[svn-commits] lmadsen: tag 1.6.2.15-rc1 r295121 - /tags/1.6.2.15-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 15 13:29:07 CST 2010
Author: lmadsen
Date: Mon Nov 15 13:29:05 2010
New Revision: 295121
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=295121
Log:
Importing files for 1.6.2.15-rc1 release.
Added:
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tags/1.6.2.15-rc1/.version (with props)
tags/1.6.2.15-rc1/ChangeLog (with props)
Added: tags/1.6.2.15-rc1/.lastclean
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--- tags/1.6.2.15-rc1/ChangeLog (added)
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+2010-11-15 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.2.15-rc1
+
+2010-11-15 18:24 +0000 [r294988-295062] Tilghman Lesher <tlesher at digium.com>
+
+ * tests/test_expr.c (added), /: Merged revisions 295026 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 Nov 2010)
+ | 2 lines Create test verifying results of expression parser
+ ........
+
+ * funcs/func_curl.c: It is possible to crash Asterisk by feeding
+ the curl engine invalid data. (closes issue #18161) Reported by:
+ wdoekes Patches: 20101029__issue18161.diff.txt uploaded by
+ tilghman (license 14) Tested by: tilghman
+
+2010-11-12 21:14 +0000 [r294904-294910] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c: Return correct error code if lock path
+ fails. The recent changes to open_mailbox actually caused it to
+ be fixed, but let's be consistent. Reported by alecdavis in
+ asterisk-dev.
+
+ * apps/app_voicemail.c, /: Merged revisions 294903 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12
+ Nov 2010) | 16 lines Fix regression causing abort in voicemail
+ after opening a mailbox with no mesgs. In order to be more safe,
+ some error handling code was changed to respect more error
+ conditions including the potential memory allocation failure for
+ deleted and heard message tracking introduced in 293004. However,
+ last_message_index returns -1 for zero messages (perhaps as
+ expected) and was triggering the stricter error checking. Because
+ last_message_index is only called directly in one place, just
+ return 0 from open_mailbox (for file based storage) when no
+ messages are detected unless a real error has occurred. (closes
+ issue #18240) Reported by: leobrown Patches:
+ bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
+ Tested by: pabelanger ........
+
+2010-11-12 02:44 +0000 [r294822] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 294821 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11
+ Nov 2010) | 11 lines Asterisk is getting a "No D-channels
+ available!" warning message every 4 seconds. Asterisk is just
+ whining too much with this message: "No D-channels available!
+ Using Primary channel XXX as D-channel anyway!". Filtered the
+ message so it only comes out once if there is no D channel
+ available without an intervening D channel available period.
+ (closes issue #17270) Reported by: jmls ........
+
+2010-11-11 21:57 +0000 [r294639-294733] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 294688 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010)
+ | 18 lines Fix problem with qualify option packets for realtime
+ peers never stopping. The option packets not only never stopped,
+ but if a realtime peer was not in the peer list multiple options
+ dialogs could accumulate over time. This scenario has the
+ potential to progress to the point of saturating a link just from
+ options packets. The fix was to ensure that the poke scheduler
+ checks to see if a peer is in the peer list before continuing to
+ poke. The reason a peer must be in the peer list to be able to
+ properly manage an options dialog is because otherwise the call
+ pointer is lost when the peer is regenerated from the database,
+ which is how existing qualify dialogs are detected. (closes issue
+ #16382) (closes issue #17779) Reported by: lftsy Patches:
+ bug16382-3.patch uploaded by jpeeler (license 325) Tested by:
+ zerohalo ........
+
+ * main/asterisk.c, include/asterisk.h, main/pbx.c, /: Merged
+ revisions 294384 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010)
+ | 47 lines Fix a deadlock in device state change processing.
+ Copied from some notes from the original author (Russell):
+ Deadlock scenario: Thread 1: device state change thread Holds -
+ rdlock on contexts Holds - hints lock Waiting on channels
+ container lock Thread 2: SIP monitor thread Holds the "iflock"
+ Holds a sip_pvt lock Holds channel container lock Waiting for a
+ channel lock Thread 3: A channel thread (chan_local in this case)
+ Holds 2 channel locks acquired within app_dial Holds a 3rd
+ channel lock it got inside of chan_local Holds a local_pvt lock
+ Waiting on a rdlock of the contexts lock A bunch of other threads
+ waiting on a wrlock of the contexts lock To address this
+ deadlock, some locking order rules must be put in place and
+ enforced. Existing relevant rules: 1) channel lock before a pvt
+ lock 2) contexts lock before hints lock 3) channels container
+ before a channel What's missing is some enforcement of the order
+ when you involve more than any two. To fix this problem, I put in
+ some code that ensures that (at least in the code paths involved
+ in this bug) the locks in (3) come before the locks in (2). To
+ change the operation of thread 1 to comply, I converted the
+ storage of hints to an astobj2 container. This allows processing
+ of hints without holding the hints container lock. So, in the
+ code path that led to thread 1's state, it no longer holds either
+ the contexts or hints lock while it attempts to lock the channels
+ container. (closes issue #18165) Reported by: antonio ABE-2583
+ ........
+
+2010-11-10 23:16 +0000 [r294571] Tilghman Lesher <tlesher at digium.com>
+
+ * main/features.c: Actually pay attention to documented settings in
+ features.conf. (closes issue #16757) Reported by: voxter Patches:
+ 20101012__issue16757.diff.txt uploaded by tilghman (license 14)
+ Review: https://reviewboard.asterisk.org/r/994/
+
+2010-11-10 12:41 +0000 [r294500] Russell Bryant <russell at digium.com>
+
+ * main/devicestate.c: Improve a debug message to be more readable
+ and consistent. (closes issue #18282) Reported by: klaus3000
+ Patches: ast_devstate2str-patch.txt uploaded by klaus3000
+ (license 65)
+
+2010-11-09 20:27 +0000 [r294429] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Detect GMime properly on systems where
+ gmime flags and libs are configured with pkg-config. (closes
+ issue #16155) Reported by: jcollie Patches:
+ 20100917__issue16155.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+2010-11-08 22:30 +0000 [r294277-294312] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_timing_timerfd.c: add missing unlock not present in
+ 294277
+
+ * main/timing.c, main/channel.c, res/res_timing_timerfd.c,
+ include/asterisk/timing.h: Fix playback failure when using IAX
+ with the timerfd module. To fix this issue the alert pipe will
+ now be used when the timerfd module is in use. There appeared to
+ be a race that was not solved by adding locking in the timerfd
+ module, but needed to be there anyway. The race was between the
+ timer being put in non-continuous mode in ast_read on the channel
+ thread and the IAX frame scheduler queuing a frame which would
+ enable continuous mode before the non-continuous mode event was
+ read. This race for now is simply avoided. (closes issue #18110)
+ Reported by: tpanton Tested by: tpanton I put tested by tpanton
+ because it was tested on his hardware. Thanks for the remote
+ access to debug this issue!
+
+2010-11-08 20:50 +0000 [r294242] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Go off hold when we get an empty reinvite
+ telling us to. (closes issue 0014448) Reported by: frawd (closes
+ issue #17878) Reported by: frawd
+
+2010-11-05 00:06 +0000 [r293969] Shaun Ruffell <sruffell at digium.com>
+
+ * codecs/codec_dahdi.c, /: Merged revisions 293968 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04
+ Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio
+ when receiving unexpected frame sizes. dahdi-linux 2.4.0
+ (specifically commit 9034) added the capability for the wctc4xxp
+ to return more than a single packet of data in response to a
+ read. However, when decoding packets, codec_dahdi was still
+ assuming that the default number of samples was in each read. In
+ other words, each packet your provider sent you, regardless of
+ size, would result in 20 ms of decoded data (30 ms if decoding
+ G723). If your provider was sending 60 ms packets then
+ codec_dahdi would end up stripping 40 ms of data from each
+ transcoded frame resulting in "choppy" audio. This would only
+ affect systems where G729 packets are arriving in sizes greater
+ than 20ms or G723 packets arriving in sizes greater than 30ms.
+ DAHDI-744. ........
+
+2010-11-03 18:31 +0000 [r293806] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293805 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03
+ Nov 2010) | 20 lines Party A in an analog 3-way call would
+ continue to hear ringback after party C answers. All parties are
+ analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A
+ flash hooks to bring C into 3-way call before C answers. (A and B
+ hear ringback) 4) C answers 5) A continues to hear ringback
+ during the 3-way call. (All parties can hear each other.) * Fixed
+ use of wrong variable in dahdi_bridge() that stopped ringback on
+ the wrong subchannel. * Made several debug messages have more
+ information. A similar issue happens if B and C are SIP channels.
+ B continues to hear ringback. For some reason this only affects
+ v1.8 and trunk. * Don't start ringback on the real and 3-way
+ subchannels when creating the 3-way conference. Removing this
+ code is benign on v1.6.2 and earlier. ........
+
+2010-11-02 23:07 +0000 [r293723] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 293722 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010)
+ | 8 lines Add enabled/disabled information for rtautoclear sip
+ show settings output. When setting to zero/"no", the numeric
+ default was shown making it not obvious the disabled setting was
+ respected. (closes issue #18123) Reported by: zerohalo ........
+
+2010-11-02 21:26 +0000 [r293647] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293639 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02
+ Nov 2010) | 6 lines Make warning message have more useful
+ information in it. Change "Unable to get index, and nullok is not
+ asserted" to "Unable to get index for '<channel-name>' on channel
+ <number> (<function>(), line <number>)". ........
+
+2010-10-30 01:49 +0000 [r293340-293417] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 293416 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some more code that serves no purpose.
+ ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 293339 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29
+ Oct 2010) | 1 line Remove some code that serves no purpose.
+ ........
+
+2010-10-28 19:54 +0000 [r293195-293196] Tilghman Lesher <tlesher at digium.com>
+
+ * main/ast_expr2.c, main/ast_expr2.h: Merged revisions 293194 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........
+
+ * /, res/ael/ael.tab.c, main/ast_expr2.y, res/ael/ael_lex.c,
+ res/ael/ael.tab.h: Merged revisions 293194 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010)
+ | 5 lines "!00" evaluated as false, which is incorrect. Fixing.
+ Reported (though the reporter did not understand he was reporting
+ a bug) on the asterisk-users list:
+ http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html
+ ........
+
+2010-10-28 16:09 +0000 [r293158] Jeff Peeler <jpeeler at digium.com>
+
+ * funcs/func_strings.c: Fix infinite loop in FILTER(). Specifically
+ when you're using characters above \x7f or invalid character
+ escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes
+ Patches: issue18060_func_strings_filter_infinite_loop.patch
+ uploaded by wdoekes (license 717) Tested by: wdoekes
+
+2010-10-26 18:33 +0000 [r293118] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 293004 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25
+ Oct 2010) | 29 lines Fix inprocess_container in voicemail to
+ correctly restrict max messages. The comparison function logic
+ was off, so the number of sessions for a given mailbox were not
+ being incremented properly. This problem caused the maximum
+ number of messages per folder to not be respected when
+ simultaneously leaving multiple voicemails just below the
+ threshold. These problems should be fixed by the above, but just
+ in case: Fixed resequence_mailbox to rely on the actual number of
+ detected number of files in a directory rather than just assuming
+ only 10 messages more than the maximum had been left. Also if
+ more messages than the maximum are deleted they are actually
+ removed now. The second purpose of this commit should have been
+ separated out probably, but is related to the above. Again, if
+ the number of messages in a given voicemail folder exceeds the
+ maximum set limit make sure to allocate enough space for the
+ deleted and heard index tracking array. A few random fixes: There
+ was a forgotten decrement of the inprocess count in
+ imap_store_file. When using IMAP storage, do not look in the
+ directory where file based storage messages may still reside and
+ influence the message count. Ensure to use only the first format
+ in sendmail. ABE-2516 ........
+
+2010-10-25 19:06 +0000 [r292867] David Vossel <dvossel at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 292866 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25
+ Oct 2010) | 27 lines This patch turns chan_local pvts into
+ astobj2 objects. chan_local does some dangerous things involving
+ deadlock avoidance. tech_pvt functions like hangup and
+ queue_frame are provided with a locked channel upon entry. Those
+ functions are completely safe as long as you don't attempt to
+ give up that channel lock, but that is impossible to guarantee
+ due to the required deadlock avoidance necessary to lock both the
+ tech_pvt and both channels involved. In the past, we have tried
+ to account for this by doing things like setting a "glare" flag
+ that indicates what function should destroy the pvt. This was
+ used in local_hangup and local_queue_frame to decided who should
+ destroy the pvt if they collided in separate threads. I have
+ removed the need to do this by converting all chan_local
+ tech_pvts to astobj2. This means we can ref a pvt before deadlock
+ avoidance and not have to worry about that pvt possibly getting
+ destroyed under us. It also cleans up where we destroy the
+ tech_pvt. The only unlink from the tech_pvt container occurs in
+ local_hangup now, which is where it should occur. Since there
+ still may be thread collisions on some functions like
+ local_hangup after deadlock avoidance, I have added some checks
+ to detect those collisions and exit appropriately. I think this
+ patch is going to solve quite a bit of weirdness we have had with
+ local channels in the past. ........
+
+2010-10-22 21:16 +0000 [r292786] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/asterisk.ldif, channels/chan_sip.c,
+ configs/res_ldap.conf.sample: Update the LDIF file for LDAP. The
+ LDIF file asterisk.ldif was quite a bit out of date from the
+ asterisk.ldap-schema file, so I've now updated that to be in
+ sync. The asterisk.ldif file being out of sync was a problem on
+ my systems where I was doing an ldapadd to import the schema into
+ the LDAP database, and the existing file would cause problems and
+ ERROR messages when registering. Additional documention has been
+ added based on feedback in the issue I'm closing. (closes issue
+ #13861) Reported by: scramatte Patches: ldap-update.txt uploaded
+ by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec,
+ rgenthner
+
+2010-10-21 13:11 +0000 [r292556] Leif Madsen <lmadsen at digium.com>
+
+ * configs/res_ldap.conf.sample: Change res_ldap.sample.conf to
+ match the schema. (closes issue #17376) Reported by: jcovert
+ Patches: res_ldap.conf.sample.patch uploaded by jcovert (license
+ 551)
+
+2010-10-21 00:05 +0000 [r292412] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * apps/app_dial.c, /: Merged revisions 292411 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct
+ 2010) | 10 lines Record priv-recordintro as sln, not gsm This
+ removes the gsm->sln step when transcoding priv-recordintro.
+ (closes issue #18176) Reported by: pabelanger Patches:
+ chan_sip.diff uploaded by pabelanger (license 224) ........
+
+2010-10-18 22:01 +0000 [r292229] Leif Madsen <lmadsen at digium.com>
+
+ * sounds/Makefile: Fix typo in the sounds/Makefile. (Issue #17426)
+
+2010-10-18 21:54 +0000 [r292226] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 292223 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18
+ Oct 2010) | 11 lines Fix improper operator key acceptance and
+ clean up temp recording files. This is a fix for when pressing
+ the operator key after recording an unavailable, busy, name, or
+ temporary message in mailbox options. The operator key should not
+ be accepted here, but should be allowed during the message
+ recording. If the operator key is pressed during ensure the file
+ is saved or deleted as apporopriate. Also, ensure removal of
+ temporary recorded files after an early hang up or when message
+ acceptance confirmation times out. ABE-2518 ........
+
+2010-10-18 21:50 +0000 [r292224] Leif Madsen <lmadsen at digium.com>
+
+ * sounds/Makefile, /, sounds/sounds.xml: Merged revisions 292222
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
+ | 9 lines Add support for the new English (Australian Accent)
+ sound files. (closes issue #17426) Reported by: camsown Patches:
+ core-sounds-en_AU.txt uploaded by camsown (license 1050)
+ add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
+ by: camsown, lmadsen, jtodd, qwell ........
+
+2010-10-16 10:03 +0000 [r292049] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * res/res_musiconhold.c, configs/musiconhold.conf.sample: Base
+ directory for MOH should be ASTDATADIR If the directive
+ 'directory' is relative, make it relative to the datadir, rather
+ than to the varlibdir. In the sample configuration it is relative
+ ('moh'). This has no effect unless you have actively set the
+ datadir explicitly (at build time or at run time). (closes issue
+ #16906) Patches: moh_datadir uploaded by tzafrir (license 46)
+ Review: https://reviewboard.asterisk.org/r/974/
+
+2010-10-15 19:35 +0000 [r291939] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291938 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, 15 Oct
+ 2010) | 2 lines Clean up formatting. ........
+
+2010-10-15 16:16 +0000 [r291904] Terry Wilson <twilson at digium.com>
+
+ * res/res_jabber.c: Don't crash or deadlock on module unload We
+ can't hold the lock while pthread_join is called since
+ aji_log_hook will attempt to lock from the other therad. We
+ reorder the pthread_join and ast_aji_disconnect so that we don't
+ do an SSL_read() while SSL_shutdown is running, causing a crash.
+
+2010-10-13 23:36 +0000 [r291655] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 291643 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13
+ Oct 2010) | 20 lines Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/ ........
+
+2010-10-13 22:58 +0000 [r291580] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c, /: Merged revisions 291577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+ | 21 lines Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+ ........
+
+2010-10-13 15:29 +0000 [r291393] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+ | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+ a frame. This fixes a crash due to a hangup race condition.
+ ABE-2601 ........
+
+2010-10-12 17:20 +0000 [r291280] Leif Madsen <lmadsen at digium.com>
+
+ * configs/phoneprov.conf.sample: Add undocumented variables to
+ phoneprov.conf.sample (closes issue #18107) Reported by: lathama
+ Patches: phoneprov.conf.sample.diff uploaded by lathama (license
+ 1028)
+
+2010-10-12 17:05 +0000 [r291264] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/acl.c: Merged revisions 291263 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Oops, incorrect range (although unallocated at ARIN)
+ ........
+
+2010-10-12 16:07 +0000 [r291229] Leif Madsen <lmadsen at digium.com>
+
+ * configs/manager.conf.sample: Add documention that mentions
+ options are defined but not used. (Issue #18101)
+
+2010-10-11 18:39 +0000 [r291073-291111] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Make exit from handle_request_do()
+ consistent.
+
+ * /, channels/chan_sip.c: Merged revisions 291109 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010)
+ | 1 line Add missing unlock to an exception condition in
+ reload_config(). ........
+
+ * main/cli.c: Fixed infinite loop in verbose/debug message output.
+ Setting the module/filename specific message level and then
+ changing it resulted in the linked list being looped on itself.
+ Traversing this linked list is an infinite loop if what you are
+ looking for is not in the list. Also plugged some CLI parsing
+ holes in the associated CLI command: * Removing a nonexistent
+ module from the list actually added it with a level of zero. *
+ Setting the non-module specific level to zero is now equivalent
+ to setting it to "off" as documented.
+
+2010-10-08 02:45 +0000 [r290863] Jeff Peeler <jpeeler at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 290862 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+ | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+ at control console. A recent change was made to avoid a race
+ condition on shutdown which only called the end functions from
+ the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls ........
+
+2010-10-07 20:57 +0000 [r290751] Jason Parker <jparker at digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 290750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+ 9 lines Allow PRI to build properly when using --with-pri. Use
+ the directories found for the parent when using lib dependencies.
+ (closes issue #17314) Reported by: tzafrir Patches:
+ 17314-withdeps.diff uploaded by qwell (license 4) ........
+
+2010-10-07 10:53 +0000 [r290712] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/
+
+2010-10-06 13:48 +0000 [r290396-290575] Tilghman Lesher <tlesher at digium.com>
+
+ * main/file.c: Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque
+
+ * res/res_jabber.c, /: Merged revisions 290392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........
+
+2010-10-05 19:54 +0000 [r290375] David Vossel <dvossel at digium.com>
+
+ * apps/app_directed_pickup.c: Fixes PickupChan() not working with
+ full channel name. (closes issue #18011) Reported by: schern
+ Patches: app_directed_pickup.c.2.patch uploaded by schern
+ (license 995) app_directed_pickup.c.trunk.patch uploaded by
+ schern (license 995) Tested by: schern, dvossel
+
+2010-10-05 17:42 +0000 [r290324] Richard Mudgett <rmudgett at digium.com>
+
+ * contrib/valgrind.supp, /: Merged revisions 290323 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r290323 | rmudgett | 2010-10-05 12:41:18 -0500
+ (Tue, 05 Oct 2010) | 11 lines Merged revision 258974 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r258974
+ | diruggles | 2010-04-26 14:05:47 -0500 (Mon, 26 Apr 2010) | 4
+ lines Line 24 missed in compatibility fix in revision 233577
+ added a "fun:" prefix line 24 .......... ................
+
+2010-10-04 23:14 +0000 [r290101-290254] Tilghman Lesher <tlesher at digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
+ pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5:
+ Change new pattern matcher to regard dashes the same as the old
+ pattern matcher -- as visual candy to be ignored. Also change the
+ AEL parser to not generate dashes within extensions, as those
+ dashes would be ignored. Update the AEL tests to match this
+ behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman
+
+ * /, configure, configure.ac: Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+
+ * /, configure, configure.ac: Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+
+2010-10-02 08:52 +0000 [r289950] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c, /: Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........
+
+2010-10-02 04:45 +0000 [r289874] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01
+ Oct 2010) | 8 lines When forwarding a message, a prepend means
+ that the filesystem will always have a better copy. (closes issue
+ #17803) Reported by: dpetersen Patches:
+ 20100923__issue17803.diff.txt uploaded by tilghman (license 14)
+ Tested by: dpetersen ........
+
+2010-10-01 23:01 +0000 [r289798] Jeff Peeler <jpeeler at digium.com>
+
+ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h:
+ Merged revisions 289797 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+
+2010-10-01 17:09 +0000 [r289704] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289703 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+
+2010-10-01 16:21 +0000 [r289700] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289699 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........
+
+2010-10-01 09:42 +0000 [r289622] schmitds <schmitds at localhost>:
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 19:51 +0000 [r289553] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Properly handle channel allocation failures
+ duing invites with replaces. ABE-2588
+
+2010-09-30 17:09 +0000 [r289501] Brett Bryant <bbryant at digium.com>
+
+ * /, res/res_agi.c: Merged revisions 289500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289500 | bbryant | 2010-09-30 13:08:20 -0400 (Thu, 30 Sep 2010)
+ | 11 lines res_agi.c:handle_getvariablefull() could recursively
+ lock a channel and not release it if an argument is the current
+ channel's name. (closes issue #17970) Reported by: mdu113
+ Patches: res_agi.c.diff3 uploaded by mdu113 (license 582) Tested
+ by: mdu113 Review: https://reviewboard.asterisk.org/r/947/
+ ........
+
+2010-09-30 15:37 +0000 [r289425] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_sms.c: Merged revisions 289424 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........
+
+2010-09-29 21:03 +0000 [r289339] Jason Parker <jparker at digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........
+
+2010-09-29 20:24 +0000 [r289334] Leif Madsen <lmadsen at digium.com>
+
+ * configs/res_ldap.conf.sample: Update sample documentation to note
+ md5secret requirements.
+
+2010-09-29 20:15 +0000 [r289332] Russell Bryant <russell at digium.com>
+
+ * res/res_config_ldap.c: Don't completely ignore md5secret from
+ LDAP if the value does not begin with {md5}. This fixes a problem
+ that lmadsen ran in to where md5secret was not working for him.
+
+2010-09-29 15:04 +0000 [r289178] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, /: Merged revisions 289177 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
+ 2010) | 8 lines Set the caller id on CDRs when it is set on the
+ parent channel. (closes issue #17569) Reported by: tbelder
+ Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
+ tbelder ........
+
+2010-09-28 18:14 +0000 [r289095] Brett Bryant <bbryant at digium.com>
+
+ * main/channel.c, /: Merged revisions 289094 via svnmerge from
[... 27610 lines stripped ...]
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