[asterisk-commits] russell: tag 1.6.0-beta7 r111668 - /tags/1.6.0-beta7/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Mar 28 11:47:43 CDT 2008
Author: russell
Date: Fri Mar 28 11:47:42 2008
New Revision: 111668
URL: http://svn.digium.com/view/asterisk?view=rev&rev=111668
Log:
Importing files for 1.6.0-beta7 release
Added:
tags/1.6.0-beta7/.lastclean (with props)
tags/1.6.0-beta7/.version (with props)
tags/1.6.0-beta7/ChangeLog (with props)
Added: tags/1.6.0-beta7/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.6.0-beta7/.lastclean?view=auto&rev=111668
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==============================================================================
--- tags/1.6.0-beta7/ChangeLog (added)
+++ tags/1.6.0-beta7/ChangeLog Fri Mar 28 11:47:42 2008
@@ -1,0 +1,39864 @@
+2008-03-28 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.6.0-beta7 released.
+
+2008-03-28 16:36 +0000 [r111662] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c, include/asterisk/strings.h: The copy_request
+ function did not take into account the necessary null terminator
+ for the string to be copied into. This resulted in parse_request
+ reading invalid memory beyond the end of the string, and in some
+ cases led to crashes. Thanks to falves11 for providing the
+ valgrind output which led to the closure of this issue. (closes
+ issue #12284) Reported by: falves11
+
+2008-03-28 16:20 +0000 [r111659] Jason Parker <jparker at digium.com>
+
+ * /, formats/format_wav_gsm.c: Merged revisions 111658 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar
+ 2008) | 8 lines The file size of WAV49 does not need to be an
+ even number. (closes issue #12128) Reported by: mdu113 Patches:
+ 12128-noevenlength.diff uploaded by qwell (license 4) Tested by:
+ qwell, mdu113 ........
+
+2008-03-28 14:37 +0000 [r111606] Tilghman Lesher <tlesher at digium.com>
+
+ * /, doc/valgrind.txt: Merged revisions 111605 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008)
+ | 3 lines Update debugging text, since Valgrind eliminated the
+ --log-file-exactly option. (Closes issue #12320) ........
+
+2008-03-28 00:55 +0000 [r111565] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_queue.c: Forgetting to unregister a manager action is
+ bad, mmmk?
+
+2008-03-28 00:12 +0000 [r111533] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fix a crash that would happen when attempting
+ to unload the app_queue module. The problem was that when the
+ refcount on the queue hit 0, the destructor was called, and
+ inside the destructor, another function was called which would
+ increase the refcount back to 1 again and then decrease it again
+ back to 0 for every member in the queue. This meant that the
+ destructor was being recursively called, leading to a double free
+ of the queue. This is now fixed by making sure to unlink the
+ queue from the queues container prior to the final unref of the
+ queue.
+
+2008-03-27 22:10 +0000 [r111500] Terry Wilson <twilson at digium.com>
+
+ * main/http.c: Fix another little http problem. In making it match
+ coding guidelines, a comparison was dropped
+
+2008-03-27 21:25 +0000 [r111497] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c: comment cleanup and iron out a really dumb mistake in
+ handling the '.'-wildcard in the new exten pattern matcher.
+
+2008-03-27 19:26 +0000 [r111443] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/acl.c: Merged revisions 111442 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008)
+ | 6 lines For FreeBSD, at least, the ifa_addr element could be
+ NULL. (closes issue #12300) Reported by: festr Patches:
+ acl.c.patch uploaded by festr (license 443) ........
+
+2008-03-27 13:29 +0000 [r111360-111410] Steve Murphy <murf at digium.com>
+
+ * main/pbx.c, /, apps/app_playback.c: Merged revisions 111391 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9
+ lines These small documentation updates made in response to a
+ query in asterisk-users, where a user was using Playback, but
+ needed the features of Background, and had no idea that
+ Background existed, or that it might provide the features he
+ needed. I thought the best way to avert these kinds of queries
+ was to provide "See Also" references in all three of
+ "Background", "Playback", "WaitExten". Perhaps a project to do
+ this with all related apps is in order. ........
+
+ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c,
+ include/asterisk/ael_structs.h: Merged revisions 111341 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) |
+ 15 lines (closes issue #12302) Reported by: pj Tested by: murf
+ These changes will set a channel variable ~~EXTEN~~ just before
+ generating code for a switch, with the value of ${EXTEN}. The
+ exten is marked as having a switch, and ever after that, till the
+ end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~}
+ instead in application arguments; (and the ${EXTEN: also). The
+ reason for this, is that because switches are coded using
+ separate extensions to provide pattern matching, and jumping
+ to/from these switch extensions messes up the ${EXTEN} value,
+ which blows the minds of users. ........
+
+2008-03-27 00:27 +0000 [r111246-111295] Jason Parker <jparker at digium.com>
+
+ * main/frame.c: But we can change the API here.
+
+ * main/frame.c, /: Merged revisions 111280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) |
+ 1 line Put this flag back so we don't change the API. ........
+
+ * main/frame.c, /: Merged revisions 111245 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) |
+ 9 lines Remove excessive smoother optimization that was causing
+ audio glitches (small "pops") after (about 200ms later) an
+ "incorrectly" sized frame was received. While it would be very
+ nice to keep this as optimized as possible, it makes no sense for
+ the smoother to be dropping random bits of audio like this. Isn't
+ that the whole point of a smoother? Closes issue #12093. ........
+
+2008-03-26 21:23 +0000 [r111213] Terry Wilson <twilson at digium.com>
+
+ * main/http.c: Stupid strcasecmp function :-)
+
+2008-03-26 20:34 +0000 [r111132-111185] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/misdn_config.c: Oops, missed one
+
+ * include/asterisk/linkedlists.h, main/config.c: Simplify new
+ macro, simplify configfile logic, now that list is sorted
+
+2008-03-26 19:56 +0000 [r111130] Joshua Colp <jcolp at digium.com>
+
+ * /, contrib/scripts/autosupport: Merged revisions 111129 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6
+ lines Update autosupport script. (closes issue #12310) Reported
+ by: angler Patches: autosupport.diff uploaded by angler (license
+ 106) ........
+
+2008-03-26 19:52 +0000 [r111127] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 111126 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500
+ (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
+ 2008) | 2 lines update UPGRADE notes to document usage of the
+ script ........ ................
+
+2008-03-26 19:39 +0000 [r111123] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 111121 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed,
+ 26 Mar 2008) | 4 lines This code change is made just for
+ clarification. It does exactly the same thing as before. It just
+ doesn't look as wrong. ........
+
+2008-03-26 19:29 +0000 [r111083] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Add expiry value to the sip show
+ subscriptions CLI command. (closes issue #12025) Reported by: agx
+
+2008-03-26 19:26 +0000 [r111067] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 111049 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed,
+ 26 Mar 2008) | 9 lines Add a lock to the vm_state structure and
+ use the lock around mail_open calls to prevent concurrent access
+ of the same mailstream. This, along with trunk's ability to
+ configure TCP timeouts for IMAP storage will help to prevent
+ crashes and hangs when using voicemail with IMAP storage. (closes
+ issue #10487) Reported by: ewilhelmsen ........
+
+2008-03-26 19:19 +0000 [r111036] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/linkedlists.h, CHANGES, main/config.c: Add a
+ linkedlist macro that maintains a sorted list
+
+2008-03-26 19:16 +0000 [r111028] Jason Parker <jparker at digium.com>
+
+ * main/dsp.c: Only try to detect silence when we actually need to,
+ instead of...always. If this is wrong, I'd love to hear why.
+
+2008-03-26 19:08 +0000 [r111025] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh (added), codecs/ilbc:
+ Merged revisions 111024 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500
+ (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
+ 2008) | 2 lines add a script to make getting the iLBC source code
+ simple for end users ........ ................
+
+2008-03-26 19:05 +0000 [r111022] Jason Parker <jparker at digium.com>
+
+ * channels/chan_usbradio.c, channels/chan_vpb.cc,
+ channels/chan_zap.c, include/asterisk/dsp.h, main/dsp.c: Large
+ cleanup of DSP code Per comments from dimas: 1. The code now
+ generates DTMF_BEGIN frames in addition to DTMF_END ones. 2.
+ "quelching" rewritten - now each detector (MF/DTMF/generic tone)
+ may mark fragment of a frame for suppression (squelching, muting)
+ with a call to mute_fragment. Actual muting happens only once at
+ the very end of ast_dsp_process where all marked fragments are
+ zeroed. This way every detector sees original data in the frame
+ without any piece of a frame being zeroed by a detector which was
+ run before. 3. DTMF detector tries to "mute" one block before and
+ one block after the block where actual tone was detected. Muting
+ of previois block is something new for this patch. Obviously this
+ operation is not always possible - if current frame does not
+ contain data for previous block - it is too late. But at least we
+ make our best. Muting of next block was already done by the old
+ code but it only affects part of the next block which is in the
+ frame being processed. New code keeps this information in state
+ structures so it will mute proper number of samples in the next
+ frame(s) too. 4. Removed ast_dsp_digitdetect and
+ ast_dsp_getdigits APIs because these are not used. 5. DSP API
+ extended a bit - ast_dsp_was_muted() function added which returns
+ true if DSP code was muting any fragment in the last frame.
+ chan_zap uses this function to decide it needs to turn on
+ confmute on the channel. This is to replace AST_FRAME_DTMF
+ 'm'/'u' (mute/unmute) functionality. (closes issue #11968)
+ Reported by: dimas Patches: v2-11968-dsp.patch uploaded by dimas
+ (license 88) v4-11968-zap.patch uploaded by dimas (license 88)
+ Tested by: dimas, qwell
+
+2008-03-26 19:05 +0000 [r111017-111021] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 111020 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4
+ lines If we are requested to authenticate a reinvite make sure
+ that it contains T38 SDP if need be. (closes issue #11995)
+ Reported by: fall ........
+
+ * /, channels/chan_iax2.c: Merged revisions 110628 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar
+ 2008) | 4 lines Add an option (transmit_silence) which transmits
+ silence during both Record() and DTMF generation. The reason this
+ is an option is that in order to transmit silence we have to
+ setup a translation path. This may not be needed/wanted in all
+ cases. (closes issue #10058) Reported by: tracinet ........
+
+2008-03-26 18:41 +0000 [r111012-111013] Tilghman Lesher <tlesher at digium.com>
+
+ * CHANGES: Oops, fix this, too
+
+ * main/udptl.c, main/dnsmgr.c, include/asterisk/config.h,
+ channels/iax2-provision.c, main/enum.c, main/rtp.c,
+ main/config.c, main/loader.c, main/cdr.c, main/manager.c,
+ main/features.c, main/logger.c, main/http.c,
+ include/asterisk/udptl.h, main/asterisk.c, main/dsp.c: Add the
+ "config reload <conffile>" command, which allows you to tell
+ Asterisk to reload any file that references a given configuration
+ file.
+
+2008-03-26 17:44 +0000 [r110963] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 110962 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar
+ 2008) | 2 lines add note that the user will need to enable
+ codec_ilbc to get it to build ........
+
+2008-03-26 17:28 +0000 [r110911-110930] Donny Kavanagh <donnyk at gmail.com>
+
+ * Makefile: revert something dumb, because i was running svn diff
+ in a subfolder not the root of trunk, before doing my commit and
+ did not see it
+
+ * Makefile, doc/snmp.txt: update documentation to reflect the
+ changes in the way configure detects net-snmp. (closes issue
+ #12067) Reported by: juggie Patches: 12067_snmp_doc.patch
+ uploaded by juggie (license 24) Tested by: juggie
+
+2008-03-26 17:10 +0000 [r110881] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
+ (removed), codecs/ilbc/syntFilter.h (removed),
+ codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
+ (removed), codecs/ilbc/StateConstructW.h (removed),
+ codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
+ (removed), codecs/ilbc/getCBvec.c (removed),
+ codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
+ (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
+ (removed), codecs/ilbc/getCBvec.h (removed),
+ codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/iLBC_define.h
+ (removed), codecs/ilbc/FrameClassify.c (removed),
+ codecs/ilbc/enhancer.h (removed), codecs/ilbc/lsf.h (removed),
+ codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
+ (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
+ (removed), codecs/ilbc/anaFilter.c (removed),
+ codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
+ (removed), codecs/ilbc/doCPLC.h (removed),
+ codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
+ codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
+ (removed), codecs/ilbc/createCB.h (removed), CHANGES,
+ codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h
+ (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile,
+ codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
+ codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
+ (removed), codecs/ilbc/iCBSearch.h (removed),
+ codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
+ codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
+ (removed), codecs/ilbc/hpOutput.h (removed),
+ codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
+ codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
+ (removed), codecs/ilbc/iCBConstruct.c (removed): Merged revisions
+ 110880 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700
+ (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
+ 2008) | 2 lines due to licensing restrictions, we cannot
+ distribute the source code for iLBC encoding and decoding... so
+ remove it, and add instructions on how the user can obtain it
+ themselves ........ ................
+
+2008-03-26 00:02 +0000 [r110831] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: This ensures that the manager interface is not
+ enabled by default. Prior to this change, it was possible to
+ start Asterisk with the manager interface enabled, then either
+ comment out the enabled option or make manager.conf unopenable
+ and the manager interface would still be enabled.
+
+2008-03-25 22:51 +0000 [r110780] Jason Parker <jparker at digium.com>
+
+ * /, cdr/cdr_custom.c: Merged revisions 110779 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) |
+ 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes
+ issue #12268. Patch borrowed from r82344 ........
+
+2008-03-25 20:02 +0000 [r110726] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: This one line change makes an if inside a
+ for loop (in realtime_peer) check all the ast_variables the loop
+ was intending to test rather than just the first one.
+
+2008-03-25 17:46 +0000 [r110689-110691] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/voicemail.conf.sample, configs/extensions.conf.sample:
+ Update sample configurations to make virtual hosting more
+ obvious. (closes issue #11969) Reported by: pprindeville Patches:
+ acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)
+
+ * configs/extensions.conf.sample: Update the sample configuration,
+ to use Macro less (since it's now deprecated). (closes issue
+ #12293) Reported by: pprindeville Patches:
+ bugid-0012293.1.6.patch uploaded by pprindeville (license 347)
+
+2008-03-25 15:44 +0000 [r110636-110639] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Oops here too. I need to stop coding for a
+ while...
+
+ * /, channels/chan_sip.c: Merged revisions 110635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar
+ 2008) | 7 lines When reverting a commit, I accidentally left in
+ this bit which was an experiment to see what would happen. It
+ passed the compile test, and I didn't notice I had left this
+ change in too. So this is a revert of a revert...sort of.
+ ........
+
+2008-03-25 15:18 +0000 [r110629-110631] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c, channels/chan_sip.c, configs/sip.conf.sample,
+ CHANGES: Add a special dialplan variable to chan_sip which will
+ cause an audio file to be played upon completion of an attended
+ transfer. (closes issue #9239) Reported by: sunder
+
+ * Makefile, /, main/app.c, include/asterisk/options.h,
+ main/asterisk.c: Merged revisions 110628 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4
+ lines Add an option (transmit_silence) which transmits silence
+ during both Record() and DTMF generation. The reason this is an
+ option is that in order to transmit silence we have to setup a
+ translation path. This may not be needed/wanted in all cases.
+ (closes issue #10058) Reported by: tracinet ........
+
+2008-03-25 10:54 +0000 [r110625] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Use the "Server" header when responding to
+ SIP requests. (closes issue #12278) Reported by: rjain Patches:
+ chan_sip.c.diff uploaded by rjain (license 226)
+
+2008-03-24 20:14 +0000 [r110619-110621] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Remove the "Event: registration" header from
+ Asterisk-generated SIP REGISTER requests. rjain points out that
+ RFC 3265 specifies that the Event: header is not a valid header
+ for REGISTER requests and that the "registration" value is not
+ defined at IANA. (closes issue #12279) Reported by: rjain
+ Patches: chan_sip.c.diff uploaded by rjain (license 226)
+
+ * channels/chan_sip.c: Merged revisions 110618 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar
+ 2008) | 15 lines This is a revert for revision 108288. The reason
+ is that that revision was not for an actual bug fix per se, and
+ so it really should not have been in 1.4 in the first place.
+ Plus, people who compile with DO_CRASH are more likely to
+ encounter a crash due to this change. While I think the usage of
+ DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
+ beyond the scope of 1.4 and should be done instead in a developer
+ branch based on trunk so that all scheduler functions are fixed
+ at once. I also am reverting the change to trunk and 1.6 since
+ they also suffer from the DO_CRASH potential. (closes issue
+ #12272) Reported by: qq12345 ........
+
+2008-03-24 17:36 +0000 [r110615] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 110614 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24
+ Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........
+
+2008-03-24 15:28 +0000 [r110610] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Only print out the set_address_from_contact
+ host verbose message if debugging is enabled on the dialog.
+ (closes issue #12280) Reported by: rjain Patches: chan_sip.c.diff
+ uploaded by rjain (license 226)
+
+2008-03-21 21:52 +0000 [r110578] Jason Parker <jparker at digium.com>
+
+ * sounds/Makefile: Update to 1.4.11 core sounds.
+
+2008-03-21 17:58 +0000 [r110542] Joshua Colp <jcolp at digium.com>
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Merge over
+ ast_audiohook_volume branch. This adds API calls for use by
+ developers to adjust the volume on a channel.
+
+2008-03-21 15:24 +0000 [r110499] Russell Bryant <russell at digium.com>
+
+ * configs/sip.conf.sample, CHANGES: Note that the TCP and TLS
+ support is currently considered experimental and is subject to
+ change while we work out the remaining issues.
+
+2008-03-21 14:36 +0000 [r110475] Jason Parker <jparker at digium.com>
+
+ * /, codecs/gsm/Makefile: Merged revisions 110474 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) |
+ 7 lines Don't attempt to do optimizations of gsm on mips
+ platforms either. (closes issue #12270) Reported by: zandbelt
+ Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33)
+ ........
+
+2008-03-21 01:44 +0000 [r110444] Tilghman Lesher <tlesher at digium.com>
+
+ * CHANGES: Add note of the added Directory options, from commit
+ 110237 (closes issue #7151)
+
+2008-03-20 23:14 +0000 [r110303-110396] Russell Bryant <russell at digium.com>
+
+ * main/autoservice.c, /: Merged revisions 110395 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008)
+ | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms
+ in the autoservice thread. This really should not make a
+ difference except in very rare cases. That case would be that all
+ of the channels in autoservice are not generating any frames. In
+ that case, this change reduces the potential amount of time that
+ a thread waits in ast_autoservice_stop() for the autoservice
+ thread to wrap back around to the beginning of its loop. (closes
+ issue #12266, reported by dimas) ........
+
+ * codecs/codec_g722.c: Use the correct buffer for
+ g722tolin16_sample. This shouldn't have caused any problems, but
+ Qwell noticed the typo here.
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 110336 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r110336 | russell | 2008-03-20 16:54:58 -0500
+ (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
+ | 6 lines Fix some very broken code that was introduced in 1.2.26
+ as a part of the security fix. The dnsmgr is not appropriate
+ here. The dnsmgr takes a pointer to an address structure that a
+ background thread continuously updates. However, in these cases,
+ a stack variable was passed. That means that the dnsmgr thread
+ would be continuously writing to bogus memory. ........
+ ................
+
+ * main/file.c: Fix a bug when using zaptel timing for playing back
+ files that have a sample rate other than 8 kHz. The issue here is
+ that format modules give a "whennext" sample value, which is used
+ to calculate when to set a timer for to retrieve the next frame.
+ However, the zaptel timer operates on 8 kHz samples, so this must
+ be taken into account. (another part of issue #12164, reported by
+ milazzo and jsmith, patch by me)
+
+2008-03-20 18:01 +0000 [r110272] Mark Michelson <mmichelson at digium.com>
+
+ * main/dial.c: Add missing unlock
+
+2008-03-20 17:45 +0000 [r110268-110270] Russell Bryant <russell at digium.com>
+
+ * apps/app_meetme.c, apps/app_minivm.c, include/asterisk/netsock.h,
+ main/netsock.c: Remove astobj.h from some places where it wasn't
+ needed
+
+ * main/channel.c, res/res_musiconhold.c: Add some fixes that I made
+ in regards to wideband codec handling to get G.722 music on hold
+ working for me. (issue #12164, reported by milazzo and jsmith,
+ patches by me) res/res_musiconhold.c: - I moved a single line so
+ that the sample queue update happened before ast_write(). The
+ reason that this was a bug is that the G.722 frame originally
+ says it has 320 samples in it (which is correct). However, when
+ the frame is written to a channel that uses RTP, main/rtp.c
+ modifies the frame to cut the number of samples in half before it
+ sends it on the wire. This is to account for the stupid incorrect
+ G.722 spec that makes it so we have to lie about the number of
+ samples with RTP. I should probably go and re-work the RTP code
+ so it doesn't modify the frame so that a bug like this won't
+ happen in the future. However, this change to MOH is harmless.
+ main/channel.c: - I made two fixes in regards to generator
+ timing. Generators use samples for timing. However, this code
+ assumed 8 kHz samples. In one case, it was a hard coded 160
+ samples, that is now written as the sample rate / 50. The other
+ place was dealing with timing a generator based on frames coming
+ from the other direction. However, that would have only worked if
+ the sample rates for the formats in both directions were the
+ same. The code now takes into account that the sample rates may
+ differ, and scales the generator samples accordingly.
+
+2008-03-20 05:06 +0000 [r110211-110237] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_directory.c, sounds/Makefile: Upgrade the sounds
+ version; add several directory enhancements: 1) Number of digits
+ to enter can now be configured 2) The digits can now match on
+ both first AND last name, instead of only one or the other
+ (Closes issue #7151)
+
+ * channels/chan_vpb.cc: Fix recent trunk breakage
+
+2008-03-19 22:58 +0000 [r110164] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 110163 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008)
+ | 5 lines Fix a bug where when calls on the trunk side hang up
+ while on hold, the state is not properly reflected. (closes issue
+ #11990, reported by anakaoka, patched by me) ........
+
+2008-03-19 22:25 +0000 [r110132-110161] Jason Parker <jparker at digium.com>
+
+ * channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_h323.c, include/asterisk/dsp.h,
+ channels/chan_mgcp.c, main/dsp.c: Rename DSP_FEATURE_DTMF_DETECT,
+ because we are *NOT* only detecting DTMF digits. This was very
+ misleading. Early cleanup for issue #11968
+
+ * channels/chan_usbradio.c, channels/chan_vpb.cc,
+ channels/chan_zap.c, channels/chan_sip.c, include/asterisk/dsp.h,
+ channels/chan_mgcp.c, main/dsp.c: Rename very poorly named
+ function to reflect what it actually does. This was causing quite
+ a bit of confusion for me...
+
+2008-03-19 21:05 +0000 [r110087] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c, CHANGES: This change adds DNS manager
+ support for registrations not referencing a peer entry. It looks
+ like there is support for DNS manager for realtime peers as well,
+ however it is not implemented correctly. The improper usage
+ occurs when ast_dnsmgr_lookup is called with one of the arguments
+ being an address from the stack to be continually updated. The
+ variable from the stack will go out of scope and dnsmgr will
+ continue to try and update the memory there, causing possible
+ stack corruption. This problem will be worked on next as well as
+ adding DNS manager support for peer entries.
+
+2008-03-19 20:34 +0000 [r110084] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 110083 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar
+ 2008) | 4 lines Add a missing unlock in the case that memory
+ allocation fails in app_chanspy. Thanks to Russell for confirming
+ that this was an issue. ........
+
+2008-03-19 19:13 +0000 [r110036] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 110035 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar
+ 2008) | 4 lines Add sanity checking for position resuming. We
+ *have* to make sure that the position does not exceed the total
+ number of files present, and we have to make sure that the
+ position's filename is the same as previous. These values can
+ change if a music class is reloaded and give unpredictable
+ behavior. (closes issue #11663) Reported by: junky ........
+
+2008-03-19 18:57 +0000 [r110023] Russell Bryant <russell at digium.com>
+
+ * /: remove svnmerge-blocked property that is not supposed to be
+ here
+
+2008-03-19 18:25 +0000 [r110020] Joshua Colp <jcolp at digium.com>
+
+ * /, main/rtp.c: Merged revisions 110019 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6
+ lines Make sure that the mark bit does not incorrectly cause
+ video frame timestamps to be calculated as if they are audio
+ frames. (closes issue #11429) Reported by: sperreault Patches:
+ 11429-frametype.diff uploaded by qwell (license 4) ........
+
+2008-03-19 17:15 +0000 [r109974] Jason Parker <jparker at digium.com>
+
+ * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
+ (added), /: Merged revisions 109973 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) |
+ 5 lines People report bugs about Asterisk crashing with DO_CRASH
+ enabled was getting a little silly... Now we only show certain
+ cflags when you run configure with --enable-dev-mode
+ (corresponding menuselect change to follow) ........
+
+2008-03-19 16:54 +0000 [r109970] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c, CHANGES: Add the ability to use a pattern match for a
+ hint. (closes issue #7767) Reported by: Corydon76 Patches:
+ 20070314__simple_hint_lookup.diff.txt uploaded by Corydon76
+ pbx-trunk-98436.diff uploaded by plack (license 365)
+
+2008-03-19 16:24 +0000 [r109942] Steve Murphy <murf at digium.com>
+
+ * /, main/config.c: Merged revisions 109908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) |
+ 72 lines (closes issue #11442) Reported by: tzafrir Patches:
+ 11442.patch uploaded by murf (license 17) Tested by: murf I
+ didn't give tzafrir very much time to test this, but if he does
+ still have remaining issues, he is welcome to re-open this bug,
+ and we'll do what is called for. I reproduced the problem, and
+ tested the fix, so I hope I am not jumping by just going ahead
+ and committing the fix. The problem was with what file_save does
+ with templates; firstly, it tended to print out multiple options:
+ [my_category](!)(templateref) instead of
+ [my_category](!,templateref) which is fixed by this patch.
+ Nextly, the code to suppress output of duplicate declarations
+ that would occur because the reader copies inherited declarations
+ down the hierarchy, was not working. Thus: [master-template](!)
+ mastervar = bar [template](!,master-template) tvar = value
+ [cat](template) catvar = val would be rewritten as: ;! ;!
+ Automatically generated configuration file ;! Filename:
+ experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
+ Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
+ [master-template](!) mastervar = bar
+ [template](!,master-template) mastervar = bar tvar = value
+ [cat](template) mastervar = bar tvar = value catvar = val This
+ has been fixed. Since the config reader 'explodes' inherited vars
+ into the category, users may, in certain circumstances, see
+ output different from what they originally entered, but it should
+ be both correct and equivalent. ........
+
+2008-03-19 16:21 +0000 [r109912-109926] Kevin P. Fleming <kpfleming at digium.com>
+
+ * res/res_phoneprov.c: ensure that res_phoneprov's HTTP handler
+ tells the dispatcher what method it can handle
+
+ * main/manager.c, main/http.c: actually implement HTTP request
+ dispatching based on both URI and method; reduce duplication of
+ data when generating responses using ast_http_error()
+
+2008-03-19 15:45 +0000 [r109910] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Fix some more breakage that I introduced when
+ changing extension state callbacks to the list macros.
+
+2008-03-19 15:41 +0000 [r109909] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/http.c: clean up code to conform to coding guidelines
+
+2008-03-19 15:22 +0000 [r109833-109907] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Remove an unneeded variable. This compiled, but I
+ missed the uninitialized warning because I always compile without
+ optimizations turned on. Sorry!
+
+ * main/pbx.c: Convert handling of extension state callbacks to the
+ list macros.
+
+ * main/pbx.c: Minor coding style changes, including adding handling
+ for memory allocation failure
+
+ * main/http.c: Minor change to use Asterisk macros
+
+ * /, main/utils.c: Merged revisions 109838 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008)
+ | 2 lines Tweak spacing in a recent change because I'm very
+ picky. ........
+
+ * channels/chan_sip.c: Set req->data to NULL after free'ing to
+ ensure that it never gets accidentally double free'd. (reported
+ by dhubbard directly to me)
+
+2008-03-18 23:32 +0000 [r109802] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_zap.c: Fix a typo which caused a double free in
+ chan_zap. This was discovered by Juggie while attempting to load
+ chan_zap. Apparently this would happen if an error were
+ encountered while trying to process zapata.conf.
+
+2008-03-18 23:22 +0000 [r109775] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/res_ldap.conf.sample, res/res_config_ldap.c: Change back
+ to using ldap_initialize() and let the user specify a URL
+ directly, instead of trying to piece it together, badly.
+
+2008-03-18 22:36 +0000 [r109764] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 109763 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008)
+ | 3 lines Fix one place where the chanspy datastore isn't removed
+ from a channel. (issue #12243, reported by atis, patch by me)
+ ........
+
+2008-03-18 22:32 +0000 [r109762] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/http.h, main/manager.c, res/res_phoneprov.c,
+ main/http.c, include/asterisk/_private.h: start the process of
+ changing HTTP request dispatching to do it based on *both* URI
+ and method, so that POST support can move into a module; move
+ http.c's private function prototypes into _private.h
+
+2008-03-18 20:59 +0000 [r109714] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 109713 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar
[... 39125 lines stripped ...]
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