked list macros. It caused corruption of
	  the list head and tail pointers. So, I fixed up the usage of the
	  linked list macros and in passing, simplified the code. I also
	  documented what the code is doing, as it was a bit difficult to
	  figure out at first. This bug showed itself with crashes showing
	  messed up head/tail pointers for the readq. However, there are a
	  couple of crashes that aren't quite as obvious, but I think may
	  be related. So, if your bug gets closed by this commit, but you
	  still have a problem, please reopen or create a new bug report.
	  (closes issue #10936) (closes issue #10595) (closes issue #10368)
	  (closes issue #11084) (closes issue #10040) (closes issue #10840)

2007-11-05 18:47 +0000 [r88671]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If a SIP channel is put on hold multiple
	  times do not keep incrementing the onHold value. (closes issue
	  #11085) Reported by: francesco_r Tested by: blitzrage (closes
	  issue #10474) Reported by: acennami

2007-11-05 17:46 +0000 [r88624]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix up datastore handling in ast_do_masquerade().
	  The code is intended to move any channel datastores from the old
	  channel to the new one. However, it did not use the linked list
	  macros properly to accomplish the task. The existing code would
	  only work if there was only a single datastore on the old
	  channel.

2007-11-05 17:19 +0000 [r88585]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Make sure we destroy the config structure on
	  configuration failure. Issue 11163, patch by eliel.

2007-11-05 16:20 +0000 [r88539]  Tilghman Lesher <tlesher@digium.com>

	* res/res_odbc.c: Don't check used pooled connections for
	  connection status, as it will cause issues for prepared queries.
	  Reported by: Nick Gorham (via -dev list) Patch by: tilghman

2007-11-04 22:38 +0000 [r88471]  Luigi Rizzo <rizzo@icir.org>

	* include/asterisk/stringfields.h, main/channel.c,
	  apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
	  Rename ast_string_field_free_pool to
	  ast_string_field_free_memory, and ast_string_field_free_all to
	  ast_string_field_reset_all to avoid misuse (due to too similar
	  names and an error in documentation). Fix two related memory
	  leaks in app_meetme. No need to merge to trunk, different fix
	  already applied there. Not applicable to 1.2

2007-11-02 20:49 +0000 [r88328-88366]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make subscribecontext behave as advertised.
	  It will now look for the presence of a hint in the given context
	  (be it subscribecontext or context). (closes issue #10702)
	  Reported by: slavon

	* channels/chan_sip.c: If an INFO request within a dialog is
	  received with a content length of 0 simply send back a 200 OK. It
	  is valid to do this and the remote side is probably using it to
	  make sure the signalling is still alive. (closes issue #5747)
	  Reported by: chandi Patches: infofix-81430-1.patch uploaded by
	  IgorG (license 20)

2007-11-02 16:51 +0000 [r88283]  Jason Parker <jparker@digium.com>

	* main/say.c: We need to make sure to specify a language to
	  ast_fileexists, otherwise it may fail for anything besides en
	  Issue 11147, fix discovered by both citats and myself
	  (independently), with input from Corydon76

2007-11-02 13:03 +0000 [r88116-88210]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
	  Patch by: ys Closes issue #11143

	* doc/valgrind.txt (added): Add some notes on using valgrind

2007-11-01 16:21 +0000 [r88078]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Make sure we set the poll fds to NULL after
	  free()ing it. Part of issue 11017, patch by tzafrir.

2007-11-01 13:27 +0000 [r87970-88026]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Fix up commit for my Zap channel with spies in
	  Meetme fix. (thanks Tony Mountifield!)

	* apps/app_meetme.c: If a Zap channel contains a spy or a spy is
	  added take it out of the conference in kernel space and make it
	  go through Asterisk so the spy gets audio from both sides.
	  (closes issue #10060) Reported by: mparker

2007-10-31 21:23 +0000 [r87906-87908]  Jason Parker <jparker@digium.com>

	* res/res_jabber.c: Make sure we free some allocated memory before
	  returning. Issue 11131, patch by eliel.

	* channels/chan_gtalk.c: Don't try to allocate memory that we're
	  just going to re-allocate later anyways. Issue 11130, patch by
	  eliel.

2007-10-31 18:03 +0000 [r87852]  Tilghman Lesher <tlesher@digium.com>

	* Makefile: Create samples for ALL of the available options in
	  asterisk.conf

2007-10-31 17:49 +0000 [r87775-87849]  Steve Murphy <murf@digium.com>

	* pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
	  save' cli command saves a file where the semicolon is not
	  escaped. Fixed this; User also wanted comments to be preserved
	  across dialplan save, but this is impossible at this point in
	  time, because comments are not stored in the dialplan. They are
	  'compiled' out of extensions.conf. The only way to preserve those
	  comments is to use the config file reader/writer that the GUI
	  uses to allow online user edits. extensions.conf is first and
	  foremost, a config file, and is read in by the normal config-file
	  reading routines. Then, it is processed into a dialplan
	  (context/exten structs).

	* pbx/pbx_ael.c: Included some verbage in the check_includes func,
	  to inform the user that included contexts that have no match in
	  the AEL, might be OK, as AEL cannot check in the extensions.conf
	  or the in-memory contexts, as they may not be there at the time
	  of the check.

2007-10-30 23:02 +0000 [r87739]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
	  Reported by: ys Fixed by: ys Closes issue #11116

2007-10-30 21:19 +0000 [r87686]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Merge the changes from
	  team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
	  race condition related to the handling of POKEing peers.
	  Essentially, a reference to a peer is held by the scheduler when
	  there are pending callbacks, but the reference count didn't
	  reflect it. So, it was possible for a peer to hit a reference
	  count of zero and have its destructor begin to be called at the
	  same time that the scheduler thread ran a POKE related callback.
	  If that happened, a crash would likely occur. (closes issue
	  #11082, closes issue #11094)

2007-10-30 20:29 +0000 [r87650]  Jason Parker <jparker@digium.com>

	* channels/Makefile: Only try to clean out h323/ if the
	  h323/Makefile exists.

2007-10-30 16:13 +0000 [r87571]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Add two more checks before printing out a
	  warning message about bridging. If either channel has hungup of
	  course the bridge will have failed. (closes issue #10009)
	  Reported by: dimas

2007-10-30 15:45 +0000 [r87567]  Jason Parker <jparker@digium.com>

	* main/editline/np/vis.c: Fix build of editline on Solaris. Issue
	  11113, patch by snuffy.

2007-10-30 15:10 +0000 [r87534]  Joshua Colp <jcolp@digium.com>

	* apps/app_followme.c: Return 1.4 to a state where it builds.
	  Changing the arguments to a function and not changing where they
	  are used is bad, mmmk?

2007-10-30 14:31 +0000 [r87514]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Fix issue where the recorded name wasn't
	  getting removed correctly. (closes issue #11115) Reported by:
	  davevg Patches: followme-v3.diff

2007-10-29 22:13 +0000 [r87460-87465]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/gsm: missed one directory

	* codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
	  codecs/lpc10, main/db1-ast, main/editline, main,
	  codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
	  codecs/lpc10/Makefile, utils, codecs, agi,
	  main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
	  clean up (and ignore) assembler and preprocessor intermediate
	  files if any are created during the build

	* Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
	  already there (used when debugging preprocessor issues) because
	  the compiler will whine about each compile command

2007-10-29 21:06 +0000 [r87427]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing a completely unnecessary quota
	  check from IMAP code.

2007-10-29 20:22 +0000 [r87373-87396]  Russell Bryant <russell@digium.com>

	* main/utils.c, include/asterisk/lock.h: Add some more details to
	  the output of "core show locks". When a thread is waiting for a
	  lock, this will now show the details about who currently has it
	  locked. (inspired by issue #11100)

	* main/astmm.c: Remove a lock that doesn't make any sense. The
	  regions lock needs to be held when traversing the list of
	  allocated chunks so that they can be printed out to the CLI.
	  (Thanks to eliel on #asterisk-dev for pointing this out!)

2007-10-29 17:20 +0000 [r87342]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix issue where if both sides of the dialog
	  cancelled the dialog at the same time chan_sip could kepe
	  retransmitting a response for no reason. (closes issue #9566)
	  Reported by: atca_pres Patches: bug9566.patch uploaded by oej

2007-10-29 17:13 +0000 [r87340]  Jason Parker <jparker@digium.com>

	* funcs/func_realtime.c, funcs/func_cut.c: Allow some function
	  modules to compile under dev mode. Issue 11104, patch by andrew.

2007-10-29 14:23 +0000 [r87294]  Joshua Colp <jcolp@digium.com>

	* main/utils.c: Fix issue with ast_unescape_semicolon going into an
	  endless loop. (closes issue #10550) Reported by: ramonpeek
	  Patches: unescape-85177-1.patch uploaded by IgorG (license 20)

2007-10-28 13:46 +0000 [r87262]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
	  funcs/func_cut.c: Add autoservice to several more functions which
	  might delay in their responses. Also, make sure that func_odbc
	  functions have a channel on which to set variables. Reported by
	  russell Fixed by tilghman Closes issue #11099

2007-10-26 16:34 +0000 [r87168]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
	  pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
	  include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
	  utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
	  pbx/ael/ael.flex: closes issue #11086 where a user complains that
	  references to following contexts report a problem; The problem
	  was REALLy that he was referring to empty contexts, which were
	  being ignored. Reporter stated that empty contexts should be OK.
	  I checked it out against extensions.conf, and sure enough, empty
	  contexts ARE ok. So, I removed the restriction from AEL. This,
	  though, highlighted a problem with multiple contexts of the same
	  name. This should be OK, also. So, I added the extend keyword to
	  AEL, and it can preceed the 'context' keyword (mixed with
	  'abstract', if nec.). This will turn off the warnings in AEL if
	  the same context name is used 2 or more times. Also, I now call
	  ast_context_find_or_create for contexts now, instead of just
	  ast_context_create; I did this because pbx_config does this. The
	  'extend' keyword thus becomes a statement of intent. AEL can now
	  duplicate the behavior of pbx_config,

2007-10-26 13:54 +0000 [r87120]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_curl.c: The addition of autoservice to func_curl
	  additionally made func_curl dependent on the existence of a
	  channel, with no real reason. This should make func_curl once
	  again work without a channel. Reported by jmls. Fixed by
	  tilghman. Closes issue #11090

2007-10-25 23:03 +0000 [r87069]  Kevin P. Fleming <kpfleming@digium.com>

	* main/channel.c, include/asterisk/linkedlists.h: appending one
	  list to another should leave the first list empty, and not
	  require the user to do that

2007-10-25 22:53 +0000 [r87067]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_cut.c: Backport alternate encoding of newline
	  delimiters from trunk to 1.4, as approved by Russell Reported by
	  blitzrage Closes issue #10903

2007-10-24 20:56 +0000 [r86982]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: Correctly respect hidecalleridname
	  configuration option. Simplify code slightly in the process.
	  Issue 11079, reported by ddv2005

2007-10-24 04:14 +0000 [r86880-86936]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
	  to specify app:spec in hint arguments

	* funcs/func_logic.c: closes issue #11052 -- where nothing after
	  the ? will allow un-initialized variable values to corrupt and
	  crash asterisk on 64-bit platforms

	* main/Makefile: this update to Makefile corrects how ast_expr2f.c
	  should be generated

	* main/ast_expr2f.c: This should get rid of a really, really
	  irritating warning generated by some 64-bit platforms from libc,
	  where free(0) is frowned upon

2007-10-22 21:36 +0000 [r86836]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: If lock tracking is not enabled, then we
	  can not attempt to log any mutex failures. If so, we could end up
	  in infinite recursion. The only lock that is affected by this is
	  a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
	  issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
	  (license 281)

2007-10-22 17:38 +0000 [r86787]  Tilghman Lesher <tlesher@digium.com>

	* main/astmm.c: Minor FreeBSD build fix

2007-10-22 16:35 +0000 [r86754-86756]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: After reading online I have confirmed that
	  Record-Route headers should be copied to 1xx responses as well.
	  (closes issue #10113) Reported by: makoto

	* apps/app_controlplayback.c: Make sure res is a positive value
	  before performing the check to determine whether the user stopped
	  it or not. (closes issue #11023) Reported by: cfc

2007-10-22 15:52 +0000 [r86726-86750]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't leak a frame in the case that an END frame
	  is received and the time since the BEGIN is less than that of the
	  defined minimum DTMF duration. (closes issue #11051) Reported by:
	  casper Patches: channel.c.86664.diff uploaded by casper (license
	  55)

	* include/asterisk/lock.h: Update the static mutex initializer to
	  include the initialization of the internal mutex used to protect
	  the lock debugging data. (closes issue #11044, patch suggested by
	  Ivan)

2007-10-22 14:48 +0000 [r86694]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Account for the fact that sometimes headers
	  may be terminated with \r\n instead of just \n (closes issue
	  #11043, reported by yehavi)

2007-10-22 14:27 +0000 [r86630-86663]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Move log message to before the frame it
	  references is freed. (closes issue #11050) Reported by: slavon
	  Patches: channel.c.86662.diff uploaded by casper (license 55)

	* pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
	  issue #11041) Reported by: jsmith Patches:
	  asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
	  (license 176)

	* main/loader.c: Fixes for building under OpenSolaris. (closes
	  issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
	  uploaded by snuffy (license 35)

2007-10-22 09:21 +0000 [r86598]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
	  DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
	  does not match after an overlap call. Also added out_cause=1

2007-10-19 16:38 +0000 [r86469-86502]  Joshua Colp <jcolp@digium.com>

	* main/app.c: When returning a DTMF digit from
	  ast_control_streamfile cast it as a char so that 0 does not
	  overlap with the success return code. (closes issue #11023)
	  Reported by: cfc

	* channels/chan_sip.c: Fix two issues with domains and transfers.
	  If a port was given in the hostname it was treated as part of the
	  hostname. If domains were configured but external domains were
	  not enabled all transfers would be considered remote. (closes
	  issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
	  uploaded by ramonpeek (license 266)

	* channels/chan_sip.c: Set port number in received as information
	  for registrations as well. (closes issue #11028) Reported by:
	  brad-x

2007-10-19 01:45 +0000 [r86438]  TransNexus OSP Development <support@transnexus.com>

	* apps/app_osplookup.c: Fixed OSP module did not report
	  source/devinfo IP in correct format.

2007-10-18 22:01 +0000 [r86405-86406]  Jason Parker <jparker@digium.com>

	* Makefile: Correct documentation. I removed the wrong line..

	* Makefile: Add documentation for options in asterisk.conf Issue
	  11029, patch by eserra

2007-10-18 21:16 +0000 [r86330-86372]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
	  commit.

	* configs/iax.conf.sample, channels/chan_iax2.c: Add support for
	  setting the maximum trunk size for IAX2 trunking

	* main/channel.c, include/asterisk/channel.h: The channel needs to
	  stay locked while running timer callbacks, as they access and
	  modify channel data that may change elsewhere. I went through
	  every timer callback in the source tree to make sure that none of
	  them did any additional locking that could introduce deadlocks,
	  and all is well. (closes issue #10765) Reported by: Ivan Patches:
	  ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
	  229)

2007-10-18 17:38 +0000 [r86328]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: If a non-existent file is specified to be
	  played either as a periodic announcement or as a hold/position
	  announcement, the caller would be kicked out of the queue. No
	  longer does this happen.

2007-10-18 15:45 +0000 [r86237-86296]  Russell Bryant <russell@digium.com>

	* codecs/codec_zap.c: Execute the RELEASE operation on transcoder
	  channels in the destroy callback. (patch from jsloan)

	* main/utils.c: Revert a change that I made for issue #10979 which,
	  as has been pointed out to me in issue #11018, doesn't really
	  make sense. There is no reason to have the base64 decode function
	  force a '\0' terminated buffer, when the result is almost always
	  binary, anyway. In fact, this caused some breakage, as some code
	  in res_crypto passed in a buffer exactly the right size to get
	  its binary result, which got stomped on by this patch. (closes
	  issue #11018, reported by dimas)

2007-10-17 21:39 +0000 [r86202]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changing the strategy field of the call_queue
	  struct to be signed instead of unsigned, since the code attempts
	  to set the strategy to -1 if you specify a bogus strategy. While
	  this isn't a huge issue in 1.4, it could be a problem for someone
	  who, say, tries to use the roundrobin strategy in trunk (despite
	  all the deprecation warnings in 1.4).

2007-10-17 17:57 +0000 [r86149]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: If Asterisk is in the middle of shutting
	  down, respond to OPTIONS with 503 Unavailable. (closes issue
	  #10994) Reported by: eserra Patches: sip-options-503.patch
	  uploaded by eserra (license 45)

2007-10-17 16:58 +0000 [r86117]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Whoops, forgot to remove the original
	  sip_scheddestroy. (closes issue #11010) Reported by: vadim

2007-10-17 15:23 +0000 [r86066]  Tilghman Lesher <tlesher@digium.com>

	* main/asterisk.c: When runuser/rungroup is specified, a remote
	  console could only be attained by root (Closes issue #9999)

2007-10-17 15:06 +0000 [r86063]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't schedule dialog destruction if a
	  MESSAGE is received using an existing dialog. (closes issue
	  #11010) Reported by: vadim

2007-10-16 23:35 +0000 [r86028-86032]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample: Since monitor-join is deprecated now,
	  remove the example from the sample queues.conf file

	* UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
	  the monitor-join queue option

	* apps/app_queue.c: Adding deprecated warning to monitor-join
	  option, since the plan is to no longer support this in favor of
	  monitor-type = mixmonitor (related to issue #10885)

2007-10-16 22:36 +0000 [r85994-85997]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: really picky formatting tweak ...

	* include/asterisk/lock.h: Some locking errors exposed the fact
	  that the lock debugging code itself was not thread safe. How
	  ironic! Anyway, these changes ensure that the code that is
	  accessing the lock debugging data is thread-safe. Many thanks to
	  Ivan for finding and fixing the core issue here, and also thanks
	  to those that tested the patch and provided test results. (closes
	  issue #10571) (closes issue #10886) (closes issue #10875) (might
	  close some others, as well ...) Patches: (from issue #10571)
	  ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
	  229) - a few small changes by me

2007-10-16 21:14 +0000 [r85958]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Trying to remove a non-dynamic queue member via
	  dynamic means can lead to some interesting (read nasty)
	  situations. This patch clears up the issue by making only dynamic
	  queue members removable via dynamic methods.

2007-10-16 19:41 +0000 [r85921]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/localtime.c: Also set up gmtoff (this is used in the
	  %z gnu extension to strftime) Reported and fixed by jcmoore
	  Closes issue #11002

2007-10-16 19:10 +0000 [r85896]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Remove a pointless lock.

2007-10-16 15:21 +0000 [r85852]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing a double free which happens in the
	  statechange thread. (closes issue #10987, reported by andrew)

2007-10-16 14:52 +0000 [r85818-85850]  Joshua Colp <jcolp@digium.com>

	* apps/app_hasnewvoicemail.c: Check to make sure a value has been
	  given to the VMCOUNT dialplan function. (closes issue #10996)
	  Reported by: marsosa

	* main/threadstorage.c: Fix memory allocation issue in
	  threadstorage. (closes issue #10995) Reported by: snuffy Patches:
	  new-patch.diff uploaded by snuffy (license 35)

2007-10-16 10:46 +0000 [r85800]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Fix the output for this channel help CLI
	  command

2007-10-15 21:10 +0000 [r85717-85720]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Ensure that no pending state changes are leaked
	  when the device state change thread gets stopped on module
	  unload.

	* apps/app_queue.c: Previously, app_queue created a thread to
	  handle every single device state change. I changed this a while
	  ago in trunk for performance reasons. However, bug 8407 points
	  out that it is actually a race condition, causing device state
	  changes to get processed in random order. So, I backported my
	  changes from trunk to 1.4. (closes issue #8407, patch provided by
	  tim_ringenbach, committed patch by me)

2007-10-15 20:29 +0000 [r85687]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_stack.c: Don't execute a gosub if the arguments is
	  zero-len (not just NULL) Reported by davevg Fixed by me Closes
	  issue #10985

2007-10-15 20:21 +0000 [r85686]  Russell Bryant <russell@digium.com>

	* main/say.c: Add a small fix for the tw version of saying dates.
	  (closes issue #7827) Reported by: sharkey Patches: say.nits.patch
	  uploaded by sharkey (license 172)

2007-10-15 20:15 +0000 [r85684]  Jason Parker <jparker@digium.com>

	* Makefile: Properly use DESTDIR in 'config' target. Do not try to
	  run chkconfig or similar if using DESTDIR. Issue 10938, patch by
	  cabal95.

2007-10-15 19:22 +0000 [r85604-85649]  Russell Bryant <russell@digium.com>

	* main/utils.c: Be pedantic about handling memory allocation
	  failure.

	* main/utils.c: The loop in the handler for the "core show locks"
	  could potentially block for some amount of time. Be a little bit
	  more careful and prepare all of the output in an intermediary
	  buffer while holding a global resource. Then, after releasing it,
	  send the output to ast_cli().

	* channels/chan_sip.c: Make the default for the srvlookup option to
	  be yes. It doesn't really make sense for it to default to off.
	  The default configuration file has it on, and proper RFC
	  behavior, as indicated by a comment in the code, is for it to be
	  on. So, let's have it on by default to make lives easier. (closes
	  issue #10954, suggested by jtodd)

2007-10-15 16:39 +0000 [r85571]  Joshua Colp <jcolp@digium.com>

	* configs/features.conf.sample: Document that DTMF based features
	  only work when two channels are bridged together. (closes issue
	  #10773) Reported by: pbayley

2007-10-15 16:34 +0000 [r85561]  Russell Bryant <russell@digium.com>

	* include/asterisk/strings.h: Make a few changes so that characters
	  in the upper half of the ISO-8859-1 character set don't get
	  stripped when reading configuration. (closes issue #10982,
	  dandre)

2007-10-15 16:22 +0000 [r85559]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Bring both DTMF begin and end frames up through to
	  the core for DTMF feature handling. (closes issue #10826)
	  Reported by: dimas

2007-10-15 15:40 +0000 [r85556]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Ensure the buffer passed to
	  ast_canmatch_extension() is properly initialized so that it is
	  null terminated. (issue #10977) Reported by: dimas Patches:
	  pbxdundi.patch uploaded by dimas (license 88) - small mods by me

2007-10-15 14:55 +0000 [r85552]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If Monitor or a spy was added to a P2P or native
	  bridged channel bring the channel back to the generic bridging
	  core so the monitor or spy operations work. (closes issue #10943)
	  Reported by: julianjm

2007-10-15 13:16 +0000 [r85540-85548]  Russell Bryant <russell@digium.com>

	* main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
	  (closes issue #10980) Reported by: casper Patches:
	  db.c.84633.diff uploaded by casper (license 55)

	* main/asterisk.c: Make sure remote consoles unmute themselves
	  again after reconnecting. (closes issue #10847) Reported by: atis
	  Patches: console_unmute_on_reconnect.patch uploaded by atis
	  (license 242)

	* main/utils.c: Make sure that the base64 decoder returns a
	  terminated string. (closes issue #10979) Reported by: ys Patches:
	  util.c.diff uploaded by ys (license 281) - small mods by me

	* pbx/pbx_config.c: Don't create the context for users in
	  users.conf until we know at least one user exists. (closes issue
	  #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
	  dimas (license 88)

2007-10-13 15:26 +0000 [r85536]  Tilghman Lesher <tlesher@digium.com>

	* configs/extensions.ael.sample: Remove deprecated syntax from
	  sample ael file Reported and patched by: dimas Closes issue
	  #10967

2007-10-13 05:48 +0000 [r85532-85533]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
	  issue with console verbosity when running asterisk -rx to execute
	  a command and retrieve its output. The issue was that there was
	  no way for the main Asterisk process to know that the remote
	  console was connecting in the -rx mode. The way that James has
	  fixed this is to have all remote consoles muted by default. Then,
	  regular remote consoles automatically execute a CLI command to
	  unmute themselves when they first start up. (closes issue #10847)
	  Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
	  by jamesgolovich (license 176)

	* main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
	  handle the case where read() may return the text for more than
	  one CLI command at once for a remote console. (closes issue
	  #10888) Reported by: jamesgolovich Patches:
	  asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
	  176)

2007-10-12 18:30 +0000 [r85523]  Tilghman Lesher <tlesher@digium.com>

	* doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address

2007-10-12 15:45 +0000 [r85515-85517]  Russell Bryant <russell@digium.com>

	* res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
	  SDMI. (closes issue #10959)

	* pbx/pbx_realtime.c: Fix the potential use of an uninitialized
	  buffer in a log message. (closes issue #10958) Reported by: dimas
	  Patches: realtime.patch uploaded by dimas (license 88)

2007-10-11 15:26 +0000 [r85397]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: When creating a new packet don't try to stop
	  retransmission of it. It was just allocated/created so it's
	  impossible for it to have already been scheduled. (closes issue
	  #10945) Reported by: flefoll Patches:
	  chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
	  (license 244)

2007-10-11 04:35 +0000 [r85356]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c: A dollar sign by itself, not indicating a start of a
	  variable or expression prematurely ends substitution (closes
	  issue #10939)

2007-10-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.13 released.

2007-10-10 15:56 +0000 [r85316]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h: I introduced a new member to the
	  ast_filestream struct in 1.4.12, but put it in the middle of the
	  struct, instead of at the end. One of the Debian folks, paravoid,
	  pointed out that this breaks binary compatability with modules
	  compiled against older headers. So, I'm moving the new member to
	  the end of the struct to resolve the situation.

2007-10-10 15:51 +0000 [r85315]  Mark Michelson <mmichelson@digium.com>

	* main/utils.c: The thread ID should be unsigned.

2007-10-10 14:42 +0000 [r85277-85280]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: If devicestate is passed a port number strip
	  it out. (closes issue #10930) Reported by: ibc

	* channels/chan_sip.c: Add support for handling a 182 Queued
	  response. (closes issue #10924) Reported by: ramonpeek Patches:
	  queued-182.diff uploaded by ramonpeek (license 266)

2007-10-10 14:26 +0000 [r85276]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: A bunch of changes from sprintf to
	  snprintf. See security advisory AST-2002-022

2007-10-10 14:14 +0000 [r85242]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Close voicemail message description file if
	  duration did not meet the minimum, or else we will eventually run
	  out of file descriptors. (closes issue #10918) Reported by:
	  brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
	  279)

2007-10-10 06:24 +0000 [r85195]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/frame.h: use a macro instead of an inline
	  function, so that backtraces will report the caller of
	  ast_frame_free() properly

2007-10-09 21:55 +0000 [r85158]  Tilghman Lesher <tlesher@digium.com>

	* main/channel.c, main/utils.c, include/asterisk/lock.h: This
	  commit fixes the following issues: - Deadlock in ast_write (issue
	  #10406) - Deadlock in ast_read (issue #10406) - Possible mutex
	  initialization error in lock.h (issue #10571)

2007-10-09 14:30 +0000 [r84990-85093]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't perform a reinvite if a transfer is in
	  progress. (issue #10915) Reported by: ramonpeek

	* main/rtp.c: Only update codec information if the channel has a
	  technology private structure. (issue #10915) Reported by:
	  ramonpeek

	* main/rtp.c: Update codec information as well as address when
	  doing hold reinvites. (issue #10868) Reported by: mavince

	* main/channel.c: Don't keep trying to native bridge if either of
	  the channels are involved in a masquerade operation to be done.
	  (closes issue #10696) Reported by: tbelder

2007-10-08 03:28 +0000 [r84957]  Russell Bryant <russell@digium.com>

	* Makefile.rules: Enable file dependency tracking for _all_ builds,
	  and not just for builds with dev-mode enabled. I have seen enough
	  problems caused by this that I don't think it's worth keeping. I
	  want to continue to encourage anybody that is interested to
	  continue to run Asterisk from svn. Furthermore, I do not want
	  their systems to break when we change a structure definition in a
	  header file. :)

2007-10-07 16:15 +0000 [r84890-84902]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: Presence packets from a client who's connected
	  with our Jabber ID are valid, therefore, those clients must be
	  considered as buddies. The resource string helps us make the
	  distinction between clients. Closes issue #10707, reported by
	  yusufmotiwala.

	* res/res_jabber.c: Prevent Asterisk from crashing when receiving a
	  presence packet without resource from a buddy that is known to
	  have a resource list. Revert a change I previously made, where
	  Asterisk could point to a freed memory location.

2007-10-05 19:42 +0000 [r84851]  Tilghman Lesher <tlesher@digium.com>

	* main/db.c: Log exactly why we can't open the database, if we fail
	  (closes issue #10887)

2007-10-05 18:55 +0000 [r84818]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Update the remembered RTP peer information when
	  putting an endpoint on hold or taking it off hold so that the RTP
	  stack does not initiate a needless reinvite. (closes issue
	  #10868) Reported by: mavince

2007-10-05 16:44 +0000 [r84783]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Do deadlock avoidance in a couple more
	  places. You can't lock two channels at the same time without
	  doing extra work to make sure it succeeds. (closes issue #10895,
	  patch by me)

2007-10-05  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.12.1 released. (This is mainly to include the 
	  app_queue fix for a memory leak on reload, but includes a couple
	  of other bug fixes, as well.)

2007-10-05 01:39 +0000 [r84742]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix a copy/paste error in the description of
	  UpdateConfig that was pointed out by JerJer on #asterisk-dev

2007-10-04 21:57 +0000 [r84692]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Don't allocate space for queue members unless
	  it's needed. You end up deleting dynamic members on a reload. Not
	  good. closes issue (#10879, reported by dazza76, patched by me)

2007-10-04 21:36 +0000 [r84690]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: callers of sig2str already add the word
	  'signalling' in the appropriate place, so don't duplicate it

2007-10-04 14:51 +0000 [r84637]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Create a duplicate of the channel's member name
	  as the tab completion stuff will free it. (closes issue #10884)
	  Reported by: adamg

2007-10-03 22:59 +0000 [r84581]  Tilghman Lesher <tlesher@digium.com>

	* main/rtp.c: When an RFC 2833 event is sent that we don't
	  recognize, ignore it, don't queue a NULL digit (closes issue
	  #10877)

2007-10-03 18:20 +0000 [r84511-84544]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
	  in a switch expr doesn't execute correctly, because the commas in
	  the function args are not converted to vertbars before the func
	  is called. I modified just the switch code to convert the commas
	  to vertbars if there, but if more of these sort of probs are
	  found, I may have to resort to something a little more
	  fundamental. We'll see, I guess.

	* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c,
	  pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
	  null input to a switch statement results in a hangup; since
	  switch is implemented with extensions, and the default case is
	  implemented with a '.', and the '.' matches 1 or more remaining
	  characters, the case where 0 characters exist isn't matched, and
	  the extension isn't matched, and the goto fails, and a hangup
	  occurs. Now, when a default case is generated, it also generates
	  a single fixed extension that will match a null input. That
	  extension just does a goto to the default extension for that
	  switch. I played with an alternate solution, where I just tack an
	  extra char onto all the patterns and the goto, but not the
	  default case's pattern. Then even a null input will still have at
	  least one char in it. But it made me nervous, having that extra
	  char in , even if that's a pretty secret and low-level issue.

2007-10-02  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.12 released.

2007-10-02 20:06 +0000 [r84474]  Russell Bryant <russell@digium.com>

	* Makefile, build_tools/prep_tarball: * Don't build the
	  menuselect-tree for the tarball, as it requires running the
	  configure script first * Change the Makefile to note that
	  menuselect-tree depends on the configure script.

2007-10-02 19:01 +0000 [r84410-84437]  Jason Parker <jparker@digium.com>

	* res/res_features.c: Fix some odd formatting I missed..

	* res/res_features.c: Finish up on transferee channel before return
	  on failure. Issue 10821, patch by Ivan

2007-10-02 14:12 +0000 [r84370]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Use snprintf instead of sprintf in one
	  place. There is no vulnerability here due to various buffer sizes
	  around the code, but I still didn't like seeing a non
	  length-limited copy of data coming off of the wire into a stack
	  buffer, as this would be a problem in the future if buffer sizes
	  elsewhere got changed or size limitations removed ...

2007-10-02 09:48 +0000 [r84345]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: terminate USERUSER String with 0

2007-10-01 21:52 +0000 [r84291]  Jason Parker <jparker@digium.com>

	* Makefile, Makefile.rules, channels/Makefile: Add dist-clean
	  support for subdirs. Change h323 to only remove the Makefile on a
	  dist-clean, rather than a clean. This fixes a bug I found with
	  trying to run make after a make clean

2007-10-01 21:25 +0000 [r84274]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/channel.c, main/manager.c, channels/chan_agent.c: moved
	  get_base_channel() code from action_redirect to
	  ast_channel_masquerade() for issue 7706 and BE-160

2007-10-01 21:18 +0000 [r84273]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...

2007-10-01 21:07 +0000 [r84271]  Russell Bryant <russell@digium.com>

	* main/utils.c, include/asterisk/lock.h: Fulfull a feature request
	  from Qwell on the "core show locks" output. It will now note the
	  lock type for each lock that a thread holds. (mutex, rdlock, or
	  wrlock)

2007-10-01 20:27 +0000 [r84239]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
	  #10777 -- by returning a null for the parse tree when there's
	  really nothing there, and making sure we don't try to do checking
	  on a null tree.

2007-10-01 19:56 +0000 [r84166-84236]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Add another sanity check in the AGI read loop. We
	  really don't care about EAGAIN unless we didn't read an entire
	  line. If there is a newline at the end if the read buffer, break,
	  because we got the whole thing. (reported and patched by bmd)

	* include/asterisk/lock.h: Show rwlocks in the "core show locks"
	  output. Before, it only showed mutexes.

	* channels/Makefile: Remove another file in "make clean". (closes
	  issue #10814, paravoid)

	* apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.

2007-10-01 14:10 +0000 [r84158-84163]  Joshua Colp <jcolp@digium.com>

	* configs/usbradio.conf.sample (removed): Remove chan_usbradio
	  config file from tree, it is not present in here. (closes issue
	  #10839) Reported by: casper

	* res/res_musiconhold.c: Fix randomness. save_pos was being set to
	  0 initially instead of -1, causing it to jump to position 0 when
	  moh started. (closes issue #10859) Reported by: jamesgolovich
	  Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
	  (license 176)

	* apps/app_dial.c: Only attempt early bridging if the options given
	  to Dial() permit it. (closes issue #10861) Reported by: peekyb

2007-09-30 20:02 +0000 [r84146]  Russell Bryant <russell@digium.com>

	* include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
	  modules. The load and reload parameters were in the wrong place.
	  (closes issue #10846, alebm)

2007-09-29 23:00 +0000 [r84133-84135]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
	  (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
	  pbx/ael/ael-test/ref.ael-test3,
	  pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
	  pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
	  pbx/ael/ael-test/ael-ntest22/qq.ael (added),
	  pbx/ael/ael-test/ael-ntest22/t1 (added),
	  pbx/ael/ael-test/ael-ntest22/t2 (added),
	  pbx/ael/ael-test/ael-ntest22/t3 (added),
	  pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
	  pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
	  that matches what I did in 84134 for AEL regressions.

	* pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
	  10786; All config files support #include with globbing (you know,
	  *,[chars],?,{list,list},etc), so I've updated the AEL system to
	  support this also.

2007-09-28 14:13 +0000 [r84049-84078]  Tilghman Lesher <tlesher@digium.com>

	* main/say.c: Correct pronunciations of numbers for .nl (Closes
	  issue #10837)

	* main/channel.c: Avoid a deadlock with ALL of the locks in the
	  masquerade function, not just the pairs of channels. (Closes
	  issue #10406)

2007-09-27 23:12 +0000 [r84018]  Dwayne M. Hubbard <dhubbard@digium.com>

	* main/manager.c, channels/chan_agent.c,
	  include/asterisk/channel.h: if an Agent is redirected, the base
	  channel should actually be redirected. This was causing multiple
	  issues, especially issue 7706 and BE-160

2007-09-27 00:01 +0000 [r83976]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: remove a todo item that has been completed

2007-09-26 23:53 +0000 [r83974]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_alsa.c: avoid the weird usage of assert() in the
	  ALSA header files that gcc 4.2 wants to complain about

2007-09-26 21:35 +0000 [r83910-83943]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: I changed my mind ... I think this should be
	  a LOG_NOTICE.

	* channels/chan_sip.c: Add a log message that was requested by the
	  masses in the developer tutorial session at Astricon. chan_sip
	  did not output any message when a call was rejected because the
	  extension was not found. This adds a verbose message (at verbose
	  level 3) to note when this happens.

	* channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
	  (please run the configure script with --enable-dev-mode so this
	  doesn't happen again ...)

2007-09-26 18:35 +0000 [r83879]  Tilghman Lesher <tlesher@digium.com>

	* channels/chan_zap.c: Remove unused 4k of memory on the program
	  stack (closes issue #10827)

2007-09-25 14:13 +0000 [r83637-83773]  Tilghman Lesher <tlesher@digium.com>

	* main/app.c: jmls pointed out that unsetting the group and setting
	  the group to the blank string aren't quite the same.

	* build_tools/make_defaults_h: In the source, keys are relative to
	  the datadir, not varlib (which is the same in most cases, but
	  it's good to be accurate). Closes issue #10811

	* doc/realtime.txt: Oops. Removed the unworkable workaround. This
	  note should never have been in the release.

	* main/app.c: Making change to group splitting, as discussed on the
	  -dev list. The main effect of this will be to permit
	  Set(GROUP([cat])=), i.e. unsetting a group.

2007-09-24 07:54 +0000 [r83620]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed round_robin group dial method, this
	  never worked well on BRI Ports (2 channels)

2007-09-22 19:39 +0000 [r83558-83589]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
	  are made here for the first arg in the for(arg1; arg2; arg3) {}
	  statement, as were done for the 3rd arg. It can now be an
	  assignment that will embedded in a Set() app, or a macro call, or
	  an app call.

	* pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
	  statement is now wrapped in Set() only if there's an '=' in that
	  string. Otherwise, if it begins with '&', then a Macro call is
	  generated; otherwise it is made into an app call. A bit more
	  accomodating, keeps the new guys happy, and the guys with ael-1
	  code should be happy, too

2007-09-21 14:37 +0000 [r83432]  Russell Bryant <russell@digium.com>

	* main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
	  channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
	  main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
	  pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
	  channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
	  main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
	  main/db1-ast/hash/hash.c, include/asterisk/channel.h,
	  channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
	  with cosnt pointers. This set of changes gets all of Asterisk
	  (minus chan_alsa for now) to compile with gcc 4.2. (closes issue
	  #10774, patch from qwell)

2007-09-21 13:34 +0000 [r83400]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix video under certain circumstances. It
	  would have been possible for the formats on the channel to not
	  contain the video format. (closes issue #10782) Reported by:
	  cwhuang

2007-09-20 21:16 +0000 [r83316-83348]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: When daemonizing, don't change working directory
	  to "/". It makes it not be able to do a core dump when not
	  running as uid=root. (closes issue #10766, xrg)

	* contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
	  ask for /bin/bash, as it uses bashisms. (closes issue #10772,
	  reported by culrich)

2007-09-20 17:09 +0000 [r83246]  Jason Parker <jparker@digium.com>

	* apps/app_disa.c: If # is pressed after dialing an extension in
	  DISA, stop trying to collect more digits. (issue #10754) Reported
	  by: atis Patches: app_disa.c.branch.patch uploaded by atis
	  (license 242) app_disa.c.trunk.patch uploaded by atis (license
	  242)

2007-09-20 16:25 +0000 [r83230-83232]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make sure the minimum T1 timer value is
	  obeyed in all cases. (closes issue #10768) Reported by: flefoll
	  Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
	  (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
	  flefoll (license 244)

	* channels/chan_sip.c: Fix a minor spelling error. (closes issue
	  #10769) Reported by: flefoll Patches:
	  chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
	  244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
	  (license 244)

2007-09-19 19:50 +0000 [r83121-83179]  Russell Bryant <russell@digium.com>

	* apps/app_system.c: The System() and TrySystem() applications can
	  take a substantial amount of time to execute while not servicing
	  the channel. So, put the channel in autoservice while the command
	  is being executed. (closes issue #10726, reported by mnicholson)

	* funcs/func_curl.c: Using curl can take a substantial amount of
	  time, so the channel should be autoserviced while waiting for it
	  to complete. (closes issue #10725, reported by mnicholson)

	* channels/chan_iax2.c: When handling a reload of chan_iax2, don't
	  use an ao2_callback() to POKE all peers. Instead, use an
	  iterator. By using an iterator, the peers container is not locked
	  while the POKE is being done. It can cause a deadlock if the
	  peers container is locked because poking a peer will try to lock
	  pvt structs, while there is a lot of other code that will hold a
	  pvt lock when trying to go lock the peers container. (reported to
	  me directly by Loic Didelot. Thank you for the debug info!)

	* main/manager.c: Fix up another potential race condition. Do the
	  loop decrementing use count on events with the eventq protected
	  from being changed. (reported on IRC by Ivan)

2007-09-19 13:47 +0000 [r83070-83074]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Protect the CDR record from modification by
	  pbx_exec so that the application data contains the Queue data.
	  (closes issue #10761) Reported by: snar Patches:
	  app-queue-mixmonitor.patch uploaded by snar (license 245)

	* channels/chan_sip.c: (closes issue #10760) Reported by: dimas
	  Patches: chan_sip.patch uploaded by dimas (license 88) Read in
	  subscribecontext option in general to be the default.

2007-09-19 09:32 +0000 [r83023-83024]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: removed comment which violates the coding
	  guidelines.

	* channels/misdn_config.c, channels/chan_misdn.c,
	  channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
	  allow configuring the asterisk dtmf detector instead of the
	  mISDN_dsp ones. also added the patch from irroot #10190, so that
	  dtmf tones detected by the asterisk detector are passed outofband
	  to asterisk, to make any use of dtmf tones at all.

2007-09-19 00:19 +0000 [r82992]  Russell Bryant <russell@digium.com>

	* apps/app_flash.c: Change the description of app_flash to note how
	  it can be a useful tool instead of just saying that it is
	  generally a worthless feature. (Thanks to Jim Van Meggelen for
	  pointing it out and providing the proposed text)

2007-09-18 23:41 +0000 [r82961]  Joshua Colp <jcolp@digium.com>

	* apps/app_queue.c: Initialize a variable to NULL to make the world
	  happy.

2007-09-18 22:42 +0000 [r82929]  Russell Bryant <russell@digium.com>

	* include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
	  interrupting the fgets() call when using FastAGI. This version of
	  the patch maintains the original behavior of the code when not
	  using FastAGI. (closes issue #10553) Reported by: juggie Patches:
	  res_agi_fgets-4.patch uploaded by juggie (license 24)
	  res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
	  mods by me Tested by: juggie, festr

2007-09-18 21:49 +0000 [r82887-82913]  Doug Bailey <dbailey@digium.com>

	* main/manager.c: Corrected patch applied in revision r82887.

	* main/manager.c: Fixed a bug where http manager sessions prevented
	  the eventq from being cleaned out because http manager sessions
	  do not have a valid file descriptor.

2007-09-18 20:56 +0000 [r82867]  Russell Bryant <russell@digium.com>

	* main/manager.c: Fix a memory leak that can occur on systems under
	  higher load. The issue is that when events are appended to the
	  master event queue, they use the number of active sessions as a
	  use count so it will know when all active sessions at the time
	  the event happened have consumed it. However, the handling of the
	  number of sessions was not properly synchronized, so the use
	  count was not always correct, causing an event to disappear
	  early, or get stuck in the event queue for forever. (closes issue
	  #9238, reported by bweschke, patch from Ivan, modified by me)

2007-09-18 20:09 +0000 [r82865]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Moving the logic for handling an empty
	  membername to the create_member function so that there is a
	  common place where this occurs instead of being spread out to
	  several different places.

2007-09-18 18:59 +0000 [r82834]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: there is no need for conditional logic to
	  select ->interface or ->membername, snince ->membername will
	  always be populated

2007-09-18 16:31 +0000 [r82802]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: When copying the contents from the wildcard
	  peer, do a deep copy instead of shallow copy so that it doesn't
	  crash when beging destroyed. (closes issue #10546, patch by me)

2007-09-18 15:28 +0000 [r82751]  Jason Parker <jparker@digium.com>

	* configs/sip.conf.sample: Correct the allowexternaldomains option
	  in SIP sample config. Issue 10753

2007-09-17 20:16 +0000 [r82594-82676]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
	  ast_localtime() instead of a couple places in app_voicemail to
	  prevent the problem everywhere instead of just a couple of
	  places. (related to issue #10746)

	* apps/app_voicemail.c: Initialize some memory to fix crashes when
	  leaving voicemail. This problem was fixed by running Asterisk
	  under valgrind. (closes issue #10746, reported by arcivanov,
	  patched by me) *** IMPORTANT NOTE: We need to check to see if
	  this same bug exists elsewhere.

	* res/res_features.c: Handle the case where there are multiple
	  dynamic features with the same digit mapping, but won't always
	  match the activated on/by access controls. In that case, the code
	  needs to keep trying features for a match. (reported by Atis on
	  the asterisk-dev list, patched by me)

2007-09-17 16:40 +0000 [r82590-82592]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: revert a change that wasn't supposed to be
	  committed... doh!

	* apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
	  where a logical member name (if specified) was not used, but
	  instead the direct interface was listed

2007-09-17 02:00 +0000 [r82514]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
	  of passing a NULL pointer into snprintf pass "". It makes Solaris
	  much happier.

2007-09-14 21:19 +0000 [r82444]  Steve Murphy <murf@digium.com>

	* main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
	  had to leave out the bit about ending the previous cdr in the
	  fork; it would destroy current implementations.

2007-09-14 21:17 +0000 [r82435]  Russell Bryant <russell@digium.com>

	* configs/zapata.conf.sample: Add a note to help clarify the value
	  set with the echocancel option. (inspired by Malcolm's blog post
	  on blogs.digium.com about HPEC)

2007-09-14 18:35 +0000 [r82396-82398]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Crap, I broke the build. Fixed.

	* apps/app_queue.c: Adding member name field to manager events
	  where they were missing before (closes issue #10721, reported by
	  snar)

2007-09-14 17:48 +0000 [r82394]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: If a channel does not have an owner, do not
	  try to set a channel variable. This will end up making the
	  channel variable global, which is not right. Closes issue #10720,
	  patch by flefoll.

2007-09-14 15:50 +0000 [r82382-82385]  Russell Bryant <russell@digium.com>

	* build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
	  checking for libusb here, so nobody has to deal with conflicts in
	  the chan_usbradio-1.4 branch every time the configure script gets
	  changed

	* channels/chan_usbradio.c (removed), channels/xpmr (removed),
	  channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
	  It can't live here because we have a strict policy to not include
	  new features in release branches. However, I'm going to merge it
	  into trunk, and I also have a special 1.4 based branch that
	  includes this module. svn co
	  http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4

2007-09-14 14:42 +0000 [r82376]  Mark Michelson <mmichelson@digium.com>

	* doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
	  (closes issue #10717, reported and patched by leedm777)

2007-09-14 01:24 +0000 [r82368]  Jim Dixon <telesistant@hotmail.com>

	* apps/app_rpt.c: Fixed problem with changes made to cdr
	  functionality

2007-09-14 00:52 +0000 [r82367]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_usbradio.c: this new driver may not live in this
	  branch for long (since it is a new feature), but it definitely
	  should not be built by default

2007-09-14 00:34 +0000 [r82366]  Jim Dixon <telesistant@hotmail.com>

	* apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
	  channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
	  channels/xpmr (added), channels/xpmr/LICENSE (added),
	  channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
	  (added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
	  channel driver for USB Radio device and support thereof.

2007-09-13 23:11 +0000 [r82358]  Jason Parker <jparker@digium.com>

	* pbx/pbx_spool.c: Fix a small typo. retrytime > waittime

2007-09-13 20:16 +0000 [r82346]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Preemptively fixing a possible segfault. It is
	  possible that queuename is NULL (meaning pause ALL queues), so
	  use q->name instead.

2007-09-13 20:11 +0000 [r82344]  Jason Parker <jparker@digium.com>

	* cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
	  mutliple threads tried to close the same file. Do we actually
	  need the locking here? What happens if you open the same file
	  twice, and two threads try to write to it at the same time? Is
	  fputs() going to write out the entire line at once? I suspect
	  that it could be possible for the second fopen to run during the
	  first fputs, so the position could be in the middle of the
	  previously written line... Issue 10347, initial patch by
	  explidous (but I removed all of the paranoia stuff..)

2007-09-13 18:57 +0000 [r82337-82339]  Russell Bryant <russell@digium.com>

	* main/astobj2.c: resolve a warning when not building under dev
	  mode

	* main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
	  in tracking astobj2 statistics if dev-mode is enabled. Also, when
	  dev mode is enabled, register the CLI command that can be used to
	  run the astobj2 test and print out statistics.

2007-09-13 18:12 +0000 [r82335]  Kevin P. Fleming <kpfleming@digium.com>

	* /, LICENSE: Merged revisions 82334 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
	  | 2 lines clarify the OpenSSL and OpenH323 license exceptions
	  ........

2007-09-13 16:25 +0000 [r82326]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Added logic to handle the unlikely case that
	  someone has two queues with the same name. Asterisk will log a
	  warning message letting the user know that one was already
	  defined with that name and is it skipping all further instances.
	  This also will work for realtime queues but in order for that to
	  happen, the user would have to trigger a perfectly timed reload
	  as a realtime queue is being looked up, which is highly unlikely
	  (but taken care of nonetheless).

2007-09-13 11:47 +0000 [r82309]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Closes issue #9401, reported and patched
	  by irrot, with slight modifications by me. Handle DTMF sent by
	  Asterisk properly.

2007-09-12 21:56 +0000 [r82296]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: Fix a check of the wrong pointer, as pointed out
	  by an XXX comment left in the code. The problem was harmless,
	  however.

2007-09-12 21:28 +0000 [r82291]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
	  files

2007-09-12 20:24 +0000 [r82286]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_meetme.c: remove a race condition for the creation of
	  recordthread's, and fix a small memory leak. This closes issue#
	  10636

2007-09-12 20:12 +0000 [r82285]  Tilghman Lesher <tlesher@digium.com>

	* main/stdtime/private.h, main/stdtime/tzfile.h,
	  include/asterisk/localtime.h, main/stdtime/localtime.c: Working
	  on issue #10531 exposed a rather nasty 64-bit issue on
	  ast_mktime, so we updated the localtime.c file from source. Next
	  we'll have to write ast_strptime to match.

2007-09-12 15:16 +0000 [r82278-82280]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Clean up the output of "asterisk -h". This
	  tweaks the wording and wraps lines at 80 characters. (closes
	  issue #10699, seanbright)

	* res/res_agi.c: revert patch from issue #10553, as someone not
	  using fastagi reported that this broke their system.

2007-09-12 14:30 +0000 [r82274-82276]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Accidentally committed changes to
	  app_voicemail which do NOT need to be in the 1.4 branch yet.
	  reverting...

	* apps/app_voicemail.c, apps/app_queue.c: We should only initialize
	  a realtime queue when it is allocated, not every time we access
	  it. This prevents the members ao2_container from being
	  reallocated every time the queue is accessed. I also removed a
	  debug message I had accidentally left in on a previous commit.

2007-09-11 22:37 +0000 [r82267]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
	  these calls was reading bogus memory ...

2007-09-11 21:41 +0000 [r82265]  Joshua Colp <jcolp@digium.com>

	* codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
	  #10679) Reported by: andrew Build under dev mode when K6OPTS is
	  enabled.

2007-09-11 20:49 +0000 [r82263]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix another missing unref of member objects.
	  This one was pointed out by Marta. When building the outgoing
	  list in try_calling(), a member reference is stored in each
	  outgoing entry. However, when this list got destroyed, the
	  reference was not released.

2007-09-11 20:36 +0000 [r82261]  Steve Murphy <murf@digium.com>

	* main/cdr.c: this change should fix issue # 10659 -- what I worry
	  about is how many other bug reports it may generate. Hopefully,
	  we can please the/a majority. Hopefully. We shall see. Calls not
	  marked ANSWERED and with only one channel name will not be
	  posted. This should eliminate the double CDR's.

2007-09-11 16:05 +0000 [r82252]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: All instances of ao2_iterators which were just
	  named 'i' have been renamed to 'mem_iter' so that when refcounted
	  queues are merged into trunk, there will be little confusion
	  regarding iterator names, especially when a queue and member
	  iterator are used in the same function.

2007-09-11 16:03 +0000 [r82250]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: The sample dundi.conf claims support for a
	  wildcard peer entry - [*], but the code did not support it. This
	  patch makes it work. (closes issue #10546, patch by dds, with
	  some changes by me)

2007-09-11 16:01 +0000 [r82249]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
	  hold/retrieve issue.

2007-09-11 15:26 +0000 [r82245]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
	  res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
	  juggie When using fastagi, fgets() can return before a full line
	  is read. Add explicit handling for the case where it gets
	  interrupted.

2007-09-11 14:56 +0000 [r82243]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
	  Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
	  (license 176) Don't leak memory when unloading DUNDi.

2007-09-11 14:34 +0000 [r82198-82240]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Add a couple more missing unrefs of queue
	  member objects

	* apps/app_queue.c: Add a missing unref of a queue member in an
	  error handling block

	* apps/app_queue.c: Document why membercount can not simply be
	  replaced by ao2_container_count()

	* main/astobj2.c: backport astobj2 race condition fix. This
	  function is the exact same as trunk so it applies here as well.

2007-09-10 18:02 +0000 [r82155]  Tilghman Lesher <tlesher@digium.com>

	* apps/app_queue.c: Convert struct member to use refcounts (closes
	  issue #10199)

2007-09-10 15:02 +0000 [r82091]  Mark Michelson <mmichelson@digium.com>

	* configs/misdn.conf.sample: Removing non-existent options from
	  misdn configuration sample. (closes issue #10678, reported and
	  patched by IgorG)

2007-09-09 02:35 +0000 [r82028]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Fix inline compiles on really old
	  compilers (who uses gcc 2.7 anymore, really?)

2007-09-08 18:41 +0000 [r81952-81997]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
	  did not free the entry it removed.

	* .cleancount: (closes issue #10672) Bump the cleancount so that a
	  "make clean" will be forced. This is needed because my fix in
	  revision 81599 made a change to a data structure in file.h, and
	  since file dependency tracking is only on with dev-mode enabled,
	  file format modules that don't get rebuilt may crash, as is the
	  case with this issue. This makes me wonder - how much faster does
	  the code build without the file dependency tracking enabled? If
	  it doesn't make much of a difference, then it may be worth just
	  keeping it on all of the time, or perhaps just not in release
	  tarballs, so that this type of issue is avoided.

2007-09-07 19:48 +0000 [r81923]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
	  as the mixmonitor filename. This moves the setting of the
	  MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
	  by sim.

2007-09-07 15:25 +0000 [r81886]  Mark Michelson <mmichelson@digium.com>

	* configs/queues.conf.sample: Moving the explanation for joinempty
	  to a more appropriate place

2007-09-06 22:28 +0000 [r81832]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: (closes issue #9724, closes issue #10374)
	  Reported by: kenw Patches: 9724.txt uploaded by russell (license
	  2) Tested by: kenw, russell Resolve a deadlock that occurs when
	  doing a SIP transfer to parking. I come across this type of
	  deadlock fairly often it seems. It is very important to mind the
	  boundary between the channel driver and the core in respect to
	  the channel lock and the channel-pvt lock. Channel drivers lock
	  to lock the pvt and then the channel once it calls into the core,
	  while the core will do it in the opposite order. The way this is
	  avoided is by having channel drivers either release their pvt
	  lock while calling into the core, or such as in this case,
	  unlocking the pvt just long enough to acquire the channel lock.

2007-09-06 22:05 +0000 [r81778-81826]  Jason Parker <jparker@digium.com>

	* Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
	  for ASTLDFLAGS. This adds LDOPTS

	* include/asterisk/astobj2.h: This should fix a build issue that
	  people building against uClibc were seeing with the addition of
	  astobj2

2007-09-06 19:40 +0000 [r81776]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: (closes issue #10122) Reported by:
	  stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
	  qwell (license 4) Tested by: stevefeinstein After looking over
	  the code I agree with Qwell. Setting the file descriptor to
	  conference each time just causes a fight back and forth.

2007-09-06 16:56 +0000 [r81743]  Philippe Sultan <philippe.sultan@gmail.com>

	* include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
	  length fixes. Removed an unused variable in aji_client structure
	  (context)

2007-09-06 16:25 +0000 [r81682-81713]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixes an issue where valid DTMF had to be
	  pressed twice to exit a queue if a member's phone was ringing.
	  (closes issue #10655, reported by strider2k, patched by me)

	* res/res_features.c: Fixes a memory leak (closes issue #10658,
	  reported and patched by Ivan)

2007-09-06 14:20 +0000 [r81650]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
	  and Google's XMPP server complaint, if set, the 'from' attribute
	  must be set to the user's full JID.

2007-09-05 20:53 +0000 [r81599]  Russell Bryant <russell@digium.com>

	* include/asterisk/file.h, main/say.c, res/res_features.c,
	  main/file.c, include/asterisk/channel.h: Fix an issue that can
	  occur when you do an attended transfer to parking. If you
	  complete the transfer before the announcement of the parking spot
	  finishes, then the channel being parked will hear the remainder
	  of the announcement. These changes make it so that will not
	  happen anymore. Basically, res_features sets a flag on the
	  channel is playing the announcement to so that the file streaming
	  core knows that it needs to watch out for a channel masquerade,
	  and if it occurs, to abort the announcement. (closes BE-182)

2007-09-05 17:18 +0000 [r81569]  Tilghman Lesher <tlesher@digium.com>

	* include/asterisk/lock.h: Solaris x86 compatibility fix

2007-09-05 15:19 +0000 [r81525]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixing the build...

2007-09-05 15:14 +0000 [r81523]  Jason Parker <jparker@digium.com>

	* channels/chan_phone.c: Do not try to unregister a NULL channel
	  tech. Also changed load_module function to use defines rather
	  than numbers for return values. Issue 10651, patch by
	  rbraun_proformatique, with additions by me.

2007-09-05 15:03 +0000 [r81520]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
	  only count members who are logged in and available. (related to
	  issue #10652, reported by wuwu)

2007-09-05 13:11 +0000 [r81492]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: (closes issue #10650) Reported by: tacvbo Only
	  print out that the spy was removed while holding the spy lock.

2007-09-04 20:54 +0000 [r81453-81455]  Jason Parker <jparker@digium.com>

	* apps/app_followme.c: Rather than attempt to play a file, we can
	  just check whether it exists. Issue 10634, patch by me, testing
	  by pabelanger, sanity checked by bweschke

	* configs/followme.conf.sample: Change default followme config file
	  to point to the correct files. Issue 10644, patch by pabelanger

2007-09-04 18:37 +0000 [r81448]  Russell Bryant <russell@digium.com>

	* main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
	  Remove the typedefs on ao2_container and ao2_iterator. This is
	  simply because we don't typedef objects anywhere else in
	  Asterisk, so we might as well make this follow the same
	  convention.

2007-09-04 16:40 +0000 [r81442]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: there is no point in sending 401
	  Unauthorized to a UAS that sent us a properly-formatted
	  Authentication header with the expected username and nonce but an
	  incorrect response (which indicates the shared secret does not
	  match)... instead, let's send 403 Forbidden so that the UAS
	  doesn't retry with the same authentication credentials repeatedly

2007-09-04 14:23 +0000 [r81435-81439]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: (closes issue #10632) Reported by:
	  jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
	  by jamesgolovich (license 176) Fix memory leak when unloading
	  chan_iax2. The firmware files were not being freed.

	* main/channel.c: (closes issue #10476) Reported by: mdu113 Only
	  look for the end of a digit when waiting for a digit. This in
	  turn disables emulation in the core.

	* main/dns.c: (closes issue #10610) Reported by: john Patches:
	  dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
	  Don't return a match if no SRV record actually exists.

2007-09-03 18:57 +0000 [r81433]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Remove a couple of calls to
	  ast_string_field_free_pools() on peers in error handling blocks
	  in the code for building peers. The peer object destructor does
	  this and doing it twice will cause a crash. (closes issue #10625,
	  reported by and patched by pnlarsson)

2007-09-01 15:57 +0000 [r81426-81428]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changed a comment to be more accurate. (really
	  this is just a test to make sure I can commit properly from home)

	* main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
	  into ao2_match_by_addr and making it available everywhere since
	  it could be a handy callback to have

2007-08-31 21:27 +0000 [r81418]  Russell Bryant <russell@digium.com>

	* include/asterisk/astobj2.h: Remove references to a debugging
	  parameter that does not exist

2007-08-31 19:48 +0000 [r81416]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixed broken behavior of a reload on realtime
	  queues. Prior to this patch, if a reload was issued and a
	  realtime queue had callers waiting in it, then the queue would be
	  removed from the queue list, but it would not actually be freed
	  (in fact, a debug message warning about a memory leak would come
	  up). With this patch, reloads do not touch realtime queues at
	  all.

2007-08-31 19:16 +0000 [r81415]  Tilghman Lesher <tlesher@digium.com>

	* funcs/func_logic.c: The IF() function was not allowing true
	  values that had embedded colons (closes issue #10613)

2007-08-31 18:44 +0000 [r81412]  Jason Parker <jparker@digium.com>

	* apps/app_dial.c: Re-order dial options to be in line with the
	  existing alpha order. Issue 10621, initial patch by junky

2007-08-31 17:38 +0000 [r81410]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
	  available. Closes issue 10548, reported by keepitcool.

2007-08-31 15:53 +0000 [r81406]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c: Make it the engine's responsible to check for
	  the presence of results.

2007-08-31 15:51 +0000 [r81405]  Kevin P. Fleming <kpfleming@digium.com>

	* codecs/codec_zap.c: add missing "transcoder show" (and deprecated
	  "show transcoder") CLI commands that were in 1.2 but never added
	  to 1.4

2007-08-31 14:38 +0000 [r81401-81403]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10618) Reported by: dimas
	  Don't pass through the stopped sounds frame.... just drop it.

	* res/res_features.c: (closes issue #10009) Reported by: dimas
	  Don't output a bridge failed warning message if it failed because
	  one of the channels was part of the masquerade process. That is
	  perfectly normal.

2007-08-30 22:05 +0000 [r81397]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Removing an extraneous (and possibly
	  misleading) log message. Firstly, if the announce file isn't
	  found, the streaming functions will report it. Secondly, not all
	  non-zero returns from play_file mean that the announce file
	  wasn't found. Positive return values simply mean that a digit was
	  pressed (most likely to skip through the announcement). (closes
	  issue #10612, reported and patched by dimas)

2007-08-30 21:23 +0000 [r81395]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10514) Reported by: casper
	  Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
	  Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
	  for it to ever be that value.

2007-08-30 21:11 +0000 [r81392]  Steve Murphy <murf@digium.com>

	* main/cdr.c: via issue 10599, where 'CDR already initialized'
	  messages are being generated. Since all channels will have an
	  init'd CDR attached at creation time, this message is now
	  particularly useless. Removed.

2007-08-30 15:38 +0000 [r81383]  Russell Bryant <russell@digium.com>

	* channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
	  define. (closes issue #10559, paravoid)

2007-08-30 15:35 +0000 [r81381]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Changed some manager event messages to reflect
	  whether a queue member is a realtime member or not

2007-08-30 15:33 +0000 [r81379]  Russell Bryant <russell@digium.com>

	* configs/modem.conf.sample (removed), configs/enum.conf.sample,
	  configs/extensions.ael.sample: Fix a typo, update a reload
	  command, and remove an unused configuration file. (closes issue
	  #10606, casper)

2007-08-30 14:53 +0000 [r81375]  Joshua Colp <jcolp@digium.com>

	* main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
	  pbx.diff uploaded by jmls (license 141) Backport changes from
	  81372. Add REASON dialplan variable for when an originated call
	  fails and the failed extension is executed.

2007-08-30 14:43 +0000 [r81373]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: Fixed some warnings.

2007-08-30 14:23 +0000 [r81369]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (issue #10599) Reported by: dimas Handle the
	  -1 control subclass during feature dialing (it indicates to stop
	  sounds).

2007-08-30 08:31 +0000 [r81367]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
	  issue where a misdn_read would lock the channel, but read would
	  not return because it blocks. later chan_misdn would try to queue
	  a frame like a AST_CONTROL_ANSWER which could result in a
	  deadlock situation. misdn_read will now not block forever
	  anymore, and we don't queue the ANSWER frame at all when we
	  already was called with misdn_answer -> answer would be called
	  twice. Also we don't explicitly send a RELEASE_COMPLETE on
	  receiption of a RELEASE anymore, because mISDN does that for us,
	  this resulted in a problem on some switches, which would block
	  our port after some calls for a short while.

2007-08-29 16:35 +0000 [r81346-81349]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This patch, in essence, will correctly pause a
	  realtime queue member and reflect those changes in the realtime
	  engine. (issue #10424, reported by irroot, patch by me) This
	  patch creates a new function called update_realtime_member_field,
	  which is a generic function which will allow any one field of a
	  realtime queue member to be updated. This patch only uses this
	  function to update the paused status of a queue member, but it
	  lays the foundation for persisting the state of a realtime member
	  the same way that static members' state is maintained when using
	  the persistentmembers setting

	* apps/app_queue.c: Changed some tabs to spaces

2007-08-29 15:57 +0000 [r81342]  Russell Bryant <russell@digium.com>

	* main/Makefile: If chan_h323 is not being built, don't use g++ to
	  do the final link of Asterisk. (in response to a question on the
	  asterisk-dev list)

2007-08-29 15:52 +0000 [r81340]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: This fix creates a more accurate way of
	  detecting whether realtime members were deleted. (closes issue
	  10541, reported by Alric, patched by me) The REALLY nice things
	  about this patch is that queue members now have a "realtime"
	  field which will be true if the member is a realtime member. This
	  means we can check this value prior to certain processing if it
	  should ONLY be done for realtime members.

2007-08-29 14:13 +0000 [r81331]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
	  rtp timeouts work even if two RTP streams are directly bridged in
	  the RTP stack.

2007-08-28 21:38 +0000 [r81226-81291]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Change the message about receiving a
	  mini-frame before the first full voice frame to a DEBUG message.

	* pbx/pbx_dundi.c: revert unintentional changes in rev 81226

	* configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
	  tones. (closes issue #7953, hanabana)

2007-08-28 14:12 +0000 [r81120-81189]  Mark Michelson <mmichelson@digium.com>

	* contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
	  res_config_mysql (closes issue #10573, reported by chrisvaughan,
	  patch suggested by chrisvaughan as well)

	* apps/app_queue.c: Resolve a potential deadlock. In this case, a
	  single queue is locked, then the queue list. In changethread(),
	  the queue list is locked, and then each individual queue is
	  locked. Under the right circumstances, this could deadlock. As
	  such, I have unlocked the individual queue before locking the
	  queue list, and then locked the queue back after the queue list
	  is unlocked.

	* channels/chan_agent.c: DTMF begin frames should be ignored so
	  that when an agent acks a call with the '#' key, he doesn't cause
	  a queue's announce file to be interrupted. Also went ahead and
	  did the same for the '*' key and for ending a call. (closes issue
	  #10528, reported by deskhack, patched by me)

2007-08-27 17:27 +0000 [r81042-81074]  Russell Bryant <russell@digium.com>

	* pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
	  of the memory it allocates on unload and should be fixed (when
	  I'm not in the middle of something else ...).

	* pbx/pbx_dundi.c: explicity define a variable as a boolean

	* res/res_musiconhold.c: (closes issue #10419) Reported by:
	  mustardman Patches: asterisk-mohposition.diff.txt uploaded by
	  jamesgolovich (license 176) This patch fixes a few problems with
	  music on hold. * Fix issues with starting at the beginning of a
	  file when it shouldn't. * Fix the inuse counter to be decremented
	  even if the class had not been set to be deleted when not in use
	  anymore * Don't arbitrarily limit the number of MOH files to 255

2007-08-27 15:01 +0000 [r81012]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
	  Patches: chan_sip-ChannelReload-20080825.patch uploaded by
	  jesselang (license 202) Remove an extra \r\n to make the
	  ChannelReload event conform with every other event.

2007-08-27 14:55 +0000 [r81010]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Found a case where the queue's membercount is
	  off. It does not take into account dynamic members on a reload.

2007-08-27 13:20 +0000 [r80974]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
	  jitter value output in the CLI to be as expected.

2007-08-26 18:11 +0000 [r80932]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Remove an extra signal_condition() for the
	  scheduler thread. (closes issue #10564, patch from casper)

2007-08-25 17:37 +0000 [r80895]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some issues with the handling of the
	  scheduler in chan_iax2. Most of the places that scheduled items
	  to be executed by the scheduler thread did not signal the
	  scheduler thread to wake up so that it could recalculate the time
	  until the next action. These changes will make the scheduler
	  thread more responsive and ensure that actions get executed as
	  close to when intended as possible instead of it being possible
	  for very long delays.

2007-08-24 22:59 +0000 [r80878]  Dwayne M. Hubbard <dhubbard@digium.com>

	* apps/app_zapateller.c: An empty string is an empty callerid ...
	  so zap it. This closes issue #10502, which was pointed out by
	  dswartz. Thank you, and may the swartz be with you

2007-08-24 21:22 +0000 [r80820-80849]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
	  available when Asterisk first starts, then don't give up on
	  poking peers. Allow the poke to get rescheduled so that it will
	  work once the dnsmgr is able to resolve the host. (closes issue
	  #10521, patch by jamesgolovich)

	* main/dsp.c: Improve the debouncing logic in the DTMF detector to
	  fix some reliability issues. Previously, this code used a shift
	  register of hits and non-hits. However, if the start of the digit
	  isn't clean, it is possible for the leading edge detector to miss
	  the digit. These changes replace the flawed shift register logic
	  and also does the debouncing on the trailing edge as well.
	  (closes issue #10535, many thanks to softins for the patch)

2007-08-24 19:52 +0000 [r80818]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: A minor correction to the available logic of
	  autofill. If a queue member is paused, they're not really
	  "available" so don't count them as such. Somewhat related to
	  issue #10155

2007-08-24 18:52 +0000 [r80789]  Steve Murphy <murf@digium.com>

	* main/cdr.c: From a complaint by jmls, I realize that the message
	  in cdr_disposition is unnecessary. To get failure disposition,
	  just return -1; no use having more than one case do that.

2007-08-24 15:51 +0000 [r80750]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.

2007-08-24 15:41 +0000 [r80747]  Tilghman Lesher <tlesher@digium.com>

	* main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
	  the code, instead of only in documentation (closes issue #10549)

2007-08-24 15:28 +0000 [r80722]  Russell Bryant <russell@digium.com>

	* utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
	  I think this would have caused a "*" to get in the
	  menuselect-tree file.

2007-08-24 14:48 +0000 [r80689-80717]  Steve Murphy <murf@digium.com>

	* utils/ael_main.c: This change addresses JerJer's complaint that
	  aelparse builds and installs even if pbx_ael is unchecked in the
	  menuselect stuff.

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
	  backport of 80649, a fix to an unreported problem in the ael
	  parser, that results in a crash on a 64bit machine

2007-08-24 11:42 +0000 [r80661]  Philippe Sultan <philippe.sultan@gmail.com>

	* channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
	  answered too early, which results in CDRs wrongly stating that a
	  call was ANSWERED when the calling party cancelled a call before
	  before being established. We must not answer the call upon
	  reception of a 'transport-accept' iq packet, but this packet
	  still needs to be acknowledged, otherwise the remote peer would
	  close the call (like in #8970).

2007-08-23 21:34 +0000 [r80601-80617]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
	  warnings

	* channels/chan_misdn.c: make chan_misdn compile without warnings

2007-08-23 20:16 +0000 [r80539-80573]  Russell Bryant <russell@digium.com>

	* include/asterisk/features.h, res/res_features.c: When executing a
	  dynamic feature, don't look it up a second time by digit pattern
	  after we already looked it up by name. This causes broken
	  behavior if there is more than one feature defined with the same
	  digit pattern. (closes issue #10539, reported by bungalow, patch
	  by me)

	* funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
	  none of these values take ms so I don't know what I was thinking

	* funcs/func_timeout.c: Fix func_timeout to take values in floating
	  point so 1.5 actually means 1.5 seconds instead of being rounded.
	  (closes issue #10540, reported by spendergrass, patch by me)

2007-08-23 17:14 +0000 [r80505-80507]  Jason Parker <jparker@digium.com>

	* /: *sigh*

	* /: use autotagged externals

2007-08-23 17:08 +0000 [r80501]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: report the actual channel number that was
	  unregistered, instead of assuming that the interface list
	  consists of channels 1 through <x> with no gaps in the sequence

2007-08-23 17:02 +0000 [r80360-80499]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some code where it was possible for a
	  reference to a peer to not get released when it should. Thank you
	  to Marta Carbone for pointing this out!

	* main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
	  This is a hack to maintain old behavior of chan_iax2. This
	  ensures that if the peers and users are being stored in a linked
	  list, that they go in the list in the same order that the older
	  code used. This is necessary to maintain the behavior of which
	  peers and users get matched when traversing the container.

	* res/res_agi.c: Revert res_agi fix that didn't quite work until we
	  get it right ...

	* include/asterisk/astobj2.h: Add some more documentation on
	  iterating ao2 containers. The documentation implies that is
	  possible to miss an object or see an object twice while
	  iterating. After looking through the code and talking with
	  mmichelson, I have documented the exact conditions under which
	  this can happen (which are rare and harmless in most cases).

	* main/astobj2.c: When converting this code to use the list macros,
	  I changed it so objects are added to the head of a bucket instead
	  of the tail. However, while looking over code with mmichelson, we
	  noticed that the algorithm used in ao2_iterator_next requires
	  that items are added to the tail. This wouldn't have caused any
	  huge problem, but it wasn't correct. It meant that if an object
	  was added to a container while you were iterating it, and it was
	  added to the same bucket that the current element is in, then the
	  new object would be returned by ao2_iterator_next, and any other
	  objects in the bucket would be bypassed in the traversal.

	* channels/chan_sip.c: Don't crash when using realtime in chan_sip
	  without an insecure setting in the database. (closes issue
	  #10348, reported by link55, fixed by me)

	* main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
	  (added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
	  channels/chan_iax2.c: Merge changes from
	  team/russell/iax_refcount. This set of changes fixes problems
	  with the handling of iax2_user and iax2_peer objects. It was very
	  possible for a thread to still hold a reference to one of these
	  objects while a reload operation tries to delete them. The fix
	  here is to ensure that all references to these objects are
	  tracked so that they can't go away while still in use. To
	  accomplish this, I used the astobj2 reference counted object
	  model. This code has been in one of Luigi Rizzo's branches for a
	  long time and was primarily developed by one of his students,
	  Marta Carbone. I wanted to go ahead and bring this in to 1.4
	  because there are other problems similar to the ones fixed by
	  these changes, so we might as well go ahead and use the new
	  astobj if we're going to go through all of the work necessary to
	  fix the problems. As a nice side benefit of these changes, peer
	  and user handling got more efficient. Using astobj2 lets us not
	  hold the container lock for peers or users nearly as long while
	  iterating. Also, by changing a define at the top of chan_iax2.c,
	  the objects will be distributed in a hash table, drastically
	  increasing lookup speed in these containers, which will have a
	  very big impact on systems that have a large number of users or
	  peers. The use of the hash table will be made the default in
	  trunk. It is not the default in 1.4 because it changes the
	  behavior slightly. Previously, since peers and users were stored
	  in memory in the same order they were specified in the
	  configuration file, you could influence peer and user matching
	  order based on the order they are specified in the configuration.
	  The hash table does not guarantee any order in the container, so
	  this behavior will be going away. It just means that you have to
	  be a little more careful ensuring that peers and users are
	  matched explicitly and not forcing chan_iax2 to have to guess
	  which user is the right one based on secret, host, and access
	  list settings, instead of simply using the username. If you have
	  any questions, feel free to ask on the asterisk-dev list.

	* res/res_agi.c: Juggie in #asterisk-dev was reporting problems
	  where fgets would return without reading the whole line when
	  using fastagi. When this happens, errno was set to EINTR or
	  EAGAIN. This patch accounts for the possibility and lets fgets
	  continue in that case.

2007-08-22 18:53 +0000 [r80302-80330]  Jason Parker <jparker@digium.com>

	* Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
	  build_tools/strip_nonapi, build_tools/prep_moduledeps,
	  build_tools/make_buildopts_h: Fix a few build issues in Solaris
	  (and likely others). Use GREP and ID variables from autoconf.
	  Reported to me in #asterisk-dev I forgot who reported this -
	  sorry. :(

	* Makefile: Change a syntax that the GNU make in Solaris dislikes.

	* build_tools/make_version: Fix a bashism (we explicitly request
	  /bin/sh). Remove some oddly placed quotes I found in passing.

2007-08-22 16:21 +0000 [r80257]  Russell Bryant <russell@digium.com>

	* Makefile: Honor the contents of the COPTS variable as custom
	  target CFLAGS. Apparently this is what openwrt does. (reported by
	  Brian Capouch on the asterisk-dev list, patch by me)

2007-08-22 16:14 +0000 [r80255]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10526) Reported by: sinistermidget
	  Revert commit from issue #10355 and return timestamp skew to 640.

2007-08-21  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.11 released.

2007-08-21 18:42 +0000 [r80183]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Don't record SIP dialog history if it's not
	  turned on. Also, put an upper limit on how many history entires
	  will be stored for each SIP dialog. It is currently set to 50,
	  but can be increased if deemed necessary. (closes issue #10421,
	  closes issue #10418, patches suggested by jmoldenhauer, patches
	  updated by me) (Security implications documented in AST-2007-020)

2007-08-21 16:39 +0000 [r80166-80167]  Steve Murphy <murf@digium.com>

	* include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
	  the diffs from ulaw.h and alaw.h for now; accidentally added them
	  in 80166

	* main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
	  This patch solves problem 1 in 8126; it should not slow down the
	  alaw codec, but should prevent signal degradation via multiple
	  trips thru the codec. Fossil estimates the twice thru this codec
	  will prevent fax from working. 4-6 times thru would result
	  hearable, noticeable, voice degradation.

2007-08-21 15:22 +0000 [r80132]  Russell Bryant <russell@digium.com>

	* channels/chan_mgcp.c: Don't try to dereference the owner channel
	  when it may not exist (issue #10507, maxper)

2007-08-21 15:03 +0000 [r80130]  Jason Parker <jparker@digium.com>

	* configs/cdr.conf.sample: (issue #10510) Reported by: casper
	  Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
	  errors in sample cdr config file.

2007-08-20 21:57 +0000 [r80088]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix the build of app_queue

2007-08-20 21:39 +0000 [r80049-80086]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: After a discussion on #asterisk-dev, it was
	  decided that this should be in 1.4 as well. (issue #10424,
	  reported and patched by irroot)

	* apps/app_queue.c: Found a pointless ternary if. member->dynamic
	  was set to 1 and has no opportunity to change between then and
	  this line, so "dynamic" will ALWAYS be output.

2007-08-20 16:08 +0000 [r80047]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample: (issue #10499) Reported by:
	  casper Patches: extensions.conf.sample.diff uploaded by casper
	  (license 55) Update CLI examples in extensions.conf.sample to
	  reflect command changes.

2007-08-20 15:34 +0000 [r80044]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Ukrainian language voicemail support.
	  (closes issue #10458, reported and patched by Oleh)

2007-08-20 02:42 +0000 [r79998]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
	  snuffy via IRC)

2007-08-18 14:30 +0000 [r79947]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Don't allocate vmu for messagecount when we
	  could just use the stack instead (closes issue #10490) Also,
	  remove a useless (and leaky) SQLAllocHandle (closes issue #10480)

2007-08-17 21:01 +0000 [r79912]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Avoid a crash in the handling of DTMF based
	  Caller ID. It is valid for ast_read to return NULL in the case
	  that the channel has been hung up. (crash reported by
	  anonymouz666 on IRC in #asterisk-dev)

2007-08-17 19:14 +0000 [r79906]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Patch allows for more seamless transition
	  from file storage voicemail to ODBC storage voicemail. If a
	  retrieval of a greeting from the database fails, but the file is
	  found on the file system, then we go ahead an insert the greeting
	  into the database. The result of this is that people who switch
	  from file storage to ODBC storage do not need to rerecord their
	  voicemail greetings.

2007-08-17 19:12 +0000 [r79902-79904]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
	  Don't send a semicolon over the wire in sip notify messages.
	  Caused by fix for issue 9938. I basically took the code that
	  existed before 9938 was fixed, and copied it into a new function
	  - ast_unescape_semicolon There should be very few places this
	  will be needed (pbx_config does NOT need this (see issue 9938 for
	  details)) Issue 10430, patch by me, with help/ideas from murf
	  (thanks murf).

	* channels/chan_local.c: Re-add the setting of callerid name and
	  number. Issue 10485, reported by and fix explained by paradise.

2007-08-17 13:37 +0000 [r79857]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix some crashes in chan_sip. This patch
	  changes various places that add items to the scheduler to ensure
	  that they don't overwrite the ID of a previously scheduled item.
	  If there is one, it should be removed. (closes issue #10391,
	  closes issue #10256, probably others, patch by me)

2007-08-17 08:22 +0000 [r79833]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: sometimes we don't need to signal dtmf
	  tones to asterisk, we just want them to go through as inband.
	  Otherwise they might be generated by the other channel partner
	  and then there is a double tone.

2007-08-16 22:32 +0000 [r79756-79792]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: Fix a little race condition that could
	  cause a crash if two channels had MOH stopped at the same time
	  that were using a class that had been marked for deletion when
	  its use count hits zero.

	* res/res_musiconhold.c: This patch fixes a bug where reloading the
	  module with "module reload" did not delete classes from memory
	  that were no longer in the config. This patch fixes that problem
	  as well as another one. Previously, if you reloaded MOH using the
	  "moh reload" CLI command, which behaved differently than "module
	  reload ...", MOH had to be stopped on every channel and started
	  again immediately. However, there was no way to tell what class
	  was being used, so they would all fall back to the default class.
	  (closes issue #10139) Reported by: blitzrage Patches:
	  asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
	  (license 176) Tested by: jamesgolovich

	* channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
	  introduced by making frame handling and scheduling
	  multi-threaded. Unfortunately, we have to do some expensive
	  deadlock avoidance when queueing frames on to the ast_channel
	  owner of the IAX2 pvt struct. This was already handled for
	  regular frames, but ast_queue_hangup and ast_queue_control were
	  still used directly. Making these changes introduced even more
	  places where the IAX2 pvt struct can disappear in the context of
	  a function holding its lock due to calling a function that has to
	  unlock/lock it to avoid deadlocks. I went through and fixed all
	  of these places to account for this possibility. (issue #10362,
	  patch by me)

2007-08-16 21:16 +0000 [r79690-79748]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Fixes a problem where agents would get
	  stuck busy due to their wrapuptime being longer than the queue's
	  wrapuptime and ringinuse=no for the queue. (closes issue #10215,
	  reported by Doug, repaired by me) Special thanks to fkasumovic
	  for pointing out the source of the problem and to bweschke for
	  helping to come up with a solution!

	* apps/app_voicemail.c: base_encode is not trying to open a log
	  file, so we should not call it a log file in the warning.
	  (related to issue #10452, reported by bcnit)

2007-08-16 09:37 +0000 [r79665]  Philippe Sultan <philippe.sultan@gmail.com>

	* res/res_jabber.c: A fix for two critical problems detected while
	  working with Daniel McKeehan in issue #10184. Upon priority
	  change, the resource list is not NULL terminated when moving an
	  item to the end of the list. This makes Asterisk endlessy loop
	  whenever it needs to read the list. Jids with different resource
	  and priority values, like in Gmail's and GoogleTalk's jabber
	  clients put that problem in evidence. Upon reception of a 'from'
	  attribute with an empty resource string, Asterisk crashes when
	  trying to access the found->cap pointer if the resource list for
	  the given buddy is not empty. This situation is perfectly valid
	  and must be handled. The Gizmoproject's jabber client put that
	  problem in evidence. Also added a few comments in the code as
	  well as a handle for the capabilities from Gmail's jabber client,
	  which are stored in a caps:c tag rather than the usual c tag.
	  Closes issue #10184.

2007-08-16 08:21 +0000 [r79642]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
	  we want to send IA5 Chars.

2007-08-15 14:40 +0000 [r79553]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10440) Reported by: irroot (closes
	  issue #10454) Reported by: flo_turc Increase maximum timestamp
	  skew to 120. 20 was apparently far too low.

2007-08-15 14:26 +0000 [r79527]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed an error in the Russian language
	  voicemail intro. (issue #10458, reported and patched by Oleh)

2007-08-15 14:18 +0000 [r79523]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: (closes issue #10456) Reported by: irroot
	  Patches: sip_timeout.patch uploaded by irroot (license 52) Change
	  hardcoded timer value to defined value. I'm doing this in 1.4 as
	  well so if it needs to be changed in the future this place would
	  not have been forgotten.

2007-08-14 18:49 +0000 [r79436-79470]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix another spot where an iax2_peer would
	  be leaked if realtime was in use.

	* channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
	  related to the use of realtime. I found these while working on
	  iax2_peer object reference tracking.

2007-08-14 15:27 +0000 [r79397]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10415) Reported by: atis
	  Revert fix for #10327 as it causes more issues then it solves.

2007-08-13 22:40 +0000 [r79363]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: memset really, really needs to be used here.

2007-08-13 21:57 +0000 [r79334]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, apps/app_speech_utils.c,
	  include/asterisk/speech.h: Instead of accepting a single DTMF
	  character accept a full string.

2007-08-13 20:37 +0000 [r79272-79301]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Don't call find_peer in
	  registry_authrequest with the pvt lock held to avoid a deadlock.

	* channels/chan_iax2.c: Release the pvt lock before calling
	  find_peer in register_verify to avoid a deadlock. Also, remove
	  some unnecessary locking in auth_fail that was only done
	  recursively.

	* channels/chan_iax2.c: Don't call find_peer within update_registry
	  with a pvt lock held. This can cause a deadlock as the code will
	  eventually call find_callno.

	* channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
	  have tracked them down to a single core issue. You can not call
	  find_callno() while holding a pvt lock as this function has to
	  lock another (every) other pvt lock. Doing so can lead to a
	  classic deadlock. So, I am tracking down all of the code paths
	  where this can happen and fixing them. The fix I committed
	  earlier today was along the same theme. This patch fixes some
	  code down the path of authenticate_reply.

2007-08-13 17:49 +0000 [r79255]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest21 (added),
	  pbx/ael/ael-test/ref.ael-test19,
	  pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
	  pbx/ael/ael-test/ael-vtest21 (added),
	  pbx/ael/ael-test/ref.ael-vtest17,
	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
	  pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
	  pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
	  fixes bug 10411. I added a new regression test, some regression
	  test cleanups

2007-08-13 15:28 +0000 [r79214]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a potential deadlock in socket_process.
	  check_provisioning can eventually call find_callno. You can't
	  hold a pvt lock while calling find_callno because it goes through
	  and locks every single one looking for a match.

2007-08-13 14:51 +0000 [r79174-79207]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, apps/app_speech_utils.c,
	  include/asterisk/speech.h: Add an API call to allow the engine to
	  know that DTMF was received.

	* channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
	  channels/chan_local.c, channels/chan_misdn.c,
	  channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
	  channels/chan_h323.c, channels/chan_gtalk.c,
	  channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
	  Don't set the callerid name and number a second time on a newly
	  created channel. ast_channel_alloc itself already sets it and
	  setting it twice would cause a memory leak.

2007-08-11 05:23 +0000 [r79142]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Ensure the connection gets marked as used at
	  allocation time (closes issue #10429, report and fix by
	  mnicholson)

2007-08-10 20:53 +0000 [r79044-79099]  Steve Murphy <murf@digium.com>

	* main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
	  a user complaint on #asterisk, I have forced pbx_spool to explain
	  what reason codes mean, when they are logged

	* main/cdr.c: Re bug behavior mentioned in #asterisk, made this
	  tweak to code, to prevent hundreds of log messages from being
	  generated

	* main/cdr.c: This will help debug; from a question asked on
	  #asterisk

2007-08-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.10.1 released.
	
2007-08-10 15:20 +0000 [r78995]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: The last set of changes that I made to
	  "core show locks" made it not able to track mutexes unless they
	  were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
	  with ast_mutex_init() were not tracked. It should work now.

2007-08-10 14:15 +0000 [r78951-78955]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Don't bother having the core pass through or emulate
	  begin DTMF frames when in an ast_waitstream. It only cares about
	  the end of DTMF.

	* configs/queues.conf.sample: (closes issue #10422) Reported by:
	  bhowell Add note to sample configuration about module load order
	  and how it can cause perfectly good queue members to be marked as
	  invalid.

2007-08-10 13:24 +0000 [r78936]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
	  information element. We send them now also in the disconnect
	  message.

2007-08-09 23:47 +0000 [r78907]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Improved a bit of logic regarding
	  comma-separated mailboxes in has_voicemail. Also added some
	  braces to some compound if statements since unbraced if
	  statements scare me in general.

2007-08-09 23:10 +0000 [r78891]  Steve Murphy <murf@digium.com>

	* Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
	  output

2007-08-09 22:03 +0000 [r78826-78860]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing some extra debug code I left in my
	  last commit

	* apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
	  1. instead of using inboxcount as the core message counting
	  function, we use messagecount instead. This makes it possible to
	  count messages in folders besides just INBOX and Old. 2.
	  inboxcount and hasvoicemail now use messagecount as their means
	  of determining return values. 3. Added a copy_message function
	  for IMAP storage. Unfortunately I don't have the means to test
	  it, but it seems like a pretty straightforward function. 4.
	  Removed a #ifndef IMAP_STORAGE and matching #endif from
	  leave_voicemail for a couple of reasons. One, we want to support
	  copying mail to multiple IMAP boxes, and two, IMAP was broken
	  because a STORE macro had been moved into this section of code.

	* channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
	  put some new code in the wrong place and so I've reverted the
	  canreinvite section to how it was and put my new code where it
	  should be.

2007-08-09 17:58 +0000 [r78717-78778]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: add a comment to indicate that inboxcount
	  for ODBC_STORAGE needs to be fixed to support multiple mailboxes

	* apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
	  ODBC_STORAGE. Also, leave a comment for this to be fixed for
	  IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
	  working on this code right now for another reason. This is broken
	  even worse in trunk, but for a different reason. The fact that
	  the mailbox option supported multiple mailboxes is completely not
	  obvious from the code in the channel drivers. Anyway, I will fix
	  that in another commit ...

	* apps/app_meetme.c: Fix a problem with the combination of the 'F'
	  option to pass DTMF through a conference and options that use
	  DTMF to activate various features. The problem was that the BEGIN
	  frame would be passed through, but the END frame would get
	  intercepted to activate a feature. Then, the other conference
	  members would hear DTMF for forever, which they didn't seem to
	  like very much. (closes issue #10400, reported by stevefeinstein,
	  fixed by me)

2007-08-08 19:29 +0000 [r78646]  Jason Parker <jparker@digium.com>

	* doc/jabber.txt: Fix mogs email address.

2007-08-08 18:16 +0000 [r78575-78620]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed some compiler warnings so that
	  compiling with dev-mode and IMAP storage would not have any
	  errors. This section of code may get changed again shortly since
	  my change uncovers a rather silly bit of logic.

	* apps/app_queue.c: Changing a bit of logic so that someone will
	  NEVER exit the queue on timeout unless they have enabled the 'n'
	  option. This commit relates to issue #10320. Thanks to
	  jfitzgibbon for detailing the idea behind this code change.

2007-08-08 13:51 +0000 [r78569]  Joshua Colp <jcolp@digium.com>

	* configs/sip.conf.sample: (closes issue #10335) Reported by:
	  adamgundy Update sip.conf to include another scenario where
	  directrtpsetup will fail.

2007-08-07  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.10 released.
	
2007-08-07 20:57 +0000 [r78488]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c: Fix the build of this module on 64-bit
	  platforms

2007-08-07 19:43 +0000 [r78450]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: The logic behind inboxcount's return value
	  was reversed in has_voicemail and message_count. (closes issue
	  #10401, reported by st1710, patched by me)

2007-08-07 19:34 +0000 [r78437]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Don't free the environment handle when the
	  connection fails, because other connections might be depending
	  upon it

2007-08-07 19:11 +0000 [r78416]  Jason Parker <jparker@digium.com>

	* channels/chan_sip.c: Allow chan_sip to build in devmode

2007-08-07 19:09 +0000 [r78415]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, res/res_config_odbc.c,
	  apps/app_directory.c: Reconnection doesn't happen automatically
	  when a DB goes down (fixes issue #9389)

2007-08-07 18:25 +0000 [r78375]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Properly check the capabilities count to
	  avoid a segfault. (ASA-2007-019)

2007-08-07 17:45 +0000 [r78371]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
	  4 lines Revert patch committed for issue #9660. It broke E&M
	  trunks. (closes issue #10360) (closes issue #10364) ........

2007-08-06 21:41 +0000 [r78275]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Add additional DTMF log messages to help when
	  debugging issues.

2007-08-06 20:44 +0000 [r78184-78242]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix an issue where dynamic threads can get
	  free'd, but still exist in the dynamic thread list. (closes issue
	  #10392, patch from Mihai, with credit to his colleague, Pete)

	* include/asterisk/linkedlists.h: Fix the return value of
	  AST_LIST_REMOVE(). This shouldn't be causing any problems,
	  though, because the only code that uses the return value only
	  checks to see if it is NULL. (closes issue #10390, pointed out by
	  mihai)

2007-08-06 16:32 +0000 [r78182]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: It is possible for a transfer to occur
	  before the remote device has our tag in which case they send none
	  in the transfer. In this case we need to not fail the transfer
	  dialog lookup.

2007-08-06 16:30 +0000 [r78180]  Jason Parker <jparker@digium.com>

	* main/config.c: Fix an issue with using UpdateConfig (manager
	  action) where escaped semicolons in a config would be converted
	  to just semicolons (\; to ;) Issue 9938

2007-08-06 15:27 +0000 [r78166-78172]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
	  we pass through RTP timestamp information we need to make the
	  allowed timestamp skew considerably less. There are situations
	  where a source may change and due to the timestamp difference the
	  receiver will experience an audio gap since we did not indicate
	  by setting the marker bit that the source changed.

	* configure, configure.ac: (closes issue #10383) Reported by: rizzo
	  Include stdlib.h so NULL gets defined for gethostbyname_r checks.

2007-08-06 13:33 +0000 [r78164]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: Fixed a mistake I made in realtime_peer
	  which caused it to return NULL every time. Thanks to Jon Fealy
	  for emailing me the correction.

2007-08-05 14:18 +0000 [r78146]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
	  bug #10382)

2007-08-05 04:15 +0000 [r78143]  Russell Bryant <russell@digium.com>

	* include/asterisk/lock.h: Fix compilation failure when
	  MALLOC_DEBUG is enabled, but DEBUG_THREADS is not

2007-08-05 03:29 +0000 [r78139]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: If peer is not found, the error message is
	  misleading (should be peer not found, not ACL failure)

2007-08-03 20:25 +0000 [r78103]  Mark Michelson <mmichelson@digium.com>

	* main/config.c, channels/chan_sip.c, include/asterisk/config.h:
	  Changed the behavior of sip's realtime_peer function to match the
	  corresponding way of matching for non-realtime peers. Now matches
	  are made on both the IP address and port number, or if the
	  insecure setting is set to "port" then just match on the IP
	  address. In order to accomplish this, I also added a new API
	  call, ast_category_root, which returns the first variable of an
	  ast_category struct

2007-08-03 20:14 +0000 [r78028-78101]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: (closes issue #10194) Reported by:
	  blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
	  blitzrage Fix a problem when you call Voicemail() with multiple
	  mailboxes specified and ODBC_STORAGE is in use. The audio part of
	  the message was only given to the first mailbox specified.

	* main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
	  improvements to lock debugging. These changes take effect with
	  DEBUG_THREADS enabled and provide the following: * This will keep
	  track of which locks are held by which thread as well as which
	  lock a thread is waiting for in a thread-local data structure. A
	  reference to this structure is available on the stack in the
	  dummy_start() function, which is the common entry point for all
	  threads. This information can be easily retrieved using gdb if
	  you switch to the dummy_start() stack frame of any thread and
	  print the contents of the lock_info variable. * All of the
	  thread-local structures for keeping track of this lock
	  information are also stored in a list so that the information can
	  be dumped to the CLI using the "core show locks" CLI command.
	  This introduces a little bit of a performance hit as it requires
	  additional underlying locking operations inside of every
	  lock/unlock on an ast_mutex. However, the benefits of having this
	  information available at the CLI is huge, especially considering
	  this is only done in DEBUG_THREADS mode. It means that in most
	  cases where we debug deadlocks, we no longer have to request
	  access to the machine to analyze the contents of ast_mutex_t
	  structures. We can now just ask them to get the output of "core
	  show locks", which gives us all of the information we needed in
	  most cases. I also had to make some additional changes to astmm.c
	  to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
	  enabled. I disabled tracking of one of the locks in astmm.c
	  because it gets used inside the replacement memory allocation
	  routines, and the lock tracking code allocates memory. This
	  caused infinite recursion.

	* channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
	  frames when the mohinterpret option is set to "passthrough". This
	  was pointed out by Kevin in the middle of a training session.

	* channels/chan_iax2.c: Don't reuse the timespec that was set to 0
	  in the previous timedwait as it will just return immediately.
	  Also, fix some logic so the thread's lock isn't unlocked twice in
	  the weird case of dynamic threads getting acquired right after a
	  timeout. (pointed out by SteveK)

2007-08-02 21:53 +0000 [r77993-77996]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
	  actually allow 6 chars to be sent. Also make note of the "A"
	  option of date format. Issue 9779, modifications by DEA, wedhorn,
	  and myself.

	* channels/chan_skinny.c: If a device disconnects, the session will
	  go away. If this happens during call setup, we need to give up.
	  Issue 10325.

2007-08-02 19:25 +0000 [r77949]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix the case where a dynamic thread times
	  out waiting for something to do during the first time it runs.
	  This shouldn't ever happen, but we should account for it anyway.
	  (pointed out by pete, who works with mihai)

2007-08-02 18:42 +0000 [r77947]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure we clear the prompt status
	  message on a hangup. Also rearrange messages to better fit with
	  what a wireshark trace shows it should be. Issue 10299, initial
	  patch and solution by sbisker, modified by me to fit with
	  wireshark trace.

2007-08-02 18:21 +0000 [r77945]  Steve Murphy <murf@digium.com>

	* main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
	  line This patch hopefully solves 10141; The user is running with
	  it, and it doesn't appear to harm asterisk's operation, and may
	  prevent a crash. I'll store it in 1.2, as we have shut down
	  support on 1.2, but since I developed the patch before support
	  finished, and it might affect 1.4 and trunk, I'm going ahead with
	  it. ........

2007-08-02 18:04 +0000 [r77939-77943]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix another race condition in the handling
	  of dynamic threads. If the dynamic thread timed out waiting for
	  something to do, but was acquired to perform an action
	  immediately afterwords, then wait on the condition again to give
	  the other thread a chance to finish setting up the data for what
	  action this thread should perform. Otherwise, if it immediately
	  continues, it will perform the wrong action. (reported on IRC by
	  mihai, patch by me) (related to issue #10289)

	* channels/chan_iax2.c: Add another sanity check to
	  vnak_retransmit(). This check ensures that frames that have
	  already been marked for deletion don't get retransmitted. (closes
	  issue #10361, patch from mihai)

2007-08-02 15:15 +0000 [r77890-77894]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure that we show the correct
	  extension if dialed from a macro "From: 5555" rather than "From:
	  s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
	  tested by sbisker

	* channels/chan_skinny.c: Put in some additional debug information
	  for softkey/stimulus messages. Issue 10291, patch by DEA.

2007-08-01 22:16 +0000 [r77887]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix some race conditions which have been
	  causing weird problems in chan_iax2. The most notable problem is
	  that people have been seeing storms of VNAK frames being sent due
	  to really old frames mysteriously being in the retransmission
	  queue and never getting removed. It was possible that a dynamic
	  thread got created, but did not acquire its lock before the
	  thread that created it signals it to perform an action. When this
	  happens, the thread will sleep until it hits a timeout, and then
	  get destroyed. So, the action never gets performed and in some
	  cases, means a frame doesn't get transmitted and never gets freed
	  since the scheduler never gets a chance to reschedule
	  transmission. Another less severe race condition is in the
	  handling of a timeout for a dynamic thread. It was possible for
	  it to be acquired to perform at action at the same time that it
	  hit a timeout. When this occurs, whatever action it was acquired
	  for would never get performed. (patch contributed by Mihai and
	  SteveK) (closes issue #10289) (closes issue #10248) (closes issue
	  #10232) (possibly related to issue #10359)

2007-08-01 22:14 +0000 [r77886]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
	  not compile cleanly (missing def)

2007-08-01 21:08 +0000 [r77883]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix an issue that caused one-way audio on
	  some newer devices (specifically the 7921), due to sending
	  packets in the wrong order during hangup. Also make sure we clear
	  tones/messages on the correct line/instance. Issue 10291, patch
	  by DEA, tested by sbisker and myself.

2007-08-01 18:08 +0000 [r77863-77871]  Joshua Colp <jcolp@digium.com>

	* main/cli.c: (closes issue #10351) Reported by: ftarz Some
	  platforms don't like it when you pass NULL to vsnprintf so pass
	  "" instead.

	* include/asterisk/threadstorage.h, channels/chan_mgcp.c,
	  apps/app_voicemail.c, main/acl.c, utils/smsq.c,
	  channels/chan_iax2.c: Add some fixes for building on Solaris.

	* main/utils.c: Whoops, I meant R_5 not R5.

	* configure, configure.ac: And for my last trick... make sure that
	  if gethostbyname_r is exported by a library that it is used.

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/utils.c: Extend autoconf logic to determine which version of
	  gethostbyname_r is on the system.

2007-08-01 14:08 +0000 [r77852-77854]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Fixes an issue I introduced to queues wherein a
	  queue with joinempty=yes would kick people out of the queue
	  because of erroneously thinking the 'n' option was in use.
	  (closes issue #10320, reported by jfitzgibbon, patched by me,
	  tested by blitzrage and me) Thank you blitzrage for all the
	  testing you've done lately with queues! It's much appreciated!

	* apps/app_queue.c: If a queue uses dynamic realtime members, then
	  the member list should be updated after each attempt to call the
	  queue. This fixes an issue where if a caller calls into a queue
	  where no one is logged in, they would wait forever even if a
	  member logged in at some point. (closes issue #10346, reported by
	  and tested by blitzrage, patched by me)

2007-07-31 21:09 +0000 [r77845-77846]  Jim Dixon <telesistant@hotmail.com>

	* apps/app_rpt.c: Much newer version, 0.70 with much additions

	* main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
	  receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
	  logic in TONE_VERIFY/RELAX mode in chan_zap.

2007-07-31 20:59 +0000 [r77844]  Steve Murphy <murf@digium.com>

	* /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
	  line This probably isn't super-general, but it's a first stab at
	  using kill -11 to generate a core file instead of gcore. ........

2007-07-31 16:17 +0000 [r77831]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, include/asterisk/speech.h: Add a flag to the
	  speech API that allows an engine to set whether it received
	  results or not.

2007-07-31 15:53 +0000 [r77827]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
	  DEBUG_THREADS or it does nothing

2007-07-31 15:21 +0000 [r77824]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: This patch makes Asterisk send 100 Trying
	  provisional responses upon receipt of re-invites. This makes it
	  so that if there are two or more Asterisk servers between
	  endpoints, the Asterisk servers will not keep retransmitting the
	  re-invites. (closes issue #10274, reported by cstadlmann, patched
	  by me with approval from file)

2007-07-30 20:17 +0000 [r77795]  Jason Parker <jparker@digium.com>

	* main/say.c: Applications like SayAlpha() should not hang up the
	  channel if you request an "unknown" character such as a comma.
	  Instead, skip the character and move on. Issue 10083, initial
	  patch by jsmith, modified by me.

2007-07-30 20:16 +0000 [r77785-77794]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix an issue that could potentially cause
	  corruption of the global iax frame queue. In the network_thread()
	  loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
	  macro. However, to remove an element of the list within this
	  loop, it used AST_LIST_REMOVE, instead of
	  AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
	  internal variables of the SAFE macro invalid. Mihai says that he
	  already made this change in his local copy and it didn't help his
	  VNAK storm issues, but I still think it's wrong. :)

	* res/res_agi.c: (closes issue #10279) Reported by: seanbright
	  Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
	  seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
	  uploaded by seanbright (license 71) Allow the "agi_network: yes"
	  line to be printed out in the AGI debug output. Also, allow
	  partial writes to be handled when writing out this line just like
	  it is for all of the others.

	* main/channel.c: file and I both committed changes for issue
	  #10301. Remove a duplicated assignment to restore the original
	  value of the previous channel.

2007-07-30 18:43 +0000 [r77783]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, res/res_agi.c: Merged revisions 77782 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
	  | 2 lines Revert change in revision 71656, even though it fixed a
	  bug, because many people were depending upon the (broken)
	  behavior. ........

2007-07-30 17:29 +0000 [r77780]  Russell Bryant <russell@digium.com>

	* main/channel.c: (closes issue #10301) Reported by: fnordian
	  Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
	  (license 110) Additional changes by me Fix some problems in
	  channel_find_locked() which can cause an infinite loop. The
	  reference to the previous channel is set to NULL in some cases.
	  These changes ensure that the reference to the previous channel
	  gets restored before needing it again. I'm not convinced that the
	  code that is setting it to NULL is really the right thing to do.
	  However, I am making these changes to fix the obvious problem and
	  just leaving an XXX comment that it needs a better explanation
	  that what is there now.

2007-07-30 17:11 +0000 [r77768-77778]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: (closes issue #10327) Reported by: kkiely
	  Instead of directly mucking with the extension/context/priority
	  of the channel we are transferring when it has a PBX simply call
	  ast_async_goto on it. This will ensure that the channel gets
	  handled properly and sent to the right place.

	* main/channel.c: (closes issue #10301) Reported by: fnordian
	  Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
	  (license 110) Restore previous behavior where if we failed to
	  lock the channel we wanted we would return to exactly the same
	  point as if we had just reentered the function.

	* /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
	  lines (closes issue #10334) Reported by: ramonpeek Pass through
	  the return value from macro_exec through the MacroIf application.
	  ........

2007-07-27 18:15 +0000 [r77571]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Missing newline

2007-07-27 17:04 +0000 [r77536-77540]  Joshua Colp <jcolp@digium.com>

	* cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
	  Patches: cdr_pgsql.patch uploaded by prashant (license 114)
	  Finish the Postgresql connection after the log messages are
	  printed so we don't access invalid memory.

	* channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
	  Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
	  julianjm (license 99) Clear ONHOLD flag when decrementing the
	  onHold peer count. If we did not do this the count may keep
	  decreasing.

2007-07-27 14:30 +0000 [r77490]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_sip.c: "re-invite" was misspelled

2007-07-26 23:19 +0000 [r77460]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: (closes issue #10302) Reported by: litnialex If a
	  DTMF end frame comes from a channel without a begin and it is
	  going to a technology that only accepts end frames (aka INFO)
	  then use the minimum DTMF duration if one is not in the frame
	  already.

2007-07-26 22:16 +0000 [r77424-77429]  Kevin P. Fleming <kpfleming@digium.com>

	* doc/mp3.txt: change protocol for downloads as well

	* doc/mp3.txt, sounds/Makefile: use new canonical name for download
	  server

2007-07-26 21:23 +0000 [r77410]  Russell Bryant <russell@digium.com>

	* Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
	  in trunk, but not in 1.4. When Asterisk is compiled under dev
	  mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
	  define it in the same way that trunk does. Also, revert the
	  change that added this define in the Makefile The advantage to
	  doing it this way is that buildopts.h gets installed when you
	  install Asterisk. Then, when building any out of tree modules, or
	  building asterisk-addons, these modules know which options the
	  rest of Asterisk was built with.

2007-07-26 20:35 +0000 [r77380]  Mark Michelson <mmichelson@digium.com>

	* Makefile, main/logger.c: Fixes to get ast_backtrace working
	  properly. The AST_DEVMODE macro was never defined so the majority
	  of ast_backtrace never attempted compilation. The makefile now
	  defines AST_DEVMODE if configure was run with --enable-dev-mode.
	  Also, changes were made to acccomodate 64 bit systems in
	  ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
	  their roles in allowing me to get this committed

2007-07-26 19:32 +0000 [r77348-77350]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/logger.c: Missed one

	* main/logger.c: Oops, that builtin define should be all-lowercase.

2007-07-26 18:30 +0000 [r77318]  Mark Michelson <mmichelson@digium.com>

	* cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
	  meaning free gets called on the same variable twice. This patch
	  sets the connection to NULL after calls to PQfinish so that the
	  problem does not occur. Also in this patch, prashant_jois
	  informed me that it is safe to pass a null pointer to PQfinish,
	  so I have removed the check for conn's existence from
	  my_unload_module. (closes issue 10295, reported by junky, patched
	  by me with input from prashant_jois)

2007-07-25 22:39 +0000 [r77191]  Steve Murphy <murf@digium.com>

	* apps/app_meetme.c: This fix solves problem with intense squelch
	  noise when someone joins conf in bug 9430; We repro'd the problem
	  with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
	  I'm applying it. It looks like playing the recorded username will
	  louse up the next thing played into the channel. Josh rearranged
	  the code so as to start things over before playing data directly
	  into the conference.

2007-07-25 22:16 +0000 [r77176]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
	  Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
	  digit to terminate a DTMF string with. If none is specified then
	  no terminator will be used.

2007-07-25 21:52 +0000 [r77154]  Mark Michelson <mmichelson@digium.com>

	* main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
	  not a signed int, so use %u instead of %d in the format string

2007-07-25 20:23 +0000 [r77116-77136]  Jason Parker <jparker@digium.com>

	* /: so are my fingers...

	* /: autotagexternals script is still obviously misbehaving...

	* /: use autotagged externals

2007-07-25 17:14 +0000 [r77071]  Joshua Colp <jcolp@digium.com>

	* configure, acinclude.m4: Fix autoconf logic for finding OpenH323
	  when it is not in the first place searched (/usr/share/openh323).

2007-07-25 09:34 +0000 [r77022]  Luigi Rizzo <rizzo@icir.org>

	* main/rtp.c: set the sequence number in a frame for all frame
	  types

2007-07-25 00:18 +0000 [r76983]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
	  line this fixes bug 10293, where the error message because
	  defaultzone or loadzone was not defined was confusing ........

2007-07-24 22:12 +0000 [r76891-76937]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
	  Jul 2007) | 2 lines Oops, res contains the error code, not errno.
	  I was wondering why a mutex was reporting "No such file or
	  directory"... ........

	* main/app.c: Found another place where we should be using the
	  umask (thanks jcmoore)

2007-07-24  Jason Parker <jparker@digium.com>

	* Asterisk 1.4.9 released.

2007-07-24 16:42 +0000 [r76803-76805]  Jason Parker <jparker@digium.com>

	* /: Blocked revisions 76802 via svnmerge ........ r76802 | qwell |
	  2007-07-24 11:32:04 -0500 (Tue, 24 Jul 2007) | 3 lines Don't
	  create the Asterisk channel until we are starting the PBX on it.
	  (ASA-2007-018) ........

	* channels/chan_iax2.c: Don't create the Asterisk channel until we
	  are starting the PBX on it. (ASA-2007-018)

2007-07-24 16:26 +0000 [r76801]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Added a membercount variable to call_queue
	  struct which keeps track of the number of logged in members in a
	  particular queue. This makes it so that the 'n' option for
	  Queue() can act properly depending on which strategy is used. If
	  the strategy is roundrobin, rrmemory, or ringall, we want to ring
	  each phone once before moving on in the dialplan. However, if any
	  other strategy is used, we will only ring one phone since it
	  cannot be guaranteed that a different phone will ring on
	  subsequent attempts to ring a phone. As a side effect of this,
	  the QUEUE_MEMBER_COUNT dialplan function now just reads the
	  membercount variable instead of traversing through the member
	  list to figure out how many members there are. Special thanks to
	  blitzrage for helping to test this out. (closes issue #10127,
	  reported by bcnit, patched by me, tested by blitzrage)

2007-07-23 22:38 +0000 [r76708]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: It was our stated intention for 1.4 that
	  files created in app_voicemail should depend upon the umask.
	  Unfortunately, mkstemp() creates files with mode 0600, regardless
	  of the umask. This corrects that deficiency.

2007-07-23 18:59 +0000 [r76656]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix some incorrect softkey labels in
	  messages. Don't try to play dialtone in some unimplemented
	  features.

2007-07-23 18:29 +0000 [r76654]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
	  2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
	  force channel format changes when a generator is present. The
	  generator may have changed the formats itself and changing them
	  back would cause issues. ........

2007-07-23 17:57 +0000 [r76620]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Don't try to queue up hold/unhold frames
	  on a non-existent channel. Issue 10276.

2007-07-23 17:48 +0000 [r76519-76618]  Joshua Colp <jcolp@digium.com>

	* apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
	  by putting the value of the digit char in an int.

	* /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
	  lines (closes issue #10236) Reported by: homesick Patches:
	  rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
	  Remote Party ID on guest calls. ........

	* channels/chan_skinny.c: (closes issue #10268) Reported by:
	  mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
	  (license 7) Add another OS that has to use the Macros for byte
	  ordering.

2007-07-23 12:25 +0000 [r76485]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Use a signed integer for storing the number
	  of bytes in the packet read from the network. Using an unsigned
	  value here made it impossible to handle an error returned from
	  recvfrom(). Furthermore, in the case that recvfrom() did return
	  an error, this would cause a crash due to a heap overflow.
	  (closes issue #10265, reported by and fix suggested by
	  timrobbins)

2007-07-22 21:42 +0000 [r76410]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /: Blocked revisions 76409 via svnmerge ........ r76409 |
	  tilghman | 2007-07-22 16:39:55 -0500 (Sun, 22 Jul 2007) | 2 lines
	  We should not use C++ reserved words in API headers (closes issue
	  #10266) ........

2007-07-21 02:02 +0000 [r76227]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
	  4 lines Backport a fix for a memory leak that was fixed in trunk
	  in reivision 76221 by rizzo. The memory used for the localaddr
	  list was not freed during a configuration reload. ........

2007-07-20 21:36 +0000 [r76211]  Steve Murphy <murf@digium.com>

	* sounds/Makefile: This patch from 10249 is worth applying! It
	  prevents downloading sound files if they are already downloaded.
	  Darn Practical, if you ask me

2007-07-20 21:03 +0000 [r76174-76178]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Allow getting a call from an existing
	  "sub" channel. Cancel ringing if endpoint hangs up before
	  answering. Fixes were backported from trunk (there was apparently
	  a bit of confusion during merge of a previous patch). (closes
	  issue #10241)

	* main/manager.c: Eliminate a compiler warning with gcc 4.2 by
	  constifying a char *

	* channels/chan_skinny.c: It's possible for sub->owner to be NULL
	  here if you cancel the call immediately after/during sending a
	  digit.

2007-07-20 18:42 +0000 [r76139]  Mark Michelson <mmichelson@digium.com>

	* apps/app_directory.c: When using users.conf for the entries in
	  the directory, if multiple users had the same last name, only the
	  first user listed would be available in the directory. (closes
	  issue #10200, reported by mrskippy, patched by me)

2007-07-20 18:22 +0000 [r76132]  Russell Bryant <russell@digium.com>

	* main/channel.c: Use the define that specifies the default length
	  of an artificially created DTMF digit in the ast_senddigit()
	  function. The define is set to 100ms by default, which is the
	  same thing that this function was using. But, using the define
	  lets changes take effect in this case, as well as the others
	  where it was already used.

2007-07-20 17:20 +0000 [r76054-76087]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
	  lines (closes issue #10247) Reported by: fkasumovic Patches:
	  chan_sip.patch uploaded by fkasumovic (license #101) Drop any
	  peer realm authentication entries when reloading so multiple
	  entries do not get added to the peer. ........

	* res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
	  Patches: res_conver.patch uploaded by fkasumovic (license #101)
	  Use the last occurance of . to find the extension, not the first
	  occurance.

	* apps/app_queue.c: Move makeannouncement variable declaration to
	  proper place.

2007-07-19 20:36 +0000 [r75980]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Remove some duplicate code.

2007-07-19 18:59 +0000 [r75969-75978]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: The diff on this looks pretty big but all I did
	  was remove a pointless if statement (always evaluates true).

	* apps/app_queue.c: Changes in handling return values of several
	  functions in app_queue. This all started as a fix for issue
	  #10008 but now includes all of the following changes: 1.
	  Simplifying the code to handle positive return values from ast
	  API calls. 2. Removing the background_file function. 3. The fix
	  for issue #10008 (closes issue #10008, reported and patched by
	  dimas)

2007-07-19 15:53 +0000 [r75928]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
	  6 lines When processing full frames, take sequence number
	  wraparound into account when deciding whether or not we need to
	  request retransmissions by sending a VNAK. This code could cause
	  VNAKs to be sent erroneously in some cases, and to not be sent in
	  other cases when it should have been. (closes issue #10237,
	  reported and patched by mihai) ........

2007-07-18 22:59 +0000 [r75807]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Need to make sure we set milliseconds and
	  timestamp - pointed out by the recent ast_ time stuff from
	  Tilghman

2007-07-18 21:09 +0000 [r75759]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
	  5 lines When traversing the queue of frames for possible
	  retransmission after receiving a VNAK, handle sequence number
	  wraparound so that all frames that should be retransmitted
	  actually do get retransmitted. (issue #10227, reported and
	  patched by mihai) ........

2007-07-18 20:40 +0000 [r75749]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
	  | 2 lines Store prior to copy (closes issue #10193) ........

2007-07-18 20:17 +0000 [r75732]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Umm, why are we transmitting dialtone on
	  cfwdall?

2007-07-18 20:00 +0000 [r75712]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
	  pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
	  Asterisk won't build under devmode using GCC 4.2.

2007-07-18 19:54 +0000 [r75707-75711]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
	  Issue 9245, patch by slimey.

	* channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
	  9887, patches by DEA

2007-07-18 17:56 +0000 [r75658]  Dwayne M. Hubbard <dhubbard@digium.com>

	* /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
	  | 1 line removed the word 'pissed' from ast_log(...) function
	  call for BE-90 ........

2007-07-18 15:44 +0000 [r75583-75623]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Few more places that needs to check for
	  onhold state.

	* channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
	  It is possible for hold status to exist without call limits set,
	  so we need to ensure update_call_counter is executed regardless.

	* channels/chan_h323.c: Don't bother reloading chan_h323 if it did
	  not load successfully in the first place. This would otherwise
	  cause a crash.

	* pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
	  the threadid of each running thread before shutting them down as
	  the thread themselves may change the value.

2007-07-18 12:29 +0000 [r75529]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_meetme.c: Using a freed frame causes crashes (closes
	  issue #9317)

2007-07-17  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.8 released.

2007-07-17 20:57 +0000 [r75441-75450]  Russell Bryant <russell@digium.com>

	* /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
	  Jul 2007) | 3 lines Properly check for the length in the skinny
	  packet to prevent an invalid memcpy. (ASA-2007-016) ........

	* main/rtp.c: cast arguments to ast_log so that it builds without
	  warnings for me

	* channels/iax2-parser.c, channels/iax2-parser.h, /,
	  channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
	  5 lines Ensure that when encoding the contents of an ast_frame
	  into an iax_frame, that the size of the destination buffer is
	  known in the iax_frame so that code won't write past the end of
	  the allocated buffer when sending outgoing frames. (ASA-2007-014)
	  ........

	* /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
	  4 lines After parsing information elements in IAX frames, set the
	  data length to zero, so that code later on does not think it has
	  data to copy. (ASA-2007-015) ........

2007-07-17 20:40 +0000 [r75439]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Ensure that the pointer to STUN data does not go to
	  unaccessible memory. (ASA-2007-017)

2007-07-17 20:33 +0000 [r75437]  Russell Bryant <russell@digium.com>

	* res/res_agi.c: (issue #10210) Reported by: juggie Patches:
	  10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
	  juggie, blitzrage Log a warning if someone uses DeadAGI on a live
	  channel.

2007-07-17 20:03 +0000 [r75405]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
	  can return -1 on failure, so I need to be sure that we only enter
	  the if statement if it is successful. Related to my fix to issue
	  #10186

2007-07-17 20:01 +0000 [r75401-75403]  Russell Bryant <russell@digium.com>

	* main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
	  10209-trunk-2.patch uploaded by juggie Tested by: juggie,
	  blitzrage In ast_pbx_run(), mark a channel as hung up after an
	  application returned -1, or when it runs out of extensions to
	  execute. This is so that code can detect that this channel has
	  been hung up for things like making sure DeadAGI is used on
	  actual dead channels, and is beneficial for other things, like
	  making sure someone doesn't try to start spying on a channel that
	  is about to go away.

	* res/res_agi.c: Remove a duplicated newline character in AGI debug
	  output. (closes issue #10207, patch by seanbright)

2007-07-16 20:53 +0000 [r75258-75306]  Kevin P. Fleming <kpfleming@digium.com>

	* main/dns.c, /: Merged revisions 75304 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
	  | 3 lines provide proper copyright/license attribution for this
	  structure that was copied from a BSD-licensed header file long,
	  long ago... ........

	* /: another fix that is not needed here (finishing up 75251)

2007-07-16 18:16 +0000 [r75253]  Mark Michelson <mmichelson@digium.com>

	* apps/app_dial.c: Restoring functionality from 1.2 wherein
	  Retrydial will not exit if there is no announce file specified.
	  This change makes it so that if there is no announce file
	  specified, the application will continue until finished (or
	  caller hangs up). If a bogus announce file is specified, then a
	  warning message will be printed saying that the file could not be
	  found, but execution will still continue. (closes issue #10186,
	  reported by jon, patched by me)

2007-07-16 18:12 +0000 [r75252]  Kevin P. Fleming <kpfleming@digium.com>

	* /: block change that is not relevant here

2007-07-13 20:36 +0000 [r75108]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
	  Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
	  load_moh_classes() could return without destroying the loaded
	  configuration. ........

2007-07-13 20:15 +0000 [r75078]  Mark Michelson <mmichelson@digium.com>

	* apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
	  2007) | 5 lines Fixed an issue where chanspy flags were
	  uninitialized if no options were passed. What triggered this
	  investigation was an IRC chat where some people's quiet flags
	  were set while others' weren't even though none of them had
	  specified the q option. ........

2007-07-13 20:10 +0000 [r75053-75067]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
	  Jul 2007) | 6 lines Ensure that adding a user to the list of
	  users of a specific music on hold class is not done at the same
	  time as any of the other operations on this list to prevent list
	  corruption. Using the global moh_data lock for this is not ideal,
	  but it is what is used to protect these lists everywhere else in
	  the module, and I am only changing what is necessary to fix the
	  bug. ........

	* channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
	  12 lines (closes issue #9660) Reported by: mmacvicar Patches
	  submitted by: bbryant, russell Tested by: mmacvicar, marco,
	  arcivanov, jmhunter, explidous When using a TDM400P (and probably
	  other analog cards) there was a chance that you could hang up and
	  pick the phone back up where it has been long enough to be not
	  considered a flash hook, but too soon such that the device
	  reports that it is busy and the person on the phone will only
	  hear silence. This patch makes chan_zap more tolerant of this and
	  gives the device a couple of seconds to succeed so the person on
	  the phone happily gets their dialtone. ........

2007-07-12 23:00 +0000 [r74998]  Mark Michelson <mmichelson@digium.com>

	* channels/chan_agent.c: Change to my previous fix regarding agent
	  logoff soft. Now uses deferlogoff instead of loginstart since
	  loginstart is used after logoff. Thanks to makoto for pointing
	  this out and suggesting the fix. (closes issue #10178, reported
	  and patched by makoto, with modification by me)

2007-07-12 20:42 +0000 [r74955]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: This patch resolves 10143; thanks to irroot
	  for the patch; looked acceptable. Let the community decide if it
	  messes things up

2007-07-12 19:17 +0000 [r74888-74922]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Whoops... didn't want this to be returned to 0
	  each iteration.

	* main/channel.c: When waiting for a digit ensure that a begin
	  frame was received with it, not just an end frame. (issue #10084
	  reported by rushowr)

2007-07-12 16:53 +0000 [r74839-74866]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: It helps if I actually add this stuff for
	  the 7921 too - otherwise it won't actually do much of anything.

	* channels/chan_skinny.c: Add device ID for 7921 wireless skinny
	  phone

	* channels/chan_skinny.c: Fix dialing in skinny that was broken in
	  some cases. Issue 10136, fix provided by DEA.

2007-07-12 15:53 +0000 [r74815]  Joshua Colp <jcolp@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
	  2007) | 2 lines Only print out a warning for situations where it
	  is actually helpful. (issue #10187 reported by denke) ........

2007-07-11 22:57 +0000 [r74767]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
	  5 lines The function make_trunk() can fail and return -1 instead
	  of a valid new call number. Fix the uses of this function to
	  handle this instead of treating it as the new call number. This
	  would cause a deadlock and memory corruption. (possible cause of
	  issue #9614 and others, patch by me) ........

2007-07-11 21:14 +0000 [r74722]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
	  Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
	  did not work...at all. Now it does. (closes issue #10178,
	  reported and patched by makoto, with slight modification for 1.4
	  and trunk by me) ........

2007-07-11 18:34 +0000 [r74657]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
	  Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
	  the condition that uses LIKE. This fixes realtime extensions with
	  ODBC. (closes issue #10175, reported by stuarth, patch by me)
	  ........

2007-07-11 18:18 +0000 [r74628-74642]  Steve Murphy <murf@digium.com>

	* Makefile: This fixes 10172, where the entire man8 dir gets
	  removed during an uninstall of asterisk

	* utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
	  further reversion of previously applied floating point stuff for
	  expr2

2007-07-11 17:16 +0000 [r74515-74590]  Joshua Colp <jcolp@digium.com>

	* /: Blocked revisions 74587 via svnmerge ........ r74587 | file |
	  2007-07-11 14:15:11 -0300 (Wed, 11 Jul 2007) | 2 lines Use some
	  Makefile magic to determine if linux/compiler.h is present.
	  (issue #10174 reported by francesco_r) ........

	* channels/chan_phone.c, configure,
	  include/asterisk/autoconfig.h.in, configure.ac: Instead of
	  figuring out kernel versions that have compiler.h and not...
	  let's just use autoconf to check for it's presence. (issue #10174
	  reported by francesco_r)

	* channels/chan_phone.c: Only check if we need to do a SIGMA based
	  tone generation if we have a card. (issue #10179 reported by
	  mikowhy)

2007-07-10 23:32 +0000 [r74476]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Forwarding a message with IMAP storage was
	  storing the message in the sender's box instead of the forwarded
	  mailbox. (closes issue #10138, reported and patched by jaroth)

2007-07-10 19:58 +0000 [r74374-74428]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
	  lines Fix an issue where it was possible to have a service level
	  of over 100% Between the time recalc_holdtime and update_queue
	  was called, it was possible that the call could have been hungup.
	  Move both additions to the same place, so this won't happen.
	  Issue 10158, initial patch by makoto, modified by me. ........

	* main/dns.c: Don't use #if to check if something is defined - use
	  #ifdef instead. Pointed out by kpfleming

	* /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
	  2007) | 4 lines Fix an issue with wrapuptime not working when
	  using AgentLogin. Issue 10169, patch by makoto, with a minor mod
	  by me to not re-break issue 9618 ........

	* main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
	  configure.ac: Merged revisions 74373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
	  lines Use res_ndestroy on systems that have it. Otherwise, use
	  res_nclose. This prevents a memleak on NetBSD - and possibly
	  others. Issue 10133, patch by me, reported and tested by scw
	  ........

2007-07-10  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.7.1 released.

2007-07-10 16:00 +0000 [r74323]  Russell Bryant <russell@digium.com>

	* res/res_musiconhold.c: fix an uninitialized variable

2007-07-10 15:38 +0000 [r74317]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
	  lines Fix a small typo in description in of Voicemail()
	  application. Issue 10170, patch by casper. ........

2007-07-10 15:31 +0000 [r74314]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
	  Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
	  #10075, this part reported by jmls on IRC, patch by me) ........

2007-07-10 14:50 +0000 [r74262-74265]  Joshua Colp <jcolp@digium.com>

	* /, main/app.c: Merged revisions 74264 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
	  lines Ensure the group information category exists before trying
	  to do a string comparison with it. (issue #10171 reported by
	  mlegas) ........

	* channels/chan_sip.c: Only spit out an inringing warning message
	  when it is applicable. Since call limits are already toast in
	  realtime let's not scare the user if they are using it. (issue
	  #10166 reported by bcnit)

2007-07-09  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.7 released.

2007-07-09 21:31 +0000 [r74162-74211]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Update the configure script to check for
	  a required function that is not present in the 1.2 version of
	  libpri. This will prevent the configure script from thinking that
	  it has compatible libpri support for Asterisk 1.4, when it
	  actually does not because the installed version is from 1.2.

	* /: Blocked revisions 74165 via svnmerge ........ r74165 | russell
	  | 2007-07-09 16:00:17 -0500 (Mon, 09 Jul 2007) | 4 lines When the
	  specified class isn't found, properly fall back to the channel's
	  music class or the default. (issue #10123, reported by blitzrage,
	  patches from juggie, qwell, and me) ........

	* res/res_musiconhold.c: (closes issue #10123) Reported by:
	  blitzrage Patches submitted by: juggie, qwell, me Tested by:
	  blitzrage When trying to find a music on hold class to use, try
	  all of the options, instead of only the first one that is set.
	  Also, change the MusicOnHold applications to not hang up on the
	  channel when a class can not be found.

2007-07-09 20:19 +0000 [r74159]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
	  lines Several chan_zap options were not working on reload because
	  they were arbitrarily disallowed when reloading some/most PRI
	  options (such as signalling) was disallowed. Options such as
	  polarityonanswerdelay and answeronpolarityswitch can safely be
	  changed on a reload. This corrects that behavior. Issue 9186,
	  patch by tzafrir. ........

2007-07-09 18:38 +0000 [r74120-74122]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Forgot to get rid of an extraneous debug
	  message.

	* apps/app_queue.c: The n option for Queue should make the queue
	  exit immediately after failure to reach any members and should
	  not be dependent on the timeout value passed to Queue (closes
	  issue #10127, reported by bcnit, repaired by me)

2007-07-09 15:32 +0000 [r74082]  Joshua Colp <jcolp@digium.com>

	* channels/chan_skinny.c: Only destroy the scheduler context if it
	  was allocated. (issue #10124 reported by gzero)

2007-07-09 14:57 +0000 [r74047]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
	  Pass the mailbox instead of the context to inbox_count when the
	  context is "default." (closes issue #10135, reported by yannj,
	  repaired by me)

2007-07-09 14:49 +0000 [r74043-74045]  Joshua Colp <jcolp@digium.com>

	* channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
	  synchronization tweaks. (issue #10124 reported by gzero)

	* configure, acinclude.m4: Use AC_CHECK_HEADER to check for
	  ptlib/openh323 to allow for cross compiling. (issue #9675
	  reported by zandbelt)

2007-07-09 04:03 +0000 [r73985]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
	  'make progdocs'. (Closes issue #10104)

2007-07-09 03:13 +0000 [r73930-73980]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c: Give Agent channel names priority when doing CDR
	  merging. (issue #10011 reported by krtorio)

	* pbx/pbx_config.c: Add a few sanity checks when writing out the
	  dialplan. (issue #10157 reported by dome)

2007-07-08 09:47 +0000 [r73849]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: While tracking down a bug, I need some more
	  history. Dumphistory is very useful, indeed.

2007-07-06 23:02 +0000 [r73769]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
	  4 lines If a sip_pvt struct has already registered an extension
	  state callback, remove the old one before adding a new one. If
	  this isn't done, Asterisk will crash. (issue #10120) ........

2007-07-06 16:36 +0000 [r73727]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixing a rare case which causes voicemail
	  to crash when compiled with IMAP storage. inboxcount has the
	  possibility of finding an "interactive" vm_state when no
	  persistent "non-interactive" vm_state exists for that mailbox. If
	  this should happen when someone attempts to leave a message, it
	  results in a crash. This patch, along with my commit in revision
	  72670 fix issue 10053, reported by jaroth. closes issue #10053

2007-07-06 16:12 +0000 [r73679-73696]  Russell Bryant <russell@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
	  Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
	  Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
	  with MSSQL 2005 by explicitly stating that '\' is being used as
	  an escape character. ........

	* /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
	  7 lines (closes issue #10125) Reported by: makoto Patches
	  submitted by: makoto This fixes a crash in chan_sip that happens
	  when the bindaddr setting is not valid on Asterisk startup, gets
	  fixed, and then a reload gets issued. ........

2007-07-06 15:27 +0000 [r73675]  Mark Michelson <mmichelson@digium.com>

	* /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
	  Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
	  (issue 9618, reported by jiddings, patched by moi) closes issue
	  #9618 ........

2007-07-06 03:34 +0000 [r73551-73629]  Russell Bryant <russell@digium.com>

	* BUGS: fix a little spelling error

	* channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
	  the monitor thread if it was never started. (closes issue #10124,
	  reported by gzero, fixed by me)

	* channels/chan_iax2.c: copy from the correct buffer when deferring
	  a full frame (related to issue #9937)

	* channels/chan_iax2.c: * Store the call number that a thread is
	  processing without the full frame bit set to ease debugging *
	  When deferring a full frame for processing, stick it into the
	  queue for the thread that is processing frames for that call, not
	  the one that read the current frame and is about to go back into
	  the idle list (related to issue #9937)

2007-07-05 22:20 +0000 [r73548]  Kevin P. Fleming <kpfleming@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
	  | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
	  like we don't support it for G.729 ........

2007-07-05 20:50 +0000 [r73512]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Pass HOLD and UNHOLD frames to the other
	  channel when they are returned from a native bridge function.
	  This fixes a problem where when two zap channels are natively
	  bridged and one does a flash hook, the other channel did not
	  receive music on hold. (Reported to me directly by Doug Bailey at
	  Digium)

2007-07-05 19:18 +0000 [r73467]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
	  lines Copy language information to the dialog structure when
	  calling a peer for situations where a PBX may be started on the
	  dialed channel. (issue #10121 reported by clegall_proformatique)
	  ........

2007-07-05 15:59 +0000 [r73400]  Mark Michelson <mmichelson@digium.com>

	* apps/app_queue.c: Correcting a minor CLI bug I found. When
	  issuing the queue show command, if you type queue show and then
	  press tab, you can continue pressing tab and it will keep
	  auto-completing queue names even though only 1 queue can be used
	  as an argument.

2007-07-05 15:28 +0000 [r73398]  Russell Bryant <russell@digium.com>

	* channels/chan_vpb.cc, channels/Makefile: Make this module build
	  for me in dev-mode

2007-07-05 14:21 +0000 [r73316-73355]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
	  lines Tweak spy locking. (issue #9951 reported by welles)
	  ........

	* channels/chan_local.c, /: Merged revisions 73318 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
	  2007) | 2 lines Actually check to make sure a PBX was started on
	  one of the Local channels instead of blindly assuming it was.
	  (issue #10112 reported by makoto) ........

	* /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
	  lines Reset ServicelevelPerf variable back to 0 if we are unable
	  to calculate it each time... otherwise we will get previous
	  values. (issue #10117 reported by noriyuki) ........

2007-07-04 14:53 +0000 [r73208-73253]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
	  Jul 2007) | 1 line bchannel configurations like echocancel and
	  volume control, need to be setuped on inbound calls too. ........

	* channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
	  Jul 2007) | 1 line bad bug in overlapdial case, we called
	  start_pbx multiple times, because the state wasn't changed..
	  ........

2007-07-03 20:17 +0000 [r73143]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
	  main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
	  expr floating patch from 1.4; too much of a behavior change. If
	  you want this fix, try trunk instead. bug 9508.

2007-07-03 15:42 +0000 [r73104-73106]  Jason Parker <jparker@digium.com>

	* /: What the heck. This should not have happened.

	* /: use autotagged externals

2007-07-03 12:38 +0000 [r73053]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
	  | 2 lines RetryDial should accept a 0 argument, but it does not,
	  because atoi does not distinguish between 0 and error (closes
	  issue #10106) ........

2007-07-03 08:17 +0000 [r73005]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
	  Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
	  be called from mISDN Source channels.. #9449 ........

2007-07-02 20:16 +0000 [r72933]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
	  main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
	  main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
	  for floating point numbers added to ast_expr2 $\[...\] exprs.
	  Fixes bug 9508, where the expr code fails with fp numbers. The
	  MATH function returns fp numbers by default, so this fix is
	  considered necessary.

2007-07-02 18:18 +0000 [r72926]  Russell Bryant <russell@digium.com>

	* main/manager.c: Remove a bogus comment and add proper locking to
	  the handler function for the CLI command to show information on
	  manager actions.

2007-07-02 17:59 +0000 [r72925]  Jason Parker <jparker@digium.com>

	* /: Blocked revisions 72924 via svnmerge ........ r72924 | qwell |
	  2007-07-02 12:58:25 -0500 (Mon, 02 Jul 2007) | 4 lines Fix an
	  issue with playing "oclock" multiple times in French with 24 hour
	  time format. Issue 10101 ........

2007-07-02 14:32 +0000 [r72888]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Added additional DTMF debug messages for when
	  emulation occurs.

2007-07-02 08:41 +0000 [r72850-72852]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 72585 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
	  1 line check if the bchannel stack id is already used, if so
	  don't use it a second time. Also added a release_chan lock, so
	  that the same chan_list object cannot be freed twice. chan_misdn
	  does not crash anymore on heavy load with these changes. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 72099 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
	  1 line simplified generation for dummy bchannels, also we mark
	  them as dummies, so they are not used later as real-bchannels,
	  optimized the RESTART mechanisms, we block a channel now on
	  cause:44, and send out a RESTART automatically, then on reception
	  of RESTART_ACKNOWLEDGE we unblock the channel again. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
	  revisions 72087 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
	  1 line simplified channel finding and locking a lot. removed
	  unnecessary #ifdefed areas. ........

2007-07-01 23:52 +0000 [r72806]  Russell Bryant <russell@digium.com>

	* pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
	  5 lines When appending lines to call files to keep track of
	  retries, write a leading newline just in case the original call
	  file did not have a newline at the end. This fix is in response
	  to a problem I saw reported on the asterisk-users mailing list.
	  ........

2007-06-30 16:50 +0000 [r72705-72766]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Tweak the configure script so that error
	  output isn't spewed to the console when searching for GTK2 libs,
	  and they aren't found.

	* formats/format_pcm.c: give format_pcm a more concise destription

2007-06-29 19:07 +0000 [r72665]  Luigi Rizzo <rizzo@icir.org>

	* main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
	  absence of the function. This was already done in trunk.

2007-06-29  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.6 released.

2007-06-29 16:31 +0000 [r72630]  Russell Bryant <russell@digium.com>

	* /: Blocked revisions 72629 via svnmerge ........ r72629 | russell
	  | 2007-06-29 11:30:56 -0500 (Fri, 29 Jun 2007) | 4 lines Backport
	  changes that make chan_iax2 not start the PBX on an incoming
	  channel until the three-way call setup is completed. These
	  changes are already in 1.4 and trunk. ........

2007-06-29 14:26 +0000 [r72597-72599]  Joshua Colp <jcolp@digium.com>

	* main/cdr.c: Minor change for older GCC versions.

	* Makefile, configure, configure.ac, makeopts.in: Backport fix for
	  GCC versions without support for declaration-after-statement.

2007-06-29 04:47 +0000 [r72554-72556]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/manager.c: Issue 10055 - Change memory allocation to use the
	  heap for a command, since the output has the potential to
	  overflow the stack (as it did here)

	* res/res_jabber.c: Fix 1.4 breakage

2007-06-28 19:44 +0000 [r72493]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in: regenerate the
	  configure script for rizzo

2007-06-28 19:29 +0000 [r72453-72489]  Luigi Rizzo <rizzo@icir.org>

	* configure.ac: add a check for gethostbyname_r so we can simplify
	  the handling e.g. in utils.c Also add comments on a couple of
	  features which are not working on FreeBSD. All the above has been
	  already done in trunk so the merge must be blocked. Can someone
	  please regenerate ./configure ?

	* Makefile, channels/chan_zap.c, main/say.c: Add
	  -Wdeclaration-after-statement to AST_DEVMODE flags to catch
	  variable declarations in the middle of a block. Fix the few
	  instances of the above spotted out by the compiler. All of this
	  has been already done or is not applicable in trunk, so the merge
	  of this change will be blocked.

	* apps/app_meetme.c: cast a time_t so that it does not conflict
	  with the print format. This change was already done on trunk so
	  this change needs to be blocked from merging.

2007-06-27 23:29 +0000 [r72383]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 72373 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
	  3 lines Reinstating patch. This actually fixes the problem,
	  however I was running a development branch without it and
	  mistakenly thought it wasn't fixed. Fixes issue #10010, and
	  #9654: 100% CPU usage caused by an asterisk console losing it's
	  controlling terminal. ........

2007-06-27 23:25 +0000 [r72381]  Joshua Colp <jcolp@digium.com>

	* apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
	  2007) | 2 lines Update documentation to clarify variable usage
	  with MixMonitor. (issue #9494 reported by netoguy) ........

2007-06-27 23:03 +0000 [r72335]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 72333 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
	  2 lines Reverted changes for earlier revisions 72259 to 72261.
	  Issue #9654, #10010 ........

2007-06-27 22:58 +0000 [r72328-72331]  Joshua Colp <jcolp@digium.com>

	* channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
	  our list. Since these are dynamic payloads the other side
	  shouldn't care. (issue #9426 reported by irroot)

	* /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
	  lines Fix issue where queue log events might be missing. (issue
	  #7765 reported by mtryfoss) ........

2007-06-27 21:08 +0000 [r72272]  Russell Bryant <russell@digium.com>

	* /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
	  5 lines Fix a minor issue with parsing the priority number. You
	  could have as much whitespace as you want around a numeric
	  priority, but you couldn't have any whitespace around a special
	  priority like "n" or "hint". (issue #10039, reported by mitheloc,
	  fixed by me) ........

2007-06-27 20:46 +0000 [r72260]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 72259 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
	  4 lines Fixes 100% load when controlling terminal disappears.
	  Issue #9654, #10010 ........

2007-06-27 20:25 +0000 [r72257]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 72256 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
	  lines I may possibly get shot for doing this... but... defer CDR
	  processing until after the channel has been dealt with. This
	  should eliminate all of the issues with channels going funky
	  (SIP/PRI) when you are posting CDRs to a database that is either
	  slow or unavailable and do not want to enable batching. ........

2007-06-27 19:13 +0000 [r72205]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c: use the proper type for storing group number
	  bits so that if someone specifies 'group=42' it will actually
	  work instead of being silently ignored

2007-06-27 18:40 +0000 [r72182-72185]  Jason Parker <jparker@digium.com>

	* /: Blocked revisions 72184 via svnmerge ........ r72184 | qwell |
	  2007-06-27 13:40:15 -0500 (Wed, 27 Jun 2007) | 4 lines Fix
	  another problem in voicemail with missing symbols. Issue 10074,
	  patch by kryptolus, extended to include #if 0'd blocks (just in
	  case) ........

	* apps/app_voicemail.c: Fix another problem in voicemail with
	  missing symbols. Issue 10074, patch by kryptolus, extended to
	  include #if 0'd blocks (just in case)

2007-06-27 17:31 +0000 [r72148]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Make the ast_read_noaudio API call behave better
	  under circumstances where DTMF emulation was happening and a
	  generator was setup. (issue #10065 reported by stevefeinstein)

2007-06-27 17:10 +0000 [r72125]  Jason Parker <jparker@digium.com>

	* channels/chan_gtalk.c: Don't modify a variable that we don't want
	  modified. Make a copy of it instead. Issue 10029, patch by
	  phsultan with slight modifications by me (to remove needless
	  casts).

2007-06-27 16:34 +0000 [r72112]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Only output debug information related to RTCP
	  timestamps when RTCP debug is turned on (issue #10066, patch by
	  me)

2007-06-27 07:58 +0000 [r72042]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
	  1 line for inbound TE calls, we setup the bchannel when we get
	  the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
	  removed some #if 0 areas which weren't used anymore. ........
	  r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
	  1 line isdn_lib.c didn't compile ........

2007-06-27 00:58 +0000 [r72006]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.

2007-06-26 23:02 +0000 [r71953]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing a pointless line. This variable
	  was already set earlier and between then and this line, there is
	  no way that the values on the right side of the assignment could
	  have changed.

2007-06-26 20:36 +0000 [r71915]  Jason Parker <jparker@digium.com>

	* main/rtp.c: Don't dereference a pointer that may be NULL here.
	  Issue 10017.

2007-06-26 19:00 +0000 [r71877]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: A few changes, the ultimate goal of which
	  is to keep better track of the number of messages that a mailbox
	  currently has. A description of the changes: 1. Changed the
	  "updated" field of the vm_state struct to act more as a binary
	  semaphore than a counting semaphore, since its current
	  implementation made the inboxcount function not work properly.
	  This change falls in line with a change made by UPenn with their
	  IMAP setup and helps to sync our changes with theirs. 2.
	  Eliminated some redundant calls to get_vm_state_by_mailbox inside
	  leave_voicemail 3. Use the play_folder variable to keep track of
	  the number of old and new messages in a mailbox as the messages
	  are deleted 4. Added an increment to the number of new messages
	  that was not there previously in the leave_voicemail function

2007-06-26 17:49 +0000 [r71848]  Jason Parker <jparker@digium.com>

	* /: Blocked revisions 71847 via svnmerge ........ r71847 | qwell |
	  2007-06-26 12:49:14 -0500 (Tue, 26 Jun 2007) | 4 lines Don't try
	  to install an init script that doesn't exist. Reported to me on
	  #asterisk on Freenode IRC. ........

2007-06-26 15:47 +0000 [r71796]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixing bug where the authuser was
	  mistakenly pulled from the mailbox string instead of the IMAP
	  user. (closes issue 10054, reported and patched by jaroth)

2007-06-26 12:27 +0000 [r71657-71751]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
	  | 2 lines Issue 10062 - Trying to move a message without
	  selecting one first results in memory corruption ........

	* /, res/res_agi.c: Merged revisions 71656 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
	  | 2 lines Issue 10035 - handle_exec returns a result inconsistent
	  with all of the other AGI commands ........

2007-06-25 14:13 +0000 [r71522-71576]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Build a peer as well when hash323 is
	  enabled in users.conf (issue #9599 reported by asagage)

	* channels/chan_agent.c: Minor tweak for queueing up the unhold
	  frame... this will teach me to do bugs while half asleep. (issue
	  #10046 reported by dimas)

2007-06-25 12:40 +0000 [r71519]  Russell Bryant <russell@digium.com>

	* doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
	  #10048, Matti)

2007-06-25 01:10 +0000 [r71412-71430]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
	  lines Ignore other URIs after the first in a 300 Multiple Choice
	  response. (issue #10041 reported by homesick) ........

	* main/cdr.c: Fix it so 1.4 actually compiles on my box.

	* channels/chan_agent.c: Check to make sure the channel pointer is
	  present before queueing up an unhold frame on it. (issue #10046
	  reported by dimas)

2007-06-24 20:16 +0000 [r71362-71371]  Russell Bryant <russell@digium.com>

	* build_tools/prep_tarball: Include the menuselect-tree file in
	  tarballs to make builds from tarballs a little bit faster

	* main/asterisk.c, /: Merged revisions 71358 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
	  2 lines Revert the patch from issue 9654 due to an unexpected
	  side effect ........

2007-06-24 17:50 +0000 [r71289-71291]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
	  peer->cdr is what we really wanted to use

	* main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
	  Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
	  be able to set variables to the empty string. ........

2007-06-23 03:29 +0000 [r71230]  Steve Murphy <murf@digium.com>

	* main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
	  where clid and src are lost via bridging.

2007-06-22 22:44 +0000 [r71214]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
	  Jun 2007) | 1 line fixed a bug that was introduced by copy and
	  paste in the last commit ..bchannels weren't cleaned properly.
	  ........

2007-06-22 16:05 +0000 [r71128]  Joshua Colp <jcolp@digium.com>

	* /: Blocked revisions 71124 via svnmerge ........ r71124 | file |
	  2007-06-22 12:02:40 -0400 (Fri, 22 Jun 2007) | 2 lines Send an
	  unhold indication when going off hold. (issue #10036 reported by
	  speedy) ........

2007-06-22 15:38 +0000 [r71096-71123]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
	  revisions 70672 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
	  1 line we activate the bchannels in TE mode on incoming calls
	  only when we want to connect the call. ........

	* channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
	  Jun 2007) | 1 line forgot one place .. ........

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
	  Jun 2007) | 1 line on receiption of cause:44 we mark the channel
	  as in use and inform the user about the situation, we need to
	  test the RESTART stuff then. Also shuffled the
	  empty_chan_in_stack function after the bchannel cleaning
	  functions, to avoid race conditions. ........

	* channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
	  Jun 2007) | 1 line when we send out a SETUP, but get no response,
	  we should cleanup everything after reception of a hangup.
	  ........

	* /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
	  1 line restart indicator 0x80 is correct, at least that's what
	  libpri does. ........

	* channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
	  Jun 2007) | 1 line if the bridged partner is mISDN too we should
	  not send dtmf tones, they are transmitted inband always ........

	* channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
	  Jun 2007) | 1 line if we have already some digits, we just stop
	  the tones. ........

2007-06-22 15:00 +0000 [r71068]  Jason Parker <jparker@digium.com>

	* apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
	  revisions 71065 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
	  lines Fix a few silly usages of ast_playstream() - it only ever
	  returns 0... Issue 10035 ........

2007-06-22 14:53 +0000 [r71066]  Brett Bryant <bbryant@digium.com>

	* main/asterisk.c, /: Merged revisions 71064 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
	  10 lines Fixed infinite loop when controlling terminal was lost
	  and return value of input function wasn't checked for errors.
	  This would cause 100% cpu to be taken up. (closes issue #9654,
	  issue #10010) Reported by: mnicholson, and eserra Idea for the
	  patch from mnicholson, patched by me ........

2007-06-22 14:10 +0000 [r71063]  Steve Murphy <murf@digium.com>

	* main/cdr.c: My conditions for merging amaflags info was naive;
	  DOCUMENTATION is the default, although null is possible; theft of
	  user-settable fields is not good. Just copy them, leave them
	  alone.

2007-06-22 03:14 +0000 [r71003]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix a small typo which ... well ...
	  completely broke chan_iax2. oops! (issue #9937, patch by me)

2007-06-21 22:34 +0000 [r70949]  Steve Murphy <murf@digium.com>

	* main/cdr.c, /: Merged revisions 70948 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
	  line This little fix is in response to bug 10016, but may not
	  cure it. The code is wrong, clearly. In a situation where you set
	  the CDR's amaflags, and then ForkCDR, and then set the new CDR's
	  amaflags to some other value, you will see that all CDRs have had
	  their amaflags changed. This is not good. So I fixed it. ........

2007-06-21 21:40 +0000 [r70899]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
	  lines Don't explode if the gain option is specified without a
	  value. (issue #9274 reported by mfarver) ........

2007-06-21 21:14 +0000 [r70866-70883]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Put the thread reading from the socket back
	  in the idle list if it deferred the processing of a full frame to
	  another thread

	* channels/chan_iax2.c: If a full frame is received while one of
	  the iax2 threads is in the middle of handling a full frame for
	  the same call, queue it up for processing by that same thread
	  later instead of dropping it. (issue #9937, patch by me)

2007-06-21 20:19 +0000 [r70841]  Steve Murphy <murf@digium.com>

	* cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
	  line it was pointed out that the cdr_custom config load could get
	  a lock, and under certain circumstances, would never release it.
	  I also noted that the situation where more than one mapping spec
	  was warned about, but did not ignore further mappings as it had
	  promised. I think I have fixed both situations. ........

2007-06-21 19:49 +0000 [r70808]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: When volgain is used don't leave a
	  temporary file behind. (Closes Issue 8514, Reported and patched
	  by ulogic, code reviewed by Jason Parker)

2007-06-21 15:22 +0000 [r70727]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Do not Packet2Packet bridge if packetization settings
	  do not allow it. (issue #9117 reported by phsultan)

2007-06-21 15:21 +0000 [r70726]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Remove a couple of duplicate unlocks

2007-06-21 13:58 +0000 [r70677]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Fix building with ODBC storage enabled.
	  (issue #10025 reported by denisgalvao)

2007-06-21 13:00 +0000 [r70656]  Steve Murphy <murf@digium.com>

	* main/cdr.c: Via complaints aired in asterisk-users, I submit
	  these changes, which allow cdr updates to see macro
	  context/exten, whether hung up or not

2007-06-20 23:32 +0000 [r70554-70612]  Jason Parker <jparker@digium.com>

	* cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
	  Issue 10020, patch by my, with credit to prashant_jois for
	  pointing out the problem.

	* cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
	  condition fix

	* cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
	  when reloading the module. Issue 10022, patch by me, with credit
	  to prashant_jois for finding the bug.

2007-06-20 22:22 +0000 [r70552]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
	  lines Don't overwrite the configured username setting upon a
	  REGISTER. (issue #8565 reported by jsmith) ........

2007-06-20 20:53 +0000 [r70494]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Make sure we clear the previously dialed
	  number if it did not exist. Issue 9958.

2007-06-20 19:29 +0000 [r70445]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
	  | 2 lines Issue 9997 - Timelimit times out the wrong channel
	  ........

2007-06-20 18:46 +0000 [r70397]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
	  5 lines Fix a problem where an established call would not be
	  properly disconnected when a PRI disconnect is received depending
	  on which cause code was received. (issue #9588, original patch by
	  softins, updated patch from jtexter3, and some additional
	  feedback from mhardeman) ........

2007-06-20 17:52 +0000 [r70198-70360]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, main/frame.c: Put the speex packetization values back
	  in but disable it when setting up the smoother.

	* main/frame.c: Don't do packetization/smoother stuff with speex,
	  it doesn't work.

2007-06-20 00:03 +0000 [r70084-70164]  Russell Bryant <russell@digium.com>

	* contrib/scripts/ast_grab_core: don't delete the backtrace in
	  ast_grab_core

	* channels/chan_gtalk.c: Only attempt to queue a hangup on the
	  owner channel if it actually exists. (issue #9795, patch from
	  zandbelt)

2007-06-19 18:23 +0000 [r70062]  Steve Murphy <murf@digium.com>

	* main/channel.c, /: Merged revisions 70053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
	  line This fixes 9246, where channel variables are not available
	  in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
	  consolidate the channel variables during a masquerade, and then
	  copy the merged variables back onto the clone, so the zombie has
	  the same vars that the 'original' has. ........

2007-06-19 17:07 +0000 [r70003]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 69992 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
	  lines Handle the CC field in the RTP header. (issue #9384
	  reported by DoodleHu) ........

2007-06-19 16:46 +0000 [r69991]  Russell Bryant <russell@digium.com>

	* /: Blocked revisions 69990 via svnmerge ........ r69990 | russell
	  | 2007-06-19 11:45:37 -0500 (Tue, 19 Jun 2007) | 12 lines
	  Backport fix for crashes related to subscriptions from 1.4 ...
	  Fix a crash that could occur when handing device state changes.
	  When the state of a device changes, the device state thread tells
	  the extension state handling code that it changed. Then, the
	  extension state code calls the callback in chan_sip so that it
	  can update subscriptions to that extension. A pointer to a
	  sip_pvt structure is passed to this function as the call which
	  needs a NOTIFY sent. However, there was no locking done to ensure
	  that the pvt struct didn't disappear during this process. (issue
	  #9946, reported by tdonahue, patch by me, patch updated to trunk
	  to use the sip_pvt lock wrappers by eliel) ........

2007-06-19 16:24 +0000 [r69987]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 69986 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
	  lines Update BRIDGEPEER variable if set to the new channel name
	  when a masquerade happens. (issue #9699 reported by dimas)
	  ........

2007-06-19 15:22 +0000 [r69944]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Fix a crash that could occur when handing
	  device state changes. When the state of a device changes, the
	  device state thread tells the extension state handling code that
	  it changed. Then, the extension state code calls the callback in
	  chan_sip so that it can update subscriptions to that extension. A
	  pointer to a sip_pvt structure is passed to this function as the
	  call which needs a NOTIFY sent. However, there was no locking
	  done to ensure that the pvt struct didn't disappear during this
	  process. (issue #9946, reported by tdonahue, patch by me, patch
	  updated to trunk to use the sip_pvt lock wrappers by eliel)

2007-06-19 13:55 +0000 [r69805-69895]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
	  lines Perform an extra hangup check just in case. (issue #9589
	  reported by bcnit) ........

	* /, res/res_features.c: Merged revisions 69846 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
	  lines Add parked call extension AFTER the parking slot has been
	  announced, otherwise two threads will try to handle the same
	  channel and it will go kaboom. (issue #9191 reported by japple)
	  ........

	* main/callerid.c: Fix for building on PowerPC under Linux.

2007-06-18 19:48 +0000 [r69796]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Issue 10005 - Segfault with missing
	  arguments, plus fix a missing define for SIP INFO channels

2007-06-18 19:00 +0000 [r69775-69794]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't count RTP timeout when involved in a
	  T38 fax session. (issue #9222 reported by ivoc)

	* /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
	  lines Set the peer name on the dialog to the one configured in
	  sip.conf and NOT the username to be used for authentication
	  attempts. (issue #9967 reported by achauvin) ........

2007-06-18 17:46 +0000 [r69744]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
	  | 2 lines Issue 9998 - Remove SIG prefix, since it's not
	  supported by ksh ........

2007-06-18 16:51 +0000 [r69708]  Joshua Colp <jcolp@digium.com>

	* main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
	  for the first time so that it does not needlessly spit out
	  changed messages when the host really didn't change.

2007-06-18 16:35 +0000 [r69689-69702]  Russell Bryant <russell@digium.com>

	* res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
	  build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
	  include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
	  To prevent 92138749238754 more reports of "I have unixodbc
	  installed, but still can't build *_odbc.so!", check for ltdl
	  directly, instead of just listing it as another library to
	  include in the unixodbc check in the configure script. This also
	  makes ltdl show up as a dependency in menuselect so people know
	  what to go install. (related to issue #9989, patch by me)

	* build_tools/prep_moduledeps: Change the use of "echo -e" to
	  "printf". On systems where /bin/sh is not bash, most of the lines
	  in menuselect-tree were getting a "-e" at the beginning of every
	  line. I'm surprised nobody noticed this, but I think the XML
	  parser was being very nice and ignoring them.

2007-06-18 16:04 +0000 [r69661-69668]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't defer the BYE till later on a transfer
	  when the transfer itself goes kaboom and has no hope of working.

	* channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
	  something we never locked, and better handle a specific scenario
	  with doing an attended transfer between two non-bridged calls.

2007-06-18 15:46 +0000 [r69660]  Russell Bryant <russell@digium.com>

	* Makefile: Tweak paths for BSD systems (issue #10001, stuarth)

2007-06-18 13:55 +0000 [r69625]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Fix issue where it would be possible for the
	  negotiated codecs to get set back to nothing. (issue #9992
	  reported by yehavi)

2007-06-15  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.5 released.

2007-06-15 20:18 +0000 [r69579]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Fix a silly deadlock in res_features that I
	  found while debugging on one of blitzrage's test machines. It was
	  one of the situations where he was seeing hung channels, and may
	  be the cause of some of the reports from other people. (related
	  to issue #9235)

2007-06-15 19:23 +0000 [r69558]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Add support for setting the maximum
	  length of acceptable DTMF in SpeechBackground.

2007-06-15 15:27 +0000 [r69518]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
	  "SUCCESS" for both an answer of the incoming call on the trunk,
	  or if the trunk reached its ring timeout. This patch changes the
	  variable to say "RINGTIMEOUT" in that case. (issue #9973,
	  reported by n00dle, patch by me)

2007-06-14 23:22 +0000 [r69434-69470]  Jason Parker <jparker@digium.com>

	* main/config.c, /: Merged revisions 69469 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
	  lines Fix an issue where the line number in an unterminated
	  comment block error message would show the wrong line number.
	  "Reported" to me on #asterisk (somebody posted an error message,
	  and I happened to catch it) ........

	* sounds/Makefile: Update to latest versions of sound files.

2007-06-14 21:50 +0000 [r69392]  Kevin P. Fleming <kpfleming@digium.com>

	* cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
	  cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
	  main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
	  apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
	  main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
	  channels/chan_iax2.c: use ast_localtime() in every place
	  localtime_r() was being used

2007-06-14 21:08 +0000 [r69358]  Russell Bryant <russell@digium.com>

	* main/say.c: Fix some problems with saying dates and times for the
	  "tw" langauge (issue #9964, ljmid)

2007-06-14 15:21 +0000 [r69259]  Jason Parker <jparker@digium.com>

	* funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
	  2007) | 4 lines Change a quite broken while loop to a for loop,
	  so "continue;" works as expected instead of eating 99% CPU...
	  Issue 9966, patch by me. ........

2007-06-13 21:19 +0000 [r69184-69222]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Whoops...

	* channels/chan_iax2.c: Let's make chan_iax2 media only native
	  transfers actually work. (issue #9376 reported by simone
	  cittadini)

	* channels/iax2-parser.c: Add TXMEDIA to list so that it is
	  properly displayed during iax2 packet output.

2007-06-13 19:57 +0000 [r69183]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Move the logic for destroying a call when no
	  response is received to a BYE outside of the block that checks
	  for FLAG_FATAL to be set. This flag is only set when the packet
	  is transmitted with the reliability set to XMIT_CRITICAL when the
	  original packet is transmitted. A BYE is always sent with it set
	  to XMIT_RELIABLE, meaning this code could never be encountered.
	  This resulted in seeing some SIP channels that would never go
	  away with the last packet sent being a BYE. (part of issue #9235,
	  patch from jcmoore)

2007-06-13 19:41 +0000 [r69181]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Contains a patch for fixing an encoding
	  problem when using Outlook to view voicemail emails and
	  attachments. This fix has also been tested on Thunderbird,
	  Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
	  patched by mutterc)

2007-06-13 19:08 +0000 [r69128-69144]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Really ignore NULL frames and check whether
	  the channel hungup or not. (issue #9912 reported by junky)

	* /, main/app.c: Merged revisions 69127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
	  lines Return group counting to previous behavior where you could
	  only have one group per category. (issue #9711 reported by
	  irroot) ........

2007-06-13 16:56 +0000 [r69016-69071]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Clarify a bit of logic. This doesn't change
	  behavior in any way, but it is helpful when following the logic
	  to debug problems like 9235.

	* channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
	  was accessed without the lock held. This issue was reported to me
	  via email by Dmitry Mishchenko. Thanks!

	* cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
	  in #asterisk-bugs. PQclear() was not called on the result
	  structure after doing a PQexec(). Also, fix up some formatting in
	  passing.

2007-06-12 19:36 +0000 [r69012-69014]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Change the full frame dropping log message
	  to debug to avoid future bug reports.

	* channels/chan_iax2.c: Schedule the sending of a PING packet a
	  second later than previously so that it does not collide with the
	  LAGRQ.

2007-06-12 19:13 +0000 [r69010]  Russell Bryant <russell@digium.com>

	* main/channel.c: In ast_channel_make_compatible(), just return if
	  the channels' read and write formats already match up. There are
	  code paths that call this function on a pair of channels multiple
	  times. This made calls fail that were using g729 in some cases.
	  The reason is that codec_g729a will unregister itself from the
	  list of available translators will all licenses are in use. So,
	  the first time the function got called, the right translation
	  path was allocated. However, the second time it got called, the
	  code would not find a translation path to/from g729 and make the
	  call fail, even if the channel actually already had a g729
	  translation path allocated. (SPD-32)

2007-06-12 14:23 +0000 [r68922]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 68921 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
	  lines Bring RTP back to Asterisk at the end of a native bridge no
	  matter what. ........

2007-06-11 21:20 +0000 [r68814]  Jason Parker <jparker@digium.com>

	* include/asterisk/time.h: Solaris 10 sometimes (?) needs this
	  include in order to have NULL defined.

2007-06-11 20:45 +0000 [r68781]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
	  used

2007-06-11 16:57 +0000 [r68733]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 68732 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
	  1 line added check for NULL Pointer when calling misdn_new.
	  Asterisk does not allow us to create channels anymore when stop
	  gracefully is used :). also modified the restart_indicator to 0
	  ........

2007-06-11 14:33 +0000 [r68683]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, /: Merged revisions 68682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
	  lines Improve deadlock handling of the channel list. (issue #8376
	  reported by one47) ........

2007-06-11 10:29 +0000 [r68644]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/ie.c,
	  channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
	  1 line fixed problem that the dummybc chanels had no lock,
	  checking for the lock now. Also fixed the channel restart stuff,
	  we can now specify and restart particular channels too. ........

2007-06-11 04:21 +0000 [r68595]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* pbx/pbx_config.c: "dialplan save" produced garbage in the config
	  file

2007-06-08 22:23 +0000 [r68527]  Russell Bryant <russell@digium.com>

	* /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
	  4 lines Don't automatically hang up after running Dictate so that
	  callers can exit cleanly using '#' (closes issue #9577, patch
	  from Thomas Andrews) ........

2007-06-08 15:52 +0000 [r68450]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: actually remember the type/subclass of full
	  frames that are in process

2007-06-08 00:17 +0000 [r68370-68401]  Joshua Colp <jcolp@digium.com>

	* /, main/say.c: Merged revisions 68397 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
	  lines Don't call ast_waitstream_full when the control file
	  descriptor and audio file descriptor are not set, simply call
	  ast_waitstream! (issue #8530 reported by rickead2000) ........

	* main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
	  lines Do a DNS lookup immediately upon calling the dnsmgr
	  function, don't wait until a refresh happens. (issue #9097
	  reported by plack) ........

2007-06-07 23:14 +0000 [r68354]  Russell Bryant <russell@digium.com>

	* /, main/say.c: Merged revisions 68351 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
	  3 lines Fix a problem where saying a character wouldn't properly
	  break out when the caller pressed '#' (issue #8113, reported by
	  patbaker82, patch from jamesgolovich (hey, long time no see!) and
	  patbaker82) ........

2007-06-07 23:00 +0000 [r68326]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix incorrect French syntax of "old
	  messages". Request for feedback was sent to asterisk-dev mailing
	  list, with little response. Issue 9118, patch by junky.

2007-06-07 22:14 +0000 [r68313]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: some improvements to the IAX2 full frame
	  dropping logic recently added: - use inaddrcmp(), since we have
	  it - output the type of frame and subclass being dropped, and the
	  type/subclass that is already being processed (which caused the
	  drop)

2007-06-07 21:16 +0000 [r68280]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
	  queue members when using realtime configuration for queues. Also,
	  remove an unneeded leading slash for the astdb family. (issue
	  #9911, patch by atis)

2007-06-07 20:25 +0000 [r68211-68249]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix an issue with newer phones which
	  require packets be padded out to the correct length. Issue 9887,
	  patch by DEA.

	* apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
	  lines Don't try to save voicemail greetings unless the user
	  presses '1' to accept/save. Issue 9904, patch by me. ........

2007-06-07 19:47 +0000 [r68198]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
	  check to make sure that greetings get stored properly. (Issue
	  8016, reported by edhorton, patched by alamantia with
	  modification by me. Thanks to Jason Parker for the advice on
	  this).

2007-06-07 19:46 +0000 [r68196]  Olle Johansson <oej@edvina.net>

	* channels/chan_features.c: Disable chan_features by default in
	  menuselect

2007-06-07 19:30 +0000 [r68192]  Russell Bryant <russell@digium.com>

	* main/strcompat.c: Include stdarg.h for build issues on Solaris
	  (issue #9381)

2007-06-07 18:39 +0000 [r68071-68157]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Fix logic when doing a name based channel search
	  for a structure when you want to start from a specific point in
	  the channel list. (issue #9324 reported by slavon)

	* apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
	  lines Allow the 'g' option to work if used with the 'S' option.
	  (issue #9888 reported by gasparz) ........

2007-06-07 10:00 +0000 [r67993-68030]  Olle Johansson <oej@edvina.net>

	* res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
	  forget.

	* res/res_jabber.c: Ok, we found out that this is not about if you
	  have any *active* clients using TLS, but if you have initialized
	  TLS at all during the lifetime of the module. So if you reload to
	  disable TLS, it won't help.

	* res/res_jabber.c: If you have a jabber client that uses TLS,
	  refuse unload. Bad fix, but will prevent crashes while we are
	  trying to find a workaround. Iksemel development seems to have
	  stalled and we might have to stop using the TCP/TLS connections
	  in that library and use our own, which would scale better from a
	  poll/select perspective I guess. It would also make it easier to
	  migrate to OpenSSL and stop Asterisk from depending on both
	  OpenSSL and GnuTLS.

	* include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
	  sure we can unload res_jabber. Patch by phsultan - thanks! Due to
	  a bug in the iksemel library, this will not work if you are using
	  GTLS in the connection. That's being investigated. If you figure
	  out a way to handle that without us having to patch iksemel, let
	  us know in the bug report. Thanks.

2007-06-07 00:10 +0000 [r67924-67941]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
	  lines Only notify the devicestate system of a peer state change
	  when the peer is built from the config file. (issue #9900
	  reported by arkadia) ........

	* main/file.c: Properly handle cases where a stream can't be
	  written to. (issue #9757 reported by junky)

2007-06-06 22:08 +0000 [r67862-67872]  Russell Bryant <russell@digium.com>

	* res/res_snmp.c: Disable reload functionality in res_snmp. It is
	  not possible to initialize the snmp library more than once
	  without completely unloading the module and loading it again.
	  (issue #9571, reported by hristo, additional helpful debug
	  information from festr, patch from me)

	* channels/chan_sip.c: Fix a crash when doing call pickups with SIP
	  phones. The code unlocked the channel when it should not have.
	  (issue #9652, reported by corruptor, fixed by me)

2007-06-06 19:26 +0000 [r67804]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
	  under a specific set of circumstances: 1. VoiceMailMain was
	  configured in the dialplan with an extension as its argument 2. A
	  message was left for this mailbox 3. Tried to call VoiceMailMain
	  but hung up before entering password. This was fixed by checking
	  that a pointer was non-null prior to trying to dereference it.
	  (Issue 9810, reported by xmarksthespot, patched by Corydon76 with
	  modifications by me).

2007-06-06 16:55 +0000 [r67716]  Russell Bryant <russell@digium.com>

	* main/channel.c, /, include/asterisk/linkedlists.h: Merged
	  revisions 67715 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
	  5 lines We have some bug reports showing crashes due to a double
	  free of a channel. Add a sanity check to ast_channel_free() to
	  make sure we don't go on trying to free a channel that wasn't
	  found in the channel list. (issue #8850, and others...) ........

2007-06-06 13:30 +0000 [r67594-67650]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 67649 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
	  lines Reinvite the RTP back to the Asterisk machine when the
	  timeout happens. (issue #9888 reported by gasparz) ........

	* main/translate.c: Fix plc_samples warning when registering a
	  translator. (issue #9897 reported by xylome)

	* apps/app_directed_pickup.c: Include macroexten while searching
	  for a channel to pick up in case they are in a macro. (issue
	  #9491 reported by jamesb63)

	* res/res_agi.c: Make the new "agi debug off" CLI command work.
	  (issue #9890 reported by eliel)

	* /, main/devicestate.c: Merged revisions 67593 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
	  lines Revert channel name splitting fix for Zap. The moral of the
	  story is don't use - in your user/peer names. (issue #9668
	  reported by stevedavies) ........

2007-06-05 23:01 +0000 [r67558]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: Fix some crashes related to the use of the
	  "meetme" CLI command. The code for this command was not locking
	  the conference list at all. (issue #9351, reported by and patch
	  submitted by Junk-Y, committed patch is different and by me)

2007-06-05 21:30 +0000 [r67526]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
	  9883, wherein macros were not allowing the includes construct.
	  fixed and tested, looks OK. Now includes can serve as an adjunct
	  to catch.

2007-06-05 20:53 +0000 [r67457-67492]  Russell Bryant <russell@digium.com>

	* include/asterisk/linkedlists.h: This bug has been hanging over my
	  head ever since I wrote this SLA code. Every time I tried to go
	  debug it by adding some debug output, the behavior would change.
	  It turns out I wasn't crazy. I had the following piece of code:
	  if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
	  AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
	  conditional statement didn't do much good at all. It always ran
	  at least all of the macro minus the first statement, so I was
	  seeing list entries magically disappear when they weren't
	  supposed to. After many hours of debugging, I have come to this
	  extremely irritating fix. :) (issues #9581, #9497)

	* channels/chan_zap.c: Suppress a bunch of debug output unless
	  option_debug is on

2007-06-05 18:32 +0000 [r67424]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
	  saved to IMAP storage using extensions other than gsm were unable
	  to be played over the phone. (Issue 9786, reporter:
	  xmarksthespot, Patched by xmarksthe spot with revisions by me,
	  reviewed by Russell Bryant).

2007-06-05 18:18 +0000 [r67421]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Correctly update date/time on devices
	  throughout the life of the device, instead of just at
	  registration. Issue 9152, yet another patch by DEA.

2007-06-05 18:17 +0000 [r67420]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: Added code to automatically add a default case to
	  switches that don't have one. In some cases, rather than fall
	  thru, it results in a goto with -1 result, which terminates the
	  extension; a sort of dialplan seqfault, sort of. This was
	  required to fix bug reported in 9881

2007-06-05 17:07 +0000 [r67360-67372]  Russell Bryant <russell@digium.com>

	* main/channel.c: Handle a failure in malloc() in
	  ast_safe_string_alloc()

	* main/channel.c: Fix a problem that showed itself by causing Zap
	  channel names to be completely bogus on my machine.
	  ast_safe_string_alloc() was broken. It called vsnprintf() on a
	  va_args list twice without re-initializing it. After the first
	  usage, va_end() and va_start() must be called again.

2007-06-05 16:14 +0000 [r67329-67334]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
	  1 line briding is a bool, fixed copy and paste issue. ........

	* channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
	  Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
	  cb_events function a lot. Commented the different possibilities a
	  bit and made functions of shared code. When the dialed extension
	  does not exist in the extensions.conf we'll jump into the 'i'
	  extension if this does exist, else we disconnect the call with
	  the cause:1 = No Route to Destination. ........

2007-06-05 15:51 +0000 [r67308]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, main/loader.c, include/asterisk/module.h: When
	  shutting down "gracefully", go through and run the unload()
	  callbacks for all of the modules. "stop now" is considered a
	  non-graceful shutdown and will not go through this process.
	  (issue #9804, reported by chrisost, patch by me)

2007-06-05 15:22 +0000 [r67304]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Only muck with the thread structure if an
	  idle one was found/created.

2007-06-05 14:35 +0000 [r67270]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: ensure that a burst of full frames
	  (AST_FRAME_DTMF being the prime example) will not be processed
	  out of order... this is a brute force fix, but seems to be the
	  safest fix for now (thanks to the Digium PQ department for
	  finding this bug)

2007-06-05 10:25 +0000 [r67210]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn_config.c, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
	  1 line added possibility to deactivate bridging per port ........

2007-06-04 23:43 +0000 [r67162]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
	  | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
	  ........

2007-06-04 23:31 +0000 [r67158]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
	  structure can disappear and the code did not account for it and
	  crashes. (issues #9642, #9569, #9666, probably others ... based
	  on the work by stevedavies and mihai, with additional changes
	  from me)

2007-06-04 23:26 +0000 [r67121-67156]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix for skinny keepalives. If there is no
	  traffic from the phone for (keep_alive * 1100) ms (arbitrarily
	  adding 10% for network issues, etc), unregister the device. Issue
	  8394, patch by DEA.

	* channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
	  to the recent fix for chan_skinny) Issue 9855, patch by DEA.

2007-06-04 22:28 +0000 [r67119]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Add comments for two functions that get
	  called with the appropriate call locked, but perform operations
	  that could result in the pvt structure getting destroyed before
	  returning again, causing numerous seg faults all over the module.
	  (inspired by issues #9642, #9569, and #9666, and the work done by
	  stevedavies and mihai)

2007-06-04 21:59 +0000 [r67073]  Steve Murphy <murf@digium.com>

	* main/cdr.c: This typo has been here since 1.4 forked. It has been
	  the source of heartburn to many a dialplan/CDR programmer.

2007-06-04 21:47 +0000 [r67071]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)

2007-06-04 19:31 +0000 [r67064-67068]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Better handle SIP devices that say they have
	  SDP content... but really don't. (issue #9398 reported by
	  mthomasslo)

	* apps/app_dial.c: Initialize cidname variable to nothing since it
	  may be used without having been touched. (issue #9661 reported by
	  dimas)

	* res/res_features.c: Returning a value that indicates the parking
	  of a call was a success when it really wasn't (because the
	  parking slot selected was in use) is the wrong thing to do.
	  (issue #9723 reported by mdu113)

2007-06-04 17:11 +0000 [r67061]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* 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.mandrake.zaptel,
	  contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
	  | 2 lines Add revision Id tags (by request of tzafrir) ........

2007-06-04 16:02 +0000 [r67026]  Russell Bryant <russell@digium.com>

	* configure, configure.ac: Change the configure script to build a
	  test program against libcurl to make sure the results from
	  curl-config can be used to compile successfully. This is intended
	  to help prevent a situation where you are cross compiling, and
	  the configure script finds the curl library installed on the
	  host. (issue #9865, reported and patched by zandbelt)

2007-06-04 15:50 +0000 [r67021]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_jabber.c: Issue 9739 - Malformed jid causes a crash

2007-06-04 15:47 +0000 [r67018-67020]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
	  handling an implicit ACK to a frame that was marked as the final
	  transmission for a call, don't call iax2_destroy() for that call
	  while the global frame queue is still locked. There is a very
	  nice explanation of the deadlock in the report. (issue #9663,
	  thorough report and patch from stevedavies, additional positive
	  test reports from mihai and joff_oconnell)

	* include/asterisk/stringfields.h: Fix some compiler warnings in
	  C++ modules. (issue #9866, reported by osk, patch by Corydon76)

2007-06-01 21:45 +0000 [r66919]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_odbc.c: On some drivers, deallocating the statement
	  handle isn't enough. We also have to clear the cursor (nice,
	  Oracle)

2007-06-01 21:31 +0000 [r66897-66917]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Removing extraneous debugging lines from
	  revision 66897. Sorry :)

	* apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
	  storage. Attachments with format specified as gsm were duplicated
	  (i.e. two attachments) were left. Thank you very much to
	  xmarksthespot for submitting the patch that fixed this. (Issues
	  9787 and 8873, Reported by xmarksthespot and jerjer, patched by
	  xmarksthespot)

2007-06-01 19:41 +0000 [r66879-66881]  Russell Bryant <russell@digium.com>

	* channels/chan_skinny.c: Changes to the way DTMF is handled in the
	  core broke dialing in chan_skinny. This patch makes chan_skinny
	  usable again. I did not end up testing this, but there are
	  multiple positive test reports listed in the bug report. (issue
	  #9596, reported by pj, testing by pj and mvanbaak, and the fix
	  was written by DEA)

	* apps/app_page.c: List app_meetme as a module that app_page
	  depends on.

2007-05-31 23:03 +0000 [r66821]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* doc/asterisk.8: Issue 9850 - update preferred command line syntax

2007-05-31 18:41 +0000 [r66775]  Russell Bryant <russell@digium.com>

	* res/res_speech.c, include/asterisk/app.h,
	  include/asterisk/speech.h: Change a couple of header files to not
	  use "new", which is a reserved keyword in C++. (issue #9830,
	  reported by osk)

2007-05-31 17:15 +0000 [r66770]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
	  | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
	  Issue 8329 will remain unfixed for pbx_realtime, but only because
	  we lack core API to do it. ........

2007-05-31 16:14 +0000 [r66768]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
	  lines It is now possible for this path of execution to have the
	  frame pointer be NULL, therefore we need to check for it before
	  trying to access it. (issue #9836 reported by barthpbx) ........

2007-05-30 23:26 +0000 [r66671]  Mark Michelson <mmichelson@digium.com>

	* apps/app_voicemail.c: Fixed seg-faults when recording greetings
	  in voicemail with IMAP enabled. (Issue No. 9735, reported by
	  xmarksthespot, patched by me)

2007-05-30 17:28 +0000 [r66602-66639]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Silly me for having out of date source! Oh
	  well... I'm still leaving my comment.

	* channels/chan_sip.c: When calling some peer/host that may not
	  exist/reply back... don't keep the dialog in memory for all of
	  eternity.

	* channels/chan_zap.c, channels/chan_features.c: Change how channel
	  names are generated a bit. (issue #9825 reported by eldadran)

2007-05-29 21:56 +0000 [r66538]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
	  | 2 lines If the value of a variable passed to FIELDQTY is blank,
	  then FIELDQTY should return 0, not 1. ........

2007-05-29 19:32 +0000 [r66474-66503]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Properly handle 408 request timeout -
	  according to the RFC, the dialog dies if a request in a dialog
	  gets this response.

	* channels/chan_sip.c: Don't issue hangup on hangup on hangup on
	  hangup (for jcmoore)

2007-05-29 16:44 +0000 [r66437]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Handle cases where a frame may have no data. (issue
	  #9519 reported by dmb)

2007-05-29 16:07 +0000 [r66404-66414]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't reset hangupcause if we already have
	  one

	* channels/chan_sip.c: Tracking down hanging channels, killing them
	  one by one. Issue #9235 and related

2007-05-29 15:43 +0000 [r66398]  Joshua Colp <jcolp@digium.com>

	* doc/datastores.txt: Update datastores documentation. (issue #9801
	  reported by mnicholson)

2007-05-29 09:41 +0000 [r66363]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
	  lines Issue #9802 - Change inuse counter on CANCEL ........

2007-05-28 23:16 +0000 [r66312]  Joshua Colp <jcolp@digium.com>

	* channels/chan_zap.c: Make the usedistinctiveringdetection option
	  work again. (issue #9823 reported by premeau)

2007-05-27 04:12 +0000 [r66244]  Jason Parker <jparker@digium.com>

	* channels/chan_zap.c: I don't know what this was trying to do, but
	  it's clearly incorrect. Issues 9808 and 9809.

2007-05-25 14:43 +0000 [r66160]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: have to check for OSP toolkit _after_
	  checking for OpenSSL

2007-05-25 14:41 +0000 [r66159]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, main/say.c: Merged revisions 66127 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
	  | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
	  ........

2007-05-25 14:28 +0000 [r66157]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
	  res/res_jabber.c: handle the GNUTLS library properly in the
	  configure script and build system don't build in OSP support
	  unless we have found and are allowed to use SSL support

2007-05-24 22:23 +0000 [r66076]  Russell Bryant <russell@digium.com>

	* main/channel.c: if the string field init fails, clean up the
	  stuff that was allocated already

2007-05-24 22:16 +0000 [r66074]  Joshua Colp <jcolp@digium.com>

	* main/slinfactory.c: Fix slinfactory logic when dealing with
	  frames coming in that may already be in the signed linear format.

2007-05-24 22:07 +0000 [r66068-66070]  Russell Bryant <russell@digium.com>

	* main/channel.c: Check the result of ast_string_field_init() in
	  ast_channel_alloc()

	* main/rtp.c: Make 1.4 build on my machine, too..

2007-05-24 20:54 +0000 [r66029-66030]  Jason Parker <jparker@digium.com>

	* configure: Rebuild configure script for previous ar fix.

	* configure.ac: Following moving strip to AC_PATH_TOOL, we need to
	  do something similar for ar.

2007-05-24 20:42 +0000 [r65978-66026]  Russell Bryant <russell@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac:
	  Checking for the strip application needs to be done with
	  AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
	  compilation environments.

	* Makefile: Clear CFLAGS before running make for menuselect. (issue
	  #9784, reported by ovi, patch by me)

2007-05-24 18:28 +0000 [r65965-65967]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_gtalk.c: oops, use #ifdef instead of #if

	* channels/chan_gtalk.c: don't reference GnuTLS headers and
	  functions unless the configure script found it

	* main/rtp.c: don't use uninitialized variables

2007-05-24 15:27 +0000 [r65902]  Joshua Colp <jcolp@digium.com>

	* main/manager.c: Add the ability to blacklist certain commands
	  from being executed using the Command AMI action. (issue #9240
	  reported by junky)

2007-05-24 15:26 +0000 [r65892-65901]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
	  core dump since the GnuTLS interface did not support
	  multithreading correctly.

	* channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
	  Patch by phsultan. Thanks!

2007-05-24 15:16 +0000 [r65877-65883]  Jason Parker <jparker@digium.com>

	* .cleancount: Update cleancount for that last commit - just for
	  good measure.

	* include/asterisk/translate.h, codecs/codec_speex.c,
	  main/translate.c, codecs/codec_ilbc.c: Fix handling of
	  zero-length frames when a codec is capable of native PLC. Issue
	  9183, patch by Mihai.

2007-05-24 15:08 +0000 [r65866]  Dwayne M. Hubbard <dhubbard@digium.com>

	* funcs/func_math.c: merged qwell's func_math patch for issue 9507

2007-05-24 15:08 +0000 [r65863]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: I like it when the RTP stack compiles myself...

2007-05-24 15:05 +0000 [r65857]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
	  when calling from gtalk to SIP over nat.

2007-05-24 15:04 +0000 [r65842-65853]  Russell Bryant <russell@digium.com>

	* apps/app_festival.c: Ensure that frames are fully initialized.
	  This will probably fix getting weird timestamp log messages in
	  logs when using the Festival app. (issue #9781, patch by me)

	* main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
	  code would result in oscillating and incorrect data.
	  Additionally, the RTT would sometimes report negative values due
	  to incorrect calculations. (issue #9601, patch from davetroy)

2007-05-24 14:48 +0000 [r65841]  Olle Johansson <oej@edvina.net>

	* channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
	  jingle

2007-05-24 14:42 +0000 [r65839]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
	  lines Allow RFC2833 to be negotiated when an INVITE comes in
	  without SDP and is not matched to a user or peer. (issue #9546
	  reported by mcrawford) ........

2007-05-24 14:38 +0000 [r65836]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
	  Fix "login" as component to jabber server. ...and, by accident,
	  fix a bug in chan_sip for stopping a loop on retransmits of BYE
	  requests.

2007-05-24 09:37 +0000 [r65768]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
	  Mai 2007) | 1 line we should only activate the generator in
	  chan_misdn, when asterisk hask not yet taken the call
	  (WAITING4DIGS state). Alerting audio will be generated fomr
	  asterisk for example. ........

2007-05-23 20:59 +0000 [r65677-65685]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: start the delayed PBX when receive voice or
	  video full frames as well, and comment this delayed-PBX activity

	* /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
	  | 2 lines ensure that variables are set on a newly created
	  channel before we start a PBX on it ........

	* channels/chan_iax2.c: clear the 'delay PBX' flag when we are
	  ready to start the PBX

	* channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
	  channel until we have some sort of response to our ACCEPT (ACK or
	  anything else)

	* /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
	  | 2 lines if we are going to set variables on a newly created
	  channel, it should be done *before* we start the PBX on it
	  ........

2007-05-23 13:07 +0000 [r65589]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
	  3 lines Revert revision 62417 as someone reported problems with
	  it to Mark. This was related to issue #9588. ........

2007-05-22 20:25 +0000 [r65541]  Kevin P. Fleming <kpfleming@digium.com>

	* build_tools/make_version: when building a version string for a
	  developer branch, include the base branch in the version string

2007-05-22 18:40 +0000 [r65501]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
	  dependency for app_voicemail and chan_zap (Thanks to mnicholson
	  for pointing it out)

2007-05-22 15:04 +0000 [r65452]  Joshua Colp <jcolp@digium.com>

	* apps/app_meetme.c: Remove a double const.

2007-05-22 14:02 +0000 [r65408]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Fix a problem with flag recognition.

2007-05-22 13:09 +0000 [r65394]  Russell Bryant <russell@digium.com>

	* /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
	  4 lines Fix a memory leak that I just noticed in the device state
	  handling in app_queue. On most device state changes, it would
	  leak roughly 8 to 64 bytes (the length of the name of the
	  device). ........

2007-05-22 08:12 +0000 [r65342]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
	  Mai 2007) | 1 line we stop the tones only when we're in the
	  pre-call phase, otherwise e.g. when in CONNECTED state we should
	  not stop tones when we receive an Information Message ........

2007-05-20 17:59 +0000 [r65250]  Joshua Colp <jcolp@digium.com>

	* res/res_agi.c: res_agi needs to export two symbols
	  (ast_agi_register and ast_agi_unregister) for usage by others.
	  (issue #9755 reported by mnicholson)

2007-05-18 22:26 +0000 [r65200-65201]  Steve Murphy <murf@digium.com>

	* main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
	  to main/cdr.c, and neither did I. This is the remainder of the
	  9717 patch, the fix for the run-away FAIL status for a call

	* apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
	  65172 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
	  line This update will fix the situation that occurs as described
	  by 9717, where when several targets are specified for a dial, if
	  any one them reports FAIL, the whole call gets FAIL, even though
	  others were ringing OK. I rearranged the priorities, so that a
	  new disposition, NULL, is at the lowest level, and the
	  disposition get init'd to NULL. Then, next up is FAIL, and next
	  up is BUSY, then NOANSWER, then ANSWERED. All the related set
	  routines will only do so if the disposition value to be set to is
	  greater than what's already there. This gives the intended
	  effect. So, if all the targets are busy, you'd get BUSY for the
	  call disposition. If all get BUSY, but one, and that one rings is
	  not answered, you get NOANSWER. If by some freak of nature, the
	  NULL value doesn't get overridden, then the disp2str routine will
	  report NOANSWER as before. ........

2007-05-18 18:16 +0000 [r65041-65123]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
	  lines Not getting an ACK to a 200 OK in the initial invite is
	  critical to the call. ........

	* /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
	  lines Issue 9235 - part of the problem, maybe not all. Please
	  retry with this patch (and no other patch) if you have problems
	  with hanging SIP channels. Thank you. A special Thank You to
	  WeBRainstorm that gave me access to his system. ........

	* channels/chan_sip.c: - Adding support for putting calls OFF hold
	  with a re-invite with blank SDP. This was a bug found while doing
	  tests at SIPit in Antwerp. - In order to not duplicate code, I
	  restructured some of the code for putting calls on/off hold.
	  Thanks DEA for reminding me. This fix has been asleep in the
	  videocaps branch until now.

2007-05-18 12:40 +0000 [r65039]  Christian Richter <christian.richter@beronet.com>

	* /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
	  revisions 65007 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
	  1 line fixed a warning regarding Keypad encoding. encode the IE
	  sending_complete at the right position. ........

2007-05-18 10:37 +0000 [r64974]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 9487 - stop media flows at hangup of
	  call

2007-05-18 08:58 +0000 [r64904]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
	  Mai 2007) | 1 line we *need* to send a PROCEEDING when
	  sending_complete is set, even if need_more_infos is requested.
	  ........

2007-05-18 02:48 +0000 [r64868]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: Fix a small bug I noticed while working on
	  something else. app_queue did not unregister its device state
	  monitoring callback in unload_module(). So, this would make
	  Asterisk crash on the first device state change after you unload
	  the module.

2007-05-17 21:19 +0000 [r64820]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, include/asterisk/linkedlists.h: Merged revisions 64819 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007)
	  | 2 lines How is it that we never caught that this is returning
	  the opposite of our documentation, until now? ........

2007-05-17 16:53 +0000 [r64761]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
	  lines If we have a negative current message, we shouldn't go back
	  even further... Issue 9727. ........

2007-05-17 16:52 +0000 [r64756-64759]  Russell Bryant <russell@digium.com>

	* contrib/scripts/astxs (removed): Remove script that is no longer
	  functional since the build system was redone. (issue #9340,
	  reported by junky)

	* apps/app_dial.c: Increase the size of a buffer to support longer
	  dial strings for channels. (issue #9291, reported and fix
	  suggested by meni)

2007-05-17 16:10 +0000 [r64720-64754]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Even more direct RTP setup fixes! Don't
	  allow a codec that isn't supported to creep into the SDP of
	  either side. (issue #9446 reported by marcelbarbulescu)

	* apps/app_voicemail.c: Fix authuser support. (issue #9740 reported
	  by xmarksthespot)

2007-05-17 06:13 +0000 [r64686]  Russell Bryant <russell@digium.com>

	* README: Update the main README to reflect the new build process
	  for 1.4 and above. (issue #9725, patch by eliel)

2007-05-16 11:01 +0000 [r64516-64609]  Olle Johansson <oej@edvina.net>

	* /: Blocking patch already in this code

	* channels/chan_sip.c: Fix auth on BYE. (Different patch than for
	  1.2)

	* channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE

	* channels/chan_sip.c: Final part of issue #9483 - fixing
	  transfer() of sip calls in the dial plan (twilson)

	* channels/chan_sip.c: Issue #9439 - properly handle username
	  parameters in SIP uri.

	* /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
	  lines Support SIP uri's starting with SIP: and sip: (reported by
	  Tony Mountfield on the mailing list. Thanks!) ........

	* /, channels/chan_sip.c: Merged following patch with a lot of
	  changes for 1.4 ------ Merged revisions 64514 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
	  lines Issue #9726 - rlister - Better logging for ACL denials
	  While at it, also added better logging and handling of peers that
	  are not supposed to register. My patch, stole the issue report
	  from Russell. My apologies, Russell :-) ........

2007-05-16 08:44 +0000 [r64515]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
	  Mai 2007) | 1 line in the case immediate=yes, we directly jump
	  into the dialplan, where people can use PlayTones to indicate a
	  Dialtone, so we don't need to to that by ourself. also we should
	  not do a dialtone_indicate for incoming calls on a TE port in
	  overlapdialmode. ........

2007-05-15 19:52 +0000 [r64353-64426]  Russell Bryant <russell@digium.com>

	* res/res_features.c: Properly fix a problem that occurs when you
	  set PARKINGEXTEN to an exten where a call is already parked.
	  (issue #9723, patch by me)

	* res/res_features.c: When someone requests a specific parking
	  space using the PARKINGEXTEN variable, ensure that no other
	  caller is already there. (issue #9723, reported by mdu113, patch
	  by me)

2007-05-14 19:26 +0000 [r64324]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit
	  more

2007-05-14 19:13 +0000 [r64306]  Russell Bryant <russell@digium.com>

	* channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by
	  just ignoring it. An unknown indication will trigger an error and
	  cause sounds to stop, which in this case, is ringing.

2007-05-14 18:52 +0000 [r64280]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Handle network errors, like host or network
	  unreachable, in a better way. This means that calls to hosts or
	  qualify (OPTION) messages will fail quicker if the TCP/IP stack
	  tells us that there is an issue. Since this is an unconnected UDP
	  socket, we will not get error messages directly in most cases,
	  but maybe on the second and third try. This is already
	  implemented in trunk.

2007-05-14 18:48 +0000 [r64240-64278]  Joshua Colp <jcolp@digium.com>

	* codecs/codec_speex.c: Properly set datalen field when doing PLC
	  in codec_speex. (issue #9722 reported by mihai)

	* /, main/devicestate.c: Merged revisions 64275 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
	  lines Only perform stripping of - strings from the channel name
	  for Zap channels. Anywhere else we might remove a legitimate part
	  of a device name. (issue #9668 reported by stevedavies) ........

	* main/channel.c: Fix scenario where if a phone that simply called
	  Echo() put itself on hold it could never get off hold.

2007-05-14 13:58 +0000 [r64193]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570,
	  worrisome CDR warnings have been removed, that are either not
	  helpful, or not relevant.

2007-05-14 10:39 +0000 [r64157]  Olle Johansson <oej@edvina.net>

	* main/channel.c: Add hangupcause when we lack codecs for
	  transcoding

2007-05-12 22:27 +0000 [r64044-64114]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: This concludes my final adventure with
	  bitmasks and the onhold flag. Would anyone care for some peanuts?

	* channels/chan_sip.c: Tweak hold flags some more. They can be of
	  three states when active: active, inactive, one direction.

	* channels/chan_sip.c: Ensure the onhold flag is set no matter what
	  when being put on hold.

2007-05-11 20:16 +0000 [r63982]  Jason Parker <jparker@digium.com>

	* main/manager.c: Hide manager password from "manager show user
	  foo". I realize that there are other ways to get this, but we
	  really don't need to just show it in plain text so easily. Issue
	  9273, patch by junky

2007-05-11 16:35 +0000 [r63905]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* contrib/scripts/safe_asterisk, Makefile, /: Merged revisions
	  63903 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007)
	  | 2 lines Issue 9121 - fixups for safe_asterisk script ........

2007-05-11 16:05 +0000 [r63886]  Russell Bryant <russell@digium.com>

	* main/manager.c: When MD5 authentication is not possible because
	  there is no challenge present, either because the Challenge
	  action was never issued, or some other reason, give a proper
	  error message and return an error instead of claiming that the
	  user wasn't found. (reported by jsmith on IRC)

2007-05-11 15:43 +0000 [r63872]  Joshua Colp <jcolp@digium.com>

	* res/res_features.c: Make the PARKINGEXTEN feature of parking
	  actually work. (issue #9708 reported by mdu113)

2007-05-10 23:15 +0000 [r63830]  Jason Parker <jparker@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
	  lines Fix an issue with trying to kill a thread before it gets
	  created. Issue 9709, patch by nic_bellamy. ........

2007-05-10 22:23 +0000 [r63804]  Russell Bryant <russell@digium.com>

	* main/manager.c: Strip terminal escape sequences from CLI command
	  output that is going to be sent out over the manager interface.
	  (issue #9659, reported by pari, fixed by me)

2007-05-10 20:48 +0000 [r63750]  Doug Bailey <dbailey@digium.com>

	* main/callerid.c: Add test for negative offsets in cid data to
	  prevent infinite loops.

2007-05-10 20:46 +0000 [r63749]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
	  lines Do not allocate SIP pvt's for PEERs we can not reach. This
	  was seen as a lot of dialogs being created then immediately
	  destroyed at reload/restart of the SIP channel. ........

2007-05-09 19:22 +0000 [r63656-63698]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: Use the DTMF frame on the channel when returning
	  a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.

	* channels/chan_sip.c: Do not prematurely go on hold if sendonly
	  was not actually set.

2007-05-09 17:25 +0000 [r63654]  Matthew Fredrickson <creslin@digium.com>

	* channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
	  lines Make sure we only create a DSP if it's requested on
	  SUB_REAL ........

2007-05-09 16:55 +0000 [r63612]  Russell Bryant <russell@digium.com>

	* main/channel.c: Modify ast_senddigit_begin() to use the same
	  assumptions used elsewhere in the code in that if a channel does
	  not have a send_digit_begin() callback, it only cares about DTMF
	  END events. (pointed out by Michael Neuhauser on the asterisk-dev
	  list)

2007-05-09 16:54 +0000 [r63611]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
	  lines Properly handle hints that point to multiple devices in
	  chan_sip. Why chan_sip is even doing this I have no idea but I
	  would rather not go into a rant. (issue #9536 reported by
	  rlister) ........

2007-05-09 16:43 +0000 [r63608]  Russell Bryant <russell@digium.com>

	* main/channel.c: Only call ast_senddigit_begin() in
	  ast_senddigit() if the channel has a send_digit_begin() callback.
	  Checking the END_DTMF_ONLY flag was the wrong thing to do,
	  because that flag indicates that a *bridged* channel only wants
	  DTMF END events coming from this channel.

2007-05-09 14:50 +0000 [r63566]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_directory.c: Merged revisions 63565 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007)
	  | 2 lines Replicate fix from 51158 (app_voicemail) to
	  app_directory (Issue 9224) ........

2007-05-09 13:24 +0000 [r63535]  Russell Bryant <russell@digium.com>

	* Makefile: I have seen multiple people post questions trying to
	  figure out what the message "The configure script must be
	  executed before running 'make'" means. So, add another like that
	  says to specifically run ./configure. If this isn't obvious
	  enough, then they should be using something like AsteriskNOW and
	  not installing from source.

2007-05-09 13:17 +0000 [r63534]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
	  channels/misdn/isdn_msg_parser.c: Merged revisions
	  62945,63402,63519 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
	  1 line when we're in state WAITING4DIGS, we use the asterisk
	  tone-generator which prods us, so we can't just return -1 in
	  misdn_write in this case. Added a MISDN_KEYPAD channel variable,
	  and fixed the sending of keypad. this enables us to modify the
	  call forward parameters in the switch. ........ r63402 | crichter
	  | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
	  application misdn_check_l2l1 which tries to pull up the L1/L2 on
	  all ports that have the layers down in a group. It waits then for
	  a timeout. This helps for scenarios where multiple PMP BRIs are
	  grouped together, or where a provider has a faulty PTP
	  Implementation, that looses the L2 after a while. ........ r63519
	  | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
	  release_chan frees ch, so we should never touch ch after
	  release_chan, this may cause segfaults. ........

2007-05-09 13:04 +0000 [r63532]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't retransmit 200 OK's on ignore status.
	  (Reported on asterisk-users)

2007-05-08 22:38 +0000 [r63478]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 63477 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007)
	  | 2 lines Issue 9602 - segfault in app_macro ........

2007-05-08 16:53 +0000 [r63403-63448]  Russell Bryant <russell@digium.com>

	* res/res_features.c: I mixed up the use of the find_feature()
	  function, so I renamed it find_dynamic_feature, and changed the
	  code to use the correct lock when using it.

	* res/res_features.c: Use a read/write lock when accessing the
	  built-in features.

	* contrib/scripts/realtime_pgsql.sql (added),
	  contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to
	  contrib/scripts to be with the rest of the sql examples. (issue
	  #9676, suretec)

2007-05-08 06:22 +0000 [r63360]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007)
	  | 2 lines Issue 9527 - upon entering a folder, no message is
	  selected (curmsg == -1), so deleting causes memory corruption
	  (beyond bounds) ........

2007-05-07 22:28 +0000 [r63329]  Russell Bryant <russell@digium.com>

	* configs/res_pgsql.conf.sample (added),
	  configs/extconfig.conf.sample, contrib/realtime_pgsql.sql
	  (added): Add a sample configuration file and example tables for
	  use with res_config_pgsql. (issue #9676, suretec)

2007-05-07 21:45 +0000 [r63283-63286]  Joshua Colp <jcolp@digium.com>

	* main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
	  revisions 63285 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
	  lines Properly handle what happens during a masquerade in
	  relation to group counting. (issue #9657 reported by ramonpeek)
	  ........

	* channels/chan_sip.c: Minor backport of revision 59083 in trunk.
	  Don't queue an unhold frame up if the call was never on hold to
	  begin with.

2007-05-07 20:05 +0000 [r63196-63254]  Olle Johansson <oej@edvina.net>

	* main/config.c: Don't remove configuration from memory just
	  because one section failed.

	* /: Guess svnmerge doesn't handle files that move around. Blocking
	  patch to ./config.c

2007-05-06 12:28 +0000 [r63152]  Olle Johansson <oej@edvina.net>

	* main/file.c: Stop the video stream when you stop playback of all
	  streams for a call

2007-05-04 20:03 +0000 [r63099]  Jason Parker <jparker@digium.com>

	* res/res_jabber.c: Fix a crash when checking version attribute in
	  an incoming XML caps element. Issue 9667, patch by phsultan.

2007-05-04 16:45 +0000 [r63047]  Pari Nannapaneni <paripurnachand@digium.com>

	* configs/manager.conf.sample: explanation for httptimeout in
	  manager.conf

2007-05-03 16:44 +0000 [r62989]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
	  lines When a peer is seeded or built tell the devicestate core to
	  update it's status. This is easier then having chan_sip load
	  before pbx_config. (issue #9658 reported by dlynes) ........

2007-05-03 16:38 +0000 [r62986]  Kevin P. Fleming <kpfleming@digium.com>

	* main/loader.c: improve loader a bit, by avoiding trying to
	  initialize embedded modules twice and avoiding trying to load
	  modules from disk when they have been loaded already during the
	  'preload' pass (reported by blitzrage on IRC, patch by me)

2007-05-03 15:23 +0000 [r62942]  Russell Bryant <russell@digium.com>

	* main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell
	  Bryant, 2007, TM, Patent Pending). This set of changes came from
	  a debugging session I had with Dwayne Hubbard. When he called
	  into his home FXO, ran the Echo application, and pressed a digit,
	  the digit would be echoed back and would never end. This is
	  fixed, along with a couple other little improvements. * When
	  chan_zap is in the middle of playing a digit to a channel, it
	  feeds back null frames, not voice frames. So, I have modified
	  ast_read to check the timing on emulated DTMF when it receives
	  null frames, in addition to where it was doing this on voice
	  frames. * Make a tweak to setting the duration on emulated DTMF
	  digits. If there was no duration specified, it set it to be the
	  minimum, instead of the default. * Instead of timing the emulated
	  digits off of the number of samples in audio frames that pass
	  through, just use time values. Now there is no code in this
	  section that assumes 8kHz audio.

2007-05-03 14:41 +0000 [r62913]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19
	  (added), pbx/ael/ael-test/ael-test18/extensions.ael,
	  pbx/ael/ael-test/ael-test19/extensions.ael (added),
	  pbx/ael/ael-test/ael-test19 (added),
	  pbx/ael/ael-test/ref.ael-test20 (added),
	  pbx/ael/ael-test/ael-test20/extensions.ael (added),
	  pbx/ael/ael-test/ael-test20 (added): updated the ael regressions
	  to match what's in trunk

2007-05-03 14:36 +0000 [r62912]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
	  revisions 61357,61770,62885 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
	  1 line some fixes for PMP Hold/Retrieve, it should work now, when
	  briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
	  (Di, 24 Apr 2007) | 1 line added lock for sending messages to
	  avoid double sending. shuffled some empty_chans after the
	  cb_event calls, this avoids that a release_complete from a quite
	  different call releases a fresh created setup by accident.
	  ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
	  Mai 2007) | 1 line fixed the problem that misdn_write did not
	  return -1 when called with 0 samples in a frame this resultet in
	  a deadlock in some circumstances, when the call ended because of
	  a busy extension. added encoding of keypad. ........

2007-05-03 13:54 +0000 [r62883]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test18 (added),
	  pbx/ael/ael-test/ref.ael-vtest13,
	  pbx/ael/ael-test/ael-test18/extensions.ael (added),
	  pbx/ael/ael-test/ael-test18 (added),
	  pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c,
	  pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7:
	  These mods fix bug 9623, where an '@' in the eswitch contents
	  causes a syntax error. I also updated the regressions.

2007-05-03 00:23 +0000 [r62797-62842]  Kevin P. Fleming <kpfleming@digium.com>

	* res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
	  May 2007) | 2 lines doh... initializing the pointer variable will
	  work just a bit better ........

	* res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
	  May 2007) | 7 lines increase reliability and efficiency of static
	  Realtime config loading via ODBC: don't request fields we aren't
	  going to use don't request sorting on fields that are pointless
	  to sort on explicitly request the fields we want, because we
	  can't expect the database to always return them in the order they
	  were created (reported by blitzrage in person (!), patch by me)
	  ........

	* res/res_config_pgsql.c: improve static Realtime config loading
	  from PostgreSQL: don't request sorting on fields that are
	  pointless to sort on use ast_build_string() instead of snprintf()
	  don't request the list of fieldnames that resulted from the query
	  when we both knew what they were before we ran the query _AND_ we
	  aren't going to do anything with them anyway (patch by me,
	  inspired by blitzrage's bug report about res_config_odbc)

2007-05-02 22:59 +0000 [r62739-62789]  Russell Bryant <russell@digium.com>

	* main/channel.c: Merge changes from team/russell/inband_dtmf ...
	  Fix some issues related to generating inband DTMF. There are two
	  changes here: 1) The list of DTMF tones in the senddigit_begin()
	  function explicitly specified 100ms of the tone followed by 100ms
	  of silence. This really broke things with the way that Asterisk
	  now wants complete control over when the digit begins and ends.
	  So, regardless of what Asterisk really wanted to do, this was
	  going to play out the tone at the length it wanted to. This
	  caused various problems like DTMF translation to inband to be
	  extremely unreliable. The list of tones has been changed so that
	  the correct DTMF tone is played indefinitely until Asterisk tells
	  it to stop. 2) ast_write() had to be modified to let a DTMF_END
	  frame get processed even when a generator is present. This is how
	  the tone will finally get stopped. (issues #8944, #9250, #9348,
	  maybe others. Thanks to mdu113 from #8944 for the testing and
	  feedback!)

	* main/manager.c: Backport the change that only went in to trunk
	  that fixes the command manager action over http. (reported
	  internally by pari and bkruse)

2007-05-02 20:46 +0000 [r62738]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May
	  2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being
	  in 'h' extension louses up the dst field ........

2007-05-02 17:43 +0000 [r62692]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007)
	  | 4 lines Issue 9638 - if a text frame is sent with no
	  terminating NULL through a bridged IAX connection, the remote end
	  will receive garbage characters tacked onto the end. ........

2007-05-02 17:10 +0000 [r62689]  Steve Murphy <murf@digium.com>

	* configs/extensions.conf.sample, main/channel.c, main/pbx.c,
	  channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the
	  clid, src fields in channel_alloc call. b)in the channel_alloc
	  func, set the cid_num and name fields from the arglist[blush]. c)
	  don't update the channel app & app data fields if you are in the
	  'h' extension. d)the load_module func in cdr_radius needs to
	  return DECLINE, SUCCESS.

2007-05-02 06:15 +0000 [r62624]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Don't unlock a channel that we already know
	  does not exist (propably isue 8228)

2007-05-01 21:57 +0000 [r62548]  Russell Bryant <russell@digium.com>

	* /, res/res_features.c: Merged revisions 62547 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) |
	  4 lines Remove an unnecessary check that makes it so if you hang
	  up after doing an attended transfer before the target extension
	  answers the channel, the transfer is not successful. (issue
	  #9338, patch by svanlund) ........

2007-05-01 21:34 +0000 [r62545]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user()
	  (found by rayjay, different fixes by me)

2007-05-01 16:26 +0000 [r62497]  Russell Bryant <russell@digium.com>

	* /, configs/indications.conf.sample: Merged revisions 62496 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) |
	  3 lines Add indications.conf information for the Philippines.
	  (issue #9525, reported and patched by loloski) ........

2007-04-30 15:58 +0000 [r62414-62419]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) |
	  4 lines This patch fixes an issue where depending on the cause
	  code, when the network sends a PRI disconnect, the call may not
	  be properly hung up. (issue #9588, reported and patched by
	  softins) ........

	* include/asterisk/http.h, main/http.c: When serving dynamic
	  content, include a Cache-Control header to instruct the browsers
	  to not store the resulting content. (issue #9621, reported by
	  Pari, patch by me)

2007-04-30 14:52 +0000 [r62371]  Jason Parker <jparker@digium.com>

	* configs/iax.conf.sample: Remove unused (and potentially
	  confusing) jitterbuffer options from sample config.

2007-04-30 14:36 +0000 [r62369]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c, /: Merged revisions 62368 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
	  lines Update copyright notice. It's now the year 2007! ........

2007-04-29 05:50 +0000 [r62299-62331]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Fix a bug that made the "language" setting
	  in zapata.conf not functional. (issue #9626, reported and fixed
	  by sergee)

	* apps/app_meetme.c: Note that the "talker optimization" option
	  will be enabled by default in 1.6

2007-04-27  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.4 released.

2007-04-27 21:10 +0000 [r62218]  Russell Bryant <russell@digium.com>

	* channels/chan_agent.c: Fix a weird problem where when a caller
	  talking to someone sitting behind an agent channel sent a digit,
	  the digit would be played to the agent for forever. This is
	  because chan_agent always returned -1 from its send_digit_begin
	  and _end callbacks. This non-zero return value indicates to the
	  Asterisk core that it would like an inband DTMF generator put on
	  the channel. However, this is the wrong thing to do. It should
	  *always* return 0, instead. When the digit begin and end
	  functions are called on the proxied channel, the underlying
	  channel will indicate whether inband DTMF is needed or not, and
	  the generator will be put on that one, and not the Agent channel.
	  (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
	  by me)

2007-04-27 16:17 +0000 [r62174]  Jason Parker <jparker@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
	  lines This transcoder message needn't be a NOTICE. I've seen it
	  cause confusion more than a few times. ........

2007-04-27 16:14 +0000 [r62171]  Russell Bryant <russell@digium.com>

	* main/pbx.c: If no variables were passed into
	  pbx_substitute_variables_helper_full(), then don't even bother
	  creating a temporary bogus channel, since that is only for
	  allowing certain functions to operate on the variables as if they
	  were on a channel. Most importantly, this fixes a crash. (issue
	  #9613, reported by callguy, fixed by me)

2007-04-27 14:04 +0000 [r62095-62137]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
	  lines Issue #7351 - SIP Cancel fails due to the wrong contact
	  uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
	  - THANKS!!!! THis was a hard one to catch. ........

	* channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
	  annoying DEBUG messages not controlled by option_debug (DEA).
	  Thanks!

2007-04-26 16:33 +0000 [r61959-62038]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
	  lines Revert previous fix for when the IAX2 channel goes funky
	  (that's the technical term). This is causing legit calls to be
	  prematurely hung up. (issue #9600 reported by justdave) ........

	* main/channel.c: Missed an ast_app_group_discard during merge.
	  Thanks blitzrage!

	* res/res_monitor.c: Don't always say that the channel is being
	  paused if it is actually being unpaused in the Manager ack
	  message. (reported by jsmith in #asterisk-bugs)

	* main/config.c, /: Merged revisions 61958 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
	  lines Don't count failed include attempts against the
	  configuration include level. (issue #9593 reported by mostyn)
	  ........

2007-04-25 22:29 +0000 [r61914]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
	  | 2 lines handle a very bizarre race condition with channels
	  being redirected before a simple switch can be started on them
	  (issue #9286) ........

2007-04-25 21:59 +0000 [r61863-61870]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
	  2 lines If the callerid= option is specified, but empty, clear
	  any previous data. ........

	* /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
	  2 lines Ensure that callerid settings are reset on a reload.
	  ........

2007-04-25 19:21 +0000 [r61805]  Joshua Colp <jcolp@digium.com>

	* main/cli.c, main/channel.c, include/asterisk/app.h,
	  funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
	  lines Merge rewritten group counting support. No more storing
	  data on the variable list of the channels. That was bad, mmmk?
	  (issue #7497 reported by sabbathbh) ........

2007-04-25 16:22 +0000 [r61799]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
	  3 lines Fix a typo where cid_num got copied instead of cid_ani.
	  (issue #9587, reported and patched by xrg) ........

2007-04-24  Russell Bryant <russell@digium.com>

	* Asterisk 1.4.3 released.

2007-04-24 21:34 +0000 [r61781-61787]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 61786 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
	  4 lines Don't crash if a manager connection provides a username
	  that exists in manager.conf but does not have a password, and
	  also requests MD5 authentication. (ASA-2007-012) ........

	* main/channel.c, include/asterisk/channel.h: Improve DTMF handling
	  in ast_read() even more in response to a discussion on the
	  asterisk-dev mailing list. I changed the enforced minimum length
	  of a digit from 100ms to 80ms. Furthermore, I made it now enforce
	  a gap of 45ms in between digits. These values are not
	  configurable in a configuration file right now, but they can be
	  easily changed near the top of main/channel.c.

2007-04-24 18:43 +0000 [r61779]  Dwayne M. Hubbard <dhubbard@digium.com>

	* channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
	  | 1 line removed #if 0 block from chan_phone, chan_zap, and
	  chan_modem restart_monitor() ........

2007-04-24 16:16 +0000 [r61774]  Russell Bryant <russell@digium.com>

	* main/dial.c: Add a few more state changes in
	  handle_frame_ownerless() so that the SLA code will get notified
	  of these changes even when an owner channel is not provided. This
	  isn't from a specific bug report, it's just something I noticed
	  while poking around.

2007-04-24 16:07 +0000 [r61772]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
	  lines Allow RFC2833 to be sent in the response SDP when an INVITE
	  comes in without SDP. (issue #9546 reported by mcrawford)
	  ........

2007-04-23 18:17 +0000 [r61763-61765]  Russell Bryant <russell@digium.com>

	* main/pbx.c: Some dialplan functions, such as CUT(), expect to
	  operate on variables on a channel. So, this little hack lets them
	  work in places where a channel doesn't exist, such as within
	  DUNDi configuration. (issue #9465, reported and patched by
	  Corydon76, testing by blitzrage)

	* main/channel.c: Ensure that digits passing through Asterisk have
	  a reasonable minimum length. It is currently 100 ms. If someone
	  thinks this should be different, feel free to speak up. (related
	  to issues #8944, #9250, and #9348)

2007-04-20 21:35 +0000 [r61705-61707]  Jason Parker <jparker@digium.com>

	* main/rtp.c: Avoid invalid seqno cycling detection. Per comment
	  from Dave Troy: This adds back in some simple typecasting I had
	  in an earlier version which I realize now may be breaking things.
	  Issue #9554.

	* main/loader.c, /: Merged revisions 61704 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
	  lines Fix an issue that I noticed while looking over issue 9571.
	  The reload timestamp was getting set after reloading the built-in
	  stuff, and before the modules. ........

2007-04-20 20:42 +0000 [r61697]  Russell Bryant <russell@digium.com>

	* main/rtp.c: Remove a stray debug message introduced by a recent
	  commit.

2007-04-20 19:51 +0000 [r61694]  Jason Parker <jparker@digium.com>

	* /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
	  lines If the '* to hangup' option is not enabled, we don't need
	  to disable * as a valid exit key. If it was enabled, this
	  statement would've never been checked in the first place. Issue
	  #9552 ........

2007-04-20 18:19 +0000 [r61690]  Russell Bryant <russell@digium.com>

	* main/config.c, apps/app_voicemail.c, main/manager.c,
	  include/asterisk/config.h: Fix the UpdateConfig manager action to
	  properly treat "variables" and "objects" differently (a=b versus
	  a=>b). (issue #9568, reported by pari, patch by me)

2007-04-19 08:37 +0000 [r61686]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
	  lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
	  Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........

2007-04-19 04:36 +0000 [r61681-61683]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/manager.c: Bug 9557 - simple reason why reading a function
	  always returned NULL

	* funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
	  funcs/func_groupcount.c, /, funcs/func_timeout.c,
	  funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
	  | 5 lines Bug 9557 - Specifying the GetVar AMI action without a
	  Channel parameter can cause Asterisk to crash. The reason this
	  needs to be fixed in the functions instead of in AMI is because
	  Channel can legitimately be NULL, such as when retrieving global
	  variables. ........

2007-04-18 22:10 +0000 [r61678]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: allow external build systems to extract the
	  required sound file versions

2007-04-18 20:46 +0000 [r61674-61676]  Olle Johansson <oej@edvina.net>

	* main/rtp.c: Clean upp formatting, add some doxygen stuff while
	  we're in cleaning mode... Thanks Kevin!

	* main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)

2007-04-16 14:47 +0000 [r61664-61666]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: #9483, half of patch by twilson to solve 302
	  redirect issues

	* /: Blocking AstHoloPatch from 1.2

2007-04-13 21:17 +0000 [r61658]  Steve Murphy <murf@digium.com>

	* main/cdr.c: This is a fix to the way CDR merge handles the data
	  that results from ForkCDR.

2007-04-13 19:17 +0000 [r61648-61656]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
	  lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
	  the same as OUTBOUND_GROUP except it will get unset after use so
	  it won't get accidentally inherited. (issue #BE-140) ........

	* apps/app_speech_utils.c: Do not bother looking for a result if
	  none are present.

	* channels/chan_sip.c: For those very verbose SIP implementations
	  that attach tons of info to the Contact header... let's increase
	  our variable sizes. (issue #9535 reported by jeffg)

2007-04-13 17:10 +0000 [r61645]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Eliminate a compiler warning with
	  ODBC_STORAGE enabled so that it will build under dev-mode.

2007-04-13 17:01 +0000 [r61644]  Steve Murphy <murf@digium.com>

	* channels/chan_oss.c: A fix for chan_oss that resulted from the
	  CDR changes; it helps to use the right info.

2007-04-13 16:32 +0000 [r61641]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't assume the callid of a dialog will be
	  set, as in some circumstances it may not. (issue #9534 reported
	  by tecnoxarxa)

2007-04-11 16:05 +0000 [r61477]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
	  5 lines If someone sets the "useragent" option in sip.conf to be
	  empty, then don't add the User-Agent header at all. It is an
	  optional header, anyway. Also, the bug report says that some of
	  Japan's SIP providers don't allow it for some weird reason.
	  (issue #9488, reported by makoto, fixed by me) ........

2007-04-11 15:39 +0000 [r61443]  Nadi Sarrar <ns@beronet.com>

	* channels/chan_misdn.c: Don't export AOCD variables on
	  misdn_hangup anymore, this was mainly a fix for trunk..

2007-04-11 15:09 +0000 [r61377-61427]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
	  6 lines Fix a bug with switching between host=dynamic and using
	  specific hosts for peers. The code would only reset the peer's
	  address when it is dynamic if it was a new peer structure. Now,
	  it will also reset the address if it was already in the peer
	  list, but before the reload, it was not dynamic. (issue #9515,
	  reported by caio1982, fixed by me) ........

	* main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
	  bkruse) In passing, constify the elements of the mimetypes table.

	* /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
	  5 lines Remove the attempt at reporting configuration errors in
	  sip.conf. This can cause a bunch of improper messages when using
	  realtime. I give up. As oej tried to convince me when I put this
	  in, there is just no easy way to do it. (inspired by a message on
	  the -dev list) ........

2007-04-11 13:40 +0000 [r61342-61373]  Nadi Sarrar <ns@beronet.com>

	* channels/chan_misdn.c: Export AOCD variables on misdn_hangup.

	* channels/chan_misdn.c: Ignore facility messages in case we don't
	  have a corresponding channel object.

	* channels/chan_misdn.c: AOCD's are now exported to asterisk
	  channel variables.

2007-04-10 16:05 +0000 [r61220]  Russell Bryant <russell@digium.com>

	* main/Makefile, main/http.c, main/minimime (removed): File upload
	  support was added to solve some needs for the Asterisk GUI.
	  However, after much discussion, it has been decided that adding
	  this to 1.4 is not in the best interests of the project. It has
	  been removed here, but will remain in trunk.

2007-04-10 12:43 +0000 [r61183]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
	  2007) | 2 lines msns config parameter defaults to '*' ........

2007-04-10 05:18 +0000 [r61136]  Steve Murphy <murf@digium.com>

	* apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
	  previous fix to overcome a compiler warning; the app NoCDR() has
	  been updated to mark the channel CDR as POST_DISABLED instead of
	  destroying the CDR; this way its flags are propagated thru a
	  bridge and the CDR is actually dropped. The cases where only one
	  channel in a bridge has a CDR was cleaned up.

2007-04-09 19:58 +0000 [r61072]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
	  lines - Don't send ActionID before Response: header. - Don't use
	  a blank in an AMI header ........

2007-04-09 19:55 +0000 [r61062-61070]  Kevin P. Fleming <kpfleming@digium.com>

	* main/minimime/mm_envelope.c, res/res_features.c: fix up some
	  warnings found using --enable-dev-mode

	* main/minimime/Doxyfile (removed),
	  main/minimime/tests/messages/CVS (removed),
	  main/minimime/tests/CVS (removed): remove some more stuff we
	  don't need

2007-04-09 19:41 +0000 [r61042-61044]  Russell Bryant <russell@digium.com>

	* main/minimime/test (removed): Remove another directory that
	  should no longer be there

	* main/minimime/Make.conf (removed), main/minimime/mytest_files
	  (removed), main/minimime/.cvsignore (removed), main/minimime/sys
	  (removed), main/minimime/mm-docs (removed): Remove various files
	  that I thought I already removed.

2007-04-09 19:05 +0000 [r61022]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Use the appropriate interface name with
	  COMPLETECALLER. Issue 9395.

2007-04-09 18:32 +0000 [r60989]  Steve Murphy <murf@digium.com>

	* channels/chan_oss.c, main/channel.c, main/cdr.c,
	  channels/chan_phone.c, channels/chan_misdn.c,
	  channels/chan_skinny.c, channels/chan_features.c,
	  channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
	  channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
	  channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
	  include/asterisk/channel.h, channels/chan_gtalk.c,
	  channels/chan_iax2.c: This is a big improvement over the current
	  CDR fixes. It may still need refinement, but this won't have as
	  many folks bothered.

2007-04-09 18:02 +0000 [r60984]  Olle Johansson <oej@edvina.net>

	* res/res_jabber.c: Add final new line after JabberEvent

2007-04-09 17:22 +0000 [r60936]  Jason Parker <jparker@digium.com>

	* /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
	  lines Allow matching on names shorter than 3 chars. This also
	  fixes the case where somebody wants to match on less then 3
	  chars. Issue 9071 ........

2007-04-09 03:01 +0000 [r60847-60850]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
	  | 2 lines Don't check for error when lowering priority (according
	  to the manpage, it should never happen anyway). It might could
	  happen, though, if another thread messed with the priority, so
	  safeguard against that (reported via -dev list). ........

	* channels/chan_local.c, /: Merged revisions 60846 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
	  Apr 2007) | 2 lines Bug 9505 - If the return value for
	  local_queue_frame is set, then p->lock is no longer valid.
	  ........

2007-04-09 01:03 +0000 [r60762-60798]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
	  lines When calling a device that then forwards us elsewhere... we
	  have to make our channels compatible if it is the only channel
	  being dialed. (issue #9445 reported by marcelbarbulescu) ........

	* apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
	  MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)

2007-04-08 14:14 +0000 [r60661-60713]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
	  | 2 lines Gosub called within a Macro resets the arguments
	  improperly and causes general weirdness. (Issue 8329) ........

	* main/http.c: Fix --enable-dev-mode

	* channels/chan_oss.c: Off by one error, resulting in a crash
	  (Issue 9500)

	* /, main/file.c: Merged revisions 60660 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
	  | 2 lines Bug 9486 - memory leak when opening a filestream
	  ........

2007-04-06 20:58 +0000 [r60603]  Russell Bryant <russell@digium.com>

	* main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
	  main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
	  main/manager.c, main/minimime/mm_mimepart.c,
	  main/minimime/test.sh, configure, include/asterisk/compat.h,
	  main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
	  main/minimime/tests/parse.c, main/minimime/mm_base64.c,
	  main/minimime/mm_mimeutil.c, main/minimime/mm.h,
	  main/minimime/tests, main/minimime/mm_header.c,
	  main/minimime/mm_error.c, main/Makefile,
	  main/minimime/mm_codecs.c, main/minimime/mm_param.c,
	  configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
	  include/asterisk/manager.h, main/minimime/strlcpy.c,
	  configs/http.conf.sample, main/minimime/mm_parse.c,
	  main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
	  main/minimime/mm_util.c, main/minimime/mm_envelope.c,
	  main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
	  main/minimime/tests/messages/test2.txt,
	  main/minimime/tests/messages/test3.txt,
	  main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
	  main/minimime/tests/messages/test4.txt,
	  main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
	  main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
	  main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
	  main/minimime/mimeparser.l, main/minimime/mm_context.c,
	  main/minimime/mimeparser.tab.h, main/minimime (added),
	  main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
	  main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
	  main/minimime/mimeparser.y, Makefile.moddir_rules,
	  main/minimime/sys, main/minimime/tests/Makefile: To be able to
	  achieve the things that we would like to achieve with the
	  Asterisk GUI project, we need a fully functional HTTP interface
	  with access to the Asterisk manager interface. One of the things
	  that was intended to be a part of this system, but was never
	  actually implemented, was the ability for the GUI to be able to
	  upload files to Asterisk. So, this commit adds this in the most
	  minimally invasive way that we could come up with. A lot of work
	  on minimime was done by Steve Murphy. He fixed a lot of bugs in
	  the parser, and updated it to be thread-safe. The ability to
	  check permissions of active manager sessions was added by Dwayne
	  Hubbard. Then, hacking this all together and do doing the
	  modifications necessary to the HTTP interface was done by me.

2007-04-06 20:32 +0000 [r60568-60572]  Dwayne M. Hubbard <dhubbard@digium.com>

	* UPGRADE.txt: clarified a sentence in the format_wav section

	* UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
	  plan to remove GAIN code from trunk

2007-04-06 19:50 +0000 [r60521-60565]  Russell Bryant <russell@digium.com>

	* apps/app_meetme.c: When a station picks up a trunk that was on
	  hold, make the hints reflect that nobody has the trunk on hold
	  anymore.

	* apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
	  reported by francesco_r, fixed by me) * The original behavior was
	  that if one station put a call on hold, another one picked it up,
	  and then hung up, the code would still consider the call on hold
	  by the first station, so the trunk would not be hung up. However,
	  to better comply with what most people seem to expect it to
	  behave, it will now hang up the trunk. * Fix a problem with
	  "barge=no". This was only intended to prevent people from joining
	  calls that are in progress. However, it also prevented other
	  people from picking up a call that was on hold. This has been
	  fixed. * When there are no active stations on a trunk and it is
	  on hold, the code now indicates the HOLD and UNHOLD conditions to
	  the trunk channel. This allows music on hold to be played to the
	  trunk when it is on hold.

2007-04-06 18:21 +0000 [r60459-60485]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we check the faxdetect option
	  before doing fax processing

	* channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
	  lines There should only be one code path for doing DTMF
	  conditionals on channels. This fixes it. ........

2007-04-06 14:49 +0000 [r60399]  Kevin P. Fleming <kpfleming@digium.com>

	* /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
	  | 2 lines remove undocumented 'cardsmode' parameter and stop
	  searching for transcoders during reload() ........

2007-04-06 01:14 +0000 [r60361]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c, apps/app_speech_utils.c,
	  include/asterisk/speech.h: Add support for returning different
	  types of results (ie: NBest).

2007-04-05 22:58 +0000 [r60325]  Dwayne M. Hubbard <dhubbard@digium.com>

	* formats/format_wav.c: modified default GAIN for issue 5823,
	  thanks jrwalliker

2007-04-05 22:35 +0000 [r60323]  Steve Murphy <murf@digium.com>

	* configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
	  some clarification to the example configs for CDRs, on how to
	  select a backend. Also, made cdr-csv the default if you 'make
	  samples', and no other changes.

2007-04-05 16:10 +0000 [r60268]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
	  lines Just because we can't find the voicemail configuration
	  file, doesn't mean that the module failed to load. The user could
	  be using realtime. Issue #9473 ........

2007-04-05 15:47 +0000 [r60265]  Russell Bryant <russell@digium.com>

	* main/http.c: Add the MIME type for gif by request from Pari

2007-04-05 12:55 +0000 [r60214]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
	  lines Only unlock our pvt and net locks if we are actually going
	  to try to lock the owner again. (issue #9472 reported by zoa)
	  ........

2007-04-04 17:40 +0000 [r60013-60137]  Russell Bryant <russell@digium.com>

	* main/manager.c, /: Merged revisions 60134 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
	  6 lines It is valid to redirect channels via the manager
	  interface that are not in the UP state. Instead of checking for
	  that to prevent to ensure a dead channel doesn't get redirected,
	  just use the ast_check_hangup() API call. (issue #9457, reported
	  by Callmewind, patch by me) (related to issue #8977) ........

	* channels/chan_sip.c: Add a Content-Length of 0 to the response
	  built by transmit_response_with_unsupported(). (issue #9454,
	  reported by makoto, fixed by me)

	* /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
	  4 lines Fix the return value of handle_common_options() so that
	  it always properly indicates whether it handled the option or
	  not. (issue #9455, reported by Netview, fixed by me) ........

	* apps/app_meetme.c: Fix a problem where if a trunk was hung up
	  while it was on hold, all of the hints would reflect the line
	  still on hold, even though it should reflect that it is back to
	  not in use. (issue #9459, reported by francesco_r, fixed by me)

	* /: Blocked revisions 60016 via svnmerge ........ r60016 | russell
	  | 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a
	  missing "\r\n" in the body of the NOTIFY that is sent to indicate
	  the status of a transfer. (issue #9388, reported by rarritt)
	  ........

	* /: Blocked revisions 60014 via svnmerge ........ r60014 | russell
	  | 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the
	  more generic check for "sed -r" support that was already present
	  in 1.4. (related to issue #9399) ........

	* /: Blocked revisions 60012 via svnmerge ........ r60012 | russell
	  | 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On
	  Darwin, the -r argument to sed is not valid. It has to be -E.
	  (issue #9399, reported by jcovert) ........

2007-04-03 19:40 +0000 [r59963]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Don't clash when a person both speaks
	  and uses DTMF.

2007-04-03 19:16 +0000 [r59853-59939]  Russell Bryant <russell@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
	  4 lines Don't attempt to report configuration errors in
	  build_user(). oej pointed out that for a "friend" entry, this
	  won't work, because all user options are valid for peers, but not
	  the other way around. ........

	* /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
	  3 lines Make chan_sip report when it encounters an unknown
	  option. (issue #9440, reported by nightcrawler) ........

	* /, main/app.c: Merged revisions 59886 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
	  5 lines When doing a built-in blind or attended transfer, restore
	  the ability to use '#' to terminate the number and immediately do
	  the transfer instead of having to dial the number and just wait
	  for the feature digit timeout. (issue #8366, xueliangliang)
	  ........

	* Makefile: Ensure that menuselect gets executed in dependency
	  check mode every time you run make.

2007-04-03 11:02 +0000 [r59804]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
	  Merged revisions 59788,59803 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
	  lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
	  or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
	  03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........

2007-04-03 07:20 +0000 [r59774]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
	  Merged revisions 59623-59624,59639 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
	  1 line we can now make 30 channels on a PRI (before we forgot
	  chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
	  (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
	  r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
	  1 line added option which allows us to accept incoming SETUP
	  Messages without automatically sending Proceeding or Setup
	  Acknowledge, this is useful with some broken switches and if you
	  want to Release incoming calls without previously having
	  acknowledged them. The new option is
	  noautorespond_on_setup=yes|no default is no, so we don't break
	  the existing behaviour ........

2007-04-02 18:58 +0000 [r59724]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
	  lines Increase the maximum size for a string of mailboxes to
	  1024. (issue #9270 reported by rtucker) ........

2007-04-02 17:31 +0000 [r59688]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
	  not the test. As per 9435, thanks to marcelbarbulescu

2007-04-02 15:39 +0000 [r59654]  Russell Bryant <russell@digium.com>

	* main/netsock.c, /: Merged revisions 59608 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
	  6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
	  This is needed by the patch that went in for issue 7874.
	  chan_iax2 needs to be able to create socket that is lisetning on
	  INADDR_ANY, but also be able to bind sockets to specific
	  addresses. (Thanks to Stevenson on the asterisk-dev mailing list
	  for explaining why this flag was needed.) ........

2007-03-30 22:50 +0000 [r59573]  Jason Parker <jparker@digium.com>

	* configure, main/Makefile, acinclude.m4: Add linux-uclibc host
	  arch..."thingy". Sorry, I don't know what it's called...

2007-03-30 17:51 +0000 [r59452-59522]  Steve Murphy <murf@digium.com>

	* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
	  include/asterisk/cdr.h: several changes via kpflemings review

	* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
	  include/asterisk/cdr.h: These mods fix CDR issues from 8221,
	  8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
	  from transfer situations.

	* configs/extensions.conf.sample: A small clarification to keep
	  bugs from being filed, and confusion from rising, if
	  clearglobalvars is set, and globals are set in the AEL file.
	  (9419)

2007-03-29 17:43 +0000 [r59363]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: When building a response to a subscription, the
	  "from" must be the full Jabber ID. This fixes some problems where
	  jabber users are not able to add their Asterisk account to their
	  user list, since they are unable to get Asterisk to approve their
	  subscription. (issue #8210, reported by caspy, and verified by
	  bradtem)

2007-03-29 17:38 +0000 [r59361]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
	  lines Keep a global array of variables indicating whether certain
	  conference rooms are in use. This ensures that two people going
	  into a new dynamic conference when the 'e' option is set don't go
	  into the same conference room. (issue #8835 reported by eliel)
	  ........

2007-03-29 17:17 +0000 [r59304-59358]  Russell Bryant <russell@digium.com>

	* main/rtp.c, /: Merged revisions 59357 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
	  5 lines If an error occurs when reading from an RTP socket, and
	  the error code does not indicate that we should try again, then
	  return NULL instead of a "null frame". This will prevent Asterisk
	  from trying over and over again, and eventually causing the
	  system to crash. (issue #8285, john) ........

	* /: Blocked revisions 59355 via svnmerge ........ r59355 | russell
	  | 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport
	  the change to chan_iax2 to return NULL instead of a "null frame"
	  from its read callback. See revision 59341 to the 1.4 branch for
	  more info. ........

	* channels/chan_iax2.c: When the IAX2 read callback gets called,
	  return NULL instead of a "null frame". This will cause Asterisk
	  to hangup the call instead of keep trying whatever it was doing.
	  Under normal conditions, this function would *never* be called.
	  However, the author of this patch says an error will occur that
	  will cause it to get called every 100 thousand calls or so. When
	  this does happen, it puts the channel in a loop that eventually
	  brings down the system. So, hangup up the call is certainly a
	  better alternative. (issue #8286, john)

	* Makefile: Export the GTK2 library and include information to sub
	  Makefiles.

2007-03-29 16:07 +0000 [r59300-59302]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
	  | 3 lines Issue 9415 - No point to getting a diagnostic field if
	  we aren't doing anything with the information. (Plus, it tends to
	  crash the Postgres ODBC driver.) ........

	* /: Blocked revisions 59299 via svnmerge ........ r59299 |
	  tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines
	  Change ENV section to use setenv, instead of putenv (Alexandru
	  Pirvulescu <sigxcpu@gmail.com>, reported via -dev list) ........

2007-03-28 03:38 +0000 [r59281-59289]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c: Another crash that I thought we had fixed already
	  - Issue 9396

	* apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
	  | 2 lines Oops ........

	* apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
	  | 2 lines Fix a few remaining bad mmap(2) return values ........

2007-03-27 23:20 +0000 [r59262-59278]  Russell Bryant <russell@digium.com>

	* /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
	  3 lines Fix the check of the return value from mmap(). Thanks to
	  Corydon for catching this one. ........

	* apps/app_directory.c: Fix app_directory to actually compile with
	  ODBC_STORAGE, and update the code to the latest res_odbc API.

	* apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
	  The Makefile did not properly ensure that this information got
	  copied from what was selected for app_voicemail. (issue #9224)

	* channels/chan_sip.c: Fix the check that ensures that the CHANNEL
	  function's first argument is "rtpqos". Thanks, Corydon. :)

2007-03-27 18:16 +0000 [r59261]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
	  asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
	  this case the proper thing to do. This change now has it doing
	  the proper thing.

2007-03-27 18:05 +0000 [r59256-59259]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
	  4 lines Fix the use of the "sourceaddress" option when "bindaddr"
	  is set to 0.0.0.0 instead of having each interface explicitly
	  listed. (issue #7874, patch by stevens) ........

	* channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
	  function to just be additional parameter of the CHANNEL function.
	  This way, it will be possible for other RTP based channel drivers
	  to expose this information in the future.

2007-03-27 15:00 +0000 [r59254]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
	  Mär 2007) | 1 line fixed #9355 ........

2007-03-26 21:45 +0000 [r59230]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Oops, this should be case insensitive

2007-03-26 21:41 +0000 [r59228]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
	  asterisk). I turned a duplicate context from a WARNING to an
	  ERROR. Now you get a module load failure, and asterisk just
	  exits. That's better than a crash, right\?

2007-03-26 21:37 +0000 [r59227]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_sip.c: Change this to a single dp function to make
	  oej happy.

2007-03-26 20:06 +0000 [r59225]  Steve Murphy <murf@digium.com>

	* main/config.c: Fix for 9257; by eliminating the globals in
	  main/config.c, we make it thread-safe, which is a minimum
	  requirement.

2007-03-26 19:34 +0000 [r59223]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Add ability to specify no timeout. This
	  means as soon as the prompt is done playing it moves on to the
	  next priority.

2007-03-26 18:33 +0000 [r59215-59217]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Somehow the code for building the email for
	  voicemail got out of sync. This change makes a few tweaks to get
	  1.4 in sync with trunk. (issue #9301)

	* apps/app_meetme.c: Fix some codec negotiation problems when
	  CallerID support is not enabled in SLA. (issue #9308, reported by
	  twilson)

2007-03-26 18:13 +0000 [r59213]  Joshua Colp <jcolp@digium.com>

	* apps/app_speech_utils.c: Make SpeechBackground obey the digit
	  timeout value.

2007-03-26 17:53 +0000 [r59207-59209]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Rename the new dialplan functions to match
	  the variable name

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
	  AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
	  some because they get set in sip_hangup. So, there are common
	  situations where the variables will not be available in the
	  dialplan at all. So, this patch provides an alternate method for
	  getting to this information by introducing AUDIORTPQOS and
	  VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
	  with some testing by blitzrage)

2007-03-26 17:38 +0000 [r59206]  Steve Murphy <murf@digium.com>

	* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
	  pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
	  and STANDALONE_AEL

2007-03-26 15:25 +0000 [r59202]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
	  include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
	  channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
	  provides a dsp pipeline for i.e. echo cancellation modules, make
	  chan_misdn use it. * add a check for linux/mISDNdsp.h to
	  configure.ac and update the autogenerated files: 'configure',
	  'autoconfig.h.in' (the 'configure' script was not in sync with
	  the latest configure.ac, so the diff is a bit bigger than
	  expected).

2007-03-26 15:16 +0000 [r59200]  Joshua Colp <jcolp@digium.com>

	* pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
	  aelparse binary! DONT_OPTIMIZE should now work once again.

2007-03-24 01:39 +0000 [r59195]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
	  lines Only try to handle a response if it has a response code.
	  (ASA-2007-011) ........

2007-03-23 16:11 +0000 [r59188-59189]  Steve Murphy <murf@digium.com>

	* /: blocking out the fix in 59187... already incorporated here

	* /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
	  line Added a few words in the Macro doc strings about the
	  behavior of macros with hangups (et al.), as per 9337 ........

2007-03-22 23:40 +0000 [r59180-59182]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't allow string input to overrun the
	  buffer to hold it (ASA-2007-010)

	* channels/chan_misdn.c: remove variables that are no longer used
	  (--enable-dev-mode is good, developers should be using it)

2007-03-22 14:40 +0000 [r59145]  Steve Murphy <murf@digium.com>

	* utils/Makefile: The stuff in utils was compiling with -O6 even if
	  DONT_OPTIMIZE is set in menuconfig. Added the include to fix that

2007-03-21 18:08 +0000 [r59081-59089]  Joshua Colp <jcolp@digium.com>

	* main/http.c: Add svg mimetype for pari.

	* res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
	  lines Indicate the filename changed when it is changed. (issue
	  #9311 reported by jsmith) ........

	* channels/chan_sip.c: Until we can do media level parsing for
	  sendrecv/etc just use the first value found. This crept up when a
	  phone was offered audio+video and returned an inactive video
	  stream. chan_sip thought the phone said to put the person on hold
	  but that was totally wrong. (issue #9319 reported by benbrown)

2007-03-20 21:04 +0000 [r59078]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/logger.c: Fix defines for inline stack backtraces (only used
	  by developers anyway)

2007-03-20 20:42 +0000 [r59076]  Joshua Colp <jcolp@digium.com>

	* channels/iax2-parser.c: Copy len variable as well, should fix
	  remaining IAX2 DTMF issues.

2007-03-20 17:48 +0000 [r59069-59070]  Steve Murphy <murf@digium.com>

	* apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
	  return it to its previous, untouched, state.

	* apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
	  The fix for the AEL <<security hole>> (bug 9316) is here...

2007-03-20 13:16 +0000 [r59064]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h: Merged revisions
	  58849-58850,59062-59063 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
	  1 line added method standard_dec for dialing out on groups, to
	  avoid conflicts, which caused issues with some ISDN providers
	  ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
	  Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
	  crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
	  avoid sending a disconnect when we already received one. ........
	  r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
	  1 line modified a loglevel ........

2007-03-19  Jason Parker  <jparker@digium.com>

	* Asterisk 1.4.2 released.

2007-03-19 22:29 +0000 [r59049]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: Oops, this should have been a %d all along

2007-03-19 15:52 +0000 [r59042]  Joshua Colp <jcolp@digium.com>

	* funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
	  reported by ajohnson)

2007-03-19 15:42 +0000 [r59040]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
	  via -dev list)

2007-03-18 20:37 +0000 [r59037]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
	  code 0 (reported by qwerty1979)

2007-03-18 16:36 +0000 [r59035]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_followme.c: Don't return a non-zero return code if the
	  profile doesn't exist, to match what the documentation says it
	  already does. (#9307 Reported by kkiely)

2007-03-16 16:12 +0000 [r58992]  Joshua Colp <jcolp@digium.com>

	* apps/app_page.c: Wait for the async thread to exit when hanging
	  up all of the paged phones under all circumstances. (issue #9181
	  reported by PhilSmith)

2007-03-16 01:42 +0000 [r58947-58957]  Russell Bryant <russell@digium.com>

	* configs/sla.conf.sample: fix a couple SLA documentation
	  references

	* doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
	  (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
	  doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
	  doc/channelvariables.txt (added), doc/ael.txt (added),
	  doc/billing.tex (removed), build_tools/prep_tarball,
	  doc/callingpres.txt (added), doc/enum.txt (added),
	  doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
	  doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
	  doc/security.txt (added), doc/imapstorage.txt (added),
	  doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
	  doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
	  doc/iax.txt (added), doc/ael.tex (removed),
	  doc/channelvariables.tex (removed), doc/enum.tex (removed),
	  doc/security.tex (removed), doc/math.txt (added), Makefile,
	  doc/imapstorage.tex (removed), doc/privacy.tex (removed),
	  doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
	  (added), apps/app_voicemail.c, doc/cliprompt.txt (added),
	  doc/chaniax.txt (added), doc/app-sms.txt (added),
	  doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
	  doc/ices.txt (added), doc/dundi.tex (removed),
	  doc/linkedlists.txt (added), doc/queuelog.txt (added),
	  doc/extconfig.txt (added), doc/radius.txt (added),
	  doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
	  doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
	  (removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
	  doc/queuelog.tex (removed), doc/configuration.txt (added),
	  doc/asterisk-conf.txt (added), doc/sla.pdf (added),
	  doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
	  (added), doc/mp3.tex (removed), doc/configuration.tex (removed),
	  doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
	  doc/channels.txt (added), doc/ip-tos.tex (removed),
	  doc/extensions.txt (added), doc/queues-with-callback-members.txt
	  (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
	  doc/misdn.txt (added), doc/manager.txt (added),
	  doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
	  doc/billing.txt (added), doc/localchannel.txt (added),
	  doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
	  (added), doc/00README.1st (added): Making these documentation
	  changes in the 1.4 branch upset various people, so these chanes
	  will only be done in the trunk.

	* build_tools/prep_tarball: Add the --pdf option to the usage of
	  rubber in prep_tarball

	* Makefile, build_tools/menuselect-deps.in, configure,
	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
	  configure script checking for GTK2 and some additional Makefile
	  targets to support gmenuselect

2007-03-15 23:52 +0000 [r58946]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
	  common syntax and update the resulting appdocs TeX file

2007-03-15 23:24 +0000 [r58941]  Russell Bryant <russell@digium.com>

	* doc/asterisk.tex: add a link to the rubber homepage

2007-03-15 23:11 +0000 [r58939]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_setcdruserfield.c, main/pbx.c,
	  apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
	  Expand deprecation warnings from simply warning on use to the
	  builtin documentation.

2007-03-15 22:51 +0000 [r58935-58937]  Russell Bryant <russell@digium.com>

	* doc/asterisk.tex, Makefile: Add Asterisk version information to
	  the generated PDF

	* build_tools/prep_tarball: have prep_tarball attempt to build
	  asterisk.pdf

2007-03-15 22:32 +0000 [r58933]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_realtime.c: Function works fine, but the documentation
	  is backwards.

2007-03-15 22:25 +0000 [r58931]  Russell Bryant <russell@digium.com>

	* doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
	  (added), doc/freetds.txt (removed), doc/odbcst