[asterisk-commits] lmadsen: tag 1.6.1-rc1 r172167 - /tags/1.6.1-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 28 17:31:08 CST 2009
Author: lmadsen
Date: Wed Jan 28 17:31:08 2009
New Revision: 172167
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=172167
Log:
Importing files for 1.6.1-rc1 release
Added:
tags/1.6.1-rc1/.lastclean (with props)
tags/1.6.1-rc1/.version (with props)
tags/1.6.1-rc1/ChangeLog (with props)
Added: tags/1.6.1-rc1/.lastclean
URL: http://svn.digium.com/svn-view/asterisk/tags/1.6.1-rc1/.lastclean?view=auto&rev=172167
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--- tags/1.6.1-rc1/ChangeLog (added)
+++ tags/1.6.1-rc1/ChangeLog Wed Jan 28 17:31:08 2009
@@ -1,0 +1,52267 @@
+2009-01-28 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.1-rc1 released
+
+2009-01-28 22:52 +0000 [r172133] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_odbc.c, /: Merged revisions 172131 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r172131 | tilghman | 2009-01-28 16:48:01 -0600 (Wed, 28 Jan 2009)
+ | 7 lines Fix how we skip fields (to avoid fields which don't
+ exist) when doing an UPDATE. (closes issue #14205) Reported by:
+ maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: blitzrage ........
+
+2009-01-28 20:56 +0000 [r172067] Steve Murphy <murf at digium.com>
+
+ * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /,
+ main/features.c, include/asterisk/channel.h: Merged revisions
+ 172063 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) |
+ 52 lines Merged revisions 172030 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) |
+ 46 lines This patch fixes h-exten running misbehavior in
+ manager-redirected situations. What it does: 1. A new Flag value
+ is defined in include/asterisk/channel.h,
+ AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
+ bridge hangup exten code not to run the h-exten there (nor
+ publish the bridge cdr there). It will done at the pbx-loop level
+ instead. 2. In the manager Redirect code, I set this flag on the
+ channel if the channel has a non-null pbx pointer. I did the same
+ for the second (chan2) channel, which gets run if name2 is set...
+ and the first succeeds. 3. I restored the ending of the cdr for
+ the pbx loop h-exten running code. Don't know why it was removed
+ in the first place. 4. The first attempt at the fix for this bug
+ was to place code directly in the async_goto routine, which was
+ called from a large number of places, and could affect a large
+ number of cases, so I tested that fix against a fair number of
+ transfer scenarios, both with and without the patch. In the
+ process, I saw that putting the fix in async_goto seemed not to
+ affect any of the blind or attended scenarios, but still, I was
+ was highly concerned that some other scenarios I had not tested
+ might be negatively impacted, so I refined the patch to its
+ current scope, and jmls tested both. In the process, tho, I saw
+ that blind xfers in one situation, when the one-touch blind-xfer
+ feature is used by the peer, we got strange h-exten behavior. So,
+ I inserted code to swap CDRs and to set the HANGUP_DONT field, to
+ get uniform behavior. 5. I added code to the bridge to obey the
+ HANGUP_DONT flag, skipping both publishing the bridge CDR, and
+ running the h-exten; they will be done at the pbx-loop (higher)
+ level instead. 6. I removed all the debug logs from the patch
+ before committing. 7. I moved the AUTOLOOP set/reset in the
+ h-exten code in res_features so it's only done if the h-exten is
+ going to be run. A very minor performance improvement, but
+ technically correct. (closes issue #14241) Reported by: jmls
+ Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer
+ uploaded by murf (license 17) Tested by: murf, jmls ........
+ ................
+
+2009-01-28 17:29 +0000 [r171966] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600
+ (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28
+ Jan 2009) | 2 lines Clarify log message (suggested by manxpower
+ on #asterisk-dev) ........ ................
+
+2009-01-28 13:21 +0000 [r171857] Olle Johansson <oej at edvina.net>
+
+ * configs/sip.conf.sample: Merged revisions 171838 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons,
+ 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2
+ lines Add a better explanation of the difference between the
+ device namespace and the dialplan for newbies. ........
+ ................
+
+2009-01-27 22:01 +0000 [r171620-171693] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600
+ (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan
+ 2009) | 39 lines Fix devicestate problems for "always-on" agent
+ channels A revision to chan_agent attempted to "inherit" the
+ device state of the underlying channel in order to report the
+ device state of an agent channel more accurately. The problem
+ with the logic here is that it makes no sense to use this for
+ always-on agents. If the agent is logged in, then to the
+ underlying channel, the agent will always appear to be "in use,"
+ no matter if the agent is on a call or not. The reason is that to
+ the underlying channel, the channel is currently in use on a call
+ to the AgentLogin application. The most common cause that I found
+ for this issue to occur was for a SIP channel to be the
+ underlying channel type for an Agent channel. If the SIP phone
+ re-registers, then the registration will cause the device state
+ core to query the device state of the SIP channel. Since the SIP
+ channel is in use, the Agent channel would also inherit this
+ status. Once the agent channel was set to "in use" there was no
+ way that the device state could change on that channel unless the
+ agent logged out. The solution for this problem is a bit
+ different in 1.4 than it is in the other branches. In 1.4, there
+ will be a one-line fix to make sure that only callback agents
+ will inherit device state from their underlying channel type. For
+ the other branches of Asterisk, since callback support has been
+ removed, there is also no need for device state inheritance in
+ chan_agent, so I will simply be removing it from the code. In
+ addition, the 1.4 source is getting a new comment to help the
+ next person who edits chan_agent.c. I'm adding a comment that a
+ agent_pvt's loginchan field may be used to determine if the agent
+ is a callback agent or not. (closes issue #14173) Reported by:
+ nathan Patches: 14173.patch uploaded by putnopvut (license 60)
+ Tested by: nathan, aramirez ........ ................
+
+ * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan
+ 2009) | 26 lines Merged revisions 171621 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan
+ 2009) | 18 lines Prevent a crash from occurring when a jitter
+ buffer interpolated frame is removed from a slinfactory
+ slinfactory used the "samples" field of an ast_frame in order to
+ determine the amount of data contained within the frame. In
+ certain cases, such as jitter buffer interpolated frames, the
+ frame would have a non-zero value for "samples" but have NULL
+ "data" This caused a problem when a memcpy call in
+ ast_slinfactory_read would attempt to access invalid memory. The
+ solution in use here is to never feed frames into the slinfactory
+ if they have NULL "data" (closes issue #13116) Reported by:
+ aragon Patches: 13116.diff uploaded by putnopvut (license 60)
+ ........ ................
+
+ * apps/app_queue.c, /: Merged revisions 171618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 |
+ mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24
+ lines Fix queue crashes that would occur after the calling
+ channel was masqueraded. The data passed to the
+ end_bridge_callback was assumed to be data which was still
+ stack'd. The problem was that with some call features, attended
+ transfers in particular, a new bridge thread is started once the
+ feature completes, meaning that when the end_bridge_callback is
+ called, the end_bridge_callback_data was invalid. To fix this
+ problem, there are two measures taken 1. Instead of pointing to
+ stacked data, we now used heap-allocated data for passing to the
+ end_bridge_callback in app_queue 2. Since bridges can end
+ multiple times on a single logical call, we wait until the final
+ bridge is broken to actually set any queue variables. This is
+ accomplished through reference-counting and the use of an
+ end_bridge_callback_data_fixup function in app_queue.c (closes
+ issue #14260) Reported by: ccesario Patches: 14260.patch uploaded
+ by putnopvut (license 60) Tested by: ccesario ........
+
+2009-01-27 15:19 +0000 [r171540] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23
+ lines Solving the same issue, but a bit different in trunk...
+ Merged revisions 171527 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13
+ lines Use the same branch tag in CANCEL as in INVITE Originally
+ putnopvut implemented some changes in revision 142079 that
+ according to the bug report seemed to have worked then, but
+ somehow fails now. I guess code, as humans, get old and forget
+ stuff. Anyway, this bug caused CANCEL not to work with picky
+ systems. Thanks Fredrik for pointing out where the bug in the SIP
+ messaging was. (closes issue #14346) Reported by: oej Patches:
+ bug14346.diff uploaded by oej (license 306) Tested by: oej
+ ........ ................
+
+2009-01-26 14:58 +0000 [r171361] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) |
+ 17 lines Merged revisions 171264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9
+ lines Don't retransmit 401 on REGISTER requests when
+ alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000
+ Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by
+ klaus3000 (license 65) Tested by: klaus3000 ........
+ ................
+
+2009-01-26 00:04 +0000 [r171190] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009)
+ | 13 lines Merged revisions 171187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009)
+ | 6 lines Correctly track the hookstate (closes issue #13686)
+ Reported by: itiliti Patches: 20081013__bug13686.diff.txt
+ uploaded by Corydon76 (license 14) ........ ................
+
+2009-01-25 13:40 +0000 [r170982] Sean Bright <sean.bright at gmail.com>
+
+ * /, apps/app_page.c: Merged revisions 170980 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan
+ 2009) | 16 lines Merged revisions 170979 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan
+ 2009) | 9 lines Resolve a logic error that was causing Page() to
+ crash when more than one channel was specified. (closes issue
+ #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
+ uploaded by seanbright (license 71) Tested by: kc0bvu ........
+ ................
+
+2009-01-25 02:52 +0000 [r170945] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009)
+ | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second
+ part of this macro is written as 0[a] instead of a[0], it will
+ force a failure if the macro is used on a C++ object that
+ overloads the [] operator. ........
+
+2009-01-24 13:57 +0000 [r170839] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 170837 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600
+ (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24
+ Jan 2009) | 2 lines Remove superfluous implementation note
+ (closes issue #14319) ........ ................
+
+2009-01-23 23:53 +0000 [r170831] Richard Mudgett <rmudgett at digium.com>
+
+ * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 |
+ rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line
+ Fix asterisk.pdf generation if branch name has an underscore in
+ it. ........
+
+2009-01-23 22:59 +0000 [r170792] Russell Bryant <russell at digium.com>
+
+ * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 |
+ russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines
+ Don't blow up if a branch name has an underscore in it ........
+
+2009-01-23 20:57 +0000 [r170693-170722] Mark Michelson <mmichelson at digium.com>
+
+ * configs/res_odbc.conf.sample, /: Merged revisions 170720 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600
+ (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan
+ 2009) | 8 lines Add notes to the idlecheck explanation in
+ res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000
+ Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by
+ klaus3000 (license 65) ........ ................
+
+ * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600
+ (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan
+ 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use
+ deprecated syntax * Convert Wait,1 to Wait(1) * Convert
+ SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
+ priorities beyond the first Also added test for Chinese numbers,
+ too. (closes issue #14320) Reported by: dant Patches:
+ i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
+ 670) ........ ................
+
+2009-01-23 20:20 +0000 [r170664] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c, /: Merged revisions 170652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) |
+ 11 lines Merged revisions 170648 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a channel is answered make sure any indications
+ currently playing stop. Usually the phone would do this but if
+ the channel was already answered then they are being generated by
+ Asterisk and we darn well need to stop them. (closes issue
+ #14249) Reported by: RadicAlish ........ ................
+
+2009-01-23 19:37 +0000 [r170637] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c, /: Merged revisions 170608 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r170608 | tilghman | 2009-01-23 13:25:10 -0600
+ (Fri, 23 Jan 2009) | 9 lines Merged revisions 170588 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23
+ Jan 2009) | 2 lines Additions to AST-2009-001 ........
+ ................
+
+2009-01-23 19:10 +0000 [r170507-170571] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 170569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) |
+ 11 lines Merged revisions 170568 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4
+ lines When a call is forwarded stop any active indications. The
+ new channel will provide an indication, if need be, itself.
+ (closes issue #14310) Reported by: RadicAlish ........
+ ................
+
+ * /, channels/chan_sip.c: Merged revisions 170505 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) |
+ 11 lines Merged revisions 170504 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4
+ lines Use the on hold flag to see if the call is on hold or not.
+ It is possible that our address for them will still be valid even
+ though they are on hold. (closes issue #14295) Reported by:
+ klaus3000 ........ ................
+
+2009-01-23 17:49 +0000 [r170502] Michiel van Baak <michiel at vanbaak.info>
+
+ * /, channels/chan_h323.c: Merged revisions 170501 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r170501 | mvanbaak | 2009-01-23 18:46:02 +0100 (Fri, 23 Jan 2009)
+ | 1 line let's use SENTINEL where needed ........
+
+2009-01-23 16:35 +0000 [r170458] Doug Bailey <dbailey at digium.com>
+
+ * channels/chan_dahdi.c: MWI messages included in CID spill was not
+ being properly handled and prevented the call from being
+ processed (issue #14313) Reported by: seandarcy Tested by:
+ dbailey
+
+2009-01-23 15:51 +0000 [r170395] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, /: Merged revisions 170393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan
+ 2009) | 36 lines Merged revisions 170392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan
+ 2009) | 28 lines Fix broken call pickup There was a subtle change
+ in ast_do_masquerade which resulted in failed attempts to pickup
+ calls. The problem was that the value of the AST_FLAG_OUTGOING
+ flag was copied from the clone to the original channel. In the
+ case of call pickup, this meant that the AST_FLAG_OUTGOING flag
+ ended up being cleared on the channel that was attempting to
+ execute the pickup. Because this flag was not set, when ast_read
+ came across an answer frame, it ignored it. The result of this
+ was that the calling channel was never properly answered. This
+ fix changes the behavior in ast_do_masquerade to set the flags on
+ the original channel to the union of the flags on the clone
+ channel. This way, if the AST_FLAG_OUTGOING flag is set on either
+ of the two channels involved in the masquerade, the resulting
+ channel will have the flag set as well. (closes issue #14206)
+ Reported by: francesco_r Patches: 14206.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut
+ ........ ................
+
+2009-01-22 20:06 +0000 [r170242] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c, /: Merged revisions 170240 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) |
+ 14 lines Merged revisions 170239 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7
+ lines Don't crash if RTCP is not enabled on an RTP structure but
+ statistics are output. (closes issue #14234) Reported by: jcovert
+ Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
+ rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........
+ ................
+
+2009-01-22 17:21 +0000 [r170178] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 170165 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170165 | tilghman | 2009-01-22 11:19:28 -0600 (Thu, 22 Jan 2009)
+ | 13 lines Merged revisions 170158 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009)
+ | 6 lines Allow global variables after substitution to be as long
+ as other variables. (closes issue #14263) Reported by: markd
+ Patches: 20090120__bug14263.diff.txt uploaded by Corydon76
+ (license 14) ........ ................
+
+2009-01-22 16:54 +0000 [r170049-170150] Joshua Colp <jcolp at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 170148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) |
+ 11 lines Merged revisions 170147 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
+ lines If we are unable to request a DAHDI pseudo channel and we
+ are using the user introduction without review option make sure
+ it gets unset so other code does not blindly assume a DAHDI
+ pseudo channel exists. (closes issue #14282) Reported by:
+ cheesegrits ........ ................
+
+ * main/pbx.c, /: Merged revisions 170051 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) |
+ 13 lines Merged revisions 170050 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6
+ lines Do a string comparison instead of pointer comparison since
+ some people specify the context they are actually in as an
+ argument to get around some funkiness. (closes issue #14011)
+ Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga
+ (license 665) ........ ................
+
+ * apps/app_parkandannounce.c, /: Merged revisions 170047 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan
+ 2009) | 4 lines Clear the autoloop flag when parsing and setting
+ the context/extension/priority to go back to. When the channel
+ executes a PBX again we want it to start out at the point we
+ explicitly say and at that point it will not yet be doing
+ autoloop. (closes issue #14304) Reported by: jcovert ........
+
+2009-01-22 00:46 +0000 [r169946] Tilghman Lesher <tlesher at digium.com>
+
+ * /, include/asterisk/linkedlists.h: Merged revisions 169944 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r169944 | tilghman | 2009-01-21 18:44:52 -0600
+ (Wed, 21 Jan 2009) | 16 lines Merged revisions 169943 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009)
+ | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really
+ wanted to ask is whether autoconf detected a static initializer
+ value. This fixes rwlocks on all such platforms (mainly, Mac OS
+ X). (closes issue #13767) Reported by: jcovert Patches:
+ 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: jcovert, Corydon76 ........ ................
+
+2009-01-21 23:28 +0000 [r169871] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c, /: Merged revisions 169869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) |
+ 11 lines Merged revisions 169867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4
+ lines Read lock the contexts to maintain the locking order when
+ we are notified that the state of a device has changed. (closes
+ issue #13839) Reported by: mcallist ........ ................
+
+2009-01-21 22:23 +0000 [r169830] Michiel van Baak <michiel at vanbaak.info>
+
+ * /, doc/tex/extensions.tex: Merged revisions 169793 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009)
+ | 2 lines remove duplicated sentence. ........
+
+2009-01-21 22:11 +0000 [r169792-169796] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/say.c: Merged revisions 169794 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169794 |
+ mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17
+ lines Fix a crash when saying certain numbers in Chinese This
+ commit fixes a crash that was occurring when attempting to say a
+ number between 10000 and 100000 due to dividing by 0. This also
+ removes some places where a "zero" is spoken when it should not
+ be. (closes issue #14291) Reported by: dant Patches:
+ say.c-14291.diff uploaded by dant (license 670) Tested by: dant
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 169791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169791 |
+ mmichelson | 2009-01-21 15:53:55 -0600 (Wed, 21 Jan 2009) | 18
+ lines Further fix some oddities in sip show users and sip show
+ peers logic ccesario on IRC pointed out that his sip peers were
+ not displayed properly when he would issue the command "sip show
+ peers." The problem was that the onlymatchonip field was used to
+ determine if the endpoint was a "peer" or "user." The tricky part
+ is that a "friend" is supposed to be treated as both a "user" and
+ a "peer" but the logic would not allow "friends" to show up as
+ "peers" since onlymatchonip was set to FALSE for friends. I have
+ modified the sip_peer structure to more explicitly keep track of
+ what type endpoint it is so that the various manager and CLI
+ commands will display the expected information Reported by
+ ccesario via IRC Tested by ccesario ........
+
+2009-01-21 21:05 +0000 [r169725] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 169723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009)
+ | 15 lines Merged revisions 169722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009)
+ | 8 lines Extra NULLs in the output cause some terminal types to
+ abort in the middle of a color code, causing terminal weirdness.
+ (closes issue #14130) Reported by: coolmig Patches:
+ 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, coolmig ........ ................
+
+2009-01-21 17:40 +0000 [r169674] Steve Murphy <murf at digium.com>
+
+ * utils/refcounter.c, /: Merged revisions 169673 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169673 |
+ murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines
+ This patch corrects a segfault reported in 14289, due to a null
+ ptr being refd. Yes, seanbright is right in the bug comments,
+ that is the fix. Sorry for this oversight; I guess my personal
+ usage didn't have this happen! murf (closes issue #14289)
+ Reported by: jamesgolovich ........
+
+2009-01-21 10:49 +0000 [r169622-169626] Russell Bryant <russell at digium.com>
+
+ * /: Merged revisions 169625 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169625 |
+ russell | 2009-01-21 04:49:00 -0600 (Wed, 21 Jan 2009) | 2 lines
+ Remove properties that erroneously got merged into trunk ........
+
+ * main/tcptls.c, /: Merged revisions 169620 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169620 |
+ russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines
+ Fix a regression in TCP support. This patch fixes a problem that
+ caused chan_sip to think that every open TCP session was to a
+ remote address of 0.0.0.0:0. (closes issue #14287) Reported by:
+ jamesgolovich Patches: bug-14287.diff.txt uploaded by
+ jamesgolovich (license 176) ........
+
+2009-01-21 00:35 +0000 [r169559-169613] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c, /: Merged revisions 169611 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 |
+ mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22
+ lines Fix device state parsing issues for channel names with
+ multiple slashes The fix being applied is a bit different for
+ trunk and the 1.6.X branches. For trunk, we only wish to strip
+ off the characters beyond the second slash if the channel is a
+ Local channel (i.e. we are removing the /n from the device name).
+ Other channel technologies with multiple slashes (e.g. DAHDI)
+ need the information after the second slash in order to get the
+ proper device state information. In addition to this fix, the
+ 1.6.X branches are receiving a much more important fix as well.
+ The problem in 1.6.X is that the member's device name was being
+ directly changed instead of having a copy changed. This meant
+ that we would strip off the second slash and trailing characters
+ and then leave the member's device name like that permanently
+ thereafter. (closes issue #14014) Reported by: kebl0155 Patches:
+ 14014_number2.patch uploaded by putnopvut (license 60) Tested by:
+ kebl0155 ........
+
+ * apps/app_queue.c, /: Merged revisions 169574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169574 |
+ mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6
+ lines Use the default timeout for a queue instead of -1 (closes
+ issue #14272) Reported by: timking ........
+
+ * /, channels/chan_sip.c: Merged revisions 169557 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169557 |
+ mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19
+ lines Convert the character pointers in a sip_request to be
+ pointer offsets When an ast_str expands to hold more data, any
+ pointers that were pointing to the data prior to the expansion
+ will be pointing at invalid memory. This change makes such
+ pointers used in chan_sip.c instead be offsets from the beginning
+ of the string so that the same math may be applied no matter
+ where in memory the string resides. To help ease this transition,
+ a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so
+ that given a sip_request and an offset, the string at that offset
+ is returned. (closes issue #14220) Reported by: riksta Tested by:
+ putnopvut Review http://reviewboard.digium.com/r/126/ ........
+
+2009-01-20 19:31 +0000 [r169488-169554] Terry Wilson <twilson at digium.com>
+
+ * /, main/features.c: Merged revisions 169510 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169510 |
+ twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines
+ Make a proper builtin attended transfer to parking work This is
+ an ugly hack from 1.4 that allows the timeout callback from a
+ parked call to use the right channel name for the callback when
+ the park is done with a builtin attended transfer (that isn't
+ completed early). This hasn't ever worked in trunk and no one has
+ complained yet, so eh. ........
+
+ * /, main/features.c: Merged revisions 169486 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009)
+ | 13 lines Merged revisions 169485 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009)
+ | 6 lines Don't play audio to the channel if we've masqueraded
+ (closes issue #14066) Reported by: bluefox Tested by:
+ otherwiseguy, bluefox ........ ................
+
+2009-01-19 20:10 +0000 [r169368] Tilghman Lesher <tlesher at digium.com>
+
+ * main/manager.c, /, apps/app_userevent.c: Merged revisions 169365
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r169365 | tilghman | 2009-01-19 14:05:52 -0600
+ (Mon, 19 Jan 2009) | 11 lines Merged revisions 169364 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009)
+ | 4 lines Truncate userevents at the end of a line, when the
+ command exceeds the buffer. (closes issue #14278) Reported by:
+ fnordian ........ ................
+
+2009-01-19 15:55 +0000 [r169213] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 169211 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r169211 | mmichelson | 2009-01-19 09:54:06 -0600
+ (Mon, 19 Jan 2009) | 21 lines Merged revisions 169210 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan
+ 2009) | 13 lines Prevent a crash in chan_local due to a potential
+ NULL pointer dereference Move the check for if both channels on a
+ local_pvt have generators to below where p->chan is checked for
+ NULLity (NULLness?). This prevents a crash from occurring if
+ p->chan is NULL. (closes issue #14189) Reported by: sascha
+ Patches: 14189.patch uploaded by putnopvut (license 60) Tested
+ by: sascha ........ ................
+
+2009-01-17 18:46 +0000 [r169154] Doug Bailey <dbailey at digium.com>
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add
+ discriminator for when ring pulse alert signal is used to preface
+ MWI spills This prevents the situation when MWI messages are
+ added to caller ID spills causing the channel to be hung up
+
+2009-01-17 01:59 +0000 [r168981-169082] Terry Wilson <twilson at digium.com>
+
+ * main/tcptls.c, /, main/http.c, include/asterisk/tcptls.h: Merged
+ revisions 169080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169080 |
+ twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines
+ Fix qualify for TCP peer (closes issue #14192) Reported by:
+ pabelanger Patches: asterisk-bug14192.diff.txt uploaded by
+ jamesgolovich (license 176) Tested by: jamesgolovich ........
+
+ * /, channels/chan_sip.c: Merged revisions 169044 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r169044 |
+ twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines
+ Fix port :0 added to SIP INVITE URI when outboundproxy used
+ (closes issue #14233) Reported by: chris-mac Patches:
+ asterisk-bug14233.diff.txt uploaded by jamesgolovich (license
+ 176) Tested by: jamesgolovich, chris-mac, otherwiseguy ........
+
+ * /, main/features.c: Merged revisions 168941 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009)
+ | 19 lines Merged revisions 168716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009)
+ | 12 lines Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy ........
+ ................
+
[... 51603 lines stripped ...]
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