[asterisk-commits] russell: tag 1.6.0-beta2 r100726 - /tags/1.6.0-beta2/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jan 28 17:47:41 CST 2008
Author: russell
Date: Mon Jan 28 17:47:40 2008
New Revision: 100726
URL: http://svn.digium.com/view/asterisk?view=rev&rev=100726
Log:
Importing files for 1.6.0-beta2 release
Added:
tags/1.6.0-beta2/.lastclean (with props)
tags/1.6.0-beta2/.version (with props)
tags/1.6.0-beta2/ChangeLog (with props)
Added: tags/1.6.0-beta2/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.6.0-beta2/.lastclean?view=auto&rev=100726
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URL: http://svn.digium.com/view/asterisk/tags/1.6.0-beta2/ChangeLog?view=auto&rev=100726
==============================================================================
--- tags/1.6.0-beta2/ChangeLog (added)
+++ tags/1.6.0-beta2/ChangeLog Mon Jan 28 17:47:40 2008
@@ -1,0 +1,35332 @@
+2008-01-28 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.6.0-beta2 released.
+
+2008-01-28 21:11 +0000 [r100679] Jason Parker <jparker at digium.com>
+
+ * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added),
+ doc/tex/channelvariables.tex, makeopts.in: Reintroduce more
+ chan_vpb stuff that was removed in r100421 and r100422
+
+2008-01-28 21:07 +0000 [r100678] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_vpb.cc (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/Makefile: Re-inserting chan_vpb into trunk.
+
+2008-01-28 21:05 +0000 [r100677] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, /: Merged revisions 100675 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008)
+ | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went
+ to 't' instead of 'T') ........
+
+2008-01-28 21:02 +0000 [r100676] Jason Parker <jparker at digium.com>
+
+ * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes
+ issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43
+ -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make
+ sure we put greeting files into the database like we do with the
+ others. Issue #11795 Reported by: dimas Patches: vmgreet.patch
+ uploaded by dimas (license 88) ........
+
+2008-01-28 20:58 +0000 [r100674] Mark Michelson <mmichelson at digium.com>
+
+ * /: Blocked revisions 100673 via svnmerge ........ r100673 |
+ mmichelson | 2008-01-28 14:55:56 -0600 (Mon, 28 Jan 2008) | 3
+ lines Undoing the deprecation of chan_vpb. It is alive and well.
+ ........
+
+2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix up some T38 state change issues. (closes
+ issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch
+ uploaded by dimas (license 88)
+
+ * channels/chan_sip.c: Fix up two scheduling issues. In one
+ instance a scheduled item was not deleted when it should have
+ been and in the other it was scheduled again when it shouldn't
+ have been.
+
+2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant <russell at digium.com>
+
+ * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge
+ of this commit was a NoOp, since res_features doesn't exist in
+ trunk. Thanks to qwell for pointing it out!
+
+ * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008)
+ | 5 lines For some reason, the use of this strdupa() is leading
+ to memory corruption on freebsd sparc64. This trivial workaround
+ fixes it. (closes issue #10300, closes issue #11857, reported by
+ mattias04 and Home-of-the-Brave) ........
+
+2008-01-28 18:27 +0000 [r100628] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/logger.c: Normalize the detection for execinfo, so that
+ Linux (glibc) and other platforms with libexecinfo will generate
+ inline stack backtraces correctly.
+
+2008-01-28 18:27 +0000 [r100627] Russell Bryant <russell at digium.com>
+
+ * /: Merged revisions 100626 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008)
+ | 7 lines Fix a crash in ast_masq_park_call() (issue #11342)
+ Reported by: DEA Patches: res_features-park.txt uploaded by DEA
+ (license 3) ........
+
+2008-01-28 18:24 +0000 [r100625] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) |
+ 1 line Correct a comment which made little/no sense. ........
+
+2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, channels/chan_local.c, /,
+ include/asterisk/channel.h: Merged revisions 100581 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28
+ Jan 2008) | 9 lines Make some deadlock related fixes. These bugs
+ were discovered and reported internally at Digium by Steve Pitts.
+ - Fix up chan_local to ensure that the channel lock is held
+ before the local pvt lock. - Don't hold the channel lock when
+ executing the timing function, as it can cause a deadlock when
+ using chan_local. This actually changes the code back to be how
+ it was before the change for issue #10765. But, I added some
+ other locking that I think will prevent the problem reported
+ there, as well. ........
+
+ * main/pbx.c: Clean up some formatting, and simplify a bit of code
+ using ast_str
+
+2008-01-28 13:57 +0000 [r100549] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Don't do a network byte order conversion
+ when setting the socket's port variable to that of bindaddr's. It
+ is already in the correct network byte order. (closes issue
+ #11800) Reported by: hmodes
+
+2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert
+ some spaces to tabs
+
+ * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few
+ little formatting changes
+
+ * main/channel.c: These readlocks always fail for me on my mac, and
+ I saw it happen again today on another mac. We ignore the return
+ value of locking operations almost everywhere in Asterisk. So,
+ ignore these, as well, so Asterisk will actually work on systems
+ where this is occurring while I look into what the issue is.
+
+2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c, include/asterisk/sched.h,
+ channels/chan_iax2.c: With the switch to the ast_sched_replace*
+ API in trunk, we lose the correction that was just merged from
+ 1.4, so this is a changeover to those APIs to use the macro
+ versions, so that we properly detect errors from ast_sched_del,
+ instead of simply ignoring the return values.
+
+ * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /,
+ channels/chan_sip.c, channels/chan_h323.c,
+ include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c,
+ channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged
+ revisions 100465 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008)
+ | 11 lines When deleting a task from the scheduler, ignoring the
+ return value could possibly cause memory to be accessed after it
+ is freed, which causes all sorts of random memory corruption.
+ Instead, if a deletion fails, wait a bit and try again (noting
+ that another thread could change our taskid value). (closes issue
+ #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan,
+ stuarth` ........
+
+2008-01-25 22:54 +0000 [r100421-100422] Jason Parker <jparker at digium.com>
+
+ * doc/tex/channelvariables.tex: Get rid of that last little bit.
+
+ * build_tools/menuselect-deps.in, configs/vpb.conf.sample
+ (removed), makeopts.in: Remove more remnants of chan_vpb
+
+2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_vpb.cc (removed), configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ channels/Makefile, .cleancount: Removing chan_vpb from the tree
+
+ * /: Blocked revisions 100418 via svnmerge ........ r100418 |
+ mmichelson | 2008-01-25 16:32:41 -0600 (Fri, 25 Jan 2008) | 4
+ lines Deprecating chan_vpb. It is now preferred that users of
+ Voicetronix products use chan_zap in combination with their
+ zaptel drivers. ........
+
+2008-01-25 21:26 +0000 [r100379] Jason Parker <jparker at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) |
+ 2 lines This would have never been true, since we're passing
+ (sizeof(req.data) - 1) as the len to recvfrom(). ........
+
+2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming <kpfleming at digium.com>
+
+ * apps/app_rpt.c: correct a real problem and silence an annoying
+ compiler warning
+
+2008-01-25 14:53 +0000 [r100344] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Insure that we are not going to pass a NULL
+ pointer to add_to_interfaces. (closes issue #11840) Reported by:
+ junky
+
+2008-01-25 02:52 +0000 [r100325] Joshua Colp <jcolp at digium.com>
+
+ * main/dial.c, include/asterisk/dial.h: Add an API call that steals
+ the answered channel so that a destruction of the dialing
+ structure does not hang it up.
+
+2008-01-24 22:58 +0000 [r100307] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as
+ the default, too
+
+2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/app.h: ummm... might be good if this macro
+ argument was actually used :-)
+
+ * include/asterisk/app.h: add the ability to define a structure
+ type for argument parsing when it would be useful to be able to
+ pass it between functions
+
+2008-01-24 22:02 +0000 [r100266] James Golovich <james at gnuinter.net>
+
+ * channels/chan_sip.c: Fix simple whitespace issue
+
+2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24
+ Jan 2008) | 2 lines make these macros not assume that the only
+ other field in the structure is 'argc'... this is true when
+ someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
+ to define your own structure as long as it has the right fields
+ ........
+
+2008-01-24 20:32 +0000 [r100245] Joshua Colp <jcolp at digium.com>
+
+ * main/features.c: Minor cosmetic change...
+
+2008-01-24 18:35 +0000 [r100224] James Golovich <james at gnuinter.net>
+
+ * main/astmm.c: Increase the size of filenames stored when astmm is
+ used. If the path length was long they would be truncated and
+ grouped together with whatever matches
+
+2008-01-24 17:47 +0000 [r100206] Joshua Colp <jcolp at digium.com>
+
+ * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp
+ branch. This adds a strictrtp option to rtp.conf which drops
+ packets that do not come from the remote party. (closes issue
+ #8952) Reported by: amorsen
+
+2008-01-24 17:24 +0000 [r100169] Russell Bryant <russell at digium.com>
+
+ * /, main/asterisk.c: Merged revisions 100164 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008)
+ | 2 lines Update main Asterisk copyright info to 2008 ........
+
+2008-01-24 16:47 +0000 [r100121-100139] Jason Parker <jparker at digium.com>
+
+ * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) |
+ 6 lines Fix compilation on Solaris. (closes issue #11832)
+ Patches: bug-11832.diff uploaded by snuffy (license 35) ........
+
+ * channels/chan_sip.c, main/features.c: Move chan_local dependency
+ into places (only one) that previously depended on res_features,
+ and used local channels
+
+2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c,
+ channels/chan_mgcp.c: Remove dependency on res_features from some
+ channel drivers. It is now part of the core and no longer exists
+ as a module.
+
+ * main/channel.c: Some more cosmetic changes.
+
+ * main/channel.c: Add some spacing.
+
+ * main/dial.c: Test hopefully over.
+
+ * main/dial.c: Testing something...
+
+2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: fix flag bit definitions to make code from
+ issue #11049 actually work; along the way, clarify comments and
+ add some dummy flag definitions for other multi-bit flags to
+ hopefully stop this from happening in the future (closes issue
+ #11049)
+
+2008-01-23 23:09 +0000 [r100039] Jason Parker <jparker at digium.com>
+
+ * res/res_features.c (removed), main/Makefile, main/features.c
+ (added), include/asterisk/_private.h, CHANGES, .cleancount,
+ main/asterisk.c, main/loader.c, include/asterisk/features.h: Move
+ code from res_features into (new file) main/features.c
+
+2008-01-23 22:00 +0000 [r100021] Russell Bryant <russell at digium.com>
+
+ * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your
+ contributions!
+
+2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7
+ lines Second attempt. Don't change invitestate when receiving 18x
+ messages in CANCEL state. (issue #11736) Reported by: MVF Patch
+ by oej. ........
+
+ * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9
+ lines Make sure we don't cancel destruction on calls in CANCEL
+ state, even if we get 183 while waiting for answer on our CANCEL.
+ (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by
+ oej (license 306) Tested by: MVF ........
+
+2008-01-23 20:26 +0000 [r99976] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23
+ Jan 2008) | 3 lines Fixing a typo. ........
+
+2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant <russell at digium.com>
+
+ * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) |
+ 8 lines ChanSpy issues a beep when it starts at the beginning of
+ a list of channels to potentially spy on. However, if there were
+ no matching channels, it would beep at you over and over, which
+ is pretty annoying. Now, it will only beep once in the case that
+ there are no channels to spy on, but it will still beep again
+ once it reaches the beginning of the channel list again. (closes
+ issue #11738, patched by me) ........
+
+ * main/tcptls.c: Fix tcptls build when openssl isn't installed
+ (closes issue #11813) Reported by: tzafrir Patches:
+ asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176)
+
+2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c: since echo canceler parameters in Zaptel are
+ now signed integers, allow them during parsing
+
+2008-01-23 16:21 +0000 [r99879] Mark Michelson <mmichelson at digium.com>
+
+ * /: Blocked revisions 99878 via svnmerge ........ r99878 |
+ mmichelson | 2008-01-23 10:18:04 -0600 (Wed, 23 Jan 2008) | 4
+ lines These flag tests were illogical. They were testing sip_peer
+ flags on a sip_pvt. Thanks to Russell for helping to get this odd
+ problem figured out. ........
+
+2008-01-23 15:23 +0000 [r99860] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_h323.c: Progress messages don't work (closes issue
+ #10497) Reported by: pj Patches: h323-announces-r99483.diff
+ uploaded by sergee (license 138) Tested by: pj
+
+2008-01-23 10:18 +0000 [r99839] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure
+ that we are aware of a pending INVITE even if we're using TCP
+
+2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Coding guidelines fixups
+
+ * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008)
+ | 8 lines When we reset the password via an external command, we
+ should also reset the password stored in the in-memory list, too
+ (otherwise it doesn't really take effect). (closes issue #11809)
+ Reported by: davetroy Patches: fix_externpass.diff uploaded by
+ davetroy (license 384) ........
+
+ * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008)
+ | 2 lines Oops, should have checked for a NULL obj, here, too
+ ........
+
+ * res/res_config_ldap.c: Coding guidelines cleanup
+
+ * /, main/acl.c: Merged revisions 99718 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008)
+ | 2 lines Just confirmed that all current platforms need this
+ header file ........
+
+ * /: Oops
+
+ * /, build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, doc/ldap.txt (added),
+ configure.ac, configs/res_ldap.conf.sample (added),
+ res/res_config_ldap.c (added), CHANGES, makeopts.in,
+ contrib/scripts/asterisk.ldap-schema (added),
+ contrib/scripts/asterisk.ldif (added): Add res_config_ldap for
+ realtime LDAP engine. (closes issue #5768) Reported by: mguesdon
+ Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon
+ (license 121) res_ldap.conf.sample uploaded by suretec (license
+ 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70)
+ asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested
+ by: oej, mguesdon, suretec, cthorner
+
+2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4
+ lines Thanks to Russell's education I realize that BUFSIZ has
+ changed since I learned the C language over 20 years ago...
+ Resetting chan_sip to the size of BUFSIZ that I expected in my
+ old head to avoid too heavy memory allocations on some systems.
+ ........
+
+ * doc/tex/channelvariables.tex, CHANGES: Documentation updates for
+ BRIDGEPVTCALLID
+
+2008-01-22 20:42 +0000 [r99646] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/acl.c: Merged revisions 99643 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008)
+ | 2 lines Fix the defines for OS X (and Solaris, too) ........
+
+2008-01-22 20:41 +0000 [r99645] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Make sure the command is not just present but is
+ also configured to be executed
+
+2008-01-22 20:35 +0000 [r99644] Olle Johansson <oej at edvina.net>
+
+ * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h:
+ Add a generic function to set the bridged call PVT unique id
+ string as a channel variable BRIDGEPVTCALLID This is important
+ for call tracing in log files and CDRs, so that the SIP callID
+ can be traced along servers. The CHANNEL dialplan function won't
+ work here, since the outbound channel is gone when we need the
+ Call-ID. Other channel drivers may now implement the same
+ function :-), but this patch only supports chan_sip.so. Inspired
+ by (issue #11816) Reported by: ctooley Patch by oej
+
+2008-01-22 20:33 +0000 [r99642] Russell Bryant <russell at digium.com>
+
+ * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change
+ the Asterisk CLI startup commands feature to read commands to run
+ from cli.conf after a discussion on the -dev list.
+
+2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_local.c, /, res/res_features.c,
+ channels/chan_agent.c, apps/app_followme.c: Merged revisions
+ 99594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3
+ lines Add more dependencies on chan_local and add a note to the
+ description of chan_local so that people don't disable it in
+ menuselect just to clean up. ........
+
+ * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5
+ lines Add dependency on chan_local to app_dial. Dial still runs
+ without chan_local, but will be missing forwarding functionality.
+ ........
+
+2008-01-22 17:15 +0000 [r99559] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/acl.c: Merged revisions 99540 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008)
+ | 7 lines Ensure that we can get an address even when we don't
+ have a default route. (closes issue #9225) Reported by: junky
+ Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
+ (license 14) Tested by: oej, loloski, sergee ........
+
+2008-01-22 16:55 +0000 [r99542] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Point out a bug in some debug counter
+ handling
+
+2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Add authentication options to the SIP
+ dialstring. Documentation follows separately (issue #11587)
+ Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by
+ sobomax (license 359)
+
+ * configs/sip.conf.sample: Documentation updates
+
+ * doc/siptls.txt: Small fixes
+
+ * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c,
+ include/asterisk/tcptls.h: Doxygen updates
+
+2008-01-21 23:56 +0000 [r99427] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_local.c, /: Merged revisions 99426 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21
+ Jan 2008) | 12 lines Fixing an issue wherein monitoring local
+ channels was not possible. During a channel masquerade, the
+ monitors on the two channels involved are swapped. In 99% of the
+ cases this results in the desired effect. However, if monitoring
+ a local channel, this caused the monitor which was on the local
+ channel to get moved onto a channel which is immediately hung up
+ after the masquerade has completed. By swapping the monitors
+ prior to the masquerade, we avoid the problem by tricking the
+ masquerade into placing the monitor back onto the channel where
+ we want it. During the investigation of the issue, the channel's
+ monitor was the only thing that was swapped in such a manner
+ which did not make sense to have done. All other variable
+ swapping made sense. ........
+
+2008-01-21 23:25 +0000 [r99424] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c: Fix distinctive ring detection. Reported by:
+ milazzo Patches: drings.diff uploaded by milazzo (license 383)
+ Closes issue #11799
+
+2008-01-21 22:32 +0000 [r99406] Mark Michelson <mmichelson at digium.com>
+
+ * configs/queues.conf.sample, apps/app_queue.c: Adding the
+ QUEUENAME variable to the variables set using the setqueuevar
+ option in queues.conf. Suggestion comes from Shaun2222 on IRC.
+
+2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_console.c: Remove compiler warning for
+ uninitialized variable
+
+ * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was
+ committed without any doxygen obviously. Tss tss.
+
+ * channels/chan_sip.c: Updating doxygen
+
+2008-01-21 18:15 +0000 [r99350] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/res_odbc.h, /, res/res_odbc.c,
+ configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21
+ Jan 2008) | 8 lines Permit the user to specify number of seconds
+ that a connection may remain idle, which fixes a crash on
+ reconnect with the MyODBC driver. (closes issue #11798) Reported
+ by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak ........
+
+2008-01-21 16:02 +0000 [r99302] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4
+ lines Bump the buffer size for Via headers up to 512. There are
+ some exceptionally large Via headers out there. (closes issue
+ #11783) Reported by: ofirroval ........
+
+2008-01-21 07:02 +0000 [r99280] Olle Johansson <oej at edvina.net>
+
+ * CREDITS: Update
+
+2008-01-21 03:54 +0000 [r99265] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Change over to using ast_debug so these
+ debug messages don't always show up.
+
+2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant <russell at digium.com>
+
+ * channels/chan_console.c: Add a "console active" CLI command,
+ which lets you find out which console device is currently active
+ for the Asterisk CLI, or to set it. Also, knock multiple device
+ support off of the to-do list.
+
+ * configs/console.conf.sample: correct the name of a CLI command
+ for getting available device names
+
+ * configs/console.conf.sample, channels/chan_console.c: Merge
+ changes from team/russell/console_devices - Add support for
+ multiple devices. All devices are configured in console.conf. -
+ Add "console list devices" CLI command to show configured
+ devices. Also, changed the old "list devices" to be "list
+ available", which queries PortAudio for all audio devices that
+ are available for use.
+
+ * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) |
+ 4 lines Fix a couple of memory leaks with frame handling.
+ Specifically, ast_frame_free() needed to be called on the frame
+ that came from the translator to signed linear. ........
+
+ * README: Add Cygwin as an "other" platform that is supported
+
+ * README: Various README updates
+
+2008-01-18 22:58 +0000 [r99128] Joshua Colp <jcolp at digium.com>
+
+ * /: Blocked revisions 99127 via svnmerge ........ r99127 | file |
+ 2008-01-18 18:57:15 -0400 (Fri, 18 Jan 2008) | 2 lines Remove the
+ __ in front of the unused variable. This causes some compilers to
+ freak out. ........
+
+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
[... 34531 lines stripped ...]
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