[asterisk-commits] lmadsen: tag 1.8.0-beta3 r281571 - /tags/1.8.0-beta3/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Aug 10 12:51:35 CDT 2010


Author: lmadsen
Date: Tue Aug 10 12:51:30 2010
New Revision: 281571

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=281571
Log:
Importing files for 1.8.0-beta3 release.

Added:
    tags/1.8.0-beta3/.lastclean   (with props)
    tags/1.8.0-beta3/.version   (with props)
    tags/1.8.0-beta3/ChangeLog   (with props)

Added: tags/1.8.0-beta3/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-beta3/.lastclean?view=auto&rev=281571
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Added: tags/1.8.0-beta3/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-beta3/ChangeLog?view=auto&rev=281571
==============================================================================
--- tags/1.8.0-beta3/ChangeLog (added)
+++ tags/1.8.0-beta3/ChangeLog Tue Aug 10 12:51:30 2010
@@ -1,0 +1,22649 @@
+2010-08-10  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-beta3 Released.
+
+2010-08-10 17:48 +0000 [r281529-281568]  Russell Bryant <russell at digium.com>
+
+	* apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+	  (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+	  | 8 lines Reset visible indication after answer. (closes issue
+	  #17641) Reported by: klaus3000 Patches:
+	  ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+	  klaus3000 (license 65) Tested by: schmidts ........
+	  ................
+
+	* channels/chan_sip.c: Ensure that the proper external address is
+	  used for the RTP destination. (closes issue #17044) Reported by:
+	  ebroad Tested by: ebroad Review:
+	  https://reviewboard.asterisk.org/r/566/
+
+	* main/cli.c: Resolve a problem with channel name tab completion.
+	  Hitting tab without typing any part of a channel name resulted in
+	  no results. This now results in getting a full list of active
+	  channels, just as it did in previous versions of Asterisk.
+	  Review: https://reviewboard.asterisk.org/r/818/
+
+2010-08-10 07:26 +0000 [r281497]  TransNexus OSP Development <support at transnexus.com>
+
+	* apps/app_osplookup.c: Fixed the issue caused by EXTEN including
+	  user parameters.
+
+2010-08-09 23:04 +0000 [r281466]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_local.c: Add some more stuff to copy from 281429.
+
+2010-08-09 20:47 +0000 [r281432]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010)
+	  | 13 lines fixes SIP peers memory leak We zeroed out the peer's
+	  addr before it was removed from the peers_by_ip container. This
+	  made it impossible to be removed from the container as the addr
+	  is the key used by the container to find the peer. (closes issue
+	  #17774) Reported by: kkm Patches:
+	  017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888)
+	  017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888)
+	  ........
+
+2010-08-09 20:43 +0000 [r281429]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_local.c, /: Merged revisions 281391 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500
+	  (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010)
+	  | 13 lines Prevent loss of Caller ID information set on local
+	  channel after masquerade. Caller ID set on the channel before a
+	  masquerade occurs when using a local channel would cause the
+	  information to be lost. The problem was that the information was
+	  set on a channel destined to be hung up. The somewhat confusing
+	  fix is to detect if any Caller ID has been set on the channel and
+	  if so preswap the Caller ID data so that basically the masquerade
+	  puts the data back. (closes issue #17138) Reported by: kobaz
+	  Review: https://reviewboard.asterisk.org/r/847/ ........
+	  ................
+
+2010-08-09 14:49 +0000 [r281358]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax.c: Validate minrate, maxrate, and modem settings
+	  before attempting a fax session. FAX-224
+
+2010-08-09 14:31 +0000 [r281356]  <simon.perreault at viagenie.ca>
+
+	* configs/sip.conf.sample: Added comment about IPv4-mapped IPv6
+	  addresses and the output of netstat.
+
+2010-08-09 12:51 +0000 [r281294-281325]  Russell Bryant <russell at digium.com>
+
+	* configs/cdr.conf.sample: Add a couple of default values to the
+	  documentation of cdr.conf.
+
+	* configs/cdr.conf.sample: Reorder some options in cdr.conf.sample.
+	  Put all of the options that affect the contents of CDRs together,
+	  instead of having the batch mode options in the middle of them.
+
+2010-08-06 18:57 +0000 [r281085]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/utils.c: Fix alignment of stringfields on the SPARC
+	  architecture (closes issue #17789) Reported by: Ian Mason
+	  Patches: 20100806__issue17789__2.diff.txt uploaded by tilghman
+	  (license 14) Tested by: Ian_Mason
+
+2010-08-05 13:16 +0000 [r281052]  Russell Bryant <russell at digium.com>
+
+	* main/cdr.c, /: Merged revisions 281051 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010)
+	  | 9 lines Cleanup default option value handling for cdr.conf
+	  [general]. The default values would differ depending on whether
+	  or not cdr.conf exists. That is no longer the case. Apply a
+	  default value to the unanswered option. Define all default values
+	  as named constants. ........
+
+2010-08-05 07:46 +0000 [r280984]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/pbx.h, main/pbx.c, /: Merged revisions 280983
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500
+	  (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010)
+	  | 8 lines Change context lock back to a mutex, because
+	  functionality depends upon the lock being recursive. (closes
+	  issue #17643) Reported by: zerohalo Patches:
+	  20100726__issue17643.diff.txt uploaded by tilghman (license 14)
+	  Tested by: zerohalo ........ ................
+
+2010-08-04 15:11 +0000 [r280909]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax.c: Initialize FAXOPT() status variables in sendfax
+	  and receivefax instead of when the details structure is created.
+
+2010-08-04 14:04 +0000 [r280809-280879]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_mgcp.c: Check cur value before attempting a deref.
+	  (closes issue #17775) Reported by: svinson Patches:
+	  20100804__issue17775.diff.txt uploaded by tilghman (license 14)
+	  Tested by: svinson (closes issue #17743) Reported by: tgruenberg
+	  Patches: 20100804__issue17775.diff.txt uploaded by tilghman
+	  (license 14) Tested by: tgruenberg
+
+	* CHANGES, funcs/func_strings.c: Sneak FIELDNUM() into 1.8. Returns
+	  a 1-based index into a list of a specified item. Matches up with
+	  FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth
+	  Patches: svn-279754.diff uploaded by gareth (license 208) Tested
+	  by: gareth, tilghman Review:
+	  https://reviewboard.asterisk.org/r/810/
+
+2010-08-03 19:54 +0000 [r280777-280778]  <simon.perreault at viagenie.ca>
+
+	* channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes
+	  issue #17663) Reported by: oej Patches: diff uploaded by
+	  sperreault (license 252) diff2 uploaded by sperreault (license
+	  252) get_domain.diff uploaded by sperreault (license 252)
+
+	* configs/sip.conf.sample: Better documentation related to IPv6.
+	  (closes issue #17737) Reported by: oej Patches: doc.diff uploaded
+	  by sperreault (license 252) Tested by: mmichelson
+
+2010-08-03 18:48 +0000 [r280742]  Russell Bryant <russell at digium.com>
+
+	* addons/Makefile, addons/mp3 (removed),
+	  contrib/scripts/get_mp3_source.sh (added): Remove the MP3 decoder
+	  source code and replace it with a small shell script. Review:
+	  https://reviewboard.asterisk.org/r/836/
+
+2010-08-03 18:42 +0000 [r280624-280740]  Tilghman Lesher <tlesher at digium.com>
+
+	* doc/asterisk.sgml, /, doc/asterisk.8, doc/Makefile (added):
+	  Merged revisions 280739 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010)
+	  | 2 lines Document -B and -W flags and regenerate manpage from
+	  sgml ........
+
+	* apps/app_voicemail.c, /: Merged revisions 280671 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02
+	  Aug 2010) | 2 lines Allow the pipe, but also allow the comma
+	  ........
+
+	* main/Makefile: Make this a little more deterministic... we want
+	  the latest value, not just a 1 somewhere.
+
+	* main/Makefile: Apparently, the values in makeopts are sometimes
+	  1:1 and sometimes 1. Compensate for this.
+
+2010-07-29 21:07 +0000 [r280557]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_fax.c: Fix regression introduced in r1664. Give the fax
+	  stack time to shutdown and populate the FAXOPT output variables.
+	  FAX-222
+
+2010-07-29 20:43 +0000 [r280552]  David Vossel <dvossel at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 280551 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010)
+	  | 11 lines fixes wrong SRV query for TLS connection (closes issue
+	  #17612) Reported by: marcelloceschia Patches:
+	  chan-sip_srvQuery.patch uploaded by marcelloceschia (license
+	  1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907)
+	  chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia
+	  (license 1079) Tested by: marcelloceschia, st, pabelanger
+	  ........
+
+2010-07-29 20:35 +0000 [r280549]  Russell Bryant <russell at digium.com>
+
+	* configs/ccss.conf.sample: Add header to ccss.conf to appease oej.
+	  (closes issue #17755) Reported by: oej
+
+2010-07-29 19:47 +0000 [r280519]  Sean Bright <sean at malleable.com>
+
+	* channels/sig_pri.c: Fix compilation error in chan_dahdi (strdupa
+	  -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches:
+	  strdupa_oops.diff uploaded by malcolmd (license 924)
+
+2010-07-29 19:13 +0000 [r280450]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c, /: Merged revisions 280449 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500
+	  (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010)
+	  | 12 lines fixes issue with translator frame not getting freed A
+	  translator frame even if it local storage so the translation path
+	  can be freed. This issue prevented g729 licenses from being freed
+	  up. (closes issue #17630) Reported by: manvirr Patches:
+	  encoder_fix.diff uploaded by dvossel (license 671) Tested by:
+	  manvirr, dvossel ........ ................
+
+2010-07-29 18:37 +0000 [r280414-280446]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* tests/test_utils.c: Remove res_crypto dependency.
+
+	* tests/test_utils.c: crypto_loaded_test depends on res_crypto,
+	  else test will fail.
+
+2010-07-29 16:25 +0000 [r280391]  Russell Bryant <russell at digium.com>
+
+	* main/rtp_engine.c: Don't blow up if get_codec() was not provided
+	  in the RTP glue.
+
+2010-07-29 16:07 +0000 [r280346]  Jean Galarneau <jgalarneau at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 280345 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r280345 | jeang | 2010-07-29 11:01:35 -0500
+	  (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) |
+	  2 lines Fix a dsp structure leak occuring when a local channel is
+	  put into a meetme conference, then masquaraded away. ABE-2422
+	  ........ ................
+
+2010-07-29 15:57 +0000 [r280307-280343]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_usbradio.c: Use PRIx64 instead of PRId64 in format
+	  string. related to r280302
+
+	* main/channel.c, channels/chan_local.c, /: Merged revisions 280306
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul
+	  2010) | 2 lines Implement support for ast_channel_queryoption on
+	  local channels. Currently only AST_OPTION_T38_STATE is supported.
+	  ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........
+	  Additionally, pass AST_CONTROL_T38_PARAMETERS control frames
+	  through generic bridges. This change appears to have been
+	  unintentionally left out of rev 203699.
+
+2010-07-29 00:45 +0000 [r280302]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_usbradio.c: Use PRId64 with format_t
+
+2010-07-28 20:49 +0000 [r280269]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/sip/reqresp_parser.c: Give test category missing leading
+	  slash
+
+2010-07-28 20:12 +0000 [r280235]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 280229 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28
+	  Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7
+	  called_nai and calling_nai config options. ........
+
+2010-07-28 20:03 +0000 [r280233]  Jason Parker <jparker at digium.com>
+
+	* sounds/Makefile, /: Merged revisions 280231 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) |
+	  6 lines Work around some silly behavior on BSD. A non-zero exit
+	  from a subshell should make the build fail. (closes issue #17621)
+	  ........
+
+2010-07-28 19:34 +0000 [r280225]  Terry Wilson <twilson at digium.com>
+
+	* res/res_rtp_asterisk.c: Do rtp/rtcp debugging when it is turned
+	  on w/o filtering
+
+2010-07-28 18:24 +0000 [r280195]  Jason Parker <jparker at digium.com>
+
+	* sounds/Makefile, /: Merged revisions 280193 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) |
+	  9 lines Remove unnecessary subshells. Attempt to make
+	  checksumming work. Also improves readability. (issue #17621)
+	  Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/
+	  ........
+
+2010-07-28 16:52 +0000 [r280161]  Sean Bright <sean at malleable.com>
+
+	* apps/app_queue.c, /: Merged revisions 280160 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul
+	  2010) | 8 lines Plug a reference leak in app_queue when adding
+	  members dynamically. (closes issue #17738) Reported by:
+	  bobwienholt Patches: issue17738.patch uploaded by bobwienholt
+	  (license 950) Tested by: bobwienholt, seanbright ........
+
+2010-07-28 13:52 +0000 [r280090]  Leif Madsen <lmadsen at digium.com>
+
+	* contrib/scripts/live_ast, /: Merged revisions 280089 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500
+	  (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28
+	  Jul 2010) | 1 line Update help text to be less confusing.
+	  ........ ................
+
+2010-07-28 13:01 +0000 [r280058]  Russell Bryant <russell at digium.com>
+
+	* res/res_crypto.c: s/init keys/keys init/
+
+2010-07-28 01:37 +0000 [r280023]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_usbradio.c: Resolve compiler warning about
+	  formatting (closes issue #17732) Reported by: pabelanger
+
+2010-07-27 22:30 +0000 [r280019-280020]  Sean Bright <sean at malleable.com>
+
+	* main/editline/el.h, main/term.c, main/cli.c,
+	  main/editline/parse.c, main/editline/tokenizer.c,
+	  main/editline/config.sub, main/editline/parse.h,
+	  main/editline/tokenizer.h, configure, main/editline/histedit.h,
+	  main/editline/sig.c, main/editline/PLATFORMS,
+	  main/editline/sig.h, main/editline/key.c, main/editline/editrc.5,
+	  main/editline/np/fgetln.c, main/editline/key.h,
+	  main/editline/TEST/test.c, main/Makefile,
+	  main/editline/configure, main/editline/Makefile.in, configure.ac,
+	  main/editline/configure.in, main/editline/readline/readline.h,
+	  main/editline/README, main/editline/editline.3,
+	  main/editline/vi.c, main/editline/sys.h, main/editline/emacs.c,
+	  main/asterisk.c, main/editline/install-sh, main/editline/term.c,
+	  main/editline/config.guess, main/editline/read.c,
+	  main/editline/term.h, main/editline/map.c,
+	  main/editline/np/strlcpy.c, main/editline (added),
+	  main/editline/config.h.in, main/editline/read.h,
+	  main/editline/tty.c, main/editline/np/unvis.c,
+	  main/editline/prompt.c, main/editline/map.h, main/editline/tty.h,
+	  main/editline/chared.c, main/editline/prompt.h,
+	  main/editline/np/strlcat.c, main/editline/chared.h,
+	  main/editline/np, main/editline/TEST, main/editline/refresh.c,
+	  main/editline/history.c, main/editline/readline,
+	  include/asterisk/term.h, main/editline/refresh.h,
+	  main/editline/search.c, main/editline/hist.c,
+	  main/editline/search.h, main/editline/hist.h,
+	  main/editline/np/vis.c, build_tools/menuselect-deps.in, main,
+	  main/editline/readline.c, main/editline/np/vis.h,
+	  main/editline/INSTALL, makeopts.in, main/editline/CHANGES,
+	  main/editline/common.c, main/xmldoc.c, main/editline/makelist.in,
+	  include/asterisk/autoconfig.h.in, main/editline/el.c: Revert
+	  r280019 for now - This was poorly executed.
+
+	* include/asterisk/term.h, makeopts.in, main/asterisk.c,
+	  main/xmldoc.c, main/cli.c, main/term.c, main/editline (removed),
+	  build_tools/menuselect-deps.in, configure,
+	  include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+	  main: Add ability to use system libedit and update bundled
+	  libedit. The version of libedit that is bundled with asterisk is
+	  old and has some bugs. This patch updates the bundled version of
+	  libedit within asterisk, and also updates asterisk to use the
+	  system libedit instead if one is available (and pkg-config is
+	  available). This review integrates several patches from other
+	  users specifically kkm and tzafrir. (closes issue #15929)
+	  Reported by: kkm Patches: 015929-astcli-editrc-trunk.240324.diff
+	  uploaded by kkm (license 888) (issue #16858) Reported by:
+	  jw-asterisk (closes issue #17039) Reported by: tzafrir Patches:
+	  0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir
+	  (license 46) Review: https://reviewboard.asterisk.org/r/807/
+
+2010-07-27 21:16 +0000 [r279953]  Russell Bryant <russell at digium.com>
+
+	* res/ais, main/db1-ast/mpool, Makefile.rules, res/snmp, cdr,
+	  formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
+	  main/db1-ast/btree, configure, main/editline, codecs/g722, main,
+	  main/db1-ast/recno, channels/sip, makeopts.in, pbx, res, res/ael,
+	  channels, main/stdtime, main/editline/np, codecs, utils,
+	  main/db1-ast/hash, cel, apps, configure.ac, main/db1-ast/db: Add
+	  --enable-coverage option to configure script. This option enables
+	  the proper compiler flags for tracking code coverage, which is
+	  useful along side automated testing.
+
+2010-07-27 20:57 +0000 [r279949]  David Vossel <dvossel at digium.com>
+
+	* main/audiohook.c, main/channel.c, /,
+	  include/asterisk/audiohook.h: Merged revisions 279946 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500
+	  (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010)
+	  | 19 lines remove empty audiohook write list on channel If a
+	  channel has an audiohook write list created on it, that list
+	  stays on the channel until the channel is destroyed. There is no
+	  reason to keep that list on the channel if it becomes empty. If
+	  it is empty that just means we are doing needless translating for
+	  every ast_read and ast_write. This patch removes the audiohook
+	  list from the channel once it is detected to be empty on either a
+	  read or write. If a audiohook is added back to the channel after
+	  this list is destroyed, the list just gets recreated as if it
+	  never existed to begin with. (closes issue #17630) Reported by:
+	  manvirr Review: https://reviewboard.asterisk.org/r/799/ ........
+	  ................
+
+2010-07-27 19:50 +0000 [r279916]  Russell Bryant <russell at digium.com>
+
+	* channels/sig_pri.c, channels/chan_dahdi.c: Fix inband DTMF
+	  detection on outgoing ISDN calls. This is a regression from the
+	  sig_pri split from chan_dahdi. When a call is first initiated,
+	  the inband DTMF detector is not enabled if it's an outgoing ISDN
+	  call. However, it needs to be turned on once the media path
+	  starts up. This handling was put back in the open_media()
+	  callback of chan_dahdi. In sig_pri, open_media() calls were added
+	  to a few places where it was needed, including handling of
+	  PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING.
+	  Thanks to rmudgett for helping me with the patch!
+
+2010-07-27 18:54 +0000 [r279887]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fix parsing error in sip_sipredirect(). The
+	  code was written in a way that did a bad job of parsing the port
+	  out of a URI. Specifically, it would do badly when dealing with
+	  an IPv6 address. In this particular scenario, there was no value
+	  from parsing the port out, so I just removed that logic. And
+	  while I was messing around in the function, I changed some
+	  variable names to be more descriptive. (closes issue #17661)
+	  Reported by: oej Patches: 17661.diff uploaded by mmichelson
+	  (license 60)
+
+2010-07-27 16:40 +0000 [r279850]  Jason Parker <jparker at digium.com>
+
+	* sounds/Makefile, /: Merged revisions 279849 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul 2010) |
+	  1 line Simply sounds/Makefile some more. ........
+
+2010-07-27 16:09 +0000 [r279817]  David Vossel <dvossel at digium.com>
+
+	* main/netsock2.c, channels/chan_sip.c: fix sip transaction match
+	  with authentication, fix confusing log message when using
+	  getaddrinfo
+
+2010-07-27 16:06 +0000 [r279815]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_dahdi.c: Support "channels" in addition to
+	  "channel" in chan_dahdi.conf. Review:
+	  https://reviewboard.asterisk.org/r/804
+
+2010-07-27 15:15 +0000 [r279785]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 279784 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul
+	  2010) | 14 lines Fix bad behavior of dynamic_exclude_static
+	  option in sip.conf. We were attempting to create a contactdeny
+	  rule based on the peer's IP address before the peer's IP address
+	  had been set. By moving the processing further down in the
+	  function, we can ensure stuff works as we expect for it to.
+	  (closes issue #17717) Reported by: mmichelson Patches:
+	  17717.patch uploaded by mmichelson (license 60) Tested by:
+	  DennisD ........
+
+2010-07-27 02:57 +0000 [r279726-279755]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_dahdi.c: If dringXcontext is null, fallback to
+	  default context value. (closes issue #17693) Reported by:
+	  iasgoscouk Patches: issue17693.patch uploaded by pabelanger
+	  (license 224) Tested by: iasgoscouk Review:
+	  https://reviewboard.asterisk.org/r/803/
+
+	* main/http.c: Use ast_sockaddr_setnull() when http is not enabled.
+	  Otherwise, ast_tcptls_server_start() will still start http.
+	  (closes issue #17708) Reported by: pabelanger Patches: http.patch
+	  uploaded by pabelanger (license 224)
+
+2010-07-26  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-beta2 Released.
+
+2010-07-26 23:29 +0000 [r279689]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* UPGRADE.txt, CHANGES: Updated documentation for FAX logger level.
+
+2010-07-26 23:03 +0000 [r279658]  Jason Parker <jparker at digium.com>
+
+	* sounds/Makefile (added), /, sounds/Makefile.380 (removed),
+	  configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+	  (removed), configure.ac: Merged revisions 279657 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul
+	  2010) | 5 lines Really fix sounds Makefile (and make it
+	  readableish). There was a rather large syntax error that should
+	  have caused ALL versions of GNU make to fail. I don't know how it
+	  worked. ........
+
+2010-07-26 21:53 +0000 [r279636]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: Ignore a control subclass of -1 in
+	  ast_waitfordigit_full().
+
+2010-07-26 21:20 +0000 [r279599-279619]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, configure, configure.ac: Merged revisions 279609 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26
+	  Jul 2010) | 2 lines Dunno why this worked on my machine, but it
+	  works better this way. ........
+
+	* res/res_config_ldap.c, /: Merged revisions 279597 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26
+	  Jul 2010) | 13 lines Apply all patches in:
+	  https://issues.asterisk.org/view.php?id=13573 (closes issue
+	  #13573) Reported by: navkumar Patches:
+	  res_config_ldap-category.diff uploaded by navkumar (license 580)
+	  res_config_ldap.patch uploaded by bencer (license 961)
+	  res_config_ldap uploaded by bencer (license 961) Tested by:
+	  suretec ........
+
+	* /: Reverting property remove
+
+2010-07-26 20:58 +0000 [r279598]  Gavin Henry <ghenry at suretecsystems.com>
+
+	* /: Merged revisions 279597 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/1.6.2
+	  -----------------------------------------------------------------------
+	  r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) |
+	  13 lines Apply all patches in:
+	  https://issues.asterisk.org/view.php?id=13573 [^] (closes issue
+	  0013573) Reported by: navkumar Patches:
+	  res_config_ldap-category.diff uploaded by navkumar (license 580)
+	  res_config_ldap.patch uploaded by bencer (license 961)
+	  res_config_ldap uploaded by bencer (license 961) Tested by:
+	  suretec
+	  ------------------------------------------------------------------------
+
+2010-07-26 19:59 +0000 [r279568]  David Vossel <dvossel at digium.com>
+
+	* channels/sip/include/sip.h,
+	  channels/sip/include/reqresp_parser.h, channels/chan_sip.c,
+	  channels/sip/reqresp_parser.c: transaction matching using top
+	  most Via header This patch modifies the way chan_sip.c does
+	  transaction to dialog matching. Asterisk now stores information
+	  in the top most Via header of the initial incoming request and
+	  compares that against other Requests that have the same call-id.
+	  This results in Asterisk being able to detect a forked call in
+	  which it has received multiple legs of the fork. I completely
+	  stripped out the previous matching code and made the comparisons
+	  a little more explicit and easier to understand. My comments in
+	  the code should offer all the details involving this patch. This
+	  patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to
+	  find multiple dialogs with the same call-id. Since the callback
+	  function was returning (CMP_MATCH | CMP_STOP) only the first item
+	  found was being returned. I fixed this by making a new callback
+	  function for finding multiple dialogs that only returns
+	  (CMP_MATCH) on a match allowing for multiple items to be
+	  returned. Review: https://reviewboard.asterisk.org/r/776/
+
+2010-07-26 19:51 +0000 [r279566]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* UPGRADE.txt, CHANGES, configs/logger.conf.sample: Add
+	  documentation for FAX logger level. (closes issue #17715)
+	  Reported by: vrban Patches: 17715.patch uploaded by pabelanger
+	  (license 224) Tested by: vrban
+
+2010-07-26 19:18 +0000 [r279562]  Tilghman Lesher <tlesher at digium.com>
+
+	* sounds/Makefile (removed), /, sounds/Makefile.380 (added),
+	  configure, include/asterisk/autoconfig.h.in, sounds/Makefile.381
+	  (added), configure.ac: Merged revisions 279561 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010)
+	  | 2 lines Use a special Makefile for noobs who still have GNU
+	  Make 3.80. ........
+
+2010-07-26 16:04 +0000 [r279504]  Mark Michelson <mmichelson at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  channels/sip/reqresp_parser.c: Allow for systems without locale
+	  support to be usable. A recent change to SIP URI comparison code
+	  added a locale-specific string comparison to the mix, and certain
+	  systems do not support such functions. This fix allows for those
+	  systems to still use Asterisk 1.8 (closes issue #17697) Reported
+	  by: pprindeville Patches: asterisk-trunk-bugid17697.patch
+	  uploaded by pprindeville (license 347) Tested by: mmichelson
+
+2010-07-26 15:43 +0000 [r279502]  Sean Bright <sean at malleable.com>
+
+	* autoconf/ast_ext_lib.m4, /: Merged revisions 279501 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ........ r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon,
+	  26 Jul 2010) | 5 lines Expand the correct value within
+	  AST_OPTION_ONLY. (closes issue #17703) Reported by: stuarth
+	  ........
+
+2010-07-26 03:27 +0000 [r279472]  Tilghman Lesher <tlesher at digium.com>
+
+	* formats/format_sln16.c, formats/format_wav_gsm.c,
+	  formats/format_siren7.c, formats/format_ilbc.c,
+	  formats/format_vox.c, formats/format_pcm.c,
+	  formats/format_h263.c, formats/format_g723.c,
+	  formats/format_h264.c, formats/format_g726.c,
+	  formats/format_jpeg.c, formats/format_siren14.c,
+	  formats/format_gsm.c, formats/format_g719.c,
+	  formats/format_g729.c, formats/format_sln.c,
+	  formats/format_wav.c, formats/format_ogg_vorbis.c: Formats need
+	  to load before apps, because some apps call
+	  ast_format_str_reduce() at load time.
+
+2010-07-25 21:26 +0000 [r279442]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* tests/test_func_file.c: Add trailing backslash to silence warning
+	  message.
+
+2010-07-25 18:21 +0000 [r279390-279410]  Tilghman Lesher <tlesher at digium.com>
+
+	* cdr/cdr_odbc.c: Don't re-register CDR module on reload. (closes
+	  issue #17304) Reported by: jnemeth Patches:
+	  20100507__issue17304.diff.txt uploaded by tilghman (license 14)
+	  Tested by: jnemeth
+
+	* main/logger.c: Don't assume qlog is open. (closes issue #17704)
+	  Reported by: vrban Patches: issue17704.patch uploaded by
+	  pabelanger (license 224) Tested by: vrban
+
+2010-07-24 23:58 +0000 [r279348]  Bradley Latus <brad.latus at gmail.com>
+
+	* doc/asterisk.8: Minor update to man page
+
+2010-07-24 20:47 +0000 [r279273-279314]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* Makefile: Remove duplicate -c flag when using $(INSTALL) (closes
+	  issue #17695) Reported by: pabelanger Patches: Makefile.diff
+	  uploaded by pabelanger (license 224)
+
+	* include/asterisk/netsock2.h: Check if ast_sockaddr is NULL then
+	  return. (closes issue #17677) Reported by: outcast Patches:
+	  issue0017677.patch uploaded by pabelanger (license 224) Tested
+	  by: elguero
+
+	* main/manager.c: Default sin_family to AF_INET for TCP / TLS
+	  Bindaddress. Otherwise, 'manager show settings' will generate
+	  errors if manager is not enabled.
+
+2010-07-23 22:20 +0000 [r279227]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_queue.c, apps/app_dial.c, /: Merged revisions 279207 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+	  ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500
+	  (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010)
+	  | 7 lines SIP promiscuous redirect could fail to dial the
+	  redirect. The ast_channel was created with one variable to
+	  ast_request() but the call to ast_call() that initiates the
+	  outgoing call was using a different variable. The two variables
+	  are not equivalent if the call_forward string included a channel
+	  technology specifier. e.g., SIP/200 ........ ................
+
+2010-07-12  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.8.0-beta1 Released.
+
+2010-07-23 18:56 +0000 [r279113]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_odbc.c: Silly 64-bit compilers (who uses 64-bit anyway?)
+
+2010-07-23 18:23 +0000 [r279056-279094]  Russell Bryant <russell at digium.com>
+
+	* /: fix up properties on 1.8 branch
+
+	* / (added): Create a branch for Asterisk 1.8.
+
+	  ___      _            _     _      _   ___
+	 / _ \ ___| |_ ___ _ __(_)___| | __ / | ( _ )
+	| |_| / __| __/ _ \ '__| / __| |/ / | | / _ \
+	|  _  \__ \ ||  __/ |  | \__ \   <  | || (_) |
+	|_| |_|___/\__\___|_|  |_|___/_|\_\ |_(_)___/
+
+2010-07-23 17:05 +0000 [r278982-278985]  Tilghman Lesher <tlesher at digium.com>
+
+	* autoconf/ast_check_pwlib.m4, /, configure, configure.ac: Merged
+	  revisions 278984 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278984 | tilghman | 2010-07-23 12:04:15 -0500 (Fri, 23 Jul 2010)
+	  | 5 lines Establish a maximum version for openh323 (i.e. not
+	  opal), because chan_h323 will fail to load, even if it links.
+	  (issue #17679) Reported by: am ........
+
+	* /, main/asterisk.c: Merged revisions 278981 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r278981 | tilghman | 2010-07-23 11:42:25 -0500 (Fri, 23 Jul 2010)
+	  | 8 lines Avoid race with consolethread on shutdown (on parallel
+	  processors). (closes issue #17080) Reported by: sybasesql
+	  Patches: 20100721__issue17080.diff.txt uploaded by tilghman
+	  (license 14) Tested by: sybasesql ........
+
+2010-07-23 16:33 +0000 [r278980]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c, channels/sip/reqresp_parser.c,
+	  channels/sip/include/reqresp_parser.h: SIP URI comparison fixes.
+	  This initially was created to work around the issue of using a
+	  string comparison instead of a binary comparison for IP
+	  addresses. It evolved a bit when test cases were created and it
+	  was discovered that comparison of URI parameters was not working
+	  exactly as it should. sip_uri_cmp() and its helpers have been
+	  moved to reqresp_parser.c and a new test has been added. (closes
+	  issue #17662) Reported by: oej Review:
+	  https://reviewboard.asterisk.org/r/792
+
+2010-07-23 16:19 +0000 [r278957]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/res_odbc.h, res/res_config_odbc.c,
+	  configs/extconfig.conf.sample, CHANGES, main/config.c,
+	  res/res_odbc.c, configs/res_odbc.conf.sample: Merge the realtime
+	  failover branch
+
+2010-07-23 16:07 +0000 [r278947]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* doc/asterisk.8: Some left-over hyphen-minus fixes in the man page
+
+2010-07-23 15:57 +0000 [r278944-278945]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: ... just kidding. Enable SIP by default. :-)
+
+	* channels/chan_sip.c: Disable SIP support by default for Asterisk
+	  1.8.
+
+2010-07-23 15:52 +0000 [r278943]  Mark Michelson <mmichelson at digium.com>
+
+	* addons/chan_ooh323.c: Well, who knew chan_ooh323 used udptl? I
+	  sure didn't!
+
+2010-07-23 15:41 +0000 [r278942]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
+	  Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
+
+2010-07-23 15:16 +0000 [r278908]  Mark Michelson <mmichelson at digium.com>
+
+	* main/udptl.c, channels/chan_sip.c, include/asterisk/udptl.h,
+	  channels/sip/include/sip.h: Allow IPv6 addresses for UDPTL
+	  streams. Review: https://reviewboard.asterisk.org/r/795
+
+2010-07-23 13:37 +0000 [r278875]  Olle Johansson <oej at edvina.net>
+
+	* res/res_config_ldap.c: Minor corrections to the LDAP realtime
+	  driver Review: https://reviewboard.asterisk.org/r/798/ Thanks
+	  Mark for a quick review!
+
+2010-07-23 13:26 +0000 [r278873]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* Makefile, agi/Makefile, sounds/Makefile: Portability updates for
+	  Makefiles. When possible, use $(INSTALL). This allows us to use
+	  the functionality within install for setting directory / file
+	  permissions, a requirement for unprivileged installation. Also
+	  move any directory we plan to create within the installdirs
+	  macro. Plus various other formatting issues. (issue #17436)
+	  Reported by: pabelanger Patches: non-root.patch.v8 uploaded by
+	  pabelanger (license 224) Tested by: pabelanger Review:
+	  https://reviewboard.asterisk.org/r/654/
+
+2010-07-23 11:01 +0000 [r278809-278841]  Alec L Davis <sivad.a at paradise.net.nz>
+

[... 21868 lines stripped ...]



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