[svn-commits] russell: tag 1.4.3 r61796 - in /tags/1.4.3:
.lastclean .version ChangeLog
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Apr 24 16:04:08 MST 2007
Author: russell
Date: Tue Apr 24 18:04:08 2007
New Revision: 61796
URL: http://svn.digium.com/view/asterisk?view=rev&rev=61796
Log:
importing files for 1.4.3 release
Added:
tags/1.4.3/.lastclean (with props)
tags/1.4.3/.version (with props)
tags/1.4.3/ChangeLog (with props)
Added: tags/1.4.3/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.3/.lastclean?view=auto&rev=61796
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--- tags/1.4.3/ChangeLog (added)
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@@ -1,0 +1,6587 @@
+2007-04-24 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.3 released.
+
+2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell at digium.com>
+
+ * main/manager.c, /: Merged revisions 61786 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
+ 4 lines Don't crash if a manager connection provides a username
+ that exists in manager.conf but does not have a password, and
+ also requests MD5 authentication. (ASA-2007-012) ........
+
+ * main/channel.c, include/asterisk/channel.h: Improve DTMF handling
+ in ast_read() even more in response to a discussion on the
+ asterisk-dev mailing list. I changed the enforced minimum length
+ of a digit from 100ms to 80ms. Furthermore, I made it now enforce
+ a gap of 45ms in between digits. These values are not
+ configurable in a configuration file right now, but they can be
+ easily changed near the top of main/channel.c.
+
+2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
+ | 1 line removed #if 0 block from chan_phone, chan_zap, and
+ chan_modem restart_monitor() ........
+
+2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell at digium.com>
+
+ * main/dial.c: Add a few more state changes in
+ handle_frame_ownerless() so that the SLA code will get notified
+ of these changes even when an owner channel is not provided. This
+ isn't from a specific bug report, it's just something I noticed
+ while poking around.
+
+2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
+ lines Allow RFC2833 to be sent in the response SDP when an INVITE
+ comes in without SDP. (issue #9546 reported by mcrawford)
+ ........
+
+2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Some dialplan functions, such as CUT(), expect to
+ operate on variables on a channel. So, this little hack lets them
+ work in places where a channel doesn't exist, such as within
+ DUNDi configuration. (issue #9465, reported and patched by
+ Corydon76, testing by blitzrage)
+
+ * main/channel.c: Ensure that digits passing through Asterisk have
+ a reasonable minimum length. It is currently 100 ms. If someone
+ thinks this should be different, feel free to speak up. (related
+ to issues #8944, #9250, and #9348)
+
+2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker at digium.com>
+
+ * main/rtp.c: Avoid invalid seqno cycling detection. Per comment
+ from Dave Troy: This adds back in some simple typecasting I had
+ in an earlier version which I realize now may be breaking things.
+ Issue #9554.
+
+ * main/loader.c, /: Merged revisions 61704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
+ lines Fix an issue that I noticed while looking over issue 9571.
+ The reload timestamp was getting set after reloading the built-in
+ stuff, and before the modules. ........
+
+2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c: Remove a stray debug message introduced by a recent
+ commit.
+
+2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
+ lines If the '* to hangup' option is not enabled, we don't need
+ to disable * as a valid exit key. If it was enabled, this
+ statement would've never been checked in the first place. Issue
+ #9552 ........
+
+2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell at digium.com>
+
+ * main/config.c, apps/app_voicemail.c, main/manager.c,
+ include/asterisk/config.h: Fix the UpdateConfig manager action to
+ properly treat "variables" and "objects" differently (a=b versus
+ a=>b). (issue #9568, reported by pari, patch by me)
+
+2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
+ lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
+ Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........
+
+2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * main/manager.c: Bug 9557 - simple reason why reading a function
+ always returned NULL
+
+ * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
+ funcs/func_groupcount.c, /, funcs/func_timeout.c,
+ funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
+ | 5 lines Bug 9557 - Specifying the GetVar AMI action without a
+ Channel parameter can cause Asterisk to crash. The reason this
+ needs to be fixed in the functions instead of in AMI is because
+ Channel can legitimately be NULL, such as when retrieving global
+ variables. ........
+
+2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming at digium.com>
+
+ * sounds/Makefile: allow external build systems to extract the
+ required sound file versions
+
+2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej at edvina.net>
+
+ * main/rtp.c: Clean upp formatting, add some doxygen stuff while
+ we're in cleaning mode... Thanks Kevin!
+
+ * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)
+
+2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: #9483, half of patch by twilson to solve 302
+ redirect issues
+
+ * /: Blocking AstHoloPatch from 1.2
+
+2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf at digium.com>
+
+ * main/cdr.c: This is a fix to the way CDR merge handles the data
+ that results from ForkCDR.
+
+2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
+ lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
+ the same as OUTBOUND_GROUP except it will get unset after use so
+ it won't get accidentally inherited. (issue #BE-140) ........
+
+ * apps/app_speech_utils.c: Do not bother looking for a result if
+ none are present.
+
+ * channels/chan_sip.c: For those very verbose SIP implementations
+ that attach tons of info to the Contact header... let's increase
+ our variable sizes. (issue #9535 reported by jeffg)
+
+2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell at digium.com>
+
+ * apps/app_voicemail.c: Eliminate a compiler warning with
+ ODBC_STORAGE enabled so that it will build under dev-mode.
+
+2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf at digium.com>
+
+ * channels/chan_oss.c: A fix for chan_oss that resulted from the
+ CDR changes; it helps to use the right info.
+
+2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Don't assume the callid of a dialog will be
+ set, as in some circumstances it may not. (issue #9534 reported
+ by tecnoxarxa)
+
+2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
+ 5 lines If someone sets the "useragent" option in sip.conf to be
+ empty, then don't add the User-Agent header at all. It is an
+ optional header, anyway. Also, the bug report says that some of
+ Japan's SIP providers don't allow it for some weird reason.
+ (issue #9488, reported by makoto, fixed by me) ........
+
+2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns at beronet.com>
+
+ * channels/chan_misdn.c: Don't export AOCD variables on
+ misdn_hangup anymore, this was mainly a fix for trunk..
+
+2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
+ 6 lines Fix a bug with switching between host=dynamic and using
+ specific hosts for peers. The code would only reset the peer's
+ address when it is dynamic if it was a new peer structure. Now,
+ it will also reset the address if it was already in the peer
+ list, but before the reload, it was not dynamic. (issue #9515,
+ reported by caio1982, fixed by me) ........
+
+ * main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
+ bkruse) In passing, constify the elements of the mimetypes table.
+
+ * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
+ 5 lines Remove the attempt at reporting configuration errors in
+ sip.conf. This can cause a bunch of improper messages when using
+ realtime. I give up. As oej tried to convince me when I put this
+ in, there is just no easy way to do it. (inspired by a message on
+ the -dev list) ........
+
+2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns at beronet.com>
+
+ * channels/chan_misdn.c: Export AOCD variables on misdn_hangup.
+
+ * channels/chan_misdn.c: Ignore facility messages in case we don't
+ have a corresponding channel object.
+
+ * channels/chan_misdn.c: AOCD's are now exported to asterisk
+ channel variables.
+
+2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell at digium.com>
+
+ * main/Makefile, main/http.c, main/minimime (removed): File upload
+ support was added to solve some needs for the Asterisk GUI.
+ However, after much discussion, it has been decided that adding
+ this to 1.4 is not in the best interests of the project. It has
+ been removed here, but will remain in trunk.
+
+2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns at beronet.com>
+
+ * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
+ 2007) | 2 lines msns config parameter defaults to '*' ........
+
+2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf at digium.com>
+
+ * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
+ previous fix to overcome a compiler warning; the app NoCDR() has
+ been updated to mark the channel CDR as POST_DISABLED instead of
+ destroying the CDR; this way its flags are propagated thru a
+ bridge and the CDR is actually dropped. The cases where only one
+ channel in a bridge has a CDR was cleaned up.
+
+2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
+ lines - Don't send ActionID before Response: header. - Don't use
+ a blank in an AMI header ........
+
+2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/minimime/mm_envelope.c, res/res_features.c: fix up some
+ warnings found using --enable-dev-mode
+
+ * main/minimime/Doxyfile (removed),
+ main/minimime/tests/messages/CVS (removed),
+ main/minimime/tests/CVS (removed): remove some more stuff we
+ don't need
+
+2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell at digium.com>
+
+ * main/minimime/test (removed): Remove another directory that
+ should no longer be there
+
+ * main/minimime/Make.conf (removed), main/minimime/mytest_files
+ (removed), main/minimime/.cvsignore (removed), main/minimime/sys
+ (removed), main/minimime/mm-docs (removed): Remove various files
+ that I thought I already removed.
+
+2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker at digium.com>
+
+ * apps/app_queue.c: Use the appropriate interface name with
+ COMPLETECALLER. Issue 9395.
+
+2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf at digium.com>
+
+ * channels/chan_oss.c, main/channel.c, main/cdr.c,
+ channels/chan_phone.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, channels/chan_features.c,
+ channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
+ channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
+ channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
+ channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
+ include/asterisk/channel.h, channels/chan_gtalk.c,
+ channels/chan_iax2.c: This is a big improvement over the current
+ CDR fixes. It may still need refinement, but this won't have as
+ many folks bothered.
+
+2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej at edvina.net>
+
+ * res/res_jabber.c: Add final new line after JabberEvent
+
+2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker at digium.com>
+
+ * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
+ lines Allow matching on names shorter than 3 chars. This also
+ fixes the case where somebody wants to match on less then 3
+ chars. Issue 9071 ........
+
+2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
+ | 2 lines Don't check for error when lowering priority (according
+ to the manpage, it should never happen anyway). It might could
+ happen, though, if another thread messed with the priority, so
+ safeguard against that (reported via -dev list). ........
+
+ * channels/chan_local.c, /: Merged revisions 60846 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
+ Apr 2007) | 2 lines Bug 9505 - If the return value for
+ local_queue_frame is set, then p->lock is no longer valid.
+ ........
+
+2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
+ lines When calling a device that then forwards us elsewhere... we
+ have to make our channels compatible if it is the only channel
+ being dialed. (issue #9445 reported by marcelbarbulescu) ........
+
+ * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
+ MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)
+
+2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
+ | 2 lines Gosub called within a Macro resets the arguments
+ improperly and causes general weirdness. (Issue 8329) ........
+
+ * main/http.c: Fix --enable-dev-mode
+
+ * channels/chan_oss.c: Off by one error, resulting in a crash
+ (Issue 9500)
+
+ * /, main/file.c: Merged revisions 60660 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
+ | 2 lines Bug 9486 - memory leak when opening a filestream
+ ........
+
+2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell at digium.com>
+
+ * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
+ main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
+ main/manager.c, main/minimime/mm_mimepart.c,
+ main/minimime/test.sh, configure, include/asterisk/compat.h,
+ main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
+ main/minimime/tests/parse.c, main/minimime/mm_base64.c,
+ main/minimime/mm_mimeutil.c, main/minimime/mm.h,
+ main/minimime/tests, main/minimime/mm_header.c,
+ main/minimime/mm_error.c, main/Makefile,
+ main/minimime/mm_codecs.c, main/minimime/mm_param.c,
+ configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
+ include/asterisk/manager.h, main/minimime/strlcpy.c,
+ configs/http.conf.sample, main/minimime/mm_parse.c,
+ main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
+ main/minimime/mm_util.c, main/minimime/mm_envelope.c,
+ main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
+ main/minimime/tests/messages/test2.txt,
+ main/minimime/tests/messages/test3.txt,
+ main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
+ main/minimime/tests/messages/test4.txt,
+ main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
+ main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
+ main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
+ main/minimime/mimeparser.l, main/minimime/mm_context.c,
+ main/minimime/mimeparser.tab.h, main/minimime (added),
+ main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
+ main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
+ main/minimime/mimeparser.y, Makefile.moddir_rules,
+ main/minimime/sys, main/minimime/tests/Makefile: To be able to
+ achieve the things that we would like to achieve with the
+ Asterisk GUI project, we need a fully functional HTTP interface
+ with access to the Asterisk manager interface. One of the things
+ that was intended to be a part of this system, but was never
+ actually implemented, was the ability for the GUI to be able to
+ upload files to Asterisk. So, this commit adds this in the most
+ minimally invasive way that we could come up with. A lot of work
+ on minimime was done by Steve Murphy. He fixed a lot of bugs in
+ the parser, and updated it to be thread-safe. The ability to
+ check permissions of active manager sessions was added by Dwayne
+ Hubbard. Then, hacking this all together and do doing the
+ modifications necessary to the HTTP interface was done by me.
+
+2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * UPGRADE.txt: clarified a sentence in the format_wav section
+
+ * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
+ plan to remove GAIN code from trunk
+
+2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c: When a station picks up a trunk that was on
+ hold, make the hints reflect that nobody has the trunk on hold
+ anymore.
+
+ * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
+ reported by francesco_r, fixed by me) * The original behavior was
+ that if one station put a call on hold, another one picked it up,
+ and then hung up, the code would still consider the call on hold
+ by the first station, so the trunk would not be hung up. However,
+ to better comply with what most people seem to expect it to
+ behave, it will now hang up the trunk. * Fix a problem with
+ "barge=no". This was only intended to prevent people from joining
+ calls that are in progress. However, it also prevented other
+ people from picking up a call that was on hold. This has been
+ fixed. * When there are no active stations on a trunk and it is
+ on hold, the code now indicates the HOLD and UNHOLD conditions to
+ the trunk channel. This allows music on hold to be played to the
+ trunk when it is on hold.
+
+2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin at digium.com>
+
+ * channels/chan_zap.c: Make sure we check the faxdetect option
+ before doing fax processing
+
+ * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
+ lines There should only be one code path for doing DTMF
+ conditionals on channels. This fixes it. ........
+
+2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
+ | 2 lines remove undocumented 'cardsmode' parameter and stop
+ searching for transcoders during reload() ........
+
+2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp at digium.com>
+
+ * res/res_speech.c, apps/app_speech_utils.c,
+ include/asterisk/speech.h: Add support for returning different
+ types of results (ie: NBest).
+
+2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * formats/format_wav.c: modified default GAIN for issue 5823,
+ thanks jrwalliker
+
+2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf at digium.com>
+
+ * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
+ some clarification to the example configs for CDRs, on how to
+ select a backend. Also, made cdr-csv the default if you 'make
+ samples', and no other changes.
+
+2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
+ lines Just because we can't find the voicemail configuration
+ file, doesn't mean that the module failed to load. The user could
+ be using realtime. Issue #9473 ........
+
+2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell at digium.com>
+
+ * main/http.c: Add the MIME type for gif by request from Pari
+
+2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
+ lines Only unlock our pvt and net locks if we are actually going
+ to try to lock the owner again. (issue #9472 reported by zoa)
+ ........
+
+2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell at digium.com>
+
+ * main/manager.c, /: Merged revisions 60134 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
+ 6 lines It is valid to redirect channels via the manager
+ interface that are not in the UP state. Instead of checking for
+ that to prevent to ensure a dead channel doesn't get redirected,
+ just use the ast_check_hangup() API call. (issue #9457, reported
+ by Callmewind, patch by me) (related to issue #8977) ........
+
+ * channels/chan_sip.c: Add a Content-Length of 0 to the response
+ built by transmit_response_with_unsupported(). (issue #9454,
+ reported by makoto, fixed by me)
+
+ * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
+ 4 lines Fix the return value of handle_common_options() so that
+ it always properly indicates whether it handled the option or
+ not. (issue #9455, reported by Netview, fixed by me) ........
+
+ * apps/app_meetme.c: Fix a problem where if a trunk was hung up
+ while it was on hold, all of the hints would reflect the line
+ still on hold, even though it should reflect that it is back to
+ not in use. (issue #9459, reported by francesco_r, fixed by me)
+
+ * /: Blocked revisions 60016 via svnmerge ........ r60016 | russell
+ | 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a
+ missing "\r\n" in the body of the NOTIFY that is sent to indicate
+ the status of a transfer. (issue #9388, reported by rarritt)
+ ........
+
+ * /: Blocked revisions 60014 via svnmerge ........ r60014 | russell
+ | 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the
+ more generic check for "sed -r" support that was already present
+ in 1.4. (related to issue #9399) ........
+
+ * /: Blocked revisions 60012 via svnmerge ........ r60012 | russell
+ | 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On
+ Darwin, the -r argument to sed is not valid. It has to be -E.
+ (issue #9399, reported by jcovert) ........
+
+2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_speech_utils.c: Don't clash when a person both speaks
+ and uses DTMF.
+
+2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
+ 4 lines Don't attempt to report configuration errors in
+ build_user(). oej pointed out that for a "friend" entry, this
+ won't work, because all user options are valid for peers, but not
+ the other way around. ........
+
+ * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
+ 3 lines Make chan_sip report when it encounters an unknown
+ option. (issue #9440, reported by nightcrawler) ........
+
+ * /, main/app.c: Merged revisions 59886 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
+ 5 lines When doing a built-in blind or attended transfer, restore
+ the ability to use '#' to terminate the number and immediately do
+ the transfer instead of having to dial the number and just wait
+ for the feature digit timeout. (issue #8366, xueliangliang)
+ ........
+
+ * Makefile: Ensure that menuselect gets executed in dependency
+ check mode every time you run make.
+
+2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns at beronet.com>
+
+ * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
+ Merged revisions 59788,59803 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
+ lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
+ or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
+ 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........
+
+2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter at beronet.com>
+
+ * channels/misdn/isdn_lib.c, channels/misdn_config.c,
+ channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
+ Merged revisions 59623-59624,59639 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
+ 1 line we can now make 30 channels on a PRI (before we forgot
+ chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
+ (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
+ r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
+ 1 line added option which allows us to accept incoming SETUP
+ Messages without automatically sending Proceeding or Setup
+ Acknowledge, this is useful with some broken switches and if you
+ want to Release incoming calls without previously having
+ acknowledged them. The new option is
+ noautorespond_on_setup=yes|no default is no, so we don't break
+ the existing behaviour ........
+
+2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
+ lines Increase the maximum size for a string of mailboxes to
+ 1024. (issue #9270 reported by rtucker) ........
+
+2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
+ not the test. As per 9435, thanks to marcelbarbulescu
+
+2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell at digium.com>
+
+ * main/netsock.c, /: Merged revisions 59608 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
+ 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
+ This is needed by the patch that went in for issue 7874.
+ chan_iax2 needs to be able to create socket that is lisetning on
+ INADDR_ANY, but also be able to bind sockets to specific
+ addresses. (Thanks to Stevenson on the asterisk-dev mailing list
+ for explaining why this flag was needed.) ........
+
+2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker at digium.com>
+
+ * configure, main/Makefile, acinclude.m4: Add linux-uclibc host
+ arch..."thingy". Sorry, I don't know what it's called...
+
+2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf at digium.com>
+
+ * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
+ include/asterisk/cdr.h: several changes via kpflemings review
+
+ * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
+ include/asterisk/cdr.h: These mods fix CDR issues from 8221,
+ 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
+ from transfer situations.
+
+ * configs/extensions.conf.sample: A small clarification to keep
+ bugs from being filed, and confusion from rising, if
+ clearglobalvars is set, and globals are set in the AEL file.
+ (9419)
+
+2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell at digium.com>
+
+ * res/res_jabber.c: When building a response to a subscription, the
+ "from" must be the full Jabber ID. This fixes some problems where
+ jabber users are not able to add their Asterisk account to their
+ user list, since they are unable to get Asterisk to approve their
+ subscription. (issue #8210, reported by caspy, and verified by
+ bradtem)
+
+2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
+ lines Keep a global array of variables indicating whether certain
+ conference rooms are in use. This ensures that two people going
+ into a new dynamic conference when the 'e' option is set don't go
+ into the same conference room. (issue #8835 reported by eliel)
+ ........
+
+2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c, /: Merged revisions 59357 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
+ 5 lines If an error occurs when reading from an RTP socket, and
+ the error code does not indicate that we should try again, then
+ return NULL instead of a "null frame". This will prevent Asterisk
+ from trying over and over again, and eventually causing the
+ system to crash. (issue #8285, john) ........
+
+ * /: Blocked revisions 59355 via svnmerge ........ r59355 | russell
+ | 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport
+ the change to chan_iax2 to return NULL instead of a "null frame"
+ from its read callback. See revision 59341 to the 1.4 branch for
+ more info. ........
+
+ * channels/chan_iax2.c: When the IAX2 read callback gets called,
+ return NULL instead of a "null frame". This will cause Asterisk
+ to hangup the call instead of keep trying whatever it was doing.
+ Under normal conditions, this function would *never* be called.
+ However, the author of this patch says an error will occur that
+ will cause it to get called every 100 thousand calls or so. When
+ this does happen, it puts the channel in a loop that eventually
+ brings down the system. So, hangup up the call is certainly a
+ better alternative. (issue #8286, john)
+
+ * Makefile: Export the GTK2 library and include information to sub
+ Makefiles.
+
+2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
+ | 3 lines Issue 9415 - No point to getting a diagnostic field if
+ we aren't doing anything with the information. (Plus, it tends to
+ crash the Postgres ODBC driver.) ........
+
+ * /: Blocked revisions 59299 via svnmerge ........ r59299 |
+ tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines
+ Change ENV section to use setenv, instead of putenv (Alexandru
+ Pirvulescu <sigxcpu at gmail.com>, reported via -dev list) ........
+
+2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_odbc.c: Another crash that I thought we had fixed already
+ - Issue 9396
+
+ * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
+ | 2 lines Oops ........
+
+ * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
+ | 2 lines Fix a few remaining bad mmap(2) return values ........
+
+2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
+ 3 lines Fix the check of the return value from mmap(). Thanks to
+ Corydon for catching this one. ........
+
+ * apps/app_directory.c: Fix app_directory to actually compile with
+ ODBC_STORAGE, and update the code to the latest res_odbc API.
+
+ * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
+ The Makefile did not properly ensure that this information got
+ copied from what was selected for app_voicemail. (issue #9224)
+
+ * channels/chan_sip.c: Fix the check that ensures that the CHANNEL
+ function's first argument is "rtpqos". Thanks, Corydon. :)
+
+2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
+ asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
+ this case the proper thing to do. This change now has it doing
+ the proper thing.
+
+2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
+ 4 lines Fix the use of the "sourceaddress" option when "bindaddr"
+ is set to 0.0.0.0 instead of having each interface explicitly
+ listed. (issue #7874, patch by stevens) ........
+
+ * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
+ function to just be additional parameter of the CHANNEL function.
+ This way, it will be possible for other RTP based channel drivers
+ to expose this information in the future.
+
+2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
+ Mär 2007) | 1 line fixed #9355 ........
+
+2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Oops, this should be case insensitive
+
+2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf at digium.com>
+
+ * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
+ asterisk). I turned a duplicate context from a WARNING to an
+ ERROR. Now you get a module load failure, and asterisk just
+ exits. That's better than a crash, right\?
+
+2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * channels/chan_sip.c: Change this to a single dp function to make
+ oej happy.
+
+2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf at digium.com>
+
+ * main/config.c: Fix for 9257; by eliminating the globals in
+ main/config.c, we make it thread-safe, which is a minimum
+ requirement.
+
+2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_speech_utils.c: Add ability to specify no timeout. This
+ means as soon as the prompt is done playing it moves on to the
+ next priority.
+
+2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell at digium.com>
+
[... 5806 lines stripped ...]
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