[asterisk-commits] lmadsen: tag 1.6.2.8-rc1 r261552 - /tags/1.6.2.8-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 6 09:46:19 CDT 2010
Author: lmadsen
Date: Thu May 6 09:46:15 2010
New Revision: 261552
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=261552
Log:
Importing files for 1.6.2.8-rc1 release.
Added:
tags/1.6.2.8-rc1/.lastclean (with props)
tags/1.6.2.8-rc1/.version (with props)
tags/1.6.2.8-rc1/ChangeLog (with props)
Added: tags/1.6.2.8-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.8-rc1/.lastclean?view=auto&rev=261552
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--- tags/1.6.2.8-rc1/ChangeLog (added)
+++ tags/1.6.2.8-rc1/ChangeLog Thu May 6 09:46:15 2010
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+2010-05-06 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.8-rc1 Released
+
+2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell at digium.com>
+
+ * tests/test_heap.c: Add test case that ensures the heap handles
+ arbitrary removals properly. (issue #17277) Reported by:
+ cappucinoking Patches: test_heap.diff uploaded by cappucinoking
+ (license 1036) Tested by: cappucinoking, russell
+
+ * /, main/heap.c: Merged revisions 261496 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 |
+ russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines
+ Fix handling of removing nodes from the middle of a heap. This
+ bug surfaced in 1.6.2 and does not affect code in any other
+ released version of Asterisk. It manifested itself as SIP qualify
+ not happening when it should, causing peers to go unreachable.
+ This was debugged down to scheduler entries sometimes not getting
+ executed when they were supposed to, which was in turn caused by
+ an error in the heap code. The problem only sometimes occurs, and
+ it is due to the logic for removing an entry in the heap from an
+ arbitrary location (not just popping off the top). The scheduler
+ performs this operation frequently when entries are removed
+ before they run (when ast_sched_del() is used). In a normal pop
+ off of the top of the heap, a node is taken off the bottom,
+ placed at the top, and then bubbled down until the max heap
+ property is restored (see max_heapify()). This same logic was
+ used for removing an arbitrary node from the middle of the heap.
+ Unfortunately, that logic is full of fail. This patch fixes that
+ by fully restoring the max heap property when a node is thrown
+ into the middle of the heap. Instead of just pushing it down as
+ appropriate, it first pushes it up as high as it will go, and
+ _then_ pushes it down. Lastly, fix a minor problem in
+ ast_heap_verify(), which is only used for debugging. If a parent
+ and child node have the same value, that is not an error. The
+ only error is if a parent's value is less than its children. A
+ huge thanks goes out to cappucinoking for debugging this down to
+ the scheduler, and then producing an ast_heap test case that
+ demonstrated the breakage. That made it very easy for me to focus
+ on the heap logic and produce a fix. Open source projects are
+ awesome. (closes issue #16936) Reported by: ib2 Tested by:
+ cappucinoking, crjw (closes issue #17277) Reported by:
+ cappucinoking Patches: heap-fix.rev2.diff uploaded by russell
+ (license 2) Tested by: cappucinoking, russell ........
+
+2010-05-06 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (×”', 06 מ××™ 2010) |
+ 4 lines When failing to configure, don't destroy 'cfg' twice
+ Fixes a crash when some config section had an incorrect channel
+ config. ........
+
+2010-05-05 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May
+ 2010) | 19 lines Merged revisions 261274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May
+ 2010) | 12 lines Registration fix for SIP realtime. Make sure
+ realtime fields are not empty. (closes issue #17266) Reported by:
+ Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick
+ Lewis (license 657) Tested by: Nick_Lewis, sberney Review:
+ https://reviewboard.asterisk.org/r/643/ ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 |
+ pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10
+ lines 'queue reset stats' erroneously clears wrapuptime
+ configuration. Resets each member's lastcall to 0 now. (closes
+ issue #17262, #16519) Reported by: rain Patches:
+ wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested
+ by: rain ........
+
+2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c, /: Merged revisions 261095 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010)
+ | 18 lines Merged revisions 261093-261094 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010)
+ | 7 lines Protect against overflow, when calculating how long to
+ wait for a frame. (closes issue #17128) Reported by: under
+ Patches: d.diff uploaded by under (license 914) ........ r261094
+ | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2
+ lines Add a tiny corner case to the previous commit ........
+ ................
+
+2010-05-04 19:01 +0000 [r260927] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500
+ (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010)
+ | 12 lines Voicemail transfer to operator should occur
+ immediately, not after main menu. There were two scenarios in the
+ advanced options that while using the operator=yes and review=yes
+ options, the transfer occurred only after exiting the main menu
+ (after sending a reply or leaving a message for an extension).
+ Now after the audio is processed for the reply or message the
+ transfer occurs immediately as expected. ABE-2107 ABE-2108
+ ........ ................
+
+2010-05-04 16:58 +0000 [r260884] Matthew Nicholson <mnicholson at digium.com>
+
+ * configs/sip.conf.sample, include/asterisk/frame.h,
+ main/channel.c, /, channels/chan_sip.c: Merged revisions 254450
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25
+ Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that
+ arrive before a T.38-capable application is executing on a
+ channel. This patch addresses an issue found during working with
+ end-users using res_fax. If an incoming call is answered in the
+ dialplan, or jumps to the 'fax' extension due to reception of a
+ CNG tone (with faxdetect enabled), and then the remote endpoint
+ sends a T.38 re-INVITE, it is possible for the channel's T.38
+ state to be 'T38_STATE_NEGOTIATING' when the application starts
+ up. Unfortunately, even if the application wants to use T.38, it
+ can't respond to the peer's negotiation request, because the
+ AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent
+ originally has been lost, and the application needs the content
+ of that frame to be able to formulate a reply. This patch adds a
+ new 'request' type to AST_CONTROL_T38_PARAMETERS,
+ AST_T38_REQUEST_PARMS. If the application sends this request,
+ chan_sip will re-send the original control frame (with
+ AST_T38_REQUEST_NEGOTIATE as the request type), and the
+ application can respond as normal. If this occurs within the five
+ second timeout in chan_sip, the automatic cancellation of the
+ peer reinvite will be stopped, and the application will 'own' the
+ negotiation process from that point onwards. This also improves
+ the code path in chan_sip to allow sip_indicate(), when called
+ for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero
+ response, which should have been in place before since the
+ control frame *can* fail to be processed properly. It also
+ modifies ast_indicate() to return whatever result the channel
+ driver returned for this control frame, rather than converting
+ all non-zero results into '-1'. Finally, the new request type
+ intentionally returns a positive value, so that an application
+ that sends AST_T38_REQUEST_PARMS can know for certain whether the
+ channel driver accepted it and will be replying with a control
+ frame of its own, or whether it was ignored (if the
+ sip_indicate()/ast_indicate() path had properly supported failure
+ responses before, this would not be necessary). This patch also
+ modifies res_fax to take advantage of the new request. In
+ addition, this patch makes sip_t38_abort() actually lock the
+ private structure before doing its work... bad programmer, no
+ donut. This patch also enhances chan_sip's 'faxdetect' support to
+ allow triggering on T.38 re-INVITEs received as well as CNG tone
+ detection. Review: https://reviewboard.asterisk.org/r/556/
+ ........
+
+2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker at digium.com>
+
+ * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260802 | qwell | 2010-05-04 10:49:57 -0500
+ (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May
+ 2010) | 1 line Fix fallout from removing from configure script.
+ Pointed out by philipp64 on #asterisk-dev ........
+ ................
+
+ * /: Fix merge props
+
+2010-05-03 17:42 +0000 [r260743] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * Makefile, /: Merged revisions 260661-260662 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May
+ 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend
+ libdir when executing mkpkgconfig allowing non-root installs to
+ work. (closes issue #17268) Reported by: pabelanger Patches:
+ issue17268.patch uploaded by pabelanger (license 224) Tested by:
+ pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41
+ -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/
+ part. Thanks Qwell. ........
+
+2010-05-03 14:59 +0000 [r260571] Leif Madsen <lmadsen at digium.com>
+
+ * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500
+ (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03
+ May 2010) | 1 line Minor typo pointed out by pabelanger on IRC.
+ ........ ................
+
+2010-04-30 22:48 +0000 [r260441] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500
+ (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010)
+ | 11 lines Ensure channel state is not incorrectly set in the
+ case of a very early answer. The needringing bit was being read
+ in dahdi_read after answering thereby setting the state to
+ ringing from up. This clears needringing upon answering so that
+ is no longer possible. (closes issue #17067) Reported by: tzafrir
+ Patches: needringing.diff uploaded by tzafrir (license 46)
+ ........ ................
+
+2010-04-30 20:22 +0000 [r260373] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_musiconhold.c, /: Merged revisions 260346 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500
+ (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr
+ 2010) | 18 lines Fix potential crash from race condition due to
+ accessing channel data without the channel locked. In
+ res_musiconhold.c, there are several places where a channel's
+ stream's existence is checked prior to calling ast_closestream on
+ it. The issue here is that in several cases, the channel was not
+ locked while checking the stream. The result was that if two
+ threads checked the state of the channel's stream at
+ approximately the same time, then there could be a situation
+ where both threads attempt to call ast_closestream on the
+ channel's stream. The result here is that the refcount for the
+ stream would go below 0, resulting in a crash. I have added
+ proper channel locking to res_musiconhold.c to ensure that we do
+ not try to check chan->stream without the channel locked. A
+ Digium customer has been using this patch for several weeks and
+ has not had any crashes since applying the patch. ABE-2147
+ ........ ................
+
+2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/app.c: Merged revisions 260292 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 |
+ tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13
+ lines Don't allow file descriptors to go above 64k, when we're
+ closing them in a fork(2). This saves time, when, even though the
+ system allows the process limit to be that high, the practical
+ limit is much lower. (closes issue #17223) Reported by:
+ dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by
+ tilghman (license 14) Tested by: dbackeberg ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 260280 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30
+ Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan
+ context. (closes issue #17263) Reported by: pprindeville Patches:
+ freenum-dialplan.patch#3 uploaded by pprindeville (license 347)
+ ........
+
+2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500
+ (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010)
+ | 26 lines DTMF CallerID detection problems. The code handling
+ DTMF CallerID drops digits on long CallerID numbers and may
+ timeout waiting for the first ring with shorter numbers. The DTMF
+ emulation mode was not turned off when processing DTMF CallerID.
+ When the emulation code gets behind in processing the DTMF digits
+ it can skip a digit. For shorter numbers, the timeout may have
+ been too short. I increased it from 2 seconds to 4 seconds. Four
+ seconds is a typical time between rings for many countries.
+ (closes issue #16460) Reported by: sum Patches: issue16460.patch
+ uploaded by rmudgett (license 664) issue16460_v1.6.2.patch
+ uploaded by rmudgett (license 664) Tested by: sum, rmudgett
+ Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA
+ AST-334 JIRA SWP-901 ........ ................
+
+2010-04-29 18:18 +0000 [r260156] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 260148 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29
+ Apr 2010) | 2 lines Pattern match fail. ........
+
+2010-04-29 15:35 +0000 [r260051] David Vossel <dvossel at digium.com>
+
+ * main/audiohook.c, /, include/asterisk/audiohook.h: Merged
+ revisions 260050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010)
+ | 21 lines Merged revisions 260049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010)
+ | 14 lines Fixes crash in audiohook_write_list The middle_frame
+ in the audiohook_write_list function was being freed if a
+ audiohook manipulator returned a failure. This is incorrect
+ logic. This patch resolves this and adds detailed descriptions of
+ how this function should work and why manipulator failures must
+ be ignored. (closes issue #17052) Reported by: dvossel Tested by:
+ dvossel (closes issue #16196) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/623/ ........ ................
+
+2010-04-28 22:36 +0000 [r259959] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 |
+ mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11
+ lines Don't override peer context with domain context. (closes
+ issue #17040) Reported by: pprindeville Patches:
+ asterisk-1.6-bugid17040.patch uploaded by pprindeville (license
+ 347) Tested by: pprindeville Review:
+ https://reviewboard.asterisk.org/r/565/ ........
+
+2010-04-28 21:26 +0000 [r259899] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, channels/chan_local.c, /: Merged revisions 259870
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500
+ (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010)
+ | 33 lines resolves deadlocks in chan_local Issue_1. In the
+ local_hangup() 3 locks must be held at the same time... pvt,
+ pvt->chan, and pvt->owner. Proper deadlock avoidance is done when
+ the channel to hangup is the outbound chan_local channel, but
+ when it is not the outbound channel we have an issue... We
+ attempt to do deadlock avoidance only on the tech pvt, when both
+ the tech pvt and the pvt->owner are locked coming into that loop.
+ By never giving up the pvt->owner channel deadlock avoidance is
+ not entirely possible. This patch resolves that by doing deadlock
+ avoidance on both the pvt->owner and the pvt when trying to get
+ the pvt->chan lock. Issue_2. ast_prod() is used in
+ ast_activate_generator() to queue a frame on the channel and make
+ the channel's read function get called. This function is used in
+ ast_activate_generator() while the channel is locked, which
+ mean's the channel will have a lock both from the generator code
+ and the frame_queue code by the time it gets to chan_local.c's
+ local_queue_frame code... local_queue_frame contains some of the
+ same crazy deadlock avoidance that local_hangup requires, and
+ this recursive lock prevents that deadlock avoidance from
+ happening correctly. This patch removes ast_prod() from the
+ channel lock so only one lock is held during the
+ local_queue_frame function. (closes issue #17185) Reported by:
+ schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel
+ (license 671) issue_17185_v2.diff uploaded by dvossel (license
+ 671) Tested by: schmoozecom, GameGamer43 Review:
+ https://reviewboard.asterisk.org/r/631/ ........ ................
+
+2010-04-28 21:09 +0000 [r259854] Leif Madsen <lmadsen at digium.com>
+
+ * config.guess: Merged revisions 259853 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010)
+ | 14 lines Merged revisions 259852 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010)
+ | 6 lines Update config.guess. Updating config.guess because
+ after installing Ubuntu Server 9.10 and running all the update
+ scripts, running ./configure would not continue because it was
+ unable to determine what kind of system I had. After updating
+ config.guess things started working again. ........
+ ................
+
+2010-04-28 20:34 +0000 [r259781-259851] Jason Parker <jparker at digium.com>
+
+ * /, configure, configure.ac: Merged revisions 259848 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259848 | qwell | 2010-04-28 15:32:14 -0500
+ (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr
+ 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so
+ systems without install can use install-sh from our source dir.
+ ........ ................
+
+ * makeopts.in, /: Merged revisions 259837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) |
+ 9 lines Merged revisions 259833 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) |
+ 1 line Missed this when removing $ID ........ ................
+
+ * Makefile, /, configure, configure.ac: Merged revisions 259760 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259760 | qwell | 2010-04-28 14:19:54 -0500
+ (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) |
+ 7 lines Remove usage of `id` since it isn't useful and was
+ causing breakge. Solaris `id` doesn't support the -u argument.
+ Instead of figuring out how to fix this to work on Solaris, I
+ decided to check why it was necessary and where else it was used.
+ It was only used in one place, and it hasn't been needed for a
+ very long time (I question whether it was ever needed). ........
+ ................
+
+2010-04-28 17:19 +0000 [r259681] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500
+ (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010)
+ | 4 lines Do not play goodbye prompt after timeout of message
+ review. ABE-2124 ........ ................
+
+2010-04-27 22:46 +0000 [r259616] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500
+ (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010)
+ | 11 lines DAHDI "WARNING" message is confusing and vague
+ "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed
+ failed: Success" Changed the warning to "Failed to decode
+ CallerID on channel 'name'". The message before it is likely more
+ specific about why the CallerID decode failed. SWP-501 AST-283
+ ........ ................
+
+2010-04-27 21:50 +0000 [r259528] Leif Madsen <lmadsen at digium.com>
+
+ * sounds/Makefile: Merged revisions 259527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010)
+ | 23 lines Merged revisions 259526 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010)
+ | 15 lines Update sounds files. * Add additional sounds prompts
+ for say_enumeration * Update the English conference sounds
+ prompts so they are better quality and all sound more consistent
+ * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files
+ to include all present sound files Both core (en, fr, es) and
+ extra (en, fr) sounds files have been updated. (closes issue
+ #16200) Reported by: murf (closes issue #17137) Reported by:
+ lmadsen ........ ................
+
+2010-04-27 21:25 +0000 [r259356-259486] Jason Parker <jparker at digium.com>
+
+ * main/editline/configure.in, /, main/editline/configure,
+ main/editline/Makefile.in: Merged revisions 259439 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Add gar to the check for AR for those silly OSes
+ (Solaris) that don't have ar. autoconf2.13 couldn't handle
+ AC_PROG_GREP, so I removed it. This is fine, since we don't need
+ to use anything that the configure script doesn't. ........
+
+ * /: Unblock revision 259439.
+
+ * /, configure, configure.ac: Merged revisions 259353 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r259353 | qwell | 2010-04-27 14:31:55 -0500
+ (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) |
+ 5 lines Support the silly OSes that don't have ar and strip.
+ Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't
+ specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to
+ AC_CHECK_TOOLS. ........ ................
+
+2010-04-27 19:03 +0000 [r259310] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
+ revisions 259307 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010)
+ | 21 lines Merged revisions 259270 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010)
+ | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue
+ #7321 implements a new chan_dahdi configuration option. However,
+ a change mentioned in the issue was never implemented. This is
+ the change that will allow the feature to work. I added a note to
+ chan_dahdi.conf.sample about the feature. (closes issue #17143)
+ Reported by: djensen99 Patches: diff.txt uploaded by djensen99
+ (license NA) (One line change) Tested by: djensen99 ........
+ ................
+
+2010-04-26 21:48 +0000 [r259103-259109] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, /: Merged revisions 259105 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr
+ 2010) | 9 lines Merged revisions 259104 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr
+ 2010) | 3 lines Let compilation succeed warning-free when
+ DONT_OPTIMIZE is turned off. ........ ................
+
+ * main/channel.c, /: Merged revisions 259023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr
+ 2010) | 19 lines Merged revisions 259018 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr
+ 2010) | 13 lines Prevent Newchannel manager events for dummy
+ channels. No Newchannel manager event will be fired for channels
+ that are allocated to not match a registered technology type.
+ Thus bogus channels allocated solely for variable substitution or
+ CDR operations do not result in a Newchannel event. (closes issue
+ #16957) Reported by: atis Review:
+ https://reviewboard.asterisk.org/r/601 ........ ................
+
+2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 |
+ lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines
+ Small error in the T.140 RTP port verbose log. (closes issue
+ #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff
+ uploaded by frawd (license 610) Tested by: russell ........
+
+2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010)
+ | 13 lines Merged revisions 258775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010)
+ | 6 lines When StopMonitor is called, ensure that it will not be
+ restarted by a channel event. (closes issue #16590) Reported by:
+ kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm
+ (license 888) ........ ................
+
+2010-04-22 22:15 +0000 [r258676] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions
+ 258671,258675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr
+ 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May
+ 2009) | 8 lines Set the proper disposition on originated calls.
+ (closes issue #14167) Reported by: jpt Patches:
+ call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+ Tested by: dlotina, rmartinez, mnicholson ........ r258670 |
+ mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11
+ lines Fix broken CDR behavior. This change allows a CDR record
+ previously marked with disposition ANSWERED to be set as BUSY or
+ NO ANSWER. Additionally this change partially reverts r235635 and
+ does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated
+ from ast_call(). To preserve proper CDR behavior, the
+ AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in
+ ast_bridge_call(). (closes issue #16797) Reported by:
+ VarnishedOtter Tested by: mnicholson ........ (closes issue
+ #16222) Reported by: telles Tested by: mnicholson
+ ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500
+ (Thu, 22 Apr 2010) | 2 lines Fix previous commit.
+ ................
+
+2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell at digium.com>
+
+ * /, main/event.c: Merged revisions 258632 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only
+ merge the bug fixes, not the unit test. ........ r258632 |
+ russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines
+ Add ast_event subscription unit test and fix some ast_event API
+ bugs. This patch introduces another test in test_event.c that
+ exercises most of the subscription related ast_event API calls. I
+ made some minor additions to the existing event allocation test
+ to increase API coverage by the test code. Finally, I made a list
+ in a comment of API calls not yet touched by the test module as a
+ to-do list for future test development. During the development of
+ this test code, I discovered a number of bugs in the event API.
+ 1) subscriptions to AST_EVENT_ALL were not handled appropriately
+ in a couple of different places. The API allows a subscription to
+ all event types, but with IE parameters, just as if it was a
+ subscription to a specific event type. However, the parameters
+ were being ignored. This affected ast_event_check_subscriber()
+ and event distribution to subscribers. 2) Some of the logic in
+ ast_event_check_subscriber() for checking subscriptions against
+ query parameters was wrong. Review:
+ https://reviewboard.asterisk.org/r/617/ ........
+
+ * /, doc/tex/channelvariables.tex: Merged revisions 258515 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22
+ Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........
+
+2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500
+ (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010)
+ | 8 lines Fix looping forever when no input received in certain
+ voicemail menu scenarios. Specifically, prompting for an
+ extension (when leaving or forwarding a message) or when
+ prompting for a digit (when saving a message or changing
+ folders). ABE-2122 SWP-1268 ........ ................
+
+2010-04-21 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/asterisk.tex: Remove missed line in previous merge.
+ (issue #17220)
+
+ * configure: Forgot to merge the updated configure script. (issue
+ #17220)
+
+ * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in,
+ doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile,
+ configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex,
+ build_tools/prep_tarball: Merged revisions 258351 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010)
+ | 20 lines Add ability to generate ASCII documentation from the
+ TeX files. These changes add the ability to run 'make
+ asterisk.txt' just like the existing 'make asterisk.pdf' commands
+ to generate a text document from the TeX files we have in the
+ doc/tex/ directory. I've also updated a few of the .tex files
+ because they weren't properly escaping certain characters so they
+ would show up as Unicode characters (like [U+021C]). Made changes
+ to the configure scripts so it would detect the catdvi program
+ which is required to convert the .dvi file generated by latex.
+ I've also added a few lines to the build_tools/prep_tarball
+ script so that the text documentation gets generated and added to
+ future tarballs of Asterisk releases. (closes issue #17220)
+ Reported by: lmadsen Patches: asterisk.txt.patch uploaded by
+ lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger
+ (license 224) Tested by: lmadsen, pabelanger ........
+
+2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 |
+ dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines
+ fixes issue with double "sip:" in header field This is a clear
+ mistake in logic. Future discussions about how to avoid having to
+ handle uri's like this should take place in the future, but this
+ fix needs to go in for now. (closes issue #15847) Reported by:
+ ebroad Patches: doublesip.patch uploaded by ebroad (license 878)
+ ........
+
+2010-04-20 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen at digium.com>
+
+ * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20
+ Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on
+ asterisk-users. ........
+
+ * configs/extensions.conf.sample, /: Merged revisions 258147 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20
+ Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers
+ (http://www.freenum.org). Minor tweaks and documentation added by
+ me. (closes issue #17058) Reported by: pprindeville Patches:
+ freenum.patch#5 uploaded by pprindeville (license 347) Tested by:
+ lmadsen ........
+
+2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500
+ (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010)
[... 23394 lines stripped ...]
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