[svn-commits] tilghman: tag 1.4.23-rc2 r160145 - /tags/1.4.23-rc2/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Dec 1 16:22:05 CST 2008
Author: tilghman
Date: Mon Dec 1 16:22:05 2008
New Revision: 160145
URL: http://svn.digium.com/view/asterisk?view=rev&rev=160145
Log:
Importing files for 1.4.23-rc2 release
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tags/1.4.23-rc2/.version (with props)
tags/1.4.23-rc2/ChangeLog (with props)
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+2008-12-01 Tilghman Lesher <tilghman at digium.com>
+
+ * Released 1.4.23-rc2
+
+2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
+ iax2_setoption(), as well, since they both have the potential to
+ send control frames in the middle of call setup. We have to wait
+ until we have received a message back from the remote end before
+ we try to send any more frames. Otherwise, the remote end will
+ consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
+
+2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/manager.c: Get rid of the useless format string and argument
+ in the Bogus/ manager channelname. Noted by kpfleming and name
+ Bogus/manager suggested by eliel
+
+2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell at digium.com>
+
+ * .cleancount: Force a "make clean" to avoid a bizarre build issue
+ ...
+
+2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/manager.c: make manager compile on OpenBSD. The last (10th)
+ argument to ast_channel_alloc here should be a pointer and NULL
+ is not really a pointer.
+
+2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/enum.c, utils/frame.c, configure, res/res_agi.c,
+ include/asterisk/module.h, main/logger.c, main/dns.c,
+ include/asterisk/threadstorage.h, include/asterisk/utils.h,
+ include/asterisk/devicestate.h, channels/chan_sip.c,
+ include/asterisk/dundi.h, main/jitterbuf.c,
+ channels/chan_agent.c, configure.ac, utils/astman.c,
+ include/asterisk/cli.h, include/asterisk/channel.h,
+ include/jitterbuf.h, include/asterisk/manager.h,
+ main/ast_expr2.c, Makefile, include/asterisk/logger.h,
+ include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
+ include/asterisk/linkedlists.h, include/asterisk/lock.h,
+ include/asterisk/strings.h, makeopts.in,
+ include/asterisk/stringfields.h, utils/check_expr.c,
+ channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
+ update dev-mode compiler flags to match the ones used by default
+ on Ubuntu Intrepid, so all developers will see the same warnings
+ and errors since this branch already had some printf format
+ attributes, enable checking for them and tag functions that
+ didn't have them format attributes in a consistent way
+
+2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_oss.c, channels/busy.h (removed),
+ channels/ring_tone.h (added), channels/chan_alsa.c,
+ channels/ringtone.h (removed), channels/busy_tone.h (added),
+ channels/Makefile: rename these files so as to avoid conflicts
+ when users update their working copies and have unversioned files
+ already in place
+
+ * channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
+ Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
+ (added), channels/Makefile: simplify (and slightly bug-fix) the
+ recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
+ and channels/ringtone.h to the repository instead of generating
+ them repeatedtly; most users do not change the settings to build
+ them, but the Makefile rules are still there if they wish to do
+ so ensure that 'make clean' removes dependency files for .i files
+ that are created in COMPILE_DOUBLE mode
+
+2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf at digium.com>
+
+ * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
+ by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
+ Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
+ fix. the change to cdr.c allows no-answer to percolate up into
+ CDR's, and feels like the right place to locate this fix; if BUSY
+ is done here, no-answer should be, too.
+
+2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
+ (closes issue #13919) Reported by: barthpbx Patches:
+ chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
+ barthpbx
+
+ * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
+ Nov 2008) | 7 lines Regression fix for last security fix. Set the
+ iseqno correctly. (closes issue #13918) Reported by: ffloimair
+ Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ffloimair ........
+
+2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
+ go along with ao2_lock() and ao2_unlock()
+
+2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson at digium.com>
+
+ * apps/app_festival.c: Add missing variable declaration in the PPC
+ code
+
+2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: System call ioperm is non-portable, so check for
+ its existence in autoconf. (Closes issue #13863)
+
+2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler at digium.com>
+
+ * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
+ issue #13786) Reported by: tzafrir When compiling against Zaptel
+ dahdi_compat will now only define all the DAHDI defines if the
+ Zaptel define is present. Also, there is no such thing as
+ DAHDI_PRI.
+
+2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf at digium.com>
+
+ * res/res_features.c: In reference to the fix made for 13871, I was
+ merging the fix into 1.6.0 and realized I missed the code in the
+ h-exten block, and didn't catch it because my test case had the
+ h-exten commented out. So, this corrects the code I missed, as a
+ preventative against another crash report. Tested with the
+ h-exten defined, all is well.
+
+2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: The passed extension may not be the same in the list
+ as the current entry, because we strip spaces when copying the
+ extension into the structure. Therefore, use the copied item to
+ place the item into the list. (found by lmadsen on -dev, fixed by
+ me)
+
+2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell at digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: When compiling with
+ DEBUG_THREADS, report the real file/func/line for
+ ao2_lock/ao2_unlock
+
+2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf at digium.com>
+
+ * res/res_features.c: (closes issue #13871) Reported by: mdu113
+ This one is totally my fault. The code doesn't even create a
+ bridge if the channel CDR has POST_DISABLED. I didn't check for
+ that at the end of the bridge. Fixed with a few small insertions.
+ Tested. Looks good. No cdr generated, no crash, no unnecc. data
+ objects created either.
+
+2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: This change had somehow gotten reverted due to
+ a completely unrelated commit. Thanks to Theo Belder on the
+ Asterisk-dev list for pointing this out.
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: There was an issue when attempting to
+ reference an embedded frame in a freed ast_filestream. This patch
+ makes use of the ao2 functions to make sure that we do not free
+ an ast_filestream structure until the embedded ast_frame has been
+ "freed" as well. (closes issue #13496) Reported by: fst-onge
+ Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
+ 60) Tested by: putnopvut Closes AST-89
+
+ * channels/chan_sip.c: We don't handle 4XX responses to BYE well.
+ According to section 15 of RFC 3261, we should terminate a dialog
+ if we receive a 481 or 408 in response to our BYE. Since I am
+ aware of at least one phone manufacturer who may sometimes send a
+ 404 as well, I am being liberal and saying that any 4XX response
+ to a BYE should result in a terminated dialog. (closes issue
+ #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
+ putnopvut (license 60) Closes AST-129
+
+ * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
+ cause on the calling channel in the case that ast_call() fails.
+ For incoming SIP channels, this was causing us to send a 603
+ instead of a 486 when the call-limit was reached on the
+ destination channel. (closes issue #13867) Reported by: still_nsk
+ Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
+ blitzrage
+
+2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+
+2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
+ channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
+ main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
+ codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
+ channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
+ res/snmp, res/Makefile: the gcc optimizer frequently finds broken
+ code (use of uninitalized variables, unreachable code, etc.),
+ which is good. however, developers usually compile with the
+ optimizer turned off, because if they need to debug the resulting
+ code, optimized code makes that process very difficult. this
+ means that we get code changes committed that weren't adequately
+ checked over for these sorts of problems. with this build system
+ change, if (and only if) --enable-dev-mode was used and
+ DONT_OPTIMIZE is turned on, when a source file is compiled it
+ will actually be preprocessed (into a .i or .ii file), then
+ compiled once with optimization (with the result sent to
+ /dev/null) and again without optimization (but only if the first
+ compile succeeded, of course). while making these changes, i did
+ some cleanup work in Makefile.rules to move commonly-used
+ combinations of flag variables into their own variables, to make
+ the file easier to read and maintain
+
+2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add some missing invite state changes
+ necessary in the sip_write function. Not setting the invite state
+ correctly on the call was resulting in the Record application
+ leaving empty files. I also have updated the doxygen comment next
+ to the declaration of the INV_EARLY_MEDIA constant to reflect
+ that we also use this state when we *send* a 18X response to an
+ INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
+ sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
+ (license 162) Tested by: putnopvut
+
+2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
+ This fix is the result of a bug fix in ast_app_separate_args
+ r124395. If an argument does not exist it should always be set to
+ a null string rather than a null pointer.
+
+2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, res/res_features.c,
+ include/asterisk/channel.h, apps/app_followme.c: Fix a crash in
+ the end_bridge_callback of app_dial and app_followme which would
+ occur at the end of an attended transfer. The error occurred
+ because we initially stored a pointer to an ast_channel which
+ then was hung up due to a masquerade. This commit adds a "fixup"
+ callback to the bridge_config structure to allow for
+ end_bridge_callback_data to be changed in the case that a new
+ channel pointer is needed for the end_bridge_callback.
+
+2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile, Makefile.rules: when an individual directory dist-clean
+ is run, run clean in that directory first, and when running
+ top-level dist-clean, do not run subdirectory clean operations
+ twice
+
+ * Makefile.moddir_rules: dist-clean should remove dependency
+ information files as well
+
+ * contrib/asterisk-ng-doxygen: major update to doxygen
+ configuration file: 1) update to doxygen 1.5.x style file, as
+ used in trunk 2) tell doxygen where are header files are, so
+ include-file processing can be done 3) make all macros that are
+ used to define variables/functions be expanded, so that doxygen
+ will properly document the resulting variable/function 4) make
+ all macros that are used to provide the contents of a variable
+ (structure) be expanded, so that doxygen will be able to document
+ the resulting fields 5) suppress compiler attributes
+ (__attribute__(xxx)) from being seen by doxygen, so it will
+ properly match up function definition and usage (for an example
+ of th effect of this, look at the doxygen docs for ast_log() from
+ before and afte this commit)
+
+2008-11-14 15:18 +0000 [r156816] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: If the prompt to reenter a voicemail
+ password timed out, it resulted in the password not being saved,
+ even if the input matched what you gave when first prompted to
+ enter a new password. This is because the return value of
+ ast_readstring was checked, but not checked properly. This bug
+ was discovered by Jared Smith during an Asterisk training course.
+ Thanks for reporting it!
+
+2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_while.c: ast_waitfordigit() requires that the channel be
+ up, for no good logical reason. This prevents While/EndWhile from
+ working within the "h" extension. Reported by: jgalarneau (for
+ ABE C.2) Fixed by: me
+
+ * main/manager.c: Provide more space for all the data which can
+ appear in an originating channel name. (closes issue #13398)
+ Reported by: bamby Patches: manager.c.diff uploaded by bamby
+ (license 430)
+
+2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configure, autoconf/ast_gcc_attribute.m4: revert this change...
+ non-functional changes don't belong here
+
+ * configure, autoconf/ast_gcc_attribute.m4: correct minor syntax
+ error... no functional change
+
+2008-11-12 21:18 +0000 [r156386] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_dial.c: When using call limits under 1 second, infinite
+ call lengths are allowed, instead. (closes issue #13851) Reported
+ by: ruddy
+
+2008-11-12 19:36 +0000 [r156297] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c: It turns out that the 0x0XX00 codes being returned
+ for N, X, and Z are off by one, as per conversation with jsmith
+ on #asterisk-dev; he was teaching a class and disconcerted that
+ this published rule was not being followed, with patterns _NXX,
+ _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
+ have been. This change, tested on these 3 patterns now picks the
+ proper one. However, this change may surprise users who set up
+ dialplans based on previous behavior, which has been there for
+ what, 2 and half years or so now.
+
+2008-11-12 19:26 +0000 [r156294] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_meetme.c: If the SLA thread is not started, then reload
+ causes a memory leak. (closes issue #13889) Reported by: eliel
+ Patches: app_meetme.c.patch uploaded by eliel (license 64)
+
+2008-11-12 19:10 +0000 [r156289] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: For whatever reason, gcc only warned me about
+ the possible use of an uninitialized variable when compiling
+ 1.6.1.
+
+2008-11-12 18:39 +0000 [r156229] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c: Revert revision 132506, since it
+ occasionally caused IAX2 HANGUP packets not to be sent, and
+ instead, schedule a task to destroy the iax2 pvt structure 10
+ seconds later. This allows the IAX2 HANGUP packet to be queued,
+ transmitted, and ACKed before the pvt is destroyed. (closes issue
+ #13645) Reported by: dzajro Patches:
+ 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
+
+2008-11-12 17:53 +0000 [r156178] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_meetme.c: (closes issue #13173) Reported by: pep This
+ change adds an announce_thread responsible for playing
+ announcements to an existing conference. This allows all
+ announcing to be immediately stopped if necessary but more
+ importantly allows other threads that need to play something to
+ not block. There are multiple benefits to this, but the actual
+ bug is for solving the scenario for a channel to be unusable
+ after hang up for the entire duration of the parting
+ announcement. The parting announcement can be extremely long
+ depending on what the user recorded upon joining the conference.
+ Reviewed by Russell on Review Board:
+ http://reviewboard.digium.com/r/25/
+
+2008-11-12 17:38 +0000 [r156167] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c: When doing some tests, I was having a crash at
+ the end of every call if an attended transfer occurred during the
+ call. I traced the cause to the CDR on one of the channels being
+ NULL. murf suggested a check in the end bridge callback to be
+ sure the CDR is non-NULL before proceeding, so that's what I'm
+ adding.
+
+2008-11-12 17:29 +0000 [r156164] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Move the sanity check that makes sure "always
+ fork" is not set along with the console option to be after the
+ code that reads options from asterisk.conf. This resolves a
+ situation where Asterisk can start taking up 100% when
+ misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting
+ me log in to his system to figure out what was causing the 100%
+ CPU problem.)
+
+2008-11-10 21:07 +0000 [r155861] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_agent.c: Channel drivers assume that when their
+ indicate callback is invoked, that the channel on which the
+ callback was called is locked. This patch corrects an instance in
+ chan_agent where a channel's indicate callback is called directly
+ without first locking the channel. This was leading to some
+ observed locking issues in chan_local, but considering that all
+ channel drivers operate under the same expectations, the generic
+ fix in chan_agent is the right way to go. AST-126
+
+2008-11-10 20:49 +0000 [r155803] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/valgrind.txt: I got tired of saying this in every single
+ bugnote referring to this file.
+
+2008-11-09 01:08 +0000 [r155553] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h,
+ apps/app_followme.c: Use static functions here instead of nested
+ ones. This requires a small change to the ast_bridge_config
+ struct as well. To understand the reason for this change, see the
+ following post:
+ http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
+
+2008-11-07 22:27 +0000 [r155398] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Clarify error message. (closes issue #13809)
+ Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: denke
+
+2008-11-06 19:45 +0000 [r155011] Mark Michelson <mmichelson at digium.com>
+
+ * configs/voicemail.conf.sample: The documentation listed the
+ ability to set 'maxmsg' per context. The truth is that you can
+ only set this in the general section or per mailbox. Thus I am
+ updating the sample config file to be more accurate. Thanks to
+ sasargen on IRC for bringing up this issue.
+
+2008-11-05 16:44 +0000 [r154724] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_agent.c: The logic of a strcasecmp call was
+ reversed (closes issue #13841) Reported by: clegall_proformatique
+
+2008-11-05 16:06 +0000 [r154685] Steve Murphy <murf at digium.com>
+
+ * main/channel.c: This fix was prompted by communication from user,
+ who was seeing thousands of error logs... looks like EAGAIN. Made
+ such uninteresting.
+
+2008-11-04 20:49 +0000 [r154365] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c: On busy systems, it's possible for the
+ values checked within a single line of code to change, unless the
+ structure is locked to ensure a consistent state. (closes issue
+ #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: kowalma
+
+2008-11-04 19:01 +0000 [r154266] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to
+ the wrong state when it receives the indication
+ AST_CONTROL_RINGING.
+
+2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_h323.c: Make the monitor thread non-detached, so it
+ can be joined (suggested by Russell on -dev list).
+
+ * apps/app_voicemail.c: Attempting to expunge a mailbox when the
+ mailstream is NULL will crash Asterisk. (Closes issue #13829)
+ Reported by: jaroth Patch by: me (modified jaroth's patch)
+
+ * main/rtp.c: Remove the potential for a division by zero error.
+ (Closes issue #13810)
+
+2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c,
+ main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c,
+ res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in
+ this branch on the first pass
+
+2008-11-02 19:51 +0000 [r153651] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/features.h: features.h depends on linkedlists.h,
+ so include it
+
+2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming <kpfleming at digium.com>
+
+ * utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c,
+ funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c,
+ channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c,
+ pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c,
+ pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c,
+ formats/format_wav.c, apps/app_festival.c,
+ main/db1-ast/hash/hash_page.c, main/translate.c,
+ res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c,
+ utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c,
+ pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h,
+ apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c,
+ utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c,
+ apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y,
+ apps/app_authenticate.c, formats/format_wav_gsm.c,
+ res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of
+ potential problems found by gcc 4.3.x, primarily bare strings
+ being passed to printf()-like functions and ignored results from
+ read()/write() and friends
+
+2008-10-31 22:36 +0000 [r153270] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c, apps/app_followme.c: Add end_bridge_callback
+ for app_follome and add AUTOLOOP flag to res_features
+
+2008-10-31 Tilghman Lesher <tlesher at digium.com>
+
+ * Asterisk 1.4.23-rc1 released.
+
+2008-10-31 16:30 +0000 [r153114] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Turn off qualify on uncached realtime peers.
+ (Closes issue #13383)
+
+2008-10-31 15:45 +0000 [r153095] Terry Wilson <twilson at digium.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h:
+ Recent CDR fixes moved execution of the 'h' exten into the
+ bridging code, so variables that were set after ast_bridge_call
+ was called would not show up in the 'h' exten. Added a callback
+ function to handle setting variables, etc. from w/in the bridging
+ code. Calls back into a nested function within the function
+ calling ast_bridge_call (closes issue #13793) Reported by:
+ greenfieldtech
+
+2008-10-30 20:58 +0000 [r152992] Sean Bright <sean.bright at gmail.com>
+
+ * bootstrap.sh: The -I argument to aclocal needs a space before the
+ include directory name.
+
+2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_h323.c: Cannot join detached threads. See
+ http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
+ (Closes issue #13400)
+
+ * channels/chan_local.c: Unlock before returning, when extension
+ doesn't exist. (closes issue #13807) Reported by: eliel Patches:
+ chan_local.c.patch uploaded by eliel (license 64)
+
+2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/cdr.c: instead of comparing the string pointer to 0, let's
+ compare the value that was actually parsed out of the string
+ (found by sparse)
+
+2008-10-29 05:23 +0000 [r152539] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
+ issue #13795) Reported by: andrew53 Patches:
+ chan_sip_sizeof.patch uploaded by andrew53 (license 519)
+
+2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy <murf at digium.com>
+
+ * configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c:
+ A little documentation cross-ref between features and dial and
+ queue... I wasted some time (stupidly) trying to get the
+ one-touch parking stuff working, because it didn't occur to me
+ that I had to also have the corresponding options in the dial
+ command! Duh! (In all this time, I never set this up before!) So,
+ to keep some poor fool from suffering the same fate, I made the
+ features.conf.sample file mention the corresponding opts in
+ dial/queue; and the docs for dial/app specifically mention the
+ corresponding decls in the feature.conf file. I hope this doesn't
+ spoil some vast, eternal plan...
+
+ * apps/app_dial.c, res/res_features.c, funcs/func_channel.c,
+ include/asterisk/pbx.h, apps/app_queue.c: The magic trick to
+ avoid this crash is not to try to find the channel by name in the
+ list, which is slow and resource consuming, but rather to pay
+ attention to the result codes from the ast_bridge_call, to which
+ I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are
+ returned when a channel is parked. If you get AST_PBX_KEEPALIVE,
+ then don't touch the channel pointer. If you get
+ AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
+ don't touch the peer pointer. Updated the several places where
+ the results from a bridge were not being properly obeyed, and
+ fixed some code I had introduced so that the results of the
+ bridge were not overridden (in trunk). All the places that
+ previously tested for AST_PBX_NO_HANGUP_PEER now have to check
+ for both AST_PBX_NO_HANGUP_PEER and
+ AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
+ parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
+ while A is getting the parking slot announcement, immediately
+ after being put on hold. 2. A calls B; B answers; A parks B; B
+ hangs up after A has been hung up, but before the park times out.
+ 3. A calls B; B answers; B parks A; A hangs up while B is getting
+ the parking slot announcement, immediately after being put on
+ hold. 4. A calls B; B answers; B parks A; A hangs up after B has
+ been hung up, but before the park times out. No crash. I also ran
+ the scenarios above against valgrind, and accesses looked good.
+
+2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Quoting in the wrong direction (Fixes
+ AST-107)
+
+ * apps/app_dial.c: Reset all DIAL variables back to blank, in case
+ Dial is called multiple times per call (which could otherwise
+ lead to inconsistent status reports). (closes issue #13216)
+ Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: ruddy
+
+2008-10-27 23:28 +0000 [r152286] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Buffer policy setting for half is not
+ needed.
+
+2008-10-27 21:32 +0000 [r152215] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_local.c: Inherit ALL elements of CallerID across a
+ local channel. (closes issue #13368) Reported by: Peter Schlaile
+ Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
+ (license 14)
+
+2008-10-26 20:23 +0000 [r152059] Sean Bright <sean.bright at gmail.com>
+
+ * funcs/func_strings.c: Since passing \0 as the second argument to
+ strchr is valid (and will match the trailing \0 of a string) we
+ need to check that first, otherwise we end up with incorrect
+ results. Fix suggested by reporter. (closes issue #13787)
+ Reported by: meitinger
+
+2008-10-25 10:59 +0000 [r151905] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Move AMI initialization to occur after loading
+ modules. This prevents a deadlock when someone tries to initiate
+ a module reload from the AMI just as Asterisk is starting.
+ (closes issue #13778) Reported by: hotsblanc Fix suggested by
+ hotsblanc
+
+2008-10-23 16:04 +0000 [r151763] Terry Wilson <twilson at digium.com>
+
+ * configs/features.conf.sample, res/res_features.c, CHANGES:
+ Backport fix from 1.6.0 that allows you to set
+ parkedcalltransfers=no|caller|callee|both, but default to both
+ which would be the equivalent of the existing behaviour. The
+ problem was that if someone parked a call, the callee and caller
+ would both get assigned the builtin transfer feature, which would
+ not only be potentially giving someone the ability to transfer
+ themselves when they shouldn't have it, but would also dissallow
+ reinviting the media off of the call. (closes issue #12854)
+ Reported by: davidw Patches: parkingfix4.diff.txt uploaded by
+ otherwiseguy Tested by: davidw, otherwiseguy
+
+2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming <kpfleming at digium.com>
+
+ * autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4,
+ configure.ac: rename this macro to properly reflect what it does
+
+ * autoconf/ast_check_pwlib.m4 (added), autoconf (added),
+ autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4
+ (added), configure, autoconf/ast_gcc_attribute.m4 (added),
+ bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added),
+ autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added),
+ autoconf/ast_c_compile_check.m4 (added),
+ autoconf/ast_c_define_check.m4 (added),
+ autoconf/ast_prog_egrep.m4 (added),
+ autoconf/ast_check_openh323.m4 (added),
+ autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4
+ (added), acinclude.m4 (removed): break up acinclude.m4 into
+ individual files, which will make it easier to maintain, easier
+ to add new macros (less patching) and will ease maintenance of
+ these macros across Asterisk branches
+
+2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke <bweschke at btwtech.com>
+
+ * main/asterisk.c: As per kpfleming's comments to the prior commit,
+ I'm reverting some of the changes here. A comment was made in bug
+ #13726 "3. The same mistake as in (2) is done in a few other
+ places in the code that check for: #if defined(HAVE_ZAPTEL) ||
+ defined(HAVE_DAHDI) Harmless, but still incorrect." In the case
+ of main/asterisk.c, this is not incorrect because without
+ HAVE_ZAPTEL defined, we're missing the include for ioctl and the
+ namespace that defines DAHDI_TIMERCONFIG which is still required
+ when using Zaptel with the 1.4 branch.
+
+ * main/asterisk.c: Fix the 1.4 branch compile again broken with
+ r150557 when using with Zaptel and not DAHDI (closes issue
+ #13740) reported by: jmls patch by: bweschke
+
+2008-10-18 01:42 +0000 [r150816] BJ Weschke <bweschke at btwtech.com>
+
+ * main/manager.c: Using the GetVar handler in AMI is potentially
+ dangerous (insta-crash [tm]) when you use a dialplan function
+ that requires a channel and then you don't provide one or provide
+ an invalid one in the Channel: parameter. We'll handle this
+ situation exactly the same way it was handled in pbx.c back on
+ r61766. We'll create a bogus channel for the function call and
+ destroy it when we're done. If we have trouble allocating the
+ bogus channel then we're not going to try executing the function
+ call at all and run the risk of crashing. (closes issue #13715)
+ reported by: makoto patch by: bweschke
+
+2008-10-17 17:18 +0000 [r150637] Steve Murphy <murf at digium.com>
+
+ * res/res_features.c: Interesting crash. In this case, you exit the
+ bridge with peer completely GONE. I moved the channel find call
+ up to cover the whole peer CDR reset code segment. This appears
+ to solve the crash without changing the logic at all.
+
+2008-10-17 15:31 +0000 [r150557] Jason Parker <jparker at digium.com>
+
+ * main/asterisk.c, main/channel.c, channels/chan_dahdi.c,
+ configure, configure.ac: Correctly allow chan_dahdi to compile
+ against older versions of Zaptel. Don't always define
+ HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor
+ cleanup to make things clear. (closes issue #13726) Reported by:
+ tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46)
+
+2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Reverting changes from commits 150298 and 150301
+ since I was mistakenly under the assumption that dialplan
+ functions *always* required that a channel be present. I need to
+ go home earlier, I think :)
+
+ * main/manager.c: And don't forget to return on the error condition
+
+ * main/manager.c: Don't try to call a dialplan function's read
+ callback from the manager's GetVar handler if an invalid channel
+ has been specified. Several dialplan functions, including CHANNEL
+ and SIP_HEADER, do not check for NULL-ness of the channel being
+ passed in. (closes issue #13715) Reported by: makoto
+
+2008-10-16 15:56 +0000 [r150124] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Fix memory leak found by customer
+
+2008-10-16 15:26 +0000 [r150056] Steve Murphy <murf at digium.com>
+
+ * cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and
+ RYM-150398 and AST-103 in internal Digium bug trackers. These
+ fixes address a really subtle memory corruption problem that
+ would happen in machines heavily loaded in production
+ environments. The corruption would always take the form of the
+ STMT object getting nulled out and one of the unixODBC calls
+ would crash trying to access statement->connection. It isn't
+ fully proven yet, but the server has now been running 2.5 days
+ without appreciable memory growth, or any gain of %cpu, and no
+ crashes. Whether this is the problem or not on that server, these
+ fixes are still warranted. As it turns out, **I** introduced
+ these errors unwittingly, when I corrected another crash earlier.
+ I had formed the build_query routine, and failed to remove
+ mutex_unlock calls in 3 places in the transplanted code. These
+ unlocks would only happen in error situations, but unlocking the
+ mutex early set the code up for a catastrophic failure, it
+ appears. It would happen only once every 100K-200K or more calls,
+ under heavy load... but that is enough. If another crash occurs,
+ with the same MO, I'll come back and remove my confession from
+ the log, and we'll keep searching, but the fact that we have
+ Asterisk dying from an asynchronous wiping of the STMT object,
+ only on some connection error, and that the server has lived for
+ 2.5 days on this code without a crash, sure make it look like
+ this was the problem! Also, in several points, Statement handles
+ are set to NULL after SQLFreeHandle. This was mainly for
+ insurance, to guarantee a crash. As it turns out, the code does
+ not appear to be attempting to use these freed pointers. Asterisk
+ owes a debt of gratitude to Federico Alves and Frediano Ziglio
+ for their untiring efforts in finding this bug, among others.
+
+2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke <bweschke at btwtech.com>
+
+ * CHANGES: Another documentation fix. (closes issue #13708)
+
+ * configs/agents.conf.sample: An update to the
+ documentation/example of agents.conf.sample with the correct
+ parameter for this feature as defined in chan_agent.c (closes
+ issue #13709)
+
+2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: fix some problems when parsing SIP messages
+ that have the maximum number of headers or body lines that we
+ support
+
+2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Change this warning to an error message.
+ Suggestion comes from Sean Bright. Thanks Sean!
+
+ * channels/chan_sip.c: Call register_peer_exten even in the case
+ that the peer's IP/port does not change. (closes issue #13309)
+ Reported by: dimas Patches: v2-13309.patch uploaded by dimas
+ (license 88)
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
+ period for sync-triggered audiohooks so that if packetization of
+ audio is close (but not equal) we don't end up flushing the
[... 20319 lines stripped ...]
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