[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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==============================================================================
--- 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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