[asterisk-commits] lmadsen: tag 1.6.2.10-rc1 r272969 - /tags/1.6.2.10-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 29 10:32:40 CDT 2010
Author: lmadsen
Date: Tue Jun 29 10:32:36 2010
New Revision: 272969
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=272969
Log:
Importing files for 1.6.2.10-rc1 release.
Added:
tags/1.6.2.10-rc1/.lastclean (with props)
tags/1.6.2.10-rc1/.version (with props)
tags/1.6.2.10-rc1/ChangeLog (with props)
Added: tags/1.6.2.10-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.10-rc1/.lastclean?view=auto&rev=272969
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--- tags/1.6.2.10-rc1/ChangeLog (added)
+++ tags/1.6.2.10-rc1/ChangeLog Tue Jun 29 10:32:36 2010
@@ -1,0 +1,25424 @@
+2010-06-29 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.6.2.10-rc1
+
+2010-06-28 21:51 +0000 [r272924-272927] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 272926 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272926 | tilghman | 2010-06-28 16:50:57 -0500 (Mon, 28 Jun 2010)
+ | 15 lines Merged revisions 272925 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Don't change ownership/group/permissions on run
+ directory, if it already exists. (closes issue #17076) Reported
+ by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+ tilghman (license 14) Tested by: stuarth ........
+ ................
+
+ * main/config.c, /: Merged revisions 272923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272923 | tilghman | 2010-06-28 16:42:52 -0500 (Mon, 28 Jun 2010)
+ | 19 lines Merged revisions 272921-272922 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010)
+ | 8 lines Change the way that we read include files, to
+ accommodate for changes in GCC 4.4. (closes issue #17472)
+ Reported by: seandarcy Patches: config2.patch uploaded by nivan
+ (license 1066) Tested by: nivan ........ r272922 | tilghman |
+ 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines Also trim
+ trailing blanks on #includes ........ ................
+
+2010-06-28 18:50 +0000 [r272882] Russell Bryant <russell at digium.com>
+
+ * tests/test_astobj2.c (added): Backport applicable parts of
+ test_astobj2 from trunk.
+
+2010-06-28 17:37 +0000 [r272806] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 272805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272805 | mmichelson | 2010-06-28 12:33:12 -0500 (Mon, 28 Jun
+ 2010) | 11 lines Merged revisions 272804 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun
+ 2010) | 5 lines Decode URI in contact header of 302 response.
+ ABE-2352 ........ ................
+
+2010-06-28 15:36 +0000 [r272685-272686] Russell Bryant <russell at digium.com>
+
+ * doc/tex/chan-mobile.tex (removed): remove accidentally added
+ file.
+
+ * doc/tex/cdrdriver.tex, doc/tex/asterisk.tex, /,
+ doc/tex/chan-mobile.tex (added): Merged revisions 272684 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r272684 | russell | 2010-06-28 10:33:32 -0500 (Mon, 28
+ Jun 2010) | 2 lines Use the underscore package so that
+ underscores do not need to be escaped. ........
+
+2010-06-25 20:20 +0000 [r272556-272577] Tilghman Lesher <tlesher at digium.com>
+
+ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 272568 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r272568 | tilghman | 2010-06-25 15:18:47 -0500
+ (Fri, 25 Jun 2010) | 12 lines Merged revisions 272562 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010)
+ | 5 lines Make the structure of the table specified before match
+ the queries and results. (closes issue #17557) Reported by: cmaj
+ ........ ................
+
+ * sounds/Makefile, /: Merged revisions 272533 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272533 |
+ tilghman | 2010-06-25 14:17:16 -0500 (Fri, 25 Jun 2010) | 2 lines
+ Symlink sounds files, to save disk space, when multiple
+ tarballs/checkouts are on the same system. ........
+
+2010-06-25 18:58 +0000 [r272531] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/_private.h, tests/test_sched.c, main/asterisk.c,
+ include/asterisk/test.h (added), build_tools/cflags-devmode.xml,
+ tests/test_heap.c, tests/test_skel.c, main/Makefile, main/test.c
+ (added): Backport unit test API from trunk. Also, update existing
+ test modules that were already in this branch but had been
+ converted to the unit test API in trunk. Review:
+ https://reviewboard.asterisk.org/r/748/
+
+2010-06-24 22:19 +0000 [r272459] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 272447 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r272447 | rmudgett | 2010-06-24 17:11:26 -0500
+ (Thu, 24 Jun 2010) | 17 lines Merged revisions 272446 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010)
+ | 10 lines ss_thread calls pri_grab without lock during overlap
+ dial Recent changes to chan_dahdi with relation to overlap
+ dialing call pri_grab without first obtaining a lock. (closes
+ issue #17414) Reported by: pdf Patches: bug17414.patch uploaded
+ by jpeeler (license 325) ........ ................
+
+2010-06-23 23:40 +0000 [r272440] Terry Wilson <twilson at digium.com>
+
+ * autoconf/ast_ext_tool_check.m4, /, configure: Merged revisions
+ 272254,272256 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272254 |
+ twilson | 2010-06-23 15:53:48 -0500 (Wed, 23 Jun 2010) | 10 lines
+ Honor the --with-${library}=path for AST_EXT_TOOL_CHECK (closes
+ issue #16991) Reported by: pprindeville Patches:
+ with_netsnmp.patch.txt uploaded by twilson (license 396) Tested
+ by: twilson Review: https://reviewboard.asterisk.org/r/739/
+ ........ r272256 | twilson | 2010-06-23 15:59:17 -0500 (Wed, 23
+ Jun 2010) | 2 lines Update configure when changing autconf m4
+ files... ........
+
+2010-06-23 23:14 +0000 [r272371] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 272370 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r272370 | russell | 2010-06-23 18:09:28 -0500 (Wed, 23 Jun 2010)
+ | 23 lines Resolve some errors produced during module unload of
+ chan_iax2. The external test suite stops Asterisk using the "core
+ stop gracefully" command. The logs from the tests show that there
+ are a number of problems with Asterisk trying to cleanly shut
+ down. This patch addresses the following type of error that comes
+ from chan_iax2: [Jun 22 16:58:11] ERROR[29884]: lock.c:129
+ __ast_pthread_mutex_destroy: chan_iax2.c line 11371
+ (iax2_process_thread_cleanup): Error destroying mutex
+ &thread->lock: Device or resource busy For an example in the
+ context of a build, see:
+ http://bamboo.asterisk.org/browse/AST-TRUNK-739/log The primary
+ purpose of this patch is to change the thread pool shutdown
+ procedure to be more explicit to ensure that the thread exits
+ from a point where it is not holding a lock. While testing that,
+ I encountered various crashes due to the order of operations in
+ unload_module() being problematic. I reordered some things there,
+ as well. Review: https://reviewboard.asterisk.org/r/736/ ........
+
+2010-06-23 22:37 +0000 [r272369] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 272368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun
+ 2010) | 16 lines Merged revisions 272367 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 This version
+ of the patch only adds AgentComplete for attended transfers. It
+ was already present for blind transfers. ........ r272367 |
+ mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8
+ lines Send AgentComplete manager events in the event of blind and
+ attended transfers. (closes issue #16819) Reported by: elbriga
+ Patches: app_queue.diff uploaded by elbriga (license 482)
+ ........ ................
+
+2010-06-23 21:54 +0000 [r272333] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 272332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r272332 | tilghman | 2010-06-23 16:53:49 -0500 (Wed, 23 Jun 2010)
+ | 8 lines If there is realtime configuration, it does not get
+ re-read on reload unless the config file also changes. (closes
+ issue #16982) Reported by: dmitri Patches: res_musiconhold.patch
+ uploaded by dmitri (license 1001) Tested by: atis ........
+
+2010-06-23 21:15 +0000 [r272263] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_meetme.c: Revert previous commit, ast_test_flag64 does
+ not exist in 1.6.2
+
+2010-06-23 21:09 +0000 [r272262] Tilghman Lesher <tlesher at digium.com>
+
+ * res/ael/ael.flex, /, res/ael/ael.tab.c, res/ael/ael.y,
+ res/ael/ael_lex.c: Merged revisions 272260 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272260 |
+ tilghman | 2010-06-23 16:06:40 -0500 (Wed, 23 Jun 2010) | 8 lines
+ Ensure a NULL file while debugging cannot crash AEL. (closes
+ issue #17215) Reported by: vazir Patches:
+ 20100518__issue17215.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+2010-06-23 21:07 +0000 [r272253-272261] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * /, apps/app_meetme.c: Merged revisions 272259 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 |
+ pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2
+ lines Fix previous merge. ast_test_flag != ast_test_flag64
+ ........
+
+ * /, apps/app_meetme.c: Merged revisions 272257 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun
+ 2010) | 19 lines Merged revisions 272255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun
+ 2010) | 12 lines First caller into a dynamic conference now enter
+ pin once. If MeetMe is configured to use dynamic conference
+ numbers, then the first caller (which creates the conference) had
+ to enter the PIN number twice. (closes issue #15878) Reported by:
+ shawkris Patches: issue15878.patch uploaded by pabelanger
+ (license 224) Tested by: pabelanger ........ ................
+
+ * main/manager.c, /: Merged revisions 272252 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272252 |
+ pabelanger | 2010-06-23 16:35:45 -0400 (Wed, 23 Jun 2010) | 8
+ lines Correct manager variable 'EventList' case. (closes issue
+ #17520) Reported by: kobaz Patches: manager.patch uploaded by
+ kobaz (license 834) Tested by: lmadsen ........
+
+2010-06-23 18:41 +0000 [r272124-272149] Terry Wilson <twilson at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 272146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 |
+ twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines
+ Don't start the sla thread unless we realy need it ........
+
+ * /, apps/app_meetme.c: Merged revisions 272109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 |
+ twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines
+ Make sure reload updates SLA config Even if there are no stations
+ or trunks defined, we need to start the sla thread to make sure
+ we get the reload event. Also, when doing a reload we need to
+ remove the existing trunks and stations or they end up hanging
+ around. (closes issue #16818) Reported by: mbonin Patches:
+ sla_reload.patch uploaded by twilson (license 396) Tested by:
+ twilson ........
+
+2010-06-22 22:14 +0000 [r272015] David Vossel <dvossel at digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 272014 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r272014 |
+ dvossel | 2010-06-22 17:11:50 -0500 (Tue, 22 Jun 2010) | 5 lines
+ fixes issue with 'dialplan remove extension blah' segfaulting
+ with tab completion (closes issue #17440) Reported by: kobaz
+ ........
+
+2010-06-22 17:37 +0000 [r271904] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 271903 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271903 | mnicholson | 2010-06-22 12:35:17 -0500 (Tue, 22 Jun
+ 2010) | 15 lines Merged revisions 271902 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Decrease the module ref count in sip_hangup when
+ SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the
+ ref count correct. (closes issue #16815) Reported by: rain
+ Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+ (modified) Tested by: rain ........ ................
+
+2010-06-22 16:30 +0000 [r271869] Russell Bryant <russell at digium.com>
+
+ * /, res/ais/clm.c, res/ais/evt.c: Merged revisions 271867 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r271867 | russell | 2010-06-22 11:28:03 -0500 (Tue, 22
+ Jun 2010) | 7 lines Resolve some errors that occur on a graceful
+ shutdown. Don't Finalize() if Initialize() did not succeed. This
+ resulted in an error about trying to Finalize() an invalid
+ handle. Also trim some trailing whitespace while in the area.
+ ........
+
+2010-06-22 15:49 +0000 [r271832] David Vossel <dvossel at digium.com>
+
+ * /, main/features.c: Merged revisions 271831 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r271831 |
+ dvossel | 2010-06-22 10:46:22 -0500 (Tue, 22 Jun 2010) | 10 lines
+ fixes attended transfer behavior when both transferee and
+ transferer hung up If both the transferer and transferee of a
+ attended transfer hangup before the new channel picks up, the new
+ channel should be hung up as well as it has no endpoint to talk
+ to. This mirrors the expected behavior used in 1.4. (closes issue
+ #17444) Reported by: corruptor ........
+
+2010-06-22 15:00 +0000 [r271691-271763] Matthew Nicholson <mnicholson at digium.com>
+
+ * configs/dundi.conf.sample, /, pbx/pbx_dundi.c: Merged revisions
+ 271762 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271762 | mnicholson | 2010-06-22 09:54:58 -0500 (Tue, 22 Jun
+ 2010) | 15 lines Merged revisions 271761 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun
+ 2010) | 9 lines Allow users to specify a port for dundi peers.
+ (closes issue #17056) Reported by: klaus3000 Patches:
+ dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+ Tested by: klaus3000 ........ ................
+
+ * include/asterisk/strings.h, configs/sip_notify.conf.sample, /,
+ channels/chan_sip.c: Merged revisions 271690 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun
+ 2010) | 18 lines Merged revisions 271689 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun
+ 2010) | 8 lines Modify chan_sip's packet generation api to
+ automatically calculate the Content-Length. This is done by
+ storing packet content in a buffer until it is actually time to
+ send the packet, at which time the size of the packet is
+ calculated. This change was made to ensure that the
+ Content-Length is always correct. (closes issue #17326) Reported
+ by: kenner Tested by: mnicholson, kenner Review:
+ https://reviewboard.asterisk.org/r/693/ ........ This change also
+ adds an ast_str_copy_string() function (similar to
+ ast_copy_string), that copies one ast_str into another, properly
+ handling embedded nulls. ................
+
+2010-06-21 20:48 +0000 [r271555] Jeff Peeler <jpeeler at digium.com>
+
+ * res/ael/pval.c, /: Merged revisions 271554 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271554 | jpeeler | 2010-06-21 15:46:53 -0500 (Mon, 21 Jun 2010)
+ | 14 lines Merged revisions 271552 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010)
+ | 7 lines Do not use sizeof to calculate size of a heap allocated
+ character array. Change left out from 271399. (closes issue
+ #16053) Reported by: diLLec ........ ................
+
+2010-06-18 21:33 +0000 [r271338-271484] Jeff Peeler <jpeeler at digium.com>
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c: Merged
+ revisions 271483 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010)
+ | 18 lines Merged revisions 271399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010)
+ | 11 lines Fix crash when parsing some heavily nested statements
+ in AEL on reload. Due to the recursion used when compiling AEL in
+ gen_prios, all the stack space was being consumed when parsing
+ some AEL that contained nesting 13 levels deep. Changing a few
+ large buffers to be heap allocated fixed the crash, although I
+ did not test how many more levels can now be safely used. (closes
+ issue #16053) Reported by: diLLec Tested by: jpeeler ........
+ ................
+
+ * channels/chan_dahdi.c, /: Merged revisions 269307 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r269307 | rmudgett | 2010-06-09 11:54:38 -0500 (Wed, 09 Jun 2010)
+ | 12 lines Eliminate deadlock potential in dahdi_fixup(). Calling
+ dahdi_indicate() within dahdi_fixup() while the owner pointers
+ are in a potentially inconsistent state is a potentially bad
+ thing in principle. However, calling dahdi_indicate() when the
+ channel private lock is already held can cause a deadlock if the
+ PRI lock is needed because dahdi_indicate() will also get the
+ channel private lock. The pri_grab() function assumes that the
+ channel private lock is held once to avoid deadlock. ........
+
+2010-06-17 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.9 Released.
+
+2010-06-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.9-rc3 Released.
+
+2010-06-10 Tilghman Lesher <tlesher at digium.com>
+
+ * Ensure signals are not blocked inside other signal handlers.
+
+ This eliminates the annoying <beep> on the console.
+
+ (closes issue 0017477)
+ Reported by: jvandal
+ Patches:
+ 20100610__issue17477.diff.txt uploaded by tilghman (license 14
+
+2010-06-09 Paul Belanger <paul.belanger at polybeacon.com>
+
+ * Fix Debian init script to not use -c.
+
+ When using the init script as-is currently, it could cause issues on Debian
+ such as high CPU usage. This fix has worked for several people so I'm
+ implementing the change. We now handle color displays properly.
+
+ (closes issue 0016784)
+ Reported by: pabelanger
+ Patches:
+ 20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+ Tested by: pabelanger, tilghman
+
+2010-06-07 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.9-rc2 Released.
+
+2010-06-07 Tilghman Lesher <tlesher at digium.com>
+
+ * Fix crash in DTMF detection.
+
+ What I did not originally see in my previous commit was that even
+ though the next digit could be detected before the previous was
+ considered ended, the detection of the next digit effectively ends
+ the detection of the previous. Therefore, the length moves in
+ lockstep with the digit, and no separate counter is needed for the
+ length alone.
+
+ (closes issue 0017371)
+ Reported by: alecdavis
+
+ (closes issue 0017474)
+ Reported by: kenner
+
+2010-06-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.9-rc1 Released.
+
+2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 266592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010)
+ | 18 lines Merged revisions 266585 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010)
+ | 11 lines Prevent CLI prompt from distorting output of lines
+ shorter than the prompt. Uses the VT100 method of clearing the
+ line from the cursor position to the end of the line: Esc-0K
+ (closes issue #17160) Reported by: coolmig Patches:
+ 20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+ Tested by: coolmig ........ ................
+
+2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * res/res_agi.c: Fix typo in documentation (closes issue #17395)
+ Reported by: pabelanger Patches: res_agi.c.patch uploaded by
+ pabelanger (license 224)
+
+2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500
+ (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29
+ May 2010) | 2 lines Reverting patch and reopening issue #16784,
+ as patch breaks color display. ........ ................
+
+2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 266337 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 |
+ tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line
+ Only report swap on platforms which can examine those statistics
+ ........
+
+2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 |
+ dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines
+ fixes crash when creation of UDPTL fails (closes issue #17264)
+ Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff
+ uploaded by dvossel (license 671)
+ issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel
+ (license 671) Tested by: falves11 ........
+
+2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher at digium.com>
+
+ * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged
+ revisions 266146 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010)
+ | 21 lines Merged revisions 266142 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010)
+ | 14 lines Use sigaction for signals which should persist past
+ the initial trigger, not signal. If you call signal() in a
+ Solaris signal handler, instead of just resetting the signal
+ handler, it causes the signal to refire, because the signal is
+ not marked as handled prior to the signal handler being called.
+ This effectively causes Solaris to immediately exceed the
+ threadstack in recursive signal handlers and crash. (closes issue
+ #17000) Reported by: rmcgilvr Patches:
+ 20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+ Tested by: rmcgilvr ........ ................
+
+2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 |
+ dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines
+ fixes failed SIP Directed pickup resulting in dead channel
+ (closes issue #17339) Reported by: one47 Patches:
+ sip_magic_pickup2 uploaded by one47 (license 23) Tested by:
+ one47, dvossel ........
+
+2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500
+ (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010)
+ | 7 lines Not finding rows in the DB does not rise to the level
+ of a warning. (closes issue #17062) Reported by: drookie Patches:
+ 20100525__issue17062.diff.txt uploaded by tilghman (license 14)
+ ........ ................
+
+ * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged
+ revisions 265894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 |
+ tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines
+ Construct socket name, according to the Postgres docs, and
+ document as such. (closes issue #17392) Reported by: dps Patches:
+ 20100525__issue17392.diff.txt uploaded by tilghman (license 14)
+ Tested by: dps ........
+
+2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Recorded merge of revisions 265842 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed,
+ 26 May 2010) | 9 lines Re-enable "always" option for videosupport
+ option in sip.conf. (closes issue #17016) Reported by: twilson
+ Patches: 17016.patch uploaded by mmichelson (license 60) Tested
+ by: devmod ........
+
+2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 |
+ tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines
+ Use configure to determine the prefixes and include directories
+ properly. This ensures cross-platform compatibility, even among
+ Linux distributions, which don't always put headers in the same
+ place. (closes issue #17391) Reported by: loloski ........
+
+2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 |
+ mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12
+ lines Properly use peer's outboundproxy for outbound REGISTERs.
+ The logic used in transmit_register to get the outboundproxy for
+ a peer was flawed since this value would be overridden shortly
+ afterwards when create_addr was called. In addition, this also
+ fixes some logic used when parsing users.conf so that the peer
+ name is placed in the internally-generated register string so
+ that an outboundproxy set in the Asterisk GUI will be used for
+ outbound REGISTERs. ........
+
+2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel at digium.com>
+
+ * channels/chan_dahdi.c: fixes build issue with zaptel (closes
+ issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff
+ uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May
+ 2010) | 15 lines Merged revisions 265610 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May
+ 2010) | 8 lines Don't mark the cdr records of unanswered queue
+ calls with "NOANSWER". This restores the behavior prior to
+ r258670. (closes issue #17334) Reported by: jvandal Patches:
+ queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+ by: aragon, jvandal ........ ................
+
+2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson at digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged
+ revisions 265320,265467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 |
+ twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines
+ Add the FullyBooted AMI event It is possible to connect to the
+ manager interface before all Asterisk modules are loaded. To
+ ensure that an application does not send AMI actions that might
+ require a module that has not yet loaded, the application can
+ listen for the FullyBooted manager event. It will be sent upon
+ connection if all modules have been loaded, or as soon as loading
+ is complete. The event: Event: FullyBooted Privilege: system,all
+ Status: Fully Booted Review:
+ https://reviewboard.asterisk.org/r/639/ ........ r265467 |
+ twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
+ Merge the rest of the FullyBooted patch ........
+
+2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon,
+ 24 May 2010) | 8 lines Print openh323 log to the Asterisk
+ console. (closes issue #17109) Reported by: under Patches:
+ logstream.diff uploaded by under (license 914) ........
+
+ * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 |
+ mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11
+ lines Allow type=user SIP endpoints to be loaded properly from
+ realtime. (closes issue #16021) Reported by: Guggemand Patches:
+ realtime-type-fix.patch uploaded by Guggemand (license 897)
+ (altered by me slightly to avoid ref leaks) Tested by: Guggemand
+ ........
+
+2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, /: Merged revisions 265273 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 |
+ dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines
+ fixes segfault when using generic plc ........
+
+2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 265316 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 |
+ tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines
+ On systems with a LOT of RAM, a signed integer sometimes printed
+ negative. (closes issue #16837) Reported by: jlpedrosa Patches:
+ 20100504__issue16837.diff.txt uploaded by tilghman (license 14)
+ ........
+
+2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fix memory hogging behavior of app_queue. From
+ reviewboard: This review request is for the patch on issue 17081.
+ A user reported that he saw increasing numbers of allocations
+ stemming from app_queue.c when he would run the "queue show" CLI
+ command. The user reported that he was using approximately 40
+ realtime queues and as he ran the CLI command more and more, the
+ memory usage would shoot up. As it turns out, there was a memory
+ leak and a separate usage of memory that, while not really a
+ leak, was very irresponsible. Both memory problems can be
+ attributed to the function init_queue(). When the "queue show"
+ command is run, all realtime queues have the init_queue()
+ function called on the in-memory queue. The idea is to place the
+ queue in its default state and then overwrite options specified
+ in the realtime backend as we read them. The first problem, the
+ memory leak, had to do with the fact that the string field for
+ the name of the first periodic announcement file was being
+ re-created every time init_queue was called. This patch corrects
+ the behavior by only calling ast_str_create if the memory has not
+ already been allocated. The other problem is a bit more
+ complicated. The majority of the strings in the call_queue
+ structure were changed to use the ast_string_fields API for 1.6.0
+ and beyond. init_queue resets all string fields on the queue to
+ their default values. Then, later in the realtime queue loading
+ process, these string fields are set to their configured values.
+ For those unfamiliar with string fields, frequent resizing of a
+ string like this is not what the string fields API is designed
+ for. The result of this constant resizing is that as the queue
+ gets loaded, eventually space for the string runs out and so a
+ new memory pool, at twice the size of the previously allocated
+ one, is created for the string fields. The reporter of issue
+ 17081 wrote a script that ran the "queue show" CLI command 2100
+ times. By the end, each of his 40 queues was taking about a
+ megabyte of memory apiece just for their string fields. My fix
+ for this problem is to revert the call_queue structure from using
+ string fields. In my patch here, I have moved the queue back to
+ using fixed-sized buffers. I ran the script provided by the
+ reporter of 17081 and determined that I no longer saw the
+ steadily-increasing memory usage that I had seen before applying
+ the patch. (closes issue #17081) Reported by: wliegel Patches:
+ 17081v2.patch uploaded by mmichelson (license 60) Tested by:
+ wliegel, mmichelson Review:
+ https://reviewboard.asterisk.org/r/651/
+
+ * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions
+ 265090 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May
+ 2010) | 15 lines Merged revisions 265089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May
+ 2010) | 8 lines Don't hang up on a queue caller if the file we
+ attempt to play does not exist. This also fixes a documentation
+ mistake in file.h that made my original attempt to correct this
+ problem not work correctly. (closes issue #17061) Reported by:
+ RoadKill ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 |
+ mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7
+ lines Be sure to set the sin_family on the proxy when allocating.
+ (closes issue #17157) Reported by: stuarth ........
+
+ * /, include/asterisk/channel.h: Merged revisions 265000 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500
+ (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri,
+ 21 May 2010) | 3 lines Fix grammatical error in comment. ........
+ ................
+
+ * main/channel.c, main/autoservice.c, /,
+ include/asterisk/channel.h: Merged revisions 264997 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500
+ (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May
+ 2010) | 32 lines Allow ast_safe_sleep to defer specific frames
+ until after the sleep has concluded. From reviewboard Background:
+ A Digium customer discovered a somewhat odd bug. The setup is
+ that parties A and B are bridged, and party A places party B on
+ hold. While party B is listening to hold music, he mashes a bunch
+ of DTMF. Party A takes party B off hold while this is happening,
+ but party B continues to hear hold music. I could reproduce this
+ about 1 in 5 times. The issue: When DTMF features are enabled and
+ a user presses keys, the channel that the DTMF is streamed to is
+ placed in an ast_safe_sleep for 100 ms, the duration of the
+ emulated tone. If an AST_CONTROL_UNHOLD frame is read from the
+ channel during the sleep, the frame is dropped. Thus the unhold
+ indication is never made to the channel that was originally
+ placed on hold. The fix: Originally, I discussed with Kevin
+ possible ways of fixing the specific problem reported. However,
+ we determined that the same type of problem could happen in other
+ situations where ast_safe_sleep() is used. Using autoservice as a
+ model, I modified ast_safe_sleep_conditional() to defer specific
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