path: root/ChangeLog

2006-12-12  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta4 released.

2006-12-12 04:13 +0000 [r48401]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
	  is the way it should have been done.

2006-12-11 23:02 +0000 [r48396-48399]  Matt O'Gorman <mogorman@digium.com>

	* sounds/Makefile: new sounds package with 100% more silence

	* /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
	  from https://svn.digium.com/svn/asterisk/branches/1.2 ........
	  r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
	  | 4 lines app_externalivr needs a real silence file, and
	  additional changes to add silence files into core instead of
	  extra patch provided by bug 8177 with minor additions. ........

2006-12-11 21:31 +0000 [r48391]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Return non-existant callerid handling to
	  that which it was before. In 1.4 and trunk callerid can be
	  allocated but not have any contents so we have to use
	  ast_strlen_zero before passing it to the relevant functions.
	  (issue #8567 reported by pabelanger)

2006-12-11 05:37 +0000 [r48382]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_strings.c: STRFTIME() does not actually require an
	  argument (issue 8540)

2006-12-11 05:36 +0000 [r48377-48381]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Merge in my latest RTP changes. Break out RTP and
	  RTCP callback functions so they no longer share a common one.

	* apps/app_meetme.c: Use the correct API call to say a device state
	  changed. (Yes, I'm a nub.)

	* apps/app_meetme.c: Don't access the conference structure after it
	  has been freed.

2006-12-11 00:47 +0000 [r48375]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
	  res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
	  apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
	  via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
	  | 5 lines When doing a fork() and exec(), two problems existed
	  (Issue 8086): 1) Ignored signals stayed ignored after the exec().
	  2) Signals could possibly fire between the fork() and exec(),
	  causing Asterisk signal handlers within the child to execute,
	  which caused nasty race conditions. ........

2006-12-10 03:04 +0000 [r48372]  Steve Murphy <murf@digium.com>

	* channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
	  line This version applies the patch suggested by stevens in bug
	  7836 (make inbound channel RINGING state consistent with other
	  channels). ........

2006-12-09 15:59 +0000 [r48362-48363]  Russell Bryant <russell@digium.com>

	* channels/chan_iax2.c: Use locking when accessing the
	  registrations list. This list is not actually used very often, so
	  the likelihood of there being a problem is pretty small, but
	  still possible. For example, if the CLI command to list the
	  registrations was called at the same time that a reload was
	  occurring and the registrations list was getting destroyed and
	  rebuilt, a crash could occur. In passing, go ahead and convert
	  this list to use the linked list macros.

	* /: Blocked revisions 48361 via svnmerge ........ r48361 | russell
	  | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use
	  locking when accessing the registrations list. This list is not
	  actually used very often, so the likelihood of there being a
	  problem is pretty small, but still possible. For example, if the
	  CLI command to list the registrations was called at the same time
	  that a reload was occurring and the registrations list was
	  getting destroyed and rebuilt, a crash could occur. ........

2006-12-07 18:17 +0000 [r48357]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
	  Dec 2006) | 3 lines Ensure that the file position is not
	  incremented beyond the total number of files available for
	  playback. (issue #8539, ulogic) ........

2006-12-07 15:33 +0000 [r48349]  Steve Murphy <murf@digium.com>

	* main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
	  killed bug 8423 -- OriginateSuccess and OriginateError incomplete
	  channel name. May it rest in peace.

2006-12-06 16:25 +0000 [r48326]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
	  retransmitted to Asterisk

2006-12-06 16:15 +0000 [r48323]  Russell Bryant <russell@digium.com>

	* configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
	  Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
	  in the sample configuration file. (issue #8526, arkadia) ........

2006-12-06 12:27 +0000 [r48316-48317]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't send Contact on MESSAGE

2006-12-05 20:42 +0000 [r48279]  Jason Parker <jparker@digium.com>

	* configure.ac: Fix curl version number testing to be much more
	  friendly to non-bash shells. Issue 8508, patch by me. This
	  *SHOULD* be POSIX compliant now..

2006-12-05 17:29 +0000 [r48264-48270]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Merging the invitestate-1.4 branch after
	  successful testing. Will check if I can solve this with less
	  changes in 1.2.

	* configs/sip.conf.sample: Add missing s from another repository.
	  (thanks jcmoore!)

	* configs/sip.conf.sample: Updating sip.conf.sample with
	  information about T38 not working when chan_local or chan_agent
	  is involved in the call. I don't know how big a fix that would be
	  to solve, but this is the current state of affairs. (Chan_sip
	  currently checks if the other side of the bridge has a SIP tech.
	  We could/should implement another check, possibly for udptl_write
	  or some flag in the ast_channel structure).

2006-12-05 01:41 +0000 [r48252-48254]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Oops, forgot to release the odbc handle

	* apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
	  | 6 lines If the recording in the database is too large, it will
	  fail to retrieve with an mmap error. Not too sure why this
	  doesn't happen when we put it in the database, also, but since
	  that doesn't seem to be broken, I'm not going to fix it (at least
	  until someone reports it). Solution is to ask for the file in
	  smaller chunks. (Bug 8385) ........

2006-12-04 21:48 +0000 [r48237-48248]  Jason Parker <jparker@digium.com>

	* apps/app_voicemail.c: Fix an issue which didn't allow
	  unavail/greet/busy/etc messages from being saved into ODBC (and
	  probably IMAP).

	* /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell |
	  2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert
	  change from 8016 - this breaks other stuff... Needs further
	  review. Tip: When you've reported a bug about something and
	  somebody has put up a patch for it.. It's not a good idea to open
	  a completely new bug and say that something is broken because of
	  the patch in the other bug - PLEASE mention something in the bug
	  where the patch was actually created. ........

	* /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell |
	  2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an
	  issue where a message isn't saved correctly when using ODBC
	  storage and reviewing a message. Issue 8016 - patch by sokhapkin.
	  ........

2006-12-04 18:16 +0000 [r48234]  Joshua Colp <jcolp@digium.com>

	* /: Blocked revisions 48233 via svnmerge ........ r48233 | file |
	  2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the
	  generic bridge tells us not to retry, and we have a frame to spit
	  out then break the bridge. Props to markit in #asterisk-bugs for
	  bringing this up. ........

2006-12-04 17:54 +0000 [r48228-48230]  Jason Parker <jparker@digium.com>

	* configs/voicemail.conf.sample: Add documentation to
	  voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
	  blitzrage.

	* doc/snmp.txt: Attempt to document some of the dependencies that
	  are needed for net-snmp Issue 8499 - initial patch by blitzrage.

2006-12-03 06:34 +0000 [r48223]  Russell Bryant <russell@digium.com>

	* sounds/Makefile: When "fetch" is in use, instead of "wget",
	  --continue is not a valid option. (issue #8451)

2006-12-02 21:45 +0000 [r48199-48219]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Removing one of two pieces of code to
	  handle 481 response on INVITE - Move handling of REFER response
	  to handle_response_refer()

	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
	  configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
	  transmission happens - Encapsulate RTP timers in the rtp
	  structure so we have one for video and one for audio The video
	  one is not used in 1.4, really. Will be used for RTP keepalives
	  when we can send something that video phones support in the RTP
	  stream. I now this is a big architectual change at this stage for
	  1.4, but decided it was needed to avoid future bug reports. -
	  Document the RTP NAT keepalive option in sip.conf.sample Issue
	  7679 in the bug tracker. Please test.

2006-12-02 03:50 +0000 [r48195]  Russell Bryant <russell@digium.com>

	* include/asterisk/utils.h: Backport the comment containing the
	  warning regarding the limitations on the usage of this function.
	  It is thread safe, but not technically reentrant.

2006-12-01 23:37 +0000 [r48193]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
	  | 2 lines if Dial() is going to send music-on-hold to the calling
	  party, it has to send PROGRESS first to ensure that the reverse
	  audio path has been setup first (BE-106) ........

2006-12-01 23:16 +0000 [r48190]  Russell Bryant <russell@digium.com>

	* Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
	  FreeBSD 6.1 does not include wget by default. However, it has
	  fetch which will work just fine for our purposes of downloading
	  the sounds packages. So, check for both wget and fetch and the
	  configure script and use what was found to download them. If
	  neither one was found, and sound packages are selected that must
	  be downloaded, the install process will print out an informative
	  error message indicating the situation. Also, fix a couple places
	  where "make" was hard coded into some output messages by
	  replacing them with the $(MAKE) variable. (issue #8451, initial
	  patch by pabelanger, with additional modifications by me)

2006-12-01 20:25 +0000 [r48184-48186]  Jason Parker <jparker@digium.com>

	* configs/extensions.conf.sample, /: Merged revisions 48183 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
	  lines Fix a small typo - issue 8848, reported by pabelanger
	  ........

2006-12-01 19:38 +0000 [r48179]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/cli.c: Double-unlock error (reported by blitzrage on IRC)

2006-12-01 17:41 +0000 [r48177]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
	  "limitonpeers" patch from trunk, to fix a lot of issues with
	  queues and SIP device states - Remove support for T.38 early
	  media, since it's impossible. (Two patches in one - extra friday
	  evening offer due to being off line from svn today... :-)

2006-11-30 21:18 +0000 [r48168]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
	  do a partial bridge for Google Talk since we need to handle STUN.
	  (issue #8448 reported by phsultan)

2006-11-30 20:51 +0000 [r48166]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue 8319 - change noncecount before
	  using it.

2006-11-30 20:28 +0000 [r48143-48162]  Joshua Colp <jcolp@digium.com>

	* /: Blocked revisions 48161 via svnmerge ........ r48161 | file |
	  2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't
	  write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel
	  driver. (issue #8390 reported by hselasky) ........

	* /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
	  lines Only print out debug message if bridged channel is not
	  NULL. (issue #8412 reported by jubilex) ........

	* /, res/res_features.c: Merged revisions 48154 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
	  lines Do not listen for DTMF on the bridge that comes into
	  existence when ParkedCall is executed. This means native bridging
	  can now occur for this. (issue #8406 reported by kebl0155)
	  ........

	* main/cdr.c, /: Merged revisions 48151 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
	  lines Print certain CDR messages out at the NOTICE level versus
	  WARNING since they can occur when used with the CDR applications
	  and are perfectly fine. (issue #8367 reported by dartvader)
	  ........

	* /: Blocked revisions 48146 via svnmerge ........ r48146 | file |
	  2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember
	  the pointer to the allocated block of memory so that we can free
	  it and not cause a memory leak. (issue #8449 reported by arkadia)
	  ........

	* /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
	  2006) | 2 lines Document 'port' for SIP peers, came up because of
	  the current mailing list thread. (issue #8450 reported by
	  blitzrage) ........

2006-11-30 14:29 +0000 [r48129-48135]  Olle Johansson <oej@edvina.net>

	* doc/manager.txt: Explain status reports and make codefreeze more
	  happy :-)

	* /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
	  GS 487 adapter without CSEQ on separate line in the REGISTER
	  request. Imported from 1.2.

2006-11-29 21:05 +0000 [r48115]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
	  mm_login. (issue #8420 reported by slimey)

2006-11-29 19:56 +0000 [r48113]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Explain the use device status system
	  implemented in SIP for subscriptions, queues and manager a bit
	  better. Like in 1.2, you will get more detailed information if
	  you set a call limit for a device. When the call limit is
	  reached, the status system will report a device as busy. For
	  queues, setting a call limit per SIP device is propably a
	  requirement. In most cases, it will work much better if you only
	  use type=peer and not type=friend. We might decide to backport
	  the new setting from trunk to apply all call limits to the peer
	  part of a friend only.

2006-11-29 16:50 +0000 [r48107]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c, /: Merged revisions 48106 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
	  lines If the frame was duplicated before writing out then we need
	  to free it. (issue #8429 reported by edguy3) ........

2006-11-29 08:03 +0000 [r48105]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.

2006-11-29 04:26 +0000 [r48101]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c: Don't crash if the mailstream was not
	  created.

2006-11-28 18:26 +0000 [r48095]  Jason Parker <jparker@digium.com>

	* Makefile: Export several more variables in top level Makefile.
	  Inspired by issue 8438.

2006-11-28 16:57 +0000 [r48054-48088]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
	  2006) | 2 lines According to the research I have done we never
	  needed to include compiler.h in the first place so let's not!
	  (issue #8430 reported by edguy3) ........

	* apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
	  lines Use the proper function to get the new message count
	  instead of always using the filesystem. (issue #8421 reported by
	  slimey) ........

2006-11-27 17:20 +0000 [r48049]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
	  Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
	  ........

2006-11-27 17:17 +0000 [r48046]  Russell Bryant <russell@digium.com>

	* main/manager.c: Remove a couple of unused variables (issue #8380,
	  casper)

2006-11-27 15:32 +0000 [r48038]  Joshua Colp <jcolp@digium.com>

	* pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
	  lines Do not reference the freed outgoing structure in the debug
	  message. (issue #8425 reported by arkadia) ........

2006-11-27 06:41 +0000 [r48031]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Change logging message

2006-11-26 00:26 +0000 [r48015-48017]  Steve Murphy <murf@digium.com>

	* funcs/func_cdr.c: might as well also document the raw values of
	  the flag vars

	* /, funcs/func_cdr.c: A little bit of func_cdr documentation
	  upgrade-- no bug# involved, although 8221 may have inspired it.

2006-11-25 09:28 +0000 [r48002]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
	  and future releases, you can disable subscription support totally
	  or per peer in sip.conf with allowsubscribe = yes | no

2006-11-24 17:17 +0000 [r47992]  Steve Murphy <murf@digium.com>

	* main/translate.c: bug 8189 posted this fix for main/translate.c
	  for PLC

2006-11-24 15:46 +0000 [r47989]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
	  channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
	  Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
	  beatufied some logs, changed some loglevels. changed the default
	  value of block_on_alarm ........

2006-11-23 11:01 +0000 [r47959]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't allocate unused variable.

2006-11-22 21:47 +0000 [r47944]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Video will never reach Packet2Packet bridging and can
	  do more harm then good.

2006-11-21 17:32 +0000 [r47897]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If we have the non standard G726-32 setting turned on
	  we want to return G726-32 to the SDP, not our AAL2 string. (issue
	  #8330 reported by voipgate)

2006-11-21 15:20 +0000 [r47892]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
	  provisional response. Let's not treat that as early media.
	  (discovered at the AVTF meeting in Paris).

2006-11-20 20:01 +0000 [r47863-47864]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Oops, merge missed release of odbc object

	* apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
	  | 2 lines Failing to trap -1 error from mmap causes segfault
	  (Issue 8385) ........

2006-11-20 19:51 +0000 [r47850-47860]  Joshua Colp <jcolp@digium.com>

	* main/frame.c, /: Merged revisions 47859 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
	  lines Don't forget to byte swap if we are exiting the smoother
	  feed early. (issue #8287 reported by arturs) ........

	* /: Blocked revisions 47855 via svnmerge ........ r47855 | file |
	  2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free
	  history items at the end of use of the temporary SIP pvt
	  structure. (issue #8383 reported by benh) ........

	* main/rtp.c: Only remove/destroy the RTCP I/O item if it exists.

	* .cleancount, apps/app_dial.c, apps/app_directed_pickup.c,
	  include/asterisk/channel.h: Use a separate variable in the
	  channel structure to store the context that the channel was
	  dialed from. (issue #8382 reported by jiddings)

2006-11-20 11:45 +0000 [r47843-47845]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Explain properly how videosupport works.
	  Committ from Asterisk Video Task Force meeting in Paris!

	* /, channels/chan_sip.c: Make sure we destroy scheduled items and
	  not use them ever again after destruction (rizzo)

2006-11-18 17:59 +0000 [r47823]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: fix bug 7450 - Parsing fails if From header
	  contains angle brackets (the bug was only in a corner case where
	  the < was right after the opening quote, and the fix is trivial).

2006-11-16 23:19 +0000 [r47781-47782]  Jason Parker <jparker@digium.com>

	* apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially
	  pointed out by mrobinson.

	* /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell |
	  2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a
	  couple of typos in applications.. Initially spotted by mrobinson.
	  ........

2006-11-16 23:00 +0000 [r47777]  Kevin P. Fleming <kpfleming@digium.com>

	* /, doc/billing.txt: update documentation regarding IAX2 transfers
	  and CDRs Merged revisions 47776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
	  | 2 lines update clearly wrong documentation regarding cdr_custom
	  ........

2006-11-16 21:11 +0000 [r47762-47764]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Compare technology using the pointers
	  instead of a straight comparison based on name. (issue #8228
	  reported by dean bath)

	* /: Blocked revisions 47761 via svnmerge ........ r47761 | file |
	  2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for
	  the header file specifically in all cases, not just the existence
	  of the directory. (issue #8358 reported by mrness) ........

2006-11-16 20:09 +0000 [r47758]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, configure.ac: check for pre-1.4 versions of Zaptel and
	  abort the configure script if found with an appropriate error
	  message

2006-11-16 19:24 +0000 [r47755]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
	  notification optional, in order to avoid a lot of extra database
	  lookups for all those realtime users out there.

2006-11-16 18:29 +0000 [r47748-47751]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47750 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
	  2006) | 2 lines Because of the way chan_local is written we
	  should be extra careful and make sure our callback functions have
	  a tech_pvt. (issue #8275 reported by mflorell) ........

	* apps/app_meetme.c: Don't unreference the SLA object if there is
	  no SLA object in the devicestate callback. (issue #8354 reported
	  by loloski)

2006-11-16 16:51 +0000 [r47733-47744]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Don't fixup if there's nothing to fixup

	* UPGRADE.txt: Warn users about change in canreinvite

	* channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
	  authenticated (according to the RFC) - Update docs on
	  canreinvite. "nonat" is the recommended setting for most users
	  with phones behind a NAT.

2006-11-15 22:31 +0000 [r47712]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c, /: Merged revisions 47711 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
	  2006) | 2 lines Make sure that the pvt structure exists before
	  trying to do fixup on Local channels. (issue #7937 reported by
	  mada123, fix by alamantia with mods by me) ........

2006-11-15 21:56 +0000 [r47709]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL

2006-11-15 21:33 +0000 [r47707]  Joshua Colp <jcolp@digium.com>

	* main/channel.c: We need to ensure timelimit stuff is included as
	  well so warnings get played. (issue #8050 reported by KNK)

2006-11-15 20:50 +0000 [r47701]  Kevin P. Fleming <kpfleming@digium.com>

	* main/file.c: don't try to call fclose() if fopen() failed

2006-11-15 20:31 +0000 [r47698]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: - Improve SIP history - Never send reply to
	  ACK (again...)

2006-11-15 20:31 +0000 [r47684-47697]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
	  | 4 lines ensure that message duration is included in email
	  notifications for forwarded messages (BE-96, fix by me after
	  corydon used his clue-bat on me) ensure that duration in the
	  message metadata is updated if prepending is done during
	  forwarding (related to BE-96) remove prototype for API call that
	  does not exist ........

	* main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
	  Nov 2006) | 2 lines clear the category's variable tail pointer as
	  well when variables are detached from it ........ r47688 |
	  kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
	  lines when appending a list of variable to a category, ensure the
	  tail pointer points to the last variable in the list ........
	  r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
	  | 2 lines when re-writing the config file, don't repeat the path
	  if it hasn't changed ........

	* main/config.c, /: Merged revisions 47682 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
	  | 2 lines ouch... don't use printf, use ast_log/ast_verbose
	  ........

2006-11-15 17:46 +0000 [r47672]  Luigi Rizzo <rizzo@icir.org>

	* main/cli.c: fix longest match search in find_cli. Trunk already
	  fixed. 1.2 not affected (well, i have no idea, the code is
	  totally different there).

2006-11-15 15:25 +0000 [r47649-47656]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Send error message when we can't allocate
	  SIP dialog, possibly due to limitation of file descriptors.
	  (imported from 1.2)

2006-11-15 04:45 +0000 [r47645]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: If NAT detection is turned on or already detected
	  then say NAT is active when setting the remote RTP peer when
	  doing early bridging. (issue #8365 reported by marcelbarbulescu)

2006-11-15 00:19 +0000 [r47641]  Kevin P. Fleming <kpfleming@digium.com>

	* main/term.c: more formatting cleanup, and avoid running off the
	  end of the string

2006-11-15 00:14 +0000 [r47639]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Turn notice about unknown RTCP packet type into a
	  debug message instead.

2006-11-15 00:05 +0000 [r47635]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
	  platforms (this variable is an 'int' anyway, comparing it to
	  'signed long' is not useful)

2006-11-14 22:17 +0000 [r47625-47632]  Joshua Colp <jcolp@digium.com>

	* apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
	  lines Update copyright information in the ADSI logo blob.
	  ........

	* channels/chan_sip.c: Only keep the video RTP structure around if
	  1. Video support is enabled and 2. A video codec is enabled on
	  the dialog

	* funcs/func_uri.c: Small documentation clarification for
	  URIENCODE. (issue #8294 reported by salaud)

2006-11-14 18:54 +0000 [r47621]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Conversion of res_odbc API to include ast_
	  prefix did not completely transition app_voicemail when
	  ODBC_STORAGE is used (reported on IRC by caio1982, not in
	  bugtracker)

2006-11-14 16:45 +0000 [r47617]  Joshua Colp <jcolp@digium.com>

	* apps/app_amd.c: Use LOG_DEBUG to print out the indication that
	  app_amd is using default settings instead of using LOG_NOTICE.
	  This stops needless logging of this information under normal
	  circumstances. (issue #8361 reported by Seb7)

2006-11-14 16:22 +0000 [r47597-47613]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Update documentation to fit the
	  implementation...

	* /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
	  retransmission system if it's an OPTION packet from peerpoke

2006-11-13 21:28 +0000 [r47584]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
	  lines Initialize global pointers for connection and result to
	  NULL. (issue #8356 reported by james) ........

2006-11-13 20:20 +0000 [r47581]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
	  | 2 lines Having more than 255 old messages caused corruption in
	  the new/old count ........

2006-11-13 19:15 +0000 [r47576]  Steve Murphy <murf@digium.com>

	* main/config.c: This solves bug 8342, whereby a crash occurs under
	  certain circumstances while reading a config file with comments--
	  a call to CB_ADD shouldn't happen if withcomments is zero

2006-11-13 19:11 +0000 [r47573]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/cli.c, channels/chan_sip.c: Re-enable old deprecated
	  commands

2006-11-13 19:10 +0000 [r47572]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: - Don't reply to INVITE already replied
	  to when we get BYE - Declare errmsg as int. Oops.

2006-11-13 18:18 +0000 [r47564]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
	  the messed if, but we all forgot to update the regressions. Until
	  now.

2006-11-13 17:13 +0000 [r47553]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
	  found... just confuses users

2006-11-13 17:08 +0000 [r47542-47551]  Joshua Colp <jcolp@digium.com>

	* /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
	  lines When sending an SMS with a user data header properly set
	  the UDH flag in the first byte. (issue #8347 reported by
	  hoffmeis) ........

	* main/cli.c: Free full command string upon unregistering of CLI
	  command. Backported from revision 47536 from rizzo.

2006-11-13 16:00 +0000 [r47540]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Only produce error message about sip history
	  once

2006-11-13 05:48 +0000 [r47527]  Russell Bryant <russell@digium.com>

	* configure, acinclude.m4: AC_PROG_SED is included in autoconf
	  2.60, but apparently it is not included in 2.59. So, to maintain
	  compatability with 2.59 since it is a small change, copy this
	  macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
	  #8345)

2006-11-13 05:46 +0000 [r47523-47526]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
	  | 2 lines If the execute fails a second time, make sure that we
	  don't pass back a stale handle ........

	* channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
	  | 2 lines Don't play dialtone if the seizing the channel fails
	  (Bug 7754) ........

2006-11-12 16:12 +0000 [r47507-47513]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
	  DEA!!!)

	* channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
	  UDPTL in sdp...

	* channels/chan_sip.c: - Don't destroy SIP dialog because of a
	  failed T.38 re-invite. Wait for a bye. Final response to a
	  re-invite does not mean that the session dies, only that the
	  re-invite fails. - Keep RTP active during processing of T.38
	  re-invite. If the re-invite fails, RTP needs to remain as before
	  the re-invite. Issue 8338 - darren1713. Please test.

	* channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
	  -Add some comments to t.38 code

2006-11-12 06:23 +0000 [r47492-47497]  Russell Bryant <russell@digium.com>

	* /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
	  4 lines Only do the check to determine whether the channel
	  calling this function is an IAX2 channel when getting the IP
	  address using the special argument, CURRENTCHANNEL. (issue #8341,
	  jcovert) ........

	* Makefile: Add the target "menuconfig" as an alias for the
	  "menuselect" target. This is just a favor to users so that if you
	  accidentally type "make menuconfig" instead of "make menuselect",
	  it still works. (inspired by a comment on IRC from wangster
	  calling me an "especially devious asterisk developer" for having
	  it be menuselect instead of menuconfig. :) )

	* main/term.c: Tweak the formatting of this new function to better
	  conform to coding guidelines.

2006-11-11 02:04 +0000 [r47490]  Matt O'Gorman <mogorman@digium.com>

	* main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
	  safe output!

2006-11-10 22:23 +0000 [r47480]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Make sure we don't use 32 bits when we only
	  need one bit.

2006-11-10 21:42 +0000 [r47463-47476]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: ...and make sure that the dialog is
	  destroyed, even if we don't get any answer on the bye... This is
	  the channel that remains dead after the SIP transfer

	* channels/chan_sip.c: Add debug output while trying to trace bug
	  in bug report

	* channels/chan_sip.c: Make sure we destroy dialog...

	* /, channels/chan_sip.c: Small cleanup of handle_request_invite()
	  - imported from 1.2 with changes

2006-11-10 19:47 +0000 [r47462]  Matt Frederickson <creslin@digium.com>

	* channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
	  callerid name for switches that bork on it.

2006-11-10 18:56 +0000 [r47454]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
	  SDP (alphaque)

2006-11-10 17:13 +0000 [r47444]  Luigi Rizzo <rizzo@icir.org>

	* build_tools/prep_moduledeps: grep -m is not available on BSD, so
	  use head -1 instead

2006-11-10 16:53 +0000 [r47437]  Joshua Colp <jcolp@digium.com>

	* apps/app_chanspy.c: Only split up extension and context if a
	  value exists. (issue #8332 reported by loloski)

2006-11-10 16:51 +0000 [r47436]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
	  channels/chan_skinny.c, channels/chan_h323.c,
	  channels/chan_iax2.c: Discussion of these CLI changes resulted in
	  more consistency (Bug 8236)

2006-11-10 16:36 +0000 [r47432-47433]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
	  removing them should be LOG_NOTICE, not LOG_DEBUG

	* apps/app_queue.c: reflect addition/removal of dynamic queue
	  members in queue_log, so that people using dialplan replacement
	  for AgentCallbackLogin can still track login/logout (issue #7736,
	  reported/patched by whoiswes but this commit was written by me
	  and covers all three paths for AQM/RQM)

2006-11-10 13:04 +0000 [r47414-47418]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Rip out half implementation of 491 response
	  support, since it wasn't implemented properly and caused memory
	  leaks in the case of us getting 491's, which Asterisk actually
	  sends... Since it is a bit too complicated to fix this, I'll rip
	  it out of 1.4 and put it on the to-do-list for future releases.
	  Now, we handle this as congestion, which it really is. Issue
	  #8331

	* channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
	  Thanks fenlander!

2006-11-10 03:44 +0000 [r47398-47405]  Joshua Colp <jcolp@digium.com>

	* channels/chan_h323.c: Fix building of chan_h323 by completeing
	  some structure definitions. (issue #8327 reported by Mithraen)

	* apps/app_voicemail.c: Do conversion in a more easier to read and
	  working way for \r, \n, and \t. (issue #8324 reported by
	  johnlange)

2006-11-09 21:26 +0000 [r47391]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c, channels/chan_zap.c,
	  build_tools/prep_moduledeps: Work around an issue that caused
	  menuselect to display a bogus description for app_voicemail and
	  chan_zap. These modules use some preprocessor directives to
	  determine what it will report to Asterisk as its description.
	  However, the way we extract this information from the source
	  files for menuselect is not smart enough to figure this out.
	  (issue #8326, #8328)

2006-11-09 16:53 +0000 [r47380]  Joshua Colp <jcolp@digium.com>

	* channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
	  2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
	  higher as, well, it's apparently going to be removed. This should
	  make all you FC6 fans happy as your Asterisk will now build
	  without any mods. ........

2006-11-09 16:28 +0000 [r47352-47377]  Russell Bryant <russell@digium.com>

	* main/cli.c: fix tab completion for "core debug channel" and "core
	  no debug channel"

	* main/cli.c: Fix "core show channel". Also, fix tab completion for
	  both "core show channel" and "core show channels".

	* main/cli.c: Fix "core debug channel <whatever>". I guess someone
	  needs to go through and audit every CLI command that changed
	  number of arguments ...

	* main/asterisk.c: revert the previous change, which actually
	  modified the deprecated command, "show profile". Now, actually
	  apply the change to "core show profile".

	* main/asterisk.c: Fix argument parsing for the "core show profile"
	  CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)

	* main/cli.c: Fix another CLI command, "core show uptime" ...
	  (issue #8323, reported by johnlange, fixed by myself)

	* main/asterisk.c: fix "core show version" to reflect the new
	  number of arguments for this CLI command (issue #8316, kshumard)

2006-11-08 23:14 +0000 [r47344-47348]  Steve Murphy <murf@digium.com>

	* main/channel.c: This update fixes 7531

	* channels/chan_skinny.c: Committed in behalf of 8190.

2006-11-08 21:46 +0000 [r47333-47338]  Kevin P. Fleming <kpfleming@digium.com>

	* main/frame.c: the battle over CLI command formats has broken
	  stuff...

	* channels/chan_sip.c: add simple fix for SDP to report proper
	  sample rate for G.722 media sessions

2006-11-08 17:03 +0000 [r47323-47331]  Russell Bryant <russell@digium.com>

	* utils/streamplayer.c: I occasionally get email from users that
	  are trying to figure out what this does, or due to some
	  misunderstanding as to what it is supposed to do, can't get it to
	  work. So, I have added some text here to hopefully explain what
	  this application does and does not do.

	* channels/chan_gtalk.c: Make this module build again

	* configure, configure.ac, acinclude.m4: Copy the macros from
	  libtool.m4 to our own acinclude.m4 such that libtool is no longer
	  required to be installed to be able to generated the configure
	  script.

2006-11-08 07:43 +0000 [r47309-47310]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)

2006-11-07 23:46 +0000 [r47303]  Steve Murphy <murf@digium.com>

	* channels/chan_oss.c, main/channel.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,
	  include/asterisk/stringfields.h, 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, main/utils.c, include/asterisk/channel.h,
	  channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
	  solve the problem in bug 7506. It's a lot of rework to solve a
	  fairly small problem... such is life.

2006-11-07 20:14 +0000 [r47284-47287]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Make MOH work as it did before in
	  chan_local, without this then it can go funky when transfers and
	  MOH are involved. (issue #7671 reported by jmls)

2006-11-07 18:56 +0000 [r47279]  Kevin P. Fleming <kpfleming@digium.com>

	* configs/musiconhold.conf.sample: clean up sample config, and make
	  native file playback the more obvious default choice

2006-11-07 18:38 +0000 [r47275]  Matt O'Gorman <mogorman@digium.com>

	* apps/app_voicemail.c: large overhaul to voicemail imap support.
	  Allows support for more imap servers, also a better
	  implementation of several parts of the original work. patch
	  provided by 8033 with major upgrades.

2006-11-07 17:30 +0000 [r47268]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
	  continue.

2006-11-07 13:13 +0000 [r47250]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Fixing the attack shield so it doesn't
	  produce attacks... Issue 8265 - never reply to an ACK

2006-11-07 01:25 +0000 [r47239]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
	  Nov 2006) | 5 lines If random order is enabled for files mode
	  music on hold, set a random initial position, instead of always
	  starting at the first file, and doing the random operation only
	  when switching to the next file. (bug reported by John Lange on
	  the asterisk-dev mailing list) ........

2006-11-04 18:32 +0000 [r47199]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
	  transfer from "john" Thank you!

2006-11-04 18:10 +0000 [r47192-47196]  Russell Bryant <russell@digium.com>

	* main/cli.c: Fix another bug in "core set debug" ...

	* main/asterisk.c, main/cli.c: Really fix the "core set debug" and
	  "core set verbose" CLI commands.

	* main/cli.c: fix the "atleast" option to the "core set verbose"
	  and "core set debug" CLI commands

2006-11-03 23:17 +0000 [r47176]  Steve Murphy <murf@digium.com>

	* channels/chan_sip.c: This fix introduced via bug 8233

2006-11-03 17:53 +0000 [r47107-47108]  Luigi Rizzo <rizzo@icir.org>

	* bootstrap.sh: align bootstrap.sh with the version in trunk (needs
	  to be blocked as it is already in trunk)

	* configure.ac: add proper environment vars to detect modules on
	  freebsd. (already applied to trunk so it needs to be blocked
	  there)

2006-11-02 23:49 +0000 [r47051-47053]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
	  channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
	  changes making the CLI more consistent with "category verb
	  arguments" (continuation of issue 8236)

	* main/config.c, main/cli.c, main/channel.c, main/manager.c,
	  channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
	  main/http.c, main/file.c, main/logger.c, main/image.c,
	  res/res_indications.c, main/asterisk.c, res/res_odbc.c,
	  channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
	  channels/chan_local.c, main/frame.c, channels/chan_sip.c,
	  res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
	  res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
	  Reverse change of "show" to "list" and make several other
	  commands more consistent with "category verb arguments"

2006-11-02 19:56 +0000 [r46992-47015]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Move check for codec translation to
	  sip_call() instead of in add_sdp. No one bothers with the result
	  of add_sdp anyway... Yet...

	* channels/chan_sip.c: Disable code for T38 over TCP and RTP since
	  there's no trace of actual functionality for it :-)

2006-11-02 17:49 +0000 [r46965]  Russell Bryant <russell@digium.com>

	* /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
	  Nov 2006) | 3 lines ignore files in a music on hold directory
	  that begin with '.' (issue #8249, cboie) ........

2006-11-02 17:17 +0000 [r46963]  Nadi Sarrar <ns@beronet.com>

	* channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix

2006-11-02 16:45 +0000 [r46937]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't send INVITE when we have determined
	  that we can't offer any audio formats due to lack of transcoding
	  support (or incorrect configuration)

2006-11-02 16:06 +0000 [r46930]  Joshua Colp <jcolp@digium.com>

	* /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
	  lines Repeat after me oej: I will at least make sure my code
	  compiles before I commit it. ........

2006-11-02 15:24 +0000 [r46901]  Olle Johansson <oej@edvina.net>

	* /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)

2006-11-02 14:02 +0000 [r46845-46883]  Russell Bryant <russell@digium.com>

	* /, main/callerid.c: Add the missing call to free described in
	  issue #8268. Also, add a bunch of missing calls to free in
	  callerid_feed_jp().

	* main/say.c: fix saying one hundred and two hundred in hebrew
	  (issue #7810, eldadran)

	* Makefile, configure, codecs/gsm/Makefile, configure.ac,
	  build_tools/strip_nonapi, makeopts.in: Fixes for
	  cross-compilation on mips (issue #8058, ywalther, with some
	  modifications)

	* aclocal.m4, build_tools/menuselect-deps.in, configure,
	  build_tools/embed_modules.xml, configure.ac: Add a check in the
	  configure script to determine whether ld is GNU ld or not. This
	  is needed because module embedding only works for gnu ld. GNU ld
	  is now listed as a dependency for all of the module embedding
	  options in menuselect. (issue #8143)

2006-11-01 20:35 +0000 [r46822]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_gtalk.c: bind address support from bug 8164

2006-11-01 19:49 +0000 [r46802]  Steve Murphy <murf@digium.com>

	* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
	  accept longer strings or mass confusion and a lot of lost time is
	  the result

2006-11-01 18:39 +0000 [r46780]  Joshua Colp <jcolp@digium.com>

	* main/Makefile: Force poll() emulation for Darwin to always be on.
	  It's too broken to consider being used. This resolves the console
	  issue OSX users have been seeing. I would have liked to autoconf
	  this but I haven't been able to come up with a test case that
	  works. Que sera.

2006-11-01 18:26 +0000 [r46778]  Russell Bryant <russell@digium.com>

	* res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
	  9 lines soxmix and Asterisk expect different file extensions for
	  certain formats. This was already handled for the wav49 format.
	  However, it was not handled for ulaw and alaw. I fixed this in
	  such a way that using the alternate extensions for ulaw and alaw
	  will only happen if we know we're calling soxmix, and not a
	  custom script defined using the MONITOR_EXEC variable. The wav49
	  processing was left alone so that external scripts will see no
	  behavior change. (issue #7550, reported by mnicholson, proposed
	  patch by junky, committed fix is a bit different) ........

2006-11-01 18:21 +0000 [r46775]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: It's another round of chan_iax2 fixes!
	  Should hopefully fix the deadlock issues people have been
	  reporting. IAXtel now has qualify turned on for 800 peers and it
	  is handling it fine.

2006-11-01 17:48 +0000 [r46760]  Steve Murphy <murf@digium.com>

	* main/config.c: Cleanups suggested by Russell.

2006-11-01 16:39 +0000 [r46744]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: Prevent an infinite loop when config
	  processing gets to a jitterbuffer option

2006-10-31 22:02 +0000 [r46716]  Jason Parker <jparker@digium.com>

	* main/translate.c: Fix "core show translation" output. Issue
	  #8243, patch by Damin.

2006-10-31 21:47 +0000 [r46711-46714]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/translate.h, main/translate.c: add an API so
	  that translators can activate/deactivate themselves when needed

	* include/asterisk/translate.h, main/translate.c: revert changes
	  that were the wrong way to address this... proper fix coming

	* main/translate.c: let's set the seen flag early enough to
	  actually make a difference...

	* include/asterisk/translate.h, main/translate.c: don't re-do setup
	  operations for translators that can dynamically register
	  themselves

2006-10-31 15:49 +0000 [r46663]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* /: Blocked revisions 46662 via svnmerge ........ r46662 |
	  tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines
	  Move thread-unsafe initializer to the module loading code; add
	  the corresponding function to the module unload to fix a memory
	  leak. ........

2006-10-31 10:56 +0000 [r46583-46631]  Olle Johansson <oej@edvina.net>

	* main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
	  #8089 - Fix the ENUM support (picking one record by number).
	  Thanks otmar!

	* /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
	  when we're supposed to support ;rport. Issue #7473.

	* /, channels/chan_sip.c: If peer fails ACL check, fail peer at
	  REGISTER

	* channels/chan_sip.c: Fix T38 too. Thanks, tgrman !

2006-10-31 06:30 +0000 [r46554-46563]  Russell Bryant <russell@digium.com>

	* contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
	  boot process to ensure it starts after stuff like MySQL (issue
	  #8253, Alric)

	* /, main/utils.c: Merged revisions 46560 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
	  3 lines When handling the case where the hostname is just an IPV4
	  numeric address, be sure to set the address type. (issue #8247,
	  alexr) ........

	* /, res/res_agi.c: Merged revisions 46557 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
	  3 lines fix some copy/paste bugs in the checking of arguments for
	  the "control stream file" AGI command (issue #8255, mnicholson)
	  ........

	* main/translate.c: Add a small tweak to the code that checks to
	  see whether destination formats are translatable based on the
	  source format. If we have already determined that there is no
	  translation path in one direction, don't bother checking the
	  other direction.

2006-10-30 22:19 +0000 [r46511-46526]  Kevin P. Fleming <kpfleming@digium.com>

	* main/translate.c: when unregistering a translator, don't rebuild
	  the translation matrix unless needed when filtering formats out
	  of an offer, ensure we check for translation ability in both
	  directions

	* include/asterisk/linkedlists.h: ensure that items removed from a
	  list are always unlinked from the list (next pointer set to NULL)

2006-10-30 21:09 +0000 [r46474-46506]  Joshua Colp <jcolp@digium.com>

	* configure, configure.ac: Don't explicitly link in crypt as it is
	  not used on some platforms.

	* channels/chan_iax2.c: We need to lock the pvt structure during
	  retransmission as another worker thread may be doing something as
	  well.

2006-10-30 16:27 +0000 [r46382-46433]  Olle Johansson <oej@edvina.net>

	* main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
	  include/asterisk/doxyref.h, channels/chan_sip.c,
	  main/ast_expr2f.c, include/asterisk/module.h,
	  formats/format_ogg_vorbis.c, main/app.c,
	  include/asterisk/channel.h, include/asterisk/lock.h,
	  include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
	  kshumard. An extra big thankyou is given to everyone that
	  contributes to doxygen! THANK YOU!

	* main/rtp.c, /: Bind RTCP to the same IP as RTP

	* /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
	  redirects (imported from 1.2)

	* /, channels/chan_sip.c: Issue #7608 - Notifications sent with
	  wrong content-type (imported from 1.2, modified)

	* channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
	  was reported for trunk, but obviously exists in 1.4 too.

	* channels/chan_sip.c: Restoring the old logic, since working
	  around it and fixing it seemed too complicated. - The
	  SIP_OUTGOING flag indicates the direction of the last transaction
	  in the dialog. - The initreq stores the last request in the
	  dialog, the request that opened the latest transaction. Please
	  now retry all the 1.4 bug reports with mixed to/from headers,
	  tags etc in ACK, BYE, CANCEL. Thanks!

	* channels/chan_sip.c: Accepting a message twice may be
	  misinterpreted...

	* channels/chan_sip.c: - 183 is not reliable message... - Error
	  should not have SDP

2006-10-28 16:37 +0000 [r46377]  Joshua Colp <jcolp@digium.com>

	* utils/Makefile: Don't build muted on OpenBSD, it is not
	  supported.

2006-10-27 19:03 +0000 [r46370]  Russell Bryant <russell@digium.com>

	* channels/chan_zap.c: move the copy of the default settings to the
	  global settings back out of process_zap, so that they aren't
	  overwritten when process_zap is called multiple times

2006-10-27 18:29 +0000 [r46367]  Olle Johansson <oej@edvina.net>

	* contrib/asterisk-ng-doxygen: Put some doxygen pressure on
	  Christian :-)

2006-10-27 17:39 +0000 [r46358-46363]  Russell Bryant <russell@digium.com>

	* main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
	  res/res_musiconhold.c: We should always be using _exit() after a
	  fork() or vfork() instead of exit(). This is because exit() does
	  some extra cleanup which in some implementations of vfork(), for
	  example, can actually modify the state of the parent process,
	  causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)

	* /: Blocked revisions 46361 via svnmerge ........ r46361 | russell
	  | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We
	  should always be using _exit() after a fork() or vfork() instead
	  of exit(). This is because exit() does some extra cleanup which
	  in some implementations of vfork(), for example, can actually
	  modify the state of the parent process, causing very weird bugs
	  or crashes. (issue #7971, Nick Gavrikov) ........

	* channels/chan_zap.c: Instead of iterating all of the options once
	  to look for jitterbuffer options, and then again for everything
	  else, move the processing of jitterbuffer options into the main
	  loop so that there are no erroneous messages about ignoring
	  unknown options. (issue #8226)

2006-10-27 10:03 +0000 [r46351-46353]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
	  Merged revisions 46350 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
	  1 line fixed a bug which caused chan_misdn to try to allocate 2
	  times the same channel on high load, which then caused
	  instability of mISDN. removed a useless function from isdn_lib.c
	  ........

	* channels/misdn_config.c: fixed not compile issue, which was just
	  introduced

	* channels/misdn_config.c, channels/chan_misdn.c, /,
	  channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
	  Merged revisions 46176 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
	  1 line added nttimeout option to configure wether we disconnect
	  calls on NT timeouts or not during an overlapdial session
	  ........

2006-10-26 17:57 +0000 [r46335-46340]  Jason Parker <jparker@digium.com>

	* /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
	  svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
	  lines oops - somebody forgot to change this - long ago, probably.
	  ........

	* CHANGES: grammar check

2006-10-26 16:38 +0000 [r46331]  Olle Johansson <oej@edvina.net>

	* CHANGES: Corrections to changes (Multiparking is not included)

2006-10-26 16:31 +0000 [r46329]  Russell Bryant <russell@digium.com>

	* main/translate.c: - If the source has no audio or no video
	  portion, do not call powerof() to get the format index. - Don't
	  run through the audio and video loops if there is no audio or
	  video portion of the source If 0 is passed to powerof, it will
	  return -1. This value of -1 was then being used as an array index
	  in these loops, which caused a crash on some systems. Other than
	  this issue, this code works as we expected it to. If a format is
	  not in the source, and we have to translation path to it, it is
	  not offered in the list of acceptable destination formats. (fixes
	  issue #8231)

2006-10-26 12:15 +0000 [r46317]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: update to reflect G.722 addition

2006-10-26 04:18 +0000 [r46298]  Russell Bryant <russell@digium.com>

	* doc/backtrace.txt: update backtrace documentation to reflect
	  changes in 1.4 (issue #8230, kshumard)

2006-10-26 01:37 +0000 [r46287]  Mark Spencer <markster@digium.com>

	* main/config.c, main/manager.c: Fix config comment code
	  preservation code (thanks murf!)

2006-10-25 20:14 +0000 [r46276]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Old todo note - Don't add Contact header on
	  BYE and Cancel

2006-10-25 19:24 +0000 [r46253-46255]  Russell Bryant <russell@digium.com>

	* configure.ac: fix error output when checking for openh323 to
	  refer to openh323 instead of pwlib (issue #8222, misaksen)

2006-10-25 19:16 +0000 [r46252]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Somewhat ugly code to try to fix issue
	  #7608. Since the problem was not very well defined, the fix is a
	  bit fuzzy too... Thanks to Luigi for accidentally spotting the
	  possible problem!

2006-10-25 19:08 +0000 [r46249]  Russell Bryant <russell@digium.com>

	* apps/app_queue.c: update warning message to include "agi" option
	  (issue #8225, jmls)

2006-10-25 18:13 +0000 [r46237-46248]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: use 1.4.3 extra sounds with corrected silence
	  files

	* sounds/sounds.xml, sounds/Makefile: add support for prebuilt
	  G.722 prompts and music on hold files

2006-10-25 15:56 +0000 [r46214-46216]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: show settings doesn't produce a list of
	  similar objects, it should stay a "show"

2006-10-25 14:32 +0000 [r46200]  Kevin P. Fleming <kpfleming@digium.com>

	* main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
	  channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
	  pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
	  main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
	  cdr/cdr_custom.c, channels/chan_mgcp.c,
	  apps/app_parkandannounce.c, apps/app_voicemail.c,
	  channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
	  res/res_adsi.c, main/utils.c, apps/app_ices.c,
	  pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
	  apps/app_getcpeid.c: apparently developers are still not aware
	  that they should be use ast_copy_string instead of strncpy... fix
	  up many more users, and fix some bugs in the process

2006-10-25 04:58 +0000 [r46165]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/pbx.c: WaitExten truncates decimals of times to wait,
	  instead of accepting them (Bug 8208)

2006-10-25 00:26 +0000 [r46152-46154]  Kevin P. Fleming <kpfleming@digium.com>

	* main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
	  channels/chan_h323.c, channels/chan_iax2.c,
	  include/asterisk/frame.h: add passthrough and file format support
	  for G.722 16KHz audio (issue #5084, original patch by andrew,
	  updated by mithraen)

	* channels/chan_sip.c, main/translate.c: code zone experiment:
	  don't offer formats in the outbound INVITE that aren't either
	  passthrough or translatable

	* main/translate.c: if multiple translators are registered for the
	  same source/dest combination, ensure that the lowest-cost one is
	  always inserted earlier in the list

2006-10-24 20:30 +0000 [r46142]  Mark Spencer <markster@digium.com>

	* res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
	  #8147)

2006-10-24 19:29 +0000 [r46130]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: We need to initialize our scheduler pthread
	  condition... yes.

2006-10-24 08:34 +0000 [r46114-46117]  Luigi Rizzo <rizzo@icir.org>

	* main/http.c: merge 45152 don't leak descriptors in http.c

	* channels/chan_sip.c: merge 45966 refer_to_domain potentially
	  containing options

	* channels/chan_sip.c: merge 46026 improper checks on get_header()
	  return values

	* channels/chan_sip.c: merge 46045 prevent NULL args to
	  ast_strdupa() in chan_sip.c

2006-10-24 05:23 +0000 [r46093]  Russell Bryant <russell@digium.com>

	* Makefile: Restore the ability to remove the firmware directory
	  without causing the installation to fail (issue #8111)

2006-10-24 03:53 +0000 [r46080-46083]  Kevin P. Fleming <kpfleming@digium.com>

	* main/translate.c: ensure that the translation matrix is properly
	  lock-protected every place it is used

	* include/asterisk/translate.h, main/translate.c: add an API call
	  to allow channel drivers to determine which media formats are
	  compatible (passthrough or transcode) with the format an existing
	  channel is already using

	* doc/imapstorage.txt: simplify and correct voicemail IMAP storage
	  build instructions

2006-10-24 03:01 +0000 [r46078]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* main/channel.c: Pass through a frame if we don't know what it is,
	  rather than trying to pass a NULL, which will segfault a channel
	  driver (Bug 8149)

2006-10-24 01:27 +0000 [r45999-46067]  Russell Bryant <russell@digium.com>

	* utils/muted.c, utils/ael_main.c: In muted.c, check the return
	  value of strdup. In ael_main.c, check the return value of calloc.
	  (issue #8157) In passing fix a few minor bugs in ael_main.c. The
	  last argument to strncpy() was a hard-coded 100, where it should
	  have been 99. I changed this to use sizeof() - 1.

	* apps/app_meetme.c: Fix the descriptions of some of the
	  MeetMeAdmin options (issue #8098, mflorell)

	* res/res_jabber.c: don't crash when an incoming message has no
	  "from" (issue #8205, jmls)

2006-10-23 00:27 +0000 [r45928]  Joshua Colp <jcolp@digium.com>

	* /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
	  r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
	  lines Don't leak memory mmmk? ........

2006-10-22 21:44 +0000 [r45916]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
	  Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
	  couldn't be initialized it would cause a segfault after 'reload'.
	  Reported by Drew/Matt thx. ........

2006-10-21 18:49 +0000 [r45818]  Russell Bryant <russell@digium.com>

	* res/res_monitor.c: Add a couple missing unregistrations of
	  manager actions and remove duplicate unregistrations of
	  applications. (issue #8194, jmls)

2006-10-21 18:48 +0000 [r45775-45817]  Joshua Colp <jcolp@digium.com>

	* main/loader.c: Don't use promotion on Darwin because it doesn't
	  seem to work quite right in all cases, this should solve the
	  unresolved symbol issue people have been seeing.

	* Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
	  installed in the proper location (reported on asterisk-dev
	  mailing list)

2006-10-20 07:44 +0000 [r45741]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Let's understand SIP: - REFER can create
	  dialog, Asterisk does not support it yet - NOTIFY can create
	  dialog in Asterisk's implementation (voicemail) even though we
	  don't support the server side of it. In this case, the standard
	  is a side issue ;-) - Added extened functionality for unsupported
	  methods (PING, PUBLISH) so we don't create PVT's for those
	  either. Russellb needs to judge what to do with this in 1.2, but
	  I think the current implementation n 1.2 is a bug since we're
	  sending bad replies to NOTIFY and REFER outside of dialogs

2006-10-19 17:24 +0000 [r45678-45694]  Joshua Colp <jcolp@digium.com>

	* res/res_jabber.c: Let's remember to unregister JabberStatus too
	  (issue #8184 reported by jmls)

	* /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
	  from https://origsvn.digium.com/svn/asterisk/branches/1.2
	  ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
	  2006) | 2 lines Respect language selection when seeing if the
	  file exists (issue #8178 reported by mnicholson) ........

	* channels/chan_sip.c: If the jitterbuffer is forced on then we
	  can't partially bridge (reported by wangster on #asterisk-dev)

2006-10-19 00:59 +0000 [r45622]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Don't leak the actual thread-specific
	  sip_pvt struct

2006-10-18 23:49 +0000 [r45621]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: don't leak memory when a chan_sip thread is
	  destroyed that has a thread-local temp_pvt allocated

2006-10-18 21:03 +0000 [r45595]  Joshua Colp <jcolp@digium.com>

	* main/asterisk.c: Don't modify things if we are using vfork as
	  this is very bad and may cause unexpected behavior (issue #7970
	  reported by Nick Gavrikov)

2006-10-18 11:54 +0000 [r45517]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: remove duplicate declarations

2006-10-18 04:09 +0000 [r45464]  Luigi Rizzo <rizzo@icir.org>

	* main/http.c: merge from trunk: move ast_variables_destroy() to a
	  better place in handle_uri() to avoid leaking memory on non
	  existing files.

2006-10-18 03:02 +0000 [r45452]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Don't segfault if you're using a channel driver that
	  doesn't turn RTCP on

2006-10-18 02:41 +0000 [r45439-45441]  Russell Bryant <russell@digium.com>

	* main/channel.c: Don't attempt to access private data members of
	  the pthread_mutex_t object, because this does not work on all
	  linux systems. Instead, just access the reentrancy field in the
	  ast_mutex_info struct when DEBUG_THREADS is enabled. If
	  DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
	  DEBUG_THREADS on as well. (issue #8139, me)

	* configs/sip_notify.conf.sample: update entry to reboot a snom
	  phone (issue #7850, pnlarsson)

2006-10-17  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta3 released.

2006-10-17 22:31 +0000 [r45408-45410]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/stringfields.h, main/ast_expr2.c,
	  main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
	  optimize the 'quick response' code a bit more... no more malloc()
	  or memset() for each response expand stringfields API a bit to
	  allow reusing the stringfield pool on a structure when needed,
	  and remove some unnecessary code when the structure was being
	  freed

2006-10-17 20:38 +0000 [r45378-45381]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't create a "real" pvt structure for
	  requests that shouldn't be able to create one. Instead use a
	  temporary pvt and fill it with enough information so we can send
	  a reply.

2006-10-17 17:39 +0000 [r45329]  Olle Johansson <oej@edvina.net>

	* configs/sip.conf.sample: Adding information about Marks
	  direct-RTP hack to the docs...

2006-10-17 17:22 +0000 [r45327]  Kevin P. Fleming <kpfleming@digium.com>

	* LICENSE: provide licensing language for IAXy firmware file

2006-10-16 20:06 +0000 [r45246-45280]  Joshua Colp <jcolp@digium.com>

	* apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
	  directed pickup (BE-85).

2006-10-16 13:59 +0000 [r45196-45213]  Olle Johansson <oej@edvina.net>

	* CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
	  your support!

	* channels/chan_sip.c: Don't destroy dialog for unexpected REFER
	  response...

2006-10-14 04:38 +0000 [r45143]  Steve Murphy <murf@digium.com>

	* funcs/func_rand.c: update the doc string for both AEL and
	  extensions.conf users.

2006-10-13 23:02 +0000 [r45125]  Kevin P. Fleming <kpfleming@digium.com>

	* main/acl.c don't drop the entire permit/deny list when an attempt
	  is made to add an invalid entry (BE-92)

2006-10-13 21:06 +0000 [r45104-45106]  Joshua Colp <jcolp@digium.com>

	* res/res_speech.c: Clear the quiet flag too since we are
	  restarting a recognition again (reported on -dev by Stephan
	  Edelman)

	* res/res_speech.c: Check return value from engine in case of
	  failure (ie: out of licenses) (reported on -dev mailing list)

2006-10-13 20:52 +0000 [r45103]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-vtest17 (added),
	  pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
	  pbx/ael/ael-test/ael-vtest17 (added),
	  pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
	  this release via these changes

2006-10-13 19:19 +0000 [r45088]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: avoiding warning, fixing potential bug

2006-10-13 18:42 +0000 [r45051-45079]  Joshua Colp <jcolp@digium.com>

	* codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
	  codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
	  codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
	  codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
	  codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
	  codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
	  codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
	  codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
	  codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
	  codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
	  codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
	  codecs/lpc10/analys.c, codecs/lpc10/onset.c,
	  codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
	  codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
	  codecs/lpc10/median.c, codecs/lpc10/encode.c,
	  codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
	  codecs/lpc10/invert.c: And file said... let the compiler warnings
	  STOP!

	* apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
	  reported by mnicholson)

	* apps/app_playback.c: Move say.conf existence check to do_say
	  function since it is called from multiple places (issue #8144
	  reported by kshumard)

2006-10-13 16:19 +0000 [r45049]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
	  we have multiple bindings (reported on asterisk-dev)

2006-10-13 16:01 +0000 [r45031-45040]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Complete merging in RPID screen changes
	  (issue #8101 reported by hristo, patch by oej in revision 44757)

	* main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
	  the background refresh item back into the scheduler if enabled
	  since it is deleted during reload. (issue #8142 reported by
	  p_lindheimer)

2006-10-13 15:41 +0000 [r45027]  Kevin P. Fleming <kpfleming@digium.com>

	* configure, include/asterisk/autoconfig.h.in, configure.ac,
	  main/utils.c: use a configure script test for PMTU discovery
	  control instead of just assuming it's available on Linux

2006-10-13 14:45 +0000 [r44994-45026]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
	  echocandisable issues when bridged. this caused a kernel panic
	  sometimes.. also some minor formatting fixes

	* channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
	  got a wrong isdn cause at RELEASE_COMPLETE

2006-10-12 22:07 +0000 [r44992]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: merge formatting and minor code
	  simplifications from trunk

2006-10-12 20:34 +0000 [r44982]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_gtalk.c: fix for bug 7764.

2006-10-12 19:14 +0000 [r44956-44971]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: we can only send one 'a=ptime' attribute per
	  media session, not one for each format

	* main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
	  main/utils.c: ensure that IAX2 and SIP sockets allow UDP
	  fragmentation when running on Linux (thanks to Brian Candler on
	  the asterisk-dev list for the tip)

2006-10-12 16:56 +0000 [r44945]  Russell Bryant <russell@digium.com>

	* main/manager.c: fix a silly typo in a comment that I saw while
	  reading the commit list

2006-10-12 16:08 +0000 [r44942]  Joshua Colp <jcolp@digium.com>

	* Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
	  #8135 reported by ssokol)

2006-10-12 12:55 +0000 [r44921]  Nadi Sarrar <ns@beronet.com>

	* main/manager.c: append_event must be called while holding the
	  session lock

2006-10-12 10:24 +0000 [r44911]  Russell Bryant <russell@digium.com>

	* res/res_jabber.c: change some debug output to use LOG_DEBUG
	  instead of verbose output

2006-10-11 16:57 +0000 [r44888]  Jason Parker <jparker@digium.com>

	* main/db1-ast/Makefile: These are already set by the parent
	  Makefile.. There is no need to have this here (it doesn't
	  actually work anyways).

2006-10-11 09:18 +0000 [r44854]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c: removed warning because of missing
	  prototype declaration

2006-10-10 19:23 +0000 [r44830]  Olle Johansson <oej@edvina.net>

	* channels/chan_sip.c: Do not set default/global values in the
	  variable declaration, set it in reload_config()

2006-10-10 17:21 +0000 [r44819]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Move some stuff around so that a NOTIFY
	  dialog won't hang around until the end of the world under certain
	  circumstances

2006-10-10 16:44 +0000 [r44809]  Paul Cadach <paul@odt.east.telecom.kz>

	* main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
	  CHANNEL() function sometime mix parameter and value

2006-10-10 16:42 +0000 [r44808]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* funcs/func_logic.c: Lost of a bit of logic when this was
	  simplified between 1.2 and 1.4 (Bug 8117)

2006-10-10 16:30 +0000 [r44806]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Bail out if we have no refer structure and
	  we get a refer response

2006-10-10 16:21 +0000 [r44805]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: more merge from trunk (comments and change a
	  static function name)

2006-10-10 15:23 +0000 [r44788]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Only set DTMF information if an RTP
	  structure exists

2006-10-10 13:50 +0000 [r44786]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
	  support of dynamically enabling hdlc on bchannels

2006-10-10 08:25 +0000 [r44776-44777]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: whitespace changes related to previous
	  commit

	* channels/chan_sip.c: merge a few code simplifications that have
	  gone into trunk during last week, to reduce differences between
	  the two branches and make porting fixes easier.

2006-10-09 16:12 +0000 [r44764]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Fix a problem where phones that go
	  "missing" never got unregistered. Issue #8067, reported by pj,
	  patch by Anthony LaMantia (with minor whitespace modifications)

2006-10-09 15:46 +0000 [r44759-44760]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
	  the deadlock

	* channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
	  (issue #8115 reported by vazir)

2006-10-08 14:14 +0000 [r44746]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: do not dereference p if we
	  know it is NULL

2006-10-07 14:39 +0000 [r44684]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
	  caller's transfer capability too

2006-10-07 11:37 +0000 [r44650-44665]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: put common code in a
	  function to avoid repetitions.

	* channels/chan_sip.c: remove hardwired usage of 5060, use
	  DEFAULT_SIP_PORT instead

	* channels/chan_sip.c: option_debug checking
	  before printing to debug channel.

	* channels/chan_sip.c: backport simplifications on sip_register,
	  usage of ast_set2_flag(), and fixes to the handling of failed
	  module loading.

	* channels/chan_sip.c: improve and document function
	  get_in_brackets(), introducing a helper function
	  find_closing_quote() of more general use.

2006-10-06 21:28 +0000 [r44629-44631]  Kevin P. Fleming <kpfleming@digium.com>

	* include/asterisk/linkedlists.h: ensure that mutex locks inside
	  list heads are initialized properly on platforms that require
	  constructor initialization (issue #8029, patch from timrobbins)

	* CHANGES: remove Jingle as per mog

2006-10-06 21:08 +0000 [r44628]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Remove the seqno check for RFC2833, the handler is
	  smart enough to not need it.

2006-10-06 21:07 +0000 [r44627]  Kevin P. Fleming <kpfleming@digium.com>

	* CHANGES: various cleanups

2006-10-06 18:46 +0000 [r44581-44605]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: When the sequence number rolls over then reset the
	  recorded sequence number for DTMF (issue #8106 reported by
	  bungalow)

	* main/file.c: Even more frames to treat as though the remote side
	  disappeared (issue #8097 reported by eldadran)

2006-10-06 15:59 +0000 [r44567]  Luigi Rizzo <rizzo@icir.org>

	* main/manager.c, main/http.c: make sure sockets are blocking when
	  they should be blocking.

2006-10-06 12:53 +0000 [r44559-44563]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c: fixed segfault which happens during
	  hold/transfer action

	* channels/chan_misdn.c: if INFORMATION Message come with keypad
	  instead of called party number, we just use the keypad as called
	  party number.

	* 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:
	  added the option 'reject_cause' to make it possible to set
	  the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
	  which is automatically rejected because chan_misdn does not
	  support that kind of callwaiting. Therefore chan_misdn supports
	  now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
	  now gets the info if the requested channel is incoming or
	  outgoing to make the 3. channel possible

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
	  removed a useless bc field, added setting of frame.delivery fields,
	  some minor code cleanups

2006-10-05 19:57 +0000 [r44502]  Joshua Colp <jcolp@digium.com>

	* main/file.c: Treat busy control frames as hangup in the file streaming
	  core (issue #8097 reported by eldadran)

2006-10-05 18:21 +0000 [r44488]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
	  Many thanks to Doug!

2006-10-05 18:01 +0000 [r44486]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
	  hanging by a thread if the other side is already setup with T.38

2006-10-05 16:10 +0000 [r44476]  Kevin P. Fleming <kpfleming@digium.com>

	* main/app.c: don't segfault when an argument without a close
	  parenthesis is found stop parsing as soon as that situation
	  occurs

2006-10-05 15:22 +0000 [r44465-44466]  Steve Murphy <murf@digium.com>

	* CHANGES: I put the accumulated changes from the commit logs and
	  inspection, into CHANGES. Hope everyone approves!

	* configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
	  install process sticks muted.conf in /etc/asterisk, so that's
	  where muted should look for it, right?

2006-10-05 02:40 +0000 [r44450]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Don't totally bail out if T.38 was
	  negotiated

2006-10-05 01:42 +0000 [r44433-44436]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: fix Polycom presence notification again

2006-10-04 22:52 +0000 [r44407-44409]  Luigi Rizzo <rizzo@icir.org>

	* utils/Makefile: as far as i can tell astman only uses newt...

	* Makefile: put linker flags in ASTLDFLAGS where they belong

2006-10-04 21:17 +0000 [r44390-44393]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
	  requests add workaround for new Polycom firmware SUBSCRIBE
	  requests (bug is known to exist in 2.0.1 firmware)

	* include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
	  work

2006-10-04 19:57 +0000 [r44380]  Steve Murphy <murf@digium.com>

	* pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
	  pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
	  pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
	  pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
	  pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
	  pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
	  pbx/ael/ael-test/ael-test16/extensions.ael (added),
	  pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
	  pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
	  pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
	  pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
	  problems reported in bug 8090

2006-10-04 19:47 +0000 [r44378]  Kevin P. Fleming <kpfleming@digium.com>

	* channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
	  main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
	  channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
	  channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
	  main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
	  include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
	  channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
	  main/devicestate.c, main/utils.c, res/res_musiconhold.c,
	  channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
	  thread creation code a bit reduce standard thread stack size
	  slightly to allow the pthreads library to allocate the stack+data
	  and not overflow a power-of-2 allocation in the kernel and waste
	  memory/address space add a new stack size for 'background'
	  threads (those that don't handle PBX calls) when LOW_MEMORY is
	  defined

2006-10-04 17:04 +0000 [r44337-44365]  Steve Murphy <murf@digium.com>

	* configs/muted.conf.sample: I've been meaning to add some
	  explanation about muted... here it is

	* configs/manager.conf.sample: CLI reverbification update to this
	  config file

	* apps/app_macro.c: In response to bug 7776, a Warning has been
	  added to the doc string for Macro().

2006-10-04 00:25 +0000 [r44322]  Kevin P. Fleming <kpfleming@digium.com>

	* main/asterisk.c, main/loader.c, main/term.c, Makefile,
	  include/asterisk.h: ensure that local include files are always
	  used avoid a duplicate function name (term_init())

2006-10-03 22:35 +0000 [r44312]  Matt O'Gorman <mogorman@digium.com>

	* channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
	  client without resource.

2006-10-03 20:18 +0000 [r44298]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: fix a logic error in my previous fix to the queue
	  reload code

2006-10-03 18:42 +0000 [r44286]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Change default presentation indicator
	  to "user provided not screened" if octet 3a missed in
	  CallingPartyNumber IE

2006-10-03 18:35 +0000 [r44284]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Use VideoSupport instead so it is considered
	  a valid XML attribute name. (issue #8075 reported by renemendoza)

2006-10-03 18:30 +0000 [r44283]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Fix preparation of type and
	  presentation of calling number

2006-10-03 00:01 +0000 [r44240]  Matt O'Gorman <mogorman@digium.com>

	* doc/jingle.txt, channels/chan_jingle.c (removed),
	  include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
	  res/res_jabber.c: updated res_jabber for even better component
	  support, soon will be jep-0100 compliant. also removed
	  chan_jingle and infromed info from jingle.txt, chan_gtalk still
	  works and should be used in this version.

2006-10-02 20:11 +0000 [r44199-44215]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Change the fd on the I/O context in case it
	  changed during the reload, which is indeed possible. (issue #7943
	  reported by eclubb)

	* contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
	  instead of hardcoding the path for the error message (issue #7942
	  reported by eclubb)

2006-10-02 18:52 +0000 [r44186]  Paul Cadach <paul@odt.east.telecom.kz>

	* configs/users.conf.sample, pbx/pbx_config.c: Missed part of
	  userconf functionality for chan_h323

2006-10-02 17:25 +0000 [r44169]  Joshua Colp <jcolp@digium.com>

	* main/io.c: Shrink when current_ioc is unused. It is set to -1 when
	  unused, not 0. (issue #7941 reported by eclubb)

2006-10-02 17:16 +0000 [r44166-44167]  Paul Cadach <paul@odt.east.telecom.kz>

	* doc/realtime.txt: Typo fix

	* channels/chan_h323.c: Optimization of oh323_indicate(): less
	  locks - less problems, plus single exit point

2006-10-02 02:38 +0000 [r44146]  Mark Spencer <markster@digium.com>

	* channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
	  you're not talking about a channel :)

2006-10-01 19:32 +0000 [r44135]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_h323.c: Do not simulate any audio tones if we got
	  PROGRESS message

2006-10-01 18:30 +0000 [r44111-44125]  Russell Bryant <russell@digium.com>

	* Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
	  be empty. The cause is that since ASTDATADIR is explicitly
	  exported using "export ASTDATADIR" at the top of the Makefile,
	  make no longer considers the variable "undefined", so the
	  Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
	  #8063, reported by akohlsmith, fixed by me)

	* configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
	  option in the sample queues.conf (issue #8065, adamg)

2006-10-01 15:01 +0000 [r44109]  Luigi Rizzo <rizzo@icir.org>

	* channels/chan_sip.c: sync with trunk - move variable declarations
	  to the beginning of a block.

2006-09-30 19:20 +0000 [r44090]  Paul Cadach <paul@odt.east.telecom.kz>

	* main/rtp.c: Allow one-way RTP streams (device->Asterisk)

2006-09-30 16:28 +0000 [r44080]  Luigi Rizzo <rizzo@icir.org>

	* codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
	  build problems: - with AST_DEVMODE, building codecs/lpc10 fails
	  because of lots of warnings, and the configure step in editline
	  fails as well. Fix this by removing the -Werror in these steps. -
	  on FreeBSD (but probably on other platforms as well), the final
	  link of asterisk fails because AST_LIBS was not exported to the
	  subdirs Makefiles. Add a proper fix in the top-level Makefile (a
	  possible alternative way is to add "export AST_LIBS" near the
	  beginning of the file). With this fix, i believe that some of the
	  platform-specific conditionals in main/Makefile are redundant
	  (because they should be already dealt with in the top level
	  Makefile) but i don't have a platform to check. Merging to head
	  will happen in a moment.

2006-09-30 16:12 +0000 [r44068-44078]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
	  of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
	  by phsultan with a small fix by me, myself or I. Thanks,
	  Philippe! (This was caused by my changes to the transaction
	  handling)

	* channels/chan_sip.c: Found some buggy SIP clients (phones Planet
	  VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
	  sends ACK not on OK message only (when remote party answers) but
	  on RINGING message too, so when we send 200 OK message, we get
	  unidentified ACK message (because INVITE acknowledged on RINGING
	  message already), so 200 OK retransmits within its retransmission
	  interval then call gets dropped. If someone else knows how to
	  provide workaround for such cases, please, fix it in correct way.
	  Thanks to ssh from #asteriskru for provide access to his box to
	  study and fix this case.

2006-09-29 22:51 +0000 [r44055-44057]  Kevin P. Fleming <kpfleming@digium.com>

	* agi, utils: ignore temporary files made by the Makefiles during a
	  build

	* codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
	  codecs/Makefile, utils/Makefile, configure,
	  build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
	  Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
	  pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
	  system bugs, and convert Makefiles to be compatible with GNU make
	  3.80

2006-09-29 22:35 +0000 [r44053]  Jason Parker <jparker@digium.com>

	* main/asterisk.c, main/cli.c: Fix a bug with the removal of
	  'atleast' argument to 'core verbose' and 'core debug'. Add that
	  argument back in.

2006-09-29 21:09 +0000 [r44022-44043]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
	  carefully when no CallingNumber IE available

	* channels/h323/ast_h323.cxx: Fake display name by called number on
	  incoming calls (until passing connected number/connected name is
	  not implemented)

	* channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
	  includes

	* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
	  pass TON/PRESENTATION information - original
	  H323Connection::SendSignalSetup() destroys Q.931 fields.

2006-09-29 18:49 +0000 [r44011-44012]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile: yet another place where we were not using the
	  correct CFLAGS by default

	* main/Makefile: missed one conversion to ASTCFLAGS

2006-09-29 18:30 +0000 [r44009]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
	  TON/PRESENTATION information too

2006-09-29 18:25 +0000 [r43952-44008]  Kevin P. Fleming <kpfleming@digium.com>

	* main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
	  main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
	  Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
	  CFLAGS and LDFLAGS for build of Asterisk components, because they
	  are also then used for non-Asterisk components (like menuselect);
	  use our own variables instead

	* configure, configure.ac: support --without-curl in configure
	  script

	* Makefile.rules: another cross-compile fix

	* Makefile: a couple more environment settings that can't leak into
	  the menuselect build

	* main/cli.c: proper fix for ast_group_t change

	* include/asterisk/lock.h: eliminate compiler warning when
	  DEBUG_CHANNEL_LOCKS is enabled and users of this header file
	  don't also include channel.h

2006-09-28 20:11 +0000 [r43944]  Jason Parker <jparker@digium.com>

	* apps/app_queue.c: Fix incorrect argument order for member names,
	  on persisted members. Issue 8047, patch by jmls.

2006-09-28 18:05 +0000 [r43932-43933]  Joshua Colp <jcolp@digium.com>

	* apps/app_playback.c, res/res_monitor.c,
	  include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
	  channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
	  main/udptl.c, main/frame.c, funcs/func_timeout.c,
	  channels/chan_sip.c, apps/app_festival.c,
	  channels/iax2-provision.c, apps/app_alarmreceiver.c,
	  res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
	  Put in missing \ns on the end of ast_logs (issue #7936 reported
	  by wojtekka)

2006-09-28 17:35 +0000 [r43919]  Kevin P. Fleming <kpfleming@digium.com>

	* apps/app_queue.c: fix buggy (and overly complex) loop used during reload
	  of app_queue for static member list updating

2006-09-28 17:34 +0000 [r43918]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Extend call establishment timeout

2006-09-28 17:31 +0000 [r43913-43915]  Joshua Colp <jcolp@digium.com>

	* channels/chan_iax2.c: Make sure the pvt exists before accessing
	  it again as it may have gone away (issue #7562 reported by Seb7
	  and issue #7939 reported by sorg)

	* main/cli.c: Warning be gone!

2006-09-28 16:41 +0000 [r43899]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: app_queue is comparing the device names incorrectly
	  while checking their statuses. It's internal list of interfaces
	  includes the dial string, while the argument passed to this
	  function does not have the dial string (/n for a local channel).
	  This causes it to ignore the device state changes because it
	  thinks it belongs to none of its members. (#8040 reported and
	  patch by tim_ringenbach)

2006-09-28 16:17 +0000 [r43893]  Joshua Colp <jcolp@digium.com>

	*  apps/app_meetme.c: Stop the stream after waitstream returns so that our
	  formats get restored. (issue #7370 reported by kryptolus)

2006-09-28 15:56 +0000 [r43877]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Fix compiler warning

2006-09-28 15:29 +0000 [r43864-43873]  BJ Weschke <bweschke@btwtech.com>

	* apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
	  tim_ringenbach reported and patched)

	* apps/app_queue.c: Autopause not working for queue members. (#8042
	  - jmls reported and patch)

2006-09-28 12:58 +0000 [r43861-43862]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
	  remote side to start media on outgoing PROGRESS message

	* include/asterisk/compiler.h: Put attribute tag at correct place

2006-09-28 11:03 +0000 [r43852]  Christian Richter <christian.richter@beronet.com>

	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
	  channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
	  when the call could not be properly established in misdn_call.
	  also removed the ACK_HDLC stuff which is not really needed.

2006-09-28 10:51 +0000 [r43843-43846]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/ast_h323.cxx: Do not open transmit channel until
	  TCS is received

	* main/file.c: Don't warn on HOLD/UNHOLD control frames

	* main/file.c: Don't treat unknown control frames as voice

2006-09-27 20:21 +0000 [r43816]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Avoid inability to lock directory log message by
	  creating the directory ahead of time. (Issue 7631)

2006-09-27 19:44 +0000 [r43801-43803]  Jason Parker <jparker@digium.com>

	* apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
	  not being set under certain circumstances. Fix a minor issue, to
	  make it use the filenames that were parsed, instead of the entire
	  argument string. Fix Background() to return -1 like Playback(),
	  if no args are specified.

2006-09-27 19:10 +0000 [r43783-43798]  Joshua Colp <jcolp@digium.com>

	* main/rtp.c: Compensate for out of order packets better if RFC2833
	  compensation is turned on.

	* channels/chan_iax2.c: Get rid of two functions from a time now
	  past (we THINK these are from pre-recursive lock time) that may
	  be contributing to two open issues on the bug tracker (7562/7939)
	  and that has the potential to just make bad things happen if the
	  timing is right.

2006-09-27 16:55 +0000 [r43779]  Russell Bryant <russell@digium.com>

	* main/channel.c,res/res_features.c: Fix a problem that occurred if
	  a user entered a digit
	  that matched a bridge feature that was configured using multiple
	  digits, and the digit that was pressed timed out in the feature
	  digit timeout period. For example, if blind transfer is
	  configured as '##', and a user presses just '#'. In this
	  situation, the call would lock up and no longer pass any frames.
	  (issue #7977 reported by festr, and issue #7982 reported by
	  michaels and valuable input provided by mneuhauser and kuj. Fixed
	  by me, with testing help and peer review from Joshua Colp). There
	  are a couple of issues involved in this fix: 1) When
	  ast_generic_bridge determines that there has been a timeout, it
	  returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
	  this result, it calls ast_generic_bridge over again with the same
	  timestamp for the next event. This results in an endless loop of
	  nothing until the call is terminated. This is resolved by simply
	  changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
	  sees a timeout. 2) I also changed ast_channel_bridge such that if
	  in the process of calculating the time until the next event, it
	  knows a timeout has already occured, to immediately return
	  AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
	  anyway. 3) In the process of testing the previous two changes, I
	  ran into a problem in res_features where ast_channel_bridge would
	  return because it determined that there was a timeout. However,
	  ast_bridge_call in res_features would then determine by its own
	  calculation that there was still 1 ms before the timeout really
	  occurs. It would then proceed, and since the bridge broke out and
	  did *not* return a frame, it interpreted this as the call was
	  over and hung up the channels. The reason for this was because
	  ast_bridge_call in res_features and ast_channel_bridge in
	  channel.c were using different times for their calculations.
	  channel.c uses the start_time on the bridge config, which is the
	  time that the feature digit was recieved. However, res_features
	  had another time, 'start', which was set right before calling
	  ast_channel_bridge. 'start' will always be slightly after
	  start_time in the bridge config, and sometimes enough to round up
	  to one ms. This is fixed by making ast_bridge_call use the same
	  time as ast_channel_bridge for the timeout calculation. ........

2006-09-27 16:24 +0000 [r43775]  Christian Richter <christian.richter@beronet.com>

	* channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
	  versioning, since Asterisk has it's own

2006-09-27 16:23 +0000 [r43774]  Joshua Colp <jcolp@digium.com>

	* channels/chan_sip.c: Make rfc2833compensate a global option.

2006-09-27 04:35 +0000 [r43756]  Russell Bryant <russell@digium.com>

	* apps/app_voicemail.c: Backport revision 43754 from the trunk,
	  which removes an unused buffer from mm_login to close bug 8038,
	  as well as addresses some formatting and coding guidelines issues
	  in passing. Originally, I did not commit this to 1.4 since it is
	  not necessarily fixing a bug. However, since the IMAP storage
	  code is brand new, I decided it would be better to make the
	  change here as well, in case someone has to work on this code to
	  address issues in the very near future. I don't want to make
	  unnecessary merge problems going to the trunk.

2006-09-27 02:32 +0000 [r43739]  Steve Murphy <murf@digium.com>

	* configs/extensions.ael.sample: This change to extensions.ael was
	  to fix bug 8031; the install scripts are causing it to be copied
	  to /etc/asterisk/extensions.ael, and because it is a fairly
	  direct conversion of the original extensions.conf, the macro and
	  context names clash with the existing extensions.conf. So, I put
	  an ael- in front of all macros and contexts, and checked every
	  goto and macro call. Also, this file compiles under aelparse.

2006-09-26 20:56 +0000 [r43710]  Russell Bryant <russell@digium.com>

	* main/asterisk.c: Back in revision 4798, this message was changed from
	  using ast_cli() to directly calling write(). During this change,
	  checking if this was a remote console was removed. This caused
	  this message about using "exit" or "quit" to exit an Asterisk
	  console to come up in times where it did not make sense. This
	  change restores the check to see if this is a remote console
	  before printing the message. (fixes BE-65)

2006-09-26 20:47 +0000 [r43707]  Joshua Colp <jcolp@digium.com>

	* .cleancount, main/cli.c, channels/chan_sip.c,
	  include/asterisk/channel.h: Use proper type to represent the group variable
	  (issue #8025 reported by makoto)

2006-09-26 20:30 +0000 [r43700-43703]  Russell Bryant <russell@digium.com>

	* channels/chan_sip.c: Add missing newline character in the warning
	  message about deprecated TOS values in configuration.

	* apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
	  mailbox definitions, don't introduce a length limit on the
	  definition by using a 256 byte temporary storage buffer. Instead,
	  make the temporary buffer just as big as it needs to be to hold
	  the entire mailbox definition. (fixes BE-68)

2006-09-26 20:19 +0000 [r43695-43697]  Joshua Colp <jcolp@digium.com>

	* channels/chan_local.c: Strip options off the argument passed for
	  devicestate in chan_local. (issue #8034 reported by pcardozo)

	* apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
	  overhaul of the whisper support. 1. We need to duplicate the
	  frame from ast_translate 2. We need to ensure we always have
	  signed linear coming in for signed linear combining. 3. We need
	  to ensure we are always feeding signed linear out. 4. Properly
	  store and restore write format when beeping on the channel we are
	  whispering on. 5. Properly discontinue the stream on the channel
	  for the beep. (issue #8019 reported by timkelly1980)

2006-09-26 18:34 +0000 [r43676]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: update to use 1.4.3 core sounds, with corrected
	  beep/beeperr/tt-monkeys files

2006-09-26 18:08 +0000 [r43650-43674]  Jason Parker <jparker@digium.com>

	* doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
	  Dan Austin. Maximum values were incorrect, which is why this is
	  being put in 1.4

	* channels/chan_skinny.c: Add proper codec support to chan_skinny.
	  Works with at least ulaw, alaw, and g729a. This is technically a
	  "new feature", but there are justifications for it. I found a bug
	  with the recent rtp packetization changes, which caused the media
	  setup to fail under certain circumstances, particularly when
	  using allow=all, or having no allow= statements (globally or on
	  the device). I could have either removed the rtp packetization
	  features, or I could add proper codec support (which, without, I
	  think most people would consider to be a bug anyways).

2006-09-25 22:07 +0000 [r43640-43642]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_voicemail.c: Should have moved these lines up in the
	  merge, instead of removing them

	* apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
	  delete=yes was ignored 2) maxmessages was ignored

2006-09-25 21:26 +0000 [r43626-43635]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
	  channels/h323/cisco-h225.asn: Fix ASN1 description of
	  non-standard Cisco extensions

	* channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
	  changes of trunk: 1) r43540: Avoid possible deadlock on channel
	  destruction 2) r43590: Disable fastStart if requested by remote
	  side

2006-09-25 15:23 +0000 [r43616]  Jason Parker <jparker@digium.com>

	* sounds/Makefile: One more fix for sounds installation - this time
	  for portability. Reported to asterisk-dev mailing list.

2006-09-25 14:52 +0000 [r43605]  Steve Murphy <murf@digium.com>

	* formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
	  crashing if trying to play an OGG moh file.

2006-09-25 06:15 +0000 [r43582]  Paul Cadach <paul@odt.east.telecom.kz>

	* channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
	  channels/chan_h323.c: Merged revisions 43472,43495 from trunk

2006-09-24 14:58 +0000 [r43553-43564]  Russell Bryant <russell@digium.com>

	* channels/iax2-provision.c: Fix a CLI command registration issue
	  where an erroneous message claiming that "iax2 show provisioning"
	  was already registered. This was because this command was
	  registering itself as both the command, as well as the command it
	  is deprecating. (issue #8022, reported by bjweeks, fixed by
	  myself)

	* channels/chan_iax2.c:Check to see if the channel that is activating the
	  IAXPEER function is actually an IAX2 channel before proceeding to
	  process it to avoid crashing. (issue #8017, reported by admott,
	  fixed by myself)

2006-09-22 23:44 +0000 [r43524]  Kevin P. Fleming <kpfleming@digium.com>

	* Makefile: don't output the 'build complete' message when the
	  target being run is already going to do an installation

2006-09-22 22:12 +0000 [r43518]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
	  properly. Remove reload support, since it doesn't
	  actually...work.

2006-09-22 21:36 +0000 [r43505-43508]  Steve Murphy <murf@digium.com>

	* pbx/pbx_ael.c: This commits a change to return
	  MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
	  goes well for bug 8004

	* pbx/pbx_ael.c: If the extensions.ael file not found, or
	  unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.

2006-09-22 17:25 +0000 [r43492]  Jason Parker <jparker@digium.com>

	* main/cli.c: Make sure we explicitly set the CLI command to not be
	  deprecated, if it isn't.

2006-09-22 16:42 +0000 [r43486-43489]  Kevin P. Fleming <kpfleming@digium.com>

	* sounds/Makefile: use rebuilt extra sounds

	* main/channel.c: all the Linux systems I have don't use
	  '__m_count' for this field, so I don't know where this came
	  from...

2006-09-22 15:47 +0000 [r43477-43484]  Russell Bryant <russell@digium.com>

	* include/asterisk/threadstorage.h: backport the compatability fix
	  to use attribute_malloc instaed of __attribute__ ((malloc))

	* channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
	  could not be configured (issue #8006, Mithraen)

	* main/frame.c: Suppress a compiler warning about the use of a
	  potentially uninitialized variable. It couldn't actually happen,
	  though.

2006-09-22 03:01 +0000 [r43469]  Jason Parker <jparker@digium.com>

	* channels/chan_skinny.c: First shot at unload_module in
	  chan_skinny.. More to come.

2006-09-21 23:50 +0000 [r43466]  Matt O'Gorman <mogorman@digium.com>

	* include/asterisk/jabber.h, channels/chan_gtalk.c,
	  res/res_jabber.c: updates for better compontent support

2006-09-21 23:24 +0000 [r43464]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
	  actually documented how the new features in res_odbc actually
	  work. (Oops)

2006-09-21 22:21 +0000 [r43454-43456]  Joshua Colp <jcolp@digium.com>

	* channels/chan_oss.c: Some more clean up in the load function for
	  chan_oss (issue #8002 reported by Mithraen with minor mods by
	  moi)

	* channels/chan_mgcp.c: Clean up chan_mgcp's module load function
	  (issue #8001 reported by Mithraen with mods by moi)

2006-09-21 21:21 +0000 [r43450]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile, build_tools/strip_nonapi (added): add another
	  attempt to strip non-API symbols from the final binary... script
	  will need to be extended to work on non-Linux systems

2006-09-21 20:22 +0000 [r43410-43445]  Tilghman Lesher <tilghman@mail.jeffandtilghman.com>

	* apps/app_url.c: Fix documentation to reflect how Url() really
	  works

	* cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates

2006-09-21  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta2 released.

2006-09-21 16:08 +0000 [r43404-43405]  Kevin P. Fleming <kpfleming@digium.com>

	* main/Makefile: remove this change... it requires binutils 2.17

2006-09-20 23:19 +0000 [r43396]  Jason Parker <jparker@digium.com>

	* build_tools/make_version: fix minor typo in the way version is
	  handled

2006-09-20  Kevin P. Fleming  <kpfleming@digium.com>

	* Asterisk 1.4.0-beta1 released.