[svn-commits] russell: tag 1.6.0-beta1 r99125 - /tags/1.6.0-beta1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jan 18 16:37:02 CST 2008
Author: russell
Date: Fri Jan 18 16:37:01 2008
New Revision: 99125
URL: http://svn.digium.com/view/asterisk?view=rev&rev=99125
Log:
Importing files for 1.6.0-beta1 release
Added:
tags/1.6.0-beta1/.lastclean (with props)
tags/1.6.0-beta1/.version (with props)
tags/1.6.0-beta1/ChangeLog (with props)
Change Statistics:
tags/1.6.0-beta1/.lastclean | 1
tags/1.6.0-beta1/.version | 1
tags/1.6.0-beta1/ChangeLog |34728 ++++++++++++++++++++++++++++++++
3 files changed, 34730 insertions(+)
Added: tags/1.6.0-beta1/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/.lastclean?view=auto&rev=99125
==============================================================================
--- tags/1.6.0-beta1/.lastclean (added)
+++ tags/1.6.0-beta1/.lastclean Fri Jan 18 16:37:01 2008
@@ -1,0 +1,1 @@
+33
Propchange: tags/1.6.0-beta1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.6.0-beta1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.6.0-beta1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.6.0-beta1/.version
URL: http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/.version?view=auto&rev=99125
==============================================================================
--- tags/1.6.0-beta1/.version (added)
+++ tags/1.6.0-beta1/.version Fri Jan 18 16:37:01 2008
@@ -1,0 +1,1 @@
+1.6.0-beta1
Propchange: tags/1.6.0-beta1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.6.0-beta1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.6.0-beta1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.6.0-beta1/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/ChangeLog?view=auto&rev=99125
==============================================================================
--- tags/1.6.0-beta1/ChangeLog (added)
+++ tags/1.6.0-beta1/ChangeLog Fri Jan 18 16:37:01 2008
@@ -1,0 +1,34728 @@
+2008-01-18 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.6.0-beta1 released.
+
+2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell at digium.com>
+
+ * CREDITS, include/asterisk/http.h, main/tcptls.c (added),
+ main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
+ main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
+ configs/sip.conf.sample, CHANGES: Merge changes from
+ team/group/sip-tcptls This set of changes introduces TCP and TLS
+ support for chan_sip. There are various new options in
+ configs/sip.conf.sample that are used to enable these features.
+ Also, there is a document, doc/siptls.txt that describes some
+ things in more detail. This code was implemented by Brett Bryant
+ and James Golovich. It was reviewed by Joshua Colp and myself. A
+ number of other people participated in the testing of this code,
+ but since it was done outside of the bug tracker, I do not have
+ their names. If you were one of them, thanks a lot for the help!
+ (closes issue #4903, but with completely different code that what
+ exists there.)
+
+ * main/frame.c, /, include/asterisk/translate.h: Merged revisions
+ 99081 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) |
+ 9 lines Revert adding the packed attribute, as it really doesn't
+ make sense why that would do any good. Fix the real bug, which is
+ to do the check to see if the frame came from a translator at the
+ beginning of ast_frame_free(), instead of at the end. This
+ ensures that it always gets checked, even if none of the parts of
+ the frame are malloc'd, and also ensures that we aren't looking
+ at free'd memory in the case that it is a malloc'd frame. (closes
+ issue #11792, reported by explidous, patched by me) ........
+
+ * /, include/asterisk/translate.h: Merged revisions 99079 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) |
+ 4 lines Since we're relying on the offset between the frame and
+ the beginning of the translator pvt struct, set the packed
+ attribute to make sure we get to the right place. (potential fix
+ for issue #11792) ........
+
+2008-01-18 16:58 +0000 [r99026] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: This should at least temporarily fix a
+ problem where the 't' Dial option is incorrectly passed to the
+ transferee when built-in attended transfers are used. There is
+ still a problem with 'T', but better to fix some problems than no
+ problems while we work on it. (closes issue #7904) Reported by:
+ k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee
+ (license 138) Tested by: sergee, otherwiseguy
+
+2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for
+ speed (closes issue #10723) Reported by: mnicholson Patches:
+ func-odbc-direct-execute1.diff uploaded by mnicholson (license
+ 96) Tested by: Corydon76, mnicholson, falves11
+
+ * res/res_odbc.c: Permit username and password to be NULL (which
+ enables pass-through from the layer above). Reported by: lurcher
+ Patch by: tilghman (Closes issue #11739)
+
+ * funcs/func_cut.c: Reset default CUT delimiter back to '-'
+
+2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant <russell at digium.com>
+
+ * channels/chan_console.c: Make the output of "console list
+ devices" a bit prettier.
+
+ * channels/chan_console.c: List which devices are inputs and
+ outputs in "console list devices"
+
+ * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for
+ ast_best_codec()
+
+ * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h:
+ Merged revisions 99004 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) |
+ 10 lines Have IAX2 optimize the codec translation path just like
+ chan_sip does it. If the caller's codec is in our codec list,
+ move it to the top to avoid transcoding. (closes issue #10500)
+ Reported by: stevedavies Patches: iax-prefer-current-codec.patch
+ uploaded by stevedavies (license 184)
+ iax-prefer-current-codec.1.4.patch uploaded by stevedavies
+ (license 184) Tested by: stevedavies, pj, sheldonh ........
+
+2008-01-17 22:22 +0000 [r99002] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue
+ #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded
+ by DEA (license 3)
+
+2008-01-17 20:51 +0000 [r98998] Jason Parker <jparker at digium.com>
+
+ * Makefile, build_tools/cflags.xml, channels/chan_zap.c,
+ main/dsp.c, configs/zapata.conf.sample: Add several busy
+ detection related defines to menuselect. Allow better busy detect
+ debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and
+ BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches:
+ busydetect_enhancement.patch uploaded by agx (license 298)
+ busydetect-r94975.diff uploaded by sergee (license 138)
+ Additional changes/cleanup by me.
+
+2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: state_interface could be NULL, so use the
+ never-NULL cur->state_interface for this check
+
+ * apps/app_queue.c: Get the device state of the state interface
+ instead of the interface when creating a new queue member. Thanks
+ to Atis Lezdins for bringing this up on the Asterisk-Dev mailing
+ list.
+
+2008-01-17 16:21 +0000 [r98992] Jason Parker <jparker at digium.com>
+
+ * /, configs/zapata.conf.sample: Merged revisions 98991 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes
+ issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600
+ (Thu, 17 Jan 2008) | 4 lines Add a clarification about the
+ immediate= option of zapata.conf Issue 11784, patch by klaus3000.
+ ........
+
+2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c, configs/zapata.conf.sample: major
+ reliability and performance improvement in VWMI monitoring for
+ FXO ports (code by markster, me and dbailey)
+
+ * res/res_config_curl.c: resolve (valid) compiler warning about
+ variable that could be used before being initialized
+
+2008-01-17 03:09 +0000 [r98988] Terry Wilson <twilson at digium.com>
+
+ * res/res_phoneprov.c, doc/tex/phoneprov.tex,
+ configs/phoneprov.conf.sample: Update res_phoneprov to default to
+ setting the SERVER variable to the IP the HTTP request for the
+ config came in on and the SERVER_PORT to the bindport setting in
+ sip.conf. I've left in the ability to override these options,
+ because I can't always guess how someone might decide to do
+ something weird with what is available to them--although needing
+ to is pretty unlikely. Documentation was updated to reflect
+ preference for not setting serveraddr, serveriface, or
+ serverport. Tested on Linux and OS X.
+
+2008-01-17 00:13 +0000 [r98987] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature
+ works, by allowing it to be a column NOT logged into the
+ database. This will allow more granularity of a decision
+ evaluated in the dialplan, then takes effect when posting the
+ CDR.
+
+2008-01-17 00:05 +0000 [r98986] Russell Bryant <russell at digium.com>
+
+ * CHANGES, main/asterisk.c: Add support for an easy way to
+ automatically execute some Asterisk CLI commands immediately at
+ startup. Any commands in the startup_commands file in the
+ Asterisk config diretory will get executed. (closes issue #11781)
+ Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt
+ uploaded by jamesgolovich (license 176) -- With some changes by
+ me.
+
+2008-01-16 23:08 +0000 [r98985] Jason Parker <jparker at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build
+ against <package>_LIB, per recommendations from Russell.
+
+2008-01-16 22:36 +0000 [r98984] Tilghman Lesher <tlesher at digium.com>
+
+ * CHANGES: Info about res_config_curl
+
+2008-01-16 22:36 +0000 [r98983] Russell Bryant <russell at digium.com>
+
+ * /: Blocked revisions 98982 via svnmerge ........ r98982 | russell
+ | 2008-01-16 16:36:24 -0600 (Wed, 16 Jan 2008) | 5 lines Add an
+ unused pointer to the ast_channel struct. This makes the
+ ast_channel structure retain the same size as it had in previous
+ 1.4 releases. Also, all of the offsets for members in the
+ structure are still the same (except for the two pointers that
+ got replaced for the new spy/whisper architecture.) ........
+
+2008-01-16 22:20 +0000 [r98981] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_curl.c (added), main/utils.c: New module
+ res_config_curl (closes issue #11747) Reported by: Corydon76
+ Patches: res_config_curl.c uploaded by Corydon76 (license 14)
+ 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: jmls
+
+2008-01-16 21:53 +0000 [r98978] Russell Bryant <russell at digium.com>
+
+ * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the
+ changes from issue #10665 from the team/group/sip_session_timers
+ branch. This set of changes introduces SIP session timers support
+ (RFC 4028). In short, this prevents stuck SIP sessions that were
+ not properly torn down due to network or endpoint failures during
+ an established SIP session. To quote some of the documentation
+ supplied with the patch: "The SIP Session-Timers is an extension
+ of the SIP protocol that allows end-points and proxies to refresh
+ a session periodically. The sessions are kept alive by sending a
+ RE-INVITE or UPDATE request at a negotiated interval. If a
+ session refresh fails then all the entities that support Session-
+ Timers clear their internal session state. In addition, UAs
+ generate a BYE request in order to clear the state in the proxies
+ and the remote UA (this is done for the benefit of SIP entities
+ in the path that do not support Session-Timers)." (closes issue
+ #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by
+ rjain (license 226) chan_sip.c.diff uploaded by rjain (license
+ 226) sip.conf.sample.diff uploaded by rjain (license 226)
+ proc_422_rsp_comment.diff uploaded by rjain (license 226)
+ chan_sip.c.cache.diff uploaded by rjain (license 226)
+ chan_sip.memalloc uploaded by rjain (license 226)
+ chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches
+ tracked in team/group/sip_session_timers, with some additional
+ fixes by russell and oej. Tested by: jtodd, rjain, loloski
+
+2008-01-16 20:36 +0000 [r98974-98975] Joshua Colp <jcolp at digium.com>
+
+ * /: Blocked revisions 98973 via svnmerge ........ r98973 | file |
+ 2008-01-16 16:34:30 -0400 (Wed, 16 Jan 2008) | 2 lines Bump up
+ cleancount due to previous commit that changed the channel
+ structure. ........
+
+ * /: Blocked revisions 98972 via svnmerge ........ r98972 | file |
+ 2008-01-16 16:33:47 -0400 (Wed, 16 Jan 2008) | 2 lines Replace
+ current spy architecture with backport of audiohooks. This should
+ take care of current known spy issues. ........
+
+2008-01-16 19:41 +0000 [r98968-98971] Jason Parker <jparker at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Partially revert r93898, because it broke the way netsnmp was
+ being detected. rizzo, do you want to discuss so we can rethink
+ this, or do you have another way?
+
+ * CHANGES: Add note about new update.log to CHANGES, by request of
+ jmls and further prodding by jsmith.
+
+ * Makefile, /: Add logging for 'make update' command (also fixes
+ updates in some places). Issue #11766, initial patch by jmls.
+
+2008-01-16 17:51 +0000 [r98967] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6
+ lines Add missing NULLs at end of two ast_load_realtimes. (closes
+ issue #11769) Reported by: tequ Patches: chaniax.patch uploaded
+ by dimas (license 88) ........
+
+2008-01-16 17:21 +0000 [r98965] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 98964 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16
+ Jan 2008) | 10 lines Fix a deadlock in chan_local in
+ local_hangup. There was contention because the local_pvt was held
+ and it was attempting to lock a channel, which is the incorrect
+ locking order. (closes issue #11730) Reported by: UDI-Doug
+ Patches: 11730.patch uploaded by putnopvut (license 60) Tested
+ by: UDI-Doug ........
+
+2008-01-16 16:06 +0000 [r98962] Terry Wilson <twilson at digium.com>
+
+ * res/res_phoneprov.c: Make users list static
+
+2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp <jcolp at digium.com>
+
+ * main/dial.c, /: Merged revisions 98960 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6
+ lines Introduce a lock into the dialing API that protects it when
+ destroying the structure. (closes issue #11687) Reported by:
+ callguy Patches: 11687.diff uploaded by file (license 11)
+ ........
+
+ * /, main/rtp.c: Merged revisions 98958 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4
+ lines Add two more SDP names for ulaw and alaw. (closes issue
+ #11777) Reported by: tootai ........
+
+ * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6
+ lines Don't drop the old record route information when dealing
+ with packets related to a reinvite. (closes issue #11545)
+ Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by
+ kebl0155 (license 356) ........
+
+ * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This
+ seems to come from way back when and I can't think of a reason
+ for it being here, plus it could cause needless DNS lookups.
+ (closes issue #10983) Reported by: jtodd
+
+2008-01-16 01:35 +0000 [r98953] Steve Murphy <murf at digium.com>
+
+ * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found
+ this problem with running the expr2 parser on OSX. Make the
+ #defines come out the same between the parser & lexer.
+
+2008-01-16 01:17 +0000 [r98952] Joshua Colp <jcolp at digium.com>
+
+ * /, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac, makeopts.in: Merged revisions 98951 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan
+ 2008) | 4 lines Add autoconf logic for speexdsp. Later versions
+ use a separate library for some things so we need to use it if
+ present in codec_speex. (closes issue #11693) Reported by: yzg
+ ........
+
+2008-01-15 23:53 +0000 [r98948] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) |
+ 11 lines Change a buffer in check_auth() to be a thread local
+ dynamically allocated buffer, instead of a massive buffer on the
+ stack. This fixes a crash reported by Qwell due to running out of
+ stack space when building with LOW_MEMORY defined. On a very
+ related note, the usage of BUFSIZ in various places in chan_sip
+ is arbitrary and careless. BUFSIZ is a system specific define. On
+ my machine, it is 8192, but by definition (according to google)
+ could be as small as 256. So, this buffer in check_auth was 16
+ kB. We don't even support SIP messages larger than 4 kB! Further
+ usage of this define should be avoided, unless it is used in the
+ proper context. ........
+
+2008-01-15 23:52 +0000 [r98947] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample:
+ Add the "filter" keyword
+
+2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant <russell at digium.com>
+
+ * main/translate.c, include/asterisk/translate.h: Clean up
+ something I did for ABI compatability in 1.4
+
+ * main/frame.c, /, main/translate.c, main/abstract_jb.c,
+ channels/chan_iax2.c, codecs/codec_zap.c,
+ include/asterisk/frame.h, main/rtp.c,
+ include/asterisk/translate.h: Merged revisions 98943 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15
+ Jan 2008) | 25 lines Commit a fix for some memory access errors
+ pointed out by the valgrind2.txt output on issue #11698. The
+ issue here is that it is possible for an instance of a translator
+ to get destroyed while the frame allocated as a part of the
+ translator is still being processed. Specifically, this is
+ possible anywhere between a call to ast_read() and
+ ast_frame_free(), which is _a lot_ of places in the code. The
+ reason this happens is that the channel might get masqueraded
+ during this time. During a masquerade, existing translation paths
+ get destroyed. So, this patch fixes the issue in an API and ABI
+ compatible way. (This one is for you, paravoid!) It changes an
+ int in ast_frame to be used as flag bits. The 1 bit is still used
+ to indicate that the frame contains timing information. Also, a
+ second flag has been added to indicate that the frame came from a
+ translator. When a frame with this flag gets released and has
+ this flag, a function is called in translate.c to let it know
+ that this frame is doing being processed. At this point, the flag
+ gets cleared. Also, if the translator was requested to be
+ destroyed while its internal frame still had this flag set, its
+ destruction has been deffered until it finds out that the frame
+ is no longer being processed. Admittedly, this feels like a hack.
+ But, it does fix the issue, and I was not able to think of a
+ better solution ... ........
+
+2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4
+ lines Based on the boundary found move over the correct amount.
+ (closes issue #11750) Reported by: tasker ........
+
+ * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4
+ lines Accept "; boundary=" not just ";boundary=" in the multipart
+ mixed content type. (closes issue #11750) Reported by: tasker
+ ........
+
+2008-01-14 22:19 +0000 [r98889] Jason Parker <jparker at digium.com>
+
+ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add
+ backupdeleted option to app_voicemail (closes issue #10740)
+ Reported by: ruffle Patches: app_voicemail.diff uploaded by
+ ruffle (license 201) 10740-voicemail.diff uploaded by qwell
+ (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak
+ (license 7) Tested by: blitzrage, mvanbaak, qwell
+
+2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_directory.c: Big improvement for app_directory. This
+ patch breaks the do_directory function up so that it is more
+ easily parsed by the human brain. It also fixes some errors. I'll
+ quote dimas from the original bug description: "app_directory
+ contained some duplicate code even before addition of 'm' option.
+ Addition of that option doubled amount of that code. Worst of
+ all, there are minor differences between these code block and
+ bugs caused by these differences. 1. There is a memory leak. In
+ the 'menu' mode, result of the convert(pos) function is not freed
+ while it should be. 2. In the 'menu' mode check for
+ OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result,
+ application works in the mode opposite to what user expect
+ (checking last name when user wants the first nd vice versa). 3.
+ select_item function plays message for user using res = func1()
+ || func2() || func3()... construct. This construct loses the
+ actual value returned by ast_waitstream() for example so at the
+ end, res does not contain digit user dialed while listening to
+ the message. 4. (also in 1.4) application announces entries from
+ voicemail.conf/realtime separately from entries from users.conf.
+ I see no reason why doing so instead of building combined list.
+ 5. Alot of duplicated code as already mentioned." This was tested
+ by dimas and I (I tested under valgrind). A word of caution: any
+ bug fixes that happen in app_directory in 1.4 will almost
+ certainly not merge cleanly into trunk as a result of this, but
+ it is well worth it. Huge thanks to dimas for this wonderful
+ submission. (closes issue #11744) Reported by: dimas Patches:
+ dir3.patch uploaded by dimas (license 88) Tested by: putnopvut,
+ dimas
+
+ * /: Blocked revisions 98849 via svnmerge ........ r98849 |
+ mmichelson | 2008-01-14 14:59:26 -0600 (Mon, 14 Jan 2008) | 4
+ lines Adding in appropriate unlocks for the locks I added. Thanks
+ to joetester on IRC for pointing this out. ........
+
+2008-01-14 20:01 +0000 [r98830] Joshua Colp <jcolp at digium.com>
+
+ * main/manager.c: Make sure the user's manager secret exists, even
+ if it is blank. (closes issue #11749) Reported by: srt
+
+2008-01-14 18:42 +0000 [r98811] Terry Wilson <twilson at digium.com>
+
+ * CHANGES: Add description of TOUPPER and TOLOWER dialplan
+ functions to CHANGES.
+
+2008-01-14 17:40 +0000 [r98776] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Add proper call forwarding (all and busy)
+ support for chan_skinny. Note: NoAnswer support is currently not
+ implemented, as it would take a significant amount of work to
+ figure out how to do correctly. Closes issue #11310, patches,
+ testing, and support by DEA, mvanbaak, and myself.
+
+2008-01-14 17:39 +0000 [r98775] Russell Bryant <russell at digium.com>
+
+ * /, main/translate.c: Merged revisions 98774 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) |
+ 3 lines Revert a change that introduces an unacceptable
+ performance hit and is causing memory leaks ... (from rev 97973)
+ ........
+
+2008-01-14 17:18 +0000 [r98773] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Fix for potential crash with vmexten
+
+2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Merged revisions 98737 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan
+ 2008) | 3 lines Fixing another compilation error. I'm a bit off
+ today :( ........
+
+ * /: Blocked revisions 98734 via svnmerge ........ r98734 |
+ mmichelson | 2008-01-14 10:30:33 -0600 (Mon, 14 Jan 2008) | 3
+ lines Oops. Last commit had compilation error. ........
+
+ * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan
+ 2008) | 8 lines Adding explicit defaults for missing options to
+ init_queue. This is necessary because if a user either removes or
+ comments one of these options and reloads their queues, the
+ option will not reset to its default, instead maintaining the
+ value from prior to the reload. Thanks to John Bigelow for
+ pointing this error out to me. ........
+
+2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c: Print out a warning when spaces are used in the
+ variable name in Set and MSet. It is extremely hard to debug this
+ issue so this should make it easier. (closes issue #11759)
+ Reported by: caio1982 Patches: setvar_space_warning1.diff
+ uploaded by caio1982 (license 22)
+
+ * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update
+ documentation. (closes issue #11763) Reported by: IgorG Patches:
+ docupd.v1.diff uploaded by IgorG (license 20)
+
+2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant <russell at digium.com>
+
+ * apps/app_jack.c: Add another small option for the JACK app and
+ JACK_HOOK function. The 'n' option tells JACK not to start jackd
+ automatically if it is not already running. Otherwise, the
+ default is that jackd will get started for you if it isn't
+ running already.
+
+ * CHANGES: - Break up the Misc. section a bit with a new section
+ for Misc. New Modules - Change spacing a bit in some places for
+ consistent indentation
+
+ * CHANGES, apps/app_jack.c (added): Bring in the code from
+ team/russell/jack/. Add a new module, app_jack, which provides
+ interfaces to JACK, the Jack Audio Connection Kit
+ (http://www.jackaudio.org/). Two interfaces are provided; there
+ is a JACK() application, and a JACK_HOOK() function. Both
+ interfaces create an input and output JACK port. The application
+ makes these ports the endpoint of the call. The audio coming from
+ the channel goes out the output port and whatever comes back in
+ on the input port is what gets sent to the channel. The
+ JACK_HOOK() function turns on a JACK audiohook on the channel.
+ This lets you run the audio coming from a channel through JACK,
+ and whatever comes back in is what gets forwarded on as the
+ channel's audio. This is very useful for building custom vocoders
+ or doing recording or analysis of the channel's audio in another
+ application. In case anyone is curious, the platform that
+ inspired me to write this is PureData (http://puredata.info/). I
+ wrote these JACK interfaces so that I could use Pd to do
+ interesting things with the audio of phone calls ...
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
+ configure script check for JACK.
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
+ Remove KDE configure script check that isn't used
+
+ * main/audiohook.c: Remove a duplicate lock of the audiohook lock
+ when destroying manipulate audiohooks. This causes an error when
+ we attempt to destroy the lock later when freeing the audiohook.
+
+ * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar",
+ which allows you to set a channel variable (or function) on an
+ active channel from the CLI.
+
+2008-01-12 18:12 +0000 [r98536] Tilghman Lesher <tlesher at digium.com>
+
+ * main/manager.c: Conversion to load manager.conf into memory did
+ not convert the password functions correctly. (Closes issue
+ #11749)
+
+2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni <paripurnachand at digium.com>
+
+ * /, main/http.c: merging a comment added in 1.4
+
+2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and
+ ensure that the AMI DNDState event always gets generated. (closes
+ issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded
+ by tzafrir (modified by me) (license 46)
+
+2008-01-12 00:17 +0000 [r98487] Tilghman Lesher <tlesher at digium.com>
+
+ * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008)
+ | 4 lines Add a connection timeout attribute, as that was what
+ was intended with the login timeout, but ODBC divides it up into
+ 2 different timeouts. (Closes issue #11745) ........
+
+2008-01-11 23:57 +0000 [r98454] Russell Bryant <russell at digium.com>
+
+ * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some
+ extra checking to help out with a potential error when trying to
+ run "make asterisk.pdf" when not all of the right packages are
+ installed. (closes issue #10763) Reported by: Corydon76 Patches:
+ 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add
+ 'auto' signalling mode for Zaptel channels. (closes issue #11690)
+ Reported by: tzafrir Patches: signaling_to_signalling.diff
+ uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded
+ by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir
+ (license 46) zap_no_default_sig.diff uploaded by tzafrir (license
+ 46) zap_signal_auto.diff uploaded by tzafrir (license 46)
+
+2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp <jcolp at digium.com>
+
+ * main/event.c: Goodbye again drumkilla.
+
+ * main/event.c: drumkilla ftw.
+
+ * main/audiohook.c: I am no longer Rockin'
+
+ * main/audiohook.c: Testing something...
+
+2008-01-11 22:52 +0000 [r98400] Russell Bryant <russell at digium.com>
+
+ * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) |
+ 9 lines Fix up setting the EID on BSD based systems. (closes
+ issue #11646) Reported by: caio1982 Patches:
+ dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
+ dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
+ by: caio1982, mvanbaak ........
+
+2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp <jcolp at digium.com>
+
+ * /, main/rtp.c: Merged revisions 98325 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6
+ lines If the incoming RTP stream changes codec force the bridge
+ to break if the other side does not support it. (closes issue
+ #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch
+ uploaded by tsearle (license 373) ........
+
+ * /, res/res_agi.c: Merged revisions 98317 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6
+ lines If the channel is hungup during RECORD FILE send a result
+ code of -1 to be uniform with everything else. (closes issue
+ #11743) Reported by: davevg Patches: res_agi.diff uploaded by
+ davevg (license 209) ........
+
+2008-01-11 19:12 +0000 [r98316] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c, /: Merged revisions 98315 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan
+ 2008) | 5 lines Properly report the hangup cause as no answer
+ when someone does not answer (closes issue #10574, reported by
+ boch, patched by moy) ........
+
+2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant <russell at digium.com>
+
+ * codecs/codec_resample.c: Kevin noted that the thing that I
+ _actually_ changed here was that I converted a value from a
+ double, to a float, back to a double. Sure enough, when I changed
+ my interim variable back to a double, it still blows up.
+ Switching all of these to a float fixes the problem. This seems
+ like a compiler bug where a double passed as an argument isn't
+ getting properly aligned, so I'll have to see if I can replicate
+ it with a small test program. (related to issue #11725)
+
+ * codecs/codec_resample.c: Fix a bus error that happened when
+ asterisk was built with optimizations on with platforms that
+ explode on unaligned access. I'm not exactly sure why this fixes
+ it, but it fixed it on the machine I was testing on. If it makes
+ sense to you, feel free to enlighten me. :) (closes issue #11725,
+ patched by me)
+
+2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to
+ adaptive_odbc, too
+
+ * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix
+
+2008-01-11 18:26 +0000 [r98267] Russell Bryant <russell at digium.com>
+
+ * /: Blocked revisions 98265 via svnmerge ........ r98265 | russell
+ | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines
+ Backport the ability to set the ToS bits on Linux when not
+ running as root. Normally, we would not backport features into
+ 1.4, but, I was convinced by the justification supplied by the
+ supplier of this patch. He pointed out that this patch removes a
+ requirement for running as root, thus reducing the potential
+ impacts of security issues. (closes issue #11742) Reported by:
+ paravoid Patches: libcap.diff uploaded by paravoid (license 200)
+ ........
+
+2008-01-11 17:27 +0000 [r98220] Joshua Colp <jcolp at digium.com>
+
+ * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4
+ lines Ensure the return value of ast_bridge_call is passed back
+ up as the application return value. This is needed for transfers
+ to function so the PBX core knows to continue execution. (closes
+ issue #10327) Reported by: kkiely ........
+
+2008-01-11 17:17 +0000 [r98218] Russell Bryant <russell at digium.com>
+
+ * codecs/codec_g722.c: At one point during working on this module,
+ I had the lin/lin16 versions of the framein callbacks different.
+ However, they are now the same again, so remove the duplicate
+ code and use the same functions for the lin/lin16 versions.
+
+2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008)
+ | 2 lines Back out changes from revision 97077, since it wasn't
+ perfect ........
+
+ * doc/manager_1_1.txt: Documentation updates
+
+2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO
+ using '!' as the 'digit'
+
+2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant <russell at digium.com>
+
+ * /: Blocked revisions 98082 via svnmerge ........ r98082 | russell
+ | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines Fix
+ samples vs. length calculations for g722 ........
+
+ * codecs/codec_g722.c, main/frame.c: - Fix the last set of places
+ where incorrect assumptions were made about the sample length
+ with g722. It is _2_ samples per byte, not 1. This was all over
+ the place, and I believed it, and it is what caused me to take so
+ long to figure out what was broken. - Update copyright
+ information on codec_g722.
+
+2008-01-11 00:54 +0000 [r98047] Mark Michelson <mmichelson at digium.com>
+
+ * main/translate.c: Fix "core show translation" to not output
+ information for "unknown" codecs. This fix was made in favor of
+ the proposed patch since it doesn't involve changing a core codec
+ define. (closes issue #11722, reported and initially patched by
+ caio1982, final patch by me)
+
+2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new
+ global and per-peer option to chan_sip, qualifyfreq, which allows
+ you to set the qualify frequency. (closes issue #11597) Reported
+ by: wilder Patches: qualifyfreq5.patch uploaded by wilder
+ (license 362) -- with some mods by me
+
+ * /: Blocked revisions 98025 via svnmerge ........ r98025 | russell
+ | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines Simplify
+ this code with a suggestion from Luigi on the asterisk-dev list.
+ Instead of using is16kHz(), implement a format_rate() function.
+ ........
+
+ * main/translate.c: Simplify this code with a suggestion from Luigi
+ on the asterisk-dev list. Instead of using is16kHz(), implement a
+ format_rate() function.
+
+2008-01-10 23:40 +0000 [r97978] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008)
+ | 6 lines 1) When we get a translated frame out, clone it,
+ because if the translator pvt is freed before we use the frame,
+ bad things happen. 2) Getting a failure from ast_sched_delete
[... 33982 lines stripped ...]
More information about the svn-commits
mailing list